SOUND ON SOUND [USA] February 2025 Issue

1985 — 2025

THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

TM

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

SSL 18 E X C L U S I V E

F I R S T

R E V I E W

The audio interface with hidden depths!

ADAM D3V

Mighty micro monitors

SOS Awards Winners announced!

Rogét Chahayed Classical virtuoso making waves in hip-hop

Neural DSP Nano Cortex Infinite guitar amps in your pocket

WIN

SLATE DIGITAL

ML-1 + SOFTWARE WORTH $ 2000 www.soundonsound.com

ON TEST: GRACE / KORG / ART / IZOTOPE / BOSS / SOYUZ / FABFILTER / YAMAHA / NEXTMIDI / NOVATION / SAMPLE LOGIC

TECHNIQUE: CONTROLLING MODULAR FROM STUDIO ONE / NOISE-FREE GUITARS

USA $11.99 / Canada C$13.25

LE ADER

DUTY CALLS F

or as long as I can remember, British musicians have been indignantly complaining about the high prices of gear imported from the USA. “How is it fair that this thing costs the same number of pounds as it does dollars?” we cry, conveniently forgetting that shipping things across the Atlantic costs money, that the distributors who handle this need to get paid, and that UK prices include 20 percent value added tax. No doubt Americans feel equally wounded when they try to buy a Coles 4038.

By the time you read this, the US will have a President who seems to favour protectionism.

It’s not entirely clear whether Donald Trump sees tariffs as an economic good in themselves, or merely as a tool with which to pressure other governments, but there’s little doubt about his willingness to introduce them. If he does, the principal targets will be massive industries such as cars, food, chemicals and semiconductors.

And, being far too small to merit any special attention, the music tech business will find itself arbitrarily rolled up into some larger category such as consumer electronics or leisure goods.

It’s quite hard to predict how this will play out in practice. On paper, tariffs could benefit manufacturers who actually do make their gear in the USA — but if retaliatory tariffs from other territories mean it costs more to import parts

SOUND ON SOUND LTD (HEAD OFFICE) ALLIA BUSINESS CENTRE KING’S HEDGES ROAD CAMBRIDGE, CB4 2HY, UK T +44 (0)1223 851658 sos@soundonsound.com www.soundonsound.com

and export finished products, those benefits might turn out to be illusory. A lot may depend on the exact wording and scope of tariff rulings.

Will products that are ‘assembled in America’ be exempt from import taxes that apply to products made in China? Will the prices of SMD components shoot up as a result?

Forward-thinking manufacturers have already tried to mitigate possible outcomes here. I know of at least one company that has moved their manufacturing base elsewhere in Southeast Asia, in anticipation of probable tariffs on Chinese-made goods. Some non-US makers are also buying up or developing manufacturing capacity in the USA. It may be that the looming tariff war turns out to be a nothingburger, and that Trump is more interested in negotiating from a position of strength than in taxing imports. But it could be that we are about to enter a period of instability in international trade, and if that happens, the prices of imported equipment seem likely to rise. Taking into account the additional burden posed by the EU’s new GPSR Directive, it may even become uneconomic for some smaller manufacturers to sell to Europe.

So, if you’ve had your eye on that boutique US-made mic preamp or Japanese synth, this might just be a good time to click Check Out.

“If you’ve had your eye on that boutique US-made mic preamp or Japanese synth, this might just be a good time to click Check Out.”

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ISSN 1473-5326 A Member of the SOS Publications Group 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 2025 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.

www.soundonsound.com / February 2025

3

“Best Flagship Synth 2023” - Alex Ball/Bo Beats

“Best Hardware Synth 2023” - Attack Magazine

“The 3rd Wave might be my favorite synth released in the last 10 years.”

- Sound on Sound

“Best New Synth 2023” - Gear News “One of the most powerful synths you could imagine.”

- Music Radar

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keyboard and desktop

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124 INSIDE TRACK

IN THIS ISSUE

www.soundonsound.com

February 2025 / issue 4 / volume 40

FEATURES

WIN

84 Modular RYK’s Jake Knight on his passion for the Roland System 100m and making FM make sense.

98 Tackling Noisy Guitar Recordings Are buzzes, hums and hisses ruining your guitar recordings?

If so, here’s how to keep them to a minimum...

114 SOS Awards: The Results Find out what gear came out on top in this year’s SOS Awards.

124 Inside Track: Rogét Chahayed Virtuoso pianist Rogét Chahayed has found a unique niche as a classically trained hip-hop producer.

132 Spotlight:

Creative Pitch-correction Plug-ins

SLATE DIGITAL ML-1 + SOFTWARE WORTH $2000 PAGE 64

Love it or hate it, pitch-correction is here to stay — but there can be way more to it than fixing performance issues.

136 How I Got That Sound Steve Hackett shares the secret behind the ‘golden tone’ on his track ‘Every Day’.

142 Q&A Your studio and recording questions answered.

138 Talkback

146 Why I Love... My Keyboards

Brueder Selke & Midori Hirano talk about inspirational cables and explain why they try to adopt a childlike perspective to music.

John McLaren-Howard tells of his life-long passion for the black and white keys.

78 PLAY ALL DAY PLAYFADER

28

ADAM Audio D3V

90

ART Pro Audio MyMonitor II

95 94

Boss GX-10 Amp & Effects Modeller

140

Dreamtonics Felicia Sample Library

86

96

70

FabFilter Pro-Q 4

16

Five12 Vector MkII & Jack Expander Eurorack Modules

50 83

Neural DSP Nano Cortex Guitar Amp Modelling & Capture Pedal

Nextmidi Divisimate 2 Noise Engineering Dystorpia Distortion Pedal

Equaliser Plug-in

82

MXR Rockman X100

Arrangement Software

Sample Library

22

Korg Keystage

66

Novation Launchkey MK4 Controller Keyboards

Grace Design m303

Perfect Surround Penteo 6 Pro+

Active DI Box

Upmixing & Downmixing Plug-in

Hexinverter Mutant Machine Eurorack Module

74

78

Play All Day PlayFader Sequencer & Controller

Roland P-6 Creative Sampler

12

Sample Logic Ultrasphere Virtual Instrument

40

Solid State Logic 18 USB Audio Interface

18

Soyuz 011 FET Small-diaphragm Capacitor Microphone

Analogue Amp Simulation Pedal

Electronic Perspectives: Vintage 34 Electronic Musical Instruments Emergence Audio Flute Textures

46

KIT Plugins BBF66 & BBF67

Controller Keyboard

Book Review

140

JMG DarkFire

Analogue Modelling Compressor Plug-ins

54

Pulsar Audio Poseidon EQ Plug-in

Saturator Plug-in

Personal Headphone Monitor Mixer

8

95

Plug-in Suite

Active Monitors

14

iZotope Neutron 5

C O V E R

ON TEST

141

Wrongtools Osmosis Sample Library

58

Yamaha FGDP-50 Finger Drum Pad

141

Zero-G Cartago Sample Library

WORKSHOPS 102 106 108 110 112

Studio One Live Cubase Logic Pro Tools

ON TE ST

Boss GX-10 Amp & Effects Modeller With some great sounding amps and effects, audio interfacing and an external effects loop, there’s plenty to admire in this compact floor unit.

PAUL WHITE

H

oused in a rugged metal case with a treadle at the right-hand side, the Boss GX-10 may resemble a typical ‘floor unit’ type digital guitar processor, but it’s both a very sophisticated piece of kit and very easy to use. Thanks to it using the same large colour touchscreen as the bigger GX-100, it’s possible to find your way around most of the functions without ever having to dip into the manual, despite there being a wealth of choices and parameter options.

In addition to the treadle, the GX-10 features three footswitches that can be configured for stepping up and down the presets, bank changing or manual bypass of effects. Power comes from a robust external adaptor that connects to its mains cable using a figure-8 type two-pin mains connector, rather than the more familiar IEC connector (don’t leave for a gig without the correct cable!).

Key Features The GX-10 is also built around the GX-100’s sound engine, with 24-bit A-D and 32-bit D-A conversion, 32-bit floating-point processing, and a 48kHz sampling rate.

The engine uses what Boss call their AIRD (Augmented Impulse Response Dynamics) technology, which models the amp and speaker as a symbiotic system. When it comes to setting up a processing chain, up to 15 locations can be populated from a library of 170 effects and 32 amp types, 23 being guitar amps and nine dedicated to bass. (Some effects consume more DSP cycles than others, so there are some scenarios in which fewer than 15 effects can be deployed.) It’s also possible to create parallel chains containing both amps

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and effects, and there’s a range of speaker models and miking options plus the facility to import user impulse responses. Up to two footswitches or an expression pedal can be connected, and for those who want to incorporate existing stompboxes, there’s an external effects loop.

The main input is mono, with the choice of mono or stereo outputs, and there are relatively few controls — just six knobs and six buttons — but in conjunction with the touchscreen interface that’s all that’s needed. With the free GX-10 driver the GX-10 can act as an audio interface for your DAW software.

Connection with your DAW is over USB-C, and this also supports communication with the free Boss Tone Studio software, which allows patches to be edited on a Mac or Windows computer. The GX-10 is also compatible with GX-100 Livesets, which are accessed using Boss’s online Tone Exchange platform. An optional Bluetooth Audio/MIDI Dual Adaptor that fits into a rear-panel dock allows editing and audio streaming using an iOS or Android device.

Input sensitivity settings can be stored for different instruments, which is handy if some of your guitars have much higher outputs than others.

Block Rocking Effects As you’d expect, all of the usual suspects are present in the effects section, but

all genres are catered for too, from the hardest of hard rock to full-on ambient floatiness, and there are esoteric treatments available that make it possible to coax almost synth-like sounds from your guitar. There’s plenty of choice in the distortion, modulation, delay and reverb departments, including an emulation of Roland’s famous Space Echo, but on top of that you’ll find a rotary speaker, with adjustable fast/slow ramping rate, the classic Slow Gear for slow attacks, octaves, harmonic generators, defretter, slicers, shimmer reverb and so on, which can hint at synths, strings and organs.

By default, the four knobs below the screen control the amp gain and its EQ parameters as shown at the bottom edge of the display, though these can be reassigned if you have other needs.

Page buttons access controls that extend beyond the four knobs. Each patch or preset can accommodate up to two amps plus effects. The amps and effects are represented as hexagonal blocks on the screen, which also indicates the routing. Blocks can be bypassed when not needed and opened for editing just by touching them. If you need to populate an empty block or replace the effect that’s already in use, a menu of effect types can be displayed along the top of the screen from where you can drag an effect directly to the block of your choice.

Blocks can also be reordered by dragging them. You can then select the variation of an effect allocated to a block. For example, after loading Distortion you can then choose from a range of modelled distortion, overdrive and fuzz pedals.

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ON TE ST

B O S S G X- 10

The controls can be edited as if you’re turning the knobs of a physical pedal — with just enough parameters to get the job done, but not so many as to make life unnecessarily complicated.

The result of adjustments made either from the front panel or using the Tone Studio software are heard in real time. In many cases there’s more than one way to achieve a specific result, which is great: you can decide on a workflow that best meets your personal preferences. The expression pedal, which includes a toe switch, defaults to switching between volume or wah, but it can also be used to activate the tuner by rocking it fully backwards in volume pedal mode.

The tuner, which is impressively large and accurate, may also be accessed using two of the footswitches together. Other pedal functions can be set up using the GX-10’s assign matrix. The saving, naming and reordering of presets follows a familiar pattern and there’s the expected global setup menu, in which user preferences can be selected. Neatly, these include output options that adjust the output sound for different amplifier types, including their main inputs and, where applicable, their loop return jacks.

The GX-10 is a mono-input, mono/stereo-output device, and also features a headphone output, an expression pedal input, an external effects loop, and USB for both the control software and audio interfacing.

combining a little gentle compression with something like a Tweed or Deluxe amp model (or, for the more UK-centric, a Vox-inspired model). I finally settled on a dual-amp setup running a Deluxe alongside a Vox model, and managed to get very close to the sound I was after in just a minute or two, and the touch sensitivity of these Boss models delivered a very comfortable playing feel. The drive pedals also somehow manage to feel very analogue. In fact, overall, I’d say that the amp and effect emulations are up there with the best available in the modelling world.

“The GX-10 could easily replace a complete pedalboard.”

Presets & Tweaking There are 33 banks of three factory presets and 66 banks of user presets, the latter being populated with copies of the factory presets that the user can then adjust or replace. Working through the factory presets reveals a wealth of responsive guitar sounds, from sparkly clean Jazz Chorus up to heavier rock and metal. There are also loads of inspirational ‘guitar but not as wet know it’ patches.

There are relatively few presets that deliver that elusive ‘amp on the edge of breakup’ tone so beloved of blues and country players, but I found it really easy to set up tones of that kind by

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In fact, it’s when you start straying away from the presets that you start to realise just what the GX-10 is capable of. For example, combining some of the octave and harmonic modes with a rotary speaker emulation produces organ-like sounds, while Slow Gear into shimmer reverb adds a string-like pad to the guitar sound. And when you start down this road, you’ll find that working from the Boss Tone Studio software makes patch creation and organisation even faster than working from the front panel. And while I’m on the subject of software, when you record these sounds direct to your DAW, they still come across very convincingly.

The Full Monty?

The GX-10’s models and effects sound great and cover almost every base. There are lots of user-adjustable functions, for example to assign new functions to the control knobs and external switches, but if you like an easy life, the default options

should suit most users. I think there would be very few occasions when you might run out of effect slots, and the parallel routing options make for useful dual-amp setups or enhanced stereo arrangements. Also, I really took to the touchscreen and control knobs for efficient patch editing and, when gigging, the displayed patch numbers are huge, so you’re not left squinting!

All of which means that the GX-10 could easily replace a complete pedalboard. Still, if you have a favourite analogue overdrive that you just can’t live without, there are the send and return jacks on the back that let you add it into the GX-10’s signal chain. That being the case, I think Boss have perhaps missed a trick here: it they’d provided a 9V power output jack to power an external pedal that would have saved having additional adaptor cables trailing across the stage. Also, my favourite live amp happens to be a Roland Blues Cube, and I would have found it useful to have that as one of the amp models on offer here — though, that said, you can get very close to a Blues Cube sound by tweaking the Deluxe amp model and its speaker/ mic parameters. Maybe one for a future firmware upgrade?

There it is then: the GX-10 is a compact and rugged alternative to the GX-100, it is very easy to use, and it offers a broad spectrum of top-drawer sounds spanning all genres for gigging or DAW recording.

And all at an affordable price.

summary A compact and rugged multi-effects floor unit, with audio interfacing capability, a good range of great-sounding amp and effects emulations, and more power than you might think.

$ $399.99 W www.boss.info

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ON TE ST

Sample Logic Virtual Ultrasphere Instrument Sample Logic rev their own engine, cinematic style.

JOHN WALDEN

M

edia composers with an addiction to cinematic virtual instruments will be well aware of Sample Logic’s pedigree in the field through titles such as Motion Keys, Cinematic Guitars Motion, Bohemian and Morphestra 2.

Up until now, all of these instruments have been built within Kontakt. However, Sample Logic have how released a first instrument using their own dedicated UltraEngine platform; Ultrasphere. The principles remain similar — Ultrasphere provides a four-core sound-design platform built very much with film or TV composers in mind, but the instrument is presented as its own plug-in rather than requiring Kontakt. I can only assume this gives the company greater control over the features and functionality they feel they can offer. So, does Ultrasphere, with its UltraEngine, go full throttle?

Plus Ultra While Sample Logic have obviously decided to take their own route here, thankfully the broad layout of Ultrasphere’s

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front end does not stray too far from their earlier Kontakt-based instruments.

So, built upon a 10GB+ library of samples, the user is presented with a four-core engine with a powerful X/Y pad blending control located at its centre. An impressive collection of supplied presets — all searchable with a neat tag-based browser — provides ample reassurance that Ultrasphere is capable of generating some very impressive sounds.

The presets will keep you entertained for some considerable time, but it’s the sound-design potential that will keep composers coming back. The UltraEngine includes many familiar features across multiple sub-pages that go well beyond the main four-core home page display shown within the screenshot here. For example, you can

deep-dive into each individual sound core for access to the sample playback settings, gain envelope, unison detuning and stereo spread, pitch, filter, filter envelope, an FM module, sends and EQ. You get very flexible send effects (each core has independent send levels) and global effects options, each with four effects slots with any of 18 different effects types available. The modulation options are impressive, with six LFOs (these can use standard waveform shapes or a gate mode for step-based options) and both key (note) and velocity modulation.

All of these offer user configuration.

The Sequencer module options are particularly impressive, with two independent arpeggiators, each of which can be enabled independently for each core. However, the Arp tab also includes a Chord option, enabling you to map a set of chords across 16 keys for easy triggering, and this can be engaged in combination with the arpeggiators.

One further neat feature is located towards the bottom of the main page and provides what are essentially low-pass and high-pass filters for each of the four cores. These are in addition to the ability to define MIDI note ranges for each core (these can be found within the individual core controls) but it’s a really useful option allowing you to quickly tailor the overall frequency balance of your sound to suit your needs.

Ultra Sound Sonically, Ultrasphere is very much within the soundscape-meets-evolving-texture ballpark but it can certainly do melodic options (for example, combined with the arps, there are some very cool Stranger Things vibes to be had) and, with appropriate modulation settings, you can also get some impressive rhythmic pulses to give all those tension cues a little extra urgency. Whether it’s mystery and suspense, lo-fi grit, gentle washes of pads, deep and subversive, or dissonant and down-right scary, Ultrasphere will have you covered.

My hat is duly doffed to the new UltraEngine.

It’s quite an ambitious move for Sample Logic to have taken but they have pulled it off with some considerable style. No, Ultrasphere is not cheap (although, at the time of writing, it was available at a discounted launch price) but it is a very impressive instrument. The bottom line here is that busy media composers will undoubtedly find plenty to love, especially once they dig into the sound-design features the new engine has to offer. These are sounds that could fit happily into any top-tier film or TV project.

summary Sample Logic’s new UltraEngine provides an impressive platform for Ultrasphere. If you want some top-tier cinematic atmospheres with plenty of sound-design potential, this is an impressive candidate.

$ $299.99 W www.samplelogic.com

ON TE ST

ART Pro Audio MyMonitor II Personal Headphone Monitor Mixer Want to hear ‘more me’? Not a problem...

HUGH ROBJOHNS

N

othing enhances a performance as much as good monitoring!

With a competent FOH or monitor mixer and decent wedges or in-ears, the ideal balance can be found quickly.

But many gigs don’t have such luxuries, making a personal monitor system very helpful.

Overview ART Pro Audio’s MyMonitor II isn’t new, but I recently acquired one and was impressed enough to write this review.

It combines a mic preamp, line mixer and headphone amp in ART’s standard, robust aluminium case, and allows the user to balance the levels of their mic with mono instrument and stereo line feeds. The mic and instrument signals are duplicated to thru outputs to feed, for example, for a FOH console or audio interface.

There’s a handful of user controls on the front, and connectors on both front and rear. Power is via an included 150mA 12V DC wall-wart with a non-latching coaxial plug.

An illuminated on/off button

14

activates the unit, and buskers will be pleased to note that it can be battery powered. The stated 15mA current drain means a theoretical life of over 25 hours for an alkaline battery and more like 80 hours for a lithium one.

When a mic is connected to the front-panel XLR, the signal is passed to a rear-panel XLR for onward connection to the PA or recording system. A silent mute button alongside the input XLR could be viewed as useful or a liability! I expected the mic input and output to be a hard-wired pass-through, but it appears to be electronically buffered, as the specs give an input impedance of 4.5kΩ; I measured a 0.9dB loss at the output and clipping distortion appeared when the mic input exceeded +9dBu (3dB above the specified maximum). Either way, there’s generous headroom, needing well over 120dB SPL into a sensitive capacitor mic or more than 150dB SPL into an SM58 to cause an internal overload. My Audio Precision system revealed that the thru output, which comes via a buffer, adds just 0.001% THD (0.004% in total) and roughly 0.1dB of noise.

That wouldn’t worry me in a studio, let alone on stage!

Phantom power isn’t provided but can be passed on from a console or interface. The maximum preamp gain through to the headphone output is +44dB, which may sound stingy but it is ample in practice, even with an SM58.

The rear-panel, quarter-inch mono TS instrument input presents a 1MΩ input impedance and accepts signals up to +12dBu, and is clearly optimised for an

electric guitar or bass. Again, there’s a buffered thru socket (quarter-inch TS) for onward routing. A line-level source can be fed to this input, thanks to a front-panel button accommodating signals up to +32dBu. In this ‘line’ mode, the volume control spans -45 to +17 dB, while the default instrument mode’s maximum gain is +35dB. The instrument input will accept an electronic keyboard’s output, but I found the volume control was easier to fine-tune in line mode, due to the higher output level compared with a guitar.

The quarter-inch TRS line input receives an unbalanced stereo mix, such as from a headphone output. Its maximum input level is also +32dBu, and separate volume trims (-15 to +14 dB) are provided for the left and right channels: useful if there’s a click on one feed, for example. Keyboard players who prefer to hear their instrument in stereo can use the stereo line input, leaving the instrument input (in line mode) for mono foldback, if required. Of course, this requires the keyboard to be split externally to feed FOH (eg. using a stereo DI box), but this reallocation of inputs worked well for me.

Helpfully, the stereo headphone output is provided on both quarter-inch and 3.5mm sockets, so in-ears can be connected without an adaptor. There’s no global volume control but adjusting the four input levels to achieve an acceptable volume is not difficult.

The headphone output clipped around 2dB higher than the specified maximum (+12dBu), which translates to almost 0.5W into 32Ω. That’s quite generous.

Being picky, I’d prefer a locking power connector or a cable clamp on the battery door, and I plugged the instrument cable into the thru socket more than once (the latter is adjacent to the stereo input socket, while the input is lower down). But this doesn’t detract from the MyMonitor II’s usefulness or effectiveness. It’s an excellent aid to personal monitoring control, and for anyone wanting their own ‘more me’ control, it’s a great solution.

summary A simple, compact and very effective personal monitor controller, the MyMonitor II boasts good technical quality and a well thought-out feature set.

The MyMonitor II features a number of thru outputs, for onward connection to front-of-house consoles, amps, PAs or recording interfaces.

February 2025 / www.soundonsound.com

$ $129.99 including VAT. T Yorkville Sound +1 716 297 2920.

E sales@artproaudio.com W https://yorkville.com W www.artproaudio.com

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If you ask the only person, who really matters. The user.

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ON TE ST

Noise Engineering Dystorpia They’ve engineered synth noises for years, but this is the

Distortion Pedal California-based company’s first foray into the guitar-pedal world.

R YA N M C C A M B R I D G E

T

he Dystorpia is the first guitar pedal to be released by Eurorack modular veterans Noise Engineering. It’s appropriately named, but while its billing as a “futuristic fuzz” accurately describes the essence of this pedal, it doesn’t really do its full capabilities justice.

Fuzz, Fold & Features Noise Engineering are known for a range of well-thought-out Eurorack modules with tailored palettes, on which any given knob-turn moves from one sweet spot to another. That same philosophy lies at the heart of the Dystorpia’s highly interactive parameters, which largely centre around two distortion circuits: Fold (wavefolding); and Pura (a multiband full-wave rectifier).

To my ears, Fold is a bit scooped but with some aggression in the high mids, and it’s unlike most other guitar fuzz pedals; wavefolding is more often associated with synth distortion. It can be brash and offensive if pushed too far, but it often works very well for guitar parts that could benefit from some added character. Pura is generally fuller-sounding, particularly in the mids and low mids, and sounds more like what you’d expect of a guitar fuzz.

Pura is clearly intended to complement Fold, and the Dystorpia’s versatility really shows when the two are combined to create unique textures. Overall, the Dystorpia sounds modern, full, aggressive, and unlike any other fuzz pedal I’ve used.

Small knob changes can make big tonal differences, though, and while you might be tempted to crank everything up to see what happens, I’ll save you the bother: absolute sonic destruction! I learned quickly that an eighth of a turn makes a big difference, and that uncovering subtle textures required a nuanced touch.

Interestingly, the Gain, Mid Band and Tone knobs are all active, even when the distortion’s Blend (wet/dry) knob is turned down. There’s a surprising amount of tone shaping here, so much so that I could achieve vintage overdrive tones with just the Mid Band and Tone knobs at around 10 o’clock, with some Gain inched in for

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some great saturation before the signal hit the other distortion circuits. Adding a bit of Pura back in, I could even find tones that resemble small-speakered valve amps, such as a Fender Champ.

Here’s an unexpected bonus: the Tone knob is mapped to the expression jack, to allow wah-wah-type effects with an expression pedal; the Tone knob sets the lowest value the expression pedal will use when heel-down. Cleverly, the expression input also works with CV signals, and when using the Dystorpia to process my Eurorack’s audio I didn’t even require an amplifier/attenuator module to convert the levels. The Dystorpia may have been designed for guitar, but it can really mangle synths in a beautiful way.

Amongst the other ‘special’ features is a Doom switch, which applies a lower octave when Sub is selected, and Blown adds the same sub at a higher volume, along with the added crunch of a distorted sub speaker. With Sub, you can craft some classic octave-fuzz sounds, which work really well for both guitars and synths. The Envelope switch manipulates the dynamics of the overall signal, its Dyn setting applying an envelope that follows the clean input, but shapes the output, lessening the overall sustain and emphasising note attacks to improve articulation. It’s a very capable solution to an issue that often plagues fuzz pedals. The Gate setting of the Envelope switch enables a noise gate that sounded pretty natural to me. Another unexpected function is the Freeze footswitch, which seamlessly loops a fragment of what you just played. The switch can be tapped to latch/unlatch the freeze or be held down for a momentary freeze. It enabled me to explore Lynchian soundscapes with interesting industrial drones that I could then play over.

I reported one small oversight: the lack of label to indicate the Gain knob’s secondary function as an output level

control, accessed by holding Freeze and turning Gain. It was explained in the manual, but Noise Engineering agreed with me, and it will be labelled on all new units.

Dystorpian Future?

The Dystorpia can do some conventional things, but ultimately it wasn’t created for guitarists looking for another Big Muff clone.

In very Noise Engineering fashion, it’s an unconventional, unapologetically digital pedal that accesses sounds that would be impossible in the analogue world. It rewards adventurous explorers who live to push through new sonic doorways and discover new frontiers of fuzz. Guitarists should rejoice that this innovative company have brought into the pedal space their years of experience and perspective from outside the walls of guitardom. I’m excited to see what they come up with next.

summary The Dystorpia exploits Noise Engineering’s modular synth nous to deliver an unusual fuzz pedal with some special features that plays as nicely with synths as with guitar.

$ $399.

W https://noiseengineering.us

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ON TE ST

Soyuz 011 FET Small-diaphragm Capacitor Microphone

Soyuz’s latest pencil mic is as easy on the ear as it is on the eye!

NEIL ROGERS

A

lthough they’re competitively priced for hand-built microphones, Soyuz’s flagship valve and solid-state models nevertheless occupy a price range that is out of reach for many of us. Some of their newer releases, however, suggest a decision to offer more accessible options. The subject of this review is one such: a miniature mic to sit alongside their well-regarded 013 series of small-diaphragm valve and FET capacitor microphones.

Out Of The Wood A pair of 011 FET mics were sent to me for review and, as a long-time SOS reviewer,

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I struggle to recall a more aesthetically pleasing package arriving at my studio.

These are small but classy-looking mics, and they come in a beautifully understated wooden case that houses the mics, -10dB pad attachments, and functional mic mounts. A case doesn’t matter that much, of course, but when you have other mics that are well over 50 years old, it feels reassuring when a modern company puts out a product with the same kind of longevity in mind.

As I hinted at earlier, the 011 FET is somewhat related to the more upmarket 013 FET, but there are some significant differences that are worth a little investigation. Aficionados of pencil mics often lament the discontinuation of Neumann’s transformer-balanced KM84 in favour of the transformerless KM184.

Although the ‘vintage is better’ mantra can be tiresome, there’s no doubt that the 184 — whilst still being a very good mic —

has a different character and can produce subjectively brighter, less forgiving results in some recording situations.

I mention this as the KM84 is often seen as a reference point for small-diaphragm capacitor mics, and in one key way the 013 and the 011 reviewed here are analogous: one of the main differences between them is that the newer mic also represents a move to a transformerless design. The other major difference is capsule size, with the 011 featuring a 19mm capsule compared to the 25mm one found in the 013. The overall size of the 011 is also smaller; these are very compact little mics, and more than one client of mine referred to them as ‘cute’. Us serious recording engineers would never describe a mic in those terms, of course, but smaller mics can have some practical positioning advantages in tight spaces such as below a hi-hat, and in circumstances

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S O Y U Z 011 F E T

where you need a very mobile or discreet recording setup.

The published specifications for the 011 FET are fairly typical of this type of mic. Sensitivity is a relatively generous 19mV/Pa, and self-noise is quoted as 16dBA, which is very acceptable given the small dimensions of the capsule. The frequency response plot shows a series of gentle peaks and troughs from about 2kHz up, but is within ±2dB of flat from 30Hz to nearly 20kHz. The 011 is said to be able to accept SPLs of up to 130dB (presumably with the pad in place).

In Use

pencil mics on drums, but there was a solidity and smoothness to the cymbals that worked well when combined with the spot mics. The -10dB pad was essential in this role, and the process of screwing and unscrewing the capsules to attach this was pleasingly smooth.

Next up was piano, which was hands-down my favourite source for the 011s during the review period. The upright piano at my studio is always well received by bands wanting that ‘darker’ Radiohead-style piano sound, and if I’m in a hurry I normally just point a pair of

at, and I tracked three or four different players over the review period with great results. In particular, the low end rarely seemed to be excessive, a feature that is often an advantage when recording acoustic instruments. I frequently find myself directing mics away from the sound hole on acoustic guitars to avoid excessive ‘boominess’ (something that is further exaggerated by the proximity effect), but I found with the 011s that I could stray a little closer to that area of the guitar if needed.

I mentioned using the -10dB pad attachment for drum overheads, but even with the pad employed, I’d be nervous to use these microphones as a top snare mic on a loud drummer. This is a shame as their smaller size makes them perfect to tuck under a hi-hat — and they sounded great when I tried them out with myself playing lightly. I used them more than once as bottom snare mics, though, and they also sounded ideal as hi-hat spot mics for providing a little extra detail and stereo placement in a mix.

“The 011s sounded wonderful... with a great sense of stereo imaging, while also sounding clean but not sterile.”

My first use for the 011s was as a pair of drum overheads for an enthusiastic young band I had in my studio for a day. It’s a job where I might typically favour a pair of large-diaphragm mics — or ribbons — but I was keen to hear if the 011s had any of that unpleasant ‘tizzy’ harshness that certain mics can exhibit, especially on cymbals. I was ready to quickly swap the mics out if needed, but was immediately reassured when I returned to the control room for a quick listen. These diminutive mics had that ‘contained’ low end and bright, forward sound that I associate with

mics straight at the exposed front strings and we’re good to go. The 011s sounded wonderful in this position, with a great sense of stereo imaging, while also sounding clean but not sterile. These also appeared to be very quiet mics, which can be one of the advantages of a transformerless design.

Recording acoustic guitar was another task the 011 seemed to excel

Summing Up As you can probably tell, I was impressed by these small-diaphragm microphones. If you crave a vintage-style mic — which many do, judging by the price of original Neumann KM84s now — these could be just what you’re looking for. And unlike a vintage mic, you won’t have to worry about it breaking down or going wrong. These small but serious little mics won’t flatter what you put in front of them, but they certainly won’t exaggerate any harsher aspects of a sound that you don’t want, as many newer pencil mics seem to do.

It would be wrong to describe a mic on the other side of £400 as ‘affordable’, but for a quality hand-built mic designed to last a lifetime, the Soyuz 011 should be a serious contender.

summary Soyuz’s new 011 FET is a small but classy-sounding affair, with a smooth yet detailed sound that suits a wide variety of acoustic instruments.

The 011 ships in a wooden box, along with a mic clip and screw-on -10dB pad attachment.

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$ $499, matched stereo pair $999.

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F A B F I LT E R P R O - Q 4

EQ sketching in action: when I let go of the button, Pro-Q creates EQ bands to realise the curve I’ve outlined with the mouse. These screens also show off the new Scaling feature, which has made the text and controls larger relative to the window size.

FabFilter Pro-Q4 Equaliser Plug-in

One of the most influential plug-ins of our time receives an impressive makeover.

FabFilter Pro-Q 4 $179 PROS • Scaling and the Instance List improve the excellent user interface still further.

• Spectral equalisation offers effective resonance suppression and control of harshness.

• Dynamic EQ now more versatile.

CONS • None, although not all of the new features will appeal to everyone.

SUMMARY Pro-Q was already one of the most powerful and intuitive EQ plug-ins on the market, and version 4 represents a big step forward in both respects.

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SAM INGLIS

W

hen we think of products with a revolutionary impact, we tend to focus on the past. It’s easy to see that the Minimoog, the DX7 and the SSL console represented turning points in music technology. Identifying products that are making a comparable difference today is harder — but FabFilter’s Pro-Q must be a pretty strong candidate.

When it appeared, back in 2011, the contrast between Pro-Q and other equaliser plug-ins was stark. This wasn’t so much to do with the way it sounded as with its radical user interface. Even if they weren’t actually emulating hardware EQs, the large majority of plug-in equalisers retained many of their design conventions: there would be a fixed number of bands,

and although you could sometimes move nodes around on a frequency plot, the primary means of interaction was to click and turn virtual knobs. Pro-Q wriggled free of this straitjacket, reimagining the plug-in equaliser using the principles of good software interface design instead of preconceptions inherited from hardware. It was efficient, intuitive and powerful, and it also sounded very good.

With the benefit of hindsight, success seemed assured, but perhaps not to the degree that it actually happened!

Pro-Q quickly became something like an industry standard, and has probably been referenced more in SOS’s Inside Track articles than any other plug-in. Its distinctive interface design has inspired numerous imitations, and FabFilter themselves have continued to develop and add new features to Pro-Q. Version 2 enhanced the interface even further, while adding new EQ shapes and an EQ ‘match’ function, and version 3 introduced dynamic EQ bands alongside many other improvements. And it’s fair to say that the new version 4 leaves the conventions and limitations of hardware EQ even further behind. As before, it’s available in all major native formats for macOS and Windows, and is authorised using a simple licence key.

Quick On The Draw Once again, FabFilter have been tinkering with the user interface, and the Pro-Q 4 colour scheme is noticeably more sombre than that of version 3. The window can still be resized or toggled to full-screen view, while there’s a new enhancement in the form of a Scaling option. Whereas the Size settings change the area of the window without altering the dimensions of the text or of controls such as sliders and dials, Scaling applies to everything.

Higher than default Scaling values are particularly helpful when engaging with the ‘expert’ pane that can be opened up when you put EQ bands into dynamic mode, making the otherwise minuscule controls much more accessible.

FabFilter have also added a new method of creating EQ settings. As before, the Pro-Q window is empty by default. Hovering the mouse pointer over the centre line will cause a yellow

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F A B F I LT E R P R O - Q 4

comparing the spectral content of a target and a reference track and generating an EQ curve to nudge the tonality of the former towards the latter. Previously, this involved a slightly un-FabFilter-like series of mouse clicks and menu selections, but the Instance List provides a quicker alternative way of accessing the same feature.

A final user interface enhancement, which applies to both standard view and the Instance List, is the ability to copy EQ bands between instances.

The drop-down menu that appears when you click on an EQ band’s node in the main graph includes Cut, Copy and Paste options, which work as expected, although you need to create a node before you can paste settings onto it.

Scrolling through the Instance List.

blob to appear, and clicking will turn this blob into an EQ band. So far, so familiar, but in Pro-Q 4, you can also create an arbitrarily large number of EQ bands in a single mouse motion.

The paintbrush icon in the lower left of the window puts Pro-Q 4 into ‘sketch’ mode. The idea here is that you can simply click to the left of the frequency graph and drag left to right across the window. As you do so, Pro-Q 4 will track any vertical movements you make and adapt them into EQ settings, creating multiple bands that interact to replicate the shape you’ve sketched. There are other EQs that allow you to ‘draw’ curves, but the Pro-Q implementation is rather different from what you’ll find in something like Harrison’s AVA Mastering EQ. Your mouse moves are treated as a loose guide rather than a precise template, and you’ll typically end up with only two or three bands created from an entire sweep. It does take a little practice to get the results you want, especially as the entire line remains ‘live’ even as you get towards the right-hand side; given how simple it is to create bands the existing way, I rarely found it quicker to sketch my curves, but your mileage may vary. I can imagine it being a boon for those using alternative input devices such as trackpads or touch screens, for example.

Shared Values The other major user interface development in Pro-Q 4 is something called the Instance List. Over the years, FabFilter have been steadily enhancing

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Ringing Endorsement the ability of multiple Pro-Q instances to communicate with each other, and this reaches its logical conclusion in version 4. Click the button at the lower centre of the Pro-Q interface, and the spectrogram view will shrink to a thumbnail version, displayed as part of a vertically scrolling carousel that also includes all the other instances in your DAW project. A zoom setting lets you easily make as many thumbnails as you like visible at once.

Click on one of them and it will grow to the full width of the window, allowing you to create, select and adjust EQ settings just as if it was the ‘home’ instance.

Helpfully, though, the thumbnails for the other instances remain visible, so you never forget you are editing via the Instance List rather than directly.

The upshot of this is that you can, in effect, do all of your EQ settings across multiple channels without having to open and close plug-in windows. Not only that, but even in thumbnail form, each instance displays a very clear animated spectrogram, so you can visually compare what’s going on. Given how powerful Pro-Q 4 is, and the fact that it incorporates some pretty sophisticated dynamic features, you could get an awful lot of mixing done by inserting an instance on every track in your project, pinning one of them to fill a second monitor and keeping it there. The only feature that seems to be missing is any way to change the order in which the instances appear in the Instance List.

Talking of communication between instances, Pro-Q has long had the ability to perform ‘matching’ equalisation,

The sophisticated dynamics features I mentioned have become significantly more sophisticated in Pro-Q 4. In version 3, EQ bands set to dynamic mode provided a choice between automatic and manually configured threshold settings, but there was no user control over time constants. Version 4 adds miniature attack and release dials. As with all of FabFilter’s dynamics plug-ins, there’s an element of programme dependence regardless of where they’re set, and hence they are not calibrated in milliseconds or other meaningful units.

The benefit of this is that compression always remains smooth and free from distortion regardless of setting and source material. The variable time constants make Pro-Q’s dynamic equalisation significantly more flexible, although personally I’d still not want to be without FabFilter’s superb Pro-MB multiband plug-in as well.

Click on the per-band drop-down menu where dynamics mode is enabled and disabled, and you’ll also find a new option entitled Make Spectral.

This enables FabFilter’s take on the concept of ‘resonance suppression’, as pioneered in products like Oeksound’s Soothe and Soundtheory’s Gullfoss. In fact, if you’re a Soothe user, the control set for a spectral band will feel instantly familiar. Like dynamic bands, spectral EQ bands are created with default settings, but clicking on a tiny double arrow icon opens up an ‘expert’ pane with manual control over threshold, attack, release and Spectral Density.

The first three are self-explanatory,

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F A B F I LT E R P R O - Q 4

while the last of these controls the granularity of the spectral equalisation.

Spectral bands also resemble dynamic bands in having a double arrow that controls the range of the dynamic action. Importantly, this can act in the opposite direction to the band’s main gain setting, so with dynamic bands, for example, you can apply a high-frequency boost with a dynamic component that introduces momentary high-frequency cuts whenever a peak in the source material comes along. This offers similar benefits with spectral bands, allowing you to simultaneously boost and smooth out a frequency range. On a solo vocal, for example, a single spectral band makes an incredibly effective and natural de-esser, and the spectral action can be combined with a treble boost to brighten up the vocal whilst retaining tight control over sibilants. I also found spectral equalisation invaluable on drums played with brushes or hot rods, allowing me to dial down the ugly upper-midrange grit without dulling the drum sound too much.

It is, of course, also possible to apply positive gain on the range control, so that Pro-Q 4 will boost rather than attenuate resonances, although I’ve yet to find a real-world use for this.

Compared with a dedicated plug-in such as Soothe 2, Pro-Q 4’s spectral equalisation doesn’t offer the same ability to adjust the core algorithm, and it seems perhaps more level-dependent, but it nevertheless sounds excellent.

(Understandably, it also introduces significant latency, even if you have the plug-in in ‘zero latency’ mode.)

For taming broadband sources such as full mixes I’d still choose Soothe, but on individual sources where you can easily identify the problematic frequency region, I found Pro-Q quicker and easier to set up. Spectral equalisation is easily my favourite new feature in Pro-Q 4, and having used it, I never want to return to a plug-in folder without it.

A spectral EQ band in action. Pro-Q is dynamically applying narrow cuts within the treble frequency band in response to resonant frequencies in the source.

but are all designed to be super-clean.

There has never been any pretence at introducing the non-linearities introduced by transformers or valves in hardware EQs. By default, Pro-Q 4 retains this clarity, but it also offers two alternative Character modes named Subtle and Warm. These are global settings, and influence the sound even when no EQ bands are instantiated.

Performing null tests with these modes makes clear how insensitive the human ear can be to some types of distortion.

all, I think it’s quite well judged, though in most circumstances I think I’d turn first to a dedicated saturation plug-in with more control. It would also be nice to have the option to switch modes for all Pro-Q instances in a project at once.

Adding new features to software products can sometimes bring costs as well as benefits. That’s especially the case with a product like Pro-Q, where extra functionality is welcome only if it doesn’t compromise the uniquely fluid and intuitive user interface. And, with Pro-Q 4, FabFilter have pulled off the notable feat of adding a ton of extra functionality whilst actually improving usability. Not all of this functionality will be useful to everyone, and personally I didn’t get a lot of mileage out of either the sketching feature or the warmth options, but neither adds any unwelcome complication. By contrast, spectral equalisation is something I’ve used on every single mix since I installed Pro-Q 4, and now wouldn’t be without. And best of all is that it comes alongside features such as Scaling and the Instance List which actually improve an already stellar user interface. Hard as I’ve tried, I really can’t fault it!

“Spectral equalisation is easily my favourite new feature in Pro-Q 4, and having used it, I never want to return to a plug-in folder without it.”

Bring The Heat The fact that Pro-Q is not a vintage EQ emulation has always been reflected in its sound as well as its user interface.

It has long offered the choice of minimum phase, linear phase and FabFilter’s proprietary ‘natural phase’ mode; these can be audibly different on some material,

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In isolation, Subtle mode is exactly that, and I could rarely hear a clear difference on individual sources when toggling it on and off. However, duplicating an entire mix and inverting the polarity of one copy whilst applying Subtle mode to the other reveals that it’s actually generating quite a bit of additional harmonic content in the high-frequency range. Switch to Warm mode and the level of these saturation products increases by several magnitudes, and their emphasis shifts to the midrange rather than the air band. Even so, Warm saturation is only marginally audible in isolation on some sources. On others, especially vocals, it adds noticeable ‘hair’ that is often quite pleasant. All in

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ADAM Audio D3V Active Monitors

How much tech can you cram into a compact set of desktop speakers? A lot, if you’re ADAM Audio.

