SOUND ON SOUND [UK] July 2025

SOUND ON SOUND THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

1985 — 2025 TM

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

JULY 2025

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M A K I N G Jung Jae-il

The composer behind Squid Game

VOLUME 40 • ISSUE 9

How to play to a click

S K E L E TÁ

Introducing...

SOS For Artists Dynaudio Acoustics M1 MkII A legend returns!

TANTRUM AUDIO

ANGRY SYSTEM SPEAKERS + SUB WORTH £1640 www.soundonsound.com

ON TEST: AUDIX / SOUNDTOYS / UJAM / EVENTIDE / TRITON / BOSS / XVIVE / BABY AUDIO / KORG / ANALOGUE SOLUTIONS / DREAMTONICS

TECHNIQUE: HOW STEREO WORKS / PRO TOOLS PLAYLISTS

July 2025 £7.99

Exclusively distributed in the UK and Ireland by Source T: 020 8962 5080 • W: sourcedistribution.co.uk/rode

LE ADER

SOS FOR... EVERYONE The idea that the ‘subscription model’ is a new thing is greeted with wry smiles at SOS Towers, where our print edition has been available on subscription for almost 40 years. We were among the first magazine publishers to offer digital alternatives, and today, more than half of our readers choose to view our articles on our website, through downloadable PDFs or in our dedicated apps. You could say that Sound On Sound knows a thing or two about subscriptions.

So when we thought about how to apply that experience beyond the world of magazine publishing, we didn’t want to offer something underwhelming. With hundreds of companies vying for a monthly fee, we knew we had to go the extra mile in order to stand out. And I think we have done exactly that with SOS For Artists.

The concept is simple: SOS For Artists should make a difference at every stage of the music-making process. For music creation, you’ll have access to thousands of samples, plus virtual instruments and even DAW software. For mixing, you’ll draw on a powerful selection of plug-ins, as well as the industry’s leading stem separation and AI mastering tools. And when it comes to getting your finished music out there into the world, SOS For Artists is a complete one-stop shop for distribution, aggregation, promotion, royalty reporting and more.

ALLIA BUSINESS CENTRE KING’S HEDGES ROAD CAMBRIDGE CB4 2HY T +44 (0)1223 851658 sos@soundonsound.com www.soundonsound.com

I’ll admit that I was initially sceptical. What could SOS For Artists offer someone like me, who rarely uses samples and has more than enough plug-ins already? I soon learned the answer, which is that each of its main strands is worth the asking price on its own.

In my case, the sample and plug-in libraries are simply a nice bonus. The ‘here, take my money!’ feature is the ability to release tracks on over 150 platforms simultaneously, with metadata, licensing and promotion all handled from one central hub.

And that’s just the start. We have ambitious plans for SOS For Artists. New content will be coming online every month, including video tutorials, and very soon you’ll be able to make real money not just by releasing music, but by sharing your unique skills and expertise.

I hope this little taster has whetted your appetite. Even if you’re one of those people who is rightly cynical about some subscription services, I’d urge you to take a minute to dive into what SOS For Artists offers. Check out the introductory article that kicks off this issue of the magazine, or point your web browser at www.sosforartists.com to find out more. I think you’ll agree that it’s a pretty compelling offering.

See you there!

EDI T O R I AL

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Managing Director/Chairman Ian Gilby

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Editor In Chief Sam Inglis

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Designer Mick Reilly

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“I’d urge you to take a minute to dive into what SOS For Artists offers.”

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The contents of this publication are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publisher. Great care is taken to ensure accuracy in the preparation of this publication but neither Sound On Sound Limited nor the Editor can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the Publisher or Editor. The Publisher accepts no responsibility for the return of unsolicited manuscripts, photographs, or artwork. © Copyright 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.

You Can’t Unhear Quality

Award Winning Audio Interface Range | Class-A Audient Console Mic Preamps Class Leading 32-Bit Converters | JFET Instrument Inputs | Headphone Outputs USB-C | Balanced Inserts* | Ultra-low Latency Software Mixer | 24bit / 96KHz Advanced Monitor Control | Audio Loop-back | 3 Year Limited Warranty Free Plugins & Software Included

*Balanced Inserts feature on iD24, iD44 & iD48 only.

110 HIGH CONTRAST

IN THIS ISSUE

www.soundonsound.com

July 2025 / issue 9 / volume 40

FEATURES

WIN

8 SOS For Artists We explore the key features and benefits of the exciting new SOS ecosystem for music creation & distribution.

44 Composer Jung Jae-il South Korean dramas Parasite and Squid Game have been worldwide hits — and Jung Jae-il’s scores have played a key role in their success.

54 How & Why Stereo Works In the first part of a new short series about stereo recording techniques, we start by exploring the most important tools of all: your brain and your ears!

62 How I Got That Sound Producer and mix engineer Stephen Power explains how it took several days and a lot of tape to create the layered drum sound on Robbie Williams’ 1998 single ‘No Regrets’.

72 Modular

C O V E R

XAOC Devices’ Maciek Bartkowiak on the potential of modular and pushing the boundaries of analogue circuit design.

74 Inside Track: Ghost Swedish hard rock band Ghost have topped charts worldwide — with an album that was mixed twice!

TANTRUM AUDIO ANGRY SYSTEM WORTH £1640 PAGE 58 110 High Contrast: Making Drum & Bass Lincoln Barrett’s musical journey started in the box but, 20 years on, he’s reinventing his trademark liquid funk sound with vintage hardware.

116 Spotlight: Multi-Tier Keyboard Stands

90 The SOS Guide To Playing To A Click

Is your synth collection spiralling out of control? These multi‑keyboard stands could help you keep your studio in order!

We look at what you need to know to program a click track and play along to it successfully.

136 Q&A

100 Talkback

Your studio and recording questions answered.

Rachika Nayar and Nina Keith, aka Disiniblud, on Hildegard of Bingen, abusing Ableton time-stretching, and Google as a mentor.

138 Why I Love... Tiny Gear Lawrence Joseph tells us why for him, small is beautiful.

64 BOSS TAE CORE

120

ON TEST

ADPTR Audio Utopia Reverb Plug-in

84

Archaea Exchange Eurorack Module

24

Arturia MicroLab Mk3

18

16

Ashton Research Tetrapatch

120 Electro-Harmonix LPB-3

36 Eventide & Third Man Records

Audix OMX Series

Fuzz Synth Pedal

34

28 Baby Audio

Smooth Operator Pro

Resonance Suppression Plug-in

Reactive Load, Speaker Simulator & Re‑amplifier

32

Book

102 22

40 92

Dynaudio Acoustics M1 MkII Passive Monitors

Triton Audio Kompressor Mic Booster & Compressor

14

UJAM UFX Filter Filter Plug-in

Void & Vista Frames

125

Flatsix Seventh Summoner

124 Westwood Instruments

Sample Library

Novella Origin & Novella Blackwater Sample Library

70

Korg microKORG2

XAOC Devices Lublin Eurorack Module

Jamstik MFC1 MIDI Floor Controller

106

Xvive More You Audio Interfaces

Synthesizer

121

Lake People G103-D-4 Dante Headphone Amplifier

12

Software Vocal Synthesizer

96

42

Ferrofish Pulse 8 AE

Sample Library

60 Dreamtonics Synthesizer V

Studio 2 Pro

Tone-shaping Plug-in

& Damage Rock Grooves 2

Drawmer MQ-1 & MQ-2 500-series EQ Modules

Tone Architect

124 Heavyocity Damage Guitars 2

Cong Burn Stacks Software Instrument

122 Teletone Audio

Eight-channel A-D/D-A Converter & ADAT Router

Eurorack Module

Complete Guitar Wiring

Soundtoys SpaceBlender Reverb Plug-in

Boost & EQ Pedal

Knife Drop

Boss TAE Core

20

Linear Power Boost

RJ45 Patching System

Dynamic Microphones

64

EastWest Ancient Kingdom Sample Library

Controller Keyboard

52

Woodwinds & Brass

Sample Library

Analogue Solutions AS500-Seq Analogue Step Sequencer

70

86 Soundpaint Solo

Newfangled Audio Articulate Envelope Shaper Plug-in

125

Song Athletics Friends Sample Library

WORKSHOPS 126 130 132 134

Studio One Pro Tools Cubase Live

TECHNIQUE

You get instant access to up to 2400 samples per year from a library of millions, covering almost every instrument and genre imaginable.

Create • Produce • Distribute We explore the key features and benefits of the exciting new SOS ecosystem for music creation and distribution.

THE SOS TEAM

C

onceived to help independent artists, producers and collaborators make music and share it with the world, we’re proud to announce SOS For Artists, a new subscription service that packs all the digital tools and services that are needed to take your music from idea to final release into a single, convenient and affordable environment. Beginners will find everything they need to make music and share it with their friends, bandmates and, if they wish, the rest of the world. But more experienced composers and producers will also find plenty here to complement the tools they already own and use.

We plan to further develop our service in collaboration with a range of commercial partners over the coming months and years, but already you get access to one of the world’s biggest sample libraries, complete with advanced sample browsing and previewing tools that integrate

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with your DAW. There’s DAW software to get beginners started, a good range of high-quality plug-in instruments and effects, and various other useful mixing and mastering tools. And when you’ve perfected your creations, you can use our comprehensive distribution service to release them on the major online streaming platforms, track your streams — and, hopefully, manage your income too!

In the coming months, we’ll be diving deeper into the features and potential applications of SOS For Artists in a series of workshop articles. But we’ll kick off this month with a quick tour of what’s on offer.

Create: Inspiration & Ideation 1. Sample Selection Made Simple. At the heart of our platform’s creative suite lies a comprehensive sample library, with over three million royalty-free offerings from acclaimed producers and labels. Updates are frequent, and new free sample packs will be offered to subscribers every week.

Use of up to 2400 samples annually is covered by your subscription, and you can preview as many as you want before downloading them.

Whether you’re trying to source specific sounds or simply searching for that creative spark, and whether you want to chase the latest musical trends or a more retro vibe, it’s quick and easy to find what you want. Samples can be filtered by tempo, key, genre, mood and instrumentation, so it’s quick and easy to find the right loop or one-shot — no need to burn time and energy wading through thousands of mismatched sounds.

Your discoveries can, of course, be marked as favourites and organised into folders for fast recall on future projects. But, better still, an advanced preview tool allows you to tempo-match and audition up to eight samples simultaneously, so that you can be sure they’ll work well together before downloading them to your DAW. And speaking of your DAW, there’s a dedicated plug-in version of this preview tool (supporting Logic Pro, Cubase, Pro Tools, Ableton Live, FL Studio and many more), so you don’t have to leave your project to find samples and quickly drag them onto the timeline.

2. Synth & Instruments Suite. As part of SOS For Artists, you get access to a suite of premium virtual instrument, synth and sample-player plug-ins.

Currently included are some popular plug-ins from the likes of Spitfire, UJAM, Arturia, Sonuscore and GForce, but while there’s already plenty of variety, covering a wide range of sounds and genres, we’re also working with a range of partners to build this collection over time. Rest assured, we’ll be actively curating this collection, carefully selecting only high-quality plug-ins and ensuring that you never face ‘overwhelm’ — you can just get on with the fun of creating.

You can install any, all or none of these plug-ins, as you prefer. All you need to do is click the install button at the side of the product listing on the SOS For Artists website, after which you’ll be given a licence key plus download and installation instructions.

songs to the SOS For Artists mastering page and perform the separation there.

With a typical song, it takes just a few minutes to deliver a handy set of tracks that can be imported into your DAW. But you can also choose to extract only what you need — an isolated vocal, perhaps — in which case it will be even quicker. Stems works especially well with drums, and will become even more useful as the technology behind it improves. Unlike some other stem separators, this one will work on the older Mac operating systems that we know plenty of SOS readers still rely on.

Some applications for this tool will be obvious, but others may be less so. For instance, not only can remixers or those who want to create backing tracks (whether for live performance or for practice) use the resulting audio files ‘as is’, but they can also be used with your DAW’s audio-to-MIDI tools to replace, say, bass lines, or to give you a starting point for sheet music

3. Stem Extraction.

A particularly powerful plug-in is LANDR’s Stems, which was developed

by AudioShake.

This tool can be opened in your DAW as an instrument and it allows you to extract stems from mixed recordings. You simply drop songs or loops into it for analysis, and it will then do a surprisingly accurate job of separating out Vocals, Drums, Bass, Instrumental and Other (ie. what’s left!) stems from the song or loop. Alternatively, you can upload your

transcription. It has the potential to speed up your sample searches too — if you find a loop with the right drums but the wrong bass line, then you can still use it. It could also be an invaluable tool for artists looking to revive or re‑balance old projects for re-release without having to record all the parts again. And yet another very useful

LANDR Stems: we reckon this powerful stem separation plug-in has rather more creative potential than you might think.

application for stem splitting is reducing the levels of unwanted spill in live recordings.

4. DAW Software. While we understand that most SOS readers will already have chosen their preferred DAW, we’re also keen to welcome new music creators.

For them — and perhaps those who’ve been thinking about switching DAW — we’ve included ‘lite’ versions of both Steinberg Cubase and Ableton Live in the sub, which can load any of the plug-ins and samples described in this article.

“You can switch distribution ownership without losing any of your play stats and followers — essential for any artist, producer or band who’s spent years building their brand!”

Just a few of the instrument plug-ins already bundled with your subscription — we’ll be adding more over time.

Produce: Shape Your Productions When you’ve got your ideas down, it’s time to polish them, and SOS For Artists provides a range of tools that you can use to produce them to a professional standard.

1. Effects & Processors. As with the instrument plug-ins, we’ve curated a great collection of effects and processors that you can use to mix and master your tracks.

Again, these can be installed in your DAW, they’ll remain active while you have a subscription, and it’s a collection we plan to grow over time. They cover the expected bases like EQ and compression, delay and reverb, distortion and saturation, along with some standout extras. Amongst the offerings you’ll find a couple of channel strips, Eventide’s CrushStation, Rev PLATE-140 from Arturia, Taip from Baby Audio, and IK Multimedia’s MixBox SE: a powerful effects rack that can run up to eight of its 24 included effects at once. You

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TECHNIQUE

SOS FOR ARTISTS

Some of the effects and processor plug-ins included in SOS For Artists. Again, we plan to add more over time at no extra cost.

also get access to Synchro Arts’ excellent vocal-alignment and pitch-correction tools.

2. Mastering. Here at SOS, we firmly believe that skilled professional mastering engineers have something unique and valuable to offer anyone making music.

But today’s content-creation environment means there are often times when artists and producers need to be able to put out a preview, get something out quickly on social media, or perhaps ‘finalise’ so many tracks or samples for libraries that it’s just not practical or affordable to send everything to a mastering engineer.

As part of SOS For Artists, you get access to the online and plug-in versions of LANDR’s AI mastering engine. This clever machine-learning-based system, which we’ve reviewed in these pages previously, has matured substantially in recent years. It analyses the frequency balance, dynamic range, tempo and stereo width of your mix, and then applies what it deems to be suitable processing for the type of music you’ve created. You can opt for different mastering styles to point it in the right direction, or let the engine make those decisions for you, based on its analysis of a vast number of commercial tracks. And if you feel the result is ‘nearly there’, then you have some controls with which you can refine the result.

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For those assembling full-length projects, the multi-song album mastering option ensures tonal and volume consistency across an entire release — it could prove useful for independent artists with limited funds, who may not have access to a mastering engineer when creating cohesive bodies of work.

Head back to the plug-ins tab and you’ll find some more mastering tools, including MixChecker from Audified, which allows you to preview your mixes on a range of emulated audio listening devices, including phones, iPads and earbuds.

Distribute: Unleash Your Creations!

When your track is polished and ready to go, SOS For Artists helps you shift into distribution mode — it streamlines the process of getting your music out to the masses, understanding who’s listening, and tracking your royalties, with no need to deal with the multiple different platforms directly.

1. Digital Distribution. We’ve teamed up with LANDR to give you access to one of the best online distribution services in the world. Subscribers can distribute unlimited music to all the major players and more — over 150 platforms, in fact, including Spotify, Apple Music, Amazon, Tidal, and Deezer — and the process is painless. Just upload

your audio files and artwork, enter your metadata, select your release date and off it goes. It’s super easy to migrate from other distribution services too.

The platform accepts the usual WAV, FLAC or AIFF files (individual stores process the files they receive to match their preferred formats). An ISRC can be added or automatically provided, and there are fields for adding featured artists, producers and remixers. You can also match your artist name to existing profiles at Spotify and Apple simply by selecting from a drop-down menu, rather than having to hunt for your URLs at these stores.

A licensed content section will scan your release for use of samples, and you can make a declaration about whether you’ve used AI-generated content — potentially a useful step in the future, as AI and creator content may well be presented separately on download stores. You can also add copyright info, label details and release artwork here.

A Release Settings tab allows you to specify which stores you want your music to appear in. The main ones are selectable individually, while the smaller stores are bundled into a single tick box, for speed and convenience. You can also choose to distribute direct to social platforms — very handy if you haven’t yet reached monetisation status on YouTube!

Releases can be set to go live within the following 2-7 days, or scheduled as part of a release campaign plan, and those who’ve licensed tracks to labels in different territories can restrict which countries can access their music. It’s then just a matter of approving all of the terms before clicking on the Submit button. Other useful time-saving options include the ability to clone releases, and a Royalty Split feature: for completed tracks, click on Manage and navigate to the Splits tab, where you can assign percentages to co-writers; anyone not on the platform already will be sent an email with sign-up details.

2. Switching From Other Platforms. For the many readers who will already have download store profiles and distribution services in place, an invaluable feature is the ability to import your catalogue to our service from other distribution platforms.

In other words, you can switch distribution ownership without losing any of your play stats and followers — something we know is essential for any label, artist, producer or band who’s spent years building their brand!

3. YouTube Official Artist Channel.

Likewise, it would be frustrating for any artist who’s worked hard to build a YouTube presence to end up with their official releases on a separate profile under ‘Artist Name – Topic’.

Link your YouTube channel to SOS For Artists, and you’ll ensure your channel subscribers and distributed content all sit in one place.

4. Cover Song Licensing. Yet another handy detail is that mechanical licence agreements for cover songs are handled automatically, with a one-off fee. There’s also a direct link to the Harry Fox database (www.harryfox.com) where you can search composer details for adding into your track metadata.

5. Promotion. When your tracks are ready to release, you’ll find various tools to help with promotion, including a linktree service, so you can send out a single URL with all your links in one place, and there’s a whole host of useful articles and videos with promotional tips.

6. Reporting Tools & Royalties. Finally, a comprehensive royalty reporting page lets you view trends, top tracks, top countries, stores and more, with filtering to allow you to dig deeper into the reporting data. You get to retain 100% of your royalties, and minimum payment thresholds are low, starting at just $1 for PayPal.

Ready To Go? That completes our initial tour. Whether you’re new to making music or just looking for good-value production tools and distribution arrangements, there really is something for everyone — it’s a comprehensive service for modern music makers that doesn’t cut corners and offers speedy workflows without diluting the artistic process.

There’ll be new features, tools and samples added in the near future, but we hope you’ll already find SOS For Artists as compelling a proposition as we do.

We’re offering early adopters a special ‘locked-in’ launch price of only £14.99 per month or £99 annually plus VAT — that’s 25% and 50% off the full price, respectively — and you have the ability to pause and resume your sub at any time without losing any of your profile, samples and projects too. Next month, we’ll be back with a more hands-on demonstration of how you can use some of the specific tools and features of this exciting new platform. But if we’ve already piqued your interest, why not head over to www.sosforartists.com, where you can learn more?

Whether you’re an individual artist or running a small label, you’ll find a comprehensive set of distribution tools for managing your releases, including the ability to import releases from other services without losing your play counts and playlist inclusions.

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ON TE ST

Newfangled Audio Envelope Shaper Articulate Plug-in

while Focused offers better-defined separation between the elements but may sound less smooth with some material — sometimes this fine control will be precisely what you need, but it’s a setting that’s definitely best judged by ear. There’s also a Mix parameter box, and this can be used to set the wet/dry balance of the processed and unprocessed signals. Further buttons bypass the plug-in, enable the side-chain input and activate a limiter, which is used to keep any output peaks (a risk with any transient‑shaping process) well under control. The four component faders range from fully off to +12dB gain, each also come with Mute, Solo and Off buttons, and there are level meters for both the input and output with peak hold readouts.

Shaping Up

With four stages to play with, Articulate offers a little more control over the attack and sustain of a sound than most transient shapers.

PAUL WHITE

N

ewfangled Audio’s Articulate, which is available through Eventide, is authorised using an iLok account, either to your computer or to a physical iLok key, as you prefer.

It supports the usual plug-in formats for Mac and Windows DAWs.

Overview Although Articulate can certainly be used to shape transients, it goes beyond the traditional transient modifier type of plug-in, and offers the user a control set that’s more akin to a synth’s envelope shaper. It works by splitting the audio info four components, which are designated Attack, Decay, Sustain and Release. The levels of each of these components can then be adjusted before they’re all recombined to create the output signal. Conceptually, then, it’s

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a simple enough idea, but as a technical achievement it’s far from trivial.

This level of control allows useful adjustments to be made to virtually any sound source that has a defined envelope of its own. For example, you can adjust the pick attack of a guitar or the thwack of a kick drum. The Release component also allows ambiences and reverb tails to be pulled up in level. Articulate can react to audio being fed to its input or to an external side-chain input, and the latter makes it possible to use Articulate to reduce mix congestion with, essentially, envelope-specific ducking.

The GUI, which is resizeable, can be set to one of three colour options and has a pleasingly simple layout, with vertical faders for Attack, Decay, Sustain and Release, plus a horizontal fader that adjusts the Separation mode, from Smooth to Focused. Smooth, as its name suggests, produces the smoothest-sounding results,

This plug-in provides a very straightforward to way to manipulate sounds that have a naturally fast attack. For example, the Attack control can be used to emphasise or pull back the impact of a kick drum or snare, while the Decay control adjusts the punchiness of the sound, and you have just that bit more control over the result than with a traditional ‘boost or cut’ transient shaper. Similarly, I found that Articulate made it easy to tame electric guitar sounds suffering from an over-strident pick attack. Pianos and other percussive synth sounds can also be treated. The Sustain and Release controls then allow the tail end of the sound to be adjusted in a similar fashion, to make it anything from dry to very airy.

There are no parameters to adjust that change the duration of the ADSR components, so you’re not able to create slow-attack sounds of the sort you can with, say, Eventide’s Physion 2, but on the whole Articulate is a very handy and easy-to-use tool for changing the character of instruments and drum loops.

So easy to use, in fact, that setting up your own treatments with just a handful of controls makes the 50 supplied presets largely unnecessary. Well worth installing the demo.

summary A transient‑shaping plug-in that provides you with more control than most over the attack and sustain portions of sounds.

£ $69.

W www.eventideaudio.com/

plug-ins-newfangled-audio

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ON TE ST

UJAM UFX Filter Filter Plug-in

modulation effects, amongst a few others.

Finisher provides an even wider choice, spanning reverbs, delays, glitch effects, distortion and various rhythmic effects. For both units, you simply pick one of these preset effects and then dial in the amount you want; it’s wonderfully simple but, again, also very effective. Finally, for those seeking sonic inspiration, UFX Filter provides randomisation options. Just hit the Surprise button and, depending on the Small/Big dial setting, the settings for your current patch will be subtly shifted or radically changed.

It’s also worth noting that all the UFX plug-in parameters play nicely with your DAW’s automation system, so you can add additional modulation via that route.

No Fuss Filter Fun

We check out the newest addition to UJAM’s UFX effects range.

JOHN WALDEN

U

JAM have a distinctive and consistent philosophy. Whether it’s their virtual instruments or virtual effects, the design intention is to combine top-notch sounds with user-friendly, intentionally streamlined control sets. This ethos applied to their first UFX series plug-in, UFX Reverb, which we reviewed in SOS December 2023, and they are now back with two further UFX titles: UFX Filter, which is the main subject of this review, and UFX Delay.

LFO waveforms but also more creative options such as stepped and rhythmic patterns. It’s a simple but surprisingly effective and flexible control set.

The right side of the GUI provides stripped-back versions of UJAM’s Shaper and Finisher effects, with the FX Input Mix knob controlling how much signal reaches these compact multi-effects options.

Shaper’s presets provide additional filter options, saturation, distortion, EQ, pan and

Familiar Format As with UFX Reverb, UFX Filter provides the user with a preset collection of processing modes. In UFX Reverb these consisted of different reverb algorithms, but this time they provide different filter combinations, including the usual low-pass, band-pass, high-pass and notch options, as well as more intriguing choices and combinations of two or three filter types. Once a preset filter mode is selected, the left/centre portion of the GUI lets you fine-tune the filter settings, adjust the depth of the input envelope modulation (allowing the filter to be driven by the amplitude of your input signal) and set the wet/dry balance using the large Mix control. In addition, the Modulation section lets you animate the filter using a range of preset modulation curves, including typical

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If you have doubts about the compact control set, UFX Filter’s ample preset selection does a very good job of demonstrating just how much filter fun is available, and you can try it all out courtesy of a 30-day free trial (for all their plug-ins). Indeed, it’s easy to find a great starting-point preset for a range of sound sources (drums, synth pads, guitars, piano or vocals, for example) and pretty much any occasion. From there, the compact control set provides you with plenty of scope to dial that effect in to suit your needs.

What this all means is that you can do some cool and creative sound design in double‑quick time, without any compromise in terms of the quality of the results. Pick a couple of elements in your project, and if you dial in just the right quantity of ear candy, the outcome can be transformative.

It’s impressive, creative, super-easy to use and a lot of fun.

The depth of control you like your creative effects plug-ins to offer is a deeply personal choice. But if the UJAM ‘keep it simple’ approach appeals to you, then great — because UFX Filter sounds great, and the incredibly modest price tag makes it a nailed-on bargain. Having also taken UFX Delay a spin (like UFX Reverb, it can do conventional effects, but holds lots of creative potential too), I have to say that UJAM’s UFX series is coming along rather nicely.

summary With new Filter and Delay options, UJAM’s UFX series is shaping up very nicely. Great fun and a bargain to boot.

The 20 different filter mode presets provide plenty of sonic options.

£ UFX Filter £26. Full UFX Bundle

(three plug-ins) £43. Prices include VAT.

W www.ujam.com

ON TE ST

Audix OMX Series Dynamic Microphones

This trio of stage mics might look the same, but they’re each suited to different roles.

PAUL WHITE

A

udix have a long-standing reputation for building good-sounding, reliable mics both for live sound and studio applications.

The new OMX-T, OMX-M and OMX-E models are dynamic microphones, each designed to meet specific use cases, differing in their frequency responses and the precise shape of their polar patterns.

The capsules employ Audix’s established VLM (Very Low Mass) diaphragm technology, which helps extend their high‑frequency reach.

Silver And Gold All three mics have a comfortable profile, measuring 19.76 x 19.76 x 7.06cm, with a zinc alloy metal body that flares towards the flat-topped basket, which can be unscrewed to access and clean the foam liner. As the physical outline of each mic is identical, a coloured identity ring is set into the body below the basket. The capsules are supported on fairly stiff, resilient mounts. Each mic comes with a soft storage pouch, a mic stand clip and a US‑UK thread adaptor.

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Identified by a silver ring, the OMX-T is a live-oriented microphone that is voiced to suit vocals, but can also accommodate loud sound sources such as electric guitars, bass amplifiers and drums.

Audix also suggest that it is appropriate for radio broadcasts. It has a tight supercardioid polar pattern designed to offer good resistance to feedback, while the basket design helps avoid problems when used by those vocalists who tend to cup the microphone. Sensitivity is quoted as 0.82mV RMS/Pa at 1kHz, with a frequency response of 52Hz to 17kHz (-6dB) measured at 0.5m. The response curve shows familiar vocal mic voicing characteristics, with some low-end roll-off below 200Hz to compensate for the proximity effect combined with a lift above 2kHz to add presence. All three OMX models boast maximum SPL handling in excess of 144dB. Billed as an all‑rounder mic, the OMX-M exhibits a slightly more neutral frequency response than the OMX-T, though it still incorporates a broad presence lift (this time centred at around 4kHz) combined with a slightly steeper LF roll-off below 200Hz, giving it

a somewhat leaner frequency response of 78Hz to 17kHz (-6dB). It has a cardioid rather than supercardioid pickup pattern, and is suited to both vocals and instruments as well as podcasting. It is identified by a gold ring. Sensitivity is quoted as 2mV RMS/Pa at 1kHz. The OMX-E, which has a green identity ring, has been voiced to add emphasis to the midrange and ‘presence’ region, and to reduce low‑end boominess. Its LF roll-off doesn’t start until around 100Hz, and the overall frequency response is quoted as 55Hz to 17kHz (-6dB). Audix’s suggested uses include vocals for small and medium PA systems, as well as acoustic and electric guitars, brass, woodwinds and drums. This time, the sensitivity is specified as 1.6mV RMS/Pa at 1kHz.

OMX... OMG? Starting with the OMX-T, I decided to make my first test a vocal recording, hand-holding the mic to check out its handling noise. Handling noise was negligible and the resistance to popping was also good, other than a couple of careless plosives very close to the mic,

and even those could be tamed with some low-cut EQ. The recorded sound was very natural, offering just enough presence boost to cut through without sounding edgy. For rock music, this mic could even be a viable alternative to a more typical capacitor model in the studio.

As a live mic, the polar pattern helps avoid spill and reduces the risk of feedback, as long as your monitors are positioned carefully. To test this, I tried recording a vocal while leaving the monitor speakers on rather than using headphones. Working facing the monitors, the amount of spill was low enough that the recordings were perfectly usable.

“The recorded sound was very natural, offering just enough presence boost to cut through without sounding edgy.”

As an instrument mic, the OMX-T takes guitar cabs in its stride and also acquitted itself well on percussion. While you can also make acceptable acoustic guitar recordings with the OMX-T, I’d still opt for a well-chosen capacitor mic in this role.

The OMX-M has a similar character to the OMX-T but with a standard cardioid polar pattern, a slightly more progressive low-end roll-off and a more subdued presence lift. It works well as a vocal mic for live use or for speech recording (for podcasts and audiobooks, for example), and it also makes a good general-purpose studio mic. Once again, the handling noise is low and the rejection of spill is just what you’d expect from a well‑controlled cardioid pattern, insomuch as the mic is still fairly sensitive at the sides but very insensitive at the rear.

That leaves the OMX-E, which again worked very well as a vocal mic, while its beefier low-end

The Audix OMX mics are outwardly identical, apart from a coloured identifying ring underneath the basket. The T, M and E models have a silver, gold and green band, respectively.

response captured a little more heft from instruments such as drums, percussion and guitar amps. It too has some built‑in presence lift, but not enough to detract from the natural sound of the voice or instrument. If using it for vocals in the studio I’d recommend a pop screen, as the extended

low end makes popping more likely on plosives, and for live use, a low‑cut filter on the desk might be needed to keep vocals sounding clean. However that extra low-end weight works in its favour when dealing with guitar amplifiers and also makes the OMX-E a highly capable snare and tom mic.

In summary, then, all three of these Audix dynamic mics can be used for vocal or instrument recording or live sound, but there are enough differences to give each model a specific focus. For live vocals I’d pick the OMX-T for its tight polar pattern and resistance to popping, whereas for guitar amps and drums the OMX-E is the more logical choice because of its more extended low end. The OMX-M is a good mic to have around when you need something that can take on most jobs.

In all, a useful and robust trio.

summary These are classy, robust mics that each offer a clear, natural sound but with enough differences to allow them to specialise in different roles.

Each OMX mic ships with a soft zip pouch, a mounting clip and a thread adaptor.

£ £135 each including VAT. T SCV +44 (0)330 122 2500 E sales@scvdistribution.co.uk W www.scvdistribution.co.uk W www.audixusa.com

www.soundonsound.com /July 2025

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and suitable articulation choices, Ancient Kingdom will — sonically, at least — put your audience right into that ancient world. I particularly enjoyed the low hulusi, double shepherd flute and various Armenian duduk options, but there are beautiful sounds to be had wherever you look.

Opus Options

EastWest Ancient Kingdom Sample Library EastWest explore the world of traditional wind instruments with characteristic style.

JOHN WALDEN

F

or film, TV or video game composers working in fantasy genres or world music styles, a virtual collection of traditional instrument sounds is an important part of their sonic toolkit. EastWest’s latest release — Ancient Kingdom — is aimed squarely at these kinds of use cases, providing a tour of some of the world’s most distinctive traditional wind instruments.

EastWest have collaborated with virtuoso wind player and composer Saulis Petreikis and the result is a deeply sampled virtual selection from Saulis’ own collection of instruments. The library uses EastWest’s excellent Opus front end and is available as

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either a perpetual licence, or as part of the ComposerCloud+ subscription.

Hold Your Breath From the Armenian duduk through to the Viking war horn, Ancient Kingdom’s instrument collection spans some wonderfully exotic choices. In each case, EastWest’s usual attention to sampling detail is fully present and correct.

This does, of course, mean plenty of expression through suitable velocity layers, but it also means Saulis’ playing abilities are represented via the extensive performance articulation options. Within each instrument group, therefore, a range of Opus presets are available focused on specific articulation choices as well as full keyswitch-based presets. As shown in the screenshot, these go well beyond sustained and staccato options, letting you create performances full of realistic detail.

The underlying sound of each instrument is excellent, capturing plenty of subtle details. With appropriate playing,

The Opus front end lets you make good use of the core sampling to further customise the sounds. This includes options for instrument doubling and control over the amplitude envelope, but the ambience options are particularly impressive. With four microphone positions (close, mid, main and surround) that can be blended, and Opus’ classy reverb (which features a huge selection of reverb types), you can place your sounds into whatever kind of virtual space you might need.

If you want to gently push the instruments away from their purely traditional sound, you also get four predefined effects signal chains that you can blend.

These offer various processing combinations, with the original signal chains including the use of guitar amps, Leslie-style speakers and megaphones, for example.

It’s all tastefully done and provides just enough to give the sounds a more modern edge if required. Opus also offers its usual Soft, Classic and Epic ‘Mood’ options.

These essentially provide preset configurations of all the sound‑shaping options to get you instantly into a suitable sonic ballpark.

World Of Sound Conceptually, Ancient Kingdom is fairly straightforward and, of course, there are other sample libraries that cater for similar types of world wind instruments. That said, the sounds here are undeniably impressive, and I could easily imagine these virtual instruments being used in any triple-A film, TV or video game context; the quality bar is very high.

I’ll leave you to your own debate about perpetual licences versus subscription services but, for busy working composers, Ancient Kingdom is just additional value added to EastWest’s already cost-effective ComposerCloud+. Indeed, it is an excellent complement to titles such as Fantasy Orchestra that are also included within that subscription and, for anyone working on world music projects, or composing for fantasy-based soundtracks, the Fantasy Orchestra/Ancient Kingdom combo would provide a versatile core sonic palette. Yes, traditional wind instruments are something of a niche musical tool, but Ancient Kingdom provides a world of beautiful breathy sound in a convenient and deeply sampled virtual format; it’s impressive stuff.

summary EastWest’s Ancient Kingdom provides a beautiful collection of traditional wind instruments from around the world.

£ $299 or via ComposerCloud+ subscriptions

($19.99/month or $149/year). Prices include VAT.

W www.soundsonline.com

ADI-2 Series

Ultra-Fidelity Reference Class Converters Experience the new benchmark for audiophile precision and professional sound quality. The combination of transparent, highly detailed audio recording and playback makes the ADI-2 series an excellent choice for studio reference playback, mastering and audiophile listening environments.

DPS-2 LNI-2 DC

ADI-2 Remote & DigiCheck NG

Low Noise Isolating DC Filter, DC Input & output

Linear Dual Power Supply

Worldwide Distribution: Audio AG, +49 8133 9181-0, www.audioag.com • UK Distribution: Synthax UK, +44 1727 821870, www.synthax.co.uk www.rme-audio.com

ON TE ST

SOUNDTOYS SPACEBLENDER

Reverb Plug-in

Soundtoys SpaceBlender

PAUL WHITE

B

oth the algorithmic and convolution approaches to artificial reverb are now capable of incredibly convincing emulations of real spaces. But ‘real’ isn’t always what we want, is it? In fact, there are often occasions when we prefer to ‘boldly go where no‑one has gone before’, and to get you there, you’ll want to reach for a reverb that strives to be a little different.

Popular examples include Zynaptiq’s Adaptiverb and Eventide’s Black Hole, for example, or perhaps you’d prefer a shimmer reverb or a Strymon Cloudburst? Well now there’s another option from Soundtoys, who’s excellent range of plug-ins has proved enduringly popular. They’ve added to their portfolio a new plug-in called SpaceBlender, which increases their repertoire of creative reverbs. Billed as an “imaginary room machine”, it supports all the usual plug-in formats and, like the company’s other plug-ins, is authorised using an iLok account.

Swarming Up While SpaceBlender is perfectly capable of delivering fairly conventional reverb

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New Soundtoys plug-ins don’t come around that often, but when they do, they’re usually well worth checking out. This one’s no exception... sounds, this isn’t its raison d’être — after all, Soundtoys already offer more suitable options when it comes to ‘normal’ reverb, in the form of their Little Plate and SuperPlate plug-ins. SpaceBlender goes far beyond that, in a way that should please many but absolutely delight sound designers and creators of ambient music, as it enables you to create intriguing imaginary spaces that can sound wonderful, but simply don’t exist in nature. Its clear and intuitive GUI can be resized in steps from 75 to 200 percent, and as is usual with Soundtoys plug-ins the control layout is so simple that there’s virtually no learning curve to negotiate.

Most algorithmic reverbs combine delay networks with multiple feedback paths involving all-pass filters and EQ to build a reverb tail that increases in density as it decays. But although SpaceBlender is indeed an algorithmic reverb, it is inspired by something known as swarm synthesis. This is a method of creating complex outcomes by combining multiple

simpler units that follow basic rules, and that can interact with each other, the overall structure emerging from the combination of these interactions. If that all sounds a bit too theoretical to take in, it might be helpful to know that some other applications of this technique are inspired by the flocking patterns of birds: each bird flies independently, but it also reacts to the behaviours of neighbouring birds. Apparently the original inspiration for SpaceBlender, though, came from Dewanatron’s Swarmatron, an interesting analogue synth we reviewed back in SOS June 2011 (www.soundonsound.com/ reviews/dewanatron-swarmatron).

Traditional algorithmic reverbs create a reverb tail that decays in a generally predictable manner, emulating a natural decay, whereas the shape of the reverb produced by SpaceBlender can be manipulated by the user to produce extraordinary results. There are no separately adjustable early reflections, as

you’d find in most algorithmic reverbs, but at minimum density settings, the output does comprise a series of discrete delays.

Examples include long reverbs with no significant decay, fairly abrupt reverb endings, very slow ‘blooming’ reverbs, reverse‑envelope reverbs, gated reverbs and so on. The maximum decay time can be adjusted from 100ms up to one minute and remains the same regardless of changes made to the reverb tail’s shape. As expected, there are several presets to get you started, and you can save your own.

Mission Control Key to the operation of SpaceBlender is the interactive Visualizer display. This shows the ‘shape’ of the reverb in blue, and also a ghost image of the sound that’s being processed (purple), moving from left to right. Controlling the reverb shape is very straightforward, being linked to the pulsating cursor spot in the Visualizer. On the vertical axis, this controls whether the reverb is generated at a constant level, after which it rapidly fades, or whether it builds up to a maximum before decaying. The horizontal position determines the point at which the level is maximum so, by manipulating the cursor, the decay shape can be set to have a slow or fast attack, and a slow or fast decay; the straightforward grab-and-drag approach makes it extremely intuitive.

The Time control adjusts the total length of the reverberant effect, either in milliseconds or, if Sync is engaged, beats (from one to 32). Some useful features that don’t clutter the GUI include right-clicking any knob to replace its name with a percentage value readout, and holding Shift while adjusting a knob to engage fine-tune mode. Also, Ctrl+Opt (on a Mac) or Ctrl+Alt (for Windows) locks a knob, so that its value stays fixed when you switch presets — not a unique feature, but a useful one that I’d like to see more widely adopted.

When Warp is active, you hear quite drastic tape-style pitch changes as the Time parameter is adjusted. When it’s switched off, the pitch remains unaffected and the level drops as Time is being adjusted.

Color essentially gives you control over the tonality of the reverb, with anticlockwise turns darkening the sound as it decays, and clockwise turns causing the sound to become brighter and thinner during the decay. Texture adjusts the reverb

density: when fully anticlockwise you see what’s essentially a multi-tap delay, albeit with a lot of irregularly spaced taps. Fully clockwise, it produces maximum density, and these changes can clearly be seen in the Visualizer.