BOB THOMAS

B

erlin-based ADAM Audio are on a roll, their new D3V active desktop monitor system being the third active monitor to be launched by the company in as many years. The D3V is designed to address the needs of musicians, producers, engineers,

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podcasters, vloggers, influencers and the like whose workflows are primarily desktop-based, often using laptop computers. ADAM already have a presence amongst this user group, courtesy of their highly regarded T5V and A4V active monitors. However, the D3V’s specific focus on desktop monitoring, combined with a wallet-friendly price,

make it an attractive choice not only for those creators whose specialist needs it directly addresses, but also for computer users in general who want to listen to high-quality streaming audio.

Opening The Box It wasn’t until I’d taken the two D3V monitors out of their packaging that I realised just how compact the enclosures are, at 240mm high (including the stand), 115mm wide and 150mm deep.

Aesthetically, the D3V follows ADAM’s

ADAM Audio D3V $299 PROS • Superb overall audio performance, particularly in the bass.

• Direct connection to your computer via USB. • Attractively priced and excellent value for money.

CONS • None.

SUMMARY A high-performance, attractively-priced, active desktop computer monitoring system that brings ADAM Audio’s two and a half decades of studio experience directly into this specialised area for the first time.

signature paradigm of angled front baffle edges, and is available as standard in either black or white.

The D3V cabinets are moulded from PC-ABS, a polycarbonate and acrylonitrile butadiene-styrene blend that is commonly used in the automotive and consumer electronics industries because of its high strength, stiffness and heat and impact resistance. I didn’t test the D3V for any of those qualities, but both enclosures felt solid and inert, as I’d expect of a monitor cabinet. The moulding process has allowed ADAM’s designers not only to soften the angularity of the front baffle’s edges, but also to curve the surfaces of the enclosure, creating, to my eyes, an attractive appearance overall. Moulded into the bottom of each enclosure is a 3/8-inch microphone stand thread together with a tapered tongue and groove slot, which accepts the supplied 15-degree-angled padded monitor supports. Also included is an external power supply with the stiffest, best-fitting connector that I’ve ever come across; a four-way cable to connect the left and right speakers; a USB-C to USB-C cable with a USB-A adaptor for connection to a computer; and six stick-on pads (three for each speaker) for mounting the D3V directly onto the surface of your desk.

At the bottom left of the ‘primary’ speaker’s front baffle (which carries a ‘Left’ label) sits a large multicolour LED indicator in the shape of ADAM’s apple-shaped logo. Next to that you’ll find a stereo headphone mini-jack socket, and beside that — and far enough away so that even my fingers didn’t catch the

headphone jack or cable — is the D3V’s combination volume knob and mode switch. The secondary speaker (labelled ‘Right’) has none of these.

ADAM Drivers The front baffles feature all-new high-frequency and bass/mid drivers.

The 1.5-inch D-ART (Accelerated Ribbon Technology) pleated-foil tweeter is hand-made and was specially developed for the D3V. A HFPS (High Frequency Propagation System) waveguide is moulded as part of the baffle, and this houses the innovative approach to the Air Motion Transformer concept (originally developed by Dr Oskar Heil) that is the signature of ADAM’s ART tweeters. In an ART tweeter, the pleated folds compress and expand, drawing air in and squeezing it out in response to the audio signal being applied to them. This movement enables the membrane to move air faster than it itself is physically moving, giving ART tweeters the ability to deliver a fast, precise transient response, which results in a clear, detailed high end. The large acoustically active area within its pleated membrane also allows an ART tweeter to produce both a highly dynamic output and a wide horizontal and vertical dispersion, which is controlled by the aforementioned HPFS. The new 3.5-inch woofer is also a bespoke design, featuring a uniquely profiled aluminium cone that combines light weight with stiffness. The woofer’s ferrite magnet, reinforced voice-coil joints and long-linear-excursion design enables it to push high volumes of air whilst avoiding breakup or audible distortion.

In addition to their tweeters and woofers — and in a first for ADAM Audio — the D3Vs feature twin passive radiators, one on each sidewall. The radiators’ 2.5-inch flat panels, which appear to be made of PC-ABS like the enclosures, sit in ribbed rubber surrounds, which provide the stable travel path essential to the delivery of a tight and controlled extension to the woofers’ innate low-frequency response.

Although passive radiators can work well to extend the low-frequency response in larger enclosures (the Focal SM9 that I reviewed in SOS January 2013 is a good example), they are more commonly found nowadays in smaller enclosures. PRs are used as an alternative to reflex ports, either where there’s no room to accommodate

a port, or where, at higher SPLs, the noise from the turbulence caused by a high-speed volume of air passing through a small-diameter port would be, to say the least, problematic.

Powering Up Power to the drivers comes courtesy of two stereo PWM Class-D amplifiers housed in the primary cabinet. These deliver 30W RMS per channel to the tweeters and 70W RMS per channel to the woofers, a total of 200W RMS for the system as a whole. The D3V’s amplifiers and crossover point (4kHz) are under DSP control and deliver, according to the published technical measurements, a creditably flat frequency response with -3dB points at 48Hz and 22.6kHz. The effect of the passive radiators can be seen in the system’s ruler-flat 60-80Hz low-frequency response.

To assist you in adapting the D3Vs to their placement in your room, the backplate of the primary (Left) D3V carries three three-position Room Adaptation switches: Position (stand, wall or corner), Desk (none, small or large) and Room Acoustics (treated, moderate or untreated). The Position settings apply a gentle shelving response reduction in the low end (0, -3 and -6 dB); Desk applies fairly wide notches in the low-mid (0, -3 and -6 dB) and midrange (0, -2 and -4 dB) frequencies; and Room Acoustics applies a gentle shelving response reduction in the highs (0, -1.5 and -3 dB).

On the backplate of the primary speaker you’ll find connectors for the supplied external 24V DC 2.5A power supply, and the four-way cable that carries the output of the treble and bass power amplifiers to their respective drivers in the secondary loudspeaker. In addition, you’ll find a pair of 6.35mm TRS jack sockets for connecting a balanced analogue input signal (maximum input level +9dBu), and the USB-C port that allows you to connect your computer, tablet or phone directly to the D3V, removing the need for an external audio interface when monitoring from those devices.

In Use Setting up the D3V was simply a matter of plugging in the connecting lead from the primary to the secondary speaker, making the necessary digital and analogue connections (USB and www.soundonsound.com / February 2025

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ON TE ST ADAM D3V

headphone out from my Mac Pro), and plugging in the power supply via its tightly fitting connector.

Although the D3V has an auto-standby function, you can put the D3V into standby manually by pressing and holding the volume knob/mode switch for two seconds. Since there is no power on/off switch, the D3V powers up more or less instantly after being plugged in or being switched out of standby by a single press of the mode switch. The D3V powers up with the last-used input mode (digital or analogue) enabled, and the indicator icon illuminates in the corresponding colour (cyan or green respectively). Switching between digital and analogue inputs is accomplished by two presses of the mode switch.

A triple-press of that switch swaps left and right channels over, allowing you to place the primary speaker on your right, should you wish to (the indicator LED flashes orange three times, signifying that a swap is in progress). The D3V can be muted by a single press of the mode switch, the indicator LED pulsing orange to show that the system is muted.

Depending on the context, a flashing red indicator LED can mean either that the D3V’s DSP-controlled limiter is being triggered or that the system is booting up. If (or when) you install a firmware upgrade, the LED flashes purple at the start of the process, and flashes green three times at the end.

Positioning & Room Adaptation The D3V provides two desktop mounting options: either on a pair of short tabletop microphone stands (for example the K&M 23320) angled towards your listening position at ear level, or on its own stands, angled upwards and inwards towards you. Sitting at my desk, I’m around 90cm away from each speaker on my desktop and, being of the tall persuasion, I used a couple of shallow wedges to increase the upward angle. I did consider table stands, but I occasionally use wired headphones and I’d rather have their cable to the D3V running off to the side than hanging down from a stand.

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With the D3Vs set up on my office desk and connected via USB to my iMac so that I was hearing the D3V’s onboard D-A conversion, I began listening critically.

Positioned as they were, with the drivers sitting forward of the computer screen and bookcases behind the speakers, I felt that the system’s low frequencies had a slight tendency to overshadow its midrange performance. Switching in the low-frequency -3dB shelving filter brought the midrange back up to where I wanted it to be. With a couple of feet or so of desktop in front of the D3V I was slightly concerned that there might be

a problem with desk reflections, but that didn’t materialise, possibly because of the upward angle required to aim them at my ears. The high frequencies also didn’t require any attenuation.

Listening Time

Since the D3V is fitted with passive radiators and an ADAM ART tweeter, the first port of call had to be a CD with plenty of low bass and lots of detail in the higher frequencies. This meant that XXYYXX’s eponymous trip-hop album from back in 2012 was first up.

Although not as detailed in the low end as Deadmau5’s Grammy Award-winning 4x4=12 (which was up next), the XXYYXX album has a lot of low-level midrange and treble percussive detail in its tracks. I’ve found this particular CD to be a good test of a two-way loudspeaker’s ability to resolve that type of detail in the heavy presence of low frequencies.

The D3V coped magnificently with both CDs, which were played in from an external CD/DVD drive. Watching the passive radiators moving in and out as they translated low bass frequencies made mesmerising entertainment. Due to its size, the D3V naturally doesn’t move enough air to deliver the bass weight that I get from the ADAM A8Hs in my studio — but the speed and accuracy of its low-end performance blew away that of the 2.1 loudspeaker system and Toslink D-A converter combination that I’ve used for years with my iMac. The D3V’s reproduction and resolution of bass transients was very precise and, even at high volume levels (I never did get the limiter to flash), that definition didn’t blur. Midrange detail was superbly delineated, again with crisp transient detail throughout.

As with every ADAM ART tweeter that I’ve come across, the new, smaller D-ART variant excelled at resolving higher-frequency transient detail in both CDs without ever sounding harsh. Unfortunately for my bank account’s prospects, the D3V’s solid, expansive and detailed soundfield outperformed that of my current 2.1 system.

The ‘primary’ speaker, nominally designated Left, houses This high level of performance all the amplification and DSP electronics, as well as the was repeated across my collection headphone output and volume encoder/mode switch.

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ON TE ST ADAM D3V

“The power and accuracy of the D3V’s low-frequency extension and delivery is much better than you’d expect from its compact dimensions.”

The rear panel of the Left speaker hosts the analogue and USB inputs, the room compensation EQ switches, and a four-pin output to pass audio onto the Right speaker.

of favourite test CDs. To name three: the natural reverb tails in Cantus’ Spes disappear slowly into the dark void of a large church’s ceiling, providing as they die away a stern test of a loudspeaker’s ability to resolve low-level detail; Malicorne’s Vox, with its full-throated, hard-edged, a cappella male and female vocals stress test a loudspeaker’s abilities to extract each singer’s individual contributions and place them precisely in place; and Galician folk band Berroguetto’s superbly-recorded 10.0, with its blend of female vocals, accordion, Galician bagpipes, bass, kit drums, low pitched drums, ethnic percussion, and bowed and fretted stringed instruments will expose any loudspeaker’s weaknesses.

Since most of the films, videos and music that I watch and listen to nowadays are streamed via computer, I also spent time listening to music from my iTunes library (which includes tracks that I’ve produced, mixed and/or mastered) and major music streaming services.

I also sampled parts of several films and videos on Amazon and Vimeo.

The D3V, connected to my iMac via

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USB, delivered superb results from both streamed music and film soundtracks.

Connecting the headphone output of my iMac to the analogue inputs of the D3V produced results that sounded good. However, switching between the D3V’s analogue and digital inputs revealed that the D-A converter in the D3V is, to my ears, capable of delivering a more detailed and expansive soundfield than its equivalent in my iMac.

Overall Overall, the D3V performs superbly as the computer desktop monitor system that it was designed to be and, I feel, delivers excellent value for money. Due to its passive radiators, the power and accuracy of the D3V’s low-frequency extension and delivery is much better than you’d expect from its compact dimensions. The new 3.5-inch woofer and D-ART tweeter work seamlessly together and in conjunction with the passive radiators to deliver an impressively detailed performance.

In A/B testing with my ADAM AH8 monitors I found that, although the D3V cannot reproduce the A8H’s

power and clarity, it does have a quite similar sonic character. Since I know that mixes that I create on the A8H translate well to other studio monitors, domestic loudspeakers and streaming services, I am confident that any evaluations or decisions that I might make when monitoring computer audio through the D3V will do likewise.

If you are looking for your first computer audio monitoring system, thinking about upgrading your monitoring when creating content, or want to improve your listening experience with the music and movies that you stream and download from the Internet, then the ADAM Audio D3V should certainly be on your shortlist.

Frankly, until I listened to the D3V, I would never have considered spending almost £300 on a dedicated active computer desktop monitoring system.

Now I am having to look for reasons not to buy a D3V system, and am failing miserably in that endeavour. $ $299.99 per pair.

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ON TE ST

Neural DSP Nano Cortex Guitar Amp Modelling & Capture Pedal

Want to take your valve amp sounds on stage without breaking your back? The Nano Cortex makes capturing and delivering those tones quick and easy.

BOB THOMAS

F

ounded in Finland in 2017, Neural DSP offer a range of software and hardware emulations of guitar-related hardware, including amps, speakers and effects. Their debut hardware product, the groundbreaking Quad Cortex, catapulted them into the limelight in 2021. We’ve since reviewed and been impressed by a number of their machine-learning-based plug-ins, not least the Archetype series, which emulate the signal chains of specific artists.

They’ve now released a new hardware device, called the Nano Cortex, and it’s both more compact and

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more affordable than the Quad Cortex.

As well as offering access to a range of machine-learning-based amp and effects emulations, it features the company’s proprietary Neural Capture technology, which uses neural networks to learn (capture) and digitally replicate the sounds of amps, drive pedals and any combination of drive pedal, amplifier, loudspeaker and microphone.

Hardware & Software The Nano Cortex is housed in a stylish, compact enclosure. One of the first things you’ll notice is that there’s no screen here, with visual feedback instead provided by four

rows of five LEDs on the top panel, but it’s also very much designed to be used with the associated control app, of which more below. The top panel carries two momentary buttons, rotary gain and EQ controls, and twin footswitches, all of which feature LED indicators, and, as on the Quad Cortex, each footswitch also incorporates a continuous rotary encoder.

The sounds are based on ‘captures’ and these are stored, along with any effects, as presets. When a preset is recalled the two LED rows above footswitch 1 display its bank and slot position. Above footswitch 2, the upper row of LEDs indicates the active effect slots, with the LEDs of the pre-capture effects glowing orange and those of the post-IR effects blue. A blinking FX LED indicates that the Amount control is active on that effect. Footswitch 2’s lower LED row displays the active IR slot, and that’s selected using that footswitch’s encoder.

Each of the front-panel control knobs is surrounded by a segmented LED ring displaying its settings within the recalled preset. Of course, since these are pots rather than continuous rotary encoders, those settings may or may not coincide with the knobs’ current physical positions when you recall a preset. By default, the LED ring jumps to the position of a control as soon as its knob is moved, but you can change this using the app to require the knob to pass through the current setting before a change is made.

The rear panel carries a series of quarter-inch balanced TRS jacks, one for the mono instrument input, two for the mono/stereo output, and one for an expression pedal that doubles up as a MIDI I/O port. There’s also a ground-lift switch, a mini-jack headphone output with a volume control, and a USB-C port. The USB port serves several functions. First, it allows the Nano Cortex to function as a 4-in/3-out 24-bit/48kHz USB 2 audio and MIDI interface for Mac, Windows, iOS or iPadOS devices, which means you can record straight from this box into your DAW. If connected to a computer, the Nano Cortex can also be programmed to send, over USB-C, up to 12 Program Change and/or continuous controller messages per footswitch press and per preset on recall, which could prove very interesting for some users. And when connected to a Mac or Windows machine’s USB-C port the Nano Cortex can draw its power from the computer, though it’s worth noting that it cannot

be powered from older USB A ports; if connecting that way, you’ll need an external 600mA, 9-12 V DC power supply.

When not connected to a computer, you can use such a pedal-style PSU, or a 5V/2A USB-C charger or power bank.

On the right-hand panel, you’ll find a dedicated combi XLR/TRS Capture Input socket, along with an adjacent switch to apply a 26dB boost. This is used during the amp or drive-pedal capture process to receive the output from the mic/amplifier or drive pedal being captured; the second (right) output on the rear serves as the send from the Nano Cortex to the device you’re capturing.

While you can obviously select the loaded presets using the hardware controls, there’s also Neural DSP’s Cortex Cloud app for Android and iOS devices. This app, which communicates with the hardware over Bluetooth, really is essential. Not only does it let you download firmware updates, captures and cabinet IRs, but it also lets you upload your own captures to the Cortex Cloud, and manage the onboard preset, capture and IR libraries. What’s more, it gives you remote access to the Nano Cortex’s settings, including the front-panel controls and the parameters for all the effects slots. You can also use it to adjust the volume of an active capture (-24dB to +12dB) and save the result in a preset. For loaded IRs, the app offers comprehensive manipulation, allowing you to select (for factory IRs only) from one of five mics and six centre to edge-of-cone positions, and (for both user and factory IRs) to

Neural DSP Nano Cortex $549 PROS • Amp-capture process is extremely fast and highly accurate.

• Very easy to navigate and use, standalone or (especially) with the Cortex Cloud app.

• Attractively chic and petite.

CONS • Some users might have preferred at least a small screen on the hardware.

SUMMARY A compact, easy to use, gamechanging AI-based amplifier and drive pedal capture and replay device, the Nano Cortex doubles up as an audio interface. It can create highly accurate captures with unrivalled speed.

invert polarity, adjust level and apply fully variable high- and low-pass filters.

Prefab Sounds I’ll get to the user-capture side of things below, but straight out of the box you can use the Nano Cortex as an amp, cab and effects simulator. It comes pre-loaded with 25 amp-only captures (20 for guitar and five for bass). These are organised in five banks of five and directly accessible using top-panel controls. The quality of the factory content is good and most user bases are covered, from clean, through blues-y breakup to high-gain metal, but you can also store up to 256 captures in an onboard user library. I suspect few

The Nano Cortex can function as a USB audio and MIDI interface — and even allows you to capture the sound of your DAW’s virtual amps to take on stage.

www.soundonsound.com / February 2025

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NEURAL DSP NANO CORTEX

of us will be tempted to create so many captures ourselves, but many factory and user-generated captures are available through the Cortex Cloud app — more, in fact, than anyone’s likely to need in several lifetimes! For loudspeaker cabinet IRs it’s a similar story, with direct access to five of the 10 pre-loaded factory IRs, as well as the ability to store and load a further 256 IRs, whether downloaded or self-created.

The signal path consists of seven slots.

Two (Adaptive Noise Gate and Transpose) are pre-capture effects slots. Next comes the Neural Capture slot, which can hold drive pedal, amp or amp-and-speaker captures, as you prefer. Then there’s the IR loader, followed by three post-IR effects slots: chorus, delay and reverb.

The order and content of the five effects slots can’t be changed, but the app does allow quite deep editing of the three post-IR effects. In addition, you can activate or bypass each effect individually from the front panel, and can set how much of each active effect is applied to the post-IR signal.

All of the available parameters for each of the five effects can be adjusted and saved on a per-preset basis using the Cortex Cloud app. The top panel’s Amount control also gives you direct access to a single operational parameter: noise reduction level on the noise gate; ±12 semitones on the pitch-shifter; and dry/wet mix for each of the three post-IR effects.

Taking the effects and processors in turn, despite having only a noise reduction level control, the adaptive noise gate is very effective. I particularly like that it doesn’t continuously open and close around the cutoff level — it just keeps on closing until a new note or chord is struck. It was more than speedy enough for me, though I’m not sure that it will be quite fast enough to satisfy djentists.

I have to say that the decision to follow the noise gate with pitch transposition puzzled me at first — but I think I get it now! Control-wise, there’s a wet/dry mix

my ears, Neural DSP have nailed the DC-2W emulation, particularly its updated SDD-320 four-mode, six-sound switching.

The app gives you access to the wet/dry mix, a variable Drive parameter that adds a bit of weight to proceedings, control over the output level (to compensate, if necessary, for the Drive level), the SDD-320’s front-panel switching, and the Standard/SDD-320 mode switch.

Next comes Neural DSP’s emulation of the warm, vintage tonality of a BBD (bucket-brigade delay) type analogue delay. Through the app, you have control of the wet/dry mix, delay time, feedback, high- and low-pass filters, modulation rate and depth, stereo width, drive and ping-pong off/on. And, lastly, there’s the Mind Hall reverb, which emulates the sound of Lexicon’s famous 224 digital reverb, and ranges from the tight ambience of a small room right up to the seemingly endless reverb of a monstrous hall. The app gives you control over the wet/dry mix, decay time, pre-delay, damping and high- and low-pass filters.

“I’ve used amp-capture systems before, but the ease and speed with which this successful capture was obtained really has opened up a whole new way forward for me.” setting, a ±12-semitone pitch-shift, and separate high-pass and low-pass filters.

The transposed pitches always tracked flawlessly, but in practice the high- and low-pass filters will probably turn out to be more useful, as they can be included in any preset either individually or together, the latter arrangement meaning you get a fully-variable band-pass filter.

The post-IR effects trio begins with Neural DSP’s model of the Boss DC-2W chorus pedal. The DC-2W itself, in Standard mode, is an emulation of Boss’ original 1980s DC-2 Dimension C, while in its SDD-320 mode it emulates Roland’s legendary SDD-320 Dimension D. To

The side-panel Capture input can accept instrument or mic signals.

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The Capture Process Clearly, then, you have access to plenty of sounds right off the bat, but for many guitarists it’s the prospect of capturing the sound of their own amps, cabs and pedals that will be the biggest draw. The neural network that Neural DSP use for this is said to have been designed to perceive sound in a similar way to human hearing, so as to produce natural-sounding captures.

Something else that sets it apart from the competition is the sheer simplicity and speed of the capture process — after initial calibration, it takes only around five minutes from start to finish, and it can be done without a computer, an Internet connection or even the Cortex Cloud app.

That said, the app does make the capture operation much slicker...

To get started, you plug your guitar into the Nano Cortex’s input and your monitor speakers into output 1L, or headphones into the headphone output.

Then connect the Nano Cortex’s 2R output to the input of the target device, and feed the target’s output into the Capture input. That target output would be a mic, if you’re planning to capture an amp and speaker combination, of course.

The Nano Cortex doesn’t supply phantom

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ON TE ST

NEURAL DSP NANO CORTEX

power — it expects a passive dynamic mic, which is the classic choice for cabs — but you could use a capacitor mic if you have a separate phantom power supply. If you’d wish to capture just the amp, without the speaker, the best output would be from either a reactive load box, or a DI box that can be connected between your amp and speaker.

With everything hooked up, a press and release of the Capture button puts the Nano Cortex into Capture Calibration mode, and automatically switches the app to its first capture screen. Then you play your guitar to check the levels, adjusting the gain of the capture signal (using the front-panel control or the app) until the gain control’s LED ring turns green, and then you’re ready to go. Now, when the capture process starts (more on that shortly), you’ll discover that the audio signals produced by the Nano Cortex are highly dynamic. It was through a process of trial and error that I discovered the ‘best’ way to calibrate the signal for my setups, which was to thrash my guitar’s open strings, and turn up the gain until louder (not the loudest) sounds just turned the LED ring green.

Having done that calibration, just press and hold the Capture button for three seconds: the LEDs will turn off, and then start re-illuminating, one segment at a time, until the capture process ends. Note that if you’re intending to capture a 50-100 Watt combo or stack at high volumes using a mic, this process can get very loud. If in the same room, I’d highly recommend ear protection... and you might also consider giving neighbours of a nervous disposition advance warning!

Footswitch 1’s LED turns green when the process is over, indicating that your new capture is active. Using the footswitch, you can then toggle between the capture and original return signal, to compare the sounds. If happy with the capture, you can store it in a Bank slot (replacing the slot’s existing contents) or in the pedal’s User capture library.

On Test The first thing to say about my experience of using the Nano Cortex is that the quality of the onboard presets is good, and I could simply enjoy the experience of playing through it. In that context, I found it liberating not having any screen calling for my attention, though the Cloud Cortex app is very

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The capture setup for the author’s miked-up Fender Vibrolux Reverb combo, with both the hardware and control app in use.

useful, not least since it gives you so much more control over the effects settings. The app’s really easy to use too.

Personally, though, I was more interested in the amp-capture side of things. Again, this can be done using the Nano Cortex as a standalone device, but it’s definitely slicker with the app.

The first amp I chose to capture was my favourite, and one which I know very well: the THD Univalve, which has a highly detailed, dynamic sound, and is capable of resolving the differences between different types of input and output valves.

I set this up using its built-in Hot Plate load box DI output to send the capture ‘return’ signal back to the Nano Cortex.

After hitting upon the ‘trial and error’ approach to calibration I discussed above, I obtained my first successful capture — and, to my great delight, it turned out to be all but indistinguishable

from the actual sound coming from the Univalve’s DI output. The only difference was that the capture sounded ever so slightly smoother overall but, honestly, that was no bad thing! Importantly, the Univalve’s playing dynamics were faithfully reproduced, and I’d be more than happy to record or gig with this capture. I’ve since created a number of captures of this and other amp setups, and they’ve all been similarly successful.

I’ve used amp-capture systems before, but the ease and speed with which these captures were obtained is truly impressive, and I have no intention of sending the review unit back — it really has opened up a whole new way forward for me. While I have no intention of abandoning my valve amps either, or my collections of IRs and amp-modelling hardware and software, I now plan on capturing my favourite sounds from each

of them into the Nano Cortex to give me instant access to them all, whenever and wherever I am.

Diving deeper into the IRs, the phase, level and high- and low-pass filtering available for the factory cab IRs (each of which contain five microphones and six mic positions, so a total of 30 IRs per cabinet) proved extremely useful in matching them to individual amp/pedal captures before saving both capture and cab in a preset.

Also, the level of control over the effects offered by the app really opened up their potential, allowing me to tailor the three post-IR effects, in particular, to my requirements and taste.

Setting up the Nano Cortex as an audio interface for Reaper on my iMac was issue-free, with the Nano Cortex showing up as a 4-in/3-out USB interface. Recording is simply a case of connecting a guitar to the Nano Cortex’s input, creating a mono audio track in your DAW to record input 1 (the dry DI guitar signal) and a stereo track set to record inputs 3+4 (the processed signal

from the Nano Cortex), arming the tracks and hitting record. If you want to re-amp the dry track, simply route it to output 3 (which feeds the input of the Nano Cortex), arm a stereo audio track set to record inputs 3+4, and hit record. It’s easy enough, but to make life easier still Neural DSP have developed templates for some of the major DAWs, including Reaper.

Capture For Masses?

My abiding impression of the Nano Cortex is of a remarkable pedal that offers exemplary sound quality and exceptionally speedy and highly accurate captures. It’s incredibly easy to use, whether you’re working standalone or using the apps to embellish the experience. That said, if you ever do find yourself stuck, there’s a very good interactive online manual.

To put it another way, with its combination of pre-loaded top-flight captures, high-quality modelled effects, and its audio interface and re-amping capabilities, Neural DSP’s Nano Cortex

ALTERNATIVES In the growing galaxy of competitively priced, AI-based amp-modelling hardware, you’ll find alternatives from the likes of IK Multimedia, Kemper and Hotone. On the other hand, if you’re exploring the NAM universe, then Dimehead is a name to check out.

really is the gamechanger that I’ve been waiting for. It lets me capture all my favourite valve amps, drive pedals, amp-modelling software and hardware presets, and store them along with my favourite cab IRs in one compact, portable package — and for a price that won’t break my bank manger’s heart.

It even has a built-in tuner... So, if you’re looking for a high-quality AI-based amp capture and replay device, I’d encourage you to check out the Nano Cortex. Having done so myself, I can assure you that this one isn’t leaving here! $ $549 W https://neuraldsp.com

www.soundonsound.com / February 2025

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Solid State Logic 18

USB Audio Interface

If you think all 1U rackmounting audio interfaces are the same, SSL have some news for you!

SAM INGLIS

F

or a few years now, Solid State Logic have had the Midas touch.

Whether it’s mixers such as the SiX, BiG SiX and Origin, outboard like the Fusion, PureDrives and Bus+, their ever-growing plug-in library, or the UF and UC controller range, SSL have produced hit after hit. And what’s most impressive is that this success has been achieved with fresh thinking and novel designs. Instead of churning out copycat products, they’ve made a case for originality, new features and deep integration. So when I was offered an exclusive chance to test the company’s first ever rackmounting audio interface, I jumped at the opportunity.

But when I removed the 18 from its box, my face fell. With its neat rows of combi XLR/jack sockets, twin headphone outputs and simple monitoring controls, it just looked so... conventional. Had SSL finally started making me-too products?

Would I have to report that their purple patch had run its course? Well, spoiler alert: it turns out there’s more to the new 18 than meets the eye.

In terms of form factor, the 1U rackmounting 18 certainly follows a well-travelled path, and you’d have to be a sharp-eyed witness to pick it out from something like the Focusrite Clarett 8Pre+ in an identity parade. Dig a little deeper, though, and it becomes apparent that

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SSL’s designers have tried to put some clear blue water between their baby and the rest of the “eight mic preamps plus ADAT” crowd.

Numbers Game SSL’s desktop audio interfaces take their names from the number of physical inputs available. The 2 and 2+ are simple two-channel models, while the 12 sports four analogue inputs and up to eight digital ins over ADAT. However, the 18 actually has either 26 or 27 physical inputs at base sample rates, depending on how you count them. The first 26 are made up of eight analogue inputs on the aforementioned combi sockets, plus stereo S/PDIF and two sets of ADAT ports, each of which can accommodate eight inputs at 44.1 or 48 kHz. The 27th, if you count it as such, is an additional analogue input with its own preamp and compressor. Officially, this is intended for talkback use, but it can be digitised and recorded instead

of the S/PDIF inputs, and has some unique capabilities (see boxout).

On the output side, meanwhile, the count reaches 28, as there are five pairs of analogue outputs on quarter-inch jacks as well as the stereo S/PDIF and 16-channel ADAT outputs. There are also two headphone sockets on the front panel, although DAW software cannot address these independently. Like those on the 12, they are unusually versatile, with three gain options to suit more or less sensitive headphones, and the ability to act as balanced or unbalanced line-level outputs. The waters are also muddied, in a good way, by the provision of a pair of balanced insert points. Using these involves repurposing outputs 9+10 as sends, and returning the processed signal to a dedicated pair of jacks that feed inputs 1+2. Finally, there’s a word-clock output on the usual BNC socket, MIDI in and out on the less usual mini-jack format, and a single USB Type-C socket for connection to your Mac or PC. The 18 has

Solid State Logic 18 $1149 PROS • Excellent audio specifications.

• Supports professional +24dBu line-up levels.

• Comprehensive and well-designed cue mixing functionality in SSL 360.

• Dedicated talkback input with Listen Mic Compressor available as a recording source.

• Many other nice touches including insert points and very versatile headphone amps.

Alpha 8 By the time you read this, the 18 will have received its official launch at the NAMM Show, where Solid State Logic are also due to announce a companion product. The Alpha 8 is described as an “audio interface expander”, and features eight balanced analogue inputs and outputs on quarter-inch jacks. This is a product category that has been rather neglected in recent years, with modern ADAT expanders typically offering built-in mic preamps rather than line-level I/O. However, in typical SSL fashion, the Alpha 8 is much more than just an ADAT expander.

It does indeed have two pairs of optical output sockets, allowing the full eight-channel input count to be maintained over Lightpipe at sample rates up to 96kHz, but that’s not the half of it. The Alpha 8 offers two-way conversion between analogue, ADAT and S/PDIF formats, with flexible routing and a headphone output that can be used to

an internal, universal-voltage PSU fed from a standard IEC socket.

Inward Bound Solid State Logic have developed a couple of different mic preamp designs in recent years. The preamps from the Origin console, which also form the basis of the PureDrive outboard units, are the successors of the older VHD design, and offer not only a colossal gain range but also a variety of ways to colour the sound, by varying the input impedance and applying different saturation options.

CONS check what’s going where. And on top of that, it’s also a USB audio interface in its own right — which, when used as such, makes all eight analogue, eight ADAT and stereo S/PDIF inputs and outputs available simultaneously, for a total of 18 ins and outs.

Mac owners who have both an 18 and an Alpha 8 can therefore create an aggregate device to use both simultaneously over USB for a theoretical 44 inputs. Those using the Alpha 8 as a more conventional expander will also appreciate the fact that it has word-clock out as well as in, meaning there’s no need to muck about with ‘T’ connectors in order to clock further units. On the analogue side, the Alpha 8 not only matches the 18 in being able to cope with professional line-up levels, but allows every input and output to be independently aligned to any of +24, +20, +18 or +9 dBu. It also offers similarly excellent audio specifications.

The SiX and BiG SiX, meanwhile, use the ultra-clean SuperAnalogue preamp, while the audio interfaces employ a circuit that is different again. I’m not sure if this has a name, but it’s somewhat like a simplified version of the PureDrive design: clean by default, but with a switched ‘4K’ option that introduces a fixed treble boost and saturation to recall the sound of SSL’s 4000-series consoles.

All of these preamp designs have two things in common, the first being superb technical specifications. As implemented in the 18, the interface

• Does not offer digital preamp gain control or related features such as auto-gain.

• No remote control or iOS functionality.

• SSL 360 mixer might present a learning curve for anyone not au fait with the conventions of mixer design and layout.

SUMMARY Solid State Logic have added their own twists to a familiar audio interface format, offering class-leading audio specs and a heap of unique features.

preamp design offers a 67dB gain range and an astonishingly low Equivalent Input Noise of -130.5dBu. SSL describe this as “best in class” and as far as I’m aware, they’re not wrong. The second feature that unites all of SSL’s current preamp circuits is that they use analogue gain controls — switched in the case of the PureDrives, but continuous on the mixers and the audio interfaces. This would have been unremarkable a few years ago, but other audio interface manufacturers are increasingly moving to digital control, which in turn permits full recall and the implementation of technologies such as automated gain setting.

Similar levels of technical excellence are maintained elsewhere, too. All the analogue inputs boast a dynamic range greater than 120dB, while the analogue outputs achieve 125dB, and THD+Noise is below -100dB on the mic inputs and below -106dB on the line ins.

The rear panel is unusual in hosting an odd number of combi jack/XLR sockets, the extra one feeding the talkback input.

www.soundonsound.com / February 2025

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S OL ID S TAT E LO GI C 18

Perhaps more pertinent to everyday use, however, are the ability of the headphone amps to deliver levels up to +18dBu, and the fact that on both the input and the output side, the 18 is designed to operate at professional alignment levels where 0dBFS corresponds to +24dBu. This is an unusual, perhaps even unique feature on interfaces in this class, which are more normally aligned to +18dBu or less, and means that you’ll never run into problems trying to integrate studio outboard with the 18. And if you run into the opposite problem, whereby the interface threatens to melt your monitors or overload downstream equipment, it’s possible to switch the first two output pairs to a more conservative +9dBu alignment instead. All of the analogue outputs are also DC-coupled, meaning you can use them to transmit control voltages to analogue synths.

Like other recent SSL interfaces, the 18 is billed as a 32-bit device, but this just refers to the converters that are used, and it’s not intended to be ‘unclippable’ in the way that some field recording devices are.

The benefits of the high-spec converters are the same whether you choose to record at 24- or 32-bit.

All Rounder I mentioned deep integration as a common theme of recent SSL products, and the glue that binds them together is a software environment known as SSL 360. If you use any of SSL’s channel strip or bus compressor plug-ins, or third-party equivalents hosted in the SSL Link plug-ins, these will appear in the Plug-in Mixer page of SSL 360, and can be manipulated using controllers such as the UC1 reviewed in last month’s SOS. Connect a 12 or 18,

Do Talk Back There’s no built-in talkback mic on the SSL 18, but you don’t have to sacrifice one of the other inputs in order to use the feature, as there’s a dedicated combi XLR/jack socket on the rear panel. Unlike the other analogue inputs, talkback can only be configured in software, but a visit to its SSL 360 channel reveals a lot more going on than you might expect.

There is switchable phantom power and three gain options, respectively applying 30, 40 and 50 dB boost to mic-level inputs, or you can tell the 18 to expect a line-level input. The biggest bonus is a button labelled LMC, which activates SSL’s notoriously aggressive Listen Mic Compressor. As the name suggests, this was originally part of the listen-back system on SSL consoles, a ‘reverse talkback’ arrangement that allowed the engineer in the control room to hear what quietly spoken drummers were muttering in the live room. Its use on a Peter Gabriel session with Phil Collins and Hugh Padgham resulted in the invention of gated reverb, the compressor bringing up the ambience on the drums after each hit before a gate cut it off again.

The ability to route a line-level signal through the 18’s talkback channel means you can use the LMC as an insert processor on previously recorded material as well as on microphones.

And, whereas Padgham had to cajole the techs at London’s Townhouse Studios into rewiring their SSL to make it possible to record the listen mic, this one is delivered straight to its own DAW input as well as the 360 mixer. That is, as long

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as you select Talkback as 9/10 Input Source in SSL 360; the other options are the S/PDIF input and a Loopback input. Note that although the talkback channel has a ‘fader’ in the 360 mixer, it does not feed the main monitor bus, which is as well, since activating the LMC in those circumstances would easily provoke feedback. Instead, the fader is essentially a global trim that adjusts the level of all the talkback sends at once, but does not affect the level going to your DAW. With my dispassionate reviewer’s hat on, there’s an element of novelty about the inclusion of the LMC on the talkback channel. As anyone who’s ever recorded drums will know, the idea makes perfect sense for listening to a drummer trying to form a coherent sentence, but extreme compression is perhaps less often needed in the other direction. However, I’m not sure that SSL really expect anyone to use it for comms. What the 18’s talkback feature really offers is a bonus recording input with instant vibe. In a busy session, there isn’t always time to muck around with outboard compressors or guitar pedals in search of that smashed room-mic sound. The LMC delivers it on a plate, without taking up an input that could be used for something sensible, as well as making it available as an insert processor at mixdown.

The talkback input has its own channel within the mixer, with three switched mic gain settings, a line-level option and the all-important LMC.

and you’ll see another SSL 360 page available. This is the equivalent of the control panel utility provided with other interfaces, and naturally, it too presents a mixer-like interface.

The virtual mixer resembles a rather gloomy SSL console — sadly for middle-aged eyes, it’s not possible to brighten it up — and allows you to independently show and hide four types of channel: analogue inputs, digital inputs, playback returns and auxiliary masters.

The talkback input also has its own dedicated channel. There are three stereo auxiliary busses, routed respectively to the two headphone outputs and to line outputs 3+4. These are fed from sends that appear above the channel faders, meaning there’s no need to tab to alternate fader views in order to set up cue mixes. As already described, gain control is out of the purview of SSL 360, but buttons on the channel strips allow you to toggle 48V phantom power, the high-pass filter and 4K mode on the analogue inputs. Inputs 1 and 2 also offer the usual high-impedance mode for direct guitar recording. This mode is only switchable in software, but the rest can be adjusted from the front panel: each channel has its own selection button, whereupon global buttons toggle these settings for the selected channels.

Three further buttons on the front panel relate to the 18’s monitor control functionality, as do separate volume controls for the main monitor and headphone levels. The buttons can be reassigned within 360, but since there are only four options for the three buttons, this is of limited usefulness. The monitor control on offer is nevertheless relatively generous compared with some interfaces in this bracket, with cut, dim, mono and alternate speaker switching. If used, the latter toggles monitoring between outputs 1+2 and 3+4, with adjustable trim to compensate for sensitivity differences between two pairs of speakers. The down side is that you lose the dedicated auxiliary bus that can otherwise feed outputs 3+4, which cannot be reassigned to another output pair. In practice, this is unlikely to be a problem, since the aux is mainly useful during tracking and speaker switching during mixing.

Odd Ones Out In general, the SSL 360 mixer page for the 18 is slick, comprehensive and easy to use, but there are a couple of quirks.

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S OL ID S TAT E LO GI C 18

The 18 Mixer tab in SSL 360 is loosely modelled after a large-format console, which is no surprise given the company’s heritage.

The most obvious of these concerns the Playback Return channels. Your DAW can address a total of 28 outputs, the last 18 being the 16 ADAT and two S/PDIF outs, which are fed directly from DAW outputs and don’t appear in the mixer. The first 10 show up as ‘analog out’ in the DAW, and although they can be set up to address physical outputs directly, this isn’t the default or the usual mode for eight of them.

Instead, DAW outputs 1+2, 3+4, 7+8 and 9+10 feed the four Playback Return channels in the 360 mixer, which in turn can feed the main monitor outputs via the faders, and the three auxiliary busses via the sends. DAW outputs 5+6, however, are hardwired to the corresponding physical output pair on the 18, and never appear in the mixer.

It turns out that although the 18 has 10 physical outputs, its DSP resources are only sufficient to support eight mono or four stereo playback channels. The choice to omit 5+6 rather than 9+10 from the Playback Returns relates to a specific use case involving hardware inserts in Pro Tools, which insists on

a master compressor, EQ or other outboard chain between outputs 5+6 and inputs 5+6. But in this circumstance, a Playback Return channel for DAW outputs 5+6 would be wasted, and one of the headphone busses would be lost.

The gapped numbering thus allows Pro Tools users to retain the full functionality of four stereo Playback Returns within SSL 360 while also using one or two channels of hardware insert processing. I suppose SSL might have avoided this compromise either by adding more DSP resources or an extra pair of line inputs, but no doubt that would have raised the cost of the 18.

The routing is made a little clearer if you hit the 360 button labelled I/O Mode.

This completely bypasses all of the mixer

“Within the ‘eight mic preamps plus ADAT’ category, this is either the most professional semi-pro interface yet — or the most affordable professional interface yet.”

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using the same-numbered inputs and outputs for this functionality. Since the 18 does not have a dedicated line input pair numbered 9+10, it’s not possible to set up a Pro Tools hardware insert on outputs 9+10. With outputs 1+2 and potentially 3+4 reserved for monitoring, the logical choice is therefore to connect

functionality, passing physical inputs directly to DAW inputs and DAW returns directly to physical outputs. In this scheme — which you might use if you preferred to set up cue mixes within your DAW — headphone outputs A and B can freely be reassigned to mirror other physical outputs.

And talking of mirroring, the 360 Mixer also has its own ‘mirror routing’ feature.

Clicking at the bottom of any input channel allows you to route it directly to any physical output on the 18, as well as feeding the monitor and headphone busses as usual.

This has obvious applications for feeding an external cue mixing system but could also, for example, be used to set up simultaneous wet/dry recording: you might record the dry signal coming into input 5, whilst simultaneously mirroring it to output 8 and connecting a processor between that and an input, though you’d have to compensate after the fact for the small amount of additional latency in the wet path.

On the topic of latency, the 18 uses macOS’s built-in Core Audio USB driver, and the familiar Thesycon offering on Windows. Signals passing through its built-in mixer also incur a tiny amount of additional latency, though this is below 1ms for all sample rates.

Selecting the lowest buffer size on a Mac allows you to obtain a round-trip latency of around 7ms at base sample rates, which feels comfortable for me, and SSL say that the Windows driver will achieve 6ms or less. If you wanted to significantly improve upon this, you’d need to either track down a Thunderbolt

alternative or something with custom USB drivers, either of which is likely to be more expensive.