I believe it was Lexicon who pioneered the use of pitch/time modulation within their reverb algorithms (Spin and Wander) to smooth out resonances, and this tactic is used here, with the Mod knob adding a chorus-like effect to the reverb tail. This can sound effective if used sparingly on vocals, but heavier applications can work well on other sources, including some guitar and synth sounds. Mix controls the wet/dry balance, just as you’d expect.

Soundtoys have also added a Freeze button that captures and sustains the reverb sound by looping the input to the reverb engine, rather than by the more common methodology of recirculating

Much of the sonic variety available from SpaceBlender comes about due to the ability to directly manipulate the shape of the reverb using the Visualizer cursor. By setting a very slow attack time, it is possible to create reverb effects that bloom and evolve or even sneak up on you several bars later in the song. Or you can push the cursor fully upwards, dial in a long reverb time and look forward to a solid wall of sound. For a plug-in with so few controls, there’s a lot to explore here, though if I put on my ‘Mr Picky’ hat, I’d have liked to see some shimmer type options built-in, to further extend the range of available effects. Having said that, you can always combine it with other plug-ins to expand its capabilities, pitch‑shifters and granular delays being particularly useful in that context.

As I’m writing this part of the review, I’m listening to a short piano phrase being processed by SpaceBlender in Freeze mode, and with the Mix 100 percent wet it sounds like a cross between a ghostly string section and distant choirs — I can’t hear any trace of a piano in there anywhere.

This is great: bounce that down, and I’ll have the start of my next ambient track!

SpaceBlender excels at ethereal reverb effects, so I suspect that sound designers, music for picture composers and those into ambient or chillout music will be the most appreciative of its abilities.

But there’s plenty here for others to investigate too. I love the immediacy of its controls, and in reality there’s almost no way to coax a bad sound out of it — SpaceBlender delivers instant gratification all the way, and can create effects that will take your compositions in new directions.

It will certainly make a very welcome addition to my reverb arsenal — with its imaginary spaces, it might well become my new best imaginary friend!

“I knew almost instantly that this reverb was something special, as I found myself playing through the default preset for several minutes.” the reverb tail. This allows frozen sounds to be manipulated by using the Visualizer cursor or other controls, which opens up some interesting possibilities for using DAW automation to create shifting beds of sound.

The sound being recirculated is also shown in the Visualizer display and the output when using long time settings is a good way to produce an evolving ambient wash or pad until Freeze mode is released.

Gold Blend?

When describing the actual sound of SpaceBlender, it’s difficult knowing exactly where to start! But I knew almost instantly that this reverb was something special, as I found myself playing through the default preset for several minutes before setting off to explore the other options.

Depending on the Texture setting, which is essentially a density control, the sound moves from a dense cluster of discrete delays to a very smooth and dense reverb with lots of useful options in between. Key to the sound of SpaceBlender is that the reverb tail doesn’t start to sound ‘ringy’ or metallic at long decay times. I tried to replicate its effects using Logic Pro’s ChromaVerb (which can generate very long reverbs and has a variable attack time) but it sounded very different and somewhat metallic by comparison.

summary An algorithmic reverb like no other, Soundtoys’ SpaceBlender can conjure up some truly enticing imaginary spaces, with none of the metallic artefacts you sometimes hear in reverbs, and it’s capable of transforming the incoming sound into something else entirely.

£ $99 E sales@soundtoys.com W www.soundtoys.com

www.soundonsound.com /July 2025

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DRAWMER MQ-1 & MQ-2

other level controls (gain only on the MQ-1, and gain and compression level on the MQ-2) have black inserts with white indicators. Black buttons with associated LED indicators control the hard bypass on both units and the signal path on the MQ‑2 (EQ pre/post the compressor).

Lastly, red buttons with LEDs switch in Drawmer’s unique Crush circuit, a harmonic saturation feature that’s been borrowed from their own 1971 four-band parametric EQ. It’s the same circuit on both modules, but the MQ-1 has an additional Harmonics LED that gets progressively brighter as the amount of saturation in the output signal increases. Other than that last LED, the MQ-1 offers no indication of the signal level, whereas the wider MQ-2 lacks the Harmonics LED but is equipped with three eight-LED bar meters. The first displays up to 20dB of gain reduction, whilst the other pair indicates the level of the stereo output’s left and right channels.

Like all Drawmer hardware, the all-analogue MQ-1 and MQ-2 are designed and hand-assembled in Yorkshire, in the UK. All of the components are full-size, through‑hole types, and the standard of workmanship is of the highest order. The ingenious mechanical design of both units creates a very stable structure, too.

The Equaliser

500-series EQ Modules

Drawmer MQ-1 & MQ-2

These modules pair a vintage-style EQ with a clever saturation feature — and the stereo MQ-2 also brings controllable compression to the table...

BOB THOMAS

F

orty-three years since the late Ivor Drawmer (1943-2023) founded his eponymous company and introduced the first frequency-conscious noise gate, the DS201, Drawmer have released what I believe to be the world’s first Motown-inspired 500-series analogue EQs. One is a single-slot, mono, seven-band EQ called the MQ-1, and its sibling, the MQ-2, is a larger double‑width stereo device that pairs the same EQ circuit with a compressor. The EQ sections are based on the

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frequency bands and bell-shaped EQ curves of a passive EQ that was developed

by Mike McLean of Motown’s Engineering Department in the 1960s: a proportional-Q, seven-band design. But neither module is a direct clone, and both have a neat trick up their sleeve that could well extend their usefulness today.

Design & Construction Largely, these modules adhere to Drawmer’s long-established black, yellow and white livery, although I noticed that the classic black knobs first seen on the DS201, which had yellow pointers, have been replaced with plain black knobs in yellow circles. To help you navigate the photos, both units’ seven cut/boost frequency knobs have light-grey inserts with black position indicators, while the

In both the MQ-1 and MQ-2, the seven bands’ centre frequencies are preset at 12.5, 5 and 2 kHz, and 500, 320, 132 and 50 Hz, just as in the original Motown equaliser. Although those centre frequencies are spread too far apart for detailed, precise tonal correction, their sizeable overlaps make them very effective for broader tone-shaping right across the audible spectrum. Each band offers ±10dB of adjustment, by the way, whereas in the Motown design it was ±8dB. To replicate the bell curves and centre frequencies of the inductor-based Motown EQ, Drawmer’s engineers have employed active gyrator-based circuits, to create virtual inductors. If you’d like to know more about the active gyrator approach, our Technical Editor Hugh Robjohns provided a good overview in this SOS Forum post: https://sosm.ag/gyrators-v-inductors.

In the Motown equaliser, the level loss caused by its passive equaliser circuitry was compensated for using a post-EQ gain stage. In the MQ-1, the gain stage sits pre-EQ, whilst in the MQ-2, it sits post the EQ/compressor Order switch. Although these gain stages act primarily on the incoming signal level (MQ-1) and the output signal level (MQ-2), each has a secondary role in controlling the level of the signal going into the Crush circuitry.

Lying at the heart of the Crush feature is a compressor with a fixed attack and automatic make‑up gain, and it’s been designed to introduce warmth and musically pleasing second-order harmonic saturation to the EQ’ed signal. This compression/ saturation is threshold‑based, of course, so applies only to the frequencies that are highest in level — in practice, that typically means you saturate the bands you’ve boosted, and not those that have been cut. The results obtained from this function, then, depend on the interaction between the frequency characteristics

of the instrument or voice that’s being EQ’ed, the frequencies being boosted and the overall signal level, including any contribution from the gain control.

As the EQ boost increases, the more apparent the Crush contribution becomes at the frequencies being boosted, and the brighter the MQ-1’s Harmonics LED illuminates. Note, though, that if you want to dial in a particularly high level of Crush by boosting a band or bands, you may actually find yourself having to reduce the signal level coming into your MQ-1, or that leaving your MQ-2, so as to avoid clipping the units’ output stages.

The Compressor In addition to its EQ stage, the MQ-2 features a one‑knob, programme-dependent stereo compressor that can deliver up to 20dB of compression.

The single knob controls the overall amount of compression, influencing both threshold and ratio in the process. Being programme-dependent, the compressor automatically adjusts its attack and release times in response to the amplitude and frequency content of the incoming signal. This automatic optimisation alters the compression envelope to suit the signal, and can help to deliver a more natural-sounding compression. The one-knob approach makes dialling in the desired amount of compression really intuitive. The post-compressor gain knob then allows you to make up for any loss in the signal level that results from the gain reduction.

As I alluded to above, the MQ-2’s compressor can be placed either pre or post EQ and this is done using the Order button. The position you choose will depend entirely on whether you want to,

for example, reduce a track’s low-end level before it gets to the compressor (which might prevent unwanted compression), or to adjust the EQ to mitigate any of the reduction in high-frequencies that can be a side-effect of compression. There’s no right or wrong, just what works and sounds best for the source you’re processing and the track that you’re working on.

Crush & Clarity To me, the defining quality of both the MQ-1 and MQ-2 is the clarity and detail of their audio performance, which is reflected in their 20Hz-20kHz (±0.5dB) frequency responses. Specification-wise, the MQ-2 has slightly higher noise levels, but only because its compressor is always in circuit — this didn’t create any issues for me.

A more consequential difference is that the MQ-1 can accept input levels of up to +21dBu, whereas the MQ-2 tops out at +16dBu, and that should be kept in mind when gain-staging signal chains.

Operating the MQ-series equaliser is intuitive, and it’s very effective too. Its wide bandwidth at lower boost/cut levels enables you to gently sculpt a source’s tonality.

Although this EQ is, to me, more paintbrush than scalpel, it can definitely reward a more aggressive approach when necessary because there’s sufficient focus available to help deal with an issue in one frequency band without creating insurmountable problems in neighbouring ones.

The EQ’s clarity means that it doesn’t change the sonic character of a source too much, of course — but that’s what Crush is for, and Crush is highly effective and pretty addictive. Given its interaction with gain and EQ boost, it rewards experimentation too, and when you become accustomed to the interaction, you’ll be able to find the ideal level of harmonic saturation for the task at hand quickly. For example, in the lower frequencies, Crush can be used to help thicken up drums and bass guitars. In the mids it can add presence and warmth to a mix or put an edge on vocals, The side panel of the MQ-1, with a graphic indicating how the seven EQ bands overlap and can thus interact.

piano or percussion. Up in the highs, you’ll find Crush useful for enhancing cymbals and string instruments. Should you want to explore the outer limits of its capabilities, the results can get quite crunchy, but avoid clipping the signal and you can certainly head on out there.

The MQ-2 boasts one of the most effective one-knob stereo compressors I’ve encountered to date. Perhaps the most effective: it just works. Turn the knob slightly and you get light compression, turn it further and there’s more; it’s that simple. At lower compression levels this compressor is perfectly suited to levelling vocals, strings, other acoustic instruments and percussion. With the compression wound up, the MQ-2 is a very effective stereo drum-bus crusher. To modify the frequencies that trigger compression, just position the EQ pre compressor and boost or cut as desired. Any drop in level due to heavy compression can be made up with the output gain control, and any loss in high-frequency content can be addressed easily if you make the EQ post compression.

Verdict Like so many Drawmer units, the MQ-1 and MQ-2 are excellent value for money.

Both offer an intuitive workflow and performance that are well above what you’d expect given their relatively modest price — but especially so the MQ-2. If you use a 500-series rack as a single or double-mono recording channel (or are considering doing so), the MQ-1 would be a great choice as your main or supplementary EQ. Its tonal flexibility and the Crush feature will give you a wider sonic palette to work with than a conventional parametric or semi-parametric equaliser.

Should you, like me, prefer a stereo setup to handle, for example, analogue synths, pedalboards, amp modellers, drum machines and stems, then the two-slot stereo MQ-2 should get your vote.

Because it could turn out to be the only EQ/ compressor you’ll need.

summary These great-sounding 500-series units deliver the essence of the famous Motown EQ, and add some very useful functionality.

£ MQ-1 £298.80, MQ-2 £454.80.

Prices include VAT.

T Drawmer Electronics +44 (0)1709 527574 E sales@drawmer.com W www.drawmer.com

www.soundonsound.com /July 2025

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Arturia MicroLab Mk3

Controller Keyboard If you’re looking for a controller keyboard that’s both compact and affordable, Arturia have got just the thing...

ROBIN VINCENT

T

he MicroLab Mk3 is quite possibly the least-featured MIDI controller I’ve ever set my fingers on. The previous version at least had a jaunty-coloured jacket and a natty recessed MIDI cable so that you would never be without it. With the Mk3, other than a dash of veneer on the ends, it is the plainest of the plain, no-frills, off-white or black, compact MIDI controller keyboard, and I’m pretty sure that’s the point.

Easy Does It To spend some moments deep diving into the facilities, you get two velocity-sensitive octaves, two touch

strips and four buttons.

It’s probably small enough to fit in your laptop bag without too much trouble, but it’s surprisingly thick. This lends it a certain amount of robustness and also makes room for the mini piano-style keys.

The keyboard is actually pretty great for its size. It has a decent feel and resistance to it, it’s relatively quiet and it plays much nicer than the Launchkey Mini I currently have on my desk.

The minimal theme is continued at the back of the unit where there’s just a quarter-inch sustain pedal input, a Kensington security slot and a USB-C port for power and connection to a computer.

The touch strips to the side of the keyboard represent the usual pitch and modulation wheels, but they also have a secondary function: if you load up the included Analog Lab Intro plug-in and hold the Shift button, you can use the mod strip to scroll through the presets and the pitch strip to change categories. Tap on the strip to load the selected preset. The octave buttons can also get involved by stepping through the presets as you hold

Shift. It’s a little hamfisted but is helpful enough to stop you from reaching for your mouse. That’s not it for the touch strips, though. Using the Arturia MIDI Control Centre, you can further repurpose them to control other things if you wish. You can also adjust the velocity curve of the keyboard and decide if you want the Hold button to toggle or gate.

One last touch is the Chord mode, where if you press Shift and the Hold button and play a chord, the MicroLab will remember it and play the same chord on every key.

The Analog Lab Intro software comes with 500 presets from Arturia’s V Collection, which has a decent selection of usable sounds, and the obligatory Ableton Live Lite is also included, so you have everything you need to start making music.

“The MicroLab Mk3 is quite possibly the least-featured MIDI controller I’ve ever set my fingers on.”

Conclusion There’s really not that much to say about the MicroLab Mk3. It’s not an exciting keyboard, but it has a good feel, takes up very little space and lets you surf presets in Arturia synths without taking your hands off the keyboard.

And at £49 it’s undeniably affordable.

It would have been great if it had a MIDI out, but otherwise, it’ll do nicely.

summary Simple, affordable MIDI controller for people who want some compact keys that don’t feel terrible.

There’s not much to see round the back, just a sustain pedal input and a USB‑C port.

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£ £49 including VAT. W www.sourcedistribution.co.uk W www.arturia.com

Discover SOS For Artists : A Complete Platform for Modern Music Creators

Your end-to-end music production suite

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Promotion tools include Linktree-style landing pages and content creation tips to help boost visibility. Detailed royalty reporting shows trends, top countries, and top-performing platforms.

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*Prices exclude VAT. Taxes will be applied at the UK / EU Country rate at checkout. ** Also available in most countries worldwide in your local currency.

www.sosforartists.com

ON TE ST

Baby Audio Smooth Operator Pro

Resonance Suppression Plug-in

Baby Audio escalate their war on bad-sounding audio with a vastly more powerful toolset for tackling problem resonances.

SAM INGLIS

B

aby Audio are the very model of a modern plug-in development company, with a readily identifiable visual aesthetic, a wealth of novel ideas and a laser focus on ergonomics. Their plug-ins usually manage to offer a usefully broad range of control without over-burdening the user with parameters, and so it was with their take on the concept of resonance suppression or ‘de-harshing’. The flip side was that Smooth Operator didn’t offer quite as much to the power user as rival plug-ins such as Oeksound’s Soothe 2.

Four years after Smooth Operator appeared, Baby Audio have returned to the idea, but rather than create a version 2, what they’ve done is more

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July 2025 / www.soundonsound.com

akin to a ground-up reinvention. And, as the addition of the term Pro to the product name would suggest, it gives the power user everything they could want and more.

Greater Operator In the original Smooth Operator, nearly every control appeared only as a graphical node on an EQ-type curve.

This was elegant on paper, but in practice was not quite as intuitive as it might have been, especially with regard to the interaction between different controls.

Baby Audio have retained the EQ-like interface in the Pro version, but they’ve made things a lot more editable, and have given up on the idea that the curve alone can accommodate all the necessary settings. Instead, an inspector-type panel

to the left of the window allows you to adjust global controls, whilst a parameter list along the bottom displays settings for the selected node.

Each node has its own set of EQ-type parameters: frequency, bandwidth and offset from the global threshold, which is represented in the same way as gain in an EQ curve. Offsetting a node downwards with respect to the global threshold makes it more likely that Smooth Operator Pro will apply processing in that band, whilst lifting it above the global threshold tends to exempt a band from processing.

The nodes also inherit numerous dynamic and spatial parameters from the global settings, which are grouped under the labels Focus, Comp(ression) and Imaging. By default, these labels

Take it anywhere. Create whenever. Trust your instincts.

ableton.com/move

ON TE ST

BA BY AU DIO S MO OT H OPE R ATOR PRO

are dimmed, meaning that the nodes follow the global settings, but parameters in these three groups can also be set independently for each node by enabling the Override Global option.

Focus Groups Two new and very welcome global controls are the Lo and Hi Preserve dials, which respectively set low‑ and high-frequency limits on Smooth Operator’s domain. As with any dynamic process, there are times when it can be a struggle to persuade a resonance suppressor to focus on the upper midrange and leave the bass region alone, since there’s usually so much more energy in the low frequencies, so the Lo Preserve option is particularly valuable.

Another interesting global control is Correction, which has just two options labelled Even and Skew. The latter is said to apply perceptual weighting to the correction curve; details are scant, but in practice I usually preferred this option.

The original Smooth Operator provided a single global slider labelled Focus, which broadly governed the granularity of the processing. Smooth Operator Pro replaces this with a panel of three parameters. The Peak/RMS switch for the detector is a reasonably common feature in a conventional compressor but not one I’ve seen in this type of processor before. Detail controls the bandwidth and number of the individual resonances that will be detected: at high Detail settings you’ll see tens of needle-like corrections being made in each band, whilst the results of using a lower setting look and sound more like conventional dynamic EQ. Isolation is a slightly mysterious control which, if I understand the manual correctly, sets the extent to which ‘out of band’ frequencies are factored into each band’s detector algorithm. Whereas the consequences of adjusting the Detail control are readily apparent both in the sound and in the visualisation, the Isolation control can be very subtle.

The Comp group of settings will be refreshingly familiar to anyone who’s used a compressor, with a hard/soft Knee switch and fully variable attack, release and ratio controls. The addition of adjustable time constants is particularly welcome, as the fixed attack and release

times in the original Smooth Operator weren’t always fast enough.

The controls in the Imaging panel relate to channel linking. Smooth Operator Pro can smoothly operate in either left/right or Mid‑Sides mode, and two circular controls allow you to offset the balance so that more correction is applied Many Smooth Operator parameters can be set on a per-band basis, in one channel overriding the global settings. than the other — which is probably more useful in at the expense of versatility, you could Mid‑Sides mode — and to adjust the perhaps accuse Baby Audio of going degree to which the detector signal is the other way with the new version. In common to both channels. Reducing this practice, I didn’t often find it necessary Stereo Link setting to zero effectively to set parameters on a per-band level, or makes Smooth Operator Pro act like dive into the Imaging settings. Thankfully, a dual-mono processor. though, you can ignore these features if you don’t need them. The PDF manual is rather terse, and doesn’t explain everything as fully as I’d like; As I’ve already mentioned, all of the in particular, the operation of the Isolation Comp and Imaging controls can be and Correction controls remains rather applied on a per-band as well as a global enigmatic. As a result, Smooth Operator basis, as can the Isolation and Peak/RMS Pro is perhaps less easy to master than options. Detail, however, remains global.

Oeksound’s Soothe 2, but unlike the original, it competes head-on with the Black Ops OG in this field, and incorporates some features that aren’t available there.

One of the notable features of the It also boasts an excellent selection original Smooth Operator was that it of genuinely useful presets. seemed to be designed for creative as As time goes by, it’s increasingly rare well as corrective applications. You could that I do a mix without incorporating achieve some very interesting results by resonance suppression on something, deliberately pushing it too hard, and that whether it be a gritty drum overhead, remains the case with Smooth Operator a harsh violin or a badly recorded vocal.

Pro. If you don’t create any nodes on the Smooth Operator Pro provides all the graph but simply pull down the global tools that are needed for this, and plenty threshold control, you’ll hear your audio more besides. The full version represents being radically rebalanced in ways excellent value, and the upgrade from that are often quite ear-catching, but version 1 is a total no-brainer. surprisingly artefact-free. You probably wouldn’t want to use it across an entire summary mix in this fashion, but hearing what it does can be a useful step in diagnosing Smooth Operator Pro lives up to its ‘pro’ the faults in those tricky mixes that sound billing, providing a massively expanded somehow ‘wrong’. set of tools for tackling harshness and unwanted resonances.

But it’s in the corrective field that the improvements in Smooth Operator Pro really make their presence felt. Indeed, if £ $129; upgrade from Smooth Operator $29 the original erred on the side of simplicity W www.babyaud.io

“It’s in the corrective field that the improvements in Smooth Operator Pro really make their presence felt.”

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July 2025 / www.soundonsound.com

PODCASTS PEOPLE &

RECORDING

& MIXING

MUSIC INDUSTRY

OLLO Audio Headphones

Creating A Filter Pan Effect

Rok Gulič, founder and CEO of OLLO Audio talks to Sam Inglis about the origins of the company, the launch of their brand new X1 headphones and working with dSONIQ to offer calibration via their Realphones software.

David Mellor discusses how to create custom effects in a DAW. Here he demonstrates his approach to creating a filter pan effect using an EQ with high-pass and low-pass filters, highlighting how custom effects can help give your music a unique edge.

Behind The Brand

Practical Tips For DAW Users

ELECTRONIC

PEOPLE &

MUSIC

MUSIC INDUSTRY

Martin Kohlstedt

Steve Dub - Producer And Mixer

Composer, pianist and producer Martin Kohlstedt talks to Caro C about his approach to combining piano and electronic instruments, improvisation in the studio and on stage, plus the work he is doing to reduce the carbon footprint of touring.

Steve Dub talks to Kevin Paul and shares insights into his workflow as a producer, discussing his ongoing work with The Chemical Brothers, his love of experimentation and how he adapts his methods to mixing in the box.

Piano & Electronica

The MixBus Interview

Follow our channels by subscribing to the shows on Apple Podcasts, Spotify, Amazon Music or wherever you get your podcasts.

All shows are mastered to the highest quality the podcast channel will support and are in stereo.

Check out our website page for further details

www.soundonsound.com/podcasts

ON TE ST

Complete Guitar Wiring

By Gerry Hayes Want to dabble in DIY? Then you’ll need a good guide!

JOHN WALDEN

L

ots of the world’s guitar playing population like to tinker with their instruments, and whether it’s electric guitars and basses or an acoustic guitar with a pickup system, you don’t have to go far down this rabbit hole before you start looking at the instrument’s electronics. You might just wish to troubleshoot a problem, maybe you want to swap out the pickups for something you hope sounds better, or perhaps you’re keen to try a completely new wiring setup. Whatever the reason, if you never took Electronics 101 and don’t know one end of a soldering iron from another (trial and error is not such

a good approach!), your projects can involve a lot of head-scratching, swearing and time-wasting (been there, done that!).

Recently, I came across a rather good solution to that possible skills gap: Gerry Hayes’ book Complete Guitar Wiring (ISBN-10: 1919649425, ISBN-13: 978-1919649429). Gerry is an experienced luthier, and runs Haze Guitars from a workshop just outside Dublin. As well as doing guitar building and repair work, he writes an excellent blog and has authored a number of books and guides on guitars and guitar setup. The latest addition to his writing portfolio, Complete Guitar Wiring, is available in printed and electronic forms.

I purchased the latter for my own use but, having also seen the hardback version ‘in the flesh’, I can say with confidence that its production quality easily qualifies it for coffee-table status.

By The Book Spanning over 400 pages, this book is comprehensive. Its contents are split into seven main chapters, plus an appendix that’s stuffed with detailed specifications of common guitar parts and components.

So Complete Guitar Wiring can serve as

From pots to pickups, the photos and illustrations are absolutely top-notch.

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a reference for those who are experienced in guitar electronics — but a beginner could pick it up without fear of being intimidated.

Gerry’s writing style is a big part of that, but it’s also because the first chapter, which explores all the key components that you might find in an electric guitar, acoustic guitar or bass guitar circuit, provides a brilliantly accessible introduction to the basic theory of electronics. That’s a topic that can easily get heavy very quickly, but here it doesn’t. With a clear and pragmatic writing style, Gerry deals with these introductory technical concepts in a very accessible and engaging fashion, exploring the concepts in a very focused way and always in a guitar-related context. So if you’re starting from a point where you’re unsure of the difference between a resistor and a capacitor, or how a volume potentiometer functions, or the basic construction of common pickup types, or what’s required to build a complete electrical circuit, fear not: by the time you have read this first chapter, you will be considerably better informed.

Chapters 2 and 3 then build on that excellent platform, to help you explore the wiring and installation of electric guitar/bass and acoustic guitar pickups. There’s a lot of interesting background material here about pickup construction, spanning all sorts of pickup types from conventional single-coils and humbuckers to various piezo designs, including the specifics of systems found on selected leading acoustic guitar brands including Taylor and Takamine. Topics such as pickup setup, polarity, phase and grounding are all explained in clear and straightforward fashion. I didn’t expect to find so much useful information in the section on mounting/installing pickups but, electronics aside, I found myself learning a lot about the use of a basic router (to expand

a pickup cavity, for example), as well as the practical problems (and some great solutions) often encountered when installing pickups and wiring on a semi-hollowbody guitar (darn f-holes!).

Chapter 4 takes the practical advice on the wiring process much further. Framed around a number of typical wiring projects, it takes the reader through the circuit designs and, by building on the concepts introduced in chapter 1, considers how those designs result in the appropriate signal flow through the circuit. However, it starts simply with wiring a guitar cable. Then it’s an output jack, before moving on to a single volume or tone pot, and eventually graduating via pickup replacement to complete rewiring tasks.

Again, the technical elements are explained very clearly, but just as impressive is the ‘introduction to soldering’ content. If you’ve ever struggled to solder on the back of a pot, the advice here will undoubtedly get you on the right track. Other tools of the trade are also discussed, from wire types to wire strippers. It’s a well thought out and very practical

discussion. As elsewhere, the text is supported by high-quality photos and illustrations.

Chapter 5 focuses on typical troubleshooting cases and, even if you have no desire to build a new circuit from the ground up, there is plenty to learn here. This includes the valuable advice of a basic troubleshooting workflow and how to distinguish between things that might be perceived as problems but actually aren’t, and genuine problems. It then walks you through the processes for testing every major element of your guitar’s wiring circuit to find whatever fault(s) might need attention. For the inexperienced, the bonus is that you will also learn the basics of using a multimeter and, again, it’s all very clearly explained.

“While I realise it might be hard for you to imagine a technical book about guitar wiring being ‘an entertaining read’ that’s exactly what this is!”

These five chapters make up the first 200 pages or so of the book. Chapter 6, which occupies the next 100, contains a huge selection of standard wiring diagrams. These span the more common Les Paul, Strat and Tele circuits (including several common variants of each), but also lots of others from manufacturers such as Ibanez, PRS, Chapman, Gretsch, Suhr, Charvel, G&L and Rickenbacker.

The majority of the diagrams are guitar based, but a decent selection of basses is represented, including a number of common Fender options. And, because the book started with such a clear explanation of how such circuits are constructed — and, importantly, how they’re represented in diagrammatic form — even a less experienced guitar tinkerer can quickly start to get their eye in.

The final chapter, which also runs to nearly 100 pages, progresses nicely into the world of mods, and is a treasure trove of ideas that should appeal to newbies and experienced DIY guitar technicians alike. Whether it’s a simple change to A huge selection of full circuit diagrams is included within the book for a range of guitar and bass instruments.

how your Strat’s tone controls operate, exploring coil splitting, or configuring series/parallel pickup options, this section will get you thinking of the possibilities that your new-found guitar electronics skills might open up. There’s also a useful reference section of the typical wiring colour schemes used by most of the mainstream pickup manufacturers.

Buy The Book?

The book’s comprehensive coverage of its subject is an important part of its appeal, and Gerry’s experience as a luthier of 20-plus years has clearly informed his decision making in terms of both the content and the way it is organised. This book is a triumph in both regards. Just as important as the knowledge and organisation of all this technical stuff, though, is Gerry’s enviable ability to explain it all in a clear, logical and entertaining fashion. Not every luthier can do that, yet throughout this book Gerry comes across as a humble and self-effacing human being. So I’ll happily say it for him: this book is beautifully written and while I realise it might be hard for you to imagine a technical book about guitar wiring being ‘an entertaining read’ that’s exactly what this is! Gerry’s sense of humour, pragmatic approach and desire to share his obviously considerable knowledge come across on every page. Oh, and the many illustrations are every bit as good as the text, not only serving to support the concepts covered in the book very well, but also making it look fabulous.

If Gerry’s guitars are as good as this book, I sincerely hope to have the chance to play one at some point.

Complete Guitar Wiring may be a niche book about a niche subject but, by a considerable distance, it is the best book on the topic that I’ve ever encountered. So if you’re lost down the rabbit hole of guitar electronics, I strongly advise you to buy it — and let it shine its considerable light on your journey.

summary An excellent resource and an engaging read, for beginners and seasoned tinkerers alike.

£ PDF version $40.

Paperback £55 (from Amazon).

T Haze Guitars +353 (0)87 9966 439 E gerry@hazeguitars.com W https://hazeguitars.com

www.soundonsound.com / July 2025

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Ferrofish Pulse 8 AE

Eight-channel A-D/D-A Converter & ADAT Router This compact converter offers an easy way to expand your audio interface.

M AT T H O U G H TO N

F

errofish now have quite a range of A-D and D-A converters, and cater for a variety of digital connection protocols including MADI, ADAT and Dante. For review here is one of their smallest models, a half-rack‑width device called the Pulse 8 AE, and it’s intended to act as an I/O expander for audio interfaces with ADAT ports.

Features & Operation We’ll start on the rear panel, since the connectivity tells you most, if not quite all, of the story. Power comes from an included inline switch‑mode supply, which attaches to the device through a 2.54mm barrel jack. Thoughtfully, this has a screw‑on connector to keep the power cord firmly attached. Once hooked up to mains power, the Pulse 8 AE boots up in… well, let’s just say instantly, because while technically there may be a short delay, it doesn’t feel like there is.

The balanced analogue line-level inputs and outputs (eight of each) are presented on quarter-inch TRS jacks, with adequate space between them for the vast majority of jack connectors to fit alongside each other. There are also two pairs of ADAT ports for the digital I/O. Technically, you need only one pair for all eight channels, of course, but the ADAT protocol limits such setups to a maximum sample rate of 48kHz. The dual ports allow the device to use the S/MUX

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protocol to convert all channels at up to 96kHz (higher sample rates up to 192kHz are supported, but that halves the channel count). More unusual, compared with other converters, is that the Pulse 8 AE can act as a signal router that can address all the analogue I/O and the maximum of 16 ADAT I/O channels individually.

The Pulse 8 AE can sync with the attached device over the ADAT Lightpipe connection, of course, but there’s also the option of word clock, for which there are both in and out BNC connectors. This allows the Pulse 8 to act as the clock master or slave to another device, and in the latter scenario it can pass word clock along to a third device. Other rear-panel I/O (whose functionality I’ll return to below) include MIDI in and out on mini‑jack connectors (fine by me; there’s no way DIN connectors would fit on this panel), a USB-C port, and a larger slot reserved for future DSP expansion.

There are not yet any current or promised expansions, but Ferrofish see this as a means of future-proofing the unit.

On the front, there’s a headphone jack on the left, and the amp seemed to have plenty of juice in reserve when feeding my various headphones (I tried it with Audeze LCD-Xs, Sennheiser HD650s and Audio‑Technica ATH-W3000ANVs). It was good to see in the on-screen menus that there’s the ability to limit the headphone output by up to 18dB below the maximum — basically this means you can set it up to ensure sensible levels with your

regular headphones when the level is set to maximum. On the far right is a backlit power on/standby button (press for on, press and hold a few seconds for off).

In between these is a very crisp and clear colour TFT display, along with two more buttons and a rotary encoder for navigating the on-screen menus and making selections. The upper button, marked with three backlit horizontal lines, brings up the menu, and you then use the encoder to navigate and the same button to make selections. The other button (a backlit ‘X’ symbol) exits the current menu layer. These controls are context sensitive, so help you do what you want that bit more easily than might otherwise have been the case. For example, when in the home screen (showing the meters when sync is established, or a No Lock warning if not), turning the encoder sets the headphone level. Do that, then hit the upper button, and you can choose the headphone source, which can be any digital or analogue input, whether single channels for mono or adjacent-channel stereo pairs. If, when in the default screen, you instead hit the three-lines button first, you’ll be presented with a radial menu, whose icons you select to access various sub‑menus.

These sub‑menus allow you to change the clock settings, set the headphone routing setup, save and load presets, access system setup options, control the channel input gains and output levels, and configure the routing. You can also

RemoteFish Sofware Accompanying the Pulse 8 AE is a version of Ferrofish’s RemoteFish software, which may be familiar if you’ve used previous Ferrofish converters but which was updated during the review period to work with the new Pulse 8 AE. RemoteFish communicates with the Pulse 8 AE using MIDI, either via the Pulse 8’s two TRS mini‑jack connectors and a MIDI interface, or over a direct USB connection with your computer.

RemoteFish is a simple and unfussy affair: it doesn’t offer any more functionality than you can access on the hardware (in fact, it offers slightly less), and it might not win awards for graphic design; but ‘focused and unfussy’ is arguably not a bad thing, and it can certainly make managing some functions more convenient. For example,

if you want to experiment with different routing options and save them as presets, then it’s a lot easier to click on a grid in the software than scroll and select using the single hardware knob. There’s one window with five tabs, which not only allow remote control of the hardware, but also update to reflect any changes you make using the hardware’s controls.

On the left, the connected unit is identified. Tab 1 is the Main view, with input and output metering for the analogue channels (graphically similar to the hardware), a level control, a mute switch and source selection for the headphone out.

Tab 2 lets you set the analogue input gain and output level for each channel. Tab 3 provides a simple channel routing matrix, with blue outlines making it easy

set any MIDI channel to respond to CC controller data separately for the mini‑jack and USB connections, and you can specify whether or not the word clock input is terminated... and much more besides. You can even lock the unit so that only the headphone level can be tweaked from the front panel (there’s a code to lock/unlock it on a sticker on the base, but I might be tempted to engrave it on the base because once locked only this code will unlock it).

The sub‑menu style will feel familiar to anyone who has used Ferrofish converters before, but there are plenty of interesting options not available on, for example, my Ferrofish A32, to allow you to configure the unit to meet your own needs. For instance, the A-D and D-A converter options include the ability to set the filter types: as Minimum Phase, Linear Phase Apodizing, Linear Phase

to spot the currently selected cell. At the time of writing, tab 4 appears a little buggy: it’s intended to save/recall presets, but while recall of presets saved on the hardware works fine, the software save facility didn’t — Ferrofish are aware, and as I write a fix is on the way. Finally, we have a Settings tab, which provides firmware details and allows you to switch word clock BNC termination on/ off. There are a few hardware functions that aren’t accessible via the software (eg. changing the converter filter type) but nothing I’d miss in normal use.

All in all, it works well, but some common OS-style touches, such as a simpler horizontal/vertical drag option for the knobs and modifier-key multi-fader selection could be useful future embellishments.

Fast Roll-Off and various others. There are eight options for each converter, and they each strike a different balance between things like phase response, impulse response and HF roll-off; they’re all made possible by the ES9017 and ES9840Q chips used. Honestly, though, these converters sounded good to me on the default Minimum Phase setting, and I don’t recommend changing it unless you know what you’re looking for here.

The gain/level of the analogue line inputs and outputs can also be changed using the menu, and you can do this for individual channels, stereo pairs, with four channels ganged, or for all eight channels at once. Again, it’s all very easy, and the feedback given on screen, with virtual faders and a dBu value for each channel, is really clear. The analogue outputs have a maximum output level of +20dBu, but the virtual faders can back

this off by up to 28dB (to give -8dBu) in 1dB steps, while the inputs can accept a maximum of +20dBu and have up to 28dB of gain available, again in 1dB steps. The connections are balanced, but there are instructions in the manual for connecting unbalanced equipment.

While we’re talking numbers, the quoted SNR and THD+N figures might not be world-beating — the chips used here are capable of better — but they’re respectable, and slightly better than my older A32, with which I remain very happy.

Verdict The bottom line is that this is a good-sounding and remarkably compact A-D/D-A converter with very little direct competition: there are now few ADAT converters out there that give you dedicated line I/O without the mic preamps and so on, and when it comes to eight-channel ones there are even fewer. The SSL Alpha 8 (reviewed in SOS May 2025: https://sosm.ag/ssl-alpha-8) is the most obvious alternative but while that boasts better technical specs and some features that the Pulse 8 AE lacks, the Ferrofish device gives it a good run for its money and offers some different features of its own. It’s also more keenly priced and half the size. Subjectively it sounds great, and one thing that Ferrofish have long got spot on and continue to do so is the great on-screen feedback and easy-access menus. All food for thought when weighing up your options...

summary A compact and decent‑sounding line-level ADAT converter, with some useful features, for a competitive price.

£ £519 including VAT. T Synthax UK +44 (0)1727 821 870 E info@synthax.co.uk W www.synthax.co.uk W www.ferrofish.com

Alongside the analogue and digital I/O are MIDI TRS sockets and a USB connector, to allow for connection to the RemoteFish control software and firmware updates.

www.soundonsound.com /July 2025

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ON TE ST

Part fuzz, part guitar synth, this quirky pedal was conveived with horror soundtracks in mind. But it’s not a one-trick pony...

TED MORCALDI

T

his pedal is nasty: good-nasty!

Half monophonic synth, half drive/ fuzz pedal, the new Knife Drop is a collaboration between Eventide and Third Man, and despite the small footprint it can deliver massive walls of sound. The pedal is themed around Italian giallo films of the 1960s and ’70s, with a seemingly fictional backstory about working closely with an Italian film director, Giovanni Bianco (consider who that might translate to in English...), to recreate the sound of his horror film soundtrack. There’s a lot of creativity going into the aesthetic and presentation of the pedal, and it complements well the sounds that it’s capable of producing.

Knife Edge The Knife Drop stands out amongst a family of mono guitar synth pedals that now dates back decades, a category full of unique oddball pedals such as the Line 6 Filter Modeler, Digitech’s Meatbox sub synth, the Alesis Deep Impact, Way Huge’s Atreides and, the grandfather of the bunch, the Electro-Harmonix Micro Synth, to name just a few. These each aim to make a guitar sound like a monosynth, but without the need for any special pickup, as is required by the ‘guitar synths’ made famous by Roland and Boss. This pedal isn’t trying to recreate the sound of any classic, unobtainable analogue synth or any of the pedals I just mentioned, and I appreciate that in a time when so many pedals are harking back to retro classics instead of creating an identity of their own. Sure, the combination of octave up, sub‑octave and synth-type square-wave distortion to create a monophonic synth/distortion pedal has shown up in different pedals over the years, but the Knife Drop’s user interface is not only very intuitive, it’s also unique.

The primary user controls are Synth/ Guitar Mix, Drive and Level, and there are secondary functions for each knob — Attack, Resonance and Cutoff, respectively — to control the filter

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Eventide & Third Man Records Knife Drop Fuzz Synth Pedal

envelope settings, with the last one setting the starting frequency of the filter sweep. There’s an octave switch too, which gives the option of adding an upper octave only or both an upper and a sub‑octave. With the Octaves switch in red mode you get three notes: sub octave, original note and upper octave.

Green indicates that you’ll hear an upper octave plus the original note.

The back panel of the pedal hosts a mono quarter-inch jack input, stereo jack output, an expression pedal socket, a TRS auxiliary footswitch or TRS MIDI socket, a USB port (for MIDI over USB and software updates), and a switch to set the pedal to instrument or line-level operation. It’s also capable of connecting to Eventide’s H90 Control for remote desktop operation of routing, preset management and more.

I found that what sets this pedal apart from others in this category is the switch that allows the filter to be pre‑ or post‑drive. This single switch morphs the pedal from being a more classic-sounding analogue synth, to a gnarly, edgy synthy dirt pedal that, when the filter is placed

pre-drive, is on the brink of destruction.

The filter envelope becomes much more extreme too, especially with the resonance cranked, in which case it delivers some wonderfully exaggerated resonant filter sweeps. An expression pedal then gives you another layer of control, opening up the sonic possibilities further. The expression input can be configured so that the pedal controls parameters individually or so that it morphs between two different sounds entirely — that’s a very effective way to create more malleable sounds with this pedal.

Presets & Pedals There are five presets for this pedal, but one drawback that I noticed when putting this pedal through its paces, and which some might view as significant, is that they are only accessible when using an external footswitch — only preset 1 can be recalled without one. Looking into this further, a reply by Eventide to a YouTube comment revealed that this was a deliberate decision: Jack White (or should that be Giovanni Bianco?) wanted

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ON TE ST

EVENTIDE KNIFE DROP

an easy way to switch back to preset 1, which is his favourite. This is achieved by holding the Octaves footswitch down until the LED for the preset at the top of the list blinks.

While I do understand that this is what the artist wanted, I still think that there should really be an option for onboard preset selection; having to buy a separate footswitch to access the five presets isn’t ideal, especially given that the front panel makes it look like they’re directly accessible, and that the manual doesn’t state clearly that they aren’t.

I thought for a while that I was doing something wrong! By the way, if you have this pedal and want to make your own footswitch, it’s an easy build and a nice first step into the DIY-world: TRS jack, small enclosure, two momentary switches (DL4 replacement style), connect one lug of each switch to sleeve of jack, the other lug of each switch connects to tip or ring.

Sounds & Surprises For those using the pedal without an external footswitch, there are five preset card overlays included in the packaging, and these outline the knob settings for each preset. I found this very helpful, not only in accessing the preset sounds, but also in learning how to create usable patches on the pedal more generally. Multiple combinations of the alternate functions with the Octaves switch or filter pre/post switch are possible within each preset, drastically changing their character.

I was pleasantly surprised to discovered two unexpected sounds when exploring this pedal’s range. The first one was when using the Knife Drop with the Synth all the way

Like most Eventide pedals, the Knife Drop can operate at instrument or line level.

down and Drive all the way up, which created a gated, Velcro-y fuzz pedal.

Bringing the Octaves switch into play turns it into a huge three-octave-range fuzz or octave-up fuzz, and the latter is more reminiscent of a Whammy or Eventide’s Harmonizer algorithms than an Octavia‑style octave-up.

With the Synth level all the way down, the pedal is polyphonic, even with the Octaves switched on, which leads me

synths benefited from the absence of analogue‑to‑digital conversion, which obviously adds some latency. Happily, the Knife Drop is fast, with very little latency, but as always with such devices, this pedal’s tracking demands clean playing technique if you’re to avoid glitches.

It’s not the easiest pedal to play though fast, then, and it feels to me as though it’s intended primarily for mid-tempo riffs, especially if the filter attack is set to have a slow rise.

So what of the nature of those glitches? When playing more than one note with the synth engaged they sometimes result in a random single note that’s unrelated to the two being played, or if playing a fifth, a lower octave note comes out. Sometimes happy ‘freak outs’ occur when playing multiple notes or allowing an open string to ring while playing another, so there’s some creative potential. But while some monosynth pedals try (and fail!) to play back multiple notes, sometimes resulting in ‘musical glitches’, that’s not possible with this pedal — it’s not capable of playing back more than one synth note at a time. It was very noticeable when I didn’t play cleanly enough for this pedal, resulting in some imperfect tracking, but I’d say that this is part of its ugly beauty!

“With the Knife Drop pedal being marketed as a horror film soundtrack in a box, I really hadn’t expected to be able to get such a ‘pretty’ sound out of it.” to the next surprising sound... With the Synth level down, turning Drive down all the way or just adding a touch of it and adding in an octave-up, you get a beautifully shimmering, slightly glitchy Whammy-type octave up of the kind famously heard on Radiohead’s ‘Sail To The Moon’, and many other tracks. This discovery was a breath of fresh air when demo’ing this pedal, as it’s very clearly intended for loud, aggressive pentatonic riff-rock, as most demonstrations out there display. With the Knife Drop pedal being marketed as a horror film soundtrack in a box, I really hadn’t expected to be able to get such a ‘pretty’ sound out of it.

Like all other guitar-to-synth modules, dating right back to the beginning with the Roland GR-500, tracking has always been an obstacle when converting a guitar signal to a synthesized sound. The early analogue guitar Overlay cards allow you to set up the presets (which can otherwise be accessed using an external footswitch) from scratch, and these double up as a helpful learning aid.

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The Drop Off Overall, the Knife Drop achieves precisely what it sets out to achieve. As a signature pedal, it’s obviously very much sonically aligned with the artist’s style of music and intent, but it does have the possibility to expand past that with some creative tweaking. It’s the perfect pedal for ferocious guitar riffs, but it also has the ability to deliver a subtle beauty that may not be the original intention, but saves it from the trap of being a one-trick pony. It would serve a film score to an Italian giallo film very well indeed.

summary An intriguing combination of fuzz pedal and guitar synth, the Knife Drop is a unique pedal, full of attitude and capable of some wonderfully original sounds.

£ £329 including VAT. T Source Distribution +44 (0)20 8962 5080 E sales@sourcedistribution.co.uk W www.sourcedistribution.co.uk W www.eventideaudio.com

20 YEARS OF ACOUSTIC RESEARCH INTO ONE SOLUTION.

Distributed in the UK by emerginguk.com sales@emerginguk.com / +44 118 402 5090

ON TE ST

JAMSTIK MFC1

DAV E LO C K WO O D

T

here was a time, not so very long ago, when a MIDI floor controller that did anything more than just send Program Change messages was considered something really sophisticated.

Increasingly, however, MIDI foot controllers are expected to be fully programmable and able to send whatever MIDI messages your devices may be able to receive and process. Which means, of course, that you have to be able to programme them, and that’s not everybody’s metaphorical cup of tea. Device-specific controllers, pre-programmed with the necessary commands to make the most of a particular unit, give the user the satisfaction of being ‘immediately usable’, but you’d always want the option to do some programming of your own. And exactly how much fun that is, or isn’t, is what divides the men from the boys in this world.

Fortunately, the MFC1, from Jamstik — the people who, in my opinion, make the world’s best MIDI guitars — is programmable from the company’s Creator control app: the same one that works with their MIDI guitars. You can program directly on the unit itself, of course, but you may not actually want to once you’ve tried doing it from the app.

Being, as I am, from an era when you had to be happy fiddling about in hexadecimal if you wanted to do anything out of the ordinary with MIDI, I count myself well versed in this stuff... but these days, I have to admit, if I know that I want to program a specific action with a unit like this, I’d much rather just find it in the Creator software ‘actions library’ and assign it to an on-screen control destination rather than having to try to recall exactly which CC does what, and is it ‘0x7fH’ for ‘On’, or is that ‘Off’? The world has moved on.

Well Connected The MFC1 features three sturdy, momentary (non-latching) footswitches, each encircled by coloured LED rings, providing visual feedback during operation, with three rotary encoders above, each with a with push-down switch incorporated. A 32-character backlit screen shows current parameters and values, as well as allowing navigation through menus and settings. The number and range of connectivity options is a clue to the MFC1’s flexibility in being able to handle a range of diverse setups:

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Jamstik MFC1 MIDI Floor Controller

MIDI pedals are more sophisticated and configurable today than they once were, which is great news. But it does mean the quality of the setup software is really important...

USB MIDI in (USB-A — the ‘flat one’) allows the MFC1 to act as a MIDI host to other devices; USB MIDI out (Type B — the ‘square one’) facilitates outgoing communication with computers and other MIDI-compatible devices; standard five-pin MIDI out offers connection for non-USB-MIDI devices; quarter-inch jack for an external expression or sustain pedal... and finally a 9V DC power input.

The MFC1 will, of course, take power from a USB connection, but it is good to have the alternative for any scenario in which you are not able to use the USB out. You might want to control a synth or an effects unit, connecting just via the ‘legacy’ five-pin port.

Using the Jamstik Creator application (standalone app for Mac and PC, and VST3/AUv3/AAX plug-in) you can select from a range of nine pre-configured onboard stored setups (aka Profiles) or save your own control assignments for particular workflows and preferences.

The footswitches and encoders can be assigned to literally anything within the lexicon of MIDI: CC messages (Continuous Controllers) to modulate parameters such as volume, pan, and effects; PC messages (Program Change) to select between presets or patches; MIDI notes, to trigger samples; and MMC transport controls to offer hands-free control of DAW functions like Play, Stop and Record.

But best of all, for Jamstik MIDI guitar users at least, the MFC1 offers direct control over Jamstik MIDI guitar parameters and device settings such as pitch‑bend, string envelope and MIDI modes, at the same time as passing through incoming MIDI data from the instrument. So, in a DAW-recording scenario, your Jamstik would be connected to the USB MIDI in port and the USB out to the host computer, with SysEx and controller messages going from the MFC1 to the Jamstik, while note data is recorded in the DAW. Of course,

“The footswitches and encoders can be assigned to literally anything within the lexicon of MIDI.” The MFC1 can be powered using USB or a pedalboard-style 9V supply, which is useful if you need the USB connection for other things.

if you’re not using a computer at all, perhaps using the Jamstik guitar to play a hardware synth, the Jamstik-specific messages available within the MFC1 will allow you to still change important Jamstik parameters without needing access to the Creator software. Of course, it doesn’t have to be a Jamstik guitar that you are connecting, it could be another hardware synth, but they tend to have onboard access to settings, whereas a Jamstik guitar does not.

The rotary controllers are used for onboard programming if you are not using the app, but their inclusion hints at this being useful as a simple desktop MIDI controller, as well as for underfoot use. If you want mouse-free control of volume and pan on whatever DAW mixer channel is currently selected, you can just set up a profile for CC7 and CC10. Logic users beware, however: Logic’s mixer doesn’t default to using these parameters, so you have to remember go into Project Settings / MIDI and enable the setting ‘Control Change 7/10 Controls Volume/Pan of Channel Strip Objects’. I’m not sure if other DAWs require something similar, but if it is not working in Logic, that’s probably why.

Another great application is using the MFC1 to control your DAW’s transport using MMC (MIDI Machine Control) messages, which is especially useful for MIDI guitar players who want to keep both hands free for the instrument. Again, Logic users will want to navigate their way down to Settings / MIDI / MIDI Sync Project Settings and enable ‘Listen to MIDI Machine Control’. Other DAWs may vary, but there’s a great resource on the Jamstik website: search for ‘MFC1 Third Party Software Profile Guides’.

are plenty that are more ambitious than the MFC1 in the number of footswitches available — a Morningstar MC6 or Singular Sound MIDI Maestro at a comparable price point are both fully programmable and offer six and eight footswitchable functions respectively.

But if you are a Jamstik MIDI guitar user, I’d say this is clearly the one to have. I know you can change all these parameters from the app, but being able to instantly transpose in octaves or semitones, or enable pitch‑bend only when you want it without having to take your attention away from the composition to look at the app, really does help to maintain a creative flow state. But the big one, for me, is being able to easily switch between MIDI modes — MPE mode is great for working with the Creator plug-in’s own synth voices, but the density of MPE data will overwhelm any virtual instrument that

isn’t set up for it. I’ll often have a project using some Creator MPE instruments and some of the integral Logic instruments.

Jumping between tracks to add parts using the different instruments requires a change of mode every time, and it’s brilliant to do it just from a footswitch.

And the user colour-coding available on the footswitches leaves me in no doubt as to which mode I’m in.

Competitively priced, easy to program and seemingly built to last, be sure to give the MFC1 a look when shopping for a MIDI floor controller, especially if you are a Jamstik MIDI guitar user.

summary A flexible MIDI controller with a wide range of applications, but especially useful for the Jamstik MIDI guitar player.

£ £259 including VAT. W www.jamstik.com

Flexible & Easy To Programme There’s plenty of choice now in the MIDI foot controller sector, and there

The MFC1 is easily programmable using the same app that’s used to configure Jamstik’s instruments.

www.soundonsound.com /July 2025

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ON TE ST

Triton Audio Kompressor Mic Booster & Compressor

This diminutive device promises instant polish when recording dialogue.

NEIL ROGERS

M

ost famous for their FEThead inline mic booster, Netherlands-based Triton Audio have carved out a nice niche for themselves with a range of miniature problem-solving tools that plug straight onto a microphone or between two XLR cables. Their range now includes several booster preamp, phantom power supply, phantom blocker and high-pass filter options, as well as some more creative tools designed to apply different analogue characters to your source.

For review here is one of their most recent releases, an intriguing combination of opto-style compressor and 25dB FET boost stage in their familiar XLR barrel format. It’s designed to partner with low-output moving-coil and ribbon mics in situations where you might benefit from some convenient dynamics control at the recording stage, whether for dialogue

Triton Audio Kompressor €95 pros • Clean, transparent compression.

• Quiet 25dB preamp booster.

• Phantom-power blocking to protect vintage mics.

• Competitively priced.

cons • Hotter sources or mics mean you may need to use an attenuator.

summary A clever, clean and convenient combination of compressor and inline preamp booster, the Kompressor is a brilliant partner for a Shure SM7 used on dialogue, but it has a range of other potential applications, and its diminutive form factor means it could prove a useful addition to any engineer’s kit bag.

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or singing applications, and while it requires phantom power to operate, crucially it doesn’t pass this on to your mic, so vintage ribbon mics aren’t at risk. Of course, this also means it’s not really suitable for use with capacitor or active ribbons, though you could use an external phantom supply, if you really wanted to.

Control-wise, the Kompressor is very simple, with just the level of the mic signal and a single, small knob ranging from ‘less’ to ‘more’ determining the amount of gain reduction. The translucent knob doubles up as a crude visual meter, a small LED lighting behind it to indicate the incoming signal level (and thus the amount of compression).

Word Up On first learning of this device, my thoughts turned immediately to the ubiquitous Shure SM7, now widely used in podcasting but a mic that has a reputation for needing a lot of gain when used for speech. I started by plugging my SM7 into the diminutive Kompressor, and the Kompressor into the Neve 73LB preamp that I’d usually choose for dialogue.

It was immediately apparent that 25dB is a sizeable boost, and while it works great with low-output mics such as the SM7, I suspect that with louder sources and a higher-output mic it could making the signal too hot for your preamp on louder sources. Triton have an answer to that, though: along with the Kompressor, Triton sent one of their Airhead variable attenuators. That’s a really handy tool to have in your kit bag in any case, but it would definitely be a worthwhile addition to the Kompressor if

you plan on using it for instrument recording, not just speech.

Returning to my SM7 test, I was struck by just how transparent the compression sounded on my voice when recording some simple spoken word material.

Indeed, I compared recordings made with and without the Kompressor, and was surprised to discover just how much gain reduction this device was actually applying. I listen to quite a lot of podcasts, and many of them sound compressed to within an inch of their lives, but even with the Kompressor’s gain reduction fully maxed out, the result still seemed to sound mostly artefact-free. Generally, with an SM7 on my own voice, I found myself liking the dial at around 60 percent — it seemed to be a superb addition to my recording chain in this setting. In other words, when it comes to its primary purpose, the Kompressor clearly delivers the goods.

One Trick Pony?

I had various bands come into the studio during the review period, so I tried to make a point of sneaking this little tool into the sessions, just to see if the Kompressor’s potential applications might extend beyond voice work, and to generally develop a better sense of how the compression behaves. With the lack of user controls and visual cues, it’s always going to be a little limiting (excuse the pun!) for music work, but I did like what it was doing when used subtly on a mono drum room mic, and it also sounded good when using a mic closer to kit, with some more aggressive compression on a ‘character’ mic pointed between the kick and snare drum.

It was a little tricky to get a firm sense of the attack and release times but, based on how it sounded on drums, I’d say the Kompressor seemed to have quite a fast-ish attack combined with a slower release. I could imagine using this tool on remote sessions for instruments like bass and acoustic guitars; depending on how good your monitoring is, you could just dial

“I could imagine using this tool on remote sessions for instruments like bass and acoustic guitars.”

in a small amount of dynamic control to help you out in an unpredictable situation.

The Kompressor, left, is adjusted using a single less/more knob that doubles up as a signal level meter. Triton also offer the Airhead variable attenuator (right) which can be handy if you want to use the Kompressor with louder sources or more sensitive mics with a higher output level.

Summing Up If you’re a podcaster or content creator who uses high-quality, low-output dynamic mics such as the Shure SM7, then this new release from Triton Audio could be an excellent addition to your recording setup. Like Triton’s excellent FEThead and competing devices like the Cloudlifter, it will raise the level of such mics to make it ‘play nicer’ with typical audio interface preamps. In that role, it sounds so clean and transparent, and the compressor can be used subtly on the way in, or more assertively to deliver a more ‘finished’ vocal sound that would be great if you’re live streaming.

In my experience of recording podcasts, it can be quite tricky to set the levels optimally, because you often need to keep things annoyingly low to cater for more strident sections and sudden laughs, yet there can be times when people naturally start to speak more quietly too. With the Kompressor in the signal chain, you can

be that bit more confident about setting the levels.

I was pleased to discover that the Kompressor has more to offer than podcast voice compression, and there’s definitely potential to use it for general instrument recording, not least on location — a couple

of these would take up very little space in your kit bag. In that role, I’d recommend taking the time to do a few test recordings, just to get a sense of the compression character, before throwing it into real-world sessions, and at this point I should reiterate that for louder sources than speech, or with hotter mics, you will likely need some kind of additional attenuation to be able to get the most from the Kompressor.

The Kompressor and its partner the Airhead were my first encounters with Triton’s range, and I was genuinely impressed with the contribution these tiny but useful bits of kit could make in my studio. £ €94.95 including shipping.

T Triton Audio +31 616 129 078 E contact@tritonaudio.com W www.tritonaudio.com

www.soundonsound.com / July 2025

43

INTER VIE W

South Korean dramas Parasite and Squid Game have been worldwide hits — and Jung Jae-il’s scores have played a key role in their success.

S O N A L D ’S I LVA

r, e s o p Comducer n Pro usicia &M

J

ung Jae-il’s score for Squid Game is so perfectly in sync with the Netflix show’s unique imagery, it’s hard to believe that the South Korean composer hadn’t visited the sets until well after the thriller had become a worldwide phenomenon. Jae-il finally got the chance before he began scoring seasons two and three “under extreme confidentiality. They had all the cameras on my phone taped up.”

This didn’t hinder the composer’s process in any way, it turns out, because his inspiration comes from an entirely different source. “For me, film is an art of editing, so I don’t get inspired as much from physically visiting a set.

What is written into the script is what inspires me more, and I match that with the scoring and the scenes. I must say, though, the deadline is what really inspires me the most!”

Creating On The Clock Jae-il has been learning to create within practical constraints since he was a young child. By the time he was four he was already playing the piano, and before he became a teenager, he had taught himself to play all of the instruments on Metallica’s Master Of Puppets. An impressive feat, especially since this was well before YouTube provided music tutorials on tap. He became a professional musician aged 12 and was soon touring the country as part of a funk band, learning about performing for an audience and what it meant to work with an ensemble.

Jae-il himself doesn’t appear to find any of this particularly exceptional, because all he was focused on at the time was earning a living as a working musician. “There were days when I was practising my guitar and I would go at it for 10 hours a day; I was completely absorbed by it and crazy for it. And then, I think anyone who has been in a band will know that because of the nature of what it’s like to be in a band,

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Jung Jae-il’s studio is a relatively Spartan affair.

you pick up on other instruments, right?

I wouldn’t say that I mastered any of them, it’s just that I can play them.

I know how to play them, but I don’t think I could be paid to play them.”

When he wanted to study orchestration, he taught himself, starting with a deep dive into a piece that had moved him as a child: Mozart’s Requiem. He has since worked on several orchestral scores, including the Baroque-inspired soundtrack for the four-time Oscar-winning film Parasite, but the lack of formal training posed challenges that Jae-il is acutely aware of.

He says of his process, “Because I was self-taught, it’s very messy. To be very honest, I always wanted to learn formally, but I was hesitant and I missed the opportunity to do so. I didn’t get into the school and that’s why I ended up having to teach myself. There is a lot that is not in line with the theories taught in school, and it takes a lot longer for me.

“I will say though, the strength of being self-taught is that I once had a director say to me, when they heard what I had created, ‘Hey, they teach

you in schools not to do this, but it actually sounds really good!’”

Jae-il progressed by learning from what he loved, absorbing lessons from the electronic sounds of Björk, the hypnotic bass of Massive Attack and the ambient magic of Brian Eno, and incorporating it all into his musical language. He scored his first feature film at 15, released a solo album at 20, and spent the next couple of years writing number one pop hits for Korean artists like Park Hyo Shin and IU. He had achieved his goal of becoming a working musician.

Left Field Then, in 2020, as the world was closing down thanks to the pandemic, Jae-il’s world was about to start really opening up. Parasite made history at that year’s Oscars as the first non-English movie to win the Best Picture award. Then Squid Game seemingly came out of nowhere to become Netflix’s most-watched show ever and a legitimate global phenomenon.

Jae-il’s multi-genre score perfectly complemented the thriller’s impossibly bright colours and stylised costumes.

Children’s instruments such as recorders and castanets sat next to traditional Korean instruments that made way for distorted guitars and delicate piano, creating a world that neither looked nor sounded like anyone expected. Take ‘Way Back Then’, the title track from season one, for example. It started out as a backup tune that Jae-il worked on but was nervous to present to the director because of its unconventional instrumentation. He performed a simple melody on a recorder, built around a 3-3-7 clapping rhythm that is popular as a sports day cheer in South Korea. His lack of recorder playing skills meant that he hit a D# when he meant to play a D, but the director liked it, and it made the final cut, eventually becoming one of the show’s signature pieces.

The standout piece ‘Vote I’ underscores one of the tensest sequences in season two: a vote, the outcome of which could literally be the difference between life or death for all

www.soundonsound.com / July 2025

45

INTER VIE W JUNG JAE-IL

As well as an actual grand piano, Jung Jae-il has a Yamaha digital piano which he uses as a MIDI controller.

involved. Jae-il eschews histrionics and lets the music hang in the air, ripe with tension. “When you think about the act of voting, it can reveal the duality of human nature. Sometimes it makes way for insanity, where the conflict between your side and the other side becomes extremely intense, and then there are also concerns of a betrayal that may take place within your own side, too, so I wanted to portray something that was more subtle and minimal rather than have something dramatic.”

The effect is powerful. When an emotional piece leaves space, the listener can sit with their emotions rather than drowning in big, sweeping strings. Jae-il’s deliberate use of silence comes from his own experience as

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a viewer of art. “When you look at certain paintings from Asia, especially those created during the Ming Dynasty, you will see mountains, a forest, and then, mysteriously, where there is nothing that has been drawn into the space, you get a feeling that you see a waterfall. It’s magical! It’s a very strange and astonishing experience and I like creating that kind of mood.”

Jae-il’s tool of choice for creating that kind of mood is reverb. “You know that feeling that you get when you enter a very old cathedral? It’s almost as if there are angels above, singing in heaven. Those sounds and feelings are created by using reverb. Sometimes when you’re at a cathedral, there will be reverberations that last for almost

15 seconds. And I believe that when I bring that element into my film scoring, you can create a space between notes, which means it gives you room for dialogue as well; that’s something that I purposely intend for at times.”

He considers the late Ryuichi Sakamoto’s score for The Revenant a great example of reverb done right.

“When you listen, in between the notes, it’s almost as if you hear a story; you hear a harmony. It is a beautiful experience, it’s very moving, and that’s what I like to aim for. I would feel too embarrassed to say that I did a great job in using that technique. However, I do believe that when you use a great reverb, that is the kind of sensation or feeling that you can create through music.

INTER VIE W JUNG JAE-IL

“Reverb was probably one of the last techniques that was computerised because it’s so hard to bring to life. I was always drawn to physical reverbs before, and I often used the Lexicon 480L, but since I started using [Audio Ease] Altiverb, I’ve been sticking with that. I am drawn to and prefer natural or organic reverberations, so Altiverb is something that I like to use often.”

Minimal Studio Over the years, Jung Jae-il has streamlined his studio to concentrate only on the gear that is essential to his music-making process. “Someone might visit my studio and think, ‘This can’t be all there is to a music composer’s studio!’ There was a time when I was into collecting synthesizers and guitars

Jung Jae-il is insistent on the importance of auditioning music on multiple monitoring systems, including his ADAM and Neumann speakers.

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but now I have only what’s necessary, so I have my laptop, a grand piano, and about three or four guitars. But because I am a film scorer, I do have a 200-inch screen in my studio.”

Far from being an extravagance, the screen is a key element of his composition process. For example, Jae-il is particularly skilled at writing very tense pieces that are also long, like the eight-minute piece ‘The Belt Of Faith’ from Parasite. Maintaining tension through such a long cue is a challenging task, which Jae-il tackles by composing directly to picture. “I will improvise on the instrument as I am looking at the screen and if something seems to work, we will capture that and then think about how we continue — how do we connect these different

pieces? What’s really important is that it cannot be boring, so I’m almost obsessive about making it dynamic.

“For Parasite, I wrote a piece for an eight-minute-long sequence and back then I thought [director Bong Joon-ho] really trained me well. However, working on Squid Game, [director Hwang Dong-hyuk] broke that record quite easily. We were having a meeting about the score with a picture-locked edit. Usually, how it works is that you watch it together and then discuss where you would like for the music to take place. But Director Hwang actually said, ‘For this sequence, I don’t think we should talk about where the music should be, let’s talk about where we don’t want music to be,’ and so he trained me pretty hard with Squid Game!

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INTER VIE W JUNG JAE-IL

“In scoring, it’s extremely “Because I use new important when the music instruments, I prefer instruments begins and when it ends, and that are intuitive and easy to there are certain directors who play. Some composers like to will control that to the very use their computer mouse to minutest detail. Also, this is draw in the notes, but I like to something very fundamental, play the instruments myself, so but I’m always thinking of for example if it’s percussion, how to use music so that it I would play that. Whether can amplify a certain emotion it’s a piano or a synthesizer or a story. What kind of or a guitar, I treat all of those musical language do I need instruments as if I am playing to use in order to create that on a piano.” effect? Honestly, I am still Piano Wires learning these things and still experimenting a lot.”

Jae-il uses a Yamaha digital He uses two DAWs to create piano as his main source of MIDI his scores: “Logic Pro when I’m notes. “Unfortunately I don’t actually composing, and Pro remember the exact name, but Tools for sound mixing. I believe I am a pianist, so I tend to play 99 percent, if not all recording well on whatever instrument and mixing studios globally, use is most similar to the keys on Pro Tools, so because there will a piano, and rather than using be many cases where we have a MIDI controller, using a digital to go back and forth with files, piano works better for me.”

I use it. It’s not only because The one thing he can’t Pro Tools is such a well-made recommend is a piano sample and great program, but I use library because, “to be very it for its compatibility as well. honest with you, I’m actually The reason I use Logic Pro for someone who’s quite numb composing is I’m not the most to sound. All it takes for me is computer‑savvy person. If I get it has to feel good. However, used to a particular tool I refuse when it comes to the piano, to change it; it’s just that I’ve I am the most sensitive, The third and reportedly final season of Squid Game has been one of the most highly anticipated TV events in recent years. been using it for such a long and so I have to tell you time, ever since I was a kid.” that I don’t use any sample Jae-il takes a similarly streamlined I have put everything into my computer. library for a piano; I can’t give you any approach to virtual instruments and Especially for Kontakt, not only is there suggestions because of that.” sample libraries, with 90 percent of the library that’s built by the company, What he can recommend is which his work done using Spectrasonics’ but also you can add virtual instruments mics to use to record a piano. “The main Omnisphere and Native Instruments’ that have been made externally and mics I use are the Neumann M149 and Kontakt libraries. “I don’t have many that means it has a huge library. I like the AKG C414. I believe the mic you instruments at home, and that’s because to just add to it and stick to that. use is extremely important. I don’t have

Uncertain Futures As a composer who works primarily with film and TV, Jung Jae-il is acutely conscious of the changes that AI may bring. “I believe a media outlet made a list of occupations that AI is going to replace, and music composers were at number three. Also, there was a music composition competition where the grand prize was awarded to a piece written by AI and none of the judges could detect that... it’s a reality that is coming. This is undeniable, so I think that AI is something that we have to be very well skilled at. We have to learn how to utilise it. Personally, I feel both fear and excitement.”

Jae-il also has mixed feelings about the technological revolution that has seen physical media superseded by streaming. “I used to collect records, I had tens of thousands of CDs, but thanks to these platforms, I now have access to hundreds of millions of CDs. As a music fan, I’m very happy about these developments and, on top of that, to have music be suggested to me, I think that’s all for the better. Algorithms and suggestions will differ by streaming platform, but I welcome these changes because the more genres you get to listen to, it’s going to bring us a step closer to achieving happiness in life.

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“But I think it does come with negativity. These changes have led to [a change in] people’s attitude toward music; it’s made it a lot lighter and I think that’s inevitable, right? Because with just tens of dollars per month, you have access to hundreds of millions of different pieces of music. Back in the day when we would buy a CD, sometimes you would gravitate towards a particular CD jacket just because you liked the artwork on it, not knowing what the music was going to be about, and then end up loving it after you listened to it. Also, because it was so expensive, you treated each track so preciously; you would listen from the first track, all the way to the last track and those experiences gave you a lot of inspiration, but I think that’s gone these days. I’m right in the inbetween generation where analogue was transitioning to digital, and so I think I experienced it in a very raw manner. In that sense, I do think that it comes with its negative aspects, but all in all, I think for me and for my life, it has more pros than cons.”

a soundcard that I prefer, but I think whatever recording studio you go to that has a great piano, tends to have great equipment to capture that great piano as well. So I like to just trust whatever the recording engineer will use. I am very picky with mics, though, so I will travel to recording studios that have these particular mics.

“When it comes to piano, I have to play the instrument to the point where I was willing to travel all the way to Norway just to record a piano piece. It was a studio in Oslo called Rainbow Studio that has produced legendary albums for ECM Records, including works by Keith Jarrett. The studio is particularly famous for its outstanding sound in jazz, classical music and piano recordings, and I worked with an engineer called Martin Abrahamssen there.”

The sound that Jae-il most associates with Squid Game is Native Instruments’ Una Corda, “which is very piano-like, but there’s one string, hence the name — Una Corda. I find this instrument invokes a wide range of different emotions: sometimes it’ll be very tense, sometimes it’ll be very serene as well. It feels very intimate, almost as if the piano chord is whispering a story to you. I used it for the ‘Vote I’ piece as well as a piece called ‘I Remember My Name’, in season one.”

Using the plug-in version of the una corda gave Jae-il one big advantage: he could control the level of instrument noise in the pieces: “Sometimes, because that kind of noise can interfere with the dialogue, I didn’t get to include that in the score for the screen itself, but we did include that noise in the OST. That kind of sound gives you a somewhat haunting feel to it, almost like a broken piano.”

compressors that are embedded within Logic Pro. Also, when I get to the mixing studio, sometimes I wouldn’t exactly know what the plug-in is in Pro Tools, but there are these techniques that only my mixing engineer knows how to do. So I like to explain in words: ‘This is what I’m looking for, this is what I want to create,’ and then

laptop and guitar and you can just create right away. I like to use [NI’s] Guitar Rig, among others.”

The other important step in Jae-il’s process is testing the final mixes on a wide variety of speakers. “As a musician, you want everybody else to listen to your music on better speakers than you have, but that can’t be the case, and so the music has to be good, whether you hear it on your iPhone, laptop, or [a speaker with the lowest specs].

That does affect [my compositions], in the sense that I try to steer away from using too intense of a low tone or intense dynamics.”

Jung Jae-il: “What’s really important is that it cannot be boring, so I’m almost obsessive about making it dynamic.” he’ll work his magic. I think he likes to use plug-ins from Waves very often.”

Jae-il rarely uses hardware effects these days, for reasons of practicality.

“When you use hardware, once you record it, it’s hard to go back in and adjust it, right? But if it’s computerised, you can go back and change the tone at any time.

That process has partly to do with the fact that I create music for clients, so if they want to change something, I can go back immediately and change the tone.

Also, there are many cases where I have to be on the go while I create music, so it makes it easier when you have your

Love Me Do The human voice in Jae-il’s compositions is often both heavily processed and layered. “I enjoy layering. Like if there’s the note ‘do’, I like to create different timbres using distortion or delay, for example, so that when you hear one ‘do’ note here, you also hear it from there; I like to use editing to create that effect.

I enjoy using the delay, distortion and

Jung Jae-il’s first taste of worldwide success came with his score for the Oscar-winning movie Parasite.

Work Ethic The most important element of Jung Jae-il’s process, though, is just the act of showing up. When stuck creatively, he says, “There are many ways to combat that but I think I push through no matter what. I just drag myself in front of the piano, sit there, and make myself push through. Sometimes I will remind myself that all because of me, this project, that hundreds of people poured their hearts and souls into, can become a flop, so I will push myself. Sometimes I just like to sit there and let time go by, but I think it’s those times when I force myself to really push through, even though I’m stuck, that I break through and create something out of it.”

With season three reportedly being the final season of Squid Game, Jae-il looks back on what the show has meant for his career: “I am very grateful for all of the different opportunities that it gave me. I was offered so many scripts, so many different projects, and I also got the opportunity to travel to many different countries as well. Having said that, because I am someone who composes music for a client and I also work alone, I don’t think it has changed anything to a point where I felt it sink in. So, regardless of success or failure, I think with every project I work on, I go back to when I first started out, almost as if I’m just beginning my career and I always feel very nervous and tense going into new projects. I thought that kind of feeling was going to go away, but apparently it doesn’t.”

www.soundonsound.com / July 2025

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ON TE ST

A S H TON R E S E A R C H T E T R A PATC H

RJ45 Patching System

The Tetrapatch TRP8 and TFP8 both feature the same complement of sockets, but arranged differently to suit rackmounting and desktop use cases.

Ashton Research Tetrapatch

Network cables don’t have to carry digital traffic. As Ashton Research’s debut range capably demonstrates, they can also work well in the analogue domain.

SAM INGLIS

T

he RJ45 connector and associated twisted-pair cable have long been ubiquitous in computer networking applications. This, in turn, has made them much cheaper to buy than standard multicore audio cables with the same number of cores. Consequently, there have been quite a few attempts to repurpose them for analogue and digital audio, perhaps the best-known to SOS readers being Cranborne Audio’s CAST system.

In principle, the advantages are obvious: up to four channels of balanced analogue or AES3 digital audio can be carried on a cheap, compact and proven physical conduit of practically any length. In practice, different manufacturers have introduced different, incompatible conventions for mapping the channels onto the wires, which has probably held back its adoption in these applications.

The Audio Engineering Society sought to smooth the path in 2019 with the publication of the AES72 standard, but this perhaps hasn’t helped as much as you might expect. It doesn’t mandate the use of any particular pinout, instead merely noting and naming the different pinouts that are already being

“The system is passive and transformerless, so every RJ45-to-jack connection works in both directions.” used. Thus, although there are already some ‘AES72 compliant’ products on the market, they aren’t necessarily compatible with each other.

The latest range of AES72 devices comes from a new company called Ashton Research. Ashton’s initial product portfolio comprises three passive boxes that pass audio between RJ45 and balanced

quarter-inch jacks, plus a rack tray and mounting bracket. All use the 1M pinout specified in AES72 and also employed by the likes of Link SRL, Wysicom and Sound Tools, but not for example by Neutrik in their AES72 stagebox.

Tet A Tet The TFP8 and TRP8 are functionally identical, and differ only in terms of physical layout. Both feature two RJ45 connectors, each with an associated bank of four quarter-inch jack sockets. On the TFP8, all of this I/O is on the front panel. The TRP8, by contrast, has the RJ45s on the front and the jacks on the top, to suit different mounting contexts. Finally, the TP3F is described as a “four-point patchbay slice”. On the rear, you’ll find two RJ45s and four balanced jacks. On the front are eight more jacks, providing break-in and break-out paths.

All can be mounted on the HF100 rack tray, which can accommodate four TFP8s or six TP3Fs in a 1U space. It

Ashton Research’s HF100 rack tray can hold multiple Tetrapatch units in various different configurations.

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could also mount up to six TRP8s, as long as you leave enough space above them for cables. A neat alternative for small installations is the HF101, which is effectively a rack ear that can mount a single Tetrapatch unit to a rack rail.

Ashton Research supplied one of each product for review, along with a couple of Cat7 (Class F) cables. Functionally, there’s nothing to stop you using the more familiar and cheaper Cat5e or Cat6 instead, but Cat7 has better noise rejection and crosstalk specs thanks to its superior shielding. The units themselves seem to be made of ABS or a similar plastic, and feel very tough despite being lightweight. And lightweight they are; although in theory you could use the TRP8 freestanding on a desktop, in practice it would be pulled around everywhere by the weight of the cables, so you probably will want to anchor it to something. The units attach to rack trays and brackets using supplied Torx bolts.

The system is passive and transformerless, so every RJ45-to-jack connection works in both directions.

For example, you could employ all eight jack sockets on a TRP8 as inputs, to be fed from synths and samplers, and use two Ethernet cables to transport these to a TFP8 which would then disgorge them as outputs on its eight jacks; but the same rig could also be employed as a four-in/four-out arrangement, or entirely the other way around.

The TP3F is even more versatile.

Because its rear panel provides two RJ45 connectors as well as a set of four jacks, you can either use it to provide a set of patch points within a cascaded Cat cable run, as a four-channel converter between the two formats with patching built in, or even as a basic passive splitter. This opens up some interesting possibilities. For example, you could station a number of TP3Fs around the walls of a live room, connect them all together with Ethernet cable, and feed the entire system from a four-channel headphone amp. Each musician could then take their headphone feed from whichever TP3F happens to be the most convenient. Alternatively, of course, you can mount multiple TP3Fs on a H100 to create a 1U patchbay with up to 24 inputs and outputs.

Tetrapatch unit is not insignificant, it should be weighed against the cost, weight and inconvenience of the multicore cable that it eliminates.

Every reorganisation of my own studio has meant frustrating hours trying to persuade heavy and uncooperative audio snakes to fit through holes or round corners, and a single high-quality multicore can easily cost as much as several Tetrapatches once you get into long cable runs. Tetrapatch has made me a convert to the joys of AES72, and I eagerly await the day the system is expanded to encompass other connector types such as XLR and DB25.

summary The Tetrapatch system is a neat, compact and affordable alternative to traditional multicore cable runs.

£ TFP8 & TRP8 £109 each, TP3F £139, HF100 £37. Prices include VAT.

W www.ashton-research.com

The TP3F is described as a “patchbay slice” and has numerous possible applications.

Patching Up The entire system worked flawlessly for me, and although the cost of each

www.soundonsound.com /July 2025

53

TECHNIQUE

Part 1: How & Why Stereo Works In the first part of a new short series about stereo recording techniques, we start by exploring the most important tools of all: your brain and your ears!

HUGH ROBJOHNS

D

espite all the contemporary enthusiasm for multi-channel immersive audio, most people still listen to recorded audio in stereo, and this obviously means that understanding and mastering stereo techniques remains a core skill for everyone involved in recording and mixing audio.

Astonishingly, it’s been almost 30 years since I last wrote at length about stereo recording techniques in the pages of Sound On Sound ! But, while it’s true to say that the principles haven’t changed in the intervening years, fashions and preferences certainly have — so I thought it about time I explored this topic once again, in a short series of articles over the next few months.

Why Stereo?

Before we get into practical techniques and practices, the first questions we must ask ourselves are: why bother to record in ‘stereo’? Just what is it that we’re actually trying to achieve? I think there are two answers to these questions, and while both are perfectly valid they have very different goals, and necessitate very different approaches.

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The role of ‘stereo’ in most popular music is generally as ‘ear-candy’, delivering exciting spacious effects.

Often, it’s a case of the wider and more obvious the stereo spread, the better.

Precise stereo imaging and accurate placement of sound sources within a real

acoustic space is usually unnecessary, and often even irrelevant in this context (although there are exceptions, of course).

Consequently, this ‘wider is better’ strategy brings enormous freedom in mic choices, placements and techniques.