18 Wheels There is a break point in the audio interface market between what we might call semi-pro and professional devices. Both price-wise and in terms of its feature set, the SSL 18 sits at the top of the former category. This means it’s significantly more versatile than many other semi-pro products, with quite a few capabilities that simply aren’t available elsewhere. The flip side of this is that it’s a little more complex than many alternatives. SSL have done a good job of making the 360 mixer page clear and logical, and I particularly like that everything happens in the one view with no tabbed pages to plough through, but you do need to be familiar with traditional mixer concepts and layout to make sense of it. To someone with little experience of working with audio hardware, the super-simple approach pioneered by Focusrite with their control panel software might be less intimidating. Focusrite have also done an excellent job of implementing remote control over WiFi from a phone or tablet, something that’s currently missing from SSL’s offering.

On the other side of the semi-pro/ pro gulf, you’ll find features such as

dedicated sockets for line-level inputs, built-in speaker calibration, surround monitor control, comprehensive metering and huge channel counts through expansion over network protocols or by cascading multiple interfaces. If you need that sort of functionality, you’re looking at devices like the RME Fireface UFX+, Antelope Orion 32+, Universal Audio Apollo X Gen 2, Apogee Symphony Studio and the Focusrite Red or MOTU AVB ranges, all of which are considerably more expensive. Yet, although it sits on the semi-pro side of the divide, the 18 actually matches most of these for sound quality and technical specs, and it also plays with the big boys in terms of being able to handle +24dBu alignment. If you’re not going into Atmos production and you’re content with a hard limit of 26 audio inputs, there’s very little compromise involved in choosing the SSL 18 over something much more costly. Within the “eight mic preamps plus ADAT” category, this is either the most professional semi-pro interface yet — or the most affordable professional interface yet. Either way, SSL are still exceeding expectations. $ $1149 T SSL +1 (818) 643 7040 W www.solidstatelogic.com

If you prefer to set cue mixes up in your DAW mixer, engaging I/O Mode simply routes inputs and outputs directly to their counterparts in your software of choice.

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Roland P-6 Creative Sampler The Roland P-6 is a pocket-sized sampling powerhouse.

SIMON SHERBOURNE

I

t’s easy to criticise Roland for milking their classics in multiple form factors, but when it comes to the Aira Compact range I don’t hear anyone complaining. They’re fun, affordable and ridiculously portable.

So far we’ve seen a drum and bass line machine (T-8), mono and polysynths (S-1 and J-6) and a voice transformer (E-4). It made sense that some kind of sampler should be next and the P-6 takes its inspiration from the Roland/Boss SP range of workstations, in particular the 303/404. As with the others in the Aira Compact series, it’s designed to get quick results in its own way, rather than to simply miniaturise a vintage instrument.

Big Picture Like the 303/404, the P-6 parks samples in pad slots across a number of banks, in this case six pads and eight pages. True to the simplicity of the original SPs there’s no separate sample pool or library: the samples that are loaded in the pad slots are what you have to work with until they’re overwritten.

Samples are triggered from the pads, though the P-6 lacks velocity sensitivity.

It does have something even the latest

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404 doesn’t: dedicated step sequence buttons. This makes sense as the device is too tiny for finger drumming.

While there’s no project structure, there are 64 pattern slots which share the palette of 48 sounds in memory. And like the 404, effects play a major role. As well as reverb and delay send effects, there’s a suite of performance effects designed to be punched in and tweaked on the fly.

Like other samplers of this type, there’s a chromatic mode (called Keyboard mode) for playing a sample melodically. Incredibly, this is polyphonic, a major win over devices like the otherwise more sophisticated Digitakt. What’s more, one sample per pattern can be played using a granular engine. The P-6 pairs really well with a MIDI keyboard. Depending on the MIDI channel you can have all 48 sounds spread across the keyboard, play the focused pad chromatically, or play the granular sampler.

Small Details The Aira Compacts are pocket portable.

They’re less than 19cm across and the plastic shell makes them ultra-light. The tiny, low-profile knobs are just on the right side of usable. Most importantly, the Compacts

have an onboard lithium-ion battery which is charged via USB-C and can keep you untethered for three hours. Unlike meatier devices such as the 404 MkII and Ableton Move, any USB charger will do, not just a high-power adaptor.

Sensibly, the connections you’d use in the studio (USB-C port and TRS MIDI I/O) are on the back, while audio connections are on the front panel and are less pokey when the P-6 is on your lap. A single audio out serves for either stereo line output or headphones. You can also plug in a headset with a mic and sample in via the same port. Otherwise, there’s a more conventional stereo audio input for capturing from other devices. The USB-C also has class-compliant stereo audio capability in both directions. Next to the audio connections are analogue sync in and out, which alongside the MIDI Clock support via the TRS or USB connections makes this a useful little box in various scenarios.

Step Up The step sequencer provides an easy route to getting things happening on the P-6. Select a pad, tap some steps and you’re off. You can also record from the

pads in real time, either quantised or freeform. Patterns can be up to 64 steps long, spanning four pages. The mod cons we’ve come to expect from hardware sequencers are mostly present: you can set trigger probability or frequency and you can print parameter automation in real time or by holding a step. There are no variable track lengths or speeds though, just the parent pattern settings.

Pressing the Keyboard button switches you into chromatic play mode for the currently selected pad, with the step buttons doubling as a virtual keyboard.

Pads can be triggered or gated, and you have both a pitch envelope and a shared amp/filter envelope per voice. Sixteen notes of polyphony are available on the device, with a limit of eight notes per step within any sample’s sequencer lane.

Another feature that Roland have absorbed from current trends is sequence reload: the ability to skip back to the last saved snapshot of the pattern.

However, this is where what I think of as ‘the curse of Roland’ starts to rain on your party. As with so many other Roland grooveboxes and drum machines, simple operations like saving or duplicating a pattern require multiple steps and can only be done while playback is stopped. This is madness.

The other slight wrinkle is that because you have 48 samples spread over four pages and two layers, and patterns can draw upon any of these, it can be difficult to hunt down the sounds used in any given pattern to edit, mute, etc. Some self-organisation is probably advisable so you can work in kits of adjacent samples.

samples from your computer on to the device (but not the other way) but there are also numerous options for direct sampling.

Sampling happens straight to a pad, and only works if the pad is cleared first. (It would be nice to have an expert preference to allow direct overwriting). When you enter Sampling mode, the P-6 starts monitoring all available audio sources, with the neat trick of level metering across the step button lights. You can then hit Sampling again and the P-6 starts capturing. When done it goes through an offline process of normalising and writing the sample. Simple.

A few settings affect sampling, mercifully set from the knobs when in Sampling mode rather than a menu. You can choose a sample rate from 44.1 down to 11 kHz, mono or stereo. These settings really matter as sample memory is very limited. Maximum sample time is just under three seconds for stereo at 44.1. You can set sampling start mode to manual, threshold detection or

Roland P-6 $219 PROS • Polyphonic chromatic/chop playback.

• Granular synth engine.

• Multiple sampling/resampling sources.

• Great value.

CONS • Stop/start, convoluted pattern operations.

• Send effects interrupted by pattern changes.

SUMMARY A feature-packed and surprisingly capable sampling groovebox that fits in your pocket.

by a couple of issues. First, you can’t turn off resampling, so any external source is always mixed with whatever the P-6 is playing.

That is, unless you go into a submenu and turn the Pattern level to zero, then sample, then go in and turn it back up again. And secondly, after sampling, the P-6 downs tools and does the normalise/write routine.

The length limit also precludes the classic technique of sampling a passage of music then chopping. However, there is chop functionality. Samples can be chopped into an equally spaced number of slices within a pad, after which they appear as separate notes in Keyboard mode. More flexibly, you can sample directly into chops.

This doesn’t use the ‘lazy chop’ method of dropping marks in real time; rather, it lets you set a number of slices before sampling, then sample into each in turn.

Again the polyphony in Keyboard mode is a real power here, because as well as classic chops you could set up a whole drum kit within a single pad, with the compromise that you lose the ability to set individual sound parameters.

“It’s the most feature-packed and deep device in the range so far, with unexpected gems like the granular voice.”

Sampling You’ll want to get your own stuff on to the P-6: there are some nice basic sounds on there but also some long vocal samples that get pretty annoying when you trigger them while browsing through the pads. A sample utility provides a means to edit and send

sync’ed; and can define a sampling duration either as free or to a set number of beats.

Grabbing a set of one-shots from another source or quickly sampling your environment with the mic works great on the P-6. Other SP-like workflows have limitations. Resampling is a key part of beatmaking on the SPs, bouncing down loops and sections onto other pads.

Resampling is always on on the P-6: the mix out will always be captured if you sample during playback, and if you use sync trigger and a set time everything is taken care of for you. Easy, but the time limit is an issue: at full quality you can just barely fit in one bar at 90 bpm.

I was also hopeful that the sync and fixed time features would allow for looper-style functionality, but it’s scuppered

Sound Design Basic sound settings are found on the right-hand side of the panel (the left side is

The P-6’s back panel is home to the power switch, a USB-C port and 3.5mm MIDI I/O sockets.

www.soundonsound.com / February 2025

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The Sample Tool is a handy editor and transfer utility, although exporting samples from the P-6 has to be done bank-by-bank in disk mode.

dedicated to effects), including Pitch and Level, and Start/End times. You can also toggle mono/poly playback, gate/one-shot modes, and looping or reverse functions.

A dedicated Lofi button adds sample-rate reduction, with intensity set somewhere in a menu.

Further settings (a rich set) are accessed via four Sample Edit menus.

Within each menu, key parameters are assigned to the knobs for easy access, while others are reached and controlled from the main encoder. Within these menus you’ll find your envelopes, filter types and settings, effects sends and mix controls.

It’s fantastic to have this much control and sound-shaping, but it’s strongly hindered by the display. I think I could live with this amount of menu diving — it’s only one level compared to the almost unusable MC-101 — it’s just that many menus and setting names are undecipherable when rendered on a four-character, seven-segment readout.

Granular The granular sampler mode is such an unexpected bonus that I wonder if it was a hackathon project where some enthusiastic developers thought, ‘I bet we could make it do this,’ and it ended up shipping. One sound per pattern can be assigned to the granular engine, and the granular settings get saved with the pattern, unlike any other sound settings.

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The Granular button itself triggers the current grain sample, but it works best in Keyboard mode. You can control the size, shape and spread of grains, and the density (number) of grains up to eight. There are even more obscure hieroglyphically labelled parameters for reverse probability and jitter. Rather than complicated modulation of the sample playback position, you can simply set a position and scanning speed (which can be backwards). The step buttons light up to indicate the ‘playhead’ position. Brilliant!

Effects & Performance Like the SP samplers, effects are front and centre on the P-6. In addition to the delay and reverb send effects that can be fed from individual pads, the front panel features six buttons for engaging the large range of master effects. Looper samples and loops a chunk of the mix output and lets you then manipulate it, while Scatter is the ubiquitous Roland stutter and reverse pattern generator.

Both of these effects ought to be great for creating fills, but are a bit hit-and-miss.

Pitch, Delay and Filter are more useful.

The MFX button holds all the other onboard processors, which are mostly great, such as the isolator filter, resonator, tape stop and vinyl effects, and regulars like chorus and reverb.

Like the SP-404 there’s a Bus system, so you can route pads either through the

effects or to one of two clean busses, and adjust submix levels (which you likely won’t as it’s too hard to find). An easy shortcut allows you to quickly toggle pads between effects and bus A, which serves as a bypass. I’m not sure if it’s a bug that this cancels assignments to bus B. When performing with the P-6, the other main thing you’ll probably do is flip patterns, which is done from the step keys in a dedicated mode. This works fine although, as on all Roland boxes, pattern changes are always queued to the end of the pattern, and therefore limit spontaneous fills. The other issue is that pattern changes reset the effects sends, so reverb and delay tails just stop dead. The one saving grace is that Roland always include a Step Loop mode, which I love for momentarily freezing and looping playback on one or more steps.

Conclusion The P-6 slots neatly into the Aira Compact range, making sampling and sample-based beatmaking fast and fun.

There’s a lot of familiar SP heritage in here, but it’s taken in a new direction with the step-sequencing-focused workflow. It’s the most feature-packed and deep device in the range so far, with unexpected gems like the granular voice. Polyphonic sample playback per channel made me start to see the P-6 as contending with way more expensive workstations. In other areas though, its scope is somewhat smaller: notably the short sampling time limits the possibilities for resampling, looping or classic SP chop workflows unless you’re OK with low sample rates, and the display is just not an adequate match for the feature set. Let’s not forget how inexpensive it is, though, and how many use cases it has. It’s an ideal travel companion, it’s a very capable sample-based drum machine, it works great as a sound module with a MIDI keyboard, and you could even justify picking one up as a multi-effects module for a desktop synth rig. $ $219.99 W www.roland.com

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Grace Design m303 Active DI Box

Did you think all DI boxes were created equal? If so, you may need to think again...

HUGH ROBJOHNS

B

est known for their superb mic preamps and monitor controllers, Grace Design also make an impressive range of instrument preamps, such as the Alix, Bix and Felix, the last of which was reviewed in SOS October 2015.

These provide varying degrees of EQ complexity, adjustable input impedance, effects loops and more, but all of them feature an active DI output. Although not everyone needs the sophistication of these ‘ix’ models, a great-sounding, reliable active DI box is a common requirement for live and recording work, so it’s not surprising that Grace’s engineers have adapted their proven DI circuitry into a dedicated standalone box.

Overview Called the m303, this single-channel, phantom-powered DI box is the size and shape you’d expect, measuring 81 x 44 x 134mm (WHD) and weighing just under 0.5kg. The strong case is manufactured from extruded aluminium, finished in

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a gorgeous anodised cobalt blue, with silver lettering that’s easily readable in low-light conditions to identify the socketry and switches. The base is covered with a rubber pad, which grips very well, helping the m303 stay where it’s placed.

A central square is cut out of that rubber pad to reveal a printed label carrying a unique serial number for each m303, so it won’t get damaged or worn off. The whole product is built in America and comes with a generous five-year warranty.

The review unit arrived in a recyclable cardboard box with a ‘Korrvu’ suspension that acts as a very effective shock-absorber. There was no printed user manual and, to be fair, it doesn’t really need one; there’s nothing unusual in the m303’s use or operation. Nevertheless, a detailed 13-page PDF manual is available, and this describes the switches, connections, and how to use the unit for novices. It also includes an unusually comprehensive set of genuinely impressive technical specifications.

High-quality components are used throughout, with gold contacts in the

switches, and comprehensive RF interference and ESD protection. All I/O connectors and switches are mounted on recessed aluminium end plates, where they’re well protected by the outer case, so I have no reason to doubt the m303 will survive decades of stage use (and abuse).

At the left (input) end of the box is a pair of quarter-inch sockets wired in parallel as unbalanced input and thru connections.

To accommodate high-level line sources, a recessed toggle switch engages an input pad. A green LED indicates when the unit is powered, while a red LED illuminates roughly 6dB below clipping.

In normal use (without the pad), the input impedance is specified as 1MΩ, which is pretty much the standard for active DI boxes and will keep all guitars and basses with passive pickups happy. Engaging the input pad reduces the input impedance, of course, in this case to a still pretty high 115kΩ. While this would affect the tone of electric guitars significantly, the m303 has such good headroom that the pad would never be needed for such sources — it will only be required when working with very hot line-level equipment (eg. DJ decks) and possibly some electronic music sources such as modular synths. For all of those, 115kΩ is more than high enough.

I mentioned headroom, and in all active DI boxes the maximum signal level that can be handled directly (without the pad engaged) is determined by the power rails feeding the active input circuitry.

To that end, the m303’s audio circuitry runs on symmetrical ±5.5V power rails, accommodating a peak-to-peak signal level of 11V (equivalent to +14dBu). This is more than most active DI boxes — the Radial J48

Grace Design m303 $299 PROS • Phantom powered.

• Massive headroom.

• Completely transparent, sonically.

• Lundahl audio transformer.

• Total galvanic isolation between input and output.

• Easy to use and extremely robust.

• Five-year Warranty.

CONS • Some might want a direct speaker connection mode.

SUMMARY The m303 phantom-powered active DI box is an impressive design in every respect, and sets a quality and practicality benchmark to which others must aspire.

A photo of the circuit board with the Lundahl audio transformer at the top and the internal power transformer at the bottom. The audio circuitry is the op-amp to the left of the Lundahl transformer. The chip between the sockets drives the level LED. All the rest is power-related!

clips at +10dBu, for example — so it has a very generous headroom margin, and is extremely unlikely to be overloaded in normal use.

Unusually, the pad switch attenuates the input signal by 13dB. It might seem an odd choice, but it means the maximum clean input level is a whopping +27dBu — more than enough for the loudest line-level equipment, even if running with its meters fully pegged (most pro equipment clips at around +24dBu). It’s not enough to handle directly the output from a beefy power amp or backline loudspeaker, though, and the user manual specifically warns you not to try that!

At the right-hand (output) end, there’s a male XLR for the balanced (microphone-level) output, and another recessed toggle switch to isolate the output ground from the input ground (to break any ground loop between the source and destination equipment).

Like many top-quality active DI boxes, the m303 can only be powered by standard 48V phantom power, with no provision for internal batteries. However, it consumes a miserly 4mA of current from the phantom supply, which is less than many mics and lower than most active DI boxes, so it won’t stress your mixer channels, interfaces, or mic preamps.

Phantom Power & Galvanic Isolation Although using phantom power to energise a DI box is a very convenient and reliable option — avoiding the issues associated with failing, exhausted, or missing batteries — it poses a significant

engineering challenge: as the output XLR ground connection is a necessary part of the phantom power supply circuit, the usual ground-lift solution of simply disconnecting pin 1 of the output XLR isn’t viable (it would stop the DI box from working). Manufacturers have evolved various workarounds to this problem, but Grace Design’s sophisticated and elegant approach ensures total galvanic isolation (meaning no direct electrical connection at all) between the input and output connections, when required — which is a good thing!

All passive DI boxes (and most active ones) use an audio transformer to provide galvanic isolation between the input and output grounds — the audio signal is passed between the primary and secondary windings inside the transformer via a varying magnetic field. The electrical isolation this provides is not only helpful in preventing ground-loop noise problems, it also provides an important level of electrical safety. For example, should a serious electrical mains fault occur on the PA system the transformer prevents the guitarist from receiving an unwelcome electric shock. So, the m303 uses a very high-quality, fully-shielded, custom-wound Lundahl

LL1975 audio transformer (with a high-nickel core content to minimise LF distortion), which provides the required electrical isolation between input and output.

However, this still doesn’t solve the phantom-power ground connection problem, because the audio electronics have a ground reference via the internal power supply. With phantom power, there will still be a ground path connection, so the internal power supply must be isolated in some way too. Like several other manufacturers, Grace achieve that by using the external phantom power only to drive an internal AC oscillator, in this case running at around 68kHz. The output voltage from that oscillator feeds the primary winding of a miniature high-frequency transformer, whose electrically isolated secondary winding passes that AC voltage into a bridge rectifier and smoothing components, to generate an isolated DC supply providing ±5.5V to power the input audio buffer circuitry. In this ingenious way, both the output XLR and phantom-power circuitry grounds are completely isolated electrically (via the two transformers) from the input circuitry ground. The ground-lift switch can couple them together, or leave them isolated, as required. The whole scheme is very neat and works brilliantly.

Most of the circuitry inside the m303 is involved in this internal powering system, and the audio circuitry components really only comprise the sockets, Lundahl transformer, input pad, and a dual FET op-amp, that serves as the high-impedance input buffer. For this, Grace chose a Texas Instruments OP1652, which has impressively low noise and distortion specs, and the ability to handle signal all the way up to the power-rail voltage (a choice that maximises headroom).

Technical Performance With the pad disengaged, which will be the normal way of working for most,

An illustration of the block diagram concept, highlighting the separation of input and output sections via transformers.

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output showed that all signal, power and distortion harmonics were comfortably below -120dBu.

Best In Class?

An Audio Precision plot of input level versus THD+N.

Another Audio Precision plot, showing the ruler-flat frequency response across a range of signal levels.

the signal level at the balanced XLR output is around 6dB lower than the unbalanced input level, though the exact attenuation depends to some degree on the destination load impedance. Thus, a typical -20dBu instrument level going into the DI box will produce a fairly hot mic level of around -26dBu at the output. With the pad engaged, the total attenuation through the unit rises to a little more than 19dB. In both cases, the maximum balanced output level is around +8dBu (dependent slightly on the destination impedance, again), from a source impedance of 330Ω.

Hooking the m303 up to my Audio Precision analyser, I found the Total Harmonic Distortion (THD) at 1kHz measured 0.0005% (for an input level of -20dBu), rising to 0.007% with an input level of +12dBu (just before clipping).

So this DI box is perfectly transparent at all practical signal levels. Intentionally overdriving the input required a level of +15dBu and didn’t sound nice — there is no graceful saturation before clipping!

All transformers introduce greater distortion at low frequencies, but with the

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same input level at 50Hz, the THD figure reached only around 0.007% (and at 25Hz about 0.03%). So, again, that’s completely unnoticeable and inaudible. Intermodulation tests were similarly impressive, with the worst case being 0.001%.

“This DI box is perfectly transparent at all practical signal levels.”

Measuring the frequency response across a range of input levels, with and without the pad, always delivered consistently ruler-flat traces between 22Hz and 40kHz (-1dB limits), or 15Hz to 70kHz (-3dB limits). There’s a fixed 12dB/octave high-pass filter which rolls off below 20Hz, and the high corner frequency is affected slightly by the source impedance.

The m303’s signal-to-noise ratio measured over -112dBu (flat) or -117dBu A-weighted (20Hz to 20kHz). While not quite as good as the published specifications, that’s still very good indeed, and an FFT spectrum of the

The m303 is a very solidly built, skilfully engineered and, most importantly, great-sounding active DI box. There’s nothing confusing about its use or settings.

Plug the output to a desk or interface mic channel, set with low gain. Turn on the phantom power, and check the green power LED. If there’s ground-loop noise, activate the ground isolation switch, and if the source is really loud and the peak light flashes, activate the pad. It’s that simple and straightforward.

I’ve long been a fan of Grace Design’s mic preamps and monitor controllers, because they sound so transparent and are easy to use. The m303 never disappoints in these departments either. It’s extremely quiet, even with low-level sources, the headroom margin is epic, and together these afford an enormous dynamic range capability. The audio path is amazingly transparent and natural-sounding, too, with a crystal clear and airy top end nicely balanced by a powerful, deep and perfectly clean low end.

I’ve owned and used a lot of DI boxes over the years, and I can genuinely say that the m303 is the best I’ve heard.

A few other designs have matched its headroom and transparency, but the big deal for me is that the m303 ticks almost all the boxes, where everything else seems to miss at least one. Others don’t also offered the total galvanic isolation on offer here. Others can’t match the 1MΩ input impedance, or can’t provide ground lifting when phantom powered... The only box left unticked by the m303 is its ability to cope with direct amplifier outputs. That’s a technique I use rarely and would in any case prefer to use a passive box for, so that’s an irrelevance.

Naturally, a fully isolated active DI box from Grace Design is never going to be inexpensive enough to give away with breakfast cereal. But the m303 is priced comparably to similar high-quality DI box offerings from Radial and Little Labs, for example, and it’s undoubtedly an investment that will last a lifetime and never disappoint sonically. $ $299 T Grace Design +1 303 823 8100 E info@gracedesign.com W https://gracedesign.com

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ON TE ST

Korg Keystage 61 With polyphonic aftertouch, MPE and MIDI 2.0, Korg’s Keystage has a lot to offer.

Korg Keystage 61 $699 PROS • It’s stylish, light and manageable.

• Polyphonic aftertouch and MPE are always welcome.

• It’s also a stereo USB audio interface.

• Having your target software map itself automatically to the controller is a treat.

• The bundled software is a bonus.

• Its value will improve as more devices and software adopt MIDI 2.0.

CONS • There are better endowed alternatives.

• The keyboard has but a single zone.

• The keybed won’t be to everyone’s taste.

• The chassis isn’t rigid.

• The wheels scrape slightly against the top case.

• It’s difficult to use in low lighting conditions.

• No PSU is supplied.

SUMMARY The Keystage is innovative but flawed.

Some players will love that it’s stylish, light and manageable, and that it offers polyphonic aftertouch and MPE, while others will be happy to sacrifice these attributes for something chunkier with a more robust action and a healthier selection of controls.

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GORDON REID

T

he genus ‘MIDI controller’ spans a wide range of species, ranging from large, heavily endowed 88-note keyboards offering sophisticated control surfaces to tiny devices that you can slip into a pocket. Between these extremes, there are all manner of products including numerous 49- and 61-note keyboards looking for their niche, and it’s into this crowded market that Korg have launched the Keystage 49 and Keystage 61, the latter of which is the subject of this review.

Let’s Be Hard Designed as both a MIDI controller and a USB audio interface, it has a smart finish, and its wooden end cheeks do much to give it an air of quality. But it’s not chunky, nor does it feel particularly robust, largely because it’s based upon a plastic chassis. Nonetheless, the modern trend is for light and slim so, in this regard, it scores well.

Unlike many MIDI controllers, it’s devoid of programmable buttons, pads and faders. This means that its sparse top panel is dominated by just nine

Controller Keyboard

knobs. By default, the Value encoder sends Program Change messages and the eight pots send MIDI CCs, but many additional functions are available in various modes, and you can see the parameters assigned to the knobs as well as their values on their associated screens. While these are bright and crisp, they’re also tiny, so you may find yourself reaching for your reading glasses to use them.

The Polytouch keybed is unweighted and springy. Back in the 1970s, we called this ‘synth action’, and we were grateful when semi-weighted and piano-action keyboards started to appear on synthesizers. It offers four degrees of expression: velocity, aftertouch, polyphonic aftertouch and MPE, but I worry when I read in the manual that: “The aftertouch sensors on the keyboard may wear out if you press into the keyboard too hard. For this reason, avoid pressing into the keyboard harder than necessary.” It feels as if Korg are getting their defence in early.

On a more positive note, the pitch-bend and modulation wheels and the octave/ transpose buttons are in their correct

positions to the left of the keyboard.

What’s more, you can connect any combination of sustain and expression pedals to the (nominally) damper and expression inputs, which adds flexibility.

Unfortunately, Korg have followed the Douglas Adams school of design, endowing the Keystage with black legends on a black background, and with little black lights that light up black on black buttons to let you know when you’ve done something. OK, it’s not quite as bad as that, but it’s not far off. For example, the rear-panel legends are embossed (so they are, literally, black on a black background), and the top-panel legends are printed in a grey that’s almost impossible to read in adverse conditions. Furthermore, while some of the buttons have backlights, these only indicate when a function is on, so you have two rows of closely

spaced buttons offering little to help you to see what’s happening.

At the rear of the Keystage, you’ll find three slots that allow you to attach an expansion plate. There are two possible positions (one in the centre and the other to the right), and two angles (flat or inclined slightly upward). The plate is a little over four inches deep but, if you mount it to the right, the total flat area (including the otherwise unused area of the top panel) is around eight inches deep. I placed a 15-inch MacBook Pro in this position and, to my surprise, everything remained in place. Nonetheless, I wouldn’t want to use anything heavier than a small tablet without a liberal application of Velcro.

Let’s Be Soft The philosophy of MIDI Controllers has changed over the past few decades.

Once upon a time, they offered up to eight zones, allowing players to control multiple audio sources simultaneously from a single keyboard. Not so here; although you can allocate different MIDI channels to various controls, the keyboard is a single zone.

Nonetheless, the Keystage has three extra facilities that help to keep things interesting. The first is an arpeggiator that’s driven by an internal clock if no external clock is applied. While not as esoteric as some, it nevertheless offers 20 patterns and 16 parameters that affect its output, including a random probability that any given step will (or won’t) play a note. If you set this up carefully, you can even create drum patterns with a degree of variation between arpeggio cycles. (Well... you can, but I’ve no idea why you would.) Happily, you can assign your choice of eight arpeggio parameters to knobs 1 to 8, which makes it easier to ‘play’ the arpeggiator in real time because you don’t have to step through pages to find the one you want.

The second is a chord mode. You can create a chord of up to eight notes under each key in the octave, giving you a maximum of 12 chords at any given moment. (If you play the same key in a different octave, you obtain the same chord but transposed accordingly.)

A strum function does as you would expect, and you can determine its time and direction. In addition, a velocity mode allows you to switch between up and down strums. A chord setup is called a Chord Set, and there are 32 factory sets as well as 32 memories into which you can save your own.

The third is MIDI 2.0, which expands the ways in which MIDI devices talk to

The Rear Panel The rear panel starts with a 9V DC power input and an on/off button.

To the left of this is a USB-B socket for connecting to your computer.

This is capable of receiving sufficient power to drive the Keystage, so you may never need an external PSU if you’re connected to a suitable host. However, when used with hardware synths and expanders, as well as tablets or phones without USB, you’ll need a 9V DC PSU or a suitable 5V DC USB power adaptor, neither of which is supplied. Next come the unbalanced stereo audio outputs and their associated headphone

output. Don’t be confused by these; the Keystage doesn’t have a secret sound generator squirrelled away inside, the outputs are just for audio from your computer. Unfortunately, there’s a single volume control for both the line-level and headphone outputs and the latter of these should, as always, be on the front panel not the rear. To the left of these you’ll find two self-configuring pedal inputs, notionally for use with a damper (sustain) pedal and an expression pedal. Next come 5-pin sockets for MIDI in and out (but not thru), and the final hole is for a Kensington security cable.

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each other. This is bidirectional so, as well as being controlled by the Keystage, a soft synth or DAW can say to it, “these are my most important parameters, and I suggest that you assign them to your knobs”. The Keystage comes bundled with three items of software that support MIDI 2.0: Ableton Live 11 Lite, Wavestate Native LE, and the Korg Gadget Producer Bundle. So, for example, if you connect it to something running Gadget, eight parameters are automatically mapped from the selected synth or drum machine to the Keystage’s knobs. Should you then switch between gadgets, it reconfigures itself to support the new one. I was quite sceptical about the value of this until I tried it, whereupon I was an immediate convert. To be fair, there won’t be much value in MIDI 2.0 until more products support it, but it’s backwards-compatible, so devices offering it should still work correctly when used with older MIDI equipment.

The Korg Kontrol Editor

In Use Before playing the Keystage, I checked the system version and found that it was obsolete. So I visited the URL shown in the manual, which turned out to be the Korg PA1000 update page. This contains a message not to try to install the upgrade on anything other than a PA1000, so I hunted for the correct page, found it, and downloaded and installed the Keystage update. The URL for the Korg Kontrol software (see box) was also wrong, so let’s hope that someone at Korg has fixed these by the time that you read this!

Next, I launched a soft synth that responds to polyphonic aftertouch and MPE and created a dual-layer Program comprising an e-piano patch with velocity directed to the amplitude and filter cutoff frequency, and a Vangelis-esque brass patch with poly-aftertouch directed to the same.

At this point, I was delighted to find that the aftertouch threshold is an adjustable parameter. This determines the amount of pressure that’s needed to activate aftertouch, which is a vital factor for anyone who likes to use this mode of expression. I was almost ready to play but, before doing so, I wanted to save

Korg’s Kontrol software allows you to configure the Keystage, and offers several parameters that are not available on the hardware itself. You can determine the MIDI channels used by various functions, the octave and transposition, the knob and wheel ranges, the behaviour of the encoder and transport controls, and more. There are also two parameters relating to MIDI 2.0, including whether Property Exchange is switched on or off. Hopefully, these point the way forward toward a more complete adoption of MIDI 2.0 in the future.

my settings, including my choice of velocity and aftertouch curves, pedal assignments and so on. One does so by saving a Scene that contains the user page, arpeggiator, chord and some of the keyboard settings. But when did the universally understood concept of ‘saving’ something turn into ‘registering’

It was now time to play. Although the keyboard is too light and springy for my taste, I was able to play the e-piano expressively, and the brass patch responded smoothly to aftertouch.

However, I must admit that I found the keybed to be a little noisy and the knobs to be a bit wobbly. But that wasn’t what worried me most. For the duration of the review, the Keystage was mounted on an X-stand with the supports about five inches from each end of the unit.

When I played a big, fat chord with both hands near the centre of the keyboard and then leaned into notes to use poly-aftertouch, I found that the whole unit flexed downward a little. It was now obvious why Korg have placed rubber feet in the centre as well as at both ends of the Keystage; it’s really designed for use on a desktop, and I suspect that its life on an X-stand will not be as long nor

“If you want a stylish MIDI controller for the studio and don’t require it to be festooned with faders, buttons and pads, the Keystage 61 could fit the bill.”

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it? The manual frequently refers to registering things, which is more than a little confusing. Anyway... there are 16 Scene memories, and you’re provided with factory presets in locations 1 to 6 that match Korg’s Wavestate Native, Opsix Native, Modwave Native, Microkorg, Triton and Polysix soft synths. Happily, you can edit and, umm... register to all 16, so you’re not saddled with a bunch of useless setups if you don’t need them.

as happy. I was also concerned with the pitch-bend and modulation wheels, which scraped very slightly against the top panel, and I found the return spring on the pitch-bend wheel to be too strong for my taste. Oh yes... and neither wheel is labelled, which is unhelpful.

Nevertheless, the Keystage has things going for it, and we shouldn’t underestimate the value of polyphonic aftertouch, MPE and MIDI 2.0, but it’s not hard to look elsewhere and find MIDI controllers festooned with faders, buttons, pads and larger screens. Do these matter? Well, yes, they can.

Suppose that you’re playing a Hammond soft synth and you want to ride the drawbars while doing so. Can you do this on something with a bank of assignable faders? Yes, you can. Can you do it on a Keystage?

No, you can’t, or I wouldn’t have mentioned it.

Similarly, you may find it more comfortable tapping in drum patterns using velocity-sensitive pads rather than keys. Of course, your needs may not include playing Hammonds or programming rhythms, so you’ll have to decide whether Korg’s choice of facilities is appropriate for you.

Now, what of that USB audio interface? The manual and Korg’s website quote this as “Audio format: 44.1kHz, 16-bit”, but I don’t understand what this means. Perhaps I’m being a trifle thick here, but it’s not an input converting analogue audio into a specified digital audio format, and I couldn’t imagine that it would only work when receiving 44.1kHz, 16-bit digital audio. So I copied a track from a CD and created copies at sample rates ranging from 96kHz to 8kHz and word lengths ranging from 24-bit to 8-bit. As I had hoped, it worked with all of them. While I doubt that the Keystage resides in the higher echelons of digital/analogue conversion, I found it entirely serviceable when monitoring my DAW as I recorded new parts.

Finally, I should note that the Keystage offers a degree of integration with many DAWs. My weapon of choice is Digital Performer, and it proved simple to connect the Keystage to this and then use the Control Surface Settings menu to map 10 functions from the DAW to the controller: play, stop, record, loop, metronome on/off, next track, previous track, rewind, fast-forward, and undo. Integration with Pro Tools, Cubase, Logic, Cakewalk, Ableton Live, Korg Gadget, GarageBand, Studio One and FL Studio is also provided, so there’s a good chance that your needs are covered.

Conclusions If you want a stylish MIDI controller for the studio and don’t require it to be festooned with faders, buttons and pads, the Keystage 61 could fit the bill.

There’s much to like about it, but the review unit also revealed a number of shortcomings, some of which could (and should) have been corrected before release. Many of Korg’s keyboards are fab, but I feel that this one is destined to be Marmite. You’re either going $ $699 to love it or... you’re not.

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ON TE ST

YA M A H A F G DP-5 0

Yamaha FGDP-50 Yamaha’s FGDP-50 drum pad is designed explicitly for the digital percussionist.

MARK GORDON

E

veryone likes to tap along to a song. Whether you’re a drummer or not, you’ll almost certainly find yourself involuntarily finger drumming to a tune on the radio or in the studio. If you fancy harnessing the true power of those digits though, a unique combination of electronic drumming and finger drumming has emerged in the shape of the Yamaha FGDP-50 Finger Drum Pad.

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Finger Drum Pad

The FGDP-50 offers the functionality of a MIDI pad controller with the sound-generating capabilities of a drum machine, in a compact package that includes a built-in speaker and rechargeable battery, enabling it to be used as a standalone instrument.

Featuring 50 preset kits and over 1500 sounds, as well as the ability to import and export WAV files and samples, the FGDP-50 can also integrate with your DAW via USB, making it something of a power-user tool for finger drummers!

Index Finger The first thing you notice about the FGDP-50 is its unusual pad layout. The

familiar design of 16 square pads in a 4x4 grid pattern seen on most MIDI controllers has been replaced by 18 pads of differing sizes, ergonomically distributed specifically for playing beats with the fingers. If you place your hands onto the FGDP-50, your fingers naturally fall into a comfortable playing position, with your thumb on the large kick drum pad, index finger on the snare pad and middle finger on the hi-hat pad, with other sounds such as open hi-hat, cymbals and toms distributed around the remaining pads in what feel like natural locations. The size of the pads and the symmetrical layout (where several sounds are duplicated on the left and right sides) also make playing with two hands very comfortable. I found the layout worked perfectly, but you can potentially assign any sound to any pad if the default layout doesn’t suit your playing style.

Eight RGB-illuminated pads are located across the top of the FGDP-50, which light up in a wide range of vivid and, as far as I can tell, random colours and perform different functions depending on the mode of the unit. In standard Kit mode, these pads are assigned to play additional sounds, such as cowbells, claps and, of course, the obligatory vocal samples.

If you use the click features, the RGB pads will flash in time from left to right, to provide a visual metronome.

A small LCD sits above dedicated buttons for Menu, Kit, Audio, Session, Registration and Click settings, with navigation arrows and +/- buttons to the right. Setting the FGDP-50 apart and putting it squarely in the instrument category is the small but surprisingly powerful 2.5 Watt built-in speaker. Along with battery operation, this allows the FGDP-50 to be used as a standalone device with no requirement for power supplies or additional amplification.

The connections at the rear of the unit are minimal: just two USB sockets and two 3.5mm jack sockets. The first USB socket is a micro-USB connection used to charge the battery and to transmit audio and MIDI data to and from a computer.

With USB-C being the connector of choice on the majority of modern devices, and offering greater power, more charging capacity and faster data transfer speeds than micro-USB, I was surprised at the choice, although (apart from charging speed) I wouldn’t expect there to be any performance difference between micro-USB and USB-C. The second USB socket accepts a standard flash drive that can be used to record audio from the FGDP-50, which is saved as a WAV file. You can also

import sounds, samples or play back audio files, again in WAV format, directly from a connected USB drive — which we’ll come to later.

If you’re more comfortable with analogue connectivity, Yamaha have provided a 3.5mm aux in jack that you can use to bring in audio from an external source to play along to. A second 3.5mm jack socket provides a stereo output you can connect to headphones or external amplification.

Trigger Finger The FGDP-50 includes 50 preset kits, spanning everything from acoustic maple, birch and oak kits to EDM, big beat and metal offerings. Percussion is also well catered for, with African, Brazilian and Arabic collections alongside orchestral and classic 808- and 909-style drum machines. A further 50 user memory locations are provided to save your own kit creations made from the 1500 onboard samples.

The pads are extremely sensitive, allowing a great deal of expression when playing with fingers. They also feature aftertouch, which is used to great effect on several preset kits, enabling crash cymbals to be choked, effects to be added and even pitch-bend to be applied to drums for authentic African talking drum and Indian tabla sounds.

A feature applied to the sounds on the RGB pads in the preset kits, but available on any pad, is the option to create repeats at set intervals. A voice assigned to a pad can be set to repeat in time with the current metronome tempo, in anything from one beat per bar to 32nd-note resolution, including triplets.

The repeats then play for as long as you

Yamaha FGDP-50 $259 PROS • Unique pad layout.

• Great sounds.

• Internal speaker and battery for standalone use.

• Impressive feature set.

CONS • Only has mini-jack and USB connections.

• Small LCD.

SUMMARY In addition to being a great excuse for finger-related puns, the FGDP-50 is great fun to play! Moreover, an incredible feature set elevates it to a serious tool that is as much at home in the studio as it is on your lap.

press the pad, as well as responding to any aftertouch assigned. This allows a single pad to play a simple shaker or hi-hat pattern, for example, or lets you create a range of EDM-style snare fills that vary in pitch or tone depending on how hard you press the pad.

Finger On The Pulse The user presets give you almost everything you might want for playing along with various genres and styles of music, but should you want to tweak a sound or two, the editing power available within in the FGDP-50 is quite extraordinary. Within Kit mode, a press of the right arrow button steps you through the Kit Edit, Pad Edit and Voice Edit screens. Here you can assign any of the 1500 samples to a pad, as well as adjusting individual volume, pan, tuning, decay, cutoff and resonance.

Three effects are available to each kit;

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Round the back we find just a pair of USB sockets, an aux input and headphone/line output, both on 3.5mm sockets.

Reverb, Chorus, (which includes a host of modulation effects) and Variation Type (featuring delays, distortions, filters and compressors). Each of these effects can be applied to any pad in varying degrees, using individual send levels, enabling you to tailor any kit to your exact requirements.

Register Interest In what I believe is a feature borrowed from Yamaha’s PSR range of keyboards, the FGDP-50 includes something called Registrations. This enables you to store four kit variations that can be instantly recalled by selecting one of four dedicated registration buttons: I, II, III and IV. This is ideal for live performance, allowing different kits to be stored together and quickly selected.

Any edits made to a kit, including tempo, trigger settings and effects, are saved in the Registration. It’s thus easy to add more reverb to a snare in a chorus, or change kit completely in

a middle eight, for example, although you must be sure to save edited kits to a user preset slot, or they will be lost once the unit is powered off.

I found this a useful, if somewhat confusing, feature. Having initially thought there were only four Registrations (which seemed more than enough), I then discovered there are 50 banks, which can each store four Registrations, giving a potential 200 kit variations!

Give It Some Stick While the FGDP-50 has the look of a drum machine, it doesn’t offer the ability to record, edit and play back parts in the same way as a conventional drum machine. What it does allow you to do, however, is record your performances as WAV files saved to a connected USB stick drive.

Pressing the Audio button and then the Record button puts the FGDP-50 into Record Standby, at which point you can start the metronome if you want

to play to a click. A second press of the Record button starts the recording process and displays a counter on the LCD, showing elapsed time. As soon as you press Stop, the file is saved to the USB drive and can be instantly auditioned via the Play button.

The audio record function not only records your performance, it can also record audio from the USB host input and aux in jack, as well as audio playback from the Session Creator function (see ‘Finger Buffet’ box). Using this method, you can record complete performances combining backing tracks and drums, or play along to backing tracks and choose to only record the drum performance.

The files on the USB drive are saved in 16-bit/44.1kHz WAV format and can be imported into your DAW or shared with other musicians.

In addition to saving recorded audio, you can also import samples into the FGDP-50, using the same USB port.

Again, audio needs to be in 16-bit/44.1kHz

Finger Buffet The Session Creator function offers the option to play along to 23 pre-recorded backing tracks stored in the FGDP-50 and features a range of styles from synth-pop and funk to metal, trance and jazz. Each ‘song’ is split into six sections: an intro section, main sections A-D (four variations), and an ending section. In Session Mode, instigated by pressing the Sess button, the RGB buttons are used to select each section and start/stop the playback, as well as muting and unmuting the pre-recorded drums on each song.