A practical example is miking an acoustic

guitar with one mic placed close to the neck/body joint and another (often a different type of mic) near the tail of the guitar, with these two mics panned hard left/right, giving a wide stereo effect. The Glyn Johns drum recording technique is another example of (often) mismatched mics placed to deliver a pleasing and spacious ‘effect’ — even though precise imaging of individual drum components within the set, both relative to each other and within the acoustic environment, is largely lost.

In contrast, recordings of classical music (and most other acoustic genres, in fact) tend to require a far more realistic portrayal of the relative spatial positions of instruments and performers, usually in a stage-like setting, and within a natural acoustic environment. Achieving that necessitates precise and rigorous microphone techniques which, in turn, are built upon a good understanding of microphone properties and a knowledge of how parameters such as mic polar patterns, capsule spacing and mutual angle, mic height and distance all interact and can be adjusted to create the desired stereo imaging and perspective.

So, in this short series I shall be exploring a variety of popular stereo microphone techniques that are typically employed to achieve ‘realistic’ stereo recordings, as well as explaining how various practical tools and procedures can help to optimise these techniques and capture realistic stereo images — as might be employed for recording acoustic ensembles from duets to full orchestras, choirs, folk, jazz, ambient spaces, and more besides.

And for those immers-o-philes who might think stereo boring and old-hat, it’s worth noting that the very concepts and techniques I’ll be discussing in relation to stereo recording can be expanded and enhanced for surround and immersive audio recording applications too!

sabre-toothed tiger’s stealthy approach through the jungle, for example. Having made our brain aware of a potential danger, our hearing then provides sufficient extra information to enable the brain to turn the head to view that danger directly (the sense of sight generally has priority in fight-or-flight decision-making).

To that end, our ears and brain are amazingly well adapted at processing the minuscule differences between sounds arriving at each side of the head to evaluate the likely position of sound sources in 3D space around us. This is called binaural hearing (and is not to be confused with the stereo recording/replay format of the same name). Like all our senses, binaural hearing is not a perfect system and it has intrinsic flaws — many animals can do far better than we can, of course. Nevertheless, our sense of hearing is still pretty impressive, and it’s definitely worth looking after!

For humans, the accuracy of positioning a broadband sound source (ie. one with wide frequency content) varies across both the horizontal and vertical planes. But it’s generally agreed

to be better than two-degree accuracy for sources in the region directly in front of us, then reduces to about a 10-degree error margin towards the sides, and is a little worse for sources behind us.

Our ears and brain rely on three distinct auditory cues to extract information about the direction of a sound source, as I’ll discuss below: inter-aural time differences (ITDs), inter-aural level differences (ILDs), and spectral artefacts (complex comb-filtering effects caused by sound reflections from the torso and outer ears or pinnae). There is a priority order between these cues in determining source location, too, with ITDs being most relied upon, then ILDs, and spectral differences being used to add more detail and deconflict ambiguities — as well as coming to the rescue where true binaural hearing is impaired for some reason.

Inter-aural Time Differences Inter‑aural time differences arise because sound travels relatively slowly through the air (roughly 340 metres per second). If you consider a sound source directly in front of a listener, the sound

What Makes Stereo Stereo?

Before we move on, I think it’s helpful to consider, at least to a basic level, how our sense of hearing actually works, because the various stereo mic techniques are all designed to exploit one or more aspects of our hearing senses, and often they succeed or fail directly because of how they try to fool our senses.

Human hearing evolved, essentially, to warn us of impending danger located anywhere around us — the legendary

Sound waves emitted from a sound source directly in front of the listener arrive at both ears at the same time because the paths lengths are identical.

www.soundonsound.com / July 2025

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TECHNIQUE

HOW & WHY STEREO WORKS

waves it emits will arrive at both ears simultaneously because the path lengths to both ears are identical. However, if that source moves off to one side, then its sound waves will reach the closer ear slightly earlier than the distant ear (because the path lengths are now different). By comparing the timing of nerve impulses from each ear, the brain can work out whether the sound source is located towards the left side or the right, based simply on which ear detects it first.

Moreover, by examining that timing difference more critically, the brain can calculate the angle of sound incidence with surprising accuracy. The diameter of an adult’s head is typically between 19 and 22 centimetres, which means, for a sound source at 90 degrees, the maximum time difference between sounds arriving at each ear is about 0.67 milliseconds. Lab tests suggest that the brain can detect differences in arrival times down to around 0.01ms (10 microseconds) — this is what gives us an angular resolution of about two degrees.

This ITD calculation is thought to be derived mainly from transients within the sound signal, which is probably why it’s often quite difficult to locate sound sources without transients — pure sine wave tones, for example. For periodic signals like these the brain has to use phase differences between the two signals instead. However, the maximum ITD of 0.67ms corresponds to a frequency of 1.5kHz, where the complete wavelength is roughly the same as the diameter of the average head. So, signals with fundamental frequencies higher than this will arrive at the distant ear with a phase difference greater than 360 degrees, and this inevitably creates confusion and ambiguous locational information because a phase difference of 405 degrees looks the same to the brain as one of 45 degrees (45 + 360 = 405), Thus, for periodic signals the ITD method is only reliable at frequencies well below 1.5kHz.

Inter-aural Level Differences The second methodology, Inter-aural Level Differences (ILDs), is caused mainly by us having an acoustic ‘baffle’ between the ears — otherwise known as a head! At higher frequencies (above about 1.5kHz, where the wavelength approaches the diameter of the average head), this causes a ‘sound shadowing’ effect for the distant ear.

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Sound waves emitted from a sound source to one side will arrive at the closer ear before the more distant ear because the paths lengths are different.

Consider a sound source directly in front of a listener, again: the sound level of high frequencies reaching both ears will be the same, because there’s nothing in the way between the source and ears to affect the sound level. If that source moves off to one side again, while remaining at the same distance from the closest ear, the level at that closer ear will remain the same.

However, the level at the distant ear will be reduced slightly because the sound waves diffract around the head, creating a ‘sound shadow’. The actual amount of signal-level reduction is a function of both the angle of incidence and the source frequency; the attenuation due to diffraction increases as the frequency rises and the wavelength decreases.

These two primary sound localisation systems, ITDs and ILDs, function largely independently of each other. In experiments using headphones, though, it’s possible to use a delay at one ear, creating a contradicting ITD, to overcome the perceived source direction generated by an ILD. That’s an effect that can’t happen in real life, of course, but it does illustrate just how easily the brain can be fooled! Importantly, turning the head towards a sound source will always reduce both the ITD and the ILD (turning away increases both), which is another important aid used by the brain to deconflict ambiguities and bring the sound source into sight. This is why people often unconsciously tilt or turn

“Spectral cues are entirely monaural — the sound source location can be calculated from the spectral distortions to sound waves detected using a single ear!”

their heads when listening to something intently: it’s an automatic response.

Spectral Distortions The third location-detection process, spectral cues, is about decoding complex spectral distortions that are created by interference patterns occurring between the directly arriving sound waves and reflections off the shoulders and from the curves of the pinna. Essentially, these are unique comb-filtering response dips, and these high‑frequency interference patterns vary with the angle of sound incidence (both horizontally and vertically). They are absolutely unique to each individual, too, because they are wholly dependent on the exact shape of the outer ears and other aspects of the person’s physique. In early life, the brain learns to correlate these unique spectral distortions with known sound-source locations, and the individual gradually develops the ability to locate sound sources just through recognising specific spectral distortion patterns.

Interestingly, whereas ITDs and ILDs are both binaural hearing processes (meaning the brain makes use of both ears to evaluate the sound direction), spectral cues are entirely monaural As high-frequency sound diffracts around the head a sound shadow is created, reducing the level reaching the distant ear.

— the sound source location can be calculated from the spectral distortions to sound waves detected using a single ear! More importantly, spectral cues are particularly important for resolving front/ back confusions and source locations in the vertical plane, which ITDs and ILDs can’t differentiate at all.

Monaural Location This mechanism explains how deafness in one ear doesn’t destroy the ability to locate sound sources. If you’d like to test it, try lying in bed with your eyes shut and one ear buried in a pillow (thus removing the option for binaural hearing) as a partner or child enters the room and moves around the bed. You’ll find you can estimate their position in the room with surprising precision, even without binaural hearing!

And, if you think about it, one reason headphone listening can sound artificial, with sound sources perceived as being inside the head rather than in the space around it, is that these spectral cues are inherently absent; the sound is essentially injected straight into the ear canal without the benefit of reflections from the pinnae and shoulders. This missing information confuses the brain to decide sounds

must be ‘inside the head’. Even with true ‘binaural recordings’ — I’ll be discussing those in a later article — the perceived ‘outside the head’ sound locations are often variable for different listeners.

These ‘problems’ can be addressed very effectively using crossfeed matrix processing, which simulates the natural ‘crosstalk’ of sounds arriving at both ears in the real world, and personalised Head-Related Transfer Functions, or HRTFs. HRTFs are a set of spectral (frequency) modifications that are applied to the binaural signal, to simulate different intended sound‑source locations. But, while generic HRTFs can certainly be helpful, if we’re to perceive accurate and stable image locations we require HRTFs that are designed specifically for each individual — that’s because we each have uniquely shaped pinnae and different physiques. Thankfully, technology is starting to make that a lot easier to achieve, with quite good results coming from analysing photographs of an individual’s ears, for example!

Auditory Cues, Mics & Loudspeakers So, our ears and brain derive the location of a point‑source sound through ITDs, ILDs and spectral cues, and we try to recreate that information by using two loudspeakers spaced apart, preferably at ±60 degree angles to our listening position. What could possibly go wrong?

Amazingly, this peculiar scheme works well, fooling our ears/brain into creating ‘phantom sound sources’ that appear to be positioned somewhere in the space between the speakers. Technically, it recreates the sound wavefronts required to work very well at low frequencies (below about 700Hz), where level differences between the two speakers create accurate ITDs at the ears (that’s right, time differences! More on that in a future article...). At higher frequencies, the system doesn’t work quite so well and, in effect, creates spatial aliases that smear the soundstage and reduce imaging accuracy.

For stereo recording, the microphones are tasked with capturing ILD and/or ITD information in lieu of our own ears, and different stereo mic techniques achieve that in different ways. I’ll be looking at these, and delving into the mysterious workings of stereo loudspeakers, in the next article in this series.

www.soundonsound.com / July 2025

57

COMPE TITION

Win! Tantrum Audio Angry System Worth £1640 S

ingle-driver monitor speakers have a proud history in the recording studio, and last year, British company Tantrum Audio brought the concept bang up to date with their Angry Box speakers. A world apart from the low-tech monitors of old, the Angry Box boasts 65 Watts of Class‑D amplification and an Analog Devices digital signal processor, allowing the user to switch between an old‑school ‘mid focus’ mode and a high‑fidelity ‘extended bandwidth’ mode, which significantly extends its frequency response to compete with conventional two‑way speakers — essentially giving you two monitors in one.

When Mike Senior reviewed the Angry Box in SOS last year, he found it to be “extremely effective at clarifying critical questions of midrange balance”

in mid focus mode, while the extended bandwidth setting made them a unique tool in the professional monitor market.

For this month’s exclusive SOS competition, we’re giving you the chance to win the Tantrum Audio Angry System: a complete monitoring solution

To enter, please visit:

ample 102dB, while the sub’s sealed‑box architecture means it doesn’t suffer from the resonances and pitch uncertainties that can afflict ported subs.

Further refining the Really Mean Sub’s performance is an integral DSP engine, which both automatically integrates the sub with your full‑range monitors, and performs full room correction for both the subwoofer and your full‑range monitors.

The result, say Tantrum Audio, is “tight, phase‑coherent bass that feels like a natural extension of your monitors — not an add‑on”.

To be in with a chance of winning this fantastic prize, simply follow the URL shown and answer all the questions there, by Friday 1st August 2025.

Good luck!

https://sosm.ag/tantrum-comp-0725

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comprising a pair of Angry Box monitors and the company’s brand‑new subwoofer, the Really Mean Sub. The Really Mean Sub is based around a pair of super‑fast, high‑excursion, 6.5‑inch woofers powered by a 400W ICEpower amplifier — a combination that yields a mighty LF extension of 15Hz at the -10dB point (and only a single decibel down at 28Hz). Maximum SPL is an

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ON TE ST

Dreamtonics Synthesizer V Studio 2 Pro Software Vocal Synthesizer

Dreamtonics take their session singer in software to the next level.

JOHN WALDEN

I

love it when a music technology product leaves me believing in magic.

Working with Dreamtonics’ Synthesizer V — which I reviewed in the March 2023 issue of SOS — was undoubtedly one such moment. Dreamtonics have now released version 2 of Synth V. This brings a whole raft of enhancements and new features that, on paper at least, seem very intriguing. To be on the safe side, I placed a pillow on the desk under my chin before exploring. Did I need it? Let’s find out...

Sing It Again I covered the basic features of Synth V in the original review so, aside from restating that v1 was already capable of generating some amazingly realistic sung vocals (realistic to the point that many listeners will not realise that they are hearing a synthesized voice until you tell them), I’ll focus on what’s new here. If you

Dreamtonics Synthesizer V Studio 2 Pro $118 pros • Enhanced workflow options well implemented.

• Undeniably impressive vocal synthesis engine.

• A fabulous virtual session singer platform for songwriters and music producers.

cons • Does require some workflow changes compared to v1.

• The psychological questions you face when the most human part of a song no longer needs to be human.

summary Synthesizer V Studio Pro 2 improves both the realism of its synthesized vocals and the ease with which they can be obtained. This is truly remarkable technology.

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Synth V’s UI has been reworked in a number of ways in v2, to reflect some significant refinements in the software’s toolset.

need to catch up, then the original review is available to read in full on the SOS website and the underlying principles still apply even if some operational changes have been made.

That said, I should also point out some highlight additions that arrived as updates to v1 of the software after that original review: a rap engine designed specifically to create rap vocals, and a voice-to-MIDI conversion option allowing you to automatically extract note/lyrical information from your own recorded vocals and turn them into the data need to generate a Synth V vocal. Spanish language support was also added. And, while running as a standalone application (which is how I did much of my own testing here), Synth V is also available as an ARA plug-in within suitable hosts.

For those new to the product, it’s also worth emphasising one key concept; Synth V is a virtual instrument. You create the melodic (and lyrical) content of the performance and Synth V’s engine then renders that information into a synthesized vocal based upon your selected virtual singer. The engine utilises AI/machine learning to process detailed samples of the actual singer from which

the selected voice library was built and to resynthesize those samples into the performance you have constructed. This includes the pitch, lyrical content, the choice of singing ‘modes’ (styles such as clear, muted, belt), characteristics (such as loudness, tension and, breathiness) and phoneme duration and pronunciation.

Make an edit and Synth V resynthesizes — not in real time, but very rapidly — the required changes.

The principle is, therefore, no different from the way you might create a virtual solo violin or guitar part; a sample/ synthesis engine interprets the MIDI and sound parameter information you have supplied into a realistic performance. It’s the same here but with the human voice.

Dreamtonics now explain very clearly on their website how the process works and how they have collaborated with (and compensated) the original human singers upon which each Synth V voice library is based. This is not AI vocals generated from a text prompt.

New For Two So, what’s new in version 2 of Synth V? There are some obvious headlines. On a technical front, the rendering of edits

(to notes, pitch curve, voice parameters, lyrics, etc) is now considerably faster. I did much of my testing on an M1-based Mac and re-rendering after any edit was pretty much instant. It’s a very slick experience.

The overall UI remains familiar from v1. The track-based Arrangement overview dominates the top (where you can house both audio tracks and Synth V vocal tracks), and the detailed view of the currently selected vocal track sits underneath. The latter is, itself, split into two zones: the piano roll, where you can edit note, pitch curve and lyrical content; and then optional lanes for automating the various vocal modes and parameters.

In addition, a series of six panels — Voice, Notes, Dictionary, Render, License and Settings — can be toggled open/closed as required to access further parameters and settings.

However, digging in to all these elements reveals numerous changes, additions and enhancements. For example, in the Notes panel, the synthesis engine options have now been consolidated into two main modes of operation — Sing and Rap — and this panel now includes the Retakes options that used to have their own panel. The Retakes feature lets the engine generate alternative versions (takes) of either a whole vocal or just selected notes. This has been rather neatly reworked with new options for just varying timing, pitch, timbre, or all three together.

There is also a new X/Y panel allowing you to quickly vary the ‘expressiveness’ used by the engine and the amount of vibrato modulation applied to longer notes.

This is all very easy to use and you can swiftly switch between your various virtual takes to see which you prefer for any given phrase.

The Voice panel, which contains parameters for your selected voice library, has also been refined. For example, while the Parameters section is as before, the Vocal Mode section — where you can adjust the level of the virtual vocalist’s various singing styles — now allows each mode to be popped open to provide individual control over the Pitch, Timbre and Pronun(ciation) associated with each mode. This provides you with a much greater degree of control over the character of the voice.

This panel also used to house controls to vary vibrato and pitch slides into and out of notes, but this functionality is now replaced by the additions to the Notes panel and by additional options to edit

Detailed pitch editing now includes the ability to define nodes on the pitch curve that can guide the synthesis engine.

the detailed pitch curve in the piano roll and automation lanes (more on this in a minute).

Changing Lanes In operational terms, v2 also brings some key changes in the editing and automation panels. Editing notes and lyrics remains very much as before. However, you now also get a Phoneme lane located directly beneath the Piano Roll editing panel. This allows you to adjust both the strength and duration of individual phonemes in an entirely graphical environment. It’s a much easier process than previously provided within the Note panel and allows you to finesse the pronunciation of words in a much more intuitive fashion. Detailed editing of the pitch curve was possible in v1 but the process has been significantly refined in v2 by the ability to add (and edit) Pitch Control Points directly on the curve. You can also draw pitch curve sections in freehand. The way both of these elements are then used is also very clever as, once a point of manual curve segment is placed, the engine will then ensure that the synthesized pitch curve does its best to pass through these points/ curves while still generating a natural result. This system provides powerful control over pitch transitions between notes or vibrato in sustained notes, for a much-improved workflow over v1.

Practice Makes Perfect The toolset changes do mean that existing users will have to make some adjustments to their established workflows, but Dreamtonics have undoubtedly streamlined and improved the overall process with these changes.

Editing is much faster, and the toolset feels much more refined. The (very welcome) improved rendering speeds contribute to the enhanced workflow.

Of course, just as (or even more?) important is whether the synthesis engine delivers improvements in the actual vocals created. The short answer is yes; once you have familiarised yourself with the changes to the toolset, the results can be incredibly realistic. I got to test a number of the updated native English voicebanks: Dreamtonics’ Kevin, Natalie, Hayden and the new Liam, for example, and Eclipsed Sounds’ (one of the main third-party companies creating voicebanks for Synthesizer V) Solaria, Saros and Astarian. As in v1, Solaria was particularly impressive but the additional details within each voice’s Vocal Modes really do let you conjure more ‘human’ in the resulting vocals, providing you are prepared to automate the available parameters. And, if you need even more voice options, Dreamtonics’ Vocaflex (see the December 2024 review) provides plenty further potential; it works just as well with Synth V vocals as it does with those of human singers.

If you want to hear some examples there are a few short clips I created during the review period on the SOS website. The most obvious use cases for this virtual instrument is for songwriters and/or music producers developing song ideas ‘in the box’. However, whether you eventually use Synth V’s vocals as a guide to a real singer, or they actually make it to your finished mix, for writing and editing vocal parts, Dreamtonics have created a brilliant creative tool.

I’ll leave a discussion about the desirability of synthesized lead vocals for another day (but it’s undoubtedly a very interesting debate). Did I need that cushion under my chin? Well, Synthesizer V Studio 2 Pro improves on what was already an impressive product. With a little practice, it’s perfectly possible to generate lead vocals that will jump well over the quality threshold required for a number of very practical use-case scenarios. That in itself is a remarkable thing to be able to say. This software makes me smile every time I use it.

Synthesizer V Studio 2 Pro is amazing technology with a surprisingly modest price tag.

Audio examples for this article can be found at https://sosm.ag/ DreamtonicsSynthV2 £ $118.80 including VAT. W www.dreamtonics.com

www.soundonsound.com / July 2025

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FE ATURE

Hear The Sound W https://open.spotify.com/

track/2h4Lh23FF8OQ8uGBnOdepy

W www.youtube.com/

watch?v=csm7MUXxb74

Stephen Power • Robbie Williams ‘No Regrets’ J O E M AT E R A

E

nglish producer and mix engineer Stephen Power has worked with artists ranging from Blur and Andrea Bocelli to Diana Ross and the Bangles.

Power co-produced some of Robbie Williams’ biggest-selling albums and, asked to select a favourite sound, Power recalls how he got the layered drum sound on Williams’ 1998 single ‘No Regrets’.

Half Power “The original demo of the song was in half time, but because we wanted to get it on the radio as a single, it meant we had to get some double-time stuff on there. So, the intro and the end are both in half time, while the body of the song is in double time.

“This was in the days of tape, and was the last Robbie album that was recorded on tape. No Pro Tools rig in this session!

I started programming with a load of loops and several of them in half time and several

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of them in double time and also some backwards loops. And then we also used the tape to play double time and then slowed the tape down to half time. Then recorded that into Logic and put that back onto the tape. What I ended up with were loads and loads of rhythm tracks, including three where Chris Sharrock played a rolling tom-tom part with the rest of the kit during the double-time body of the song. You can’t really pick that out, because the idea was to make it epic-sounding — kind of like a Phil Spector thing. The idea was to get everything doubled and repeated and just try and make this collage of something that would sound epic out of lots of sounds.

“At the beginning and during the track, you can hear the reverse hi-hat going.

And it was at a time when you couldn’t line everything up, because it was done on tape. So, it was more a case of filtering out the low end of the loops so that the real bass drum is the one that you hear. And basically, that was it. All these collages were made out of half-time loops, reversed loops, half-time playing full speed, and a tom-tom part. And then when it came to mixing it, I didn’t want it to sound clear. I wanted it to sound like a magical blend of stuff that you

couldn’t really figure out what the hell was making the rhythm track. It took us a whole day of picking the loops, a whole day of the drummer playing parts, and then a lot of messing around in the mix to get a blend together of everything working correctly to try and make a motoring sort of epic backing track.”

Going Epic “The loops were all recorded through the SSL at Trident Studios. The live drums were recorded with a Neve console, and it was all mixed on an SSL. I used some compression on each loop as I recorded them, though didn’t use any compression on the live drums. Everything then went through an SSL quad compressor to try and hold the whole track together. As I didn’t do any group compression of the loops, the only group compressor used was the quad SSL over the whole mix and individual EQ’ing and compression again on all the loops. So that’s why it sounds kind of like a steamroller thing, where you can’t really figure out what’s playing what.”

ON TE ST

Boss TAE Core

Reactive Load, Speaker Simulator & Re‑amplifier

Offering all the essential facilities at a lower price point, could the new Core version of the Tube Amp Expander be a serious competitor to the original flagship model?

DAV E LO C K WO O D

T

he Boss division of Roland announced their original flagship Waza Tube Amp Expander at 2019’s Winter NAMM show, putting it head-to-head with Universal Audio’s OX tube-amp load box and speaker simulator, and we reviewed it in SOS September 2019 (https://sosm.ag/ boss-tae). Five years on, that ‘big box’ version is now joined by the TAE Core, which is half the size and half the price, but offers by no means half the functionality. The TAE Core, like the original TAE, facilitates direct recording of a tube amp without the need for a guitar speaker. The amp feeds the TAE’s dummy load, where a line-level signal is derived to pass through an A-D converter. A range of digital effects can then be activated, before the signal is split into two paths: one to feed an impulse-response host,

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where you can access both the integral cabs and any other speaker IRs you may choose to load; and the other to feed an internal solid-state power amp to use with a real speaker.

Overview The Core’s reactive load is switchable between 4, 8 and 16 Ω, with power-matching options of 10, 50 and 100 Watts, primarily I’d imagine for internal level optimisation. Maximum power input is stated as 100W, and there is a ‘safety load’ in place if you forget to power up the TAE before switching the amp on. Once you are all hooked up, the Speaker Out socket on the back of the TAE Core, which is fed from the internal 30W power amp, can then be used to hear your safely-loaded amp at whatever volume setting you like, via a fully variable (not stepped) level control on the front. A pair of balanced XLRs

convey the DI path’s speaker-emulated signal to a PA mixer or recording system.

Apart from the fact that it was possibly the first integrated re‑amping solution, as opposed to an attenuator, the original TAE’s ‘big thing’ was that it offered a user-adjustable impedance curve, rather than the fixed response offered by most dummy load/attenuator boxes.

It offered 16 possible response shapes via its Resonance Z and Presence Z front-panel switches. In the guitar world, we tend to think of speakers just in terms of 4, 8 or 16 Ω units, but an actual impedance will always vary significantly across the whole frequency range. The significance of this parameter is that the impedance curve determines the power transfer at different frequencies, which, of course, becomes a factor in the actual sound that we hear. The purpose of including the user-adjustable reactive load is that it allows the user to approximate the impedance curve of a number of different types of speaker, thereby making the connected amp behave more accurately like it would if it were connected to that specific driver.

Now, the reason I’m dwelling on these controls on the original model within a review of the new one is that the TAE Core doesn’t have them... But I think it is none the worse for it. The load settings for the Core are just a choice of Combo or Stack, perhaps replicating a single-driver open-back cab and a closed-back 4x12.

Whilst the more sophisticated control setup could allow you to fine-tune an exact the response, it could also allow you to get it a bit wrong. A single, fixed impedance curve is likely to be optimised for just one type or the other, but a choice of the two primary types really has got you covered without having to think too deeply about it. Yes, the impedance curve matters, but so do all the other variables — the virtual mics, the effects and EQ, and the room ambience — as well.

Speaker Emulation You can use the TAE Core’s speaker-emulated outputs straight out of the box without firing up the dedicated software editor — there’s a speaker and effects setup stored in every one of the 10 settings of the front-panel Rig control — but you’d be missing out on quite a lot.

The editor, available now in both desktop and mobile formats, and connecting over USB-C or (using an optional adaptor) Bluetooth, allows you to store your own Rigs (whole setups of routing, IR, EQ and effects), with 10 of these able to reside within the TAE hardware for instant recall via the front-panel control, the app, or MIDI, whilst many more can be stored

in your library via the app. The editor improves on the original TAE editor in that it now also includes the IR loader, which makes a lot more sense than having separate apps. Apart from the Speaker Out level and reactive load setting, the only other control knobs on the front are the 10-way Rig selector, line out, phones level and ‘Air Feel Level’. This last control is a tight, stereo ambience applied to the line out, headphones and USB digital outputs, but, logically, not the speaker feed. Air Feel opens out the close-miked dryness of the IRs into something more like an in-the-room sound that is far more comfortable to play with, without having to make the sound noticeably reverberant unless you turn it up a long way.

Round the back, power connection is from a 12V DC external adaptor, with a chunky right-angle connector that inspires a lot more confidence than the standard DC connector used for pedals.

There’s now a single speaker output; the ‘big’ TAE had a paralleled pair. The power amp specs at 30W into 8Ω, with a minimum impedance of 4Ω (which would obviously give you a bit more output given that this is a solid-state power stage). Driving a single 12-inch speaker in an open‑back combo cab, I could comfortably get up to what counts as ‘performance volume’ for me these days. I guess it might not be enough in a large band line‑up with a hard-hitting drummer, but who still gets to really crank it up on stage these days without being told to ‘turn it down’ and ‘get some

Boss TAE Core £605 pros • Simplified reactance setting.

• IR loader now integrated into the editor.

• Improved software editor with connection via Bluetooth and USB-C. • Comprehensive remote control via MIDI. • Fully variable ‘attenuated’ volume.

cons • Poor scaling of software controls.

• No gain-reduction metering or EQ-curve graphic in software editor.

• MIDI footswitching required for stage use.

summary The Boss Tube Amp Expander Core is a great little package — dummy load, speaker sim/IR host, integral power amp, onboard effects and MIDI control, at a significantly lower price point than the original. Whilst the control app is undoubtedly better than the original TAE editor, this version still feels a bit more like a ‘point-one’ tweak than a ‘five-years-later’ revamp.

in-ears’? At lower levels, it’s great for just practicing with an authentic overdriven amp sound and, of course, it’s perfectly valid to still mic the speaker for recording rather than using an IR. It’s not exactly the same as having the output stage drive the speaker directly because you’ve broken the interactive feedback loop that gives that relationship some of its feel, but the moment you stop thinking about

The rear panel has impressive connectivity options, including send and return jacks for an effects loop.

www.soundonsound.com / July 2025

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BOS S TA E COR E

With a block diagram printed on the top of the TAE Core, you’ll never be in any doubt about the signal flow.

it and listen to what you’ve actually got, it’s perfectly fine!

The analogue line outputs appear as a stereo pair of XLRs. You can’t pan the mics, though, so the only stereo elements are the delays, the reverb and Air Feel effects, and the output from any stereo IRs you may have loaded. If there’s nothing important going on in the stereo field, you can of course just use one output for mono. An analogue send and return loop, switchable for -10dBV or +4dBu operation, allows you to patch in a hardware effects device, with the loop status (on/off, series/parallel) set as part of the stored data of a Rig. For silent practice, the headphone output has independent level control, but there’s no obvious way of connecting a backing-track player in the analogue domain, other than using an external mixer, as the effects loop input returns before the speaker-sim stage, and you’d want the backing track to remain full-range. The USB connection will let you do it, however, and you can also stream into it via Bluetooth. The TAE offers two USB audio output streams: Line and Pre-Effect, the latter taking a mono

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signal from immediately after the external effects loop and thus before the internal effects and speaker sim. You can record from both simultaneously, so if you like the performance but not the sound, you can route the Pre-Effect signal back through the TAE and try a different IR. The revised version of the TAE editor is an improvement on the one for the original TAE, which was functional at best. In the Cab/Mic screen we’ve now got a simple pictorial representation of mic positions that change as you alter the distance and offset-from-centre parameters. It does help, especially when you are ‘positioning’ two mics.

You can also change the cab type at this point, with a comprehensive choice of 4x12s, 4x10s, 2x12s, 1x12s, 1x10s and even three 1x8s.

As in the original TAE, some virtual cabinet setups are tagged as ‘for FOH’. These are usefully pre-tweaked specifically for feeding a PA system,

perhaps taking the mic slightly off-centre to reduce harshness and taming a bit of the low end. I like what it does to the 4x12s even for recording, whilst some of the smaller speakers tend to lose a bit of the detail and idiosyncrasies that make them interesting, which is, of course, the point for FOH, where that stuff would just be lost anyway.

An interesting addition to the processing line-up, along similar lines to the ‘FOH’ tweakery, is the new Mix Assist feature. This applies user-variable degrees of EQ and dynamics processing, aiming to curate the final output, like a good engineer might, without you having to do it for yourself. Setting 1 “reduces the high-frequency range that overlaps with the hi-hat and the low-frequency range that is difficult to distinguish from the kick or bass guitar, resulting in a sound that blends better when mixing” (basically a ‘bracketing’ EQ), while settings 2 and 3 combine this with varying degrees of mid push. The EQ works in conjunction with the Transient control, which you can use to target high-level spikes or give the sound more attack. None of it is too extreme, so it is

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ON TE ST

BOS S TA E COR E

onboard Rig slots (a TAE Core Rig is a single stored configuration of your cab and mic choices and effects settings) that you can then recall using simple MIDI Program Change messages. Sets of Rigs can be saved as LiveSets, perhaps for grouping Rigs together for a specific set list or a recording session.

The Last Word

Air Feel is a clever effect that can counter the dryness of close-miked IRs, leaving you a sound that can make playing feel more comfortable.

actually quite useful, and used to good effect in a number of the presets.

The mic choices of the original TAE (Shure SM57, Sennheiser MD 421, Neumann U87 and the slightly oddball AKG C451 — was that ever anyone’s first choice for miking a guitar cab?) remain, but are joined now by a Royer R-121 ribbon. Yay! That’s great for pairing with one of the dynamic models.

Flipping from Preset to IR mode lets you access the included set of very nicely curated Celestion IRs, or anything of your own you may have loaded. You can even tweak the time alignment of any double-miked IRs, but not the distance and position, which is baked into the IR. User IRs need to be WAVs at 44.1, 48 or 96 kHz, 16-/24-bit linear or 32-bit floating point.

You can use mono or stereo IRs of up to 500ms — personally, I’ve never been able to discern any benefit in going that long, but I know others think differently so having the option makes sense.

or graphic) for the speaker and line outputs, as well as speaker and line output level controls. The EQ can be employed as a useful Solo Boost facility, but there is also a global EQ that lives outside the Rig settings which you can use for ‘environmental compensation’, adapting the sound to the acoustics of your venue, perhaps. Like the line and

The new Core edition of the TAE represents a well-thought-out rationalisation of the facilities of the ‘big’ TAE. It lacks nothing that I’d call essential, and ‘tidies up’ in a few places, as well. It’s significantly more affordable, easier to transport (the big one weighs a ton) and has a better app. One perceived limitation in this configuration, for some at least, is going to be the 30W power amp. On its own, it’s not quite enough to serve as ‘backline for the room’ with a loud band, but I guess the thinking is if you’re using a re‑amping attenuator to control volume, you probably don’t want to get it back up to the volume you started with. I think having just a bit more output would have been nice for the scenario where you are using the TAE Core to boost a little amp like a Champ up to stage levels, or indeed using a modeller into the effects return as an emergency backup rig.

For recording, it’s fine. It might seem perverse to mic a re‑amped speaker when you’ve already got a line-level IR recording output available, but it is sometimes just more comfortable for the player to be able to interact with a speaker in the room.

Record both and see which one you like best. There’s a polarity flip on the speaker output to get it ‘round the right way’ with the IR if you need to, and you can switch in a compensating Fletcher-Munson correction to boost low and high frequencies for when the re-amping volume is set very low.

Both the line and speaker paths go through the 32-bit/96kHz converters — there’s no all-analogue path through to the speaker — so there’s no digital latency offset between the two paths to worry about there.

Unless you’ve messed up your gain staging somewhere, the fundamental sound of the TAE Core is as sweet, clean and low-noise as you could wish for.

There is a fan, but while I hate to have

“You can switch in a compensating Fletcher-Munson correction to boost low and high frequencies for when the re-amping volume is set very low.”

Editor The Editor page is a simple block diagram, with the ability to set parameters for the onboard compressor (modelling a dbx 160 or UREI 1178), delay (mono, stereo, faux-analog, and Roland SDE-3000), reverb (Halls, Rooms, Plate and Spring), plus separate EQ (parametric

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speaker EQs, the global EQ can be both set and switched separately for each output, or tied together.

The effects appear in both the line and speaker outputs, as will anything patched into the effects loop, which gives you the option of ditching a pedalboard if you only have fairly basic effects requirements. You will have to use MIDI to control everything though, as there’s no option for using one of the nice Boss multi-footswitch units that connect with just a TRS cable. You can get at all the parameters via MIDI CC messages, but if that’s a bit daunting you can save different effect statuses using the 10

The TAE Core can be configured using a desktop or phone app. Shown here are some of the screens from the mobile version.

anything with a fan in a control room, other than a little ‘check turn-on’ when you first fire it up, I didn’t manage to get it to spring into cooling action. Running at full-whack for a few minutes, I got too hot before it did! I’m sure it would have done if I’d fed a more powerful signal into it requiring more load dissipation.

Whilst the app is certainly better than the original TAE editor, personally I still find it disappointing that there’s no gain-reduction metering for the compressors, and no visual interface for the EQs — that’s pretty standard stuff in any digital audio software these days.

And the control scaling is still oddly dysfunctional in my opinion. You’ve got up to 2000ms of delay available, but the range from 0 to 400 or so — that’s the

bit that most players are likely to actually use — is crammed into the bottom quarter of the rotation. Similarly, the halfway point on the reverb decay time is five seconds. Anyone used more than that on a guitar recently?

Finished whinging? Not quite.

Having a desktop editor with a wired connection is great, but why does it have to require a ‘direct’ connection to the computer to work at all? My ports are all full of essential things, including hubs that are supposed to be for connecting ‘other things’... And the app takes an age to connect via Bluetooth — long enough to have you seriously wondering if it has stopped trying, and definitely something that would get you agitated if you needed an urgent remedial

adjustment during a soundcheck. But, credit where it is due, it has never once actually failed to connect for me.

So, is there any reason to still buy the ‘full-fat’ TAE? Well, a lot more power is the obvious one, and the ability to fine-tune the load impedance curve, but there are also little useful facilities like the Amp Control feature for channel switching that can be set as a parameter within saved rigs, and the option to use a dedicated Boss footswitch. So there are plenty of reasons for both models to coexist. But if none of those things are a ‘must have’ for you, the new TAE Core is an absolute winner. £ £605 including VAT. W www.boss.info/uk

www.soundonsound.com / July 2025

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Archaea Exchange Eurorack Module

T

he Archaea Exchange falls into what is possibly my favourite category of modules: essentially very simple but with massive scope. It has eight inputs on the left and eight outputs on the right. Patch a signal into one of its inputs and then, with that input selected via its own light‑up button, simply push the buttons corresponding to the outputs you want to send that signal out of. I suppose the very simplest application of this is to patch a single input and use it as a buffered eight-way mult — and with no complaints; it’s very useful being able to switch various outputs on and off. For instance you might audition or combine different oscillators being controlled by the same sequencer, or use it to performatively send an LFO to various destinations without having to keep un-patching and re-patching.

The Exchange, of course, can go further than this. With all eight inputs patched, you can select any combination of outputs for each — although be warned you can’t use it to sum several input signals to one output. I’m sure the developer would have a good answer for why this parallel mixer functionality wasn’t possible (in fact I’d imagine it was one of the first questions on the R&D table), but at the end of the day there are plenty of Eurorack summing mixers available, some as small as 2HP, and I daresay it wouldn’t be a bad idea to have one nestled alongside the Exchange for when the need arises.

Once you have a patch in place, you can save it to one of 64 preset slots, any of which are accessible in two button pushes (eight banks of eight presets, if it wasn’t clear). Here’s where the Exchange shines, in my opinion. Once saved,

you can use the Copy and Clear functions to move presets into different locations if desired, and once they’re there, you can sequence them in all manner of ways — which is another way of saying you can sequence patches. The Exchange supports an astonishing variety of methods to do this: you can use USB MIDI and send notes corresponding to different presets, or you can assign input 8 to accept triggers or CV. If set to expect triggers, it will progress incrementally through a designated range of presets with each pulse, representing the progression with its light‑up buttons. If set to expect CV, however, it will divide 5V by the number of presets in the range; so you can sequence them in any order from a CV sequencer, randomise them with stepped voltage or for that matter send an LFO to have them sweep back and forth. Very, very fast if you want to. It made for some wild and unpredictable results, and in honesty challenged my patching technique a bit too, since it required not a little parallel thinking to create a patch and then sufficiently vary it into sequence-able sub-patches.

Now is probably a good time to say that, of course, things like the Erica Synths Desktop Matrix Mixer are juggernauts in the patch-recall department. That unit can handle 256 connections, accept MIDI Program Changes, and so forth.

What it can’t do, though, is sit in the very middle of a system, patched in and ready to go, and it can’t change patches with CV. Also, in my opinion the linear ‘Y-Y’ layout of the Exchange (as opposed to X-Y) suits it perfectly to being sequenced. Simple enough to use at speed, complex enough to facilitate deep-diving. In this way

XAOC Devices Lublin Eurorack Module

T

he Lublin 1957 from XAOC Devices is a two-channel pingable VCA module with three ping responses and a gain limiting circuit to prevent peaking or distortion. For anyone who doesn’t know, a pingable VCA has a Ping input which

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it can either be a type of ‘set and forget’ module or it can be a highly performative and dynamic one, playing a key role within a patch as well as external to it. It’s also worth mentioning that Archaea have purposefully created a substantial expansion system to support multiple Exchanges in a setup, if you do have need to exponentially increase the size of your matrix.

Archaea are a fairly new face on the scene, but they mean business. The Exchange somehow manages to be elegant, creative and useful all at the same time, and on more than one occasion gave me genuine pause for thought when I realised the amount of things I might be able to do with it. Whether your system is big or small, I guarantee it has a place for this lovely little design.

William Stokes £

£249

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www.archaea.co.uk

makes it possible to trigger short envelopes, useful for creating shortened sounds with a slightly rounded attack while retaining their harmonic value. The Lublin also has an Exponential input, meaning it can perform more conventional VCA duties.

As XAOC describe on their website, the Ping inputs use an analogue circuit to “model the dynamic behaviour of vactrols, allowing you to trigger a carefully crafted gain response that mimics the natural decay of plucked and struck acoustic

ER-Range

instruments”. The technical definition of a vactrol is an electronic component that combines a light-dependent resistor and an LED, and it is what allows the smooth plucked and struck sounds to be achieved.