Once a song is selected, you can use the RGB buttons to choose the section of the song you wish to start with. When you press the RGB Play button, the currently selected song section will begin playback and the Play button will flash at the song tempo. During playback, the LCD shows the number of bars in the section, as well as which bar is currently playing, which is a very handy way to locate where you are in the section. To move to another

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section of the song, you simply press the corresponding RGB button and the FGDP-50 waits until the current section ends before switching. Each section will continue to loop until you select a new one, with the exception of the ending section, which will play once and bring the song to a close. Playback will continue even if you move to a different mode (other than Session Creator), so you can easily switch to Kit mode and audition different kits while playing back. You can even mute individual parts (or voices), such as the bass or piano, to change the feel of a song.

Session Creator includes the usual slightly cheesy backing tracks you find as demo songs in many keyboards and modules, and isn’t something I personally would use, but it is a fun part of the FGDP-50 that allows you to practice your finger drumming techniques in a range of styles. You can even record your performance using the USB audio recording features and impress your friends!

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eventideaudio.com/h9000 Eventide and Harmonizer are registered trademarks of Eventide Inc. © 2024 Eventide Inc.

ON TE ST

YA M A H A F G DP-5 0

WAV format but can be anything from a single drum sample or loop to a complete instrumental phrase. Once imported, user samples can be assigned to any pad within a kit and manipulated in the same way as the factory samples. This means you can import your own snare, kick drum or keyboard stabs and create a completely unique kit, which hugely expands the sonic possibilities of the FGDP-50. Unlike audio recorded into the FGDP-50, imported samples are saved to the internal 50MB memory of the FGDP-50, which equates to 600 seconds (mono) or 300 seconds (stereo), so any kits you create using your own samples can be instantly recalled without having a USB stick connected.

Virtual Handshake These days many people are choosing virtual instruments as their preferred source for drum sounds, playing in parts via a piano keyboard, electronic drum pad or MIDI controller. The USB connectivity provided by the FGDP-50 makes it a great option for integrating with your DAW and drum software running on macOS or Windows.

The USB connection provides MIDI and audio communication between the

The FGDP-50 measures 223 x 223 x 51mm and weighs 1.1kg.

FGDP-50 and computer, offering stereo audio in and out along with 16 channels of MIDI. Integration with various VIs installed on my Mac was seamless, enabling me to play them using the comfortable pad layout provided by the FGDP-50. In most cases, the mapping of pads to instruments was perfect, with

only a couple of tweaks required on Addictive Drums. Toontrack’s EZ Drummer 3 offered particularly impressive results, with aftertouch translating into the same choked crashes offered by the FGDP-50 itself. Audio recording also worked flawlessly, the stereo output of the FGDP-50 immediately becoming

Little Finger: The FGDP-30 Retailing for considerably less than the FGDP-50 and slightly smaller in size is the FGDP-30. Resplendent in white, the FGDP-30 includes the same ergonomic pad layout as the FGDP-50 but most notably misses off the RGB pads and LCD. The built-in speaker and battery are still present, along with the USB host port, so for use primarily as a standalone instrument or MIDI controller it would be ideal.

Featuring 39 preset kits and 1212 Instruments, the FGDP-30 still offers a big enough sound palette to satisfy the most demanding user, but the lack of a USB flash drive port does mean you can’t import your own samples.

It also precludes recording your performances and exporting to a DAW. The lack of LCD, while not necessarily a big issue for those using the FGDP-30 as a simple instrument or controller, might be significant if you

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wanted to edit in detail. The process is made slightly more irritating, in my opinion, by the fact that Yamaha have chosen to include a voice prompt to help you navigate the menus. As you select a kit or function, a female voice announces ‘10, Modern Jazz Kit’ or ‘Pad Volume 5’. Whilst it’s impressive that the FGDP-30 offers an almost identical level of editability, in terms of kits, pads and effects, as its bigger brother, navigating menus and activating functions by way of voice prompts isn’t ideal.

If performing drum and percussion accompaniment, working on your finger-drum technique and controlling a virtual instrument via the ergonomic pads are what interest you most, the FGDP-30 is a decent option, but for me the FGDP-50’s LCD and ability to import and export WAV files are must-have features, so I’d go with the 50.

available as a track input source in my DAW. The option to play back audio via the FGDP-50 is also available, so you can potentially use it as a basic audio interface.

Thumbs Up? First things first: the FGDP-50 is great fun! The battery operation and internal speaker mean you can pick it up and play, anytime, anywhere, without having to think about audio connections, headphones or power sockets.

Playing through the small 2.5 Watt speaker sounds surprisingly good, and at full volume it’s more than loud enough to accompany an acoustic guitar player.

The pad layout is extremely well thought out, feels very natural to play and, in conjunction with the sensitivity of the pads and aftertouch, makes creating dynamic and naturalistic patterns a breeze. The sonic palette is extremely wide, with great-sounding acoustic drums and percussion alongside classic electronic beats. The ability to import

your own samples and integrate them into kits is a feature I hadn’t expected but is very much appreciated. The ability to export recordings to your DAW as WAV files takes the FGDP-50 to another level, enabling you to create beats anywhere and integrate them into your studio-based projects.

At around $260, the FGDP-50 isn’t exactly a casual purchase, but it does provide a lot of functionality that songwriters and live performers will find fun and creative, as well as offering studio professionals a very well thought-out and fully featured MIDI input device. The pad layout makes creating a passable rhythm very easy, even for someone who has never played drums, and with a little practice it’s possible to play some very complex and impressive-sounding beats.

I’ve even managed to integrate it into my acoustic drum-kit setup as a way to add some additional percussion.

If you don’t need all the features of the FGDP-50 but you like the concept, it may be worth looking at the FGDP-30 (see the ‘Little Finger’ box), which offers a slightly reduced feature set for around half the price.

“With a little practice it’s possible to play some very complex and impressive-sounding beats.”

If I’m being really picky, the small LCD makes navigation around the menus a little tricky, with only limited information visible. Ironically, this is only an issue because the FGDP-50 offers such an extensive level of editing.

I have a similar feeling about the Finger Drum Pad having only a stereo mini-jack output and no dedicated MIDI ports.

The lack of connectivity feels at odds with the generous functionality Yamaha have crammed into this unit.

$ $259.99 W www.yamaha.com

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COMPE TITION

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B

ack in 2016, Slate Digital took the microphone market by storm with their revolutionary VMS, or Virtual Microphone System. Designed to allow engineers the opportunity to audition and record with a huge collection of faithfully modelled vintage and esoteric microphones, the VMS was unique at the time in that it combined hardware and software to achieve its highly accurate microphone models. By pairing modelling DSP with their own high-quality ML-1 microphone, Slate Digital were able to tune the software to work perfectly the raw analogue signal from the mic, with impressive results. And it wasn’t just microphones being modelled: two virtual preamps are

“The system has evolved considerably since 2016, and now boasts a whopping 26 microphone models...”

also available, based on the legendary Neve 1073 and Telefunken V76.

When we first reviewed the system, we were wowed by the quality of the modelled mics, which included some of the most revered valve models in the world. And as well as the sound, the ability to change the microphone model after recording had huge potential in terms of recording and mixing workflows.

To enter, please visit: https://sosm.ag/ml1-comp-0225 The VMS has been much imitated since, but Slate Digital haven’t been resting on their laurels: the system has evolved considerably since 2016, and now boasts a whopping 26 microphone models including all the add-on packs.

Some of these bundles are based on the specific microphones owned by world-famous studios, giving you access to Blackbird Studio’s collection of valve Neumanns, the AKG C414s and Coles

ribbon mic found at Strongroom Studios in London, and many, many more.

For this month’s exclusive SOS competition, we’re giving you the chance to win a Slate Digital ML-1 of your own, plus a year’s subscription to the Slate Digital Complete Access Bundle. As well as including the complete collection of VMS mic and preamp models, the Complete Access Bundle includes some of the finest analogue-modelled plug-ins in the world, plus cutting-edge EQ and dynamics processors, the ANA 2 Ultra virtual instrument, creative effects, and much more — a grand total of $15,000 worth of software! And even if you don’t manage to win the ML-1, you could still end up one of our five runners up, who will each get a year’s subscription to the Complete Access Bundle.

To be in with a chance of winning one of these fantastic prizes, simply follow the URL shown and answer all the questions there, by Friday 7th March 2025.

Good luck!

Prizes kindly donated by Slate Digital W www.slatedigital.com

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Novation Launchkey MK4 Controller Keyboards ROBIN VINCENT

T

he Launchkey MK4 range certainly made an impression on my doorstep when an enormous box containing six MIDI controllers in all sorts of shapes and sizes arrived. The forms fit into the most popular and fought-over area of decent MIDI keyboards. And like their contemporaries they are plastic, chunky, smart and offer a similar level of connectivity and controllability for a reasonable price.

The Range Navigating the range is relatively easy.

At the top you have the comprehensive Launchkey 61 and 49. These are the only ones with faders, associated buttons and a semi-weighted, piano-lipped keyboard.

The faders have been rearranged from the MK3, appearing now on the left of the controller, with the pads and encoders to the right of a new, central OLED display.

The pitch and mod wheels have been sucked into the same space rather than being, more traditionally, to the left of

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Novation’s updated Launchkey range introduces improved creative features and DAW integration.

the keyboard. Other than the physical control differences, the 61 and 49 also have a facility for splitting and zoning the keyboard, which the smaller ones do not.

The Launchkey 37 and 25 lose the faders and keep the full-sized keys but with a synth-action feel. Whereas the Launchkey Mini 37 and 25 reduce down to a more compact form, with comfortable mini-keys and the simplicity of pitch and modulation touchstrips.

Otherwise, the guts and the intelligence inside the controllers are the same throughout the range, as is the single MIDI output on the back and the smart, plastic but solid feel.

The only missing feature is any form of aftertouch on the keyboards; however, you will find polyphonic aftertouch on the velocity-sensitive and brightly lit pads.

Arp With the MK3 Launchkeys, Novation introduced Mutate and Deviate functions, which did interesting things to the melody and timing of the arpeggiator. These have been expanded with some creative

options and are so much easier to use than they were before. The Arp button now stays lit, and while the encoders are not labelled, all the controls are displayed on the nice but tiny OLED display, showing you what the encoders do.

Mutate adds up to 128 melodic variations to the notes you originally held.

Whereas the newly renamed Rhythm gives you eight deviating steps of note triggering. The pattern is shown on the OLED screen and can be edited on the top row of pads, while the second row can add ratchets, accents or ties.

This is a lot of unexpectedly good stuff and pushes the arpeggiator towards something approaching a sequencer.

Hidden at the end of the arp types is another tasty titbit of a feature called Strum. Nothing happens until you move the modulation wheel, at which point it cycles up and down through the held or latched notes. It’s very nicely done. In fact, if you have the smaller Launchkey with the modulation touchstrips rather than a wheel, then you have a slight advantage in that you can access

different parts of the arpeggiation depending on where you place your finger.

Scale & Chord The scale count has been increased from eight to 30, giving you more scales than you’ve ever heard of. There are three modes: the first one snaps every wrong note to the nearest right one, the second filters out the wrong notes and doesn’t let them play, and the third mode puts the whole scale, regardless of what the notes are, across only the white keys.

‘User Chord’ lets you assign any chord or collection of notes you play to a pad for instant replaying. ‘Fixed Chord’ is similar, but this time, you are assigning a chord to the keyboard that can be played and transposed on every key.

That’s all good and useful. ‘Chord Map’ is something quite different.

Chord Map gives you a set of eight chords to complement your chosen scale.

These are available on the left half of the 16 pads. There are 40 banks of eight chords that can be reached with ‘Adventure’ and ‘Explore’ encoders, which can take you on a fascinating journey into relative chords and progressions. There’s also a Spread function that pushes the chords nice and wide, and a Roll function that behaves as I thought Strum would, which leads us nicely into the playing modes.

On the right of the complementary chords, you have six illuminated pads that change how the chords are performed.

The first pad gives you an arpeggiation up, but you have to tap the chord pad for each note. There’s also one for arp down.

Then, you get cycles of inversions with each tap, a swap between left and right hands, and a bass note played on the first tap, followed by the chord on the second. You can hold the mode pads to add the performance feature momentarily or latch it permanently.

The Chord Map is an enormous amount of fun that can keep you fascinated for hours and is ultimately a very useful compositional tool. The only slight disappointment is that you can’t route the chords to the arpeggiator, which feels like a missed opportunity.

controllers now offer a similar level of control and clip launching built into the hardware, so it seems right that Novation should actively expand into DAWs other than long-term collaborators Ableton.

With the MK4, we get scripts for Reason, Logic Pro and Cubase, as well as some enhanced control options in FL Studio, along with a range of controls applicable to any DAW. As you’d expect, the integration with the included Ableton Live 12 Lite is flawless. The pads become a colour-accurate 2x8 grid of clips that can be launched, enabled, navigated and activated with ease. The encoders can take on a number of roles. Using the Shift button and a pad you can switch between Plug-in, Mixer, Sends and Transport control. The faders naturally fall to level control. I appreciated how the OLED display shows a slightly cryptic abbreviated reminder of what each knob is assigned to, because otherwise, the lack of labelling would be a bigger issue.

Plug-in mode maps the encoders

Novation Launchkey MK4 From $119 PROS • Solid, good-looking controllers.

• There’s a size for almost every scenario.

• Creative arp and chord features.

• Tight Ableton Live integration.

• Decent integration with other DAWs.

CONS • Screen is very small.

• Needs better documentation on how to use it outside of the basics.

• The 49/61 are a bit on the chunky side.

• No off button or sleep mode.

SUMMARY Novation update the Launchkey with creative arp and chord modes and wider DAW integration. With six sizes in the range there’s a controller for almost everyone.

experience. You have Capture MIDI, Undo/Redo (which is always handy) and Quantise and Metronome. There’s also a row of buttons under the faders that arm tracks for recording, and the faders also have their own custom modes.

I really like how the Launchkey is unfazed by mode switching or going to different functions, and how it lets me use the encoders and pads in whatever way I wish without ever feeling like you are dropping out of Ableton control.

I found the Reason integration to be a little quirky. On an audio track, control defaults to the mixer, giving the encoders control over the compressor, gate, the first two knobs of the EQ and then panning and level. That’s fine, but the mixer has a lot more knobs, and there were no additional encoder banks to address them or any way of remapping them that I could find. There are eight send knobs which would be perfect for the eight encoders in send mode, but sadly whichever mode you choose, it sticks stubbornly to the same mixer controls. Faders, if available, didn’t seem to do anything, which is very odd.

If you add a Reason instrument to a track, then the encoders grab control over a bunch of parameters and, if it has a button matrix to enable modulation and effects, the pads light up helpfully to reflect their state. The faders now come into play and offer control over additional parameters. This is all great, but there doesn’t appear to be any way to shift control back to the mixer to control the levels.

“Novation have done a lot of good work in making the internal functions of the Launchkey more useful and entertaining.”

DAW Control The MK4s come with enhanced DAW control features, opening up deeper integrated control. Other MIDI

directly to the selected device with the assured confidence of a long-time collaborator. Or, for third party plug-ins you are free to map using MIDI Learn or the Configure function on the plug-in’s Device panel.

Transport mode is best used in the Arrangement view, which throws up one misstep in this otherwise fluid hardware control experience. And that’s the absence of any way to switch views from the Launchkey.

The pads have an extra feature in Live called Sequencer Mode. This turns the pads into a bar of 16 steps that are perfect for drum programming and not bad for melodies either. Hold the note or drum sound you want to play and tap the pad to store it in the step. It feels a little ham-fisted at first but once you get the hang of it, you find yourself crafting rhythms and tunes that you might not have thought of playing live.

The Launchkey 49 and 61 have additional buttons that add to the Ableton

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The full range. To give you an idea of the relative sizes, the Launchkey Mini 25 measures 338.5 x 176.8 x 49mm, while the Launchkey 61’s dimensions are 895 x 263.5 x93mm.

Getting control over a third-party VST instrument also has some challenges. You can ‘Remote Override’ to map control, but this only works on the first bank of knobs, and it replaces the automatic control you were getting over Reason devices. I found that if I created another generic controller and gave it the Launchkey MIDI input (ignoring the complaint about a clash), I could then use any custom modes and encoder banks outside of the first eight without any bother. If you think that sounds a bit fussy and confusing, you’d be right.

I had an entirely different experience with FL Studio. If you have faders then they control the mixer, the encoders jump all over any loaded FL Studio instruments, and the pads light up with the colours of the tracks, letting you select the track you want to work on. If you switch to Mixer mode, the encoders become banks of channel control, mixer control and EQ. You can leap around in blocks of eight and get your fingers into more or less everything.

FL Studio also has an active community who like to get their hands dirty. They’ve created an unofficial script that covers nearly 300 third-party plug-ins, giving you instant, easy control without having to use custom modes or manual MIDI mapping.

Stepping outside the scripts, I wanted to see how well it works with something that’s not preconfigured. For that, we’re going to have to use our old friend HUI. The Launchkey comes with two sets of MIDI drivers: one for regular MIDI and one for DAW control. This brilliantly enables

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us to create both an emulated HUI controller and an external MIDI controller for playing and controlling synths. The controls worked perfectly.

The smoothness of all this is very impressive. I can use every aspect of the Launchkey, both DAW control and MIDI control, without any bother, and I wonder, not for the first time, whether all the effort put into scripting is as necessary or as helpful as we’re led to believe.

Conclusion Novation have done a lot of good work in making the internal functions of the Launchkey more useful and entertaining.

The arpeggiator has been taken to a very creative level, the chord features stand well above what we’ve come to expect, and the Chord Map, in particular, can take you off into very welcome flights of fancy. It gives you an inspirational alternative to digging around for MIDI effects and tools within your DAW. The range of sizes gives an option to almost every situation without sacrificing too many features. The overall quality of the hardware is good, the OLED display is nice if a little small, and the general navigation is well-lit and competently done. If I was looking for a complaint, it

Pricing Launchkey Mini 25 Launchkey Mini 37 Launchkey 25 Launchkey 37 Launchkey 49 Launchkey 61

$119.99 $149.99 $179.99 $209.99 $249.99 $299.99

would be that there’s no off switch, and so the brightly lit pads don’t turn off with your computer.

In terms of DAW control, it’s great to see Novation venture out from Ableton Live and that’s been largely successful.

They could improve their documentation on how to get things to work smoothly beyond in-built devices and into the third-party plug-ins that probably make up most of what we use.

Overall, the Launchkey MK4 is a solid upgrade over the previous version, and the consistency across the range is top notch. It has some great tools, good integration, and the Mini 37 could easily stick around as my day-to-day controller. $ See Pricing box.

W www.novationmusic.com

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Nextmidi Divisimate 2

The Arpeggiator in Basic mode: step lengths can be freely shortened or lengthened to tie across as many steps as required.

When activated, the link icons create a brief overlap from one step to the next, enabling legato transitions to be triggered in those instruments that support them.

Arrangement Software

Divisimate 2 brings a host of improvements and new features to Nextmidi’s orchestration software.

NICK MAGNUS

U

p until four years ago, composers dreamed of being able to play chords on the keyboard and have each note of those chords go to completely different instruments, all in real time.

The ability to instantly audition different instrument combinations and voicings this way would save so much donkey work, rather than splitting out a ‘sketch’ part to separate tracks, or recording one instrument at a time just to see if an arrangement worked. In 2020, the impossible dream came true with the arrival of Nextmidi’s Divisimate. Put simply, Divisimate intercepts MIDI data from your keyboard, splitting simultaneous notes into individual data

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streams, then sends them to your DAW where they can be routed to whatever instruments you please via 32 virtual MIDI ports. Divisimate evolved considerably since its v1.0 release, reaching v1.4 (see the ‘Previously, On v1...’ box); our 2020 review further explains Divisimate’s essential principles, which remain pertinent four years on: www.soundonsound.com/reviews/ nextmidi-divisimate.

Divisimate 2 is a major update, bringing new features and many ‘quality of life’ improvements, requiring a complete rebuild of the core program (see the ‘Version 2 Installation Notes’ box). The first change of note is that Divisimate 2 now works as either a standalone app or a VST3/AU/ AAX plug-in within your DAW. We’ll get on

to the plug-in implementation later. The most significant feature additions from both creative and tailored control perspectives are the Arpeggiator and the Transformer — two new modifier plug-ins to complement the Transposer, Trigger and Repeater.

Arpeggiator The Arpeggiator could be considered as an evolution of the Repeater plug-in, and whilst it can perform the same function, the Repeater is a simple tool by comparison.

The Repeater has been retained to ensure backwards compatibility with any Divisimate 1 presets that used it. The Arpeggiator provides two modes of operation: Basic and Expert. Basic mode offers all the standard features of your typical arpeggiator; note order, note values ranging from 64th to half dotted, swing, single/double/triple repeats and octaves. Up to three sequencer pages each offering up to 24 steps in length provide a possible total of 72 steps — more than enough for creating interesting, evolving patterns. Step velocities can be drawn individually or with a sweep of the mouse across the page, and each step’s duration can be shortened or lengthened to tie across as many steps as desired. Even monophonic legato instruments are catered for — small link icons connecting each

Nextmidi Divisimate 2

Version 2 Installation notes If upgrading from Divisimate 1, Divisimate 2 should install alongside it... but only, it seems, as long as you are on v1.4. That update introduced the Nextmidi Hub port management system, which replaced the previous method where Divisimate created and managed its own ports. Perhaps because of this change, installing v2 caused my installation of v1.39 to be unilaterally removed. No matter, there’s nothing that v1 did that v2 doesn’t do, and then some. To make the transition painless, upon running v2 for the first time, it asks if you want to import all your v1 performances and presets. That would be a yes, and you can carry on using Divisimate just as before.

step can be activated to smoothly connect specific steps; adjusting the Legato Delay setting compensates for the time it takes for legato transitions to play out, so those steps always fall bang on the grid. The pattern’s overall velocity range is affected either by key velocity or any MIDI controller; this can be continuously biased towards either the steps’ programmed velocities or keys/CC with the slider at top right. On the far left are slots relating to four additional lanes to which any MIDI CC can be assigned; curves of any shape drawn in these lanes follow the pattern, adding dynamic animation — whether it’s vibrato, breath control or filter cutoff — or whatever else your instrument responds to. Pattern Reset ensures that however many or few notes are played, it always resets to the correct starting point when the pattern cycles around.

If Basic mode seems like a pretty capable arpeggiator, things go up several gears of complexity in Expert mode. Each voice of an Orchestration (up to eight) is represented in its own pattern lane; a lane can only play one note at a time, and its steps are activated (ie. notes will sound on that lane) by clicking on the crosshair at each step. Steps are coloured white by default and play at a single pitch, determined by the chord played and the current divisi mode. Clicking and dragging up or down on a step will transpose it by up to +/-1 octaves; however, the transposition is not by semitones, but by scale degrees, as determined by the Global Scale. So if we’re in D major, and an untransposed step plays a D, transposing it up by one plays an E, up by two plays an F#, and so on.

Since multiple lanes can play simultaneous notes, it’s possible to craft polyphonic arpeggios or entire musical phrases that lock to the musical scale as you move around — no illegal notes! Furthermore, if Follow Low is selected in the Global Scale editor, you can change key as you play, and the pattern will follow suit. The creative possibilities of Expert mode are huge, from simple arpeggios or musical phrases where all voice lanes are routed to a single instrument, to multi-part polyphonic patterns

that dance between, say, all the instruments of a woodwind section. That capability alone is impressive, but here’s the biggie — every port can host its own completely independent Arpeggiator, doing its own thing. With multiple Arpeggiators routed to different instruments, we have something that is highly reminiscent of Sonuscore’s Engine that powers The Orchestra, Elysion and The Score, but with the potential number of Arpeggiators being limited only by the number of ports. Tempting as it may be to pile on the Arpeggiators, the mounting complexities could well fry your brain, if not bring your system to its knees!

The Transformer Most DAWs have some kind of logical editor that transforms one type of MIDI event into another, for example changing CC7 into CC11, altering all your G sharps to G naturals, or applying changes to a particular note range. The new Transformer plug-in performs all these types of task in real time — and like the Arpeggiator, separate instances of the Transformer can be applied to individual ports. Typical applications would be to restrict the range of an incoming CC in order to avoid an instrument’s upper or lower dynamics, or to focus on a particular CC range, whilst

$249 PROS • The unique Arpeggiator.

• Real-time transforming of MIDI events.

• Dynamic preset changing via the Multipads.

• 512 outputs possible via multitimbral ports.

CONS • Apart from the time spent getting to know it, absolutely none.

SUMMARY Not so much an update as a complete rebirth, Divisimate 2 brings a wealth of creative inspiration to composing for multiple instruments, not to mention making the process of arranging and orchestrating a delightful and satisfying voyage of discovery.

still having the full range of movement on the mod wheel, knob or fader. Transformer provides up to eight slots for defining the conditions; these might operate on the MIDI serially depending on the task. For example, to make any incoming notes play a cymbal whenever the velocity reaches 127, the first condition would force all incoming notes to play one specific note (the cymbal); the following condition would be to block all note velocities below 127. Source event types that can be transformed are note range, note, CC, Program Change, channel pressure and pitch bend; destination event types include the same choice as the source types, plus the blocking of specific events and/or their ranges, and the option to quantise incoming notes to the Global Scale. Such a vast array of permutations makes this plug-in a real problem solver — if

In Expert mode, the Arpeggiator becomes a phrase generator. Polyphonic patterns and phrases can be constructed using multiple voice lanes — especially creative when each lane is assigned to a different instrument. Note also the freely drawn CC1 controller curve adding dynamic movement to the pattern.

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you can think of an application, you can most likely achieve it with the Transformer.

Any Port In A Storm The original Divisimate’s 32 virtual ports would seem more than enough to cope with most situations. Making the not unreasonable assumption that the demographic of Divisimate users largely comprises those working with orchestral templates, many of them will be film and games score composers. So what if you need to include numerous synths and other esoteric instruments in your orchestral score template, but you’re running out of ports?

The solution is to give every port the option of multitimbral capability, where each MIDI channel is sent to its own discreet sub-port — that’s a massive total of 512 ports at your disposal. Not only does this satisfy the most megalomaniacal user, but it makes life tidier for those who like to keep groups of similar instruments (eg. all strings, all woodwinds or all percussion) each in their own instance of a plug-in such as Kontakt.

It also keeps the UI cleaner as the contents of multitimbral ports are treated like folders that can be collapsed to a single line when not the focus of attention. In a similar vein, section headers can be inserted anywhere on the Orchestrator matrix, which in turn can be collapsed for neatness. I’m sure no-one is foolhardy enough to attempt playing 512 instruments at once, each with their attendant Arpeggiator, but you never know...

As well as providing a selection of preset scales, the Global Scale Editor allows for custom scales: the Follow Low option enables the player to change key on the fly.

having to route the Low range to each instrument port; the Global Keyswitch key range is also customisable. You can choose to customise Divisimate’s Humanisation settings for a particular Performance, or have it follow the plug-in’s Global setting.

It would be useful to have this option available on a per-preset basis — maybe it might be considered for a future update?

The way presets change while playing has also received a facelift. Normally, if a preset is changed while notes are sounding, the new orchestration won’t come into effect

until all notes (or the sustain pedal) have been released. Now there are three new transition modes: Merge, Overlap and Retrigger. So as not to turn this review into a major novel, it’s recommended to watch Nextmidi’s walkthough video which illustrates these modes’ differences far more concisely than I can here. www.youtube. com/watch?v=-xkenDWQlHg Naturally, all these Performance settings are included with the Performance when it is saved.

It’s also worth mentioning that Nextmidi offer some very useful Performance

Performance Settings In Divisimate 1, a Performance consisted of five pages, each page containing 20 presets — a total of 100 presets. In Divisimate 2 you can now define the number of pages as anything from one to 20 — potentially up to 400 presets per Performance. A new Global Keyswitch Area can be optionally engaged (highlighted in green on the UI’s keyboard), removing the need to dedicate the Low range to keyswitch duties and repeatedly

When you need to integrate preset changes into your playing, the highly versatile Multipads can do this using a variety of conditions such as key velocity and CC controls — all without the need for Program Change messages.

Previously, On v1...

When reviewed in 2020, Divisimate was at v1.1; the final revision was v1.4, by which time it had evolved with new features. Here are some of the highlights: • The Transposer received a major overhaul: Low, Medium or High ranges can be excluded from transposition; especially useful if using the Low range to perform keyswitches. Notes can be transposed locked to a Global Scale — when active, all notes are transposed by scale degrees, so intervals such as major and minor thirds always follow the scale and key.

A number of preset scales are provided, with the ability to create custom scales. The ‘Follow Low’ option enables a scale’s key to be changed on the fly by playing root notes in the Low zone when it is active.

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• Three Divisi modes: Top Down and Bottom Up determine the order in which chord notes are analysed; Fill mode deals with situations where fewer notes are played than the selected number of voices by distributing those notes evenly across all active ports.

• Further enhancements include DAW tempo sync added to the Repeater modifier plug-in, improved Performance Manager, polyphonic aftertouch and MPE support. Version 1.4 also introduced the Nextmidi Hub, a port management tool that runs in the background, accessible from the system tray. Nextmidi Hub solved a Mac issue where ports would disappear on closing Divisimate — it also laid the foundations for future port management developments as found in Divisimate 2.

In this Orchestration, the string sections are sent to port 1, which is multitimbral, serving MIDI channels 1 to 5. The woodwind and brass each occupy a port of their own, whilst the piano is assigned to the Melody range. Each instrument group has a section header — these can be collapsed to a single line for a neater UI. Note the Global Keyswitch Area on the keyboard, shown in green.

templates for free, on their website. These include DAW project templates for Cubase/ Nuendo, Studio One, Reaper, Logic Pro 10.7, Digital Performer and Ableton Live, as well as the Divisimate Performance templates.

Current offerings are Synchron Prime for the Vienna Symphonic Library (Legacy and new v2 versions), SWAM Big Band and SWAM Orchestra. For other à la carte libraries there’s Universal Orchestra (a full orchestral layout), and the eminently inspirational Universal Voicings.

Multipads This powerful feature provides another way to integrate preset changes into a performance. Any preset slot can be designated as a Multipad, which itself hosts nine preset slots to which you assign existing presets from the preset manager. In the Multipad editor are two bars representing X (horizontal) and Y (vertical) planes. Logically enough, the

Supporting Cast Seasoned users will welcome other refinements: preset attributes can be imported from one preset to another, or across all presets of an entire Performance; multiple routing nodes can be selected and moved, copied or deleted together; Global Scale and Tempo settings are unique to each preset, but can also be locked to ignore preset changes. And lastly, the Nextmidi Hub gains a Merger Port, enabling Divisimate to receive the input from multiple sources, for example a keyboard and a breath controller.

X plane changes presets horizontally, the Y plane changes them vertically. Both planes respond to several event types: velocity, CC, key range (either highest-note or lowest-note priority) and pitch bend. The point at which presets change is governed by adjusting the white boundary markers in the bars. This is an excellent system for making dynamic orchestration changes whilst playing, without having to organise Program Change messages to do the same job. As long you assign otherwise unused CCs to control the Multipad, the CC messages you normally use for dynamics are not compromised. Nextmidi’s Multipad tutorial video demonstrates practical examples of creative ways in which Multipads can be used: www.youtube.com/ watch?v=qA5iDRbfCy8.

Keep It In The DAW As an alternative to the standalone app, Divisimate can now work within your DAW as a plug-in; VST3, AU and AAX formats are supported. Note that you can’t use the

plug-in and the standalone simultaneously — only one instance of Divisimate can be run at a time. Whether you choose the standalone or plug-in depends on your preferred workflow and the task at hand.

The plug-in allows for the automation of Divisimate parameters, and is well suited to recording and editing a complete performance on a single MIDI track, but less so for complex arrangements with independent movement and counterpoint between different sections of the orchestra.

The standalone offers a more flexible approach for working in shorter segments and recording overdubs as you can target specific instruments on their own MIDI tracks. Different DAWs will have their own take on how to set up and use the plug-in; in my case the VST3 plug-in runs inside Cakewalk, inserted as an instrument with its associated MIDI track. It also requires additional MIDI tracks, one for each instrument, all listening to Divisimate, and routed to the appropriate ports. Once you’ve recorded a performance and edited it to your liking, you can then record the results to those individual MIDI tracks for further forensic per-instrument editing.

Conclusion It would be futile attempting to exemplify every use case scenario, due to the literally endless possible routings, assignments, plug-in modifier settings and many other under-the-hood options that affect how Divisimate 2 behaves. Although it can be deep and complex, it’s not difficult to grasp the basics — and having spent time getting to know it well, I’d venture to say that Divisimate could change the way you compose, arrange and orchestrate forever. It’s hard to imagine what else Nextmidi could do to develop it further, but whatever that may be, it’s likely to be exciting — if not surprising. $ $249 W www.divisimate.com

Here, the Transformer confines minimum key velocity to 40, transforms aftertouch to CC12, while notes C-1 and C#-1 send CC23 messages to toggle between two fixed values.

www.soundonsound.com / February 2025

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TREVOR MICHAEL

P

enteo is an upmix/downmix plug-in that started life way back in 2012 as a stereo to 5.1 upmixer.

It quickly became something of a standard in the world of film and TV audio, and has since expanded to include many more channels and features to make it just as useful in today’s multi-format immersive-sound universe.

Penteo is available in three versions: Penteo 360 ($299) offers 5.1 upmixing and downmixing to 13 audio formats; Penteo Pro ($499) provides 16 channels of upmixing and downmixing to 43 audio formats, so can be used for Dolby Atmos formats up to 9.1.6; and, finally, Penteo Pro+ ($799) allows you

Perfect Surround Penteo 6 Pro+ $799 PROS • Upmixes and downmixes so you don’t need two plug-ins • Excellent sound quality • Full ITU downmix compatibility • Built in binaural renderer • Options to suit budgets from $299.

CONS • None.

SUMMARY Unlike cheaper tools, Penteo upmixes and downmixes with a minimum of artefacts — it stands up well in comparison with Nugen’s Halo Upmix and Downmix plug-ins.

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Perfect Surround Penteo 6 Pro+ Upmixing & Downmixing Plug-in

If you do much work with surround or immersive formats, upmixing and downmixing are a fact of life.

With Penteo, good results needn’t cost the Earth. to work with up to 64 channels of upmixing and downmixing to an impressive 62 audio formats, and is the version I was sent for review. (There’s a helpful comparison table for the three version on Perfect Sound’s web store.) Formats are divided into Stereo (stereo LFE. LtRt), Surround (3.0, 3.1 through to 7.1), Atmos (2.1.2 to 9.1.6), Immersive (Auro 9.1 to 13.1 and DTS:X 11.1), 360 Audio (Sony 5.0.2+2B to Sony 7.0.7+2B and Cube 0.0.4+4B) and Ambisonics (AmbiX first-, second- and third-order, FuMa first-, secondand third-order, and UHJ 2, 3 and 4 formats).

All the plug-ins support macOS (running on Intel or Silicon processors) and Windows 10 and 11. They’re available as AAX, AU for Logic (up to 7.1.4) and VST3.

Getting Started A pretty self-explanatory GUI makes it simple to get going fast, but there are some ‘beneath the hood’ options and some things you might need to configure, so

it’s definitely worth reading the manual or watching some of Penteo’s helpful YouTube tutorials (www.perfectsurround.com/videos).

For example, on first inserting an instance of Penteo on a stereo track, I noticed a shift of signal to the left. That didn’t seem correct, but on further investigation I discovered that in the settings pane you must select the in and out channel order to suit your DAW workflow. By default, it’s set to ITU (L, R, C, LFE, Ls, Rs, Lr, Rr, Lw, Rw, Ltf, Rtf, Lts, Rts, Ltr Rtr) but Pro Tools, which I usually use, uses the Film layout (L, C, R, etc). In and out channel order can easily be changed and stored as a new default, so it’s something you need do only once, but it serves as a good reminder to look over the manual!

The main controls are shown in the centre, with the input channels displayed on the left and output channels on the right. Audio always flows from left to right whether you’re upmixing or downmixing, which in my opinion is the logical way to

approach it. Controls are available to adjust Centre, Lss-Rss, Lsr-Rsr, LFE, and Upper.

Each set of controls is colour-coded, and the colours can be changed in the settings pane if they’re not to your liking. Each control has a few options or parameters that can be adjusted, helpfully displayed in the same colours. So it’s pretty simple to see what’s going on.

Input gains can be adjusted individually by dragging the thick fader-like line at the top of each channel. The channels also feature a level meter, keeping the GUI concise and clutter-free. Grabbing the triangle icon to the right of the input level display allows you to adjust all the input channels together. The input section also allows soloing and muting of each input channel as well as a digital value display at the bottom.

On the output side things look very similar. One difference is as default here, all the levels are ganged together. This is deliberate. One advantage of Penteo when upmixing is that it guarantees ITU downmix compatibility (more about that later). Obviously, that’s going to fall apart if you start changing the levels of the individual output channels but, if desired, selecting the gain button above the output section will allow you to tweak the individual channels in the same way that the input side is set up.

Input and output meters are again colour-coded to match the central controls (Screen 1), and various channels can be ganged together (eg. Lss and Rss, or the Upper channels), making muting and soloing these zones easier.

Operating the plug-in is as simple as finding your way around. Select your input format from the drop-down menu in the

input section on the left, and your output format from the output drop-down on the right, and you’re good to go.

Three upmix modes are available though a drop-down menu. In Normal Upmix mode, the input channels are distributed around the surround array according to the input signal format.

Stem Upmix mode takes the information completely out of the L and R channels, and allows the new upmixed audio in the surround channels to be blended in with the original signal. Finally, Stem Passthrough mode does the same, but puts the unaffected original signal back into the L and R channels for you. Further tweaking of these modes can be made with the fader controls in the centre section of the plug-in.

Centre Fader The Centre fader changes the level of the centre channel by redistributing energy to other channels rather than simple

and any phantom centre info is eliminated when the faders are set at unity. Pulling back the Centre fader in this mode will start to bring back some of that phantom centre information, while increasing it above unity gain results in a slightly out-of-phase phantom centre that you perceive as a narrowing effect. Composite mode is a gentler affair, in which the Centre fader is linked to the side surround fader. Finally, there’s Phantom mode, which does away with the centre channel completely.

Tops, Back & Sides The Lss-Rss, Lsr-Rsr, and Upper faders operate in a similar fashion. Lss-Rss distributes more or less energy to the side surrounds, while Lsr-Rsr affects the rears, and Upper obviously the top channels.

Again, rather than add or subtract level in the traditional way, these faders rebalance energy across the channels. For instance, reducing the Rear Surround fader does reduce the sound in the rears, but increases it in other channels.

The side surrounds have a useful Boost feature, which automatically takes some of the energy from the rears and adds it to the sides, while the Surround and Upper channels each have a single-band shelving EQ. By the way, note that while the surround shelf EQ’s colour-coding indicates it’s for the Rear Surround, it also gets applied to the side channels. Not a big deal, but definitely worth knowing.

A drop-down menu above the Upper fader accesses a further four modes, similar to those for the centre channel, and these affect how the signal is upmixed to the upper channels. Low and Raise modes throw more of the off-phase energy into the Upper channels, keeping the in-phase energy in the horizontal plain, while Max mode sends the more in-phase energy to the upper channels. Penteo say Low mode is best used to create subtle vertical effects, and is excellent when used on high-frequency, transient sounds. Raise mode is useful when the user wants to create a greater size of the overall 360-degree sound field around the listener, and Max Lift is best used when a strong, more intense emphasis to the overhead channels is needed. Finally, Mute mode mutes all the upper channels.

The Upper section also has a Tilt option, which subtly shifts the upper information towards the front upper channels and, finally, a Drop Vox option basically takes the

“The centre fader changes the level of the centre channel by redistributing energy to other channels rather than simple attenuation.” attenuation, so that reducing the centre channel never alters the overall signal level. This is how Penteo maintains that 100-percent ITU downmix compatibility I mentioned above.

Just above the Centre fader is a four-mode drop-down menu, which affects how the centre channel operates.

The default mode is pretty standard: the centre channel partly replaces the phantom centre from the input channels. In Discrete mode, a discrete centre channel is created

Screen 1: The meters are colour-coded to match the corresponding controls.

www.soundonsound.com / February 2025

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PERFECT SURROUND PENTEO 6 PRO+

centre channel energy out of the upper channels. That’s potentially a very useful function, especially in the film world where a stereo music track needs some gentle upmixing (due to an immersive or multi-channel version being unavailable); this would ensure centre information like the lead vocal doesn’t ‘come at you’ from above.

Inserting another instance across a keyboard pad spread it around the room subtly, and the side surrounds’ Boost brought it nicely to the sides while still maintaining that sense of it ‘being around you’. These first impressions were just reinforced by subsequent experiments with other sources, which were similarly successful.

A final point to mention is that LFE the Output Visualiser (see Screens 2 and 3) provides a fantastic visual The final parameter in the centre representation of what’s going on section is the LFE, which is adjusted at all times, from all angles. There’s using a knob/switch control rather a 2D view on the left and a 3D than a fader, and has a drop-down view on the right, which can auto menu to select the crossover rotate if that’s your thing. (It is mine.) frequency (63, 80, 100, 125, 160 There are options here to show all or 200 Hz, or full range). Another Screen 2: The Output Visualiser, with a 2D view on the left and a 3D speakers, or horizontal or upper drop-down, Effect, offers a few one, which can automatically rotate, on the right. planes, keeping things visually options. When set to off, the LF pleasing and impressing all the clients that section is switched out and has no effect. than just moving the energy around the come in the studio.

The Boost option adds LF content without soundscape, so it will have an effect on how All in all, then, I really can’t fault Penteo: changing the main channels, whereas it all folds down too. That may not matter it does what its developers claim, and when Split mode is selected the LFE to you, but there’s a very helpful Downmix does it well. In fact, I’d go as far as to channel will share the workload of the Compatibility Monitor section (top right of say that there’s nothing to dislike about main channels. Essentially, this means in the GUI) that warns you when this happens. this plug in! I’ve tried a few upmixing Split mode with LFE set to max, the main Again, it’s displayed in percent, making tools at the more affordable end of the channels get high-pass filtered, as the LF it easy to tell how far away you are from market and have never been happy with energy is distributed between the LFE and perfect ITU downmix compatibility. the result. A better point of comparison main channels. Reducing the LFE setting Perfect Upmixing? for Penteo would be the combination will result in more of that LF content being of Nugen Audio’s excellent Halo Upmix added back into the main channels. When OK, that’s how it works, but what are the and Downmix plug-ins, and I can’t really using Penteo with Sony 360 applications, results like in the real world? The demo recommend one option over the other: a Bottom fader is introduced, so that energy version arrived just as I was mixing a live both are simple and intuitive to use, laid can be distributed around this format’s track in Atmos, and I immediately inserted out well, and sound excellent, with little three planes. is across a stereo stem I’d been sent of to no audible phase artefacts (depending, A Diffuse parameter can be set (in the recorded room ambience. I was easily of course, on the material and amount of percent) individually for the Rear, Upper able to spread the room sound around the upmixing required), and they are clearly and Bottom channels. It obviously tends to immersive sound field, favouring it towards the ones to beat. smear the signal a little, but if used sparingly the back and sides but with a hint in the One advantage Penteo may have it can be just the right thing for some jobs.

Upper channels too. Penteo performed this over Nugen’s offering, though, is that Tweaking the Diffuse parameters, or function perfectly: I could hear no phasing it upmixes, downmixes and works in Surround and Upper shelving parameters artefacts (unlike with other upmixing plug-ins the 3D audio environment. Halo Upmix in the Rear and Upper channels, obviously I’d tried!), and the fold-down to stereo and upmixes only to 7.1 unless you pay extra changes the sound of the signal rather binaural formats worked really well. for their 3D Immersive Extension licence (which still only takes you up to 7.1.4 channel widths) and for downmixing you must use the separate Halo Downmix.

Penteo’s mid-priced option puts all this functionality in one place, and is significantly more affordable. Oh, and Penteo also includes a binaural renderer.

What’s not to like? If you’re still not sure, then a 14-day fee trial is available from the Perfect Surround website. $ Penteo 360 $299, Penteo Pro $499, Screen 3: The Output Visualiser also allows selective views of different planes.

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Penteo Pro+ $799.

W www.perfectsurround.com

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ON TE ST

Play All Day PlayFader Sequencer & Controller

Play All Day’s debut outing takes a novel approaching to hands-on hardware control.

ROBIN VINCENT

T

he PlayFader stands out against a backdrop of serious synths and black and silver modular, and that alternative vibe is most certainly deliberate. It draws you in and invites you to play. With buttons arranged like an assortment of Chewits and a pair of big DJ-style faders, it feels like you know exactly what to do.

But while it may look like a DJ effects box, it is actually many other things.

The PlayFader is a two-channel MIDI and CV sequencer, controller and button-laden fun box adventure. It’s unexpectedly deep, well thought-out, inspirational and, at times, frustrating.

You can come up with a fabulous groove in moments, fiddle it about, forget your button combos and accidentally overwrite yourself all with a huge smile on your face. It’s a lot of fun.

It was designed by Play All Day founder Andrew ‘Hadj’ Hadjiantoni, who left a corporate career at Apple to run Ambient Sound & Music retreats in Mallorca and design fascinating controllers. For the PlayFader, he

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partnered with Jason Hotchkiss of 64 Pixels to provide the coding.

Physicality The unit itself is a decent Game Boy-sized device encased in recycled plastic, which can be easily swapped out for a more subdued colour. The top cover that protects the controls also fits on the bottom to raise it up a bit and provide a chunkier and grippier form. The Bourns faders have a pleasing amount of resistance over their 60mm throw, but there’s quite a bit of wobble.

On the top end, you will find a row of mini-jack connections. If you turn the PlayFader over and remove the cover that’s now covering the bottom, you’ll find the corresponding labels.

You have two lots of CVs/gates for Play A and Play B, and MIDI or sync in and out, which can be set with a couple of recessed switches. The power socket is on the same row, and in the box you’ll find a braided myVolts Ripcord cable that can plug into any USB socket and supply power to the PlayFader.

Plug it in, and the centre space between the faders lights up with a lovely

green segment LED display. The display runs at right angles to the flow of the PlayFader, making you feel you should turn the device into landscape and use the faders as crossfaders. But that’s not how this is supposed to work. This slight disorientation is somehow part of the charm of what we’ll discover is quite a quirky device, and you’ll be surprised at how little it matters.

So What Does It Do? Either side of the two-channel controller can sequence or control things, quantised or unquantised. This can be CV and gate or translated into MIDI notes and CC numbers. You can have both the analogue outputs and MIDI working simultaneously.

So, grab the fader to select the note or control the voltage level and hit the white Play A or Play B button to squirt that into your gear. The idea is that you have a running loop into which you capture notes and modulation on the fly.

In a couple of taps you’ve got a looping bass line and a melody, or a bass line and modulated filter, or looping modulation running to whatever you want. You can

add to it, overwrite it, clear it, rework it or throw on some ‘Fun’ effects.

As a sequencer the fader acts rather like a trombone. You are sliding through all the notes to find the one you want.

With a scale enabled, the display will show note letters (sideways) alongside the fader, but, as with the trombone, you won’t hear the note until you blow or, in this case, hit the white Play button.

So, to write your melody, you skip about with the fader to find a note on the display and only hit Play when you have the right one. Making it sound good in a live performance is going to take some practice.

Each channel has two recording buffers, so you can have two things captured and flip between them. You can transpose, reset, hold notes or punch in live over the top. You can define the range of octaves or depth of voltage, set up user scales and timing quantisation.

It’s brilliantly simple, clever, and thoroughly engaging. However, getting it to the state of controller bliss can take some novel finger gymnastics and a great memory for button combos.

Preparation, Recording & Fun I’ve found that to get to the point of PlayFader fluidity you need to do some prep. The first thing is to enable quantisation and change the scale.

There are a number of built-in options, but the most useful is the user scale, where you can choose your own notes and set a root note and octave range.

You should now be sliding about over a handful of intentional notes, which will give you a much better playing experience. Scale is also where you can set a voltage range for CV control if you intend to use the channel for modulation rather than sequencing.

Sequencing on the PlayFader happens in real time; there’s no step entry, so you are going to be performing the notes live into a looping sequence of up to four bars, so some help with the timing might be useful. The PlayFader has a flashing metronome, but it’s so snug up against the Fun button that you can’t see it unless you are looking directly from above. However, there is a timing quantisation grid from four steps per bar up to 32nd-note triplets, which should keep everything together.

With scale and time sorted out you’re in a much better position to pull off a decent fader performance.

The PlayFader’s rear-panel connections include 3.5mm ports for MIDI/sync and two sets of CV and gate outputs. At the farthest right is an input for a 9V power supply.

Recording takes place only when the red Loop and white Play buttons are used in combination. Tap the red button while you’re playing to capture the next note, or hold it to capture a few in a row. Or you can start by holding the red button and then add a single note with the white button, let it loop around and add another. Other button combos let you erase notes and clear the buffer to start again. Tapping in a quick sequence or building one up, provided you’ve done your prep, is really quite effortless. It’s an enjoyable way to work with your synths.

With another quick button combo, you can switch to the second buffer and record another sequence or copy the other buffer across and drop in some variations. You do, of course, have a whole other channel to play with.

Recording CV works in exactly the same way but without a scale or grid.

It’s worth pointing out that CV is only transmitted when you press the white Play button. You can sometimes wonder why the filter you’re trying to control stays resolutely unmoved when you move the fader and see the display move in response.

Under the orange Fun button, we have a range of effects that will either transform or replace our loops with bursts of performance-friendly happenings. The effects in question are Roll, Slice, Sustain, Slew, Random, Bend and LFO. Roll is your regular ratcheting, Slice loops chunks of your sequence, Sustain holds the gate open, and Slew adds drag to the notes. Random throws in a sample & hold-style random stream of values, Bend applies a pitch-shift to

whatever is going on, and LFO adds modulation on top of what’s there.

As the name suggests, these can be a lot of fun in performance but also benefit from being prepared beforehand.

They are not recordable, so they only happen in the moment. I found that Rolls and Slices are great on sequences, Slew is good in small doses, Sustain and Bend weren’t that useful and LFO and Random were brilliant with CV control.

MIDI The MIDI out and in ports also double up as analogue sync ports, but you can’t use both, so you will need to choose which one you want to use via two switches on the back of the unit. This may trip you up the first time you try to use MIDI, as might the fact that you can turn MIDI off in a menu buried deep within the machine.

Play All Day PlayFader $339 PROS • Enormously fun and freeing.

• Feels different.

• Sparks creativity.

• Playful loop sequencing.

• Fun performance controls.

CONS • Lots of button combos to learn.

• Changing modes can be a handful.

• Some MIDI issues need fixing.

• Wobbly faders.

SUMMARY The PlayFader is a novel fader-based sequencer and performance controller that’s fun to use once you’ve learned your special move combos.

www.soundonsound.com / February 2025

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Annoyingly, I found this to be the default setting. At a basic level, the MIDI mirrors the CV generation and replaces 1V/oct with MIDI notes and CV with CC, but there are a few extras.

From the MIDI menu (Fun+Clear+Play A or B and then double-tap Latch to keep it open), you can choose between Note, three Chord modes, two CC modes and two Bend modes. Notes are obvious enough, but the Chord modes are interestingly quirky. Chord-A mode is ‘automatic’ and quantises triads to the chosen scale. With a user scale it will quantise diatonically depending on what notes are present and add in other notes to compensate. It is odd at times but also very interesting and you can push the chords about by changing the notes in the scale. Chord-C mode is chromatic and gives you major triads regardless of your scale, and Chord-S mode makes chords based purely on the notes in your user scale. It works well enough once you get the hang of what’s going on, but I do feel like it could do with being able to give you more control over the chord types or maybe even store chord progressions.

The CC option has two modes. The first one lets you allocate a CC number and map it to control a knob on your MIDI synth. The second is the same, but they’ve crowbarred in a second CC that fires out either 0 or 127 when you press the white button. So, in practical terms, you can simultaneously control the filter and turn a delay effect on or off when you hold the button. Note CC sends out the root note at the same time as the gate to add a drone note beneath the control to complement the melody on the other channel. It’s an unusual feature that you might find useful but I end up feeling like I’m in conflict with myself over what I’m trying to do.

Finally, the Bend mode is a regular pitch-bend mapped to the pitch wheel, and Note Bend also sends out the root note. I found that in Bend mode, the pitch wheel would just flicker as I moved the fader, but if I recorded the movement, it worked fine on playback.

This is probably just a glitch that could be fixed in a firmware update.

The PlayFader measures 174 x 108 x 40mm, this last figure increasing to 55mm with the cover on.

versatile and even witty at times. For two channels of quick and easy sequencing or controlling that’s in your hand, ready for performative music-making, it’s hard to beat. It has some similarities to the Bastl Instruments Neo Trinity, which does similar note and controller recording on

but this currently only saves the loop data, not the fader configuration. It also pauses playback briefly when loading, which means you can’t smoothly load up another sequence.

Overall, the PlayFader leaves me feeling very happy. The loop-based sequencing is very immediate and enjoyable; I like how it’s not hardwired into my Eurorack case and I can take it all over my studio for some quick playing in any space and on any device.

It’s very versatile; you can see what’s going on, even if it is sideways, and it encourages you to fiddle. You may feel at times like you are battling with the buttons, but with a little bit of patience, you will emerge victorious.

More colours are coming in the next batch so you can have something more tasteful, in case that was a barrier, and the price has also come down to a more realistic €299.

“For two channels of quick and easy sequencing or controlling that’s in your hand, ready for performative music-making, it’s hard to beat.”

Conclusion The PlayFader gets all the fundamentals right. It is fun to play with, useful,

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the turn of a knob. But the PlayFader is released from the confines of Eurorack, and you definitely feel that freedom in the faders and the interface.

One thing I think would ease the button combo learning burden would be to have some sort of profile system, where you could store and recall fader setups and instantly switch between them. Currently, to switch from sequencer to controller, I have to find and change the scaling and the time quantisation and, if it’s MIDI, change the function and set the CC number. There is a patch saving system,

$ $339.99 W www.goplayallday.com

RELEASE ROUNDUP 2024

As we close the books on 2024, we proudly look back on 41 new titles spanning a wealth of styles. Out of all these releases, one that stands out a little extra is of course EZmix 3.

As for 2025, we promise to keep up the same pace. See you on the site!

WWW.TOONTRACK.COM

ON TE ST

Five12 Vector MkII & Jack Expander Eurorack Modules

I

t was Five12’s fault I caved and bought a Eurorack case.

Their Vector sequencer’s mix of MIDI, USB and CV connectivity was perfect for use with my desktop synths.

And then there was a gap in the case that needed filling...

The Vector is a digital sequencer with eight parts assignable as quantised mono or chord tracks, or four-lane drum tracks. Recallable Projects, Presets and Scenes make the Vector ideal as the brain for a live or DAWless rig. But what makes it interesting to me is its capacity for self-modulation. Multiple threads of chance, step operations and cyclic variation can weave complex results from simple patterns.

The original Vector has been around for five or six years and was conceived as a physical manifestation of Five12’s modular sequencing software Numerology. All the core sequencing features of Numerology were teased into a 42HP module with nine encoders and a generous OLED

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display strip. The discipline of reformatting the elements into a hardware controller made the Vector a more focused, interactive and fun environment.

The MkII Vector maintains the same heading as the original, but adds more memory, extra ports, an updated and slimmed-down jack expander and, perhaps most significantly, an extra layer of chance.

Work on the Vector centres around the eight endless push encoders, setting values on multiple pages selected from the top button row. The master encoder mostly scales all values.

Note sequences can be dialled in or you can capture in steps or in real time from a MIDI source, which handily can be a directly connected USB keyboard or Novation Launchpad.

Often the initial creation process in modular sequencing is semi-random: you dial in some notes on knobs, listen to the result and iterate, rather than programming a melody you had in your head. The creative input is then a reaction to that initial seed, and sound design. The Vector provides a shortcut to fun with Generate and Evolve algorithms that work in various styles like Acid, Berlin School or Euclidean.

Gate, Velocity and Slide get their own pages, as does Groove, where you can nudge each step’s timing. The Length, Repeat

and Ratchet pages are what piqued my interest in the Vector.

These work similarly to the RYK M185 and Intellijel Metropolis/x sequencers, providing individual control of step duration, repeats and divisions. A great feature is that as you adjust these a display keeps track of the total sequence length.

A Numerology superpower that the Vector inherits is subsequencers. Each part has two extra eight-step subsequences (plus external sources) that can modulate parameters in the main sequence. Pitch obviously, but also fun things like sequence start, rate and direction.

Then there’s Chance, of which there are now two lanes per Part in the MkII. This is a suite of step operations that can affect far more than just gate probability.

You can, for example, adjust pitches, trigger ratchets and jump to different steps, and can set both a chance and frequency with which an operation will occur. The only things missing are the accumulators found on the Metropolix, TE OP-XY and Cirklon, which can walk notes up or down incrementally over multiple cycles.

I/O was already generous on the MkI, with two Pitch/Gate/ Velocity sets, two TRS MIDI outputs, external mod inputs and both host and device USB. The MkII gains two more assignable outs, and the mod ins are now bipolar. The next-gen Jack Expander has been shrunk to 8HP, swapping MIDI DIN for TRS. Its velocity ports have been upgraded so all can be used for full-range, calibrated Volt/ octave output.

As well as the four additional CV voices, the

expander is especially useful for the multi-channel drum parts, which eat up trigger outs. But even when idle they can all generate various useful clock divisions.

While you can get seriously deep into programming, the Vector can shift gears and become a live performance/ improvisation hub, supported by all that connectivity. On the Part level you can store and recall multiple patterns, then store combinations as global Scenes. Fills are a new feature that MkI users also get, providing template pattern modifiers that you can trigger with different strengths.

Fills take a bit of setting up and are currently mainly effective on drum parts.

Melodic parts can be bent in real time with the Skip and Mute buttons and you can jump between variations in step, but I wish there was a step/loop repeat feature where you could freeze sequences on a single step or range.

My favourite feature — unique to the Vector, I think — is direct support for Novation’s Launchpads. Plug into the USB host port and many of the Vector’s features automatically spill out onto the pads. There’s a command page with track

mute/arm/select and scene launching.

A second layer gives you an Ableton-style view of your project. There are even gate and piano-roll sequencing views, and a note mode for playing and recording.

The Vector is deep, and requires some learning and muscle memory development, but it’s an outstanding sequencer that can integrate your whole system, in and out of the rack.

Simon Sherbourne $

Vector MkII $759.99, Jack Expander MkII $289.99.

W

www.five12.com

Hexinverter Mutant Machine Eurorack Module

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don’t know what it is about drum modules, but I will never not be intrigued when one is announced. The original Mutant Machine was released not far off a decade ago, and it’s worth mentioning a little of the context for this iteration of the module. In September 2022 Hexinverter Électronique founder Stacy Gaudreau announced that all Hexinverter operations were to cease, citing the post-pandemic manufacturing climate as “the final nail in the coffin for us at Hexinverter”. Enter Ģirts Ozoliņš and Erica Synths, who put out a statement of their own: “We strongly believe the effort and innovation incorporated in Hexinverter Électronique Eurorack modules is far too great to prematurely become history.” For that I commend them.

One immediately visible difference between this edition of the Mutant Machine and its forebear is its finish, which is completely gorgeous. It really does look the part of a premium electronic musical instrument, with textured black-on-black patterning to differentiate its different sections and an intricate gold design around its Output Drive knob. Hexinverter Électronique, as the name suggests, are French-speaking; but calling this the Édition Noire is fully becoming of its sleekness regardless of nationality. The original Mutant Machine was grey, silver and orange; a finish clearly inspired by the classic Roland drum machines from which its circuitry draws much inspiration.

‘Another Roland-inspired drum module?’ I hear you yawn. True, it seems everywhere you turn nowadays there’s some sort of homage being paid to the sonic signatures of the 707, 808 or 909; but not only does the Mutant Machine do it better than most, that’s not the only thing on offer here — not by a long shot. Apart from anything else, it’s fully analogue — or, as the official spiel declares, ‘microcontroller-free’ — which somehow sounds like the healthier option. Just don’t tell Noise Engineering I said that, whose digital BIA drum module I am a huge fan of.

In many ways, the Mutant Machine is a little like its own miniature modular system for drums, with a markedly discrete design encouraging individual manipulation of pretty much all of the elements of its overall sound. This it offers down to a fine level of detail. It also makes sense in

light of the fact that, allegedly, the Mutant Machine originally started life as a snare drum synthesizer to work with Hexinverter’s other Mutant drum modules, but became too big and expensive to justify that sole purpose.

Indeed, the pursuit of the perfect snare drum sound is a good way to think about the Mutant Machine and its workflow: you might say that within the sonic signature of a snare are more or less all the ingredients of the drum spectrum: there’s the initial click of its transient, the thump of its lower frequencies, the sizzle of the snare itself, the mid-tone crack, and so on.

Explode these all out and you can indeed build a kit of sorts with a thumping kick, sizzling high hats and the rest. It occurs to me as I write how even conventional

snares can sound like almost any of these, depending on the context. Some are so beefy they almost sound like bass drums; if you don’t believe me, just listen to the snare in Dire Straits’ ‘Money For Nothing’.

The left two-thirds of the panel, ostensibly, is taken up by the Membrane section, while on the right is the (less technically-named) Snappy section. Two analogue VCOs constitute the core of the Membrane section, each with their own level and relative tuning controls. These offer sine, triangle or square wave shapes and while, disappointingly, their CV input doesn’t track at one Volt per octave, it’s possible to get things passably melodic at the Pitch CV input if you’re careful on the attenuator. www.soundonsound.com / February 2025

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Fundamentally, these work in much the way you’d expect, with a combination of amplitude and pitch enveloping to create drum sounds high and low, and this it does beautifully. Things get much more off-kilter and interesting, however, with the Wave Scanner, which has one or both oscillators cycle through the three waveforms. This it can do slowly, in time if desired like a kind of timbral arpeggiator, or at audio rate to create snarling, squealing, metallic textures that can be brought in and out by sending CV triggers to the Enable input. Excellent.

On the other side of things is the Snappy section. This, ostensibly, is a noise generator, but it also has a Click control for more pronounced transients.

This by itself was just the ticket to achieve, for example, that little dose of attack on an otherwise soupy kick drum; but together with the Pitch and Decay knobs there’s a huge amount to play with, and with just a little reverb I found some very realistic rim-style sounds alongside those more ostentatious 909-style white noise bursts.

A lovely touch here is the ability to process other audio beyond white noise through the Snappy circuit, and a dedicated Noise output to send the white noise by itself elsewhere — perhaps to another envelope. In fact, the I/O of the Mutant Machine is generous to say the least: it’s also possible to exchange both the internal oscillators with other audio sources via the Wave 1 and Wave 2

inputs. Beyond its Main Out, it has outputs for both the Membrane oscillators and a clock output, as well as a CV output for the pitch decay envelope. While the Main Out is by far the one you’ll use most often, the other outputs are primed for achieving stereo width, perhaps, or for sending elsewhere in a system for further sculpting.

I must mention the celebrated Basimilus Iteritas Alter at this point. As a multi-function drum heavyweight, it may well have met its analogue match in the Mutant Machine. Like the BIA, this module doesn’t just hit, it hits hard. Erica Synths have hit a home run. William Stokes $

$389

W

www.ericasynths.lv

W

www.hexinverter.net

Modular Profile: Jake Knight

J

ake Knight founded RYK Modular back in 2009, hitting the scene with the excellent M185 sequencer; a module originally designed for the Roland System 100m format but later adapted for Eurorack, and which later became the stimulus for Intellijel’s wildly successful Metropolis and subsequent Metropolix sequencers.

The recent ALGO Quad Complex Oscillator Bank, by contrast, is an impeccably designed Eurorack module for creating complex FM and additive sound textures, with a rather unique algorithm display to boot.

On his entry into modular I first got interested in modular synthesisers as a teenager, growing up in the late ’80s picking up Roland catalogues from a local music shop. One catalogue had the whole range of the Roland System 100m products, which I studied meticulously, but could never afford. I built small modular projects from electronic magazine articles like ring modulators, a compander noise gate and a spring reverb. These I housed in cardboard boxes

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with 3.5mm patch sockets for interconnection. So I had a kind of cardboard modular setup for a few years.

On his go-to modules (aside from his own!)

The reason I started designing modules was to create modules that I wanted to use that did not really exist. So mostly the only modules I use are my own! Probably the only one module that stays in my Eurorack rack is a [ALM/ Busy Circuits] Pamela’s Pro Workout. I often like to use it as a four-channel rhythm sequencer into a Vector Wave set up as four drum synth channels. In the System 100m, I really like the inverter on the 132 mixer, it is great for adding an upside-down keyboard CV to a patch whilst playing.

Also the gate delay on the 172 is really nice for adding an offset syncopated note trigger to sequences.

On the RYK ALGO The idea behind releasing the ALGO was to try and debunk the myth that FM synthesis was hard to understand, to make a module where all the key controls for 4-op FM were available without menus, and to display the

February 2025 / www.soundonsound.com

algorithm in such a way that it is easy to understand the intermodulation between oscillators. Hopefully it has succeeded in this goal.

On the Roland System 100m The first real modular synth I owned was the Roland System 100m, mainly due to a childhood obsession, and also the fact that it used to be available at a reasonable price. I love the design continuity that Roland made with this series; both the graphic design, label terminology and panel layout. Most importantly, the form factor is a perfect balance between size and ergonomics. There is some clever signal routing and normalisation, the power connectors also carry CV from the keyboard input to each

module, and of course every input has an attenuator!

On the culture of modular The use of modular has seen a dramatic resurgence in the last decade or so. I think this is partially due to a desire to experiment and make sounds or music that is perhaps less predictable than working from a computer screen. Modular and hardware systems can also remove some of the modern problem of ‘seeing music’ as you make it. DAWs tend to encourage a visual approach, where every note or loop is visualised and snaps or conforms to a neat and tidy structure. Perhaps with modular people are more inclined to use their ears rather than their eyes, and trust in what they are hearing.

William Stokes

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Electronic Perspectives: Vintage Electronic Musical Instruments Hardback Book Tom Rhea’s colourful, comprehensive and insightful history of electrical and electronic music machines includes CDs with audio examples.

PAUL D LEHRMAN

W

hen I was a kid, among my greatest joys was getting one of those Christmas toy catalogues in the mail. While my sister wanted the dolls and games and my friends all wanted the baseball gloves and football helmets, I salivated over techie stuff: transistor radios, electric trains and cars, building and highway construction sets, and walkie-talkies. Opening Electronic Perspectives: Vintage Electronic Musical Instruments is a lot like poring over one of those catalogues, only looking backwards.

It’s a huge and beautifully put-together compendium of electrical and electronic musical toys, going back well over a century.

Overview Tom Rhea was the right guy to write this book. He is one of the sages of the electronic music era, having penned a column called Electronic Perspectives for the late, lamented Contemporary Keyboard (later just Keyboard) magazine from 1977 to 1981. His PhD thesis, back in the days before the Minimoog (for which he wrote the owner’s manual), was on the evolution of electronic instruments. He had a 30-year career teaching college courses in synthesis and synthesizers.

The book reflects the enormous knowledge Rhea has accumulated: it’s over 400 pages, and weighs upwards of seven pounds. All 51 of his columns are here, reproduced with their original layout and typefaces, accompanied by over 300

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pages of illustrations, and bound in stiff cardboard with a hard slipcover.

Attached to the back cover are two CDs containing some 53 musical examples totalling 1.5GB, some of which have never been heard before, and there are extensive notes about each track. It is, in one oversized volume, the history of electronic music, in words, pictures and recordings, from the pre-vacuum-tube era to the dawn of digital audio workstations.

What might be most remarkable about the columns, besides how well they hold up today, is that Rhea did all of the research for them the old-fashioned way, long before there was an Internet. As musician and author Brian Kehew, who helped on the project, says in his foreword, Rhea used “card catalogues, engineering and music indices, telephone books, letters written, books in library stacks, old newspapers and magazines, [and] flights to interview pioneers using his cassette tape recorder”, as well as the New York Times on microfilm.

Much of the information in his columns would still, even in the age of Google, not be easy to find. Stylistically, though, the

columns don’t feel at all dated — Rhea’s conversational voice remains as pleasant to read as it is informative.

But while the columns form the basic framework of the book, they are enormously augmented by the work of graphic artist

Electronic Perspectives $99 PROS • A comprehensive history.

• Covers both familiar and less familiar instruments.

• Includes two CDs with audio examples.

• Gorgeous graphics.

CONS • None.

SUMMARY A wonderfully compelling and thorough history of electrical and electronic musical instruments from the pre-valve era to the dawn of computer music.

Joseph Bastardo, whose contributions include restorations of vintage ads, product brochures, concert programs, photos, patent documents, newspaper and magazine articles, and many other materials. As you can see from the sample spreads included in this review, they are simply gorgeous.

Some of the instruments the book covers will be familiar to today’s readers: the Theremin, the Wurlitzer and Rhodes electric pianos, the Hammond organ, and Robert Moog’s analogue synthesizers.

Some others will be less well-known: the RCA Mark II synthesizer, for example, which, with its 1700 valves, took (and still takes) up an entire room on 125th Street in New York City, and to which Rhea devotes six columns; the Ondes Martenot, invented in the 1920s but which still has active players in France and Canada; the Ondioline, conceived by Frenchman Georges Jenny in 1939 and used to great effect by Al Kooper on his seminal 1968 album Super Session; and Don Buchla’s keyboardless modular synths. There are also a lot of instruments and systems in the pages and on the discs that will probably be brand-new to most readers, such as the Hellertion, the Electronic Sackbut, the Pianorad, the Clavivox, the Multi-Track Tape Recorder (not what you think it is!) and the Cellulophone. And there are instruments made by well-known inventors and companies that never got popular, such as: Hammond’s Solovox (presented here on disc by the legendary Sun Ra) and Novachord; Theremin’s Terpsitone and electric cello; Telefunken’s Trautonium; and an early electric piano from the German piano maker Bechstein.

and popular tunes of the day. But despite huge investments in the instrument, and a formidable marketing campaign, it was a colossal failure. According to some who heard it, the instrument’s sound was impressive (Mark Twain pronounced himself a fan) but, alas, no recordings of it have survived.

Rhea organises the book in the chronological order in which his columns were published, and these don’t necessarily reflect the chronology of history. Yet this serves him well when he follows the Telharmonium columns with two on the Hammond Organ, introduced in 1933, which was in a technical sense a direct descendant of the Telharmonium: the dynamos were replaced by small metal wheels spinning alongside miniature electromagnets, whose output was amplified by now-ubiquitous vacuum tubes.

Unlike its predecessor, of course, this invention was a great success.

Later in the book he goes back earlier in time to what he calls the first ‘electronic’ instrument — that is, one that used nothing but circuitry to produce sound, with no moving parts: the Singing Arc of 1899. An English physicist named William Duddell discovered that if you attached a shunt circuit to a high-voltage DC carbon-arc lamp, you could make it oscillate in the audible range. The frequency of the oscillation depended on the components in the shunt circuit (resistor, inductor, capacitor) and thus you could design a keyboard that would play different pitches by varying the values of the components. There was no need for a speaker—the arc itself produced plenty of sound! (The same principle is used today in the plasma tweeters found in some high-end speakers.)

Following a column on Harald Bode’s Vocoder and Klangumwandler (frequency shifter), and audio demonstrations of both on the discs, are four pages on Bell Labs’ 1939 Voder keyboard-controlled speech synthesizer, which they nicknamed Pedro (“About a year’s practice enables an operator to make Pedro talk glibly,” states an article from Popular Science magazine), and four more pages on the Sonovox, the first talkbox (which could also be used as an artificial larynx) from 1942.

This book also covers instruments that Rhea didn’t get a chance to write about in his columns. For example, he devotes six pages to the Choralcelo, a variation on a piano that used electromagnets to vibrate the strings so they could sustain tones, not unlike today’s EBow for guitars. Along with two pages of colour photographs, he

Structure & Highlights The book starts out with three essays about Thaddeus Cahill’s 1900 invention of the Telharmonium, an electro-mechanical monster of an instrument that used dynamos (electrical generators) to produce powerful audio signals that were transmitted to subscribers in New York City in real time over telephone wires. It was, arguably, the first electric instrument to enter the public eye. It also pre-dated other subscription music services by about 100 years! The size of 10 railroad freight cars, it occupied an entire building in midtown Manhattan, and required two musicians to play it.

Although its means of tone production was revolutionary, its repertoire was hardly so: in order to attract subscribers, Telharmonium programs consisted mostly of light classics www.soundonsound.com / February 2025

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E LECT RONI C PE R S PECTI V E S : V IN TAGE E LECT RONI C M USI C A L INS T RU M E N T S

reproduces an article from 1916 that calls it “the most wonderful musical instrument ever thought out by the human mind”.

A sampling of it is the first track on disc 1, and it is indeed lovely, a cross between a piano and a seven-octave flute choir.

The illustrations and the recordings on the discs expand greatly on the columns.

Rhea’s essay on Leon Theremin’s various non-keyboard instruments is followed by no fewer than 25 pages of illustrations (including a high-school photo of Robert Moog playing his home-built Theremin), and then a column on the Ondes Martenot, which used very similar electronics but presented a completely different user interface, is accompanied by 18 pages of pictures. On the discs are three Theremin tracks (including a multitrack performance

by Theremin cousin and student Lydia

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Kavina of a piece by Percy Grainger), four Ondes Martenot tracks, and a demo of a 21st Century polyphonic descendant of the Ondes Martenot, the Haken Continuum Fingerboard.

The RCA Mark II, introduced in 1956, was the first digitally controlled synthesizer to get widespread recognition, if not use — only one was made and only a small handful of composers knew how to work it. Rhea gave it no fewer than six columns, in which he took the opportunity to explain sampling theory, binary numbers, how various electronic instruments approach the issue of portamento differently, and what is often lacking in the performance of electronic music. The corporate motivation behind the Mark II was not to create a new platform for experimental music, which is where it ended up, but to replace conventional

musicians and studios with a single machine for playing and recording music. The tracks on disc 2 hardly support this notion: they are laughably wimpy and sterile!

Many, many more instruments and their inventors are profiled. The brilliant and secretive Raymond Scott and his Rhythm Modulator, probably the world’s first sequencer, and his automatic composing machine the Electronium (which was to be adopted by Motown Records head Berry Gordy, who also had visions of replacing humans), gets five pages and three CD tracks. Oskar Sala and the Mixtur-Trautonium, used to great effect in Alfred Hitchcock’s horror film The Birds, is represented by four tracks. There are three columns about instruments that used light and photocells to produce music, and a total of 41 pages devoted to the evolution of the electric piano. And, of course, there are 24 pages about Robert Moog and his many innovations, from his first prototype to the Polymoog.

The final text entry is Rhea’s farewell column from June 1981, which for the first time talks about the role of computers in making music. “In my opinion,” he wrote hopefully, “the artistic — and commercial — success of the instruments produced in the next 10 years will depend more on the willingness of their designers to research and read existing literature on psychoacoustics and ergonomics (human engineering) than on startling technological breakthroughs.” But then came MIDI, and the musical instrument paradigm changed completely: the physical characteristics of an instrument no longer needed to bear any relationship to what it sounded like.

Ergonomics, for a while, took a back seat to making sure we had the latest sounds, but as the recent explosion in alternative controllers shows, they’re getting attention once again.

A Book Worth Owning?

Electronic Perspectives: Vintage Electronic Musical Instruments ranks as one of the most valuable resources ever about the history of electronic music. Its combination of friendly text, beautiful pictures and unique recordings make it both useful as a reference and highly enjoyable to peruse.

It’s not cheap, but it will live in your library for a very long time, and although it covers scores of obsolete technologies, it will never itself become so. $ $99 W electronicperspectives.com

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ON TE ST

iZotope Neutron 5

Can the improved, AI-based Mix Assistant make iZotope’s already powerful collection of mixing plug-ins more useful?

JOHN WALDEN

i

Zotope now offer several plug-in ‘suites’, including the Ozone mastering suite and Nectar, which focuses on vocal production. The third is Neutron, which comprises a range of tools aimed at general mixing duties. In the recent iterations of all three suites, iZotope have introduced AI-based ‘assistant’ features, which have the potential to help both those who are still learning the crafts of mixing and mastering, and those with more experience who perhaps need to be able to work more quickly.

As well as allowing their individual processing modules to be loaded directly in your DAW, all three products offer a ‘mothership’ plug-in that provides access to them all in one place. Ozone and Nectar were updated fairly recently (they were reviewed in SOS January 2024

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and April 2023, respectively), and now it’s the turn of Neutron, which has been updated to version 5.

New ’Tron Neutron 5 is available as the full version, or a streamlined Neutron Elements.

The latter provides less experienced music producers/mixers with access to Neutron’s key processing elements, through the AI-assisted Mix Assistant and a compact macro-style control set.

Inserted on a suitable track or bus, the Mix Assistant will audition the audio before offering processing suggestions that, hopefully, can improve the sound.

The Mix Assistant is also available in the full version of Neutron, but here the user can choose to switch to a Detailed View, and dig much deeper into the control set of Neutron’s individual processing modules. The full version also includes

Plug-in Suite

Neutron 5’s Mix Assistant can help you quickly find suitable processing options for individual instrument tracks within your mix.

a Visual Mixer plug-in, designed to help with creating an initial static level balance of your various tracks (or busses). And, as with the top-of-the-line versions of Ozone and Nectar, its component processors are also supplied as individual plug-ins that can be used outside the mothership plug-in.

With the AI genie now very much out of the bottle, it’s no surprise to see that the Mix Assistant features have been improved. From the user perspective, the most obvious demonstration of this is the revamping of the macro control set, which is now split across four panels called Tone, Dynamics, Saturation and Width.

In their documentation, iZotope refer to these as Intent Controls, which is a good description: both the panel labels and the controls themselves now make their effect, if not the processes behind them, much clearer for the novice user.

Once you move beyond the Mix Assistant’s Assistant View in the full version, the Detailed View includes three new modules: Clipper, Density and Phase.

The previous line-up remains, so you also have Equalizer, Compressor, Gate, Exciter, Transient Shaper, Sculptor (a ‘smart’ dynamic EQ to improve the tonal balance of instruments) and Unmask (to reduce unwanted frequency-based competition between specific instruments). Taken as a whole, this suite provides plenty of options for a whole range of mix-related processing tasks, but note that I said ‘processing’ — you won’t find effects such as reverb or delay here. As before, the Neutron 5 package also includes the very useful Tonal Balance Control 2 plug-in.

Clipper’s purpose should be obvious from the name, and it’s a useful addition to Neutron’s complement of dynamics processors. Density is particularly interesting: it uses upwards compression to raise lower-level parts of your audio signal, making it easy to prevent such elements disappearing in a busy mix.

Finally, Phase attempts to fix any phase issues you might have within your audio, whether that’s asymmetrical waveforms on an individual track, or time-based issues across multiple tracks, such as you might experience with multi-mic drum recordings. Essentially, it automates

what can be a complex task, and then lets you make manual adjustments if you feel some fine-tuning is required.

All modules now offer a very useful ‘delta’ option — the delta signal is the difference between the dry and processed sound, so you can use this to hear exactly what a module’s current settings are adding or taking away.

Finally, as in the latest version of Ozone, a number of Neutron’s modules offer both Mid-Sides and transient-sustain modes of operation alongside the normal stereo mode. For more experienced users, these alternatives might provide some interesting new processing possibilities.

Neutron Workflows The streamlined feature set of Neutron 5 Elements makes it very straightforward to use, but in the full version the user could settle on one of a number of potential workflows. First, as with Elements, you can employ the Mix Assistant on individual tracks or busses (or both) to offer initial advice on optimising those individual mix elements, before tweaking the Mix Assistant’s macro controls. But you could also use the Mix Assistant as a ’quick start’ option, and then use Detailed View to access the component modules that the Assistant added to the signal chain; as well as tweaking parameters, you

iZotope Neutron 5 $249 PROS • A powerful suite of plug-ins.

• Improved Mix Assistant is genuinely useful...

• But there’s plenty of control for those who want it.

• New Clipper, Density and Phase modules work well.

CONS • Ambience or reverb treatments could be helpful.

• Visual Mixer workflows could be better explained.

SUMMARY Neutron 5 is a powerful suite of plug-ins, but the refined Mix Assistant makes harnessing that power and potential incredibly quick and easy.

can change the module order or add/ subtract modules to suit your needs.

More experienced users might choose to bypass the mothership plug-in entirely, and simply construct their own signal chains to process different tracks and busses from the individual component plug-ins, just as you might with any other plug-ins in your DAW. Even without the Mix Assistant, the plug-ins are all very well featured, and they provide modern takes on both core (for example, EQ or compression) and more specialised (for example, spectral balancing, unmasking or phase management) mixing tasks.

Finally, Neutron 5’s Visual Mixer plug-in, which has its own dedicated Assistant, caters for a different aspect of the mixing process: level balancing and/or panning of your various tracks and busses. This is made possible by iZotope’s Inter Plug-in Communication (IPC) technology that allows iZotope plug-ins inserted on different DAW tracks to pass data to each other.

It’s used in Neutron’s unmasking analysis/ processing, naturally, but it’s also integral to the Visual Mixer and its ability to suggest appropriate mix levels for the various tracks/ Neutron 5’s module selection includes three new options — Phase, Clipper and Density – and as shown here, all three can busses within a project. also be used as individual plug-ins. www.soundonsound.com / February 2025

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Again, it’s easy to see the appeal of this option for less experienced users.

Mix Fixer Experimenting with each of these Neutron 5 workflows was an interesting experience for me. For example, applying the Mix Assistant to individual tracks is both easy and rapid, and in the vast majority of cases I was suitably impressed with Mix Assistant’s initial suggestions. The processing was, mostly, appropriately restrained, gently nudging each instrument in a suitable direction without overhyping anything, and I often found that the macro controls were more than adequate when I felt the results needed adjustment to deliver on my specific intentions for a given source’s role in the overall mix.

Perhaps unsurprisingly, I was generally happier with the Mix Assistant’s suggestions at the more granular track level (an individual guitar, drum, vocal, bass sound...) than when attempting the same thing at a group/bus level.

For example, while Neutron processed individual drum elements such as snare or kick very sensibly, I felt that its suggestions for a drum bus were less successful and, on occasions, a little too aggressive. The plug-in uses a relatively short analysis time prior to offering its processing suggestions, and I suspect that with a bus this window may not always be representative of the combination of instruments it contains, which can change through the length of a project. Having said that, though, even with just the macro controls I always found it pretty easy to rein things back in.

Clearly, if you were planning to use Neutron 5 on every track in a complex project, you would need to put in considerable configuration time, and there could be quite a load on your computer’s CPU, but used on a per-instance basis it’s not particularly onerous in either respect.

Finally, it’s worth noting that Neutron does introduce some latency. It’s designed for use while mixing, where this isn’t really an issue (your DAW will compensate for it automatically) but, should you need to record anything after you’ve started using it, it’s worth noting that the individual module plug-ins do offer a zero-latency option, albeit with some adjustments made to the processing.

Once you switch to Detailed View you get full access to Neutron 5’s signal chain and, as shown here for the Compressor module, sophisticated control options within each processor.

Moving beyond the Mix Assistant into Detailed View rewards you with a significant step up in terms of the amount of control you’re given. Inevitably, since the individual processors are both so powerful and so fully featured, there is a learning curve to travel if you’re to capitalise fully on the potential offered here. Less experienced users may, at first, find this a little intimidating, but it’s a journey worth undertaking. Taking the Compressor as an example, this offers three compression styles (Punch,

options for reducing artefacts, it provides clear visual feedback, and you can audition the delta signal, adjust the knee of the clipping process, and operate on the stereo, Mid, Sides and transient or sustain portions of a signal.

I found the new Density module particularly interesting. By boosting lower-level parts of your signal — those that fall below the threshold setting — it reduces the overall dynamic range, without any impact on the high-level parts of the signal, including transient peaks.

It’s a great additional option to have and can help avoid some of the tedious manual volume automation that can be otherwise required to ensure that quieter sections of a track are not lost within the mix. Again, it is well specified with up to three bands available, delta monitoring and good visual feedback, stereo, Mid/ Sides and transient/sustain modes, and control over the upward compression threshold, maximum applied gain, ratio and speed of attack. Upward compression can, in theory, raise any noise in the signal along with the wanted details, but it’s not something I noticed being a problem in this implementation, so I wonder if there is something clever happening behind the scenes to prevent that?

“iZotope’s Inter Plug-in Communication technology... allows iZotope plug-ins inserted on different DAW tracks to pass data to each other.”

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Modern and Vintage), up to three-band operation (with each band featuring an independent set of threshold, ratio, attack, release and mix controls), as well as an overall mix control, internal or external side-chain support, three different input level detection options (RMS, peak and true peak), real-time visual feedback on the gain reduction, knee adjustment and automatic gain adjustment, and it can be used on mono, stereo or surround sound channels.

All three of the new components/ plug-ins allow a similar level of detailed control. For example, the Clipper also offers up to three-band operation, and features a useful Learn function that can suggest appropriate band crossover frequencies. There are oversampling

Balancing Act What about Neutron’s other ‘assistant’ role in setting your overall mix balance?

This option is built around the Visual

imagine those newer to the mixing process finding this a less intimidating starting point, while they’re still becoming familiar with the ins and outs of their DAW’s virtual mixing console.

Is Neutron The Bomb?

For those new to the mixing process, the Visual Mixer provides a very appealing way to approach initial level and pan settings for the tracks within your mix.

Mixer plug-in and IPC functionality mentioned earlier. It has been part of Neutron for several iterations, and Paul White described the process in some detail when reviewing Neutron 3 in SOS September 2019. In essence, the Visual Mixer offers two key features.