Traditionally vactrols contain cadmium, which is banned in EU countries, hence the modelling.

The three settings of the switches — ‘Shortened’, ‘Natural’ and ‘Prolonged’ — allow a fine‑tuning of the ping response in designing your sound. The Shortened setting can provide an easy solution to creating percussive sounds. I have always enjoyed the possibility of shaping a sound with a VCA to create hi-hats or other percussion while patching.

Usually we think of the main function of a VCA as processing the audio output signal, but the Lublin is DC-coupled to process both CV and audio, which is a patching possibility that might not have been evident at first.

And while it creates exponential decay curves, it cannot exceed unity gain, hence there will not be any distortion coming from this module, unless the signal of the sound patched into it is already distorted. As we have found, some modules can be brutal and gritty without the built‑in unity gain control, and I like them like that, but it’s also nice to have one like the Lublin, which allows both deep and delicate sounds to be clean without distorting or peaking.

The sound through the Lublin can be soft and light for creating subtle percussive and atmospheric sounds, or it can be heavy for kicks and bass lines.

It is very versatile in the range of audio responses it can help to achieve.

VCAs are an important tool for all modular systems; by now, we all understand we should have several in our systems to creatively and effectively patch with. I have found, when VCAs have several output settings, they can be inspiring tools that can also bring new ideas to light. The choice between the Exponential and Ping inputs along with the switch settings gives a range of options in one module, all with an intentional and clean-sounding response.

This is a great module that is highly recommended for everyone who finds their rack in need of a new VCA, and as we all agree, the VCA component is one of the most important factors in building a custom synthesizer and patching it.

The Lublin, with its ability to maintain control signal while giving the user plenty to explore, is entirely consistent with the high quality we’ve come to expect from XAOC Devices. Dr Chelsea Bruno £

£149

W

www.xaocdevices.com

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ON TE ST MODULAR

FlatSix Seventh Summoner Eurorack Module

F

latSix captured my imagination last year with the Arp Of Darkness. I was so enraptured by the simple yet fascinating functionality that I made it my Sound On Sound pick of the year. FlatSix have released another module based on the same hardware but with a distinct firmware. It’s called the Seventh Summoner, and I was keen to see if it could hit the same high notes.

The unsettlingly named Seventh Summoner is a really simple sequencer, and as with the Arp Of Darkness, it does just enough to make itself fabulously useful. The 26HP, 1U module is quite striking, yet other than the button keyboard stretching for a single octave, there’s not much else going on.

You have a couple of buttons, a trimmer knob for Glide and time division and two patch

points — that’s it. The idea is that you punch in the notes in an SH-101 step-sequenced style, and the Seventh Summoner plays them back.

The programming requires a little bit of finger choreography. The Shift button on the right does a lot of heavy lifting. Hold and keep holding Shift, press Play (octave up), select one of the six sequence slots (white notes), and then press Rec (octave down), release Shift. The sequence is now cleared and ready for notes. You can enter up to 32 notes in a sequence or stop when you’ve had enough. Hold Shift and press Play to save it. The sequence will instantly start playing with reference to the pulse coming into the gate input. But hang on; is there no gate output? No, the Seventh Summoner only sequences notes, not gates, so you’ll have to use something else to fire envelopes or define step lengths if that’s what you want.

Using the keys and octave buttons, you can transpose, and there are a handful of playback modes you can access with the popular Shift

button. These include forwards, backwards, ping‑pong, random and Golden Ratio. Golden Ratio uses the Fibonacci sequence to calculate which step to go to next. The result is a seemingly endless variation on the programmed sequence that feels somehow more natural than pure randomisation.

Lastly, we have the Summoner mode, where you can chain up to 16 sequences.

You can specify transposition and playback mode for each sequence as you add them to the chain. It’s simple but quite powerful in its own way.

There are many things it lacks. An LED would be useful just so you know it’s clocked and working. The loss of the top C to form the Shift button is mildly annoying. The lack of gate output is perhaps inconvenient, but I actually like to do gating separately

Modular Profile: Maciek Bartkowiak

X

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£

£208

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www.flatsixmodular.com

blocks? That view changed completely when I met XAOC. I realised that modular is not about following the book on synthesis, but about rewriting it.

AOC Devices (pronounced ‘chaos devices’ if you happen to be reading this to your kids) are without a doubt one of Eurorack’s most respected developers, with a host of functional and beautiful modules to their name. One of the central figures behind that success is Maciek Bartkowiak, a veritable heavyweight in electronic instrument design and a charming fellow to boot.

On his entry into modular I first encountered synthesizers through a TV show at the age of 13, and that moment sparked a lifelong obsession. Even before formally studying electronics, I began collecting magazine articles about them. At 18, I built my first polyphonic synth, and by 21 I had designed a hybrid digital/analogue sampler. I went on to study electronics and

and it encourages me to find patterns elsewhere that are not tied to the same workflow.

I’d like to see a real-time mode where you could add notes as the sequence loops, and a play‑through mode so you could override the sequence.

For me, there’s just enough to make the Seventh Summoner a superbly valuable use of 1U space. Eurorack sequencers are often big, chunky, multi-channel affairs that overflow with features.

Whereas here, you have a bunch of sequences snuggled away in a 1U row that you can pull from your back pocket any time they’re needed. Once you’ve committed the finger functions to memory, it becomes a terribly useful module.

Robin Vincent

telecommunications and earned a PhD in Signal Processing.

Over more than two decades in academia, I contributed to the MPEG Audio standard and now hold several DSP-related patents. All this provided a solid background to what I do today. For a long time, modular synthesizers struck me as outdated and unnecessary — why break down subtractive synthesis into separate, cumbersome

On his circuit design philosophy Above all, I value innovation and have little interest in copying classic designs. I’m proud of the many small inventions embedded in our circuits, even if they remain invisible to the user. My colleagues often set ambitious standards for new modules, and meeting them can be a significant challenge. It usually requires multiple prototype iterations and careful refinement, but the result is always deeply rewarding.

I enjoy working with upper-shelf modern components — even when they might seem excessive — because they bring a sense of elegance, long-term reliability, and subtle sonic advantages that some users genuinely appreciate. That said, I also occasionally draw on simple, old-school circuit techniques that introduce

intentional imperfections — essential in shaping what has become the XAOC signature sound.

On the story of XAOC Devices Designing and building synthesizers for a living had been my dream since the early days, but for a long time there were no synthesizer companies in Poland. That changed in 2012 when Marcin and Tomek founded one with the release of the Moskwa sequencer. I first heard about them through our mutual friend Michał and thought, how many of those could they possibly sell — 50? Then I discovered the rapidly growing Eurorack scene and realised the scale of what was happening.

We eventually met, and Marcin invited me to join. At first, I was hesitant to leave my academic career, but once I took the leap, everything accelerated. From the start, the focus was on ambitious projects. Praga, our mixer, took five years to develop — but it was worth every moment. Odessa was another demanding endeavour, developed in collaboration with my colleague Adam who is a specialist in FPGA design.

XAOC began as a team of three. We now have 12 people on board — and I believe that’s the right size.

On the XAOC Lublin In the Lublin VCA module, the Ping input is often seen as its standout feature — but to me, it’s just the cherry on top. The project was initially motivated by the success of Tallin, our first VCA, which was intentionally designed to be imperfect, nonlinear, and sonically coloured. Lublin was conceived as a counterpoint to that: a clean, transparent VCA built around a 2180-type Blackmer gain cell.

Still, basic VCA functionality can feel somewhat uninspiring, even though it’s essential in every system. The pinging feature gave Lublin a unique character. I modelled its nonlinear time response using a network of diodes, based on how real vactrols behave in various classic modules. The result captures that initial bell-like curve, a transition phase, and a long, natural-sounding tail reminiscent of reverb. It was a first step into vactrol modelling, a kind of trial run, and I hope it will lead to a fully featured LPG-style module in the future.

On the future of modular synthesis I believe that modular synthesis as a method has not yet reached its full potential — but I worry that its popularity may wane before it gets there. As the format becomes more mainstream, it becomes increasingly influenced by short-term trends and fleeting fads, which shape demands for new modules. It’s easy to foresee a rise in digital modules that encapsulate specific combinations of building blocks, offering instant access to compelling sounds with minimal effort. Machine learning will gradually make its way into modular synthesis as well, enabling highly complex, ‘intelligent’ behaviours that may further reduce the need for patching creativity.

Personally, I remain passionate about analogue design and still hope to see the development of more advanced analogue ICs from manufacturers like Alfa RPAR and SSI. Such components could enable the creation of sophisticated yet affordable devices. A filter bank on a chip, an oscillator array, or a precision arbitrary function generator would be true game changers. William Stokes

www.soundonsound.com / July 2025

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INSIDE TRACK

The members of Ghost are, notoriously, anonymous.

Max Grahn & Dan Malsch: Producing & Mixing Ghost Swedish hard rock band Ghost have topped charts worldwide — with an album that was mixed twice!

PAUL TINGEN

“O

ne of the most interesting things of my career happened in November last year,” reports Dan Malsch. “Andy Wallace and I had mixed the entire Ghost album, Skeletá, over three weeks in August 2024 at my Soundmine Recording Studio in Pennsylvania, on my SSL 4000 desk.

A few weeks later I received a call asking whether I could come to Stockholm and touch up all the mixes. This turned out to be a huge learning curve, because I ended up in a studio that I had never worked in before, with monitoring and a console I’d never used before, so I could not do recalls.

“It was a tall order for a mix engineer.

Over two and a half weeks I was at the studio, IMRSV, with 16-hour daily marathon sessions and no breaks. It was a lot of pressure, and very stressful.

I mean, I didn’t want to be the one

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messing up a bunch of Ghost mixes!

The entire Ghost team wanted the record to be great, and there was no shortchanging anything. They could have gone with the Pennsylvania mixes.

They were great mixes. But they still wanted to go that extra mile, trying to get the mixes to another level.”

How far the commitment to excellence of the Ghost team and Malsch’s upgraded mixes helped steer Skeletá to major success is hard to say, but the fact is that Ghost’s sixth studio album is the theatrical Swedish heavy metal band’s most successful to date, in terms of chart positions. It was the band’s first album to top the Billboard 200 in the US, it went to number one in many other countries as well, and it reached number two in the UK.

Ghost Story Naturally, a commitment to excellence starts right at the beginning. In the case of Skeletá this involved band leader

Tobias Forge writing two of the album’s 10 songs alone, another two songs with Salem Al Fakir and Vincent Pontare, and the remaining six with Max Grahn. In all cases, these Swedish songwriters also co-produced the songs they co-wrote, with the exception of the two tracks written by Forge alone, which were co-produced by him with Pontare.

Max Grahn belongs to the colossally successful Swedish pop songwriting scene that started with ABBA and spawned Max Martin. Grahn has co-written songs for Conan Gray, Kim Petras and Lewis Capaldi, and performed on music by Justin Bieber (drums), the Weeknd (guitar), and more. Grahn is signed to Max Martin’s MXM Music, and works from the publishing company’s studio complexes in Sweden and the US. “Max [Martin] is a mentor and a colleague, and we have written a few songs together,” explains Grahn. “Before that, I played in 30-something bands.

Photo: Nikola Stankovic

One of them was a hardcore band, and the singer was Karl Schuster, who left to sign with Max and write pop music under the name Shellback. He quickly had big hits with Pink, Taylor Swift, Justin Timberlake and so on. But I thought, and still think, that being in a band is the coolest thing you can do.

I would record the bands I was in, and gained some merit from being on the computer. But it turned out that being in a band with multiple grown men, creating, rehearsing and aligning your goals and sacrifices is harder than I thought.

So I started going my own path in production, writing pop music, which I’ve always loved. Turns out writing actually good pop is harder than I thought.”

Conducting Traffic

House Of Hits Max Grahn first connected with Tobias Forge in 2021, and the two co-wrote the lead singles for Ghost’s fifth studio album Impera (2022): ‘Hunter’s Moon’ and ‘Call Me Little Sunshine’. The latter was nominated for a Best Metal Performance Grammy.

“Tobias and I hit it off right away.

We went into the studio, sat down with a guitar each, started playing and writing, and found this really good synergy.

We fed off each other’s ideas and kept building stuff, and wrote those two songs.

At some point in 2024, we decided to go back in the studio again to see what might happen. We kept coming out of the studio with new songs and material.

I actually don’t think we’ve written anything that hasn’t been released.

We have had no misses yet, only ‘hits’.”

Tobias and Grahn conducted their songwriting sessions for Skeletá at MXM’s studio complex, designed by Jochen Veith, who was interviewed in SOS May 2022. “The whole house is filled with a group of 10 people or so who are signed to MXM, and each of us has their own studio, plus there’s a big live room. Next door to me is Elvira [Anderfjard] who’s crushing it with the new Addison Rae stuff now, and she’s done Taylor Swift, Ed Sheeran and so on.

Next door is Oscar Görres, who’s done Ariane Grande, Troye Silvan and so on.

Then it’s LIOHN, Holter, Mattman & Robin and more. I can go out of my door and hear this cutting-edge pop music from the best people in the world. I’m fantastically spoiled.

“Gear-wise, I have ATC SCM45A Pro monitors, built into the wall, with a pair

box. My accumulated outboard so far is an Alesis Midiverb II, which I heard Aphex Twin uses, an Alesis 3630 compressor, an Evol Audio Fucifier Bloodmaster Edition Distortion, and I got a Roland GP‑8 guitar effects processor, because Kee Marcello, the guitarist of Europe, used that in the ’80s, and we wanted that kind of guitar tone for the Ghost album. We ended up not using the GP‑8, but I’m still experimenting with it. I also have some synths and cool guitars.”

Max Grahn co-wrote and co-produced six of the tracks on Skeletá.

of 1x12 custom ATC subs. I have a UA Apollo x8p, and then the magic part: a Xilica XP 4080 audio system processor, which coordinates the speakers and allows me to tune the room to my liking.

And I use Yamaha NS10s with a sub as alt listening. My setup is essentially in the

For the writing sessions he did with Tobias Forge, Grahn says: “We were mostly in the big live room, where we had drum kits and guitars and stuff hooked up. For us, playing instruments is the cornerstone of every song we write. We both play drums, guitar, bass and synths. It’s very important that things feel fun and, weirdly enough, both intuitive and unintuitive to play.

Writing should not only be fun, but also be the right amount of challenging.

“We both grew up in rehearsal spaces, where we learned to write songs. It’s a way of writing that automatically creates an identity, because playing is so physical and personal.

During writing we didn’t have engineers in the room. I have kindergarten level knowledge of how stuff works in a live studio. I don’t know my way around a patchbay, but I can press Record. Once in the box, in Pro Tools, I’m at home.

Dan Malsch at Stockholm’s IMRSV studios, where Skeletá was remixed.

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INSIDE TRACK

MAX GRAHN & DAN MALSCH • GHOST

“While writing we used a Shure SM7 going into a Fairman and then an 1176 for vocals. We’d play electric and acoustic guitars, the former through a Rockman, made by Tom Scholz from Boston.

They’re like a portable cassette player that you can clip to your belt and plug into your computer. With headphones you sound like you’re a rock star on a rock stage.

“All of the songs that we’ve done are 50/50, him and me. When we hammer out songs and arrangements, we are both like little conductors. We like to have a full overview of the full ‘score’ in our heads.

We always come back to the guitars when writing, but we are all over the place with instruments, creating the basic arrangements. Once we had written and mapped out the songs in enough detail, it was time to take them into the big studio for proper tracking, which was Atlantis Studios in Stockholm.”

Pin Sharp Famously, the members of Ghost are anonymous, calling themselves ‘Nameless Ghouls’. Even Forge’s identity was not revealed until 2017. As a singer on Skeletá, his name is Papa V Perpueta, while he’s credited as songwriter ‘A Ghoul Writer’ and producer Gene Walker. With band proceedings shrouded in secrecy, Grahn has to be discreet in how he describes the tracking process.

“Let’s just say that all music appeared magically in Atlantis Studios! But I can tell you some technical details. We started by building a drum kit, with cymbal shootouts and stuff like that. At the same time we built guitar and bass rooms with amp stacks. Once we knew what the drums were going to be, we recorded them, while still figuring out the guitar tones, which was a whole chapter in itself.

Atlantis Studios is legendary, for example many ABBA songs were recorded there.

The piano used for ‘Dancing Queen’ is in the middle of Studio A, and we used it on Impera and on the new album. It’s a magical place.

“The guitar sounds are never just one sound. For this album we had a few combinations of sounds. For the main guitars on most of the album we used Vincent’s Rockman amp, modified by his guitar guy, Göran Elmqvist. That went into a Roland Jazz Chorus amp and

Many of the guitar sounds on Skeletá were tracked at Atlantis Studios using a combination of the Roland JC‑120 and Mesa Boogie IIC+ amplifiers, miked with Sennheiser MD421s.

a Marshall Plexi with a Hiwatt 4x10 guitar cabinet. For a more distorted sound, we also had a Mesa Boogie IIC+, the same one that James Hetfield of Metallica used in the ’80s. The blend is pretty special and gives a weird guitar sound that I’m super happy with. For the solos we used a Boss OD-1 from between 1977-1979, which is important because it had a 14-pin chip, which was replaced by an 8-pin chip in 1979. They’re all played on Fredrik Åkesson’s custom PRS guitar, sometimes by him, sometimes by me.

one of the songs in particular, ‘Cenotaph’, we had an emergency situation where we liked the song but not the shape the song was in. It ended up in a different bpm, different time signature, and in a different key. And now it’s the best song on the album.”

Legends Of Atlantis During the tracking sessions at Atlantis, four engineers were in action, recalls Grahn. “We were blessed to have Martin Sandmark Ericksson, who is Tobias’ guy; Mark Rankin, who’s a brilliant person as well as a Po Tools ninja; and two Atlantis house engineers, Niclas Lindström and Calle Gustavsson. It was essential to have these technical people there, because as I mentioned, I do not know my way around a patchbay! But with regards to the microphones, I do recall that we used two Telefunken ELA M 251s for the overheads of the drums. Quite luxurious!

“For the vocals we’ve worked out a certain way to go about it. ‘Papa’ [Forge] has like five frequency spikes in his voice that are impossible to turn down. It’s

Dan Malsch: “One interesting thing about Ghost is that they have a subkick that changes with the chords, around 40, 50 Hz. It’s quite unique for a hard rock metal band.”

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“Although we had mapped out the songs, they shape-shifted a bit once we got into the real studio and got the final takes. This allowed us to hear new possibilities, like using eight toms, or there’d be a cool new guitar lick. With

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INSIDE TRACK

MAX GRAHN & DAN MALSCH • GHOST

outside in the historical live room of Atlantis. And we printed all these separately. We used an EMT 251 for many things. For instance, on a percussive organ that’s doubling the bass guitar in the verses of ‘Lachryma’. ‘Kih Kah Kih Kah.’ It sounds like a giant clock ticking towards your imminent death.

“Engineer Mark Rankin and I kept an overview of all proceedings at Atlantis, until we started feeling like we had an album. Also, managers and label people were hammering in some deadline dates. This meant that we needed to get started with final mixing. We finished a few songs so that we could start rolling them to Dan and Andy.

In the beginning Mark and I did rough mixes together, but then I rough mixed and mix-prepped the six songs I had co-produced in my own studio.

“Vincent [Pontare] would do the same for the songs he co-produced, ‘Peacefield’ and ‘Satanized’, and Martin Erickson Sandmark and Tobias did the rough mixes for the two remaining songs, ‘Massilia Amori’ and ‘Marks Of The Evil One’. I wanted to listen to what we had in my own room, so I knew where everything’s at. Like the relation between kick, snare, vocal and making sure the wetness of the all reverbs sits right and so on. I’m really used to my ATCs in this ridiculously acoustically well-designed room by Jochen [Veith], who is a genius.”

Perfecting Perfection

A Neumann KU100 binaural ‘dummy head’ was used at Atlantis to track sources such as choir and glockenspiel.

almost like an autonomous polyphonic overtone chanting. There is no other singer that sounds like him. And the SM7: if Michael Jackson can record the whole Thriller album on an SM7, we should be able to make do with that. It just delivers.

Also, sometimes the first demo take is irreplaceable, so we keep that. I always go for character over perfection.

“The bass sound is a combination of five components. We had a SansAmp pedal, and an 8x10 SVT rig. We also sent the bass through a Tandberg

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Båndopptaker 2, a small Norwegian tape recorder from around 1954 with a built-in speaker. The speaker sounds beautifully broken and kind of lo-fi, and we recorded it with a mic. And we had a mysterious chorus pedal. Also, Atlantis has a renowned echo chamber that we used on the bass.

“In general during the sessions we used a healthy mixture of synthetic reverbs and actual room mics. We opened the doors a little bit to the guitar rooms, and then put mics

On the other side of the Atlantic, Wallace and Malsch received the rough mixes as Pro Tools sessions. “Andy and I spent about three weeks mixing Skeletá during August 2024,” says the latter. “I started out as Andy’s assistant six years ago, and our process has grown as we got to know each other. It’s really cool and works really well. I was mix engineer on the previous Ghost album, Impera, which Andy and I also mixed at my studio, Soundmine. With Skeletá, Andy and I began with our current routine, which starts with me trying to better the demo, and Andy then coming in and doing his thing. We’re a kind of tag team.

“In this case, I’d first of all make sure I had every plug-in. My Pro Tools sessions needed to sound exactly like what I was sent. I can’t have missing plug-ins. There’s no room for interpretation at that point.

Max and I worked directly with each other, sending tracks and notes back and forth, and checking every plug-in.

I think in the end I only had to buy a Baby Audio plug-in. Each session had lots of tracks, well over 100, with pretty much every instrument and mic separated and many effects printed.

“After I’d made sure that what I had matched what they had, I spent probably five or six hours trying to beat the rough mix. This was not easy. The production and rough mixes of the Ghost tracks that we received were great — in my opinion, probably good enough to release. Andy and I would look at each other, like, ‘How are we going to make it sound better than this?’”

“That’s way too flattering!” laughs Max Grahn. “Of course, we take it as far as we can. But what happens when we get the mixes back from Andy and you, is that there’s more 3D. Even though the mixes are stereo, they sound 3D. You add depth and extra flair that we would not be able to achieve alone.

Also, we’re a solid gang of control freaks, so we were not looking for surprises

Skeletá is Ghost’s sixth album and their most successful to date.

from your mixes, even as there were a few. We simply wanted you to perfect the rough mixes, which you did.”

Desk Job Wallace and Malsch’s process still centres on a hardware mixing console. “I have a 56-channel SSL 4040 EG+ with Total

Recall from 1986,” elaborates Malsch.

“It came from Backstage in Nashville.

I have a great tech, so my SSL is very well maintained. Andy uses every single knob — it needs to work flawlessly! My studio also has tons of outboard, too much probably, and I have an Atmos room. My desk is old, and has character, and many bands and artists still request that we mix on the 4k, like Avenged Sevenfold, whose recent album [Life Is But a Dream...] we mixed here.

“I like mixing on a desk because of the tactileness, the fact that you’re working with your hands and touching buttons. You’re not looking at a screen, it’s an intuitive way of working. It’s about the process. We use the EQ a lot on the console, and I can get basic levels and EQ on a mix in 45 minutes to an hour.

It’s a lot quicker on a console. We’re also driving the mix into the 4k master bus, which has a particular sound. You can hit it pretty hard, though not too hard, or it gets squishy. It’s about that overall sound

Bass guitar was tracked using no fewer than five separate rigs.

This photo shows one of them.

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INSIDE TRACK

MAX GRAHN & DAN MALSCH: GHOST

For the second mix of Skeletá, Dan Malsch worked on an SSL Duality rather than his own 4k.

that the SSL brings, just like a Neve will bring a certain sound, and an API will bring a certain sound.

“The way we mix is that I first stem the Pro Tools session out to the SSL. There are a lot of tracks, and I try to get as many tracks separately to the console as I can: every guitar mic, every backing vocal, every printed effect. We want maximum control. I don’t want to miss anything.

I don’t want to miss a guitar that has a specific reverb on it. When there are more Pro Tools tracks than desk channels, I create groups in Pro Tools. In fact, I have a mix template, and I usually have two kick channels, inside and out, and one for a low-frequency kick. So there may be three different kicks in a mix. Heavy rhythm guitars are grouped and we take them to a stereo channel on the console, or two mono channels. Overdubbed guitars, any kind of jangly guitars or ethereal guitars, we take to another two channels of the console. Whatever effects

are on the guitars, like rooms, digital reverbs or delays, go to another two channels. And so on.”

In The Clear Malsch stresses repeatedly that most of Wallace’s and his mix process “takes place on the console. We want to have control. We do a lot of EQ’ing,

mixes to the main mixes is more space and a depth and a width.

“Clarity is a huge thing for me. I want to have as much bottom end and sub as I can, but if I hear any mud in a mix, I don’t enjoy it. One interesting thing about Ghost is that they have a subkick that changes with the chords, around 40, 50 Hz. It’s quite unique for a hard rock metal band. I think that comes from the pop world, and it really helps with the bottom end.

I’ve been called a clinical mixer because I need to hear every instrument cleanly.

If it’s there, I think you should hear it. Sometimes you’ll have to shave off a little more than you might want to, because there might be three or four different guitar parts or different key parts, and to make sure they are all heard, you have to take out some 300Hz, 400Hz or 500Hz, which are problem frequencies in a lot of music that I work with.

“I do all this on the desk. The only thing we’d really change in Pro Tools

Max Grahn: “For the solos we used a Boss OD-1 from between 1977-1979, which is important because it had a 14-pin chip, which was replaced by an 8-pin chip in 1979.”

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compression, level riding and panning.

Both Andy and I are very detailed tweakers, looking for every space and separation that we can get, because we’re trying to open up the mix as much as we can. We use the SSL desk EQ to filter certain frequencies out. It’s very delicate because you don’t want to upset what’s there. They are delicate moves.

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INSIDE TRACK

MAX GRAHN & DAN MALSCH • GHOST

is if a track is pushing a little too hard into the console or not pushing enough into the console. For instance, if a kick drum is hitting the input of the SSL too hard, there’ll be weird compression and distortion, so I’ll have to lower the kick drum 4 or 6 dB in Pro Tools and then raise it another 4-6 dB on the console.

Actually a lot of what I do is bringing all the clips in a session down by perhaps 5-6 dB because they’ll distort the console. It’s tricky because sometimes you all of a sudden lose some of the energy, and then you have to bring that back with the console. But if you hit any console too hard going in, the mix just collapses. There’s an art to bringing tracks out of a DAW into a console.

Gain structure is huge. It’s the same thing with balancing levels. If you’re volume riding in Pro Tools before going into the console, you’re going to distort the console. We do the bigger rides on the console, and tiny little volume tweaks in Pro Tools after that.”

In The Balance Once Malsch has taken the mix as far as he can, Wallace comes in. “He’ll say, ‘How’s everything going?’ and then he’ll sit down at the desk. Andy is a balancing genius. He’s a tweaker, tweaker, tweaker. He is very, very detailed. He’ll listen quite loud for a while. And then he’ll listen quite low for another few hours. That’s our process.

I’ll get the song to where I could almost sign off on it, and then he still finds something to improve.

“While we mix, the master bus compressor on the 4k is always in, and it’s almost always set the same: auto release on, the slowest attack possible, and a threshold of no more than 3dB. On the 4k even the slowest attack is fast. We also have outboard on the master bus, which is the DW Fearn VT7, a big, clunky, great stereo tube compressor. We’re not compressing with it, just running the signal through the tubes. And then a Bettermaker Stereo Passive Equalizer, which is a solid-state version of the Pultec. The settings on the DW Fearn never change, but we’ll be boosting on the Bettermaker. It has a really broad boost. So if I’m boosting at 30Hz, it’s boosting a good amount at 60 also. And then we often boost top‑end air, like 16kHz. It’s really broad strokes.

“I can’t send a mix for review to clients at -12 LUFS because every other mix

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they’re listening to is, unfortunately, -6, -7 LUFS. I’ve seen mixes at -5 and even less, which is crazy. It’s tough because if I send out a mix at -10 with some dynamics, clients are probably going to think there’s something wrong with the mix, because it’s 4dB quieter. So before sending mixes off I’ll have to use a limiter in the box, often the FabFilter Pro‑L 2. Especially if it’s a loud rock mix, I’ll mix into the plug-in limiter, because I want to know what it’s doing. The Ghost record is probably -6.5 LUFS. Andy and I work almost exclusively with Ted Jansen from Sterling. When I send him the mix, I’ll take our limiter off.”

Epilogue As mentioned at the beginning of the article, Wallace and Malsch spent three weeks mixing Skeletá in Pennsylvania with their regular process. There were the usual recalls and mix adjustments, and once they delivered the final mixes at the end of August 2024, all parties signed off on them, and the two mixers assumed that it was a job well done and completed. So the request in late October to re-adjust the mixes came as a big surprise to Malsch, and led to one of the most unusual and challenging mix experiences of his career.

“I got a call from Tobias’ manager, asking if I could come over to Stockholm in less than a week. I was told that there were a couple of sonic things that Tobias, after living with the mixes for a while, wanted to change. Also, he had not been able to be there for the mixes in Pennsylvania, and wanted to be present for the mix reviews.

“Andy could not make it to Stockholm, so I went alone. The first thing we did was try to find a studio in Stockholm with an SSL 4k, so I could bring the floppy disks and do recalls. We could not find a studio with a 4k that was available, and booked IMRSV Studio 1, where they have the newer SSL Duality Pro Station Delta. It has 48 split channels, so I did not have to regroup things in the Pro Tools sessions.

“In hindsight, I’m glad that we didn’t find a 4k, because the Duality suited the record a little better. It sounds more open and cleaner than the 4k. As soon as you put a session on a 4k, it automatically does you some favours, especially in the rock world, as it adds some dirt and glue.

You can’t get around that. The Duality doesn’t have that. The Duality is a little more open and much cleaner. So you have to work a little harder. But if I had

Dan Malsch’s Soundmine Studio, where he and Andy Wallace undertook the first mix of Skeletá.

to choose between the two at this point, I would mix on a Duality over a 4k. It’s also quicker and more efficient to use, and easier to maintain.

“It was the first time I used a Duality.

So I was unable to recall our existing mixes, on a different desk in a different studio with monitors I did not know.

It was a huge challenge and quite stressful. I’ve been mixing on ADAM monitors with ribbon tweeters for the last 15 years. ADAMs have a very distinct sound. You love them or you hate them. At IMRSV I was mixing on Genelec The One monitors, which are completely different. And the room took some getting used to. Initially I did not understand the monitoring at all. Basma Jabbar, the assistant at IMRSV, was

a lifesaver for me. She was exceptional.

She knew the Duality inside out, and was positive and kept my energy up when I was weighed down by the 16-hour daily marathon sessions.

“My first target at IMRSV was to get back as close as possible to the existing mixes. I had some memories of what we had done, and for the rest I had to use my ears! After that, with Tobias in the room, we were able to take the mixes further sonically. Tobias wanted specific changes.

Snare drum changes were a big one. We swapped around some keyboard parts, and changed the balances of certain solos and keyboard parts. Adjusting the panning, particularly of the guitars, was one of the bigger changes. Also, I knew what the flaws were in the original

mixes, so I did my best to iron these out.

I always second-guess my own work! For example, the original mixes may have been slightly over-compressed, and I could make the new mixes punchier.

And the EQ on the Duality is a little sharper and cleaner, which helped.

“Another problem to solve was the master bus chain. The Duality bus compressor is pretty much the same as that of the 4k. Understandably, they kept that signature SSL sound. But when I tried to get a DW Fearn, which is part of the signature sound of Andy and I, I could not get one. Not having that was a big deal. I was sweating quite a bit, and in the end substituted the Fearn with a UAD Pultec EQ, because it has a tube-like sound. It’s cleaner and

wider-sounding. That was the ‘analogue’ piece for me, and the only thing on the master bus. So no Bettermaker. And none of the mixes you hear on the album were touched by the 4k.

“I sent every mix to Andy, management, the producers, and everyone would give their feedback.

Andy gave detailed comments on the mixes, so he was involved. I was supposed to be in Stockholm for 10 days, but it ended up being two and a half weeks, without any breaks. It was gruelling! I have to hand it to Tobias and the producers that they insisted on going that extra mile. I’m very satisfied with the end result. And obviously, the album is doing really well. I think it’s a great record!”

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Analogue Solutions AS500-Seq

Analogue Step Sequencer Analogue Solutions’ AS500-Seq is defiantly old-school, eschewing quantisation, MIDI and encoders...

WILLIAM STOKES

“Y

ou spoke, we listened,” declares the promotional video for Analogue Solutions’ AS500-Seq 64-step sequencer. It might look big and bulky — an impressive 38 by 29 centimetres — but by Analogue Solutions’ standards it’s pint-sized.

Perhaps I say this because the AS500-Seq is in fact lifted straight out of a corner of the staggeringly huge EMS Synthi 100-inspired Colossus; although it would actually be more accurate to say it skips a generation, since the Colossus’

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sequencer circuit, so far as I can tell, is itself a faithful reimagining of the lauded Megacity ‘teardrop’ step sequencer.

I’ll stop, before we end up in Old Testament-length lineages; the Megacity was reviewed in SOS by Paul Nagle back in 2017, who found it “an endless and evolving world of polyrhythms.”

A 64-step analogue sequencing titan that could be split into two 32-step sequencers if desired, each step offered a choice of two different gate outputs, X or Y, alongside some very inventive and creative pattern configurations. If you filed a cutting of that review, you won’t

be blown away by a raft of new features here; indeed, the latter does not offer one or two things the Megacity did, such as MIDI. In fact I wouldn’t dissuade you from reading Paul’s great piece to get something of a different angle on the AS500-Seq, so similar is its functionality to that of the Megacity. In any case, since user demand was the apparent driving force behind the AS500-Seq’s release, I suppose it’s my job to ask whether or not that demand has been satisfied.

Along with the AS500-Seq itself, I was grateful to have been sent a companion synth, the similarly-proportioned

Analogue Solutions AS500-Seq £1198 pros • A proper piece of old-school analogue goodness!

• Unique and original sequencing workflow.

• Solidly built.

cons • Some might find it too big.

• Hefty price tag for a standalone sequencer.

summary A statement piece for your synth setup, the AS500-Seq is a slice of classic electronic music-making brought elegantly into the present day — with a price tag to match.

Analogue Solutions Ample. The Ample I have also been keen to try for some time, and while this review isn’t principally concerned with it, it’ll be our case study if nothing else. And what a gorgeous thing it is: a 16-step sequencer of its own with a host of syncopation options and even a ‘snare drum’ noise output, a row of CV touch pads, three oscillators, two envelopes, a ring modulator and an elegant little pin matrix, it’s ample indeed, as futuristic-looking as it is retro.

Simply marvellous.

Simplicity Itself If you’re at all familiar with Analogue Solutions then you’ll already know what to expect from the AS500-Seq: this is old-school synthesis, in almost every sense. A 64-step all-analogue machine, it has no MIDI, no quantisation of any kind (again, unlike the Megacity) and certainly no USB. Its encoder-less, preset-less workflow consists of knobs, toggle switches and the odd button, and that’s more or less it. And here I was thinking my Doepfer Dark Time was a nod to old-school sequencing.

Like its predecessor, the AS500-Seq centres on a vertical ‘teardrop’ layout whose transport moves from top to bottom, each step’s knob lighting up in sequence column-to-column as if ‘dripping’ downward. It is very pretty.

There’s no Direction control here: instead, sequence variation is achieved by flicking various step switches to Jump, meaning the sequence will hop from one column to the next at that point. The topmost steps in each column have a Reset option instead of Jump for variable sequence

lengths, and variously combining these two can make for some very interesting patterns. You can set the second lot of 32 steps to operate at divisions of the main clock, and you can also set it to act as a ‘fill’ for a sequence ordinarily operating across steps 1-32, either with a button or trigger signal. Combine all of this along with the X and Y gate options and you’ve got some considerable and unique creative scope.

Fair to say, with so many steps available and with no scale quantising, you might find yourself spending a good amount of time tuning and preparing your sequence, but once a selection of values are in place and the clock is running, it’s essentially for the step switches to take over with their selectable gate outs and Jump settings. This is a lot of fun.

In one simple but effective patch I sent one side of the AS500-Seq to one of the Ample’s oscillators and the other side (with a sequence of different length and division) to another. I then used the AS500-Seq’s clock to sync the Ample’s

been taken. That, in my experience, is generally worth paying for.

As for the second, this is something Analogue Solutions founder Tom Carpenter himself joked about with me at SynthFest, wryly quipping that people must surely not have any need for such a physical beast of a sequencer, let alone one with basically none of the mod cons we’ve come to rely on. It’s one thing to remortgage your house and jump feet-first into the land of Colossus — you know what you’re getting yourself into there. But I’d say that adding this one ‘Colossal’ module into an existing setup requires a different mindset entirely.

My argument for its intrinsic value is fairly simple. Having but one genuine piece of no-stabilisers analogue gear is not only a rarity these days but a highly valuable piece of creative stimulation; on top of this, it’s even more valuable that it isn’t a sonic component like an oscillator or a filter — any of which, if we’re honest, can basically be whipped into shape with a modern control signal. It’s a controller

“Having but one genuine piece of no-stabilisers analogue gear is not only a rarity these days but a highly valuable piece of creative stimulation.” own onboard sequencer and trigger its filter and VCA envelopes (while it certainly does not disappoint overall, the Ample curiously doesn’t have a way to trigger either of its envelopes externally).

In any case, I then patched the Ample’s Interval Generator CV touch pads to the filter cutoff with the pin matrix so I could performatively offset its response to the envelope. I dialled in a little portamento for the AS500-Seq’s second side (there’s a Glide knob for each), and in no time I was composing some rich and dynamic sounds.

Back To Basics Two understandable questions might arise at this point: firstly, why is it so expensive? And secondly, why would I want that in my setup? To address the first: not only do Analogue Solutions operate, from what I can tell, with fairly short runs, the components in their designs are at a premium: durable, sealed potentiometers, a steel and aluminium enclosure, and so forth. Where a manufacturing shortcut has been available, nine times out of 10 it has not

that requires direct interaction to work, and occasionally a little bit of calculation.

Even writing this review, using the AS500-Seq felt like reading a physical book after, ahem, staring at a computer screen all day. Or recording to tape without the visual feedback of a DAW. Its physical size meant I had to move my body to interact with the machine, the steps’ unquantised values meant that improvising on the knobs created as many weird microtonal intervals as it did more musical ones.

As for the question of whether or not the user demand for a rebooted Megacity has been satisfied — if you are a MIDI devotee then I daresay you’ll be disappointed with the AS500-Seq. Retaining the signature Megacity workflow with a more compact form factor, however, along with more tactile and durable light-up step knobs and a far more ergonomic overall shape, it would be remiss not to call it an enormous success. £ £1198.80 including VAT. W www.analoguesolutions.com

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Sample Library

Soundpaint Solo Woodwinds & Brass Soundpaint’s affordable wind instruments contain some hidden treasures.

DAV E ST E WA RT

A

nyone engaged in contemporary virtual orchestration will likely be familiar with the name Troels Folmann. Since 2009 this voluble Danish composer has created hundreds of sample libraries, initially under the Tonehammer banner in conjunction with sound designer Mike Peaslee, then in creative partnership with producer Tawnia Knox, who co-founded the snappily-named 8Dio Productions with Folmann in 2011.

Based in West Hollywood, California, 8Dio’s extensive catalogue features epic choirs, cinematic percussion and a sprawling range of orchestral sample products that run the gamut from a solo oboe to the 240-piece Majestica mega-orchestra. The Legion series’ 66-player bass, cello, trombone and tuba ensembles are similarly gargantuan, while the musically detailed Century Strings, Century Brass and Adagio string series feature more traditional ensemble sizes.

In October 2021 Folmann and Knox launched Soundpaint, a free, easy‑to‑use

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sample player jam-packed with features (see box for more info). Accompanying the new engine was a free deep-sampled 1928 vintage Steinway grand piano and an intriguing assortment of high‑quality new instruments. The remarkable thing about these products was their price — in a world where sample developers routinely charge hundreds of pounds for their wares, many of them cost $20 or less.

After nearly four years of development, Soundpaint’s catalogue has grown into an Aladdin’s cave of affordable instruments that promise many hours of fun for the acquisitive sample user. I found much to admire in this astonishingly diverse collection, so to get started I’ll shine the spotlight on Soundpaint’s solo woodwinds and brass.