First, after an ‘audition’ process, during which it analyses the signals on different tracks, it can create an initial static mix balance for your project. That could be helpful for even experienced mixers when faced with making sense of a big project quickly, but the concept will no doubt appeal even more to less experienced users. I’m not sure, though, that the current documentation (or the Neutron 5 learning resources on iZotope’s website) really makes the required workflow clear, in terms that such users would find particularly accessible. Yes, if you have a grasp of mixing console concepts you can probably work out a number of ways to approach Neutron is available in bundles with iZotope’s Ozone and Nectar, and together they form a very impressive mixing and mastering toolkit.

the process. But if you know your way around a mix console already, then arguably you stand to benefit less from the Visual Mixer in the first place. More generally, iZotope are exceptionally good when it comes to the provision of online learning resources to support their products, so I’m sure this is something they could easily address with another ‘how to’ video or two.

Second, the main Visual Mixer allows you to adjust the level and pan of the individual elements in your mix manually, in the rather neat graphical display. This is a side of the Visual Mixer that I found very appealing, and I can certainly

Neutron 5 is definitely a worthwhile evolution of the Neutron concept.

It has the potential to both improve the experience for novice mix engineers and offer useful processing options that should appeal to more experienced users. You could consider it simply as a very capable collection of sophisticated processors that you can use to do general mix processing, from tonal shaping to more surgical work on almost any element in your mix. With both routine tasks (EQ, compression, and so on) and more specialist ones (for example, spectral balance, upward compression, unmasking) catered for, there’s plenty to get your teeth into. The new processing additions (Clipper, Density and Phase) all bring something useful to the table in this context too, and judged on this basis alone, it’s perfectly possible to argue that the Neutron 5 suite offers good value for money.

Alternatively, Neutron 5’s Mix Assistant offers less experienced users access to the same powerful processing, but in a more user-friendly fashion, and I found it genuinely impressive. As with Toontrack’s EZmix 3, it provides an accessible entry point to the often puzzling world of mix processing, and novices should be able to learn plenty from what the Assistant is attempting to do.

If you find the detail of the full version intimidating, Neutron 5 Elements is a low-cost option that can get you started, and there’s an upgrade path from that to the full version that you might consider as your confidence grows. The improvements to the Mix Assistant also make a strong case for existing Neutron users to upgrade to v5.

We can only really guess at where AI will eventually take ‘assisted mixing’. But today Nectar 5 is as clever, competent and accessible as it gets, while at the same time being deep, surgical and powerful — if you want it to be. $ Neutron 5 $249, Neutron 5 Elements $49.

Also available in various iZotope bundles.

Upgrade discounts available.

W www.izotope.com

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KIT Plugins BB F66 & BB F67 Compressor Plug-ins There are now quite a few plug-ins that emulate the famous Fairchild 660 and 670 compressors but, as with their other recent creations, KIT Plugins have based theirs on specific units in John McBride’s Blackbird Studios in Nashville — in this case, they’ve modelled the units used for, amongst other things, Martha McBride’s vocals. Authorisation is via iLok Cloud or iLok 2/3 dongles (computer hardware authorisation isn’t permitted), and all the usual plug-in formats for Mac and Windows hosts are supported, but for 64-bit systems only. Curiously, the download for my Mac-based system was pretty large (almost 1GB), though the installed plug-in takes up only about a third of that space. Other than that, installation was hassle free.

Two separate plug-ins are installed: the BB F66 (a 660 emulation) and the BB F67 (a 670 emulation), and with both channels always ‘in view’ for the latter you’re never in any doubt as to which you’ve loaded. Starting with the ‘single-channel’ BB F66, this can be used as a mono, stereo or mono-to-stereo plug-in. All the anticipated Fairchild controls are there, with continuously variable input gain and threshold knobs joined by a five-position time-constant switch, a virtual moving-coil meter that can be set to display the input or output level or gain reduction, and an on/off switch. Despite the vintage look, there are nods to modernity here: a panel at the bottom hosts controls for a 0-500 Hz side-chain filter, a wet/dry blend facility and up to ±20dB of make-up gain. At the top, you have separate, clean (ie. proudly digital and not modelling any hardware) input and output level controls, along with buttons for undo/redo and A/B comparison, as well as a generously stocked preset menu, with settings created by various famous names. The GUI is resizeable to small, medium or large, with medium being the default.

The BB F67 can only be used on stereo material, and it duplicates the main compressor controls of the BB F66, one channel appearing above and the other below the panel that hosts the side-chain filter. As well as the other controls there, this middle panel on the BB F67 includes a stereo mode selector: in Linked mode, the top channel controls apply to both channels, in Stereo mode

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the two channels’ control sets are unlinked, while Mid/ Sides mode implements the Lat/Vert setting of the hardware, with the top channel manipulating the Mid signal and the bottom one the Sides.

So it’s all familiar enough, but what about the sound?

Of course, with Fairchilds The BB F66, which can be used on mono or stereo sources. having been around for three quarters of a century now, and individual units having aged or been modded, individual specimens will vary, not least with things like the attack and release times. KIT say the settings they measured on these treasured hardware devices differed from those given in the manual. Still, I thought I’d compare both plug-ins on a range of material with some of the other Fairchild emulation plug-ins I often turn to. I started by exploring and tweaking some presets The dual-channel BB F67. which seemed to have been duplicated in the BB F66, with two entries an effect that I’d choose — I just couldn’t for each setting (I’m sure that’s a bug that get them to sound quite as upfront, solid will be ironed out quickly), but there are or alive as the equivalent plug-ins I’ve some excellent starting points for those been using recently, even with the help not used to a Fairchild interface, even of the side-chain filter. To be fair to KIT, if you obviously need to tweak settings I’ve also heard much worse, and I should to suit the material you’re working on. probably point out that one of those The (deliberately) limited menu of attack other plug-ins is based on a hardware and release pairings means that on the recreation that itself isn’t quite a direct right material, you can often simply adjust copy of a vintage Fairchild, while the other the top strip’s input and output level uses a different technology to capture knobs to get the signal in the sweet spot the hardware sound (and although that for a given preset. one sounds great I do find its GUI clunkier There’s definitely a Fairchild-ish sonic to use!) character on offer here: a sweetening Do these two compressor plug-ins of the sound and a bit of an EQ curve sound like Martha McBride’s Fairchilds? applied even when there’s no gain Well, some big-name engineers who’ve reduction. When actually compressing, worked at Blackbird — and have enjoyed they both performed admirably on a range far greater success than I ever will — are of sources. On sounds with some sustain, quoted as saying so. Obviously, without and on female vocals in particular, it access to the Fairchilds at Blackbird, could sound effortless and seductive, I can’t vouch for that, but what I can say and I suppose that shouldn’t come as is that they’re certainly capable of doing a surprise given the units that they are some sweet-sounding things to vocals, modelled after. But it delivered the goods and have the potential to be equally when it came to smooth control of bass pleasing on most — if perhaps not quite and acoustic guitar, and served decently all — other material. Matt Houghton on the mix bus of an Americana track. $ Perpetual license $99. Also available Used on a drum bus, the KIT plug-ins through KIT’s Noiz Hub subscription, seemed to pull the image backward in the $9.99 per month or $99.99 per year. sound stage a touch. Not terrible, but not W https://kitplugins.com

JMG DarkFire Saturator Plug-in Authorised by a personal code that permits use on multiple machines, JMG’s DarkFire (AU, VST, AAX) is a harmonic enhancer whose added harmonics can be shaped by a three-band EQ to put them precisely where you want them.

The input is processed by variable highand low-pass filters with switchable slopes, before being sent to the saturator, which, if the Advanced button is clicked, can be set to Tube or Tape.

A large Input Drive knob sets the level before the saturator, so determines the degree of saturation.

Harmonics Level then sets how much of this enriched signal is mixed back in alongside the dry sound. The Harmonic Shaper is a straightforward three-band EQ, with two frequency options for each band. The Press switch applies compression prior to saturation, so lower-level signals still receive some saturation.

A lower row has knobs for Comp, Stereo, Depth and Mod. Comp

“Concentrating the harmonics in the high end produces more of an exciter-style effect.”

is modelled on classic analogue compressors, with FET and VCA options. Stereo is a width adjuster, ranging from mono to 100 percent, which can be applied to the full signal or only the high frequencies. As well as widening/narrowing stereo material, it can generate pseudo-stereo content from a mono source. Depth generates multiple short delays to create a psychoacoustic sense of depth, while Mod acts in a similar way to a chorus.

Further high- and low-pass controls, again with switchable slopes, are available to process the output.

A Listen button mutes the dry signal so you can hear the harmonics in isolation, and spectral displays of the input and output reveal useful information on where harmonics are being added. Other buttons bring in a bypass, an automatic gain compensation function (making the output level nominally the same as the input level) and a switchable limiter that avoids overs at the output.

There are also the usual options to compare A and B settings, undo/redo, and load or save settings. A small library of presets covers all the main applications, and tweaking them is easy.

In my view, DarkFire straddles the disciplines of saturator and exciter, and I found it very effective in beefing up bass and drum sounds in

particular — as long as you don’t set the Input Drive too high, the added harmonics tend to sound smooth and natural. Concentrating the harmonics in the high end produces more of an exciter-style effect, which can be great for adding definition to things like snare drums and guitars, as long as you don’t overdo it.

There’s plenty of choice now if you’re looking for saturator plug-ins, but this one is very easy to use and, whether you need to add a bit more bite to the highs or warmth to the lows, it can be surprisingly effective.

There’s a demo mode, and it’s worth checking this one out for yourself.

Paul White $ $72 (Introductory price of $16 applied when going to press)

W https://unitedplugins.com

Pulsar Audio Poseidon EQ Plug-in The puntastic name and the faders on the GUI of Poseidon leave you in little doubt as to which famous console EQ it sets out to emulate, but there’s more to this thing than mimicking the Trident A-Range. As with other Pulsar EQs I’ve used and, it has to be said, enjoyed, there are various thoughtful ‘utility’ features. Along the top are the usual undo/redo buttons, a preset manager and A/B compare buttons. You also get a GUI view selector, with three options for the main display: you can show only the pane containing hardware-inspired knobs and faders; only the pane with a modern ‘nodes on a curve’ workflow, which includes a frequency analyser and metering; or you may show both at once, which is the default view. A three-dots icon accesses more settings, including up to 8x oversampling and the ability to

switch the EQ frequency controls from stepped to continuous as you prefer, with options for the resolution of those steps too, which is a lovely touch.

The signal path starts with a switchable Drive control that governs the amount and nature of harmonic distortion (aka saturation) imparted by the plug-in. There are two settings for this

(Trident or Poseidon), as well as off, and Drive also influences the contribution of the modelled output transformers. You have a choice of two transformers (or neither), one of which is modelled after the famous Marinair transformers used

by Neve in years gone by.

In between these stages comes the EQ section itself. As you’d expect of www.soundonsound.com / February 2025

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a Trident-style EQ, this comprises four main bands (high and low shelves, with two bell bands in between) and these are sandwiched between high-pass and low-pass filters.

Starting with the hardware-esque control panel, the frequencies you’d expect to find are here but, since the controls can be continuous (and are so by default), you can set them to anything within their range and those ranges have in some cases been extended too, for greater versatility. The high-/low-pass filters are controlled with a vertical fader, while the main bands’ gains (±15dB) use horizontal faders. You can also switch each band on/off, and choose whether it operates on the stereo signal, or just the Mid or Sides. (There seems to be no option to EQ only the L or R signal; not something I missed personally, but some might.) Holding down Command on a Mac (presumably Ctrl for Windows users) allows you to make finer adjustments with the mouse.

At the end of the chain comes an output level knob, with an auto-gain button that allows you to keep the overall signal level consistent when making EQ adjustments. That could be useful for mastering and stereo mix-bus work, where ‘louder is better’ means boosts can otherwise skew your perception, even if it’s not usually something I like to use on individual sources in a mix (where I like to hear the result of my EQ moves in context). Another neat feature is the Scale knob, which allows you to shrink or magnify the overall EQ curve. This can be really helpful for fine-tuning when all four bands are active; rather like a wet/dry blend in practice, but without the risk of phase cancellation coming into play.

On the curve-plus-analyser pane, you can manipulate the same settings using nodes on the curve. A hidden feature in this view is a band solo: hold down Shift and click on a node, and you hear only the frequencies on which that band is operating. There are also controls to refine the behaviour of the frequency analyser, and level meters that show peak and RMS for the input and output.

While this ‘curve panel’ operates in a familiar way, it’s important to note that it’s not controlling a regular parametric EQ, but rather manipulating the controls of the emulated Trident EQ. So, for example, there are no bandwidth controls, and each band’s frequency range is limited. You’ll soon notice that the bands don’t all exhibit

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“Hold down Shift and click on a node, and you hear only the frequencies on which that band is operating.” quite the same characteristics (I couldn’t set gains on the two central bands at the same frequency and fully cancel, for example) and the bands seem to be somewhat interactive. All of which means that if you’re unfamiliar with Trident EQs, you may at first find that they don’t do quite what you expect. Happily, the results are usually very musical-sounding, and with a little practice you start to find that it’s very quick to dial in characterful, broad tonal shifts on a sound, with fewer clicks and drags than on a regular digital EQ. I have to say that Pulsar seem to have done a decent job of capturing this complexity and interaction, just as they did on their model of the Massive Passive that I reviewed a couple of years ago. It’s not all about the curves, though: there’s definitely a sound to this thing, and between the input and output stages, you have access to a surprisingly broad saturation palette that can add a lovely character to vocals and guitars, lend thickness to a bass guitar or synth part, and even make kicks squelch. It’s tricky to convey in words the differences in distortion between one plug-in and another, but unless you go deliberately over the top, charm, subtlety and smoothness are some terms that spring to mind.

All in all, then, it’s another great-sounding EQ model from Pulsar Audio, and one that sports one of my favourite GUI designs for analogue-modelling plug-ins: it’s a clever interface that should please both those who like skeuomorphism and those who prefer the visual feedback of digital EQs. The world and your DAW may not be short of EQ plug-ins, but this one’s a little different. If you’re after a Trident-style EQ, or just enjoy hardware-modelling plug-ins generally and want to upgrade your go-to ‘channel EQ’, this one is doesn’t cost the Earth given its sound and feature set, and the free trial would be well worth auditioning. Matt Houghton $ €99 (about $103), discounted to €59 () when going to press.

W https://pulsar.audio

MXR Rockman X100 Analogue Amp Simulation Pedal The original Rockman series started life around 1982, and included models both with and without an onboard echo effect. Originally designed as a headphone practice tool, the Rockman X100 offered four preset sounds, plus chorus and echo effects and battery operation. It was designed by Tom Scholtz to recreate his guitar sounds from the band Boston, and was really the precursor to the sophisticated amp modellers we see today. There was no variable drive control so, other than selecting one of four presets, the only control you had was over the output level, and even that was simply switched in three steps. The chorus and delay were created using analogue delay lines, while the guitar sounds themselves also benefited from some compression (again non-adjustable). The chorus and delay effects could be used together or you could select one or the other (though you couldn’t switch both off, which I recall being annoying!) and, again, there were no adjustments. Despite the limited user control, these pedals had a sound of their own that won them plenty of fans.

Having acquired the Rockman brand, Jim Dunlop set about recreating the Rockman X100 under their MXR brand, but this time in pedal form and with a bypass footswitch. The use of

surface-mount technology made it possible to cram the necessary circuitry into a standard pedal, which now features input and output jacks plus a Ctrl jack for hands-free stepping through the four presets. A button and four status LEDs are used to navigate through the presets, which are designated Clean 1, Clean 2, Edge and Dist.

While the Rockman sound it delivers is familiar, and seems accurate, the feature set is actually slightly different, and there’s no battery power option or headphone output. The only effect available this time is chorus and, like the original, this is preset, but it can now be bypassed using a button, which is helpful. I asked about the delay, and was told that to recreate the vintage BBD circuit accurately would have upped the price considerably. That’s fair enough, I think; there are plenty of separate delay pedals available if that part of the sound is important to you.

Other useful additions not present on the original are sliders to control the input gain and output level, which is again helpful, even if the input gain setting seems to have little effect on the amount of overdrive that’s added.

As this pedal is designed to sound like an ’80s unit, it comes as no surprise that the sounds have a definite retro character to them, but then recreating nostalgic Rockman sounds is the whole point! Clean 1 provides a generic clean sound, while Clean 2 is slightly brighter and less full in the lows, and both include a little compression. Edge gives a raunchy rock tone that can be dialled back using the guitar’s volume control, and it responds fairly well to playing dynamics. Dist is a little more aggressive but neither stray into heavy metal territory. As with the original, here you can hear a slight background hiss too, but it’s nothing too obtrusive.

The pedal works fine plugged into a clean guitar amplifier, and in that scenario you can use the amp’s own tone controls if you want more tone-shaping

“The sounds have a definite retro character to them, but then recreating nostalgic Rockman sounds is the whole point!”

options. Used for direct recording into a DAW, the clean sounds work well with the integral compression, giving them a ringing sustain, and the chorus sounds sweet too. Both distorted sounds are a little raw-sounding compared with what today’s modellers can give us, but they’re not unappealing, and they definitely evoke that ’80s Boston sound.

If you hanker after those ’80s Rockman tones without paying silly money for a used original, you need look no further. Paul White $ Price not announced when going to press.

W www.jimdunlop.com

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Tackling Noisy Guitar Recordings Are buzzes, hums and hisses ruining your guitar recordings? Here’s how to keep the noise to a minimum.

PAUL WHITE

F

or as long as we’ve had electric guitars, guitarists have had to grapple with unwanted noise making its way into the guitar sound. When

playing electric guitar live, hum, buzz, fizz and hiss are certainly not ideal, but you may still feel that a little of it is acceptable.

But recorded performances will be heard over and over again, so such noises are likely to be much more irritating. This article takes you through the main causes of electric guitar noise, offers some solutions, and should also help you make the best of things and manage expectations in those situations where at least a little noise is inevitable.

Guitar MOT

A cheap multimeter allows you to check that there’s a zero-resistance path between the guitar’s hardware and its output jack .

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Lots of clever plug-ins attempt to address noise after the fact, but they can’t take care of everything and the more you ask them to do, the greater the side-effects. In short, the best results always come from eliminating or minimising annoying noises before your record. There are several potential issues that can cause and exacerbate noise problems, and some are more obvious than others, but the best place to start is probably the guitar itself.

The first thing to check is that there’s a zero-resistance path between the guitar’s hardware and its output jack. To check this, use a multimeter set to Ohms, plug in your regular guitar cable and measure the resistance between the strings and the barrel of the jack plug at the amp end of your lead. The meter should read very close to 0Ω. By the way, checking in this way rules out problems with the guitar lead, which is helpful — if there’s a problem, change the lead; if it goes away there’s your issue, if it doesn’t, the problem is with the guitar...

Usually, the guitar’s bridge is grounded, and this means that the strings are also grounded. Grounding is normally handled by a length of wire inside the guitar that, in the case of a Strat-type guitar, links the vibrato spring claw to the ground wiring on the control panel. The backs of the pots are common termination points for ground wires, as there’s plenty of area for soldering and most wiring regimes will ground these parts, either using wires or via a metal-foil lining to the pick guard, through which all the controls are fitted.

A less immediately obvious potential cause of poor grounding is something I experienced when Even slight corrosion on the guitar’s jack could be the culprit — a quick squirt of Deoxit should address the problem.

playing a guitar with ‘noiseless’ pickups (stacked humbuckers that are designed to sound like single coils) at a noisier than usual live venue that had lighting dimmers. As you may know, these produce radio-frequency emissions for which the pickups in your guitar act as antennae. My guitar wiring was all good, the noiseless pickups were Fender stacked humbuckers, and I had a really good guitar cable but I was still picking up a huge amount of interference. My amp was properly grounded too, so where was all that buzz coming from? The culprit turned out to be a slight layer of corrosion on the grounding prong of the guitar’s output jack. This added resistance to the ground path, which can be identified as outlined above.

A quick clean and a spray of Deoxit D5 soon had me sorted.

To get the best from noiseless pickups, it’s also a good idea to screen the control cavity of the guitar using either conductive paint or self-adhesive copper strip, all of which should be interconnected and linked to the guitar ground.

Another option is to fit active pickups.

These tend to have a lower impedance, which makes them less susceptible to induced electromagnetic interference, and their active circuitry matches the pickups to your amp or audio interface. However, many players are wary of them, partly because they need power (typically from either a 9V battery or a USB chargeable lithium battery), and partly because they may not sound the same as the passive pickups to which they’re accustomed. Also, it’s often necessary to partly disassemble the guitar to change the battery if the pickups are a retrofit. So there are pros and cons, but they can be effective.

you start to play. You’ll often find that your recordings then sound much cleaner than before. If they do, perhaps consider using some sort of remote control option to operate your DAW from a more distant recording position.

You may also have noticed that when the source of unwanted hum is a nearby mains transformer (for example the one in your amp), rotating your playing position also changes the level of any noise. In such cases, a common and often effective tactic is to rotate your playing position until you find a ‘null’. Experiment to find the sweet spot at which the noise is lowest, and then stick to it while you’re recording.

Earth Calling

Shielding a guitar’s circuitry with copper tape can sometimes help, but not as much as you might hope — the hundreds of metres of copper in the pickups are more likely to be the source of your woes!

Location, Location Although screening the inside of the guitar can help, if you don’t have active, traditional humbucking or noiseless pickups, it will probably make less difference than you hoped. Why? It’s simply because the half-metre of unscreened wire linking your pickups to the controls is not the main problem: the real culprit is the several hundred metres of wire wrapped around the magnets inside your pickups, which has been configured in such a way as to respond to the smallest changes in magnetic field.

Single-coil pickups can be arranged to noise cancel when they’re used in pairs — they are, for example, in positions two and four of a modern Strat’s selector switch. But

There are now some very good ‘noiseless’ pickups designed to look and sound like single-coil types.

when used singly they’re very susceptible to stray magnetic fields, especially from mains transformers, lighting dimmers or basic wall-wart power supplies.

It’s convenient to sit close to your computer when recording isn’t it? After all, this gives you easy access to things like your computer keyboard and your DAW’s transport controls. But sitting close to a computer often results in excessive noise pickup, even with humbuckers. The first thing to try if this is an issue is, just after hitting record, moving just a couple of metres away from the computer before

For a grounding scheme to work, there should also be minimal resistance between the guitar’s ground and the mains electricity ground in the building — and a problem might not be on the guitar side of that equation. In practice, building grounds aren’t always as good as they might be. If the soil around the grounding spike dries out, for example, then the ground resistance will go up, which means any grounding-related noise problems will likely grow worse. A simple solution to that state of affairs is to empty several buckets of water around the ground spike to moisten the soil — though note that it may take a long while for the water to soak down to the desired depth.

Another situation I often encountered during Studio SOS visits in days gone by is where the guitar is correctly wired, the cable is of good quality and even the building ground is also fine, yet there’s still a lot of unwanted noise. This can easily occur in studio systems that are built around a laptop, in which all the connected equipment (audio interfaces, small monitors and so on) is powered using either external power supplies or two-core mains cables, with no ground wire. If nothing is directly grounded, yet everything’s connected to each other, then the entire recording system can be left ungrounded, and that may leave you with noise problems.

Thankfully, this common problem is generally very easy to fix: you just need to run a grounding wire to the mains ground from a piece of equipment that has a metal case. You can do this using a special grounding mains plug that has plastic prongs for the live and neutral connections (this avoids any nasty shocks should the ground wire get pulled accidentally out of the plug and touch the live conductor). www.soundonsound.com / February 2025

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a single point, is probably the best you can do without resorting to exotic studio wiring schemes.

It’s not perfect, but it definitely reduces the risk of ground-loop hum. (By the way, please never be tempted to remove ground connections from a mains lead to cure hum, as that increases the risk of electric shock!)

When you need to plug lots of gear in, it’s generally better to configure your distro boards as a star, fanning out from a single socket, than to simply daisy-chain them.

Skimping on pedal power supplies can be yet another source of noise. Simple, unregulated pedal power supplies may be fine for the odd low-current pedal, such as an analogue overdrive that isn’t too fussy about the actual voltage it runs from. But some PSUs, especially cheaper, poor-quality switch-mode supplies, can put noise back onto the mains — this can manifest itself as a whine (as opposed to hum or hiss) and may affect other devices plugged in close to the offending PSU. (Note that switch-mode power supplies aren’t inherently a bad option; I’m talking specifically about cheap ones here!)

In any event, you should avoid using those cheap daisy-chain pedal power cables to supply multiple pedals from one PSU, and that’s especially the case where one or more of your stompboxes is a digital pedal. These can become noisy if sharing

supplies with other pedals, even if there seems to be plenty of current available.

You can recognise this by a faint whining sound in the background, and I suspect it’s because some of the digital clock noise makes it back out through the PSU. As analogue pedals require much less current than typical digital pedals and also tend to draw a consistent amount of current, they tend to be less problematic.

By far the best option is to buy a good-quality, multi-output regulated supply with fully isolated outputs — ideally ones that don’t share a common ground, and where each output has its own voltage regulator. These are not ‘cheap’, but relative to a big pedal collection they’re inexpensive, and they will allow your pedals to give of their best. Note, though, that some cheaper PSUs claim to have ‘isolated’ outputs but actually will only have separate output regulators and still share a common ground. With such PSUs,

Some studio systems based around a laptop and external PSUs may need to be separately grounded — something that’s easily and affordably done with an earth-bonding plug.

Cheap pedal power supplies can also be a cause of problems — and the same goes for daisy-chain power cables like the one above.

Pedal Pushing As the name implies, a ‘grounding spike’ goes in the ground, but if that ground becomes too dry it may be the root cause of unwanted noise.

If there’s no exposed metal to which you can connect this ground wire, you could instead ground the barrel of your guitar lead’s jack plug — the most elegant way to do this is to make up a very short cable with a jack plug at one end and a jack socket at the other so you can attach your grounding cable to the ground side of the jack socket. Put this in series with your guitar cable and it’s job done.

Look To The Stars We’ve covered studio mains wiring lots of times, but it’s worth revisiting the concept of ‘star wiring’ here, because this approach reduces the likelihood of ground loops, which are yet another source of hum and buzz. Most studios, even simple home setups, require a relatively large number of power sockets, and it may seem tempting either to plug into different power points in different parts of the room, or to ‘daisy-chain’ a number of distribution strips. But both approaches leave your system open to ground-loop issues.

For a more detailed explanation of this, check out Hugh Robjohns’ excellent Ground Control article in SOS August 2023 (https://sosm.ag/ground-control), but in practice, plugging a single distribution strip into the mains and then plugging further ones into that first strip, so that the various sockets effectively ‘fan out’ from

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It’s possible to use software to tackle many problem noises, such as hum (being removed here using iZotope RX), after recording, but the more you ask of such software, the more audible and unacceptable the side-effects will become.

“When using a compressor, 10dB of gain reduction typically means 10dB of make-up gain, making any noise 10dB louder!” noise issues can still ensue, and unless you have fully isolated outputs, you can’t use a polarity invert cable to power the odd pedal that needs a pin-positive rather than pin-negative supply. If you use many high-current digital pedals, then you also need to ensure that your prospective power supply can provide the requisite number of high-current outputs!

More Gain More Pain It’s a fact of life that most guitar players like to add a lot of gain to their signal, either when using a compressor (10dB of gain reduction typically means 10dB of make-up gain, making any noise 10dB louder!) or when using some form of overdrive. All overdrives work by boosting the signal and then squeezing it through a circuit with deliberately limited headroom, so that the signal gets clipped. The clipping may be harsh or smooth depending on the type of pedal being used, but the more gain you pile on, the more any residual noise (including that generated by pedals earlier in the chain) will be boosted with the wanted sound.

Modern metal drive tones tend to lie at the extreme end of the gain

spectrum, and you’ll find that the noise can be very significant, even when using a well-sorted guitar fitted with humbuckers.

In that scenario, using a noise gate or expander to keep pauses between notes or phrases silent may be the only solution. The only solid tip I can offer here is to place the gate/expander before any added reverb, delay or ambience. That way, the effect tails will tend to cover any unduly abrupt note endings caused by heavy gating.

Fixes in the Mix? There are numerous plug-ins that can reduce noise in recordings, and they can often produce good results as long as you have realistic expectations. If you’ve made every effort to keep your recording as free from unwanted hum and noise as possible, then they can really help to finesse the end result. To combat hum and buzz, there are plug-ins that create very narrow notch filters at the frequencies corresponding to the local mains supply (50Hz or 60Hz, depending on which country you live in) and to the harmonics of that frequency.

More elaborate ones may actually track the mains frequency. If you use these to

reduce the level of hum by around 10 to 15 dB, you should find a good compromise between noise level and the tonal integrity of the overall sound.

It’s a similar story when it comes to broadband noise removal plug-ins. The more basic ones need a noise-only sample or ‘fingerprint’ to work with and only work very well if the level of noise is reasonably constant. If you’re recording yourself and plan on using this sort of processing, just make sure you record some ‘silence’ before or after your part so you have somewhere to take your noise-only fingerprint from. Again you can claw back maybe 10dB in this way before side effects become apparent. There are some more elaborate plug-ins that can adapt to changing noise levels but, as with gates, make sure you apply any de-noising before adding delay-based effects.

Obviously, that’s trickier if those effects are part of your pedal chain — if you have access to a decent re-amping kit (ie. one that doesn’t change the sound), then it could make sense to record a clean DI signal and re-amp it. This would allow noise gates, de-noisers and the like to be fine-tuned after recording. Note, though, that the effect of your pedals would still need to be audible while recording, as that affects the way you play — creating a split at the start of your chain, so that both clean and effected parts are recorded, would be a good plan. Even if the track recorded with effects isn’t used, it would serve as a reference when tweaking the re-amped sound.

A Lot of Noise About Nothing?

Finally, while it’s good to tackle unwanted noise at source by ensuring you have a good guitar cable and power setup, I would urge you not to stress too much about it — by which I mean, don’t feel you have to get rid of every last shred of noise. Some noise is an inherent part of the driven electric guitar sound, and in most cases it will be masked by whatever else is happening in the mix.

More exposed sections can often then be treated using a combination of noise/ hum reduction and level automation.

It can also be helpful in the case of high-frequency noise on a note with a long decay to automate a high-cut filter so that it closes as the guitar note decays.

Some engineers even choose to actively embrace some noise, because they find it adds a sense of energy to certain more aggressive rock styles! www.soundonsound.com / February 2025

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ROBIN VINCENT

O

ne of the most interesting and yet slightly under-the-radar new features in Studio One 7 is the CV Instrument plug-in. It sits very unassumingly in the instrument plug-ins list, and is easily ignored. For modular fanatics such as myself, who enjoy nothing more than complex interconnected workflows, it is a source of exciting Studio One-to-Eurorack action.

However, there seems to be very little information out there on how to use it. As I mentioned in my Studio One 7 review, the manual is factual rather than instructional; hopefully, I’ll be able to help with that. So, couple your DCs, grab some patch cables, plug in some mini-jack adaptors and let’s get Studio One talking to your modular.

Studio One With the introduction of CV Instrument, you can control your modular rig from Studio One!

Nuts & Volts CV Instrument is a virtual instrument plug-in that can send note pitch and gate information out of Studio One and into your Eurorack synthesizer in a form it will understand. That form is CV, or Control Voltage. The pitch of a Voltage Controlled Oscillator (VCO) is... controlled by voltage, which means that as you change the voltage, the pitch or frequency of the oscillations changes. As standard, a change of 1V equates to an octave shift in pitch. So, CV Instrument takes a standard MIDI note from your MIDI controller or sequence and converts it into a voltage to send to your VCO. Easy, right? The process is easy enough, but the physicality can be a bit of a struggle.

CV happens along the same pathways as audio signals. You’re not sending MIDI or other digital data between machines, but varying voltage signals down jack

cables. CV is the same as analogue audio, except it’s usually moving too slowly for us to hear. To get CV out of our DAW, we will need to use an audio interface — more than likely the same audio interface you’re using for audio. However, many audio interfaces have a high-pass filter built into the signal path to remove exactly the sort of voltages we are trying to generate.

This ‘AC coupling’ can have benefits for audio applications, but makes the outputs useless for CV.

Before you can send CV to your modular from CV Instrument, you need to tell it which analogue outputs on your audio interface to use.

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The new CV Instrument plug-in will take your MIDI data and output it as control voltages and gate signals through your DC-coupled audio interface.

So the first piece of the puzzle is that your audio interface must not be AC-coupled; it must be ‘DC-coupled’ so that it can let slow-moving or stable voltage signals through. We’re only going in one direction, from Studio One to our modular, so our audio interface only needs to have DC-coupled outputs.

Fortunately, this is becoming more common in modern audio interfaces.

Most interfaces from PreSonus, MOTU and RME are DC-coupled, and you should find it mentioned in the specifications of other brands and models. However, not all interfaces are the same, and you may find that some will work better than others. The Komplete Audio 6 from Native Instruments, for instance, is DC-coupled but can only output a range of 2V, and so won’t give you more than two octaves of notes.

My audio interface is the excellent SSL 12, which has DC-coupled outputs, but for reasons I can’t yet explain, it refused to work properly with CV Instrument, whereas it works fine with

the very similar HW CV Instrument device in Bitwig Studio. I swapped the interface for the MOTU M4, and that’s working fine. My point is that if you encounter problems getting CV Instrument to work properly, it’s likely to be a problem with your audio interface.

Connections Usually, you’d need two DC-coupled outputs for each instance of CV Instrument, although it depends on what you’re doing. So, sticking to the most likely scenario of running a sequence to a VCO while triggering an envelope patched to a VCA or filter, you need one output to carry CV pitch and one to carry gates or triggers.

It’s vital that you remove these outputs from your mix, as DC voltages are not going to do your speakers any good. Check any monitoring settings in your audio interface’s software mixer if it has one.

In Studio One, you can set up the outputs in the Song Setup page and label them up all nicely. However, CV

For pitch to track correctly, you must calibrate CV Instrument using a steady tone (a VCO in your modular is an excellent choice). Connect it to an input on your interface, and select that input in the Audio In menu within CV Instrument.

Instrument will do this for you, so you can hold off on this little bit of studio routing management for now.

To make the right connections, I’m taking two nice long Eurorack patch cables, putting small-to-big jack adaptors on one end, and plugging

them into outputs 3+4 on the back of the MOTU M4. I’m plugging the other end of the cable from output 3 into the 1V/oct input on my VCO to control the pitch. The cable from output 4 is being patched into the trigger or gate input of an envelope module, which is patched to open the VCA that the VCO is going through.

The idea is that when a note is pressed or sequenced in Studio One, it will

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Studio One activated. S-Trig is the opposite and was used in vintage Moog Modular systems.

You’ll also find it on the Behringer Eurorack Moog Modular clones.

If you’ve made the right connections, you should be able to play your MIDI keyboard and CV Instrument will change the pitch of the VCO and open the VCA. The sound will, of course, be coming out of your modular, not Studio One or your interface. You also don’t need to use the CV Gate output and a VCA; you could have the VCO droning and simply use CV Pitch Out to change the pitch. Modular is nothing if not versatile.

Calibration

Once you’ve assigned your CV I/O in CV Instrument, the Song Setup page will update to reflect that those I/O are not to be used for audio.

send pitch to the VCO and trigger the envelope so that we’ll hear our modular play that note.

CV Instrument With the connections in place, we need to set up CV Instrument to use them.

Drag the CV Instrument plug-in into space in the arrangement window and it will create an instrument track, just like with any other VST instrument. If you check the Song Setup options, you should see that CV Instrument has automatically created a couple of output busses for CV Pitch and CV Gate. It doesn’t allocate them an output until you’ve selected them in the instrument itself.

So, from the drop-down menu for CV Pitch Out and CV Gate Out, select the appropriate outputs. These choices are reflected in the Song Setup page. On the CV Pitch Out panel you have a choice between 1V/oct and Hz/V. Eurorack and most modern modular synthesizers use 1V/oct. Hz/V is used by some vintage synths, such as the Korg MS-20.

On the CV Gate Out side you have a couple of choices. Choose Gate, and a high voltage will be sent for as long as you hold a key, or for the length of a sequenced note. Choose Trigger, and CV Instrument will send out a short

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pulse on every key press or note. You can manually set how long you’d like that pulse to be. V-Trig is most commonly used by almost everything; it maintains 0V and then puts out a high voltage when

You’ve probably noticed that it doesn’t sound very good. The pitch is all over the place and the intervals are nothing like semitones. This is because CV Instrument needs to be calibrated to the range of CV levels available through the audio interface.

To calibrate, CV Instrument needs a reference pitch to measure voltage against. Handily, we have a VCO sitting right there, generating pitch.

So, we need to take the audio output of the VCO and plug it into Studio One via an input on your audio interface.

You need to select that input on the left under Audio In on the CV Instrument panel. This is only needed

Not just for notes: you can also use CV Instrument to generate modulation signals from your MIDI data.

for calibration, so you don’t need to set up a special input or anything. I’d recommend using an input that has a gain control, as you might need to adjust the input level to get the calibration to work. The Audio In has a useful level meter that shows us that we’re getting an input. If you are triggering an envelope, then you’ll need to play some notes to see it.

I find it useful to monitor the VCO during calibration and for testing to make sure it’s done it correctly. There’s a little Monitor knob to help you with that.

Hit Calibrate, and you should hear a fairly random range of pitches as it tries to home in on a usable voltage range. After 5-10 seconds, the Calibrate button will either be bordered in red to indicate failure or get a nice green tick that tells you everything is fabulous and that you can now play your modular from your keyboard.

For the best chance of success, here are my calibration tips: • Use a sine wave as the source against which CV Instrument is calibrating. If you only have a sawtooth and square, try the square. Failing that, try a square through a filter to get as pure a sound as possible.

• Input level should be just above -12dBFS on the Audio In meters.

• Your VCO should be set to around C3.

If it has an octave switch, try octave 2.

for the CV Instrument track, you can select Insert Pattern and the editor will open with an empty pattern ready for your sequence. You can change gate times, and add probability and delay to the notes to craft the grooviest of bass lines.

If percussion is more your thing, then you can use the gate output of CV Instrument to trigger percussion modules.

You’ll need a separate CV Instrument with its own physical output for each percussion module you intend to play.

That may make generating patterns a bit more challenging than using MIDI, as each kick, snare, and cymbal will be on a different track.

Modulation possibilities are quite good fun. You can use MIDI notes to send out

stepped voltages to control filter cutoffs, wavefolding, or anything you like. To get smoother steps or more LFO-like behaviour, you can use the Glide function on the right of the CV Instrument. To get Glide to work, the MIDI notes need to overlap so that they become legato.

With a pattern of overlapping notes and the right Glide time, you can create quite a convincing LFO. If you have lots of DC-coupled outputs, then there’s no telling how much modular you could control from Studio One. Even with the MOTU M4, I have potentially four channels of CV, which could be running a pair of sequences or one sequence and some modulation, or I could use it as an elaborate MIDI-to-CV converter.

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If the calibration fails, try the following: • Use different octaves or adjust the tuning up or down. If you’re not hearing the pitch change during calibration then you are not starting at the right pitch.

• Try it without the gate so the VCO is droning.

• Adjust the input level to ensure it’s not distorting.

If it calibrates correctly, then the range available to you will be highlighted on the CV Instrument keyboard. The MOTU M4 can output about 3.7 Volts, so I get three octaves and eight semitones’ worth of usable melodic range. That’s certainly enough for a bass line.

CV Instrument In Use The new clip launcher is perfect for sending sequences or patterns to your modular. If you right-click an empty scene

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TECHNIQUE

OLI FREKE

W

ith Live 12 came the groundbreaking Tunings feature that has the potential to change the sound of modern music.

A bold claim perhaps, but the ability to easily try different tuning systems means it’s now simple to experiment. Despite the ease of changing tunings, the feature can seem daunting at first and that can be offputting, which would be a shame as, in my opinion, alternative tunings lend themselves particularly well to electronic music.

It’s true that some artists have been doing this for years. I’ve always felt that one of Aphex Twin’s ‘secret sauces’ is his liberal use of detuned melodies, creating a frisson against more conventionally tuned elements. Experimental artists like Wendy Carlos, Pauline Oliveros and La Monte Young have also been composing in alternative tunings for decades.

And users of vintage analogue synths have inadvertently been dabbling in microtones for many years!

To use the old cliché, the standard Western tuning system is like the water we musical fish swim around in. We might notice when singers are ‘out of tune’, but we barely think about what that ‘in tune’ is referencing. In comparison to equal temperament (the Western default), all of these new Live 12 tunings are ‘out of tune’... but if you immerse yourself in them, they can be as refreshing as a wild swim in new harmonic waters!

In this article I will outline the main categories of tunings available in Live 12, and the uses to which the creative musician might put them — from the simple but effective to the outright wild and bizarre. Perhaps the future of music may lie somewhere within these systems... and there’s only one way to find out!

Implementation & Use Head over to the Tunings menu under Library, and double-click on any one of the 152(!) different tunings available. Once selected, all MIDI parts will conform to that tuning system (and the Scale Aware feature will disappear for those that are non-compatible).

It must be said that 152 tuning options is completely overwhelming and one would be forgiven for simply giving up at this point! One saving grace is the fact that they are organised into 11 categories, which can be thought of as belonging to four main types as described below.

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Pro Tools Live Live 12’s new Tunings system is your invitation to explore alternative tunings and microtones. move existing MIDI notes to the closest match in the new system — but be warned, notes can be removed if more than one are mapped to the same target note.

Also worth noting is that you can exclude a MIDI track from a global tuning setting by selecting Bypass Tuning. Beneath that button, the Custom Controller menu offers further options, including the ability to restrict notes to just the black or white keys if the scales happen to have only five or seven notes in them. Handy!

All Ableton instruments work with alternate tunings, as do external hardware and third-party VSTs, if they are MPE-enabled and have their pitch-bend range set to 48 semitones.

Equal Divisions Of The Octave (EDO) All EDO tunings divide the octave into equal intervals, and the number in front of each EDO indicates how many notes the octave has been divided into.

12-EDO is another name for equal temperament, which has been the standard tuning system of Western music for at least two hundred years. It’s what all synths, pianos and keyboards use.

Tunings are available from the Tunings menu and filterable EDO options that start with by type; changes made in the table below the list can be saved the numbers 19, 31, 41, 43, 55, and the ‘link’ button will take you to the online manual, which and 72 contain the standard 12 will load extensive information about the selected tuning. notes of the octave, but they have additional microtones in between the When rolled over with the mouse, each semitones. Tip: use the Tuner to identify the tuning comes with informative notes in the ‘normal’ notes, and then the microtones in Help box, but if you don’t happen to know between, to help find your way around. your quintal syntonic comma from your The EDOs that start with 5, 7, 11, 13, quarter comma meantone, I’m not sure and 35 do not include a perfect fifth so it’s going to help you very much. In fact it tend to sound even less harmonic than may be even more offputting! It’s incredibly the previous set. They are pleasingly useful for the scholar of tunings, perhaps, referred to as ‘xenharmonic tunings’. less so for your average bedroom producer.

Use for more exotic results!

However, if you’re brave enough to Fans of drone music might select a tuning, each note of enjoy the beat frequencies its scale is mapped to each that are generated with MIDI note, and you can play some of these unusual them directly from a connected microtone intervals... keyboard (or from a MIDI clip or arrangement part). There is The ability to bypass the current an option for ‘Retune Set on tuning and extended keyboard control Loading System’, which will settings are available in MIDI tracks.