Solo Woodwinds: Flutes Soundpaint’s seven solo woodwinds made their debut some years ago as the rather expensive 8Dio Claire Woodwinds Kontakt library. The collection has now been re-formatted for Soundpaint and can be bought as single instruments at decidedly affordable prices. Recorded from close, Decca Tree and wide microphone positions in the amazing concert hall of the Cultural and Congress Centre, Zlín,

Czech Republic, these woodwinds feature some exceptional specimens.

The flute family is represented by the 2004 Flute Platinus, 2001 Piccolo Shire and 1987 Alto Flute Savana. An unnamed flautist sets the tone with some beautifully expressive performances — her platinum instrument’s sweet-sounding high register is perfect for romantic themes and dramatic swirling note flurries. The versatile piccolo can also tug the heartstrings with romantic melodies, evoke visions of a Hobbit-ridden English countryside with rustic melodies, or burst your eardrums with piercing fortissimo shrieks. All good stuff, but if you want one flute to rule them all, the alto flute’s sultry, breathy mellow tone, superbly consistent timbre and infinitely stretched range can sound like a complete flute family when played chordally.

Solo Woodwinds: Reeds In the reed corner we have the 1967 Clarinet Paris, 1982 Oboe Amore, 1995 English Horn Lorium and 2010 Bassoon Firebird. Played with no vibrato in the orchestral style, this Bb clarinet is unsuitable for hot Dixieland jazz but will function perfectly in a traditional score. I like the instrument’s effortless smooth deliveries and liquid, rounded tone, the only small

flaw being the noticeably unclean attack of its mf sustains’ E5-F5 register.

Formerly known as Claire Oboe Virtuoso, the Oboe Amore is not an actual oboe d’amore, but has a strikingly bright, plaintive and distinctive sound of its own — great for underscoring cinematic love scenes and panoramic nature shots, and also effective for incisive stabs. The bassoon is a tour de force: its lyrical vibrato legatos render soulful melodies with uncanny realism, and it excels at supportive bass lines, blasting marcatos and staccato rhythm ostinatos.

Much as I enjoyed these instruments, I have to reserve my highest praise for the English horn (aka cor anglais). It is absolutely stunning: warmer and darker than the oboe with a wonderfully poignant low register, it sounds equally magical high up in the oboe range. An ultra-expressive legato style incorporates emotive dynamic swells, while ‘Legato FX’ programs use quiet breath noises and key clicks to enhance realism. A playable instrument, and a brilliant example of woodwind sampling.

Woodwind Artics The woodwind players perform a near-identical set of articulations consisting of unlooped regular and legato sustains, marcato long notes, staccatissimo shorts, major and minor second trills and major and minor up and down octave runs. Each instrument plays between four and eight types of ‘arc’ (medium-length notes with expressive built-in dynamic swells). The oboe and flute also perform straight crescendos.

While the excellent legato mode can handle very fast playing, its note durations seem a little short — a reminder to program occasional pauses for breath! Though the instruments’ vibrato is baked in, you can use Soundpaint’s time-stretcher to adjust the vibrato speed, and also to alter note durations and attack speeds.

Three Duduks The duduk is an Armenian double reed woodwind instrument, described by ethnic winds specialist Dirk Campbell as “a cool, transcendent sound, best used with a minimum of background”.

The instrument (a wooden pipe punctuated by a row of finger holes with a mouthpiece inserted at one end) might appear rudimentary, but its sound has an unforgettable, haunting voice-like quality. Skilled performers can generate a range of sophisticated and expressive performance techniques, including bent notes, multiple grace notes and an impassioned vibrato.

Soundpaint’s three duduks were built in the keys of A, Bb and C, but thanks to chromatic sample mapping each one can play in all 12 keys. You can choose between sustain, marcato and legato articulations, the latter offering simple unornamented notes, single and double grace notes and slow inter-note portamento slides.

These legato variants are presented in keyswitchable four-way programs, so you can switch between straight legatos, grace notes or slides in real time.

Eight years in the making, the Soundpaint engine contains a wealth of creative features.

Soundpaint Solo Woodwinds & Brass From $20 pros • An extremely affordable selection of high‑quality, ultra-playable solo wind instruments.

• Contains some stellar solo woodwinds.

• The woodwinds were recorded from three mic positions in a fabulous European concert hall.

• The three duduks sound lovely.

• The Soundpaint player is fun to operate.

cons • None to speak of.

summary Showing a keen awareness of musicians’ budget constraints, Soundpaint make their excellent products available at knockdown prices. There’s nothing cheap about the sound; some of these solo woodwinds are among the best I’ve heard, and deserve serious consideration by all orchestral sample users.

Also included are layered ensemble and sound design programs, which work well for chordal passages and bass drones. Overall, an enchanting and beautiful-sounding trio of instruments, a worthy addition to Soundpaint’s catalogue.

Solo Brass Recorded in the aforementioned Zlín scoring hall, Soundpaint’s Century Solo Trumpet and Solo Flugelhorn are 100 percent musically and sonically compatible with the solo woodwinds.

Both were originally part of 8Dio’s Century Solo Brass, a seven-instrument Kontakt collection now beginning a migration to the Soundpaint format.

The solo trumpet is a winner, offering a large menu of bright, super-confident, well-tuned performances including looped sustains and polyphonic legatos with and without vibrato, marcato shorts, staccatissimos, swells (called ‘soaring’ for some reason), long and sfz crescendos, ‘arcs’, trills, flutter tongue, and double and triple tongue, some of which come in Harmon and straight mute variants.

The flugelhorn, a magnificently fat-toned, mellow instrument often associated with jazz, matches the trumpet for quality but omits the fancier artics. At the current prices, both are an absolute snip.

Jazz, Latin and pop producers might prefer Soundpaint’s Fire Brass, a brass trio drily recorded at Feeling Studios in Cali, Columbia. Operating under the names of 1984 Trumpet Fire, 1989 Fire

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SOUNDPAINT SOLO WOODWINDS & BRA SS

Trombone and 1992 Fire Saxophone, these instruments collectively cover the idioms of big band jazz and pop horns, whipping out unrestrained, in-your-face artics such as the overblown ‘frullatos’. While suitable for hot rhythm work, they also perform conventional artics (including legatos), so could be used in an orchestral setting.

Saxophones Our final instruments were recorded at Gorbals Sound Studios, Glasgow, from close and room positions. The fancifully named 1980 Tenor Sax Spectralius, 1999 Soprano Sax Moonlight and 1989 Baritone Sax Barry deliver a lorry load of articulations (including runs, scoops, triplets and ‘daa da da’ mini-phrases), but I’ve a feeling most people will take one listen to their utterly realistic legatos before compulsively reaching for their credit cards and buying all three.

I slightly preferred the tenor and baritone instruments’ timbre over the soprano — however, I very much enjoyed the slow, dreamy fade of the soprano sax’s ‘Somber Arcs’ program, a great textural backdrop for meditative melodic explorations.

Conclusion Troels Folmann has long been an advocate of hybrid sounds, so it’s no surprise that the Soundpaint factory programs include many sound design and effects-based creations. These atmospheric, sometimes transformative concoctions are not separately presented in the browser, so don’t be surprised when looking for a close-miked oboe if you end up with something resembling the distant piping of the god Pan. In that

The concert hall of the Cultural and Congress Centre, Zlín, Czech Republic, venue for the recording of many of the libraries on review here.

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Soundpaint Engine The free Soundpaint engine uses a proprietary audio format called Real-Time Samples to play back its recorded material.

All instruments can have 127 discrete dynamic velocity layers, with musical realism enhanced by the UDS (Ultra Deep-Sampled) versions supplied for some instruments. These contain identical material to the standard instruments, but introduce an additional set of real-life round robin for repeated notes.

Soundpaint’s HAL III (Hyper Acoustic Legato) technology offers limitless polyphonic legato. This means you can play cohesive, joined-up, single-note melodies on (say) a woodwind instrument, then instantly switch to chords on the same instrument and enjoy equally smooth inter-chord note transitions.

Eight years in the making, the Soundpaint modular interface contains the core engine, a downloader-cum-installer (which handles updates), and a user sample import (USI) with a built-in editor.

You can drag and drop your samples onto the keyboard, adjust their ranges and stretch them across multiple keys, then save your work as a single instrument ‘part’. Soundpaint also supports keyswitching and time-stretching, has a built-in arpeggiator and boasts a comprehensive effects section which includes a beautiful Lexicon-style reverb.

An intriguing feature of this sample player is ‘part morphing’. This enables you to modulate the sound of one part using another part, with a slider controlling the morph mix. A single program can have up to four sets of morphing parts, which opens up brain-boggling possibilities. Having heard Troels Folmann modulate a grand piano with an orchestral recording of Grieg’s ‘In The Hall Of The Mountain King’, I’m already a fan!

The Soundpaint player works as a plug-in and in standalone mode on Windows 10+ and Mac OS X 10.14+ systems. For full specifications, see the company website.

vein, I enjoyed the program ‘Memory MJ’, a mystic, tranquil flute echoing from some far-flung galaxy.

Also worth mentioning are Soundpaint’s ‘hidden gems’, factory programs featuring instruments from different libraries. When I installed one of the woodwinds the display suddenly shouted, “Congratulations Dave, you’ve unlocked new hidden gems!” (A bit sinister really — how did it know my name?)

Nevertheless, it’s a fun marketing tool, and some of the layered multi-instrument programs sound great.

If you’re looking to accompany the above‑mentioned melodic instruments with filmic ambiences, textures, synths, pads and vocals, I can recommend Dunescape 2.0,

a much larger and somewhat more costly library, which spans the sonic spectrum from delicate to cataclysmic. To spice up your rhythm tracks, you might also fancy the pulsing textures of Rhythmic Aura Acoustics, a former 8Dio product. I mention these additional libraries to give an idea of the wider capabilities of Soundpaint, a forward-looking company whose instruments have caught my imagination and inspired my creativity. £ Solo woodwind and Solo brass

instruments $20 each, duduks $30, Fire Brass bundle 2.0 $50, saxophones $30 each.

W www.soundpaint.com

TECHNIQUE

The SOS Guide To Playing To A Click We look at what you need to know to program and play to a click track.

Creating a tempo map in your DAW enables you to introduce small tempo variations, perhaps increasing the speed slightly in a chorus or decreasing in a verse, adding a natural pull and push that playing to a rigid tempo click can remove.

MARK GORDON

W

hen it comes to recording music in the studio, one of the most fundamental tools at your disposal is the click track. However, being asked to play to a rigid metronome can strike fear into even the most experienced musician. The idea of adhering to a click while attempting to give your best performance can be quite intimidating, and a stark reminder of your imperfections. So why use one at all?

There are certainly pros and cons to working with a click. It not only helps you stay in time and record tighter performances, it can also streamline the entire production process. From a production point of view, if a song has been played to a click, that allows you to do precise editing, easy looping, accurate drop-ins and seamless overdubs, especially when collaborating with other band members remotely or over extended periods. It also helps align any pre-recorded or machine-based elements

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such as sequencers, drum machines and loops, which are often a part of many modern productions, as well as ensuring that tempo-based effects such as delays remain in time.

The other side of the argument is the potential for natural ebb and flow to be eliminated, and for the song to feel sterile and cold. Of course, for some genres, particularly dance music, a strict tempo with each instrument locked to the grid is part of what makes that music sound the way it does, but sometimes using a click really isn’t appropriate. I have engineered several recording sessions where it became obvious within the first few minutes of introducing a click that it really was not going to be a productive method! At that point you must make a choice: can you use other techniques to achieve your production and editing goals, or do the band need to take some more time in the rehearsal room practicing with a click.

For the purposes of this article, we’re going to assume you’ve decided the best option is to use a click. How can

you, as a producer or musician, make the process work for you and be as painless as possible?

Clicks & Tones & Metronomes Not all click tracks are created equal.

Different musicians and genres require different approaches, and there are many ways of keeping in time that don’t require a cowbell banging in your headphones at ear-bleeding volume. Using the click in a way that complements the creative and emotional energy of the song, and choosing the right type of click track, can make a big difference.

1. Standard Audio Click The most common click is a metronome-style beep (or, indeed, cowbell), typically playing quarter notes at a pre-defined tempo. Often the downbeat (or the ‘one’) might be accented or assigned to a different sound to emphasise the start of each bar. It’s possible to make significant adjustments to even this basic type of click, to aid musicians. The sound choice

can make a big difference — some prefer softer sounds like shakers or rimshots to harsh digital beeps. Instead of just counting quarter notes, a click can be subdivided into eighth notes or 16th notes, making it easier to stay in time during slower or complex passages of music.

2. Visual Metronomes Some musicians find it distracting to play along with an audible click and prefer a visual metronome instead. The original incarnation of this was, of course the orchestral conductor, who would typically use a baton in their dominant hand to indicate the tempo and beat pattern, while the other hand was used to indicate dynamics, phrasing and expression.

Most hardware devices that feature an internal metronome (such as drum machines) will include some form of visual reference, often a flashing LED, and these metronomes can easily be sync’ed to your DAW . Many phone and tablet metronome apps also include sync facilities and feature all manner of visual feedback options, from a full-screen flash to numerical bar and beat counters. Interestingly, I have not come across many DAWs that include click options with strong visual feedback, but you can purchase a third-party plug-in for Ableton Live that features a fully customisable visual metronome with flashing lights and numerical bars and beats display.

3. Haptic Metronomes A relatively new phenomenon is the haptic metronome, which provides a physical vibration that you feel rather than see or hear. Several phone and tablet apps include the option to add a vibration on each beat, and this is taken to a more practical conclusion with vibrating smart watches enabling the wearer to feel a pulse on their wrist. In my experience, the haptic feedback offered by metronome apps on the Apple Watch is very weak and, although not unusable, it does require some level of concentration.

Better options are the Soundbrenner Pulse and Core, dedicated wearable vibrating metronomes

designed for musicians, which I reviewed in SOS a few years ago (www.soundonsound. com/reviews/soundbrenner-core and www.soundonsound.com/reviews/ soundbrenner-pulse). Not only do these devices provide physical feedback in the form of user-configurable pulses, they also offer visual feedback by way of an illuminated display. Designed for musicians, they also include the ability to sync to your DAW via the Soundbrenner app, which makes them a great option for both live and studio work.

4. Groove-based Clicks All the above methods rely on single pulses (whether they be audible, visual or physical) playing on each beat, quarter note or subdivision, but this can feel quite unnatural if you are used to playing along with a drummer or other musicians. Creating a simple drum pattern or using a drum loop that mimics the feel of the song can often be a much better option that feels far more natural to play to. A programmed drum pattern can also introduce dynamics, accents and even swing that can help musicians lock into a groove in a more organic way. I’ve found this to be a great method when working with musicians that are not used to playing to a click — it feels far more natural and closer to playing in a regular band context.

Move With The Times Once you’ve established the type of click you want to play with, you need to give some attention to tempo. For most bands, the tendency is to speed up during exciting bits, slow down into the dreamy bits, and for members to cast glancing smiles at each other during the cool solo bits.

In the majority of cases the click track would retain a strict tempo, but almost every DAW has the option to create a tempo map or conductor track, which allows you to introduce tempo changes, and even time signature changes, into your song to mimic these fluctuations.

The Soundbrenner Pulse and Core are wearable metronomes that provide a physical pulse that you can feel, which offers an interesting alternative to an audio click.

Loosely Tight Depending on the musical genre, the click can be viewed as an indicator of the tempo, rather than something you must stick to like glue. This approach will be more suited to guitar-based pop and rock bands who are looking to achieve a natural, expressive feel, while still maintaining solid tempo. It can mean some of the finer editing techniques you may want to employ further down the line become challenging, as everything isn’t always perfectly aligned to the grid, but it can produce outstanding results.

Playing ‘around’ the click is a skill that takes a fair amount of practice and experience, with the drummer doing most of the heavy lifting.

Try intentionally placing your notes slightly ahead of the click for a more driving, energetic feel, then try sitting exactly on top of it for a tight, mechanical feel. Finally, practice sitting just behind the click to create a laid-back, relaxed vibe. You can see exactly where your beats fall against the click by looking at the waveforms in your DAW and comparing this with how the rhythm and tempo feel.

By creating small tempo variations, perhaps increasing the speed slightly in a chorus or decreasing in a verse, you can add a natural pull and push that playing to a rigid tempo click can remove.

A good way of creating a tempo map is to base it on a rehearsal recording of the song. Many DAWs offer built-in automatic tempo detection, which can quickly create a tempo map based on transient detection, but you could also use a tap-tempo feature, or even manually map the tempo. Once a tempo map has been created, you can edit and adjust to suit your requirements while still retaining the original feel of the song. It’s unlikely you’ll want dramatic shifts in tempo and, with a little practice, playing along to the variable click will feel quite natural.

Hit Them With Your Rhythm Click Playing to a click isn’t being forced to listen to a quarter-note pulse in your headphones and instantly producing a top performance. It’s about experimenting with what works for you and, like any skill, it becomes easier with practice. When paired with dynamic tempo maps, musical cues, or rhythm-based clicks, the experience can feel as organic and expressive as playing live. In fact, many musicians (and I include myself) discover that playing to a click becomes enjoyable in its own right, and an essential and even gratifying part of the recording process.

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Synthesizer

Korg microKORG2 Updated for a new generation, can the microKORG2 live up to the huge success of its predecessor?

WILLIAM STOKES

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he microKORG. Where to begin?

With its legion of long-time users, it’s not the kind of instrument to be upgraded without some serious consideration. And I dare say the time is right: you hardly need me to tell you that the marketplace is now far more congested and competitive than it was in 2002, particularly in the realm of small-but-mighty synths. What was once a reliable slam-dunk year on year for Korg is now in a position where it very much needs to reassert its relevance for fear of being left behind.

The original microKORG managed to slake the needs of two very different parties equally: preset-flicking users, with an idea of what sound they want but little to no interest in electronic sound sculpting, and then

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more experienced players needing a convenient instrument with which they can craft and tweak sounds down to a reasonably fine level of detail. Parallel to this was its modest selection of big, accessible physical controls and, above all, a really rather excellent-sounding DSP analogue‑modelling system ported straight over from the significantly more esoteric MS2000. All of that really constitutes the criteria for this review: if the microKORG2 doesn’t similarly mediate between depth and accessibility, it simply has no raison d’être.

Next In Line I’ll try to avoid simply listing the differences between the microKORG2 and its predecessor, but in terms of spec it simply offers more, and in almost every way. It doubles the amount of voices from four to eight, and also the capacity for preset

storage from 128 to 256. There’s a new sound engine, more effects and more modulation possibilities, more types of arpeggiator, a significantly upgraded vocal processor, and a handful of new features such as the loop recorder.

The panel’s most conspicuous indication of all this, perhaps predictably, is a lush 2.8-inch colour screen, which immediately upends the traditional microKORG workflow. I should say straight out of the gate that the original microKORG’s workflow was by a mile the weakest thing about it. Assignable knobs clunkily programmed by way of two encoders; a cramped three-digit, seven-segment display doing its best to display words, reams of parameters written on the body of the synth itself... it strays perilously close to the likes of the supremely clunky Korg Poly-800 (tattooed up to the neck with on-body parameters — though we’ll forgive it, because, well, ’80s), so I’m glad to see the back of most of it.

Here on the microKORG2, though, there is blissfully little text on the body of the thing. “Don’t worry,” goes Korg’s spiel, “it’s all about enhancing your experience, not complicating it.” Glad to hear it. The fundamentals of the microKORG panel

remain, of course: five assignable Edit knobs, a large central Program Select dial and a row of big Program buttons. A Mic knob (to adjust level after A‑D conversion) has appeared next to the Volume knob, and of course the centre of the panel has been all but overhauled with the inclusion of the screen. Impressively, despite feeling sturdier and featuring an improved, slightly bigger keyboard, the whole thing is actually slightly lighter than the original.

You’ll be either pleased or dismayed to know that cheesy genre labels still orbit the Program Select dial, though these seem to have been updated for 2025 (to reflect, erm, Korg’s finger on the cultural pulse?), displaying ‘Pop/Rock’, ‘Funk/Soul’, ‘Hiphop/R&B’, ‘House/Disco’, ‘Ambient/Electronica’, ‘Techno/Trance’, ‘DnB/Dubstep’, and ‘Game/SFX’. Then as now, I honestly cannot think of anybody who would seriously choose to organise their sounds as if they were programming muzak; and the moment you start editing those sounds, you’re likely to deviate from those descriptors anyway.

Flow State Nonetheless, the architecture of the microKORG2 is intuitive and immediate.

If you’re preset-flicking you can simply turn the Program Select dial, choose one of the four banks below and use the buttons below that to choose a preset.

You can also save a set of eight Favourite presets to the buttons for quick access or to craft performance sets, which is handy. Once in a preset, it’s essentially a case of following the buttons on the right side of the panel from left to right — from oscillators, to filtering, to modulation, and so forth. Moving through each button’s screen, the five Edit Control knobs (physically colour‑coded a little subtly but nice and clear on the screen) take on different roles as specified by the GUI, along with live parameter readouts and in most cases quirky little icons and animations to accompany them. Atop the aforementioned Favourite presets, once any preset is edited and written you can save a set of assignments to

the five Edit knobs for quickly accessible performance parameters. Lovely stuff.

Belied by this slick workflow is the sheer amount of variation on offer with the microKORG2. For starters, like its predecessor, it’s bi-timbral; a simple and effective way to allow sounds to become more than the sum of their parts, though of course this halves the voice count from eight to four. Each timbre offers three oscillators and a noise generator, each of which presents simple saw, square, triangle or sine waveforms, as well as DWGS (Digital Waveform Generator System), which creates FM-style tones with single-cycle complex waveforms, and finally OneShot, which delivers a selection of 32 fairly basic PCM waves for passable pre-packaged percussive or metallic sounds. The Shape parameter allows for further variation within these waveforms, offering one predetermined parameter for quick waveshaping. With a sine wave, for instance, it’ll clip the wave into saturation; with a square wave it’ll adjust pulse width. For DWGS and OneShot it’ll cycle through samples. Osc Mod allows for cross-modulation, oscillator sync, ring modulation and variable phase modulation between oscillator 3 and oscillators 1 and 2; there’s also a highly flexible noise generator, with selectable noise ‘colour’, its own filter or decimator and keyboard tracking per oscillator. Whew!

The filter section has all the parameters you’d hope for — different modes, key tracking and the like — but once again there’s a very generous amount of variety on offer, for instance 24dB/oct or 12dB/oct curves for the high- and low-pass filter modes, and also a rather nice-sounding filter drive.

Overall the filter itself doesn’t have much character, but in this instance I wouldn’t expect it to. Its role, like much of what’s on offer here, is to fit into a frictionless workflow, and in this respect it does well.

Micro But Mighty These are solid building blocks by any metric, but with instruments like this I tend to look to the modulation and

Round the back we find full‑size MIDI I/O ports, inputs for condenser and dynamic mics, an aux input, damper switch input, stereo audio outs, headphone out and a USB‑C port.

Korg microKORG2 £519 pros • A slick and spacious workflow that takes very little getting used to.

• Huge range of sounds on offer.

• Generous amounts of modulation and effects options.

• A well-designed arpeggiator with a few tricks up its sleeve.

• Sturdy build quality — yet still lighter than the original.

cons • With so many assignable functions, endless encoders would be preferable to knobs.

• Needless ‘Trophies’ screen.

summary Korg have gone back to the drawing board where it counts and done well to keep the charm of the original microKORG intact. A worthy successor to one of the world’s most popular electronic instruments.

effects sections to see what the scope for going deeper really is, and here I was not left wanting. The amount of modulation on offer is quite amazing, with six ‘virtual patch routes’ allowing for up to two modulation sources per patch to control a litany of destinations, from fairly conventional parameters like filter cutoff to more off-piste ones like arpeggiator octave-jumping or vocoder formant. The selection of sources itself is also wide, from keyboard tracking to the pitch-bend wheel.

On the subject of effects, the routing here is highly flexible as well: they’re not just on or off the master bus, they can be discretely assigned to each timbre layer, or to audio coming via the mic jack.

Effects are also applied to audio coming in via the aux in jack: while I couldn’t see a way for this to be switched off without disabling the effects themselves, it does mean the microKORG2 can effectively become a relatively powerful effects processor for just about anything coming down a 3.5mm cable. The effects are organised into modulation effects, delay and reverb. Each of these has generous variety within: there are nine modulation effects, from the expected chorus, phaser

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KORG MICROKORG2

The microKORG2 measures 542 x 238 x 65mm and weighs a portable 2.1kg.

and flanger, to other effects that perhaps stretch the category of ‘modulation’ a little confusingly, like compression and even an amp simulator with different amp types. There are six distinct types of delay, from classics like Tape Echo to the more experimental Pitch Shift; and six types of reverb, from Room to the intriguing Rust — a surprisingly characterful lo-fi style of reverb.

On the vocal side of things, the microKORG2’s included gooseneck microphone is fit for purpose. I’ve seen many online complain at its rough and ready character, but I have to say I quite like it — even if you basically have to kiss it to move the diaphragm. Overall, though, I deem it vital that a mic be included; if for no other reason than the fact that many users — particularly at the less experienced end of the spectrum — would be unlikely touch the synth’s vocal processing capabilities if they had to provide their own. The microKORG2 offers a vocoder, hard tune and harmoniser, each with their own buttons and all of which sound both fun and very useable, particularly when used in conjunction with the effects. The harmoniser I particularly enjoyed, quickly achieving some pleasant chordal stylings straight out of the Bon Iver playbook. It all beds in nicely with other aspects of the microKORG2’s architecture, and as such can be as core a part of the experience as you want it to be.

Interestingly the microKORG2 lacks a sequencer of any kind, though in many ways it makes up for this with a creative array of other, similar features. I was

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pleased to discover that the arpeggiator — as you may have come to expect by now — capably covers the conventional bases of latching, swing, octave jumping and various pattern configurations, but it takes things further by incorporating a handful of options you’d be more likely to find on an actual sequencer, such as variable gate time and also last step. This clever little feature works in conjunction with the program buttons, which allow steps in the arpeggiator to be skipped for some interesting syncopation. Add to this a bpm-sync’able delay and the fact that you can choose to only have one timbral layer arpeggiate if desired, and from a very simple starting point you can find yourself in startlingly complex and creative territory.

Work & Play Fun graphics? Colour-coded knobs?

Trophies? I agree: it does all lean rather heavily toward the Elektron and Teenage Engineering schools of synth design, and away from the distinctly Moog-ish visual cues of the original microKORG with its faux-wooden sides. I mean, Korg didn’t think they could include playful graphics and a row of colour-coded assignable knobs on the upper right of the panel and not draw comparisons to the OP-1, right? In this respect the only truly puzzling screen page I came across on the microKORG2 is the Trophies page: a series of greyed-out icons which are then coloured to reward certain ‘achievements’: “To celebrate you exploring the functionality of the microKORG2,” says the manual. I thought that the genre labels around the Program

Dial were the cheesiest thing about the microKORG2. I was wrong. It’s certainly a brave stab at the aesthetic those Swedish developers manage to incorporate so naturally, but here executed with all the nuance of David Brent singing ‘Free Love On The Free Love Freeway’.

The only thing it did leave me wondering is if winning all the trophies somehow unlocks a new bit of functionality. But if that is the case then, in short: I’m fine, thanks. Because, just like its predecessor, the microKORG2 is actually a serious instrument and should be considered as such. It simply masks its seriousness somewhat with a deceptively simple interface — and that, in short, is the magic of the microKORG. It’s finely tuned, flexible and road-ready.

It can cover the basic needs of touring musicians or seasoned producers while allowing scope for surprisingly in-depth sound sculpting; but it can also coax less experienced players into new territories of synthesis, facilitating genuinely rich and inventive sounds without feeling overly pre-packaged or, by contrast, daunting.

As I said at the beginning of this review, that was my rubric, and by that rubric the microKORG2 passes with flying colours — and it even manages to do so while disposing of many of its predecessor’s niggles. If you’re in the market for an original microKORG, you’re in luck: I wouldn’t be surprised if the secondhand market is flooded with them this year. £ £519 including VAT. W www.korg.co.uk

88C DUAL DESKTOP DYNAMICS

Iconic Sound Quality | Versatile Control | Legendary Character

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Dynaudio Acoustics M1 MkII

Passive Monitors In updating their classic M Series, Dynaudio Acoustics show that passive doesn’t need to mean low-tech.

PHIL WARD

D

ynaudio, together with studio designer and acoustician Andy Munro, played a really significant role in the development of the contemporary professional studio monitor — particularly in Europe. The story goes that Andy Munro, dissatisfied with the horn‑loaded compression tweeters found in the vast majority of main studio monitors of the 1970s and 1980s, hunted around Europe’s specialist driver manufacturers to find one that could supply high‑performance, non-compression tweeters able to survive the rigours of studio use. Munro alighted on Dynaudio in Denmark, and a deal was done that brought the company’s renowned drivers into a range of Munro‑designed and ‑manufactured studio monitors. The monitors would be marketed by Munro, but with Dynaudio Acoustics branding.

That monitor range was the original Dynaudio Acoustics M Series, and over

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the last three and a half decades, M Series monitors have been installed in countless studios the world over.

All Change The original M Series speakers ranged from relatively compact nearfield/midfield models right up to huge, soffit‑mounted systems incorporating multiple bass drivers, and although they were incrementally developed and improved over the decades, there was never a root and branch redesign — until now. Once again, Munro Acoustics have taken the best of Dynaudio’s driver technology and built around it a range of new M Series passive and hybrid active studio monitors.

The new M Series again encompasses models from the relatively compact M1 MkII right up to the enormous, soffit-mount, four-way M4 MkII. Furthermore, to partner with the new M Series, Dynaudio Acoustics have also developed and launched a range of power amplifiers designed to complement the monitors.

So, along with a pair of M1 MkII monitors,

I was also loaned a pre-production sample of the new DA-2 power amplifier.

The M1 MkII is the smallest of the new MkII M Series. Dimensionally it inhabits a territory that, to my way of thinking, is nearfield bordering on midfield. To put some numbers on that, it’s 256 x 476 x 393mm. It is designed exclusively to be installed in landscape orientation. The M1 MkII is of course heavy, at 14kg, but considering some of the monitors I’ve had to heave onto my monitor shelf, not outrageously so. As far as the published data goes, Dynaudio Acoustics specify a frequency response of 50Hz to 20kHz, a maximum SPL of 122dB from a pair of speakers at 2m, and total harmonic distortion of 1% at 90dB SPL. The specifications are written generally without constraining limits, so are of limited value.

While we’re talking specs, the DA-2 is a substantially constructed and satisfyingly heavy 2U unit rated at 500 Watts per channel into 4Ω. It provides both balanced and unbalanced input options, and output options of binding posts and Speakon sockets. The amplifier is convection cooled so generates no bothersome fan noise. Somewhat unusually, the DA-2 combines a conventional 500VA

linear power supply with Class-D output topology, the latter based on Ultra-Sigma technology. The DA-2 is clearly a very well‑specified and high‑performance power amplifier.

Returning to the M1 MkII, its cabinet is manufactured in 22mm and 18mm MDF panels and finished as standard in traditional, pro-audio textured black paint; oak, cherry, walnut and maple veneer finishes are optionally available. The cabinet is not internally braced and sounds marginally live to the knuckle tap, but it carries foam panels on its internal surfaces that will provide some damping. The cabinet is not rectilinear, but reduces in width significantly towards the back. This is likely to bring acoustic benefits, helping to suppress internal cabinet resonances that might otherwise escape out of the reflex port or through the relatively thin material of the bass/mid driver diaphragms.

Architecture In electro-acoustic terms, the M1 MkII is a conventional, reflex-loaded, passive two-way system. It incorporates two custom-specified Dynaudio 180mm bass/midrange drivers and a Dynaudio Esotar3 28mm soft‑dome tweeter, arranged in a MTM (mid-tweeter-mid) D’Appolito array. This will result in relatively narrow horizontal dispersion through the upper midrange and will generally have the effect of suppressing side wall reflections and shifting the transition between the acoustic nearfield and reverberant fields further back into the listening space — at least at those upper midrange frequencies. Being reflex loaded, the M1 MkII sports a reflex port.

The port is located beneath the tweeter on the front panel and is notably short (not much longer than the front panel is thick) while still being relatively generous in diameter. The port is flared on its exit but not on its entrance.

Now, the length of a port defines the reflex tuning frequency, which will be chosen by the designer to achieve the desired low‑frequency response characteristics for the monitor, and the port diameter effectively defines how loud the monitor will play before port compression and distortion becomes a problem. The challenge for the designer is that the larger the diameter of a port, the longer it must be for the same tuning frequency.

But when a port length approaches around five times its diameter it will likely fall prey to secondary (and highly undesirable)

organ-pipe resonance effects. So, all that considered, the low‑frequency sums have clearly been well managed to enable such a short yet large‑diameter port. I would have preferred to see some flaring on the port entrance, but generally, the M1 MkII’s reflex port implementation augurs well.

The two bass/mid drivers work in parallel before handing over to the tweeter at a relatively low crossover frequency of 1650Hz. The crossover itself employs nominally second‑order (12dB/ octave) filter slopes and comprises only air-cored inductors, polypropylene film capacitors and wire-wound resistors.

It is implemented on a glass-fibre printed circuit board with generous, high‑current‑capacity copper tracks.

Driver Technology On the surface, then, the M1 MkII is a relatively uncomplicated traditional passive loudspeaker; there’s none of the active electronics and complex DSP that characterises many of today’s monitors.

However, that surface hides some pretty sophisticated engineering in the drivers. I’ll kick off with the bass/mid driver. Dynaudio have long followed a somewhat atypical path in bass/mid driver design. While the unusual external voice-coil configuration that characterised early Dynaudio drivers has mostly been jettisoned (and isn’t employed on the M1 MkII driver), there are still a number of design and constructional techniques not often found elsewhere.

One example is the integrated diaphragm and dust cap, and a second is the use of a unique mineral‑filled polymer diaphragm material, christened MSP. While the current trends in bass/mid driver diaphragm material seem to be either towards traditional paper and aluminium items, or very high-tech materials such as carbon fibre/graphene composites, Dynaudio have stuck resolutely to their MSP solution.

Dynaudio Acoustics M1 MkII £7578 pros • Great integrity, consistency and clarity.

• Insightful, detailed midrange.

• Exceptional high‑frequency performance.

• Suited to compact rooms.

cons • None.

summary The original M Series became a byword for reliable, consistent high‑performance monitoring in busy, professional studio facilities. On the basis of the new M1 MkII, there seems no reason why that should change.

They argue that MSP’s particularly beneficial combination of rigidity, density and damping can’t be had by any other means. And while I’m on the subject of the MSP diaphragm, in its particular application in the M1 MkII bass/mid driver, the material is just 0.4mm thick. It’s a very light diaphragm.

Hidden behind its diaphragm, the M1 MkII bass/mid driver employs a voice coil comprising aluminium wire wound on a glass-fibre former. The use of aluminium wire is relatively unusual, but it offers significantly lower moving mass than the much more common copper wire option.

And the glass fibre voice-coil former of the M1 MkII bass/midrange driver offers two advantages over the the aluminium more often chosen for the role. Firstly, it is not an electrically conducting material, so its movement through the magnetic field of the driving magnet doesn’t result in distortion‑inducing eddy currents, and secondly it conducts less heat to the thermoplastic diaphragm. At 55mm in diameter, the voice-coil is significantly

Diagram 1: The M1 MkII’s on-axis frequency response, measured at 90dB SPL at 1m. Second- and third‑harmonic distortion is shown in green and blue, respectively.

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DYNAUDIO ACOUSTICS M1 MKII

Diagram 2: Comparing the axial frequency response (red trace) with measurements taken 20 and 40 degrees horizontally off‑axis (green and blue, respectively).

larger than is typically found in a 180mm driver, and this bodes well in terms of power handling and resistance to thermal compression. And speaking of driver‑borne compression, the generous voice coil enables a large‑diameter flared vent hole to be located in the pole piece, so that air pressure under the diaphragm dust cap is relieved. Further bass/mid driver refinements include suspension spider and rubber surround components optimised through finite element analysis (FEA) to maximise linear excursion and thus reduce distortion, and a copper pole-piece cap that reduces voice-coil inductance and, perhaps more significantly, reduces the degree to which that inductance is modulated by voice-coil excursion. While we regularly read from monitor manufacturers that improving driver performance is all about big technological advances such as new diaphragm materials, the truth is that the devil of driver performance is very much in the detail. Designing a better bass/ mid driver is about optimising all the small details, and the M1 MkII bass/mid driver looks to be a good example of that principle.

Tweeter performance is also all about the small details. Esotar tweeter models from Dynaudio go back to the 1990s, and the Esotar3 is the latest in the line. It is a 28mm‑diameter soft driver designed, say Dynaudio, for very high‑performance applications, particularly those that demand high sound pressure levels. The Esotar employs a particularly powerful neodymium-iron-boron magnet system that means its inherent sensitivity is so high it is able to reach the sound pressure levels required of the M1 MkII while accommodating a low crossover frequency; the latter parameter is

additionally aided by the tweeter’s unusually low fundamental resonance frequency. The Esotar3 also incorporates a sophisticated absorptive element, christened Hexis, that dissipates the dome’s rearward acoustic radiation to ensure it doesn’t reflect forwards.

Measuring Up I took the M1 MkII to my usual large and empty speaker measuring venue (a local church) to capture and analyse a little electro-acoustic data. The four diagrams in

format of the M1 MkII would result in narrow dispersion through the upper midrange, and that’s revealed very clearly in Diagram 2, which shows the axial response curve overlaid with curves taken at 20 and 40 degrees horizontally off-axis. This phenomenon is likely to have some significant subjective repercussions and I’ll write more about that further down the page.

For Diagram 3 I placed the measuring microphone right at the mouth of the port to investigate whether anything undesirable finds its way into the outside world. The result is a clean bill of health for the M1 MkII port. It always surprises me how little of the intense acoustic energy bouncing around inside a speaker enclosure finds its way out through a well‑designed port, and in the case of the M1 MkII, I’d hazard an educated guess that its non-rectangular cabinet helps in this respect.

Finally, Diagram 4 shows input impedance: data that I rarely get to include in monitor reviews, because so few of them cover passive speakers. Now, those readers used to reading speaker input impedance curves might be surprised to see three curves overlaid, and those not used to reading speaker impedance curves might be a bit surprised that speaker input impedance is so complex. Either way, a bit of explanation is probably in order.

Broadly, the input impedance of a passive speaker varies widely over the audio band, partly because driver voice coils are inductive and they move (which results in multiple dynamic impedance phenomena), and partly because passive filters are created from networks of inductors and capacitors which themselves present impedance that varies with frequency (which is how they

“The M1 MkII sounds wonderfully consistent and, in monitoring terms, trustworthy.”

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this article are the result.

Diagram 1 shows the M1 MkII axial frequency response measured at 90dB SPL at 1m. The response curve appears nicely flat, with a smidgeon of midrange emphasis and gently drooping high‑frequency response. Overlaid on Diagram 1 are second‑ and third‑harmonic distortion curves, and both look very well controlled at typically better than -50dB compared to the fundamental — that equates to 0.3%, and that’s a good result.

I mentioned earlier that the D’Appolito

Diagram 3: A close-mic measurement of the port output.

do their filtering job). The result, as shown in the red curve of Diagram 4, is that the M1 MkII input impedance magnitude varies from a high of around 28Ω at 35Hz to a 3Ω low at around 200Hz. The two peaks (35Hz and 78Hz) and central trough (55Hz) are, by the way, characteristic of the particular arrangement of cabinet volume, driver parameters (moving mass, suspension stiffness and so on) and reflex tuning frequency, the latter being defined by the central trough. So the impedance magnitude curve reveals that the M1 MkII port is tuned to 55Hz, which ties in with the published 50Hz low‑frequency cutoff.

The green phase curve of Diagram 4 is associated with the red magnitude curve in that it shows the phase angle, in degrees, between the input voltage and input current. If the M1 MkII, or any passive monitor, were to display an entirely resistive input impedance, the phase angle between voltage and current would be zero.

However, as discussed, speakers display multiple inductive and capacitive effects, so the angle is typically anything but zero.

This is significant because amplifier output stages are typically more stressed as phase angle increases, and this is especially the case if a large phase angle, say greater than 45 degrees, is combined with a low impedance. The impedance phase angles displayed by the M1 MkII are entirely typical and reasonable, and I’ve overlaid a third curve to Diagram 4 (one that’s beginning to be used more commonly in the speaker design community) that displays the effective load the amplifier sees from the combination of impedance magnitude and phase. The curve is known as the Equivalent Peak Dissipation Resistance (EPDR for short), and it shows a calculated resistive load that would give rise to the same peak amplifier power. The M1 MKII EPDR curve shows that the equivalent resistive load seen by the amplifier drops to pretty low values for a significant portion of the audio band, so it’s reassuring that the DA-2 amplifier is stable down to 2Ω and is able to deliver up to 35A.