The piano roll updates according to the scale selected to help inform its use. Shown here from left to right: Hijaz 2, 24 Helmholtz, 2 — Dastgah-e Abuata, and 11-ED3_2 Wendy Carlos.

Wendy Carlos created EDOs that she named Alpha, Beta and Gamma. She has written that the “melodic motions of Alpha are amazingly exotic and fresh, like you’ve never heard before”. Her Beauty In The Beast (1980) uses it, and is a worth a good listen. The new tunings sound both odd and strangely familiar at the same time.

European Historical I imagine most of you are probably scoring a film about Henry VIII at this precise moment, so choose amongst this set of tunings for that fully authentic 1580 flavour! These historical examples include such things as the Tertial-Quintal (Erlanger Traktat), which is close to 12-EDO, but has what is called a ‘wolf interval’ between A and E (play that as a fifth and the wolf

Advanced options include the option of restricting notes to just the black or white keys — useful for five- or seven-note scales.

howls!). Another tuning system from 1545 has a name that wouldn’t be out of place in Harry Potter — the Henricus Grammateus!

There are 23 Meantone, Tertial, Helmholtz and Well Tempered tunings in this category and most are only very subtly different with slightly flattened or sharpened fifths and thirds. Use a simple synth tone like a sine or square wave to make it easier to hear the different beat frequencies those intervals produce. It can indeed be very subtle. But you can try writing in them and see if you like the results.

Just Intonation & Rational Intonation Another set of tunings are those built with the ratios of the harmonic series. Many composers have experimented with these and Live 12 includes the following sets: HS (Harmonic Series), Quintal, Tertial (different from the historical tertials), Undecimal, Bohlen, and also Sruti (an Indian system with 22 notes per octave). Some of these use 12 notes per octave, so those you can test using ‘normal’ melodies or chords to see how they work out. For the others... a more experimental approach is needed, ie. play some notes and see what appeals. Or use the techno method — use a random note generator to find a pleasing sequence.

Arabic, Persian & Gamelan Tunings Another rich source of tuning systems is music from around the world. If

you are a musician or composer with musical training from these regions, you can now tune your electronic music to match in a way that’s never been (easily) possible before. In my view this is a groundbreaking development and Ableton should be fully commended.

There are many different tunings under the headings of Arabic Maqam, Turkish Makam and Persian Radif. As I’m unfamiliar with these musical traditions, my approach is to read the Help notes, and where it might describe, for example, Dastgāh-e Bayāt-e Esfahān as one of the “most poetic and romantic” modes, I’ll load it up and see if that quality can be detected (to my untrained ears). Even if not, it provides a starting point for exploration.

Sudanese and Javan Gamelan tunings are extremely well represented (the categories of Degung, Pelog and Salendro), and the excellent online manual provides an extensive exploration of this fascinating family of instruments.

This global selection should be considered a starter pack — notable by their absence are tunings from India, China, Japan and more. But there are over 5000 global and microtone scales in the long-established Scala format which can be imported directly into Live 12.

It’s also possible to make your own tuning systems. Again the excellent online manual takes you through a step-by-step guide, allowing you to easily export your files to Live 12.

There are many ways to engage in this new world of alternative tuning systems: pick one and simply start playing the keyboard; try your existing melodies in the many alternative ‘12 notes per octave’ options; recreate accurate historical tunings for authentic mediaeval experiences; experiment with world music tunings; or even create your own tunings and work out a new theory of music yourself!

Either way, taking a plunge in the bracing waters of the adjacent tuning realities we so often overlook is well worth the effort. You might even change the face of music as we know it!

Useful Links Ableton’s online manual (also available from the ‘tuning window’ under the main menu via the link symbol): https://tuning.ableton.com Scala microtune database: www.huygens-fokker.org/scala Spotify playlist of microtonal music: https://spoti.fi/3Cz0etM

www.soundonsound.com / February 2025

107

Pro Tools Cubase

TECHNIQUE

Some software amp sims can struggle with clean guitar tones — is Cubase’s stock VST Amp Rack up to the challenge?

JOHN WALDEN

T

he plug-in marketplace is awash with more guitar-rig simulations than I care to count, but might you already have what you need in your DAW? Cubase’s stock VST Amp Rack (included in Cubase Pro, Artist and Elements) is not only easy to use, but also very capable. To demonstrate this, I thought it would be fun to try and create something inspired by a couple of classic clean (well, clean-ish) guitar tones. So, with a few suitable audio examples to accompany us along the way (available on the SOS website: https://sosm. ag/cubase-0225), let’s get started.

VST Amp Rack can get you a nice clean(ish) Fender sound but, unfortunately, matching SRV’s playing style is entirely up to you!

Jimi Meets Stevie Meets John Inspired by Jimi Hendrix (think ‘Little Wing’), and a source of inspiration for countless others not least John Mayer, it doesn’t get any more classic than the glassy cleans of Stevie Ray Vaughan’s ‘Lenny’. In a hardware world, the most obvious choice would be a Strat-style guitar into a Fender amp — unless you have a superstar budget, in which case perhaps a Dumble amp. In practice, though, plenty of combinations can work, and the key element is finding the sweet spot for the amp’s input gain. This needs to be set just high enough that the dynamics of the playing can control: the sound is super-clean when played softly, but becomes lightly overdriven when the playing is harder, as the valves start to generate more harmonics.

So, let’s assume you have a suitable guitar that’s equipped with single-coil pickups. Can VST Amp Rack get us in the right ballpark? It certainly offers a number of Fender-inspired amp models. The other obvious element of SRV’s ‘Lenny’ sound is a healthy dollop of a fairly expansive reverb (more ‘hall’ than ‘spring’?), and again the plug-in ought to have us covered here.

The first screenshot is a composite created from VST Amp Rack’s virtual amps, mics and post-effects tabs, and shows the key settings I used to capture the essence of the overall sound. I picked the Tweed

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amp model and its associated cab, and to my ears it does the Fender clean sound very well, but it would also be well worth you experimenting with the Blackface or Deluxe models.

Note that your mileage may vary depending upon the output of the pickups on your guitar, just how hard you play, and your gain staging. At first, I found that I had to crank pretty much every control on the amp to get the Tweed model to start to overdrive when I played more forcefully!

Of course, in the virtual world, you can easily fine-tune the input gain of your DI signal: you can use your audio interface’s instrument preamp gain to drive the amp model harder, or to back things off a little.

You could also look to the gain at different stages inside the plug-in. For instance, you can put the Overdrive pedal into the Pre-Effects section, set with a low gain, and then set the Level control so that it provides a ‘clean boost’ that drives the amp that little bit harder. However you achieve it, finding that ‘just about to go into overdrive’ tone is based on getting the right balance between the input signal level, the amp’s gain/volume controls, and how hard you play. So do be prepared to experiment.

The cabinet and microphone choices are perhaps secondary elements to the overall sound in this case. I simply went with a position nearer the speaker edge and backed away from the speaker slightly,

as I felt this let the sound ‘breathe’ in a more natural way than when positioned right up against the cone centre.

Finally, to emulate the other key element of the tone, I added both delay and reverb effects in VST Amp Rack’s Post-Effects section. The delay is not actually doing a great deal here: with a short delay time setting, low feedback and low mix settings, it’s just there to give the faintest of slapback-style effects. In contrast, the reverb’s Type and Mix controls have been adjusted to add a very healthy dollop of ambience to the sound. The Type control lets you move between Studio, Plate, Room and Hall (which was my choice here), while the Mix control allows you blend in less/more of the reverb effect, as required. I constrained myself to creating the complete tone inside VST Amp Rack, but you could, of course, add the reverb as a send effect using one of Cubase’s various studio-style reverb plug-ins if you wanted further options.

The audio examples that accompany this workshop step through each of the stages involved in creating this tone, and explore how changing the input gain influences the clean/overdriven balance.

Obviously, I’m no Steve Ray Vaughan, and the result is not identical to the iconic original, but it does capture the essential characteristics of the sound in a way that lets you explore that mood and playing

style. Incidentally, for comparison, I also provided three more examples of the same tone that I recreated using well-known third-party virtual guitar rigs. They do all have their own character but, importantly, the version captured through VST Amp Rack is not shamed by the comparison. For a stock plug-in, that’s really impressive.

On The Edge For my second example, I chased a sound with the overall character of U2’s The Edge, as heard on their classic track ‘Where The Streets Have No Name’. This is also a clean(ish) tone that results from a Strat-style guitar being plugged into a finely balanced (in terms of the input gain) tube amp. However, The Edge is all about sound design and, in this case, the ‘iconic’ element is delivered as much by his use of effects as it is the guitar/ amp combination.

The second screenshot shows the settings I ended up with. At this time, The Edge was noted for his use of Vox amps, so I’ve picked the British Custom model (and matching speaker cabinet) in VST Amp Rack. Again, I’ve dialled in the gain and master volume controls so that the amp is pretty clean when the guitar is played more softly (for example, arpeggiated chords) but starts to break up when played more aggressively (strummed chords). There is perhaps more obvious contrast here between the ‘clean’ versus ‘overdriven’ elements of the sound, and maybe that’s a characteristic of the amp choice. Again, experimentation is required to find a sweet spot where the dynamics of the sound feel right with your guitar choice and playing style.

In the Pre-Effects section, a Compressor has been added to provide a little additional ‘always on’ sustain to the sound, while an Overdrive pedal is available as an optional extra. The settings are designed just to add a little extra push to the overdriven element of the sound as and when required, rather than pushing things into full-on distortion.

It’s in the Post-Effects section that the magic happens, though. A Reverb has been included but this is set to add only a very modest amount of studio-style reverb. The key element of the sound is achieved through the use of two delays, each with a different delay time. VST Amp Rack’s effects selection includes three different delay options, and in this case I combined the Tape Delay

(which has a nice analogue-style sound) and Delay (a bit more clinical) pedals in the Post-Effects section.

In terms of the feedback and mix settings for both delays, I aimed for about three repeats before the signal fades away — the best way to set this by ear is simply to isolate each delay in turn as you tweak the controls. The fun part is then setting the delay times for each pedal using their respective Delay (time) controls. I used Beats mode (engage the red LED next to the Delay knob) so that the

section! If that bothers you, you could always use delays outside of VST Amp Rack — Cubase has plenty to choose from. Finally, with plenty of ‘space’ added by these effects, I didn’t feel the need for additional ‘room’ from the microphone position, so I simply placed the virtual mic right against the cone edge.

I’ve provided a similar set of audio examples that step through the development of the VST Amp Rack configuration. It’s a super-cool sound, but it’s also one that requires you to play ‘with’ the effects, by which I mean that if you overplay the part then the dual delays can soon get a bit messy. Again, I found that VST Amp Rack stood its ground very well when compared with some popular third-party amp emulations.

“VST Amp Rack stood its ground very well when compared with some popular third-party amp emulations.” delay times sync to time divisions based on Cubase’s tempo. In this case, I set the Tape Delay to quarter-note repeats and the Delay time to dotted eighth notes.

It’s a classic combination: whether you play an arpeggiated chord/note pattern, or a very simple rhythmic strummed part, the result sounds almost instantly Edge-like.

Incidentally, it would have been nice to have these delays panned left and right, but although VST Amp Rack does support dual amp signal chains (check out the Configuration page), the signal is summed again before reaching the Post-Effects

No Pain, No Gain As these examples hopefully demonstrate, Cubase’s humble VST Amp Rack is perfectly capable of generating some very usable clean(ish) guitar tones, and it can do so with the minimum of fuss too. So, that’s the box ticked for clean tones. What of the higher gain options, I hear you ask? Well, that’s a topic I might well return to another day.

Until then, may all your cleans be sparkly, courtesy of VST Amp Rack!

Getting close to The Edge with dual delays in the VST Amp Rack’s Post-Effects section.

www.soundonsound.com / February 2025

109

Logic Logic

TECHNIQUE

We go beyond the headlines to explore Logic 11’s less obvious new features.

PAUL WHITE

B

ack in SOS September 2024 we explored the new intelligent keyboard and bass players that were introduced alongside an updated Drummer in Logic Pro 11. We also checked out the new stem-splitting facility in SOS December 2024, but there are actually lots of other little workflow enhancements in version 11, which, while perhaps not as glamorous, are no less useful (such as the ability to use a text search to locate effect plug-ins or to interrogate busses — for example, to see which busses send to a delay). This month I’d like to look at a few of these overlooked new features, starting with the ability to bounce in place external MIDI instruments, or indeed to have them bounced during the course of a full mix.

External Instruments To use external instruments, the procedure is much the same as before.

Create a new track, and from the window that pops up, select External MIDI Instrument. Here you can also select which of your interface’s audio inputs will carry the audio from your MIDI hardware instrument. You can set the MIDI channel your synth responds to here, too (this applies whether you’re using traditional 5-pin MIDI or MIDI over USB).

Once you create the External MIDI track, an External MIDI Instrument plug-in window shows up. This too allows you

to change the MIDI and audio input settings if required. There’s a latency compensation button that can be ticked to ensure your timing stays spot on.

You also get the option to send Program Change and Bank Change messages from the plug-in window if you need to.

From this point onwards in previous versions of Logic, to capture your synth performances or sequences you’d have to play the MIDI track while recording the output of the external instrument as a new audio track, and then trim the region to length. Now you can just use the same Bounce In Place command you use for software instruments, and Logic will automatically recognise that an external instrument is in use and do the bounce in real time. The same goes for bouncing a full mix. When bouncing a mix, the new Bounce mode, Automatic, will automatically force a real-time bounce if external instruments are involved, or do an offline bounce if no external instruments are in use. Note that you can do a Bounce In Place involving several selected regions in different tracks all at the same time by selecting One File per Track in the Bounce menu that opens

Logic can now bounce external instruments in the same way as soft synths, albeit in real time only. You can bounce external instruments individually, or as part of complete mixdowns.

once you go to Bounce Regions in Place.

The other new option is One File per Region, which is useful if you want to bounce several regions on the same track but still keep them separate.

Another little nicety is that Logic Pro 11 has added an input monitoring button for MIDI tracks, so you can now send MIDI to that track and hear the result (but without recording it) while at the

Quantec Room Simulator is a new reverb plug-in based on a classic rack unit. It comes in two flavours: the Yardstick models a newer revision...

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...while the QRS models the original ’80s unit.

same time recording to a different MIDI track. A related function is that it is now possible to send the MIDI data from one track to an external instrument on a different track by setting its internal MIDI source to the track you want it to follow. This saves having to copy MIDI data or create aliases.

Before getting onto the next biggie, you can now also drag mixer channels into a new order and the main page tracks will follow. These little workflow enhancements are thoughtful additions, and there are many more that I haven’t even stumbled across yet.

notable that the Quantec Room Simulator was the first commercial reverb unit to include a Freeze function, which is also replicated in the plug-in.

The controls are fairly intuitive, though there are three operational modes for the Quantec Yardstick: Complex, Medium and Simple. These appear to relate to

replaces the reverb tail with a series of early reflections, which can be adjusted in density using the Room Size slider.

The Reverb Time control is greyed out in this mode and the subjective result is somewhat like playing inside a large, slightly metallic tank. Using the Reverb Delay slider, it is possible to create metallic-sounding slapback delay effects that can be effective on drum loops.

To my ears, the original QRS sounds rather like a plate reverb, with its very high-density, slightly resonant-sounding tail.

The Quantec Yardstick is rather more versatile, not least because of its Reverb Density slider and choice of modes, and leans more in the direction of other early digital reverbs when used in its Simple or Medium modes, especially at at lower density settings. It has a more natural, almost plate-like sound in Complex mode with a greater sense of density, but again, reducing the Density Slider can work well on some types of material. I found that both versions added a really distinctive character to drum tracks and drum/percussion loops, creating a sense of space and richness without diluting the impact of the original sound. I also had good results on guitar, and of course it functions as a practical vocal reverb too. Both versions make useful and characterful alternatives to the existing SilverVerb, Chromaverb, Enverb and Space Designer, so I’d recommend that you explore their potential.

“To my ears, the original QRS sounds rather like a plate reverb, with its very high-density, slightly resonant-sounding tail. The Quantec Yardstick is rather more versatile...”

QRS The other major addition I’d like to cover is the new Quantec Room Simulator reverb plug-in, which was developed after examining the hardware and operating code of two original versions: the QRS and the Quantec Yardstick, a second-generation update of the original QRS launched in 1997. The Quantec Room Simulator models acoustic reverberation and its essential resonances using a proprietary methodology that is quite different to the more common algorithmic approach of generating multiple delays and then recirculating them via all-pass and EQ filters. This gives it a unique character that can be heard on many records, and it is now available on both the macOS and iPadOS versions of Logic Pro 11. It is also

reverb density and, on some sources, the Simple mode produces a more flattering result than the more dense, more natural room emulation of the Complex mode.

All the key controls are available as soon as you open the window — room size, decay time, high- and low-frequency delay adjustment, first reflection level and delay, reverb delay and so on — but there are also tabs for Secondary and Extended functions, which access further controls for tweaking the first reflection spread and delay as well as adjusting the reverb correlation (stereo width) plus more EQ functions and left/right channel swapping options for both the dry and reverb signals.

For the older QRS model, the control set is similar but with fewer Secondary functions, and rather than offering Complex, Medium and Simple modes, it has instead an Enhance button that

www.soundonsound.com / February 2025

111

Pro Tools

TECHNIQUE

Pro Tools has many features aimed at post-production.

Could they help you with music production too?

JULIAN RODGERS

P

ro Tools stands out as a DAW that is widely used in both music and post-production. While some workflows and features might seem exclusive to the latter, the line between these two disciplines is often blurred, and features aimed at post can prove useful for music projects too. This was illustrated by a recent project I worked on, which involved creating promotional video material for social media. The brief was to capture a band’s live performance, and although there was a single full-length video track in the session, multiple camera angles and B-roll footage were also available for the video edit. Once the video was sync’ed with the multitrack audio, the mix itself was essentially a straightforward music mix. However, many of the tasks and techniques I used owed something to post-production workflows.

Post-production projects can throw up organisational challenges because they often combine material gathered by different people at different times on different devices, and all of this needs to come together at some stage. In this project, audio was captured by the band’s regular sound person using a Soundcraft digital mixer and recorded into Reaper over USB. Since this was effectively ‘location audio’ recorded on a separate device rather than directly into the DAW, there was an extra step

The Import Audio dialogue: better than drag and drop.

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of ingesting the audio into Pro Tools, a common task in post-production.

Importing Audio To ensure consistency between sessions, I created a bespoke template, as I wanted to import audio for each track into identical sessions. Consistent sound across multiple sessions was critical, as extracts of each performance were going to be cut into a single showreel. The files I received were neatly organised: stereo files with track names and sequential suffix numbers, which made it easy to identify which files belonged to which song. While many post-production users might rely on AAF imports through the Import Session Data window, in this case, the Import Audio window was more suitable. This approach allowed me to audition audio files and decide whether to copy them into the session or reference them directly from their original location.

While dragging and dropping files into the session is an option, I find the Import Audio window much more versatile.

It provides better control over how files are handled, especially when working with larger projects or organising multiple sessions. To access this window quickly, use the shortcut Command+Shift+I on a Mac or Control+Shift+I on Windows.

Stereo Versus Mono A recurring issue when exchanging raw multitrack audio files is dealing with stereo bounces of mono sources. For this project, I chose to import audio into the Clips List rather than directly to new tracks. The Clips List provides a more flexible way to handle ‘stereo’ files that contain mono sources, as frequently encountered in live recordings or exchanges between DAWs. If you import directly to tracks in Pro Tools, you do have the option of the Split Into

Importing audio into the Clips List is much more flexible than importing directly to a track, especially where you need to split stereo files into mono.

Mono command, which is accessible by right-clicking a track name. However, using this on stereo files requires additional steps: importing to a stereo track, splitting onto two mono tracks, and then deleting the stereo and one of the mono tracks. By importing into the Clips List, I could simply reveal the two mono components of a stereo file using the disclosure triangle and drag one directly onto a mono track, which is ready with the required plug-ins in place as inherited from the template session. For stereo sources like overheads or keyboards, you can drag the stereo clip itself onto an already created stereo track. This method saves time and keeps the session cleaner.

Gain, No Pain In live recordings, it’s common to leave generous headroom to avoid clipping. This is exactly what you want during capture, but it often results in levels that are too low for optimal plug-in performance during mixing.

Many plug-ins that emulate analogue hardware are designed to operate with relatively hot input levels, so some gain adjustment is usually necessary.

While Clip Gain is a standard approach for this, I opted to insert the Trim plug-in into the first insert slot of each track.

My reasoning was that the material was consistent across files, and applying a fixed gain adjustment to all tracks would be faster and more efficient. For this, I created a Mix Group for all the tracks, disabled Follow Globals, and linked the plug-in parameters for insert slot A. By instantiating the Trim plug-in in this slot across all tracks, I could adjust the gain for all similarly formatted tracks in one operation.

One detail to note is that Pro Tools treats mono and stereo Trim plug-ins as different plug-ins, so you cannot link parameters across mono and stereo tracks in a Mix Group. The

You can adjust gain across multiple tracks by creating a Mix Group for them, loading a Gain plug-in on the same insert slot for each channel, and then linking the controls for that insert slot.

workaround is to adjust a mono and a stereo Trim plug-in manually to ensure all tracks are balanced. If more than 12dB of gain is needed, Clip Gain, which offers up to +36dB of adjustment, is an alternative. For those looking for a similar third-party solution, Blue Cat Audio offer a free Gain plug-in with extended boost capabilities and built-in grouping functionality, which is ideal in this application.

Keeping Sync One of the most post-production-centric tasks in this project was sync’ing the multitrack audio to the video. In a typical music DAW workflow, sync’ing isn’t much of a concern, as the audio tracks are usually aligned during recording. However, this

project required sync’ing live audio to a single video track, which needed to stay locked to the audio throughout.

I was working in Pro Tools Ultimate, which supports multiple video tracks, though Pro Tools Studio, which supports one video track, would have been sufficient for this task. Pro Tools Artist and Intro do not support video. To keep the video track and its corresponding audio in sync, I created a Clip Group by selecting both the audio and video and hitting Command+Option+G (Control+Alt+G on Windows) to group them. This grouping ensures that the two elements stay aligned if moved.

Professional workflows often rely on timecode for sync’ing, but for this project, a simpler method sufficed. As long as camera sound is available, it’s easy to identify a transient event (such as a snare hit or guitar chord) to use as a sync reference. The key is to avoid manual adjustments in Slip Mode, which can be imprecise and time-consuming.

Tabbing Out To align the multitrack audio and video quickly, I use Tab to Transients. This tool locates the start of transient peaks, such as drum hits, making it ideal for pinpointing sync points. After identifying a suitable transient in the video’s audio track, I’ll use the A key to trim the clip at that location (I can always trim back later to reveal material preceding the sync point). I’ll then repeat the process on the multitrack audio, selecting the clearest track for reference, marking this point so I can find it easily in the next step.

To move the video’s audio clip to the correct location, place the insertion point at the point you want to move

the clip to and use the Grab tool with Control+click (Start+click on a PC) to align the clip to the insertion point precisely. Fast and accurate.

Spot Mode A more elegant alternative uses one of the most overlooked features among music users, but which is indispensable in post-production: Spot mode. This mode allows placement of clips based on their precise timeline ruler position.

While it’s less commonly needed in music workflows, it can save significant time when sync’ing or repositioning clips.

For instance, if a clip is accidentally moved out of its original position and is out of time with the rest of your audio, you can quickly restore it to its original time stamp position. Simply grab the clip in Spot mode, select the Original Time Stamp field in the dialogue box, and hit OK. This restores the clip to its exact original location.

Using sync points — special markers created within clips — in conjunction with Spot mode is really fast and accurate. After identifying the desired location for your clip’s start, end or sync point, place your insertion point and copy the exact time from the main counter. You need an extended keyboard for this: hit * on the number pad to copy the location from the main counter, and paste this into the Sync Point field in the Spot mode dialogue box to align the clip perfectly.

Post Hoc Although these techniques originated in post-production workflows, they are equally valuable in music projects. For example, Spot mode is excellent for placing reverse swells and risers, where the start time of the clip isn’t what’s important. The project that inspired this article reminded me how versatile these techniques can be. By borrowing from post-production workflows, music producers can tackle complex tasks with greater efficiency and accuracy. Whether it’s sync’ing audio to video or managing multitrack imports, these techniques are powerful additions to any Pro Tools user’s skill set. And at the end of the day they are all just features of your DAW of choice.

If a feature of Pro Tools helps you get a project done, learn it and use it.

Spot mode is extremely useful for keeping audio and video in sync, and can also help you out if you’ve gone too far in your editing and need to move audio back to its original time stamp position.

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Chosen By The Readers Of Sound On Sound The SOS Awards throw the spotlight on innovation and contemporary design in music technology, with nominations limited to products launched or reviewed in the period from September 2023 through to the end of August 2024. The winners in each category are the products that received the most votes through our online voting portal.

Congratulations to all the winning products — as well as the very worthy close contenders in each category, who have been awarded our Highly Commended accolade.

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RUPERT NEVE DESIGNS MBT www.rupertneve.com

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WINNER

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NATIVE INSTRUMENTS GUITAR RIG PRO 7 www.native-instruments.com

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www.soundonsound.com / February 2025

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SOFTUBE CONSOLE 1 MKIII 116

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FE ATURE

S OS AWA R DS 2025

HIGHLY COMMENDED SOS AWARDS 2025

GUITAR & BASS TECHNOLOGY

ABLETON LIVE

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STEINBERG SPECTRALAYERS PRO 11

www.toneprojects.com

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KEYBOARD/SYNTH

NATIVE INSTRUMENTS S88

YAMAHA MONTAGE M8X

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Introducing the MKII

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DRUM MACHINE

ELEKTRON DIGITAKT II www.elektron.se

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HARDWARE AUDIO RECORDER

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AUDEZE MM-100 www.audeze.com

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STUDIO MONITOR

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FOCAL ST6 TRIO6

ELECTRO-VOICE EVERSE 12

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PODCASTS ELECTRONIC

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Emulator II 40th Anniversary Iconic Sampling Keyboard

In celebration of the 40th anniversary of the groundbreaking E-mu Systems Emulator II sampler, Dave Rossum, Kevin Monahan and Paul Wiffen share inside stories of the development and manufacture of the instrument.

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Check out our website page for further details

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INSIDE TRACK

Rogét Chahayed Virtuoso pianist Rogét Chahayed has found a unique niche as a classically trained hip-hop producer.

PAUL TINGEN

“T

here are videos on YouTube of me playing piano pieces

by Rachmaninoff, Liszt, Chopin and so on, and someone commented, ‘How did this guy go from this to “Broccoli?”’ I spent years of my life studying the rich, respected, sophisticated art form of classical piano, and to then get away with making smash hit songs with just two chords is a relief in many ways, because sometimes the best things are simple. And today I’m really happy that my discography reflects the many different kinds of music that I love, from R&B to hip-hop to jazz to classical.”

Rogét Chahayed has worked with household names like Drake, Kendrick Lamar, Travis Scott, Bruno Mars, Jennifer Lopez and many others. The most recent major hit that he was involved in is ‘APT.’ by Bruno Mars and Rosé, which was the biggest song worldwide of the end of 2024. Chahayed has also received nine Grammy nominations, including for Producer of the Year, Non-Classical, in 2022.

As one of the world’s top pop writers and producers, Chahayed’s unique selling point is his classical music background, which comes with serious keyboard chops, an abundance of chordal and melodic imagination, and a knowledge of orchestral arrangement. This classical background comes to the fore on projects like Jennifer Lopez’ This Is Me… Now (2024), but is much less apparent in many other cases, like his two-chord breakout hit ‘Broccoli’ by DRAM in 2016, and the simple Korean pop-rock of ‘APT.’

Photo: Matt Ty

Crossing Borders

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Chahayed was born in 1988 in Los Angeles to a father from Damascus, Syria, who grew up in Lebanon, and a mother from Argentina. “He sings in Arabic and plays the darbouka, and my mother listened to a lot of music. When I was seven, my dad enrolled me and my sister, who was four at the time, in piano lessons. I didn’t really

Rogét Chahayed’s Virtuoso Studio.

have a choice, so I played piano as a kid, but did not really have a passion for it.

“This changed when I was 15 or 16, and saw the movie The Pianist and heard Adrien Brody’s character play Chopin’s Ballade No 1 in G minor. I said to myself, ‘I really want to learn this.’ I took studying and practising the piano much more seriously, and got deeper into classical music and discovered jazz and Bill Evans. I saw how French impressionism, like Debussy, and jazz related to each other, and got into the stylistic and theoretical aspects of music.

I decided I wanted to do this full-time, and went to the San Francisco Conservatory of Music, where I studied Liszt, Rachmaninoff, Ravel, Chopin, Prokofiev and many other great composers. I was extracting the little moments that made their compositions special harmonically and melodically. That became the foundation of my approach to pop and hip-hop.

“I’ve always loved hip-hop, but wasn’t really allowed to listen to it as a kid because my parents were pretty strict, and many rap CDs had these parental advisory stickers. So in secret friends of mine burnt me CDs with music by Eminem, Cypress Hill, Dr Dre and so on. When I was in

conservatory I had friends who were opera singers, and who loved listening to rap and pop music. I wondered who was playing these sounds and chord progressions and riffs, and discovered Scott Storch, the Neptunes and other producers who inspired me sonically.”

Learning From The Masters A chance meeting then changed the course of Chahayed’s career. “While studying, I would come back to LA for my Summer breaks and started playing in bands, and was a hired gun for anybody who needed keyboards on a track. I also got together with a group of friends to write songs. Those were my beginning stages.

After I graduated I moved back to LA, and I was doing jazz gigs and teaching piano to support myself, and I’d go to the studio at night and make beats. I had a friend who owned a studio, and he gave me my own setup and taught me how to use Logic.

I was fascinated by the fact that I could write a four-chord progression and then add tons of instruments to it.

“My real journey started in 2013, when I was introduced to producer Mel-Man, who had been a key contributor to Dr

Dre’s album 2001 [1999] and Eminem’s The Marshall Mathers LP [2000]. Mel-Man trained Scott Storch and Mike Elizondo.

I worked with Mel-Man for about a year, and one day he brought me over to Dr Dre’s Aftermath Studio. Dre immediately put me on the spot and asked me to add keys to a track. A few months later he hired me as his in-house keyboard player, which I was for most of 2014. It was very demanding, but also an incredible learning experience.

“These two incredible mentors, Mel-Man and Dr Dre, showed me the ropes and helped me channel my classical knowledge and expand my knowledge of sounds, and realise how powerful an accidental thing you play can be. I learned many elements of making music that I to this day take into every session I walk into. A lot of my time with Dr Dre was spent trying to find sounds, with people hovering over my shoulder, and that was very nerve-racking at times, with a lot of anxiety. People pull up a drum beat with no music, so as a musician there’s a lot of pressure, because you have to provide the chords that the hook will be written to, using the right sounds.

“It made me realise the importance of having a massive library of sounds. www.soundonsound.com / February 2025

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A few of Chahayed’s many keyboards. From left, top row: digital Mellotron, Sequential Prophet X, Sequential OB6; middle row: Korg Kronos, Korg Prologue, Nord Stage 3; bottom row: Sequential Prophet 08, Korg SV-1, Moog One.

I learned about Native Instruments’ Kontakt, and the many different kinds of synths.

I was obsessively studying what sounds and keyboards and patches people like Dre and Pharrell and Scott Storch were using on songs. So I really learned about equipment, and then got into analogue synths. And I learned that when you use a sound, if you play or phrase it differently, or maybe put an effect on it, you won’t even recognise it as the same sound. It’s not necessarily about what are you playing and what sound are you using, but also about how you’re using it and in what context.”

Switching Modes Chahayed’s next big moment occurred in 2016, when he co-wrote and co-produced the aforementioned hit ‘Broccoli’ by American rapper DRAM, which went on to go quintuple platinum in the US and receive a Grammy nomination for Best Rap/Sung Collaboration. The song is based on an ultra-simple piano part that in no way reflects Chahayed’s advanced musical skills, but gave Chahayed his “first recognition. It was my first placement. Many people discovered who I was through that song.”

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More placements soon followed, with tracks by artists like Travis Scott, Miguel (‘Sky Walker’), Kesha, Halsey (‘Bad At Love’), Calvin Harris, G-Easy and more. Chahayed’s next big moment arrived in 2018 with ‘Sicko Mode’ from Travis Scott’s Astroworld album, a track on which his classical music background is very much in evidence.

Chahayed and Hit-Boy were responsible for only the first minute of ‘Sicko Mode’.

Hit-Boy gave his version of events in our 2024 July issue (www.soundonsound.com/ techniques/inside-track-hit-boy).

“For me,” says Chahayed, “the most harmonically pleasing composers for piano are Rachmaninoff, who really evokes romanticism and big cinematic moments, and uses almost jazzy Gershwin-like chords, and Scriabin, who is all about mysticism. These guys are such big influences on me, because they have normal tonal chord progressions, and then there’s always one or two notes that stick out, and once they resolve, it gives you this sense of relief. I was thinking about these two composers when I was playing the ‘Sicko Mode’ chords. I was experimenting in the studio with Hit-Boy and happened

to come across that organ sound. I played the part, and he flipped and chopped it.

“It’s interesting that that song ended up so big. For me that was a win for people who are really musical and into music theory. I was like, ‘Wow, amazing that something as crazy and experimental as that song could go to number one!’ But I had no idea that the song was coming out.

One day Hit-Boy texted me saying that we had an intro or something on Astroworld, but he did not know what. So I’m listening to the album the night it comes out [August 3rd, 2018], and when I clicked on ‘Sicko’ and I heard the organ, I was like, ‘OMG, they used 25 seconds of just me playing, and then Drake comes in, this is insane.’

Then I realised that the song is made up of three different beats. That song became one of the biggest rap songs of all time and opened many doors for me.”

Executive Suites Since ‘Sicko Mode’, Chahayed has cowritten and co-produced songs by the likes of Carly Rae Jepsen, Doja Cat, Kali Uchis, Drake, Big Sean, Nas, 21 Savage, Mary J Blige, John Legend, Lil Wayne and the Kid Laroi. Recent big projects include executive

production on albums by Jack Harlow (Come Home The Kids Miss You, 2022) and Jennifer Lopez (This Is Me... Now, 2024), and his work on the mega-hit ‘APT.’ “When executive producing, I’m not just playing and writing in the room. Sometimes I wasn’t at the actual sessions, but listened to songs afterwards, seeing what they need, arranging strings, outros, intros, things like that, making sure the verses and the hooks were better. J.Lo really trusted me to be in charge of everything, which was a big responsibility. In addition to writing and playing, I spent a lot of time on phone calls and planning and stuff.

“But I was also very hands-on, for example in the title song and opening track, I did the harps and the flutes and the strings and everything. Obviously, a lot of that album was about her and Ben Affleck and their long relationship history, and she said, ‘I want to write something for Ben.’ I was like, ‘Why don’t we start with the B chord for Ben?’, and started playing a B chord, and literally, just in the moment from having that conversation with her, I played that harp part on the

spot and started adding things to it, and that became the intro.

“That’s a really good example of being on the keyboard, where all great composers and all great symphonies and orchestral pieces, any really great ensemble piece is going to start, because of the range the piano has. It also shows

It was his work on Travis Scott’s hit ‘Sicko Mode’ that brought Rogét Chahayed to the attention of many within the music world.

how much I love to be an orchestra; I love harps, strings, flutes, pizzicatos, all kinds of things. I studied orchestration at the conservatory, and I know where to add the flute, I know where to add the cello, I know where to add the high strings, the tremolo, whatever. It’s years and years of exposure and listening that becomes part of you. So I’m very grateful for that.

“The Jack Harlow album was a lot of work, with lots of trial and error, and collecting ideas from different people, and really seeing it through from start to finish. Jack also took on the role of executive producer, and brought in people.

On the music side, he really trusted us to make sure that all the nuts and bolts and things were tied up. We did a lot of post-production on songs. For example, we spent months producing the song ‘First Class’, trying to nail it and making sure it was perfect, so it would be the major hit that it became.

“For the session of ‘First Class’, I had three keyboards including my Nord Stage 3 hooked up to Logic, and also used a number of VSTs. I would record straight

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Effects pedals as well as keyboards are important sources of inspiration for Rogét Chahayed.

into Logic and then I would AirDrop whatever files or chord progressions or things to Jack’s engineer [Nickie Jon Pabón, covered in our March 2024 issue], and he would blend everything together.

Most of my sounds on that song are from the Nord, including the fairytale-sounding string part, but the Rhodes at the end comes from Logic’s standard electric piano, which I put through Cableguys’ HalfTime plug-in and the Retro Color RC-20.”

Pulling Together ‘APT.’ was another hit that came about through collaboration. “In September 2023, I was in a session with Cirkut, Omer Fedi, Theron Thomas and Amy Allen. Rosé was there and we were talking about the vibe she wanted. We started making a track, but it felt a little too soft. We were about to finish the session when Rosé started talking about a drinking game that they have in Korea. So Cirkut pulled up some drums, and Omer suggested we just do something simple, so we started by playing those hits in the beginning. It was very open, there weren’t even any chords.

“I said, ‘We need a pre-chorus, and some hook chords,’ so I put in the four-chord progression. I was using very bare-bones stuff, just the Mellotron, with

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a brass sound for the stabs and a piano sound for the chords. So there were Cirkut’s drums, Omer played some acoustic guitar, and I think I also added a bass sound from the Korg Kronos 2. Plus there’s of course the hook which is inspired by the ‘apateu’ chant of the drinking game. It was a very skeletal track, I didn’t really know what to make of it, but we knew it was cool.

“I didn’t know that Bruno Mars was going to be on the track until about a week or two before it came out. Rosé and he had been working together. The chords, riffs and drums are the same, but he added more rock elements to it, more guitars, more synths and things like that.

I think he also put in that bridge at the end.

That really added so much to the song.

So you’ve got this Korean drinking game element and Bruno Mars, making it a huge cultural moment. K-pop is big because they’re taking a lot of elements from American music and pop music and putting their own spin on it, but in this song there’s a Korean element that’s brought into Western pop, and it became a huge smash.

“It’s crazy to think how far our basic idea went. When you write a song, you never know what will happen with it. It was just an incredible moment. About the reference to Tony Basil’s song ‘Mickey’

from 1982, to be honest, I wasn’t thinking about that at all when we were writing.

When I heard the song later, I could see some similarities in the pocket of the drums and a little bit the cadence. There are only so many notes you can use, so this is bound to happen, and when it does, the smart thing is to get it cleared.”

No Distractions When working on ‘APT.’, Chahayed was accompanied by his engineer, Julian Vasquez, Logic and a few of his many keyboards. “Julian is fantastic, and he records what I do, so I can focus on writing, playing and producing. We discovered a new process when working on the J.Lo album. I have doubles of all my favourites, so I have two Nord Stage 3, two Stage 4, two Sequential Prophet X, two Korg Kronos 2 and two Mellotron Mini model D keyboards, and two laptops with Logic.

When we get called for a session, Julian brings what we decide we need, and makes sure everything is ready to go.

“I am in Logic, and have been using it since 2009/10. It is home for me, I love the simplicity and cleanliness of the interface, the quality of the audio, and so on. Julian works with Ableton, and I sometimes use it as well to mess with drums and manipulate

time. Julian also is comfortable with FL Studio and Pro Tools, so he’s got every DAW under his belt, but all I need are my keyboards and a computer with Logic. I use Logic as a separate keyboard. I may have 40 or 50 sound patches pulled up from libraries I trust and want to mess with in the moment. As soon as I’ve played something, Julian has got it recorded, and then he loops or cuts it or adds fades, or whatever, and sends it to Cirkut or Omer or whoever is driving the session from their computer.

AirDrop is a big part of our workflow.

“Some friends tell me I need to get more into engineering, but Julian is great at it, so why should I? Making music is what I do. Having someone else with expertise in recording, organising, mixing and so on is one of the best investments I can make. It’s very distracting when I’m sitting in a room multitasking and spending most of my time looking at a screen. It compromises the magic that could be happening. I’m there to connect with the artist, and to get a song started and finished. I’m there to get the track popping and to make it work.

Cirkut is the only producer I know who can connect with an artist while he’s cutting vocals and EQ’ing something and talking

to me all at the same time. Watching him work is pretty incredible.”

On The Arc Which of his keyboards Chahayed turns to first “depends on the artist or the style of music or the vibe. My go to is the [Nord] Stage 3, because it has literally everything I need covered. It is like what the Yamaha Motif ES8 was to Scott Storch or the Korg Triton to the Neptunes and Timbaland.

But if I’m going to a rap session, I’m going

the chord progressions I’ve come up with happened in 10 seconds. So I obviously love presets. If I need a quick piano sound, or a quick synth or a quick Rhodes, I can find them in the blink of an eye on the Kronos or Stage 3. When you buy a good keyboard and a good sound bank, half the work is done. These guys took months to perfect these sounds, and then it’s up to people like us, performers, composers and producers, to play and express them.

I also have a guy who upgrades my sounds and helps me with new patches for my synths. And I do my research online to find new sound banks and upgrades for every keyboard that I have.

“I’m not much of a sound designer, I prefer to call myself a sound manipulator. You tailor your sounds to who you’re working with: a rapper, a songwriter, a singer. If you have a rapper with a lighter voice texture, you may use smoother sounds, or you have somebody who has a tone that punches through, you can amplify that with more aggressive sounds. You try to give a rapper or singer a good chord progression, a good mattress, a bed we call it, to float on that suits them. Producing is not about you.

In the beginning I wanted to put as many

Rogét Chahayed: “In the beginning I wanted to put as many sounds as I could in a track, and create this symphony of sound, but that’s really not important. You have to think about the artist and the listener.” to try and use the Prophet X as well, because it has more cinematic things, with darker strings, and so on. If I’m going into a pop session or something upbeat with an electronic vibe, I’ll bring the Kronos because it has so many great organic sounds, guitars, basses, synths, funky things. If I’m going for a more playful pop vibe, I’ll bring the Mellotron.”

Whatever keyboard he plays, Chahayed stresses that speed is of the essence. “The best music is made in seconds. Many of

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Virtuoso Studio

Chahayed’s drum kit is permanently miked up, with a pair of Coles 4038s on Triad Orbit stands as the main overheads.

sounds as I could in a track, and create this symphony of sound, but that’s really not important. You have to think about the artist and the listener. If you can find a way to make it something you like as well, then you’ve really done your job.

“You also have to let go of trying to be a virtuoso show-off, which a lot of musicians are. Instead, Mel-Man and Dr Dre both explained the importance of having an arc to me. When a great composer is writing a piano piece, chamber music, whatever it is, opera, aria, there will be phrases. Hip-hop is taking one phrase that you really like and looping it over and over. It’s about finding that golden loop.

I find so much satisfaction in the perfect two, three, four-chord progression, or even just a riff, that helps create an arc.

“You really have to limit the amount of notes you play, and put emphasis on the sound, because sometimes all people need is one chord, one sound, one thing. You have to serve the purpose of getting the artist excited. I need to put something into a track that they’ve never had before. Often, I get tired of

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sitting at the piano and trying to be a riff machine. Instead I may pull up a sample, or something I’ve made at my studio, and I’ll put it through a pedal or half-time it or speed it up. When you’re experimenting, you have to be open to what you’re not used to doing. In other cases, I sit back and people start something and I try to find something that fits.”

Blank Canvas Rogét Chahayed also makes beats, either alone or with others. “We may create a sample or a loop or a vibe, and sometimes I run things through a pedal.

I use a lot of Strymon pedals, and have the Eventide Space, a Rainbow pedal and the Red Panda Tensor. The latter allows you to change time almost like a vinyl record, slow it down or pitch things up, and so on. You can create three or four samples out of one with these pedals.

“I also use VSTs. There are a few things that I love in Kontakt. One of my favourite string packs is called Vivace.

For a good 808, I love the Ramzoid 808 Cooker. There’s some great pianos in

Although Rogét Chahayed prefers to hire engineers rather than record himself, he is the proud owner of an impressive studio, informally known as Virtuoso Studio. The huge number of keyboards there are the result, says Chahayed, of “my love of Guitar Center when I was a young kid. My entire style of working with analogue keyboards came from that.

Visiting Guitar Center was like going to Disneyland.

I’d be looking at all the stuff they had, with tons of keyboards that I did not think I could ever afford.

I was fascinated with the sounds I could get from them, and thinking, ‘One day I’m going to have a studio just like Guitar Center, with shelves and shelves of keyboards.’ Now it’s a reality for me!

“In addition to the keyboards I mentioned earlier, my Nords, Kronos, Prophet Xs and Mellotrons, I have a Korg SV2, and Oberheim OB6 and OB8, Korg Prologue, Moog One, Prophet 8, Juno 106, Kodamo Mask 1, Expressive E Osmose, Yamaha DX7, a Fender Rhodes, and an Akai MPC Live 2. I also just got a Steinway grand piano last year.”

Virtuoso is also fitted with an impressive set of studio gear that includes a Mac tower with Logic and tons of plug-ins, PMC speakers, Ferrofish A32 Pro Dante and RME Digiface Dante interfaces, RME 12Mic Dante 12-channel network-controllable microphone pre, and a Dante-enabled Neve 1073 OPX preamp. Chahayed recently had a live room added, which hosts a drum kit provided by Anderson .Paak, and a vocal booth, and the studio’s mics include a Telefunken ELA M251 stereo set, Neumann U87Ai stereo pair, Myburgh M1, a pair of Coles 4038 ribbons, Sennheiser MD 421-II and MD 441-U, Shure SM57 and SM7, and a Triad Orbit stand system.

Imagiro Piano that you can tweak. I also really like a plug-in called the Initial Slice loop slicer, which is fun. It has royalty-free samples that you can chop and pitch and manipulate into whatever beat you’re making. And I love the Native Instruments Duets plug-in. It has incredible vocal runs and really cool things that can add a nice texture to a lot of songs.

“In addition, I’ve also been working on my own artist project, which is more ambient in nature, like piano film score.

‘Aurelius’ is my first release. I have just finished my second piece. So I am working on my own music and would like to get into film score eventually. I’m still very much in love with the piano, but classical musicians are very much historians that carry on a tradition of performance from hundreds of years ago. As fun as this is, I don’t want my life and my musical identity to be based off of somebody who died 100 or 200 years ago. I want to compose myself, and producing hip-hop and pop has been the perfect canvas. And you can do whatever you want nowadays.

It’s an exciting and interesting time.”

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SPOTLIGHT

Antares Auto-Tune Pro 11 LUKE WOOD

O

riginally, pitch-correction processors were developed to fix the slight tuning issues that are so common in recorded vocal parts, but engineers and artists soon discovered that by pushing those early processors’ settings beyond their intended limits, they could make them behave more like an effect — the almost vocoder-like sounds that grace so many modern productions.

The technology has come a long way since then, and there are now plenty of natural-sounding pitch-correction plug-ins that can be used to nudge notes discreetly into place. In fact, you’ll probably have such a tool bundled with your DAW. But there’s also a growing range of pitch-processing effects designed specifically with creative applications in mind. Some combine pitch-correction with effects such as formant-shifting and ring modulation, while others use the corrected signal to trigger synth engines or pass MIDI data to other instruments. In Spotlight this month, we’ll take you through a selection of plug-ins that were purposely designed with creative use in mind, whether that’s extreme hard-tuning or mangling a sound beyond all recognition.

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Creative Pitch-correction Plug-ins

Love it or hate it, pitch-correction is here to stay — but there can be way more to it than fixing performance issues.

Antares Auto-Tune & Mutator For many, the first name to spring to mind when you mention pitch-correction will be Auto-Tune. Launched back in 1997, it was intended to tackle minor tuning issues, but it wasn’t long before producers and engineers realised that extreme settings created an ear-catching effect. Antares have expanded the software significantly over the years, and Auto-Tune Pro 11 includes a wealth of features, including a four-part harmony player, tuning to MIDI notes, graphical pitch and timing

editing and more. There are also no fewer than five versions choose from!

The flagship offering is joined by Artist, a variant optimised for real-time tracking and live performance. EFX+ pairs the pitch-correction algorithm with a range of built-in modular effects. Access is a stripped-back, entry-level version, while Hybrid runs on Avid’s DSP-based Pro Tools HDX and VENUE systems.

Outside the Auto-Tune range, Antares also offer Mutator, a sound-design plug-in created specifically for transforming voices into aliens, monsters and other weird and wonderful noises. In addition to

pitch-shifting and ring modulation, Mutator features Antares’ innovative throat-modelling technology, and even includes an Alienize section that separates and reverses segments of the audio signal to create a new language! As well as allowing you to purchase individual plug-ins up front, Antares offer Auto-Tune Unlimited, a subscription bundle that includes all versions of Auto-Tune alongside their entire collection of vocal-processing plug-ins. $ Auto-Tune $49 to $459, Mutator $79.

W https://sosm.ag/antares-autotune-pro-11 W www.antarestech.com

Devious Machines Pitch Monster

Baby Audio Humanoid

Baby Audio Humanoid

Described by Baby Audio as an “over-the-top pitch corrector”, Humanoid has been designed specifically for creating hard-tuning effects and carrying out radical voice transformations. Its Pitch section makes it possible to instantly quantise incoming notes and tune them to a user-defined scale, and also allows vocal parts to be ‘played’ using either MIDI notes or automation data. Some additional shaping of the signal is provided by Robotify and Formant controls, but it’s in the following Synthesize section that things get really interesting.

Auburn Sounds Graillon 3 Along with its pitch Auburn Sounds’ Graillon 3 is correction, Humanoid a pitch- and formant-shifting plug-in features a built-in wavetable that’s capable of both natural-sounding engine that resynthesizes the correction and more extreme creative incoming signal, creating a effects. It features three pitch engines: new, blendable layer that pushes the vocal G2, which retains access to the sound sound even further into synthetic territory. of this plug-in’s predecessor; G3, an Yet more synth-style control is provided by updated and enhanced version that a Filter section, which acts on the spectral boasts improved pitch detection; and content gathered during the plug-in’s I1, which utilises the engine found in analysis process — a move that allows the company’s Inner Pitch plug-in, and for much narrower cuts than traditional offers more natural-sounding correction filters. A Utility module on complex source material. A built-in offers onboard de-essing, effects section provides low-latency gating and control over the compressor, gate, chorus, preamp and pitch-detection behaviour, bit-crusher processors, all of which while an Effects section have been optimised for vocals. Best adds some finishing of all, Graillon 3 is free to download touches like stereo — though if you choose to pay for the Full widening, vibrato and Edition you’ll also get a Pitch-Tracking tempo-sync’able stutters.

Modulation feature, along with support for non-standard tunings and a built-in $ $129. vocal doubler.

W https://sosm.ag/baby$ Free. Full Edition $38.67. audio-humanoid W www.auburnsounds.com W babyaud.io/humanoid

Auburn Sounds Graillon 3

Despite its relatively unassuming GUI, Devious Machines’ Pitch Monster packs in a trio of sound engines (Granular, Formant and Vocode) that allow it to extend far beyond the capabilities of typical pitch-based processors. Eight-voice polyphony means that incoming monophonic sources can be played much like an instrument, and it’s possible to generate up to eight unison voices to reach a grand total of 64 parts. In addition to its standard pitch-shifting behaviour, the plug-in features a Chord mode that allows users to create up to eight-part harmonies and chords using an on-screen piano keyboard, and a MIDI mode that tunes the incoming signal based on note data from a MIDI controller or DAW track. When using MIDI mode, Pitch Monster will respond to glide and pitch-bend messages too.

Although primarily designed for vocals, Devious Machines say that their plug-in is also capable of working with other clean monophonic sounds. $ $99.99.

W www.deviousmachines.com

Devious Machines Pitch Monster

MeldaProduction MAutoPitch Available as part of MeldaProduction’s free MFreeFXBundle, MAutoPitch is a simple plug-in that’s capable of both subtle corrective tuning and

MeldaProduction MAutoPitch

www.soundonsound.com / February 2025

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C R E ATI V E PITC H - COR R ECTION PLU G - INS

more heavy-handed effects. Incoming monophonic audio signals can be locked to chromatic, major, minor and pentatonic scales, while an Automatic Tuning section houses Depth, Speed, Detune and Base parameters that can be used to fine-tune the pitch-correction’s behaviour. An Effects section then offers control over formant-shifting and stereo width processors. As with everything in the MFreeFXBundle, MAutoPitch is free to use, but for a modest fee all of the bundle’s plug-ins gain additional functionality, including oversampling and the ability to create and organise your own presets. $ Free. Extended features $51.

W www.meldaproduction.com

Native Instruments The Mouth Another option that combines pitch-correction with synthesis is Native Instruments’ The Mouth. Created by British producer and performer Tim Exile, this plug-in is capable of creating melodies and harmonies from any input source, and keeps latency low enough to allow for live use. Incoming signals are automatically tuned, either to a user-defined scale or in response to MIDI notes, before triggering built-in synth, bass and vocoder modules.

Alongside the sound-generation sections, the plug-in includes an effects section that houses delay and compression processors. A simple mixer section then offers independent control of Input, Synth, Vocoder, Bass and FX signals, while a set of eight macro knobs provides control over the likes of tone, resonance and stereo width. The Mouth was built using NI’s powerful modular Reaktor software, and will run happily in the free Player version — but those with a full licence are

Native Instruments The Mouth

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able to take a look behind the scenes and see exactly how the effect works. $ $69.

W https://sosm.ag/ni-the-mouth W www.native-instruments.com

Slate Digital Metatune

Slate Digital boldly claim that their offering in the world of creative pitch effects is “the best automatic tuner on Earth”. Capable of everything from subtle, natural-sounding tuning to extreme, robot-like sound effects, Metatune promises to have users up and running in seconds thanks to a simple and intuitive GUI. On-screen piano keys make it possible to quickly select a target note, chord or scale, while a handy ‘heat map’ display provides a visual indication of the input signal’s pitch both before and after processing.

Metatune’s entire range of effects can be created and fine-tuned using just three simple controls: Speed, Sustain and Amount.

There’s also a Note Stabilizer mode that allows wanted elements of the input signal such as vibrato or small variations between notes to go unprocessed, and, thanks to a Groups feature, Slate Digital Metatune it’s possible to link multiple instances of the plug-in so that changes can be applied across multiple tracks instantly, rather than relying on copying and pasting settings. $ $199.

W https://sosm.ag/slate-digital-metatune W www.slatedigital.com

Soundtoys Little AlterBoy

Soundtoys Little AlterBoy

Soundtoys know a thing or two about pitch-shifting: their founders were part of the team behind Eventide’s iconic H3000, and were also involved in creating a number of pitch- and formant-shifting tools for Pro Tools TDM systems both as Wave Mechanics and Soundtoys.

Pitch-correction duties in their current line-up, though, fall to Little AlterBoy, a simple and intuitive plug-in dedicated to hard-tuning and vocoder-style effects, with some saturation thrown in for good measure. Little AlterBoy sports a minimal set of controls, with Pitch and Formant dials joined by three operation modes: Transpose, Quantize and Robot. The first mode simply shifts the pitch of the incoming signal without applying any automatic correction, while the second tunes it to the nearest semitone, resulting in the ever-popular hard-tuning effect.

Finally, Robot locks the signal to a single

Waves OVox

note that can be tuned in either direction using the Pitch parameter. It’s also possible to control the Pitch dial from a MIDI keyboard, essentially turning the plug-in into a vocoder. Lastly, there’s a Drive control derived from the tube saturation models found in Soundtoys’ Decapitator, and a Mix control for blending the effect with the unprocessed input signal. $ $99.

W https://sosm.ag/soundtoys-little-alterboy W www.soundtoys.com/product/littlealterboy

Waves Vocal Bender & OVox

There are two creative pitch-correction plug-ins by Waves: Vocal Bender and OVox. The former is the simpler option, with real-time pitch- and formant-shift controls providing a fast and easy way to achieve the usual hard-tuning effects, as well as more subtle doubling effects.

A Flatten control facilitates robot-like vocal tuning, and there’s a section equipped with four modulation sources (pitch and amplitude envelopes, an LFO and a step sequencer) that can be used to create effects such as vibrato, pitch drops and sequenced pitch changes. OVox, on the other hand, is a voice-controlled instrument that provides independent

Zynaptiq Pitchmap

Waves Vocal Bender

control over a vocal’s pitch, amplitude and formant. It has two independent engines that can generate up to eight voices, making it possible to create chords from a monophonic input, and there’s a whole host of built-in filtering, modulation and effects processing. Along with the usual vocoder-style effects, OVox can transform vocals into entirely new sounds, and it’s even capable of converting the incoming signal to MIDI data, allowing users to trigger other instruments with their voice in real time. $ $149 each.

W https://sosm.ag/waves-ovox-0520 W https://sosm.ag/waves-vocal-bender W www.waves.com

Zynaptiq Pitchmap & Pitchmap Colors Zynaptiq are well known for their innovative approach to processing, and Pitchmap is no exception. It’s capable of processing the pitch of individual sounds within a mixed signal in real time, making it possible to change the melodies and harmonies of any source by either playing a new part on a MIDI keyboard, or manually tuning notes in the GUI. Some clever behind-the-scenes processing splits

incoming signals into their component parts, with the resulting pieces displayed above an on-screen keyboard. From there, it’s possible to apply tuning correction to individual notes, harmonics and transients, as well as ‘re-mapping’ them to a new pitch, or even isolating/suppressing them.

Used sparingly, Pitchmap is capable of subtle corrective work — there’s a Natural Mode that recognises vocal sounds and processes them separately to help avoid artefacts — but pushed a bit harder, it’s capable of far more, from quickly trying out new harmonies and variations to creating synth-style sounds from live inputs. Based on the same technology, Pitchmap Colors is intended to make more obvious transformations to the source material. It splits incoming audio into chromatic pitches, then shifts them to match notes on an on-screen keyboard, in a MIDI track, or in a live MIDI input. So any piece of music, in any key, can be conformed to fit the notes or chords you choose. Interestingly, Pitchmap Colors can apply harmonic characteristics to atonal or noise-like sounds, allowing even ‘industrial’ sound recordings to be turned into something melodic. $ Pitchmap $169. Pitchmap Colors $99.

W www.zynaptiq.com

Zynaptiq Pitchmap Colors

www.soundonsound.com / February 2025

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Hear The Sound W https://open.spotify.com/

track/2jGde7Y7Ee3EMLb6HxPsHo

W www.youtube.com/

watch?v=SQaGCNGgfe4

Steve Hackett • ‘Every Day’ J O E M AT E R A

S

teve Hackett rose to fame as guitarist with prog-rock legends Genesis, and was one of the very first practitioners of the revolutionary two-handed guitar technique that would later be popularised by Eddie Van Valen.

After he left the group in 1977, the innovative and virtuosic guitarist embarked on a prolific solo career which continues to this day. His sonic calling card is a guitar tone he created in 1979 on his album Spectral Mornings.

Pure Gold “My favourite sound is the guitar sound I achieved on the track, ‘Every Day’, which I call my ‘golden tone’ as it’s become part of my signature sound.

It’s best described as a Les Paul guitar drenched in echo that floats, flies and

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cries, and at times sounds a bit like a distant violin or string section.

“When it came to achieving that sound, at that time of recording, the Roland Jazz Chorus JC-120 amps were first appearing, and they were the only thing that gave you that chorus-y sound, which came from the amp’s onboard circuit. I was using a couple of those amps and an extra speaker as well, so it gave me a spread of three different cabinets. The thing that accompanied all of that, too, was a Macari’s Coloursound fuzzbox, which was a British-made fuzzbox

track, which was created by the use of an EBow and a bottleneck slide, and sliding down to the bottom.

Dutch Courage “I remember that we were working at the Phonogram Studios in Hilversum in the Netherlands in early 1979. Being Phonogram’s headquarters, which included this extraordinary studio, they also had a radio station too. I can’t remember what mics were being used at that time, but I know that it was a big space so I was able to get a big guitar sound when I cranked it up. Those miking things were left up to my co-producer and engineer John Acock, who was a great engineer, and I credit him with getting me this wonderful sound on that track.

“It was quite spontaneous when it came to capturing that sound. And I really don’t how we managed to do that at that time. I seem to remember that I was quite fuelled up on Scotch and Coke as well, because it was literally freezing cold in Holland when we were recording it. So, all of those elements combined in creating that sound. “‘Every Day’ ended up being the opening track on Spectrum Mornings, an album that saw me return to the charts and allowed me to put together a touring band that saw us tour all over Europe.”

Steve Hackett: “At that time of recording, the Roland Jazz Chorus JC-120 amps were first appearing, and they were the only thing that gave you that chorus-y sound.” that duplicated the sound of the Marshall Super Fuzz. And using the chorus through the amps, I also added an Echoplex and a shallow volume pedal to the chain. And finally, for guitars, I used my 1957 Les Paul goldtop exclusively. Another thing that ended up solidifying that sound was this one long sustain note that I added to the

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TALKBACK

Brueder Selke & Midori Hirano WILLIAM STOKES

B

rothers Sebastian and Daniel Selke and Kyoto-born Midori Hirano are Berlin-based producers and artists who have come together to produce a collaborative album called Split Scale. The concept behind the record is simple — each piece follows the Western scale of A to G from beginning to end — but the results are deep and varied.

At the moment I can’t stop listening to Midori Hirano: If I had to pick one, Lucrecia Dalt’s soundtrack album The Baby would probably be my choice. I love the weirdness of her sound, which still manages to sound pop. Even though it was made as a soundtrack for the TV series, it is full of her signature sounds — her weird vocals, warm merengue-like rhythmic grooves.

Sebastian Selke: Twin Color, a solo project from Mexican producer Murcof. I’ve been following Murcof’s modular-experimental soundscapes for many years now. On his first EP as Twin Color he built this futuristic-nostalgic ambience around the voice of his young daughter Alina, and that really impressed me. My daughter Alma is currently trying her hand too, mostly at the cello. The EP was very inspiring for both of us.

Daniel Selke: The Guatemalan artist Mabe Fratti’s Será Que Ahora Podremos Entendernos is, like all of her previous albums, a revelation for us. I think she succeeds at the almost unique balancing act of giving accessible song structures a sufficiently exciting experimental aesthetic.

On top of this is her rudimental cello playing which follows an ideal all its own.

Just recently, Sebastian, his five-year-old daughter Alma and I were lucky enough to attend a concert by Mabe in Berlin. All three of us were blown away.

The artist I’d most like to collaborate with MH: After making this collaborative album

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Split Scale with the wonderful Brueder Selke, I’d like to keep myself open to more unknown possibilities rather than hinting at my musical expectations by mentioning any artist’s name!

S: We’ve been admiring Laura Cannell for a long time. We love music that is able to transport the audience to another time while remaining absolutely contemporary. Just her changing repertoire of instruments — and her voice — move us every time.

D: We’re also big fans of the Lippok brothers. They founded the band and platform Ornament & Verbrechen in East Berlin in 1983. With their groundbreaking art and electronic music, they influenced not just the underground music scene but the entire art scene of the late GDR years, as well as contemporary movements in Europe after 1990.

The first thing I look for in a studio MH: It really depends on what I want to do that day. It can be modular synths, which I’ve been into for a couple of years, if I want to try something new. Or it could be a piano, if I just want to practise or if I have some specific ideas about what I want to record with it. It could also be a keyboard synthesizer like the Casiotone 701 for some warm analogue organ-like sounds, or the Microkorg XL for a sharper digital bass sound.

S: We pursue everywhere and always a free creative spirit. Sometimes a special cable is all it takes to bring inspiration.

Or custom-made pots on an analogue console. Or an engraving on an instrument!

D: Yes, we are particularly interested in instruments. We also love studio technology, but ultimately it’s the instruments that ‘do the talking’. Just recently we acquired a beautiful old harmonica which now adorns our new studio table.

The person I would consider my mentor MH: I would say my friend Frank Bretschneider, who I’ve known for over

15 years and have admired his music for even longer. I first got to hear his music in the early 2000s through his releases on Raster-Noton, and I found his 2007 album Rhythm fantastic. Quite minimalistic and sharpened sound, but still humorous. I got to know him shortly after I moved to Berlin in late 2008 and we have been going to the same concerts a lot since then. He always has an opinion about what he hears or sees, and I never get bored listening to him. He often gives me his thoughts when I send him my new music; he can be very critical sometimes, but that way I know that the praise he gives me is not fake!

S: My daughter Alma is now five years young. When the three of us went to the aforementioned Mabe Fratti concert, Alma just started dancing quite naturally to the beat of the quite experimental songs. And she expected me to join in, which of course I did. As a classically trained musician, these moments give me new energy.

D: Our dear parents aren’t themselves musicians but they introduced us to music early. We remain grateful to them to this day that we could immerse ourselves in this universe.

My go-to reference track or album MH: Lucrecia Dalt’s The Baby, which I already mentioned! I’ll say that’s also my go-to reference. I think it’s well mixed, especially how she places each percussion and instrumental sound. Alternatively, utp_

by Alva Noto and Ryuichi Sakamoto with Ensemble Modern. I often find it difficult to do the mixing for music that is quite minimalistic but consists of real instruments and digital sounds. I think utp_ is one album that successfully manages to sound wonderfully atmospheric, while also having enough of that crispness.

S: For me, Severant by Kuedo is a brilliant album. The combination of atmospheric retro-futuristic synth pads and complex rhythmic figures — I’ve listened to that over and over and I still listen to it. It’s great for

it would be great if I had a chance to work or even play together with him. His 2001 album Insignificance is one of my long-time favourites and I still listen to it, over 20 years later.

S: Carsten Nicolai, whose pseudonym is Alva Noto, came from what is now Chemnitz, formerly Karl-Marx-Stadt. So he was also born in the former GDR and as an artist, composer and musician he has done everything!

D: Exactly. Be it film projects, album production, installations for the Guggenheim Museum, the Venice Biennale or the documenta... Carsten Nicolai looks for the energy in pared-back representations.

A minimalism that is infinitely powerful.

waking up in the morning, for motivation at lunchtime and for driving home after work in the evening when the stars light up the night sky!

D: The collaboration between Alva Noto and Ryuichi Sakamoto is also very, very inspiring for me and for us. Their debut album Vrioon was released in 2002 as part of their Virus series and it opened worlds for us.

My secret weapon in the studio is MH: I would say the Roland RE-301 Chorus Echo, which I took over from my good friend and great vibraphonist Masayoshi Fujita before he moved back to Japan from Berlin a couple of years ago. A beloved tape delay from the ’70s. I use it when I want a sound played on a synth or piano to sound a bit wobbly. When I have time and want to really immerse myself in sound waves, that machine gives me good inspiration.

S: The Vermona E2010 filter bank is such a special graphic equaliser. In contrast to the usual RC filters, it has LC filters. The peculiarities of its resonance behaviour produce beautiful, overtone-rich sounds with a warm vintage character. It’s ideal for transforming cold, tinny samples into warm and powerful analogue sounds. If you want more extreme, experimental results, you can connect the two stereo channels in series with a cable; then you can work with a 24dB filter slope and with extreme settings modify the audio material in a really hardcore way that dramatically changes the character of the input signal. Clean mastering filters aside, it is an analogue equaliser with a sound that is all character!

D: We should also mention that we had to

sell a lot of equipment during the pandemic!

But we really never want to be without our monitoring system. Geithain RL940s are, for us, absolutely indispensable.

The studio session I wish I’d witnessed MH: CocoRosie’s debut album La Maison De Mon Rêve is still my favourite album of theirs, and I remember listening to it over and over again after it came out in 2004.

Later I stumbled across their live video on YouTube, where they were playing and singing the song ‘Madonna’ with just a guitar and a toy keyboard. It seemed they were playing in a small house, which I suppose was still possible before they became big.

I really wished that I could be there. It was a very simple concert that showed the pure beauty of that song.

S: In GDR times, electronic music for us consisted mostly of Vangelis records. So many times we lay awake enfolded in his inimitable sound, dreaming that we were in his Nemo Studios. We could see it so clearly in our imagination. It awakened in us the longing to record similar music ourselves, with our own instruments, in our own studio, to produce and release music.

D: I’d also like to mention that for us, there has never been a better harmony between image and music than in Godfrey Reggio’s experimental film Koyaanisqatsi with music

by Philip Glass.

S: That’s right! Powaqqatsi and Naqoyqatsi were good successors, but Koyaanisqatsi is simply a masterpiece.

The producer I’d most like to work with MH: I have appreciation for Jim O’Rourke’s arrangement and mixing skills. I think

The studio experience that taught me the most MH: I couldn’t tell you. The gear and instruments I have in the studio are quite minimal and I work mostly on my own. But when it comes to the process of composing and recording, I have learnt that I just need to be patient and concentrate — on what I am doing and what I am hearing.

S: Long before the Berliner Funkhaus in Nalepastrasse, which is right on the Spree [river], was flooded with concerts, we regularly visited our dad at work there. An almost family feeling, and Santa Claus even visited at Christmas. The countless studios were a whole universe, and to this day we still dream of the giant RFZ mixing consoles!

D: And the Wuhlheide and the Pionierpark was just nearby, with its recording studio workshop. A paradise for children.

The advice I’d give myself of 10 years ago MH: I would tell myself: learn more production techniques and try to stick with each instrument that you want to master, rather than looking for a new one too quickly. And don’t be disappointed if things don’t develop as quickly as you expected! Many good things are worth taking time to develop.

S: Through Alma I’m reliving my childhood again. What a gift! I am infinitely humble and grateful. And so, among other things, I try to teach her a healthy understanding of mistakes at an early age. They’re a part of life.

D: That’s right. Alma is my niece, so I also try to help her with my experiences: to never lose focus, and at the same time to learn joy and humour! Be patient, especially when starting out in music. There is a universe hidden in every element, a treasure waiting to be found and unearthed. www.soundonsound.com / February 2025

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Emergence Audio Flute Textures Kontakt Instrument

★★★★★ Emergence Audio apply their Infinite Motion Engine and Non-Static Sampling technology to another member of the woodwind family in their latest release, Flute Textures.

As with their clarinet, piano and string instrument releases, this Kontakt instrument offers pristinely recorded source material enhanced by sound-design techniques, making it a versatile and evocative tool for a creative composer.

The core sounds, performed by Vienna-based flautist Veronika Vitazkova, have a beautiful, sweet tone, and this wonderfully clear and open tonal quality holds throughout the entire range of the instrument.

It’s not all sweetness and light with this library, though. Once you load the Grace Notes preset, you’re ready to unleash increasingly dissonant compositions upon your audience. The preset (and its variations) combine whole-step and half-step intervals to produce an unsettling, slightly unhinged effect. The creepy continues with the Multiphonics, Abyss and Astral presets, which are great starting points and provide feedback for musical ideas to develop as the sounds evolve. To explore some minimalist randomness, try the Triple Tongue preset.

Once you venture into the Experimental presets and beyond, your composition becomes cinematic, whether you like that oft-overused term or not. Background Radiation, for example, immediately brings to mind Ludwig Göransson’s work in Oppenheimer. Elsewhere you hear hints of Trent Reznor and Atticus Ross, Tom Holkenborg aka Junkie XL, Hans Zimmer, and the late, great Bernard Hermann. This is not to say that the sounds are derivative at all; their textures simply evoke feelings in a way that is similar to what the best and brightest contemporary musical minds do.

The UI and performance of the library is impeccable. Those familiar with Emergence Audio releases will be pleased to see the familiar layout — a clear, well-organised Main Page with drop-down menus to select presets, ADSR control, LFO parameters, panning knobs, high- and low-pass filters, and a crossfade dial, all easily assignable to your MIDI keyboard. The Effects Page

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contains controls for convolution reverb, delay, chorus, tape saturation, distortion and other effects that can be applied to both layers in a patch, or to individual layers, with the simple click of a button.

What this library demonstrates, and what you as a composer can explore with great ease, is the world of expression that lives within the humble flute. One moment it evokes the calm of a dewy winter morning, the next, a jolt of terror from movement in the dark where none should exist. As a composer, it’s wonderful to have access to it all in a well-designed, smoothly functioning library like Flute Textures.

Sonal D’Silva $99 www.emergenceaudio.com

Dreamtonics Felicia Voice Database For Synthesizer V Studio Pro

★★★★★ Music technology has given us some truly amazing virtual instruments and, in 20+ years of writing product reviews for SOS, I’ve been lucky enough to have my jaw dropped by some of the very best of them. The most recent addition to that list would undoubtedly be Dreamtonics’ Synthesizer V Studio (see the March 2023 issue), which, somehow, manages to generate genuinely realistic lead vocals in software. Dreamtonics have continued to add further voice database expansions to their product line. These are produced in collaboration with an actual singer whose voice is appropriately sampled, and those samples are then used to create the data needed by Synth V’s engine to sing whatever melodic or lyrical combination you choose to compose, along with options to fully customise the style of vocal delivery (soft, tense, powerful, etc). Each voice offers different combinations based upon the original singer’s vocal styles.

The latest voice database addition is Felicia. This is based upon a native English singer and, while the engine lets any of the voice databases sing in English, Japanese, Chinese or Spanish, it’s undoubtedly easier to get the best from Synth V when it is generating vocals in the original singer’s own language. Alongside the singing expression controls offered for all voices (loudness, tension, breathiness, gender and tone shift), Felica offers five voice-specific vocal modes; airy, bright, musical, operatic and powerful.

Dreamtonics suggest the voice is suitable for a number of genres such as pop, soul, electronic, opera, classical, alternative rock and metal. Having experimented with the voice database, I think that’s a pretty good summary. Lots of the voice databases can easily be used for more electronic music styles (where vocal production styles are not necessarily striving for a natural sound), but acoustic pop meets singer-songwriter styles are also well within Felicia’s capabilities, while ‘show tune’ styles are also easily dealt with. In terms of more aggressive styles such as punk, indie rock or metal, Felicia voice modes perhaps don’t take her into full-on grit or growl, but she can certainly be made to belt and sing with power. And, if you like your metal to be symphonic, she can do that too.

If you want to hear some examples, check out Dreamtonics’ excellent demo song (‘Show Me’ was created in collaboration with UK producer/songwriter Nick Trepka) or the cover of Paramore’s ‘All I Wanted Was You’; a quick YouTube search will easily find both. Yes, as with any virtual instrument — piano, drums, bass or orchestra — it’s perfectly possible to end up with something that reveals its ‘virtual’ origins but, equally, once you understand that software’s capabilities, you can also create results where the vast majority of listeners will not even question its ‘human’ status. Given just how complex the human voice is, that’s a remarkable thing to be able to say. Each of the new voice databases does seem to nudge the level of realism

forwards. The Felicia voice database does just that and Synth V continues to amaze me every time I use it. John Walden $79 www.dreamtonics.com

Zero-G Cartago

Kontakt Instrument

★★★★ Released by Zero-G, Cartago is a collaborative production from Giuseppe Caiazzo of Silence+Other Sounds and Stefano Maccarelli. Both have extensive experience with sample libraries aimed at composers and Cartago, which is built for the full version of Kontakt (6.8 or above), features a collection of cinematic-flavoured drum and percussion sounds, and is aimed squarely at that same audience.

The instruments include both acoustic sources and some ‘designed’ options, drawn from the likes of orchestral bass drums, timpani, snares, frame drums of various types, taikos, darbukas, djembes and a number of more exotic options. The 13GB of sample data means you are getting some deeply sampled instruments, featuring multiple round robins, articulations, dynamic layers and microphone positions.

Inside Kontakt, Cartago actually provides two very different front-end experiences for the user; the Acoustic Percussions Sequencer (as shown in the screenshot) and the Designed Drums Kit. The APC presets let you load and layer three of the deeply sampled drum/percussion sounds.

These can sound pretty epic when triggered manually but, some comprehensive effects options notwithstanding, the really interesting element if you need quick results is found in the Sequencer panel. This offers four ‘scenes’ (different sequence patterns) that you can keyswitch between and control the overall dynamics of via a combination of the Expression, Scale and Tight controls.

The sequencer engine has its own preset library and user sequences can easily be created. It’s effective, easy to use and makes for instant performance inspiration.

The Designed Drums Kit lets you build your own drum kit across a conventional 4x4 pad grid with over 2200 one-shot (and some multisample) options to choose from. These samples are divided into five categories — cinematic impacts, low-end kicks/thumps, designed snares, percussions/raw sound and whoosh/rise/ SFX — and, whether you want traditional

sounds or something more modern, there is plenty to explore. Again, the features and functionality of the front end are actually rather impressive and provide all sorts of interesting ways to map and layer the sounds, as well as a whole slew of effects options for further sound design.

Both producers know their way around the sampling process so, as ably demonstrated by the impressive collection of presets for both engines, Cartago most certainly delivers in terms of sounds. Some care is needed with some of the APS presets as you can easily end up triggering quite a lot of samples (so things can get pretty demanding in terms of computer resources) but, if you like your UIs functional rather than super-slick (this is perhaps not the prettiest Kontakt instrument I’ve ever used), the functionality is equally impressive and both engines offer very accessible performance options that will appeal to the busy composer.

Yes, the cinematic drums marketplace is a crowded one, but Cartago can certainly stand its ground sonically and is also very competitively priced (even more so if you pick it up at the discounted launch price).

Well worth auditioning. John Walden $119 www.zero-g.co.uk

Wrongtools Osmosis

Kontakt Instrument

★★★★ While there are plenty of cello samples for those who want to recreate orchestral arrangements, Osmosis takes a somewhat different direction, providing a wide palette of articulations and experimental techniques that inspire creativity. Apparently the library takes its inspiration from experimental film scoring, with different takes and articulations being layered to create sounds that inspire the user to seek new compositional directions. The ensemble size can be from six to 30 celli, though there are also single-instrument patches, all recorded by musicians from the Stavanger Symphony Orchestra. Articulations can be changed by keyswitching or MIDI CCs, with the option of adding random sample start positions,

and the Pendulums category of patches can be crossfaded using the mod wheel.

The mic setup is as impressive as the concert hall in which the recordings were made: Close, Decca Tree, ST-MS, Flank and Distant. Weighing in at around 5GB, the library comprises 80 factory patches, organised into four categories (Multi-mic, Main, Pendulums and Tainted) created from over 1550 samples.

In some instances a patch comes over like a short musical performance or motif, while others may be evolving textures, unusual bowing techniques, ghostly flautando harmonics or stuttering bow bouncing. This variety provides scope for pads or embellishments to more conventional string parts, spot effects, shifting soundscapes or inspirational starting points for new compositions. I suspect that there was a lot of improvisational playing and experimenting going on during the recordings as the players interacted instinctively — the results are truly organic.

From delicate to diabolical, you should find something to suit the required mood.

Osmosis is more than a simple sample player, however, as it also includes Wrongtools’ Sculpturing Engine that can transform sounds in creative ways via layering and morphing and other techniques, plus there’s provision for filtering. There are so many different styles of performance that you really need to visit the Wrongtools website to check out a few examples before you can appreciate the breadth of this library. Despite the experimental nature of the content, the essence of the cello still comes across very strongly (though some patches in the Tainted section stray rather a long way from what you might expect from a cello) and slotting some of these sounds into a composition really does give the impression that a real performer was part of your session. There’s a lot to explore here, and that’s before you start making your own changes, so if your compositions need an experimental edge that still retains the organic character of real celli, Osmosis is most definitely worth taking seriously.

Paul White €188 www.wrongtools.com Audio examples of this month’s libraries are available at www.soundonsound.com.

www.soundonsound.com / February 2025

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Q

How should I connect one mixer’s stereo line output to another’s inputs?

The safest way to connect your line-level devices to a front-of-house mixer is to feed them through the mic preamps via a good DI box. For stereo devices such as the mixer discussed here, a stereo DI box such as the Radial Pro D2 is convenient.

I sat in on a session last night with some guys who use a Soundcraft UI24R mixer. They used a TRS-TRS cable from my personal mixer’s line output to route my vocals into the Soundcraft. All worked well, but it seemed like there were far more XLR inputs than TRS ones, and I wondered if they’d have been happier if I’d given them a TRS-XLR cable instead?

This got me thinking: are mixers smart enough to know when a signal coming into the XLR is at line level and to adjust accordingly?

SOS Forum Post Hugh Robjohns, Technical Editor The Soundcraft UI24R has 10 combi XLR inputs with TRS sockets at their centre, and 10 more straight XLRs. They can all accept line-level signals (up to +19.5dBu), with the appropriate gain being set manually with the corresponding channel controls. I don’t know of any mixers that can switch automatically between mic and line input levels, but plenty can accommodate a vast range using the input gain control and/or pad or line switches.

But there are some important differences between the TRS and XLR inputs on the Soundcraft, and most other desks. For a start, the TRS inputs in the combi XLRs present a moderately high 12kΩ input impedance, whereas the XLR inputs are 6kΩ — but both values are perfectly acceptable in the context of your personal mixer outputs. A more concerning difference is that phantom power can be enabled on any of the XLR inputs, whereas the TRS inputs are isolated from phantom power. It’s significant because a lot of line-level equipment is not designed to tolerate 48V DC across its outputs, and applying phantom power

can cause catastrophic damage to the source equipment in some situations.

On that Soundcraft desk, phantom power has to be manually enabled for each channel individually so, technically, you could connect your mixer’s line-level

“A lot of line-level equipment is not designed to tolerate 48V DC across its outputs, and applying phantom power can cause catastrophic damage.”

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output to an XLR input if the phantom is switched off. But can you guarantee that, in the heat of the action, someone won’t accidentally turn it on? Are you prepared to risk your mixer going ‘phut’? Connecting via the TRS inputs avoids that potential problem completely, and would be the simplest solution.

If you do a lot of sessions with different bands, or perform on stage with different PA systems and operators, I would strongly urge you to invest in one or more passive DI boxes and feed your mixer’s outputs to that, and the DI output to the other mixer. This will protect your system from phantom power and other electrical faults with the other people’s gear. It also provides easy ground-lift switching, should that prove necessary to stop ground-loop noises. What’s more, the output of a DI box is a standard, balanced mic-level signal that can be plugged into any stage box and everyone knows exactly how to deal with it at the mixer — there’s no confusion over line level or mic level, high or low impedance, phantom or no phantom. Just plug it into a channel mic preamp alongside all the other mics and it’s job done.

If you’re working in stereo or need to pass two channels, there are two-channel DI boxes that are very convenient. For electronic outputs, such as from your mixer, a passive DI box is absolutely fine and less expensive than an active box. Active boxes are fine too, though they need battery or phantom power, which a passive box doesn’t (but it won’t care if phantom is present).

In my own rig, the outputs always go via Radial Pro D2 passive two-channel DI boxes. I wouldn’t risk direct connections to any system I wasn’t 100 percent confident about — which means on any live stage or rehearsal room!

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MY KEYBOARDS JOHN MCLAREN-HOWARD

P

iano lessons from seven years of age and pipe organ fascination from the age of nine began a passion for keyboards. Now, aged 82, I love my keyboards, They include several synthesizers and two arranger keyboards, and access to a grand piano and three-manual (electronic) church organ. I play the piano badly and am no more than an amateur organist.

Along the way, I have built divider organs and embraced synthesis and sampling, but it is the keyboard that gives me the way in to express and experiment with the sound world in my head.

Korg’s OpSix, WaveState and ModWave, Roland’s Jupiter-XM and Gaia-2, and an ASM Hydrasynth each have a place in the expression of what I hear in my head. MIDI controls these instruments — most recently driven by a Yamaha CK61, which contributes welcome bread-and-butter sounds.

I record my ‘dreams’ to a Rode Podcaster. I do not crave or need a high-end keyboard.

A humble synthesizer keyboard or the simple CK61 are more than adequate. But, I do love my keyboards, where I can

easily lose hours of my time exploring seemingly endless sound possibilities.

Whether I really succeed in getting down what I hear in my head is a matter that only I can judge, but just occasionally, I do get it right. Most of the time it is a near miss, but at least it is out of my head and satisfies something basic in me.

Something impossible without the keyboards I love.

Over the years, Sound On Sound has been a major source of useful, reliable reviews of new gear and introductions to new techniques. However, the excitement of trying these innovations has almost universally depended on the ‘humble’ keyboard and my not-so-brilliant skills. For a brief period I considered a change from hardware to software, but my computer skills proved limiting, and anything that removed me from the hands-on, keyboard-driven approach never seemed to quite satisfy me. I can really understand why so much creation and production of music now takes place in the computer/software world, and I don’t decry any of this, but I just love my keyboards and the hands-on approach using hardware.

Manufacturers continue to tempt us with hardware

“There is much to admire and marvel at when one hears the remarkable keyboard skills of today’s performers...”

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synths and I am astounded by the range of techniques and remarkable user interfaces developed. Many excellent composers expect to drive these creations from their DAWs, and the available software is indeed remarkable.

So is there still room for simple, direct keyboard input to these new synthesis engines? There is for me, but am I in a dying minority? Yet, there is much to admire and marvel at when one hears the remarkable keyboard skills of today’s performers, spanning everything from classical music to the most extrovert modern composition or improvisation.

Surely, we have to appreciate and admire the skill and the often unimaginable amount of time that goes into the preparation and presentation of music in its many forms. It is great to have opportunities to observe these processes, even greater to follow that long path in our own musical journey. A good way to realise a little of what is involved is to set oneself the task of preparing a new composition and demanding excellence in the process.

It would not matter what equipment or computer programs were used, but a nod to keyboard skills remains a learning process in its own right.

Just recently, at the invitation of a female vocalist and her producer, I had the privilege of observing a recording of a new work in a high-end studio. What

captivated me most was the fanatical attention to detail and the ‘density’ of concentrated effort by everyone involved.

It was punctuated by short episodes of manic humour, but the professionalism shone through it all.

Readers will not be surprised that my interest centred on the major contribution of the keyboard player and his interaction with the other musicians (mainly strings).

His constant interchanges with and support of the vocalist were a joy to behold.

Recorded versions of this four-minute piece spanned everything from rubato-rich, exciting free-and-easy to a highly restrained ‘quantised’ pristine interpretation of the score. While it was clear to see (and hear) that the most extrovert version was favoured by all of the musicians, I was shocked to hear the producer announce that the most restrained version was his choice. My friend, the vocalist, simply shrugged her acceptance and said that her fee was the same whatever the chosen version.

It was the keyboard player who lost his cool, which he did in a very vocal way, illustrated by a keyboard demonstration that took my breath away.

He did what I would love to achieve. He took his deepest feelings and expressed them through his playing.

I have only a tiny fraction of his talent but, like him, I love my keyboards.

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