Listening Tests With the M1 MkIIs on my monitor shelf I began listening to a playlist of regular favourites. This is always a slightly nervous moment for me with a new monitor because, well, what if it’s obviously flawed? There was no such worry with the M1 MkII, though, because it was clearly very capable and well sorted from the start.

Diagram 4: The M1 MkII’s input impedance. Magnitude is shown in red, phase rotation in green, and the EPDR (equivalent peak dissipation resistance) in blue.

The overwhelming characteristics of the M1 MkIIs for me are balance and integration. Its overall tonal balance is spot on, and bearing in mind the very limited scope in passive speaker engineering for finely targeted EQ, this speaks firstly of very high‑performance drivers and secondly of great skill in their integration and voicing. And it sounds very ‘together’; there’s a consistency of character and performance all the way through the audio band. Some speakers, for example, might combine an explicit, insistent high‑frequency character with a slightly soft and warm midrange character, and that kind of contradiction rarely sounds comfortable. The M1 MkII sounds wonderfully consistent and, in monitoring terms, trustworthy. Bass is well extended downwards, but not so ambitiously that it trades bandwidth for dynamic or pitch accuracy. And it remains consistent from low to high volume levels.

The M1 MkII’s midrange performance is characterised both by the usefully neutral balance and a natural and insightful ‘hear-through’ clarity, especially in the lower midrange/voice band. I’d wager that this is a direct result of the low‑distortion driver engineering and that MSP diaphragm, but it is also I think emphasised by the narrow dispersion that suppresses side wall reflections. It feels to me like the M1 MkII would be pretty well suited to use in relatively small mix spaces.

Having said that, in larger spaces where more listening potentially happens in the reverberant field, the M1 MKII might not be so well suited, because the reverberant field perhaps will lack a little upper midrange energy.

Moving up the audio band, the Esotar3 tweeter is obviously an extremely high‑performance driver. One subjective criterion for me when considering subjective high‑frequency performance is to consider how individually noticeable

ALTERNATIVES There are relatively few passive monitors available in the M1 MkII’s price band, but considering a budget that also covers a DA-2 power amplifier, active monitors such as the Telegrapher Gorilla, ATC SCM45A, PMC6-2, PSI A23M and Dutch & Dutch 8C would all be worth considering.

the tweeter is. Does it draw attention to itself? In these terms the Esotar3 implementation in the M1 MKII is pretty much faultless. Continuing the theme of consistency and integration, the Esotar3 itself is all but inconspicuous, but the level of natural high‑frequency detail is extraordinary. With brushed cymbals, for example, it’s almost as if each strand of the brush is made audible. And this is achieved without any sense of unnatural high‑frequency emphasis.

It never ceases to amaze me just how many ways there are to conceive, configure, design and manufacture a speaker that can play convincing music and effectively carry out the monitoring function. For example, in engineering and technology terms, the M1 MkII couldn’t be much further away from, say, the Kii Seven, yet they both sound great and can do an excellent monitoring job. Back in 2020 I wrote a Why I Love column about passive speakers and the element of artistry that I feel is perhaps taken away from speaker design when almost limitless electronic and DSP resources are available. The M1 MkII is a perfect example of the kind of passive speaker I love. It’s full of skill and artistry, and it sounds really great as a result. £ £7578 per pair. DA‑2 £4779.

Prices include VAT.

T Synthax Audio +44 (0)1727 821870 E info@synthax.co.uk W www.synthax.co.uk W www.dynaudio.com

www.soundonsound.com / July 2025

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TALKBACK

Disiniblud

D

of choir practice taken by someone’s grandmom in northeast Philadelphia, and it’s the most precious music I’ve ever heard.

I just want to make more music that sounds like that. If anyone is working recreation at a retirement home in LA or something, hit me up!

At the moment I can’t stop listening to Rachika: Kashiwa Daisuke’s Re: and Piana’s raula. They tear me to pieces. Everyone in the FLAU / Virgin Babylon scene from Japan’s late-oughts is a huge inspiration for me. I met a few of those artists in Tokyo last year and was fangirling! Also, I’ve been listening to the singles from the 1975 Live At MSG album.

Nina: The song ‘Car Alarm, Turn Signal’

by Lia Kohl featuring Ka Baird. I love Lia’s music so much. Something about this song, when the car alarm starts going off it moves me to tears. She’s so incredible at tying found sounds and life’s minutiae with these crushing harmonies in the most potent way.

I always feel so much clearer after listening.

The first thing I look for in a studio Rachika: Guitar pedals. Then a synthesizer.

I’m considered a guitarist, but I actually often don’t pick up the guitar in the studio.

I process guitar slowly over time, more than I play guitar. I am bad at guitar and approach guitar as a producer, not as a guitarist. I think I just get bored with what a guitar can do.

It’s an instrument with a very hard attack and you can’t control the sustain. It’s kind of limited. I wish I could play the violin or cello, but guitar is what I spent time learning as a kid. I’m in a different place to where I was when I was 12, but it remains the instrument I know best and have the most connection with. Putting it through Ableton is how I get the sounds I actually want out of it.

Nina: Usually the piano, if there is one.

I’m always immediately taking it apart also. I get weirded out by playing a piano with all the panels closed. I don’t like that there’s a big wooden box holding all the resonances like a coffin. I have to hear all the details in the high end or I lose my internal compass completely.

WILLIAM STOKES

isiniblud are producers, composers and multi-instrumentalists Rachika Nayar and Nina Keith, whose self-titled debut LP is released this July on Domino imprint Smuggler’s Way. The duo’s otherworldly music features guests including Julianna Barwick and Tujiko Noriko.

The artist I’d most like to collaborate with Rachika: Most of those people I’ve asked, and have now collaborated with! Life can be so surprising if you put your heart on your sleeve and give it to the people you admire. As for which one meant the most... probably this album with Nina. I can’t really put any of them on a scale, but making music with Will Wiesenfeld the first time we met was very special to me. His album as Geotic, Mend, has taken care of me in all my darkest and most intimate moments since high school.

Nina: I want to write music for a senior choir. I would like to make long choral pieces for like, a 60-and-up squad. I have some ancient cassettes I’ve found in thrift stores, like, handheld cassette recordings

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The person I would consider my mentor Rachika: So many of my gay little friends.

I think we’re all each other’s mentors and mommies. Nina obviously. Walt McClements.

Will Wiesenfeld. Aside from that, there’s my childhood guitar teacher Jeff. Walt is one of the few older people I know who, like me, is both from a radical queer community since a young age and also been around quote-unquote ‘music industry’ stuff for a long time. So I look up to him as a model of how to maintain integrity with your values as you’re pulled into different worlds to those you’d usually choose to inhabit. That

stuff comes to mind more right now than whatever production things I might think of.

Nina: I don’t think I have one. I dropped out of school when I was 16 and have just been Googling it or whatever! I’m lucky to have an abundance of gays that I can call for guidance... audiobooks... dads at the hardware store... I would love a mentor, though.

My go-to reference track or album Rachika: I should really start using reference tracks! I only end up using them when I come up against a very specific issue or goal with a song, I guess. Usually they’re songs way back from my teenage years era that showed me some big songwriting idea — some magical thing that can happen if you get the elements just right. If I want a sub-bass to really shake your chest, I’ll bring up James Blake’s ‘Limit To Your Love’ or Haxan Cloak’s ‘Consumed’. If I want a climax to blossom a certain way, I might bring up M83’s ‘Beauties Can Die’.

If I want the song to suddenly pivot into the underworld, it’s maybe the Jamie XX remix of Nosaj Thing’s ‘Fog’.

Nina: It’s constantly changing. But I think ‘Surf Solar’ by Fuck Buttons has been

Photo: Allegra Messina

sound design. It’s as if Hildegard von Bingen was given every musical tool from the year 2450 and ascended to the astral plane.

Nina: Probably Cynthia [Erivo] and Ariana [Grande] recording ‘Defying Gravity’. I was a lifelong hater of musicals until I saw Wicked a few months ago. And like, four times since then. Also, I just remembered someone said they recorded all their vocals live as they were filming? If I could witness Cynthia belting from inside the tower in the Emerald City, that would be ideal.

The producer I’d most like to work with Rachika: It’s the same answer as I had for ‘the artist I’d like to work with’!

Nina: Frank Ocean. ‘White Ferrari’ is my favourite song. The last verse where he’s like, “I’m sure we’re taller in another dimension...” really ruins me. I love that there were like 50 versions of this song and that it took forever to finish. I relate to that way of working when it calls for it. Like when the jewel of the song is too precious and just needs some sort of miracle that eventually happens after working on it for like two years.

a go-to for big intense moments. I think with more neoclassical stuff I try to stay away from reference tracks because I’ll just go insane! I’m always multitracking myself playing cello or flute or something, with 40 little stems going at once, and the more I try to make it sound like a string quartet in a room, the more frustrated I get.

My secret weapon in the studio is Rachika: Automating the Ableton Texture Warp Mode parameters. That’s the main way I got all my ‘Rachika guitar sounds’ on my first two albums. I’ve never owned or used much gear. Ableton just has a few different algorithms for its time-shifting and pitch-shifting engine. The different modes exist so that you can stretch samples more transparently depending on the type of audio — say, a synth pad versus a drum break. But if you abuse it, stretch the audio to its limit, and then mess around with automating the parameters, you can get a lot of interesting freaky sounds. Each mode has its own specific artefacts.

Nina: I use low-pass gates a lot, like the Make Noise Optomix and Tiptop-Buchla 292t Eurorack modules. I feel like that’s one of the few things that makes all the

electronics feel more alive and organic.

I think that’s what makes everything I do on the modular feel cohesive and blend with all these acoustic textures. There’s something kind of romantic about vactrols. The way that circuits communicate with each other by sending flashes of light inside a tiny dark hallway so that they can stay isolated and protect each other from their own voltage.

I like how this is just magically happening inside these components but strictly hidden from the human eye so that it can stay pitch dark inside the little cavity until you ask it to make a sound and blink its light for no‑one.

The studio session I wish I’d witnessed Rachika: My friends KMRU and Aho Ssan making Limen or Dear Laika making Pluperfect Mind. I’m like: how did you make any of these sounds? The former two are ambient-electronic artists. The album is this massive, swirling, splintered landscape of volcanic noise and bass that is so moving and emotional. I don’t know, it just sounds so untamed and wild and cosmic. Izzy [Dear Laika] is the most galaxy-brained composer and songwriter I’ve ever met. There’s so much nuance in Pluperfect Mind, both in the chord arrangements themselves and the

The studio experience that taught me the most Rachika: Every single time I sit down with Nina. So many little ideas! High-passed tape delay on a close‑miked piano, Serum FX patches of filters with daisy-chained LFOs... We’ve also just together developed so many little production ideas while making this album that became pivotal sounds in our toolkit. One of my faves is the particular percussion style we came up with by processing field recording sounds with particular Max for Live granular plug-ins.

Nina: Pretty much any time that Will Wiesenfeld is talking about mixing or writing, it’s like the most valuable jewel of information. I feel like for both Rachika and me, he’s influenced so much of our practice and so many of the little decisions we make in Ableton. He’s always game for the stupidest questions I have and so good at explaining the more unexplainable things about mixing.

The advice I’d give myself of 10 years ago Rachika: There’s nothing to prove. Learn to be your own witness. It’s OK to be pretentious! Aim below your threshold.

You can’t save them. And there’s no one coming to save you, either.

Nina: Start playing cello, get on oestrogen, you have fibromyalgia, stop overworking yourself for people you hate.

www.soundonsound.com / July 2025

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ON TE ST

Cong Burn Stacks

If you want proof that original independent music software is still flourishing, you need look no further than Stacks.

ROBIN BIGWOOD

S

tacks is a new virtual instrument for macOS, Windows and iPadOS

by Cong Burn, the software development arm of British electronic artist and programmer John Howes.

It follows on from Strokes, which I reviewed most recently in the June 2024 issue of SOS, and shares a similar left‑field, minimalist, modular-inspired character, and is similarly affordable.

If forced to summarise what Stacks actually is, you’d have to say it’s a four‑track looper with an onboard synth, audio input,

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step sequencers, modulation sources and effects. Particularly in its standalone app form, on the iPad, it has sufficient scope to create exploratory long-form pieces all by itself, and perhaps even a longer ‘set’: sequencers and modulators drive the synth, the output of which (or another incoming live signal) can be captured and layered in the loopers.

The same and more is possible with Stacks running in a DAW or other host, in plug-in form (AUv3 for iPadOS, AU for Logic, and VST3 for everyone else on macOS or Windows). However, Stacks is open-ended, and I think it’s likely some

Software Instrument

Stacks is a busy environment visually, but easy enough to understand, with no cryptic labelling. This is the VST3 version, with its resizable window, and with three looper tracks loaded.

users will have it for its multitrack looping alone. It’s also not strongly deterministic, in the way a linear timeline-based DAW is, but more of a platform for experimentation and for generating that old friend, the happy accident.

Stacked A visual scan of Stacks’ main interface is potentially a bit overwhelming at first — it’s a very ‘busy’ design — but the main sections are easy enough to spot. Two step sequencers (one primarily for notes, the other for modulation) towards top left. Synth and effects top right. Various modulation

sources plus mode/view buttons (and, on the standalone app, transport controls) at bottom left. And then at bottom right the four parallel loopers. However, this is only the ‘Main’ layout, and others reconfigure the interface, opening up new panels or increasing the size of existing sections. For example, the Loopers mode increases the size of all four looper tracks and hides the sequencers and the synth almost entirely.

The way elements interrelate remains very clear at all times though.

Such is the depth of functionality in Stacks I dare not try to describe all its features. But I’ll focus in on a few to hopefully give a flavour of the what, how and why.

Let’s take the uppermost step sequencer, Seq A, for starters. Its main role is to send notes to the onboard synth (and, for good measure, across virtual MIDI too). You specify note pitch by dragging faders, with the option to generate single notes or many types of chords.

An eight‑step design might look rather limiting, but these are actually eight ‘stages’, individually adjustable in length and with varying numbers of notes triggering (like a ratchet) within that length.

Ensuing rhythmic subdivisions can range from downright predictable to exotically discombobulating. There’s even a way for the note triggers within a stage to be skewed, for a ‘bouncing ball’ effect.

Equally, you can dial in rests, or skip stages

completely. An overall Steps parameter sets the ultimate musical length of the loops you create here. It doesn’t alter the number of stages but forces a reset to stage 1 every time the whole cumulative caboosh of note triggers and durations plays out, helping to maintain useful musical structures — or not... Finally, Seq A can run forward, backwards, in palindrome fashion, or completely randomly.

Notes from Seq A feed into a pitch quantiser section to the right. This constrains raw note data to any musical key and one of many preset scales (or indeed to a live MIDI input), all of which means you needn’t be deadly accurate with note programming in the sequencer. The quantiser will also alter chord inversions and voicings, apply pitch offsets, allow for control and modulation of velocity and note length, stereo spread, and much more.

Stabs Stacks’ synth is essentially a subtractive design, but has many unconventional tricks up its sleeve. The Oscillator section gives you 26 wavetables to choose from, and within them there’s proper smooth interpolation of waveforms via two control knobs. FM, AM and phase modulation are available, as well as a noise source and sub-oscillator. It’s a potent harmonic palette, unashamedly digital-leaning.

The filter section offers a HPF/LPF combo with either a notch or digital

Cong Burn Stacks £19/£49 pros • A really sophisticated multi-stage sequencer and wavetable synth.

• A brilliant multitrack looping environment.

• Not difficult to learn, despite being very unconventional.

• Good‑quality onboard effects.

• Fantastic value for money.

cons • Nothing important.

summary Stacks takes all the most interesting bits of step sequencing, modular-inspired and granular synthesis, and multitrack looping, and blends it into an exciting, unique whole.

degrader stage, and drive. A ‘rubberband’ option links the cutoffs, for a sweepable band-pass response, or you can switch the whole thing to become a six-band filter bank, with control over the balance and spread of centre frequencies.

The final Amp stage is also unusual.

It defaults to a Buchla-esque Low Pass Gate, which is to say it’s an amp and filter combined, with level and resonance called Ping and Bite. With only a decay envelope it works best for percussive sounds. A Clean mode is more familiar, and offers a fuller ASD envelope. Tucked away here is also a compressor and a really nice‑sounding

While the default Main window layout gives an overview of all four looper tracks, an alternative view opens up all the parameters for pairs of tracks.

www.soundonsound.com / July 2025

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ON TE ST

CONG BU R N S TACK S

chorus with three different depths.

it forwards or backwards at different rates, freezing it in time, and transposing Loops it. These modes are similar, but where Slice I expect for some users keeps generated material the loopers will be the sync’ed to clock, Grain is main draw of Stacks. more nebulous. Neither, There are four of them, incidentally, performs any capable of running sort of transient detection in musically coherent or formal ‘beat-slicing’, sync’ed fashion, or as but with some material it independent entities. can sound like they do.

It’s possible to load All modes offer a lot each track with an audio of creative playback file: mono or stereo WAV, parameters, alongside AIF and OGG worked basic level, pan and for me, not FLAC or sends to the effects.

MP3. They’ll also record So much so that it’s and overdub material For getting non-synth audio into easy to end up with in real time, either from Stacks’ looper tracks there’s an Audio audio looping around Stacks’ internal synth or In option in the Oscillator section. that bears little or no live-input oscillator, or the resemblance to what output mix of the loopers seeded it in the first place. Essentially the themselves, allowing resampling and loop tracks become tone generators — Portastudio-style three-into-one bouncing. granular oscillators. What’s more, there’s By default, loop recording and playback is a way for their pitch parameters to follow quantised to a four-beat bar but that can be the current sequencer pitch quantising, changed to longer or shorter time divisions, so they can be heavily modulated but or indeed quantising turned off altogether. retain musical relevance. Stacks allows Using default settings you can, if you for a drag-and-drop from a looper track to wish, use these loopers in a traditional, a DAW audio track, to ‘print’ its sound, and orthodox way. A MIDI configuration tab, in intriguingly, usefully, the resulting audio fact, is completely given over to setting up file includes any results of modulation. So MIDI control for conventional record, stop, you can use a looper as a laboratory, and play and overdub commands, from a foot capture your work as you go. There’s not controller perhaps. the grain density of a dedicated granular A whole other deeper, weirder character synth, but some pretty remarkable sounds lies just underneath the surface, though. can be produced.

Each looper can work in one of three modes. The first, Reel, recreates both Stacking Up the sound and varispeed characteristics of magnetic tape. Grain and Slice can You know you’re in for a mind-expanding essentially resample material in the loop experience with a Cong Burn product, buffer, and are capable of scanning through and Stacks is no different. It’ll take

Effects & Modulation In addition to the built-in chorus and compressors, Stacks’ delay and reverb are flexible and of good quality. The delay has stereo and ping-pong options and a nifty, elegant graphic that communicates instantly how it’s set up. The reverb is, unexpectedly, two separate reverbs: a clean-sounding algorithmic type and a convolving spring with normal and dark modes. Either one or both can be in play, routed serially or in parallel, but through experimentation I discovered that the springs are essentially preset, and nearly all parameters relate only to the ‘Algo’ reverb.

As for modulation, there are many sources, including two multi-mode LFOs. In those, Wobble mode is almost conventional, with familiar waveshapes, but you’ll also find Flutter, based on the ‘controlled chaos’ Lorenz Attractor principle, and Ripple, which mimics lapping of water waves. The second step sequencer, Seq B, can be programmed for short cycles or longer shapes, while an X/Y pad gives real-time control options. Also, each stage in the main note sequencer has a ‘lock’, allowing it to have radically different synthesis and effect settings from all others around it.

To set up modulation, you click one of the many little ‘dial’ buttons, whereupon all parameters capable of receiving it are equipped with an additional ‘collar’ around their circumference. Modulation depth is then set with a mouse or finger drag, and thereafter parameter values update in real time.

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time — hours, perhaps a few days — to absorb all the possibilities on offer and find fluency, and let’s just say that Stacks might not be ideal for a user taking first tremulous steps beyond GarageBand, for example... But if you’re an experimental type who relishes getting stuck into a Eurorack case full of funky modules, or an out-there Max for Live patch, this is going to be right up your street.

For what is such a unique instrument, clearly a labour of love by a brilliant, free-thinking developer, I hardly feel I should criticise anything in Stacks.

But there are a few things... Even on an 12.9-inch iPad Pro the user interface is so crammed that it’s sometimes difficult to operate. That’s especially true for knobs and controls right near the edge of the screen, which can end up triggering system gestures if you’re not careful. Testing the VST3 (in Studio One 7 on macOS) some pop-up menus took a few clicks to open (though I would fully expect glitches like this to get ironed out). Amongst the multitude of oscillator waveforms I could see no way of generating simple narrow pulse waves, and hence any kind of PWM tones. The reverb control scheme is a bit odd currently: workable, but just not very intuitive. I also wish the loopers’ Reel mode Pitch control wasn’t quantised to semitones, to allow for really gloopy pitch‑bends. Finally, I’d love to see a looper track parameter reset, so it’s possible to return to normal playback after localised, temporary destruction, as it were. The current Recall button does something else.

These are all rather trifling concerns, however, and in every other way there is absolutely masses to enjoy in Stacks, whether you’ll use all it has to offer or only bits. Interesting things seem to flow out of it whether you’re approaching it as a completely self-contained creative ecosystem or just as a scratchpad to generate or deconstruct material to be assembled elsewhere. The loopers alone, with their ability to support both utterly mainstream and many radically creative approaches, are worth the price of admission. Yes, Stacks is unusual and idiosyncratic, but it’s also brilliant and unique, inspired and inspiring in equal measure. £ Apple/Windows £49.99, iOS £19.99.

Prices include VAT.

W www.congburn.co.uk

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ON TE ST

Xvive More You

USB Audio Interface

Xvive’s ingenious modular interface puts foldback in the hands of the performers.

ROBIN VINCENT

T

he Xvive More You is one of those rare finds: an audio interface with a unique feature. It has the uncommon ability to expand and daisy-chain itself into a whole band’s worth of personal monitoring and recording. While the name may make it feel playful and toy-like, the reality of the system is far from it, and the more you get into it, the more problems it appears to solve.

The More You is about collaboration, and it manages to remove a lot of the stress and chaos you often find around recording and rehearsing with other people. It’s perfect for ad hoc situations or home

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and project studios that don’t have much hard‑wired in.

DAW Recording Scenario Typically, when you want to record with a few musicians into a DAW, you’ll have one audio interface that everyone has to go through. There will be lots of cables dragging about the place, you’ll be fighting over headphone sockets, and only one person can really see what’s going on.

It tends to feel fraught and fussy and can be more about getting things to work than making music together.

The More You system releases that tension by pushing the audio interface out into your room. Starting with the two‑in/

two‑out Hub, you chain together up to eight More You Hubs, or the simpler two-channel 2X units, so that everyone has one. Your guitarist, bassist and vocalist all have their own connections, right next to them, in a little box, with a pair of inputs, controls and headphone sockets. The main Hub

Xvive More You £299/£229 pros • Brilliantly expandable recording system.

• Gives each musician their own connections and monitoring.

• Reduces cable clutter in multi-musician settings.

• A complete rehearsal monitoring system.

cons • You’ll need an extra power supply.

• No loopback driver.

• Currently no remote mixing.

summary The More You system could radically simplify the recording and rehearsing of bands or other scenarios involving multiple musicians. It removes the drama and replaces it with solid connections and happy players.

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is still the primary interface handling all the input gains and level control, but you can use the Mix knobs on your own box to blend your input with the mix locally in your headphones, which is really all a musician is interested in. The other knobs add reverb per channel, again locally. The brilliant thing is that neither of these affects the mix or what anyone else hears. So, your singer can drown themselves in reverb, and the guitarist can make their monitoring all about them, and no‑one else has to care.

As for the actual recording, the Hub supports up to 24 inputs; two of its own on the front with combi XLR connections and an ADAT port on the back for eight additional inputs. The remaining 14 inputs can come from any combination of Hubs or 2X boxes.

The boxes connect via a regular XLR cable plugged into a digital port that carries power and multiple channels of digital audio. They daisy-chain from one box to another rather than spanning out from a single box, which can massively reduce clutter and tangles. With a maximum cable length of 25 feet you can occupy quite a large space.

The system works like a shared bus, so no additional latency is added when you connect another box. Audio arrives at each 2X box within one sample cycle, and if they’re all set to direct monitoring, then there will be no appreciable latency. Like any audio interface, if you want to monitor through the DAW, which you can choose to do with any input, then there will be the usual round-trip latency. Still, I had it

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ON TE ST

XVIVE MORE YOU

When lit red it is sending only the output of the DAW. The headphone mix remains unaffected.

2X Expanders

You can daisy-chain up to eight More You Hubs or 2X expanders using the XLR ports on the back of each unit.

working down to a couple of milliseconds without any trouble.

There’s no software mixer or routing app to pull up; it’s all done on the hardware. This means that whether you have a computer connected is largely irrelevant because the mixing and monitoring is the same either way. The only slight disappointment is the lack of any loopback driver for bringing the output of other software into the DAW, which has become a very common scenario.

Using The Hub The Hub is a solid and chunky desktop box with two XLR/jack combi inputs, stereo line outs, a headphone out, a display, and a friendly bunch of roomy knobs set on top. It ticks all the right boxes with audio up to 24-bit/96kHz, MIDI I/O and an ADAT port for eight additional inputs. It’s weighty, covered in rubberised protection, and has an integrated mic stand mount hidden behind a panel on the side.

The top section of the Hub contains everything you need to control the entire system. You have a little screen, two clickable encoders and a handful of buttons. The top encoder switches between Gain mode and Mix mode, and selects which channel you wish to control.

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The bottom encoder changes the value.

For every input, the Hub can offer up to 60dB of gain, phantom power, polarity inversion, a high-pass filter, a Hi-Z switch for instruments, and direct monitoring.

Any connected boxes appear as a pair of channels on the Hub display. These are usefully colour-coded so you always know you’re adjusting the right input. Then you simply go from musician to musician, setting the gain for their instruments and microphones. You can also use the very welcome Autogain feature. Just get everyone to play, and with the tap of an encoder, the Hub will do a decent job of setting the gain up for you. It’s the sort of feature that blasts through all the difficulties of getting lively, creative people to set things up when they just want to play some music.

In Mix mode, you use the encoders to set the monitoring levels and panning for each of the inputs. This is the mix that everyone will hear in their headphones and against which they can balance their own input signal on their own box.

The last button on the Hub is the monitor switch. When the switch is lit blue it is sending the whole headphone mix, including playback from the DAW, to the stereo output on the back of the Hub.

The 2X boxes are the same size and shape as the Hub but with some very key differences. The 2X has no outputs other than the headphones and no USB port for computer connection or power. All of the audio data is streamed directly to the Hub and power comes down the same cable.

Otherwise, you have the same two inputs, mix and reverb knobs, a big headphones control and an intriguingly large Talk button.

The Talk button is a talkback circuit designed to let you communicate with the other musicians via a little microphone recessed into the front panel. Just push the button and everyone can hear you ask for more cowbell. It’s a very useful feature; however, it’s so noisy and loud that it’s more likely to be used to abuse your fellow band members.

The 2X is not an audio interface in its own right and can only operate connected to a Hub and so you’d buy them with the express purpose of expanding your system. Xvive have two larger expansions coming soon, called the 4X and 8X, which probably need no further explanation. One rather neat feature is that each Hub can also act as a 2X box. So, if all your musician friends owned a Hub for their own systems, they could bring them along to a session and plug them all in together without anyone having to fork out for an expander.

In this mode, they operate simply as a two-in interface and lose the ADAT input and mixing ability. However, Xvive told me they are working on firmware that will allow anyone with a Hub to create their own local monitoring mix while the primary Hub handles the gain and overall mixing to output.

Power None of the Xvive units came with a power supply or USB cable, and this threw up some unexpected problems because the power requirements are very specific.

When you plug your Hub into your computer, you get a readout that tells you how many expanders it can support. On my slightly ageing desktop computer with regular USB-A ports, the Hub reported that it couldn’t support any 2X units at all, leaving me sitting here with two expanders and a spare Hub and no way of using them.

When I moved to my Surface Pro 9, which has USB-C connections, the

Hub gleefully told me it would support a single 2X unit.

So at least I could test out the concept. To support more units, you have to provide extra power, which you can do via a second USB-C port on the back labelled ‘power’.

However, nothing I possessed in my home or studio could summon up enough power to convince the Hub that it could support more than one 2X. It turns out that what you need, ideally, is a 65W USB-C supply like the one that comes with a MacBook Pro. Xvive explain all this in detail on the website and in the manual, but it is slightly frustrating to have to buy a whole power supply just to use the system as intended.

Ag

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Rehearsal Scenario Once I had everything powered up and got the system to work in the relative calm of my studio, it was time to introduce it to the chaos of other people. Xvive gave me a few of these boxes, and so I took them to a band rehearsal.

I had it all set up, plugged in, attached to mic stands and ready to go before the bassist returned with a sandwich from the shop up the road. There was no computer involved this time, but everything worked in exactly the same way. When we were all ready, I hit Autogain to set the levels and we were off.

As a band, we’re used to playing with in-ear monitoring via a Soundcraft Ui24R system controlled by an iPad app.

It works, but it’s always a faff. The iPad never seems to be charged, no‑one can remember the login for the mixer, the personal in-ear monitoring boxes are always running out of battery or getting trodden on, and there are cables absolutely everywhere. In comparison, the More You system was a refreshingly smooth, tidy and competent experience.

With the Hub clipped on to my mic stand, I could get to all the controls, set up the general mix and then adjust my own level via my own interface, all with my guitar hanging around my neck. The others did the same without any fuss or mess, and no‑one had to ask for any adjustments. It was an absolute breeze.

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Conclusions Xvive have really dug into the challenges of playing and recording together and come up with an elegant and robust solution. While expandable interfaces are not entirely new, the way the More You system does it is pretty unique and solves more problems than simply offering additional channels. The forthcoming 4X and 8X units look really useful and will let you do things like wire in your tower of synthesizers via a single box with a single cable. And if you wanted to go off and do some music on your laptop, or drop round a friend’s for a jam, then you just need to take the perfectly portable Hub. The technology is solid, and the ease of setup and the practicalities of use as a monitoring system made for some very happy bandmates, and that in itself makes it a winner.

If you’re in a band or often have people round to record, then the More You system is definitely something to be reckoned with. £ More You Hub £299, More You 2X £229. Prices include VAT. W www.jhs.co.uk W www.xvive.com

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www.soundonsound.com / July 2025

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INTER VIE W

CARO C

I

Producing Drum & Bass

Lincoln Barrett’s musical journey started in the box, but 20 years on, he’s reinventing his trademark liquid funk sound with vintage hardware.

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nspired by the original pioneers of jungle and drum & bass, Lincoln Barrett aka High Contrast found his own way into production. Since signing to Hospital Records in 2000, he has become one of the genre’s biggest names. Drawing on an early obsession with film soundtracks and sound design, he developed a signature liquid funk sound, pumping out anthems with epic vocals and drum & bass remixes of household names like Adele, Kanye

cinematic and haunting. But there was no context to it for me, I was coming at it completely blank.

“When I was 16 I went to college, and friends I met there were already into jungle and buying vinyl. I realised there was this whole scene where people are making music, often in bedrooms with very simple setups, and they’re sampling films that I love, and people are actually dancing to it. I felt this was tailor-made for me, because the one thing I do know about is movies.

“At the same time, I got a free CD-ROM on the cover of a computer magazine with a demo version of Cubase. I didn’t know anything about music production, but it did allow me to throw some samples together and start very basic explorations of making drum & bass. This was all at the start of the Internet, so I was able to download things like the Amen break and bass sounds but there was no real sense of an online community around drum & bass for me.

“It was only when I was 18 and I was able to actually get into clubs that I heard sub-bass for the first time! Back then the sub-bass was just pure sub 808, so there was no way I was going to hear that on my home stereo system. I think that has stuck with me, because I focus more on the melodies, the samples and the sounds rather than the bass and even the drums sometimes. That’s kind of my contrarian approach to things within drum & bass. I actually try and de-emphasise the drums and bass element!”

Out Of Step

West, White Stripes and Coldplay. Even nowadays, a new High Contrast track just has to be premiered on BBC Radio 1 on a Friday night for audiences to be singing the track back as he DJs the next day in a massive club. His recent album Restoration is heavily inspired by the old Roland and Akai hardware samplers used by the hip-hop pioneers.

Sound & Vision “I had no interest in music until I was about 16,” admits Barrett. “Certainly not contemporary music. I was interested

in film from the earliest possible age, so I collected film soundtracks. My sister got me Goldie’s Timeless album and I thought ‘What the heck is this?!’ I’d never heard jungle before. It was the most alien thing, and I couldn’t get my head round it. So I was immediately turning it off but kept going back to it, trying to work out exactly what this music was. Hearing those cut-up, sped-up drum loops being edited with a lot of sound effects. I also noticed there were sounds in there that I recognised from films, and then the strings were quite

“I started very basic production around 1996, 1997 without any help from anyone. It took me a long time to figure out what I was doing, but by around 2000 my tunes were starting to sound... not professional, but they were sounding interesting, I think because drum & bass was so dark at that time.

There was the whole techstep era in the late ’90s and I really didn’t connect with that. At the same time, I started to listen to house music, specifically Chicago house, the deep house sound, and New York garage. I loved all those sounds. At the same time there was Daft Punk happening and the whole French filtered house thing, which I loved. And I fell in love with disco. I wondered, why aren’t we getting this in drum & bass?

So it kind of became my mission to try and do that whilst also tying it in to

www.soundonsound.com / July 2025

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INTER VIE W

HIGH CONTRAST

For many years, Lincoln Barrett created music entirely in the box. When he decided to embrace hardware, he chose Isla Instruments’ modern recreation of the classic E-mu SP‑1200 as his main sampler.

hip-hop — the way you would have those funk, jazz and soul samples in hip-hop.

“I was studying film at university but the music took over my life. I was being very prolific, knocking out track after track and honing the craft. It wasn’t club-ready so people didn’t want to sign it, but they could see that I was making this kind of groundbreaking stuff. The liquid funk movement was starting to happen, as pioneered by DJ Fabio. Then it just so happened that London Electricity played in Cardiff and they ran Hospital Records.

I gave them a demo when they played at the club because I was a resident DJ at the club. We just hit it off because I was playing soulful drum & bass and they were too. They listened to the demo and then they just rang me up and said ‘We want to sign you for an album deal.’ Going from a complete unknown to getting signed for an album was just fantastic.

“That immediately threw me into the scene, getting my stuff out on vinyl. Then that led to gigs and doing my first international gigs, and that led to a residency at Fabric and then getting played on Radio 1 by Fabio. It all happened very quickly off the back

of that. And I also like the fact that they signed me for an album, because usually in dance music, and especially in drum & bass, the focus is on singles.”

Learning By Doing At the same time, Barrett was still teaching himself the art of production.

“I guess it’s good to have mentors and people, but I think nothing can beat figuring stuff out yourself and finding your own way of working. Like my first album. The production was very raw,

more interested in repetition and loops, finding the moment of something that I love, and repeating that. I would say it’s somewhere in between like proper songs and what other people normally do in drum & bass.

“I think the other thing that creates that distance from normal songwriting is that I love vocals, but I try and use them just as another element in a track, and not give them the kind of dominance that a proper song would have. I treat the vocal as if it’s a Rhodes keyboard or the bass. There’s been some repeat uses of vocalists over the years, but generally I’m trying to find a new sound each time. But the majority of what I do is sample-based, so that I’m sampling a track first of all, whether that is an a cappella or from a full song.

Back in the day, I had to find creative ways of isolating a vocal sample from the rest of the song, whereas today, there are AI tools that will take vocals out for you cleanly. So most of what I do starts off as a sample of something else, and then I will get a vocalist who can accurately recreate it. So that is usually the determining factor.

Lincoln Barrett: “I built my whole career on sampling and yet I’d never used a sampler!”

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some of the tracks could barely get mastered, because the production was so raw. But it just reinforces my philosophy, which is: if the ideas are good enough and if it makes you feel something powerfully enough, then the production almost doesn’t matter.

“The way I look at it is I don’t actually make songs, but I make things that kind of have the impression of a song.

There are a lot of people making drum & bass now who are getting full vocals and it’s proper song structure. But I’m

“For me, it’s finding someone who sounds like the the existing sample. And that’s a very specific skill set. Not every vocalist wants to do that. There are people who specialise in that and there’s a company called Replay Heaven who are just incredible at recreating samples and do it to a world-class degree, to the point where often I think the recreations actually sound better than the original samples. Doing it that way, I also get the stems of the sample. So I have so much more freedom in exploring that musical idea than if it was just the sample.

Sometimes I will, however, maintain it as the solid sample to keep the integrity of the idea of sampling. I think sometimes that limitation of just having the sample and not having the separate stems has a particular sound in itself. I am a believer in limitations as being actually essential to creative processes.”

Coming Around Again Restoration is the seventh High Contrast album, and marks a conscious return to Barrett’s roots. “At the outset, the Restoration album was me wanting to get back to basics and go back to the liquid funk sound that was introducing those soulful and house elements to drum & bass. I always wanted to come back to this liquid sound, but I knew I needed to leave enough time. I think 20 years is actually a good amount of time when it comes to cycles within music. I think it’s enough time that there’ll be a new generation of listeners who aren’t aware of the original sound.”

The album also saw Barrett take an old-school approach to production — which, ironically, wasn’t the approach he took first time around. “When I started out, 25 years ago, I was all in the box and I didn’t use any hardware. Still to this day, I’ve never used a proper mixing desk.

I have a little mixer but I’m not using it for mixing. It’s just literally to colour the sound, running the whole mix through it. It wasn’t until almost 10 years into releasing music that I bought one synth that I occasionally used. So it’s relatively recently that I’ve started to get more into using hardware, outboard gear and things like guitar pedals, as a way of challenging myself and wanting to try something different with my production. I built my whole career on sampling and yet I’d never used a sampler! It was a eureka moment about six years ago when I thought I should really get a sampler

and have a go with that, because I love sampling. And so then that’s started a whole process and chain of events with getting into different samplers.

“I got an old Akai S950 that would have been used in the early days of jungle and drum & bass. I was going back to the roots of trying to get that sound. I like the sonics you get from it, even though using floppy drives is painfully slow. So I started looking at other samplers and I got the Isla S2400, which is a new recreation of the old E-mu SP‑1200. The SP‑1200 was something I was very interested in getting, because Daft Punk used it on their first few albums, and of course it was used for lots of classic hip-hop.

But it’s really expensive. People I know who already had one said it was painful using it. It’s slow with the disks and you only have two seconds’ sample time.

So that put me off it. With the S2400, you’ve got all the modern conveniences of USB connections and much longer sample time, but it does have the 12-bit, 26kHz sample rate, so you do have that authentic old sound from the SP‑1200, where you get the aliasing on the samples and that lo-fi quality.

“So, I started making hip-hop beats on this sampler. I was very influenced by people like DJ Premier, RZA and ’90s hip-hop. It made sense to me to make this liquid funk drum & bass album using this sampler, as it ties in with the hip-hop

feel and the French house thing as well.

Having that as the focus of the album helped galvanise it and give it a particular sound and a direction for me to go in.

“I wanted to keep it sample-based because I think in the digital age, you’re fighting against infinity. So I was trying to take the approach of making tracks with as few individual tracks as possible, and building things around a sample.

I’ve got the sample: how many ways can I mess with this sample across the length of a piece where it feels interesting and the track is evolving? But I’m not actually adding new things to the track. Instead of ‘Here comes the string section!’ or a synth solo, I was trying to take a sample and then blow it apart to little pieces and rearrange it to try and find new sections for the song. J Dilla was the master of doing that.”

Ideas Man Barrett’s writing process is unusual for the genre in that it’s somewhat ‘top down’.

“I think most people in drum & bass start a track by building the drums and then the bass line, and then it seems that other elements such as a vocal or the melodic parts are almost an afterthought, in response to the growing base. I mean, it does make sense because the genre is called drum & bass! So I’m not knocking that approach. But for me, I don’t think I’ve ever made a track where I started with the drums or bass. For me, it’s

This small mixer is used mainly to colour the sound, rather than for mixing in the conventional sense.

www.soundonsound.com / July 2025

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INTER VIE W

HIGH CONTRAST

always the musical idea that is the key.

I try and make the drums and the bass fit around the musical idea and complement it. A lot of the time I’m using sampled drum breaks. I love the texture of them, there’s just something magical about the sound of those ’60s and ’70s drum breaks from the funk and soul records.

There’s something about the recording of those drums. There is a reason why they have been used so much in hip-hop and drum & bass, and that people still use the Amen break to this day, because it’s lightning in a bottle. It’s magic.

“For this album specifically, what with it having this sampled feel and a lot of soul samples, it just made sense to go back and use some of those classic drum breaks. But I am layering those with one-shots. There’s generally at minimum 10 drum tracks on one of my songs. To get the weight that you need in a drum & bass track, especially at the moment, it does need this kind of very careful balancing of drum layers and a lot of saturation. Clipping basically has become the name of the game. Everything is clipped to the nth degree to squeeze every bit of loudness out of it. I try not to push it to the point where it sounds uncomfortably squashed. I try and get the balance where it still feels kind of pleasant to listen to, but then also has the weight where it will compete with other drum & bass tracks in a DJ set.”

Although hardware samplers have become central to the High Contrast production approach, software is also important. “I am generating ideas sampling vinyl or my little modular rig into the Isla S2400 sampler, usually at the classic lo-fi setting of 26kHz/12-bit. Once I have a groove or main loop, I record it into Ableton Live via the Prism Sound Lyra 2 interface. After adding drums and bass in Ableton and doing some arranging and processing, I’ll often then send the sampler recordings and other parts made in the DAW back out via the Lyra into my patchbay, where I run the audio through some effects, which is often another sampler like the Roland SP‑303 or the Zoom Sampletrak, which are both handheld devices from about 25 years ago and have great effects, or through guitar pedals or a Roland Space Echo. This signal then goes back through the Lyra into Ableton.

“With vocal recordings, I find the takes I want to use in Ableton and then send them to the S2400 where I lower the

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Since embracing hardware, Lincoln has built up a small collection of units including (from top) Pioneer SR-202W spring reverb, Samson patchbay, Alesis Microverb reverb, Oberheim Matrix 1000 and E-mu Vintage Keys synths, Alesis 3630 and SSL compressors and Louder Than Liftoff Silver Bullet preamp.

quality to 26kHz/12-bit and usually do any pitch-shifting I want there. I find the pitching done by hardware to be sonically more pleasing most of the time than doing it in a DAW. Then I run that through maybe the echo unit, either with delay or dry just for some saturation, and record that back into Ableton. Once the track

is basically done, the main mix will likely get a pass through my little Mackie 1202 mixer for a touch of saturation. There might be an irony in using something as high fidelity as the Lyra interface when I’m using samplers at really lo-fi settings, but my philosophy is that I want to record at the highest possible quality, even

Remixing At the time of writing, Lincoln Barrett has just finished remixing his Restoration album into what he describes as a “lofi hip-hop‑style mixtape”, and says that “Remixing is fun to me, because the hardest part has been solved. If you love the original song, then you’ve already got a great melodic idea there, so it’s just fun then playing around with that and arranging it.

I’m not sure what the total count is, but it is a big part of what I do. I almost approach everything as a remix, because I’m sampling so much that you could argue that my original tracks are remixes in some way. Especially in my early productions, when I was just learning the craft, I was effectively remixing songs.

“One of the huge appeals of drum & bass to me was the drum & bass remixes of non-drum & bass based songs. The big one for me was DJ Zinc’s bootleg remix of the Fugees ‘Ready Or Not’ [1996, white label]. That track just blew my mind when I heard it, it was so magical especially because it was

and especially when the audio source is very lo-fi, because then the sound of the original gear is preserved and the audio is coloured as little as possible by the recording process. This feels important when going back and forth between hardware and software, potentially multiple times.

“If I’m cutting up a melodic sample on the sampler, I’ll record in the pattern that I’ve played on the sampler so that it’s recorded as audio. Then it’s like having a guitar performance and I can tweak it from there. Working with recordings of sampler patterns is a good way of making me commit to an idea.

“I am wary of nostalgia and being too tied to the past, and I’m not interested in just recreating the past, but I think we can take things from the past and try and reimagine them for today. I think you have to respond to the moment you are in as well, you know? So I like to kind of come up with these rules or limitations for making things. But at any given moment, I’m willing to abandon the rule if I think a better idea comes along. I guess I have my cake and eat it, because I’m using old samplers and some old things. But then, if something’s bothering me, I’ll fix it in Ableton or use some high-tech tool. I am fine with that because it’s the end product that is the most important thing. At the end of the day, nothing beats your ideas, your emotional responses to things.

“When you’re doing something outside of the historical moment, it’s just not going to hit in the same way. Like you know, when Daft Punk were doing that side-chain house thing, 20-odd years ago, that was new and exciting and it was something that naturally happened through their process. But like now kind

a bootleg. I loved the original Fugees track, but then this drum & bass version took it to another level. I really struggle remixing drum & bass tracks, so I almost always turn them down now, because a big part of the appeal to me is taking something from another genre and turning it into drum & bass. Again, I like that contrast of pop with drum & bass.

“Drum & bass still is quite an abrasive music. A lot of people just cannot handle the tempo of it and the relentless nature of the drums and then the sub‑bass kind of blowing your head off constantly. The perfect example to me is like the Adele ‘Hometown Glory’ mix, where the original song is just piano and vocal, and then a little bit of strings come in. You’ve got this very kind of gentle, basically acoustic track. But it just so happened to be around drum & bass tempo, so it was a natural fit for me to put some slamming drums and bass over the top of it. I love the contrast of the the softness of the musicality and then the hardness of the drum & bass part.”

Restoration is High Contrast’s seventh album, and leans heavily on vintage sampling technology.

of fetishising it and recreating it, it’s like, well done. You’ve made something that sounds exactly like Daft Punk from 25 years ago. So that’s kind of like my thing with jungle. I’m playing with those ideas, but it is also me today making something and hopefully it has my personality in it and whatever historical moment I’m in.”

Forward Motion Drum & bass as a genre has demonstrated an impressive ability to survive and evolve without losing its distinctive qualities. To conclude, I asked Lincoln Barrett where he sees it developing in the future. “Since the lockdowns, there’s been a real explosion of female DJs. It’s great to see that things have opened up because it did feel like such a boys’ club for so many years. I think it can only be good for the continuation of the genre that we have more diverse voices in it. There are people who have blown up from those lockdown streams and now have careers as DJs. I think we still need more female producers coming

through. It’s great that there’s EQ50 [a collective of women working towards fairer representation within drum & bass] doing their mentorship schemes.

“I don’t know if it’s even possible, but I’m interested in what other ways of working can there be. Traditionally, dance music is made by one person working alone, just focused on that one task. Can the whole approach to music be reimagined? That’s what I would like to see from the new blood. Different perspectives. I’m an old man now, so I’m quite stuck in my way of working in isolation. But I don’t think everyone actually enjoys that. I’m intrigued to see if people can invent their own way of making music, maybe in a more collaborative way or something. That could be a path that leads to new sounds and maybe new genres. And I guess technology would be aiding that.

“The AI question is going to be a major thing to consider, just seeing how that plays out. If people can just type ‘High Contrast’ into an app and they get hundreds of tracks generated that sound like me, it could mean that we will be stuck with generating existing artists and no one can break through.

Or maybe people will completely reject AI-generated music because on some incalculable level, it just lacks the humanity that people can connect with.

I really don’t know how all that is going to play out. I have no problem with AI tools, because they are being used in service of the human trying to create something.

But when you’re delegating everything to a large language model to create something for you from a text input, that’s just less interesting to me. At the moment, it still feels like the AI music I’ve heard is just not good enough.”

www.soundonsound.com / July 2025

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SPOTLIGHT

Multi-Tier Keyboard Stands

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Fortunately, there are plenty of stands around that aim to solve such issues, ranging from simple two-tier solutions to large frame systems designed for those who are blurring the lines between a studio and a museum!

Gator Frameworks

Gator’s Frameworks family includes stands, clamps and holders for just about every bit of kit you can think of, and keyboards are no exception. First up is the GFW-KEY-4000, a two-tier column-style stand aimed at both studio and stage use. Its heavy-duty frame is capable of supporting up to 54kg, and both tiers are height-adjustable, allowing the stand to be configured for seated or standing players. Helpfully, the top of the central column features a mounting point that can be used to

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attach an included microphone boom arm. The GFW-KEY-5100X then offers a double-braced X-style stand that comes fitted with two tiers as standard, with a third available thanks to an optional add-on — the standard configuration will support up to 79kg, while the additional tier will hold an extra 34kg. Finally, there’s the GFW-KEY-8500, a framed design that includes three height-adjustable tiers and will support up to 58kg. All of the models in the Frameworks family are fitted with ergonomic fittings that have been designed to make setups, adjustments and pack-downs as efficient as possible. £ From £98 including VAT. W www.gatorco.com

includes two pairs of height-adjustable arms, and includes a threaded mounting point for attaching a mic boom or other accessories. £ From £79.99 including VAT. W www.gear4music.com

Gibraltar Double Keytree

Gear4Music

Gear4Music’s own range of hardware includes three multi-tier keyboard stands.

The most basic is the X Frame stand, a foldable design that will support 30kg on its first tier and 15kg on its second, while the table-style Deluxe 2 Tier is an adjustable solution that, thanks to a 105kg rating, will support even the heaviest 88-note pianos along with an additional keyboard. The company also make a column-style stand that

Gear4Music Deluxe 2 Tier

Gibraltar

Gibraltar are a well-known name in the world of drum hardware, but their range also includes a trio of keyboard stands, two of which are designed to accommodate a pair of instruments.

The smaller of the two, the Keytree 2, boasts a lightweight yet sturdy design built around a curved central column, with the frame held securely in place by the company’s substantial memory-lock clamps. The Double Keytree 2, as its name suggests, doubles up the design with a pair of legs joined by horizontal braces, offering enough support for large 88-note keyboards. Helpfully, both designs break down into multiple parts for transportation. £ From £638.50 including VAT. W www.gibraltarhardware.com

Hercules Stands

Gravity Stands KS MKS 01 B

Gravity Stands

Gravity Stands have a few options in their range, starting with their KSX line-up.

Both the KSX 1 and KSX 2 — X-frame designs capable of supporting up to 55 and 65 kg respectively — can be fitted with the optional KSX 2 T attachment, which adds an angle-adjustable second tier capable of holding keyboards weighing up to 15kg. For those with larger instrument collections there’s the KS MKS 01 B, a rack-style stand that boasts adjustable height, tilt and arm spacing for each of its four tiers.

It features a rugged but lightweight frame that’ll support up to 25kg on each tier, and comes supplied with an addition set of stopper attachments that ensure even the largest, heaviest instruments are held securely in place. £ From £72 including VAT. W www.gravitystands.com

Hercules KS210B

The Hercules Stands line-up currently includes two models capable of accommodating a pair of keyboards. The first, the KS210B, falls into their X Stands range, and boasts a compact, foldable design that can be securely fixed at four set heights and will support up to 120kg — 80kg on the lower tier and 40kg on the upper tier. The KS410B then combines a pair of Z-shaped supports held together by a set of locking arms, and will happily accommodate up to 130kg on its lower tier, and 40kg on the upper tier. Both models feature fast and secure setup and adjustment thanks to the company’s patented EZ-LOK system. £ From £85 including VAT. W www.herculesstands.com

Jaspers 170-4-120RB

K&M

K&M are well known for the high quality of their stands and mounting hardware, and there are a number of options in their range that cater to those with multi-keyboard rigs. Although their X-Stand series is mainly populated by single-tier designs, both the 18930 and 18990 models can be fitted with the 18941 Stacker. This is an optional add-on additional tier that has full height, depth and tilt adjustment and the ability to support instruments up to 20kg — in addition to the original stands’ 50kg. The Omega, Omega Pro and Omega E then offer table-style designs capable of supporting up to 80kg, and can be combined with Stacker attachments that add an extra tier rated at either 15 or 25 kg. All three models are height-adjustable, and the Omega E comes fitted with an electric lift. The company’s take on the column stand comes in the form of the Spider Pro, which will support up to 105kg, comes with two tiers as standard and can be fitted with a third set of arms if required. There’s also the Baby Spider Pro, a packed-down version that can accommodate up to two keyboards weighing a total of 70kg. Both feature a threaded mount for securing a mic boom arm or other mounting accessory. As you might expect from K&M, there’s a huge range of optional extras

Jaspers

If you’ve got a large collection of keyboards to house, then it’s well worth taking a look at the Jaspers range.

Their collection of stands ranges from single- to six-tier designs, all of which are available in a selection of widths and are fully height- and angle-adjustable thanks to a versatile frame and clamp system. They also offer a selection of square-framed racks that are available in four- and six-tier versions, again with a fully adjustable design. Both the stand and rack offerings are rated for between 15 and 40 kg per tier depending on their height, and the company also manufacture a range of accessories that makes it possible to attach the likes of mic booms, tablet and laptop holders, music stands and more. £ From £158 including VAT. W www.jaspers-alu.de

K&M Spider Pro

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Liquid Stands Workstation Three-Tier

that allow users to kit their stand out with laptop or tablet holders, keyboard trays, mic booms, cable management add-ons and much more. £ From £103.60 including VAT. W www.k-m.de

Liquid Stands

There are two styles of multi-tier stands in Liquid Stands’ product line-up, both of which are available in different configurations. Their Z Style range includes a two-tier model with an adjustable height range that caters for seated or standing playing, and is capable of supporting up to 113kg on its main, lower tier plus an additional 27kg on its upper tier. There are three versions available, with the main model joined by a variant with adjustable arms that allow the top tier to accommodate deeper instruments, and a version fitted with locking wheels. The company’s Workstation range then comprises two square-frame designs that are height- and width-adjustable, and come supplied in two- or three-tier versions. £ From £144 including VAT. W www.liquidstands.com

On-Stage

First up in the On-Stage range is the KS1365 Z, a simple two-tier stand capable of supporting a substantial 180kg, 27kg of which can be placed on its upper tier. If you need something more portable, then there’s the KS7365-EJ, a foldable version of the Z-frame design, or the KS7292, an

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X-frame stand that’ll support up to 145kg.

The KS9102 then offers On-Stage’s take on a column stand, and features two pairs of height-adjustable arms that allow the stand to hold up to 56kg on each tier. £ From £109 including VAT. W https://on-stage.com/keyboardpiano

Quik Lok

Quik Lok’s extensive range includes a selection of multi-tier stands, starting with the T-22, a simple X-style design with an adjustable second tier and a maximum capacity of 81kg. The QL-742 and QL-723 then offer two- and three-tier designs that will each support up to 90kg, while the two-tier QL-642 and three-tier QL-623 will each accommodate a more substantial 120kg. If it’s a column stand that you’re after, then Quik Lok’s answer is the SL930, which comes with two pairs of keyboard arms that can be fixed at 11 heights and five different angles. The Z-726 and Z-726L, as their names suggest, feature a Z-shaped frame with an adjustable second tier — both will hold up to 113kg, but the latter is a wider stand designed to support larger keyboards.

Finally, there’s the MKS/4, a four-tier A-frame design that’ll support 40kg on its base level and up to 20kg on each of its upper levels. £ From £99 including VAT. W www.quiklok.com

On-Stage Three-Tier A-Frame

RockNRoller RockNRoller specialise in carts designed to transport your existing music gear, but one of the latest additions to their line-up is a two-tier keyboard stand aimed at gigging musicians. The RCK-1 Compact Keystand Keyboard Stand is made from a lightweight but rugged steel frame, and will support up to 90kg on its first tier and up to 34kg on the angle-adjustable second tier. The frame is height-adjustable to allow for sitting and standing playing, and there’s the option of

Quik Lok Z-726

RockNRoller RCK-1 Compact Keystand

adding a mic boom arm or a shelf kit that can be used to accommodate mixers and other gear. As the focus is on gigging, the RCK-1 can be packed into a compact carry case, and the company say that setup and tear-down only takes around 15 seconds! £ £115.90 including VAT. W www.rocknrollercart.com

StandTastic

There are three models to choose from in the StandTastic range. The first, the 122KS, is a two-tier stand that comes

supplied with four adjustable-length arms that can be fixed at a range of different heights and angles. The 102KS then offers a similar design but with a taller frame, while the 103KS combines the taller frame size with a third tier.

All three models boast a lightweight design, and can be folded down for easy transport — all models are supplied with a carry bag. The company also offer the 100TK, an add-on attachment that can be used to kit any of their stands out with an additional tier. £ From £225 including VAT. W www.lm-products.myshopify.com

Stay

StandTastic 103KS

There are a few options in the Stay range, all of which are column-style designs. First up is the Slim 1100/2, a lightweight two-tier stand designed for smaller, lighter rigs. It’ll support up to 10kg per tier, and is also available in a ‘C’ variant, which features a curved top tier designed to accommodate smaller devices. Stepping up to the Intruder 02 gets you a similar lightweight design but with enough support for 76-note keyboards weighing up to 60kg, while the more substantial Tower 1300/2 takes the maximum capacity up to 120kg.

For those with substantial full-sized keyboards there’s the Piano 1200/2, which boasts a double-braced design and will happily support rigs weighing up to 130kg. £ From £149 including VAT. W www.staystands.com

Stay Piano 1200/2

Ultimate Support The first three options from Ultimate Support fall into their IQ-X series.

The IQ-X-1000, IQ-X2000 and IQ-X-3000 are all single-tier stands in their default form, but they can be turned into two-tier designs thanks to optional IQ-X-200 and VSIQ-200B attachments. In the Apex series you’ll find the AX-48 Pro, a portable column-style stand with two sets of keyboard arms that’ll support up to 113kg, along with the AX-48 Pro Plus, which comes supplied with Ultimate Support Apex AX-48 PRO a mic boom arm and a carry bag. Finally, there’s the V-Stand Pro, an ultra-compact folding stand capable of supporting up to 68kg, which can be fitted with an additional tier using the optional VSIQ-200B attachment. £ From £70.06 including VAT. W www.ultimatesupport.com

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ADPTR Audio Utopia Reverb Plug-in The Utopia reverb from ADPTR Audio is marketed under the Plugin Alliance umbrella and supports all the mainstream plug-in formats for macOS and Windows. While most reverb devices and plug-ins either take an algorithmic or convolution approach, Utopia uses a similar spectral manipulation technique to Zynaptiq’s Adaptiverb so, like Adaptiverb, Utopia can dynamically reshape the spectrum of the reverb to remove any elements that might clash harmonically with the input signal. This spectral approach to reverb generation also produces a very dense and satisfying reverb tail that can be made very long, without any ringing artefacts. If special effects are your thing, you might be intrigued to know that in this case, the reverb time can be set up to a whopping 500 seconds (after that you hit the Infinity mode).

The decay time can be sync’ed to DAW beats and there’s also a Freeze button to extend the current reverb sound indefinitely by freezing the reverb buffer.

Two rows of controls address all the key functions but more depth is often available if you click on the control name — for many controls, this opens up a parameter box at the bottom of the GUI for adjusting settings in more depth. There are preset A, B and C variations, as well as manual adjustments available for Early Reflections, Spectral, Stereo, Detune and Shimmer.

The Mix, Early Reflections and Tail functions each have separate controls, and there are further controls for Pre-delay, Stereo and Shape, the latter adjusting the onset of reverb. A Send control below the Mix knob comes into its own when using the reverb as an insert

Electro-Harmonix LPB-3 Linear Power Boost Boost & EQ pedal The first pedal offered for sale by Electro-Harmonix (or EHX, as they’re often called), some 57 years ago, was the LPB-1, which was a simple gain boost pedal.

They’ve now revisited the concept to create the LPB-3 Linear Power Booster, which incorporates an active EQ, complete with a parametric mid control, and I find it can be incredibly useful in tailoring the sound before it hits a drive pedal or amp. The

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“If special effects are your thing, you might be intrigued to know that in this case, the reverb time can be set up to a whopping 500 seconds...” effect, as if this is automated it allows the tail to fade out naturally, as it would if fed from a mixer’s aux send.

The lower row of controls include Detune and Shimmer, as well as controls governing Decay EQ, Post-EQ and Colour. Behind the scenes, Colour adjusts what ADPTR Audio term the Focus and Atmosphere parameters, and is different for each of the factory presets. Focus reduces the contribution of noise and transients while Atmosphere can either bring out detail or create a smoother tail. Colour settings (which comprise spectral and EQ settings) can be loaded separately from any preset.

pedal can also add up to 33dB of boost, making it well suited to pushing the front end of an amplifier to coax out its natural saturation characteristics. This impressive headroom is achieved by internal circuitry that increases the power supply to 30V. True or buffered bypass operation is available, as you prefer, courtesy of an internal switch that’s accessed by removing the bottom panel, held in place by four crosshead screws. A standard footswitch bypasses the effect, with a red LED showing that the effect is active. (LEDs were barely out of the experimental stage when the LPB-1 came out!)

Although the control set looks fairly straightforward, many of the GUI controls are actually macros controlling multiple parameters. For example, Shape affects decay, sustain and swell. Furthermore, each of those parameters also links to a spectrum map to determine which frequencies are affected by the parameter and by how much. Volume Shaping is available with a choice of Duck, Gate (which can be used as a type of LFO volume modulator) and Pump modes, sync’able to tempo or to a side-chain input. There’s also the option to tempo-sync a buffer-clear feature, to introduce rhythmic truncations of the reverb tail that can sound very effective on rhythmic parts. Auto Clear reduces harmonic clashes by suppressing frequencies in the reverb tail that clash with those of the input signal.

Applications naturally include the conventional uses of reverb in music production, but they also extend further into the territory of sound design, with impossibly long reverbs, spectral coloration, shimmer and rhythmic gain pumping. The shimmer feature sounds very smooth and the sync’able rhythmic effects could be put to good use when mixing dance music too.

Other settings introduce unusual spectral textures, adding an otherworldly quality to the sound, so the possible applications of Utopia span almost all musical genres, from ambient and pop to experimental and EDM. The plug-in comes with a comprehensive set of presets, organised by category, demonstrating the range of possible effects, and there are helpful touches like parameter locks, to maintain selected settings when loading new presets. There’s a 14-day free trial and I’d definitely recommend taking it for a spin. Paul White £ $190 (discounted to $83.99 when going to press). Prices include VAT.

W www.plugin-alliance.com

Housed in a standard, pedalboard-friendly Nano-format die-cast alloy case, the LPB-3 draws 120mA. Battery operation is not supported, but the pedal runs from a standard 9V power supply and one is included in the price. There are six knobs, four of which relate to the three-band EQ: this has treble and bass knobs, plus adjustable mid frequency and mid gain. Treble affects frequencies above 3.2kHz while bass works below 240Hz. The mid band can be adjusted anywhere between 240Hz and 3.3kHz. All three EQ stages offer both cut and boost. Mid Q (Bandwidth) is switchable

“The ability to adjust the guitar’s tonality before it hits the amplifier opens up a world of useful tonal options.”

between low and high rather than being fully variable. The Boost knob adjusts the overall output of the pedal, and the Max switch offers a choice of 20 or 33 dB maximum boost. A Pre-gain knob adds up to 20dB of gain before the EQ stage, which suggests that the post-EQ boost switch adds either no further gain or 13dB, to reach the 20 or 33 dB maxima.

Having so many knobs on such a small panel does make them a little fiddly to adjust, but the pots have a pleasing resistance so tend to stay in position once

set. I found the circuitry to be quiet, even at higher gain settings, and the two Q options are very well chosen, the lower setting adding a gentle hump or dip, and the higher setting a little bite and attitude to the mids, without sounding overblown.

Used before an amp or indeed an amp emulation, the ability to adjust the guitar’s tonality before it hits the amplifier opens up a world of useful tonal options, from adding a Tube Screamer-like mid hump to warm up drive sounds, to thickening up the sound of low-output, single-coil pickups.

Alternatively, thick‑sounding humbuckers can be thinned and brightened. Used before a drive pedal the LPB-1 makes it possible to conjure up a wide range of drive sounds, while the additional gain can be useful in pushing a drive pedal harder, in the same way as it can be used to push the front end of an amp. With all that boost on offer, no high-gain sounds are out of reach, though for most of us just adding a few dB of boost will probably be enough to warm things up nicely.

In the studio, putting the LPB-1 between the guitar and the input to an audio interface offers both level and impedance matching, as well as the tonal shaping that in many cases can help you fine-tune the sound of modelled amplifiers. There are other boost pedals out there — lots of them! — but, for me, the flexible midrange EQ of the LPB-3 sets it apart from much of the competition. Paul White £ £119 including VAT. W www.ehx.com

Lake People G103-D-4 Headphone Amplifier I’ve looked at several of Lake People’s headphone amps in these pages before, and I have been impressed by their no-nonsense, engineering-led design.

The G103 and G105 MkII models reviewed back in SOS August 2022 (https://sosm. ag/lake_people_g103_2_g105_2) were all-analogue devices, with inputs on XLRs and RCA phonos, and were designed to be powered directly from a mains supply. But the latest variant to join the range is a bit different.

The ‘D’ in the G103-D-4 stands for Dante, and you’ll find neither analogue inputs nor an IEC connector on its rear panel.

Instead, there’s an RJ45 connector and a pair of USB sockets. The former is the only option for getting audio into the unit, and if you connect it to something capable of supplying Power over Ethernet, then a single Cat5 or Cat6 cable is all you’ll need.

However, PoE is far from standard on Dante gear, so you also have the option of using

either the Type‑B or Type‑C USB socket to deliver the necessary DC voltage. (Neither of these can be used to pass audio, by the way).

Although the other G-series headphone amps can drive two pairs of cans at once, they are stereo-only devices that feed the same signal to both pairs, at the same volume. The G103-D-4, by contrast, is in effect two completely independent headphone amps, each offering a volume

control plus mini-jack and quarter-inch headphone sockets. Plug it into a Dante network and you’ll see two separate pairs of receive channels, so you can send different signals to each if you wish. (If you only need a single stereo feed, there’s also a smaller model called the G103-D-2.)

Like the other G-series models, the G103-D units feature Lake People’s ‘pre-gain’ options, accessed by opening up the unit and making selections on DIP

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switches. These allow the gain level to be optimised for different impedances and sensitivities, so that the volume control operates in a sensible range for your headphones. Why Lake People use Allen bolts at one end of the unit and Torx at the other I don’t know, but a Torx driver did the job with both.

I tested the D103-D-4 with Dante Virtual Soundcard. The experience was completely trouble-free, latency was imperceptible, and the sound quality was as good as I remember from the other G-series units. Powering the G103-D-4 over USB doesn’t seem to compromise its ability to put out the sort of ear-melting levels that some drummers require, and if your network carries PoE, the

Teletone Audio Tone Architect Tone-shaping Plug-in Teletone Audio’s Tone Architect might, at first glance, look like just another take on the old channel-strip idea. But in reality it is quite different, and its main focus is on tonal shaping. Tone Architect, which supports all the mainstream plug-in formats for macOS and Windows, offers a sensible balance between flexibility and ease of use, and combines three key processing sections called Saturator, Tone Shaping and Compressor. The Saturator section itself can be switched between three analogue-flavoured saturation types too: Tape, Tube, and Cassette. The amount of saturation is controlled by the large Saturation knob, but you also have separate controls for Pre- and Post-gain. Separate Input and Output gain controls at the bottom of the GUI come at the start and end of the overall signal path and include level metering. Oversampling can be turned on to improve fidelity. That comes at the expense of greater CPU overhead, of course, but this plug-in certainly isn’t greedy when it comes to processing power, and it sounds perfectly fine to me with oversampling turned off. There’s an overall bypass button and you can bypass the individual sections by clicking on the label of each.

The Tone Shaper section comprises a rather unusual three-band EQ, with adjustable crossover frequencies — as

“This is one audio tool that will get a lot of use once you’ve tried it.”

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Power can be received over the Ethernet connection, or via the USB A or C ports.

convenience of having the entire thing driven from one Cat5 cable is hard to beat.

The adoption of audio‑over‑IP in music recording still seems quite slow compared with adjacent businesses such as broadcast and live sound, and I think one of the factors in this has been a dearth of useful utility devices. If you

expected the Mid control’s range is constrained by the two crossover controls.

The main Lows and Highs controls are not the usual shelving filters but instead have what might be described as a Pultec-style character — for example, boosting the low end by turning the Lows control anticlockwise simultaneously creates a dip at a higher frequency to keep the sound well-focused. By contrast, if turned clockwise, a dip is created below the crossover frequency, so these filters also include an element of ‘tilt EQ’. Where this tilt occurs is set by adjusting the crossover frequency, and the curve graphics alongside the controls help show what is happening here. Separate 12dB/octave low- and high-pass filters are each adjustable all the way from 20Hz to 20kHz, and allow you to focus this effect’s contribution to your sound pretty much anywhere you want in the audible spectrum.

There’s also a Band Compress control, which was apparently also inspired by the Pultec: it acts as a three-band compressor to dynamically pull back where you’ve boosted each band in the Tone Shaper section.

Although the Tone Shaper doesn’t behave at all like a conventional cut/boost three-band EQ, it does make it incredibly easy to dial in sounds by ear, once you’ve figure out what’s actually going on here!

To the right is a transparent-sounding soft-knee compressor, with variable Attack and Release times, Ratio and Mix controls, with a large central knob to set the amount of compression. Apparently this was inspired by the API 2500, and if set to its lowest attack time it’s a really fast compressor.

There’s no gain-reduction ‘meter’ as such, but there is a threshold LED that modulates

“If your network carries PoE, the convenience of having the entire thing driven from one Cat5 cable is hard to beat.” can’t find a simple headphone amp or monitor controller that works for you, the ability to carry millions of channels over a single cable is somewhat redundant.

The G103-D-4 fills a gap very neatly.

Sam Inglis £ £560 including VAT. W www.synthax.co.uk W www.lake-people.de

in brightness when compression is taking place, and the amount of gain reduction is displayed numerically. An Auto Gain button is located below the Compress knob. Compressor control values are shown whenever a control is being adjusted, as is also the case for the various gain controls and the Tone Shaper HPF and LPF controls.

Tone Architect is a little different from the norm, but it really doesn’t take long to get used to it, and in most cases it could well be all that’s needed to finesse the tonal qualities of a track: I found I could use it to add depth to drums, give more definition to an acoustic guitar or lend weight to a voice, for example. Tone Architect also works exceptionally well as a bus processor or mix processor, with an ability to act as ‘audio glue’, making all the different parts sound like a more cohesive whole. Its saturation section works smoothly and sounds to me pretty much like the real thing. On those rare occasions that you do need more control over the EQ balance, adding another plug-in with high‑ and low‑shelving EQ controls will take care of it, but in the majority of applications, Tone Architect will do all that’s needed without such assistance. I think that this is one audio tool that will get a lot of use once you’ve tried it. Paul White £ $89 W https://teletoneaudio.com

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Written in the accessible, no-nonsense style of the Sound On Sound team of authors and editors, Recording Technology: Basics & Beyond covers: ■ What to buy ■ Studio setup ■ Computers for audio ■ Audio interfaces ■ Monitoring ■ Acoustic treatment ■ Mic techniques ■ How digital audio works ■ Understanding your DAW software ■ Upgrading your system

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Heavyocity Damage Guitars 2 & Damage Rock Grooves 2 Kontakt Instruments

HHHHH I suspect many heavy rock and metal fans will have enjoyed Heavyocity’s Damage Guitars library (reviewed in SOS December 2023). Happy to say there’s a sequel: Damage Guitars 2 continues the metal mayhem with a collection of leads, riffs, grooves, motifs, shredding licks and a host of extras, enabling you to (cough) “channel your inner metal god”. The library is 7.95GB installed and runs on full Kontakt and Kontakt Player 7.10.5 or later.

Heavyocity’s Neil Goldberg returns to perform the samples on a tasty collection of era-spanning six-string guitars. The library features three distinct tone flavours: ‘Metal’ and ‘Damaged’ are aggressively modern, while the classic ‘Hard Rock’ was created with Marshall and Victory Amps Duchess tube amps. A variety of overdrive and effects pedals were also employed.

The cornerstone of the library is a total of 432 tempo-sync’ed guitar riffs with matching bass lines, all played in the keys of D, F# and A in the 85-115 bpm range. Each tone flavour provides 24 guitar and bass riffs in all three keys, with keyswitches available to change their central pitch. In addition, 36 motif loops in each key function as melodic ostinato phrases.

Immaculately played and brimming with aggression, the riffs encompass chugging eighth-note grooves, furious 16th‑note repetitions, syncopated staccato rhythms, doom-laden bass lines, low-lying melodic patterns and simple chord movements. The motifs are generally more tuneful, with occasional forays into Black Sabbath-style heaviness. Taken together, these performances show great musical imagination while remaining essentially supportive, an admirable achievement.

For hot melodic action you can choose from 648 tempo-sync’ed guitar leads and licks (216 in each key, 72 per tone type). Mr Goldberg dispatches these with enormous flair and commitment, employing bends, heavy vibrato, Van Halen-esque squealing pinch harmonics and exciting flashes of virtuosity — the perfect modern lead guitarist resident on your hard drive, available 24/7 with no outlandish rider demands.

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You can construct convincing guitar solos simply by stringing the licks together, or create your own rocktastic performances with guitar and bass sustains, distorted EBow long notes, bends (three types), feedback, major, minor and bar chords, single-note and fifths chugs and trills (which sound amazing when played chordally!) 16th‑note runs, divebombs, scrapes and dead strokes.

Guitar and bass slides are also supplied in all three keys.

As in the original library, the Designer instrument transforms the raw samples into lush, layered cinematic textures, viciously syncopated rhythmic sequences and epic end-of the-world soundscapes. I particularly enjoyed the wonderful floating ‘Six String Cloudscapes’ pad.

In another welcome return, the affordable Damage Rock Grooves 2 contains a collection of beats written specifically for Damage Guitars 2. Based on the Damage drum kit, it mirrors the guitar riff bank names, making it easy to pair grooves with corresponding riffs. Let’s rock! Dave Stewart Damage Guitars 2 $299, Damage Rock Grooves 2 $49. www.heavyocity.com

Westwood Instruments Novella Origin & Novella Blackwater Kontakt Instrument

HHHH

Unique, but designed to fit together, Westwood Instruments’ Novella series provides diverse toolkits for scoring and songwriting.

Origin’s 38.7GB brings strings, alto choir, electric guitar, cotton-draped piano, synth sounds and a drum kit, while Blackwater’s western-inspired 32.6GB includes strings, gut violin, electric and acoustic guitars, mandola, harmonium, trumpet, harmonica, piano and percussion.

These libraries are inspiration machines and get you writing — fast.

Origin’s strings feel bright and sharp; the Improv and Sul Pont articulations are vivid, with harmonics poking out here and there.

The Random Rebows provide some lovely movement, and the Short Picks sound crisp — a great alternative to normal pizzicato.

Blackwater’s lush‑sounding strings come with the ‘Motion’ function which moves through dynamics for you, so you can play two handed with swelling movement. The Drifting Tremolo allows morphing between normale and sul pont measured tremolo performances. Origin’s Vivid Legato has some moving texture in each sustain, and is great in combination with conventional longs. I prefer Blackwater’s slightly more agile legatos, though. Blackwater also comes with a gut violin — a raw, open sound; great for writing brooding melodies and for folk productions.

The libraries’ guitars have a lovely tone, though on the Sustains I would have liked more round robins. Each of Blackwater’s acoustic strummed instruments comes with four rhythmic loop options, which are great cue starters. My favourite of the guitar sounds on offer in Blackwater are the Splangs and Slides, providing slow strummed chords and bends that summon Americana immediately, while the Wends in the nylon acoustic provide beautiful textures of random plucked moving parts.

Origin’s alto choir provides some beautiful, haunting improvised textures with vowel morphing, alongside some crisp short ‘nahs’ and ‘ohs’. The Wanderlust synth fits into Origin’s moody aesthetic, while Blackwater’s harmonium sounds soulful, and the ability to dial in the mechanical sounds is welcome. The harmonica is great.

The Wild Longs have fun bending to the extreme, and the trumpet gives us three distinctive longs, alongside some great ‘motifs’ — jumping up fourths/fifths, for example. Giddy up!

Origin’s Cotton Upright piano sounds gorgeous — nostalgic and mellow. Using the Transpitch feature to transpose the samples from two to 12 semitones down provides really moody results. Blackwater’s Hickory piano is the exact opposite, a brilliant sound based on drumsticks hitting the strings.

Origin’s drum kit sounds cinematic, and is especially great for quiet, ticking parts, while Blackwater’s percussion section includes a jaw harp among a grab bag of oil drum hits, snares, toms and cowbells.

Mashing the keyboard in anger produces some great results.

Both libraries run in Westwood’s familiar engine (in Kontakt Player Version 6.6.1 or up), which includes delay, EQ, and velocity sensitivity and round robin customisation.

Whether you’re scoring to picture or want some unique sounds, the Novella series provides a diverse range of options to expand your palette at a very reasonable price. Thomas Field £299 each, £498 together. www.westwoodinstruments.com

Void & Vista Frames

Kontakt Instrument

HHHH Continuing their reputation for unique and inspiring Kontakt instruments, Void & Vista’s latest release is framed as “an exploration of experimental sounds, prepared instruments and extended techniques”. That’s quite a statement, but by design, V&V’s sonic offering is simplistic and engaging, thanks to the uncluttered interface which un-hides parameters as you go.

At the heart of each Snapshot (patch) is a two-partial construct. Partial/sample selection is revealed by clicking at the bottom of the window, at which point you are presented with five categories, each containing eight samples. At the simplistic end, the Waves category offers a collection of subtractive-style waveforms, along with noise. The inclusion of these is useful, as some of the sounds from the four other categories have less of a sustaining character, featuring moments of random accentuation.

Case in point: the String Piano sample has been recorded by way of a piano note being struck, before being excited by a guitar string. There is a natural decay, exaggerated by the chaotic, random attacks of the guitar string. This unique sound may, in some settings, benefit from tonal bolstering, and those subtractive waves are ideal for the task.

Moving to the other less obvious sample categories, the Instrument category offers samples of string strikes, erratic bowings, drums and the sound of a clarinet overblowing; this latter sample is very stylised and effective, but the audio equivalent of a clarinetist having a nervous breakdown! Other categories include Objects, Effected and Textures, which all contain a diverse collection of raw samples, coupled with annotated descriptions. There are some gems here, from heavy magnets

being dragged across a wooden table, to a food tub being repurposed as a speaker enclosure, while filled with paper!

You have to applaud the depth of imagination, but this is only the beginning of the adventure, as the two partials combine to form an expressive patch, which lends itself to numerous forms of modulation. These mods and wider sonic controls are accessed via the seven dials located at the top of the instrument. Clicking on each reveals a substantial number of settings, providing everything from basic filtering and application of delay and reverb, through to full-on and sometimes wild modulation excesses.

Void & Vista provide 300 curated Snapshots, residing alongside the 16GB of sample data, and implore you to use the modulation wheel when auditioning. This underscores one of the unique details of Frames, which is that it thrives upon varying degrees of movement, often generated by the mod wheel. Even the most basic process of morphing from one partial to the other can create unique interest and space, and when placed under automation (either from within the instrument or from DAW control) provides an instantly unique backdrop.

As a playable instrument, it lends itself to the more static and elongated side of things, but it’s certainly plausible that some of the more percussive elements could be used melodically, which would sound exceptionally expensive in a layered scenario. It’s another unique and very classy instrument, plucked from the Void... & Vista!

Dave Gale £129 www.voidandvista.com

Song Athletics Friends Kontakt Instrument

HHHHH With a quick browse of the Song Athletics website, you will quickly realise that this is a sample library, virtual instrument and plug-in development team that walks a somewhat individual path. Their compact product catalogue provides an eclectic suite of options. The latest addition to this range is Friends, a Kontakt-based sample library that provides a collection of fairly conventional piano, keyboard, guitar, bass and drum options. All of which doesn’t really seem to fit the somewhat quirky nature of many of the other Song Athletics offerings. So, is there a suitably creative twist?

Well, yes... and no... but I suspect the ‘no’ bit is very much by design. The library provides 14 instruments in total, built from just over 15GB of samples and, once purchased, can be downloaded through Native Access.

The instruments themselves span various pianos (grand, upright, felt, electric), four different synth patches (drawn from Roland SH‑101 and Juno‑106), an acoustic drum kit featuring Ludwig, Rogers and Paiste components, a Gibson acoustic guitar, a Nash T-style electric guitar, a Jazz-style electric bass and a Paul Claudot upright bass. It’s here that we begin to see some of the Song Athletics aesthetic as you get the very distinct impression that these instruments have been carefully chosen for their classy sound. They have certainly been carefully sampled. All the recording was done at Snap! Studios in London through a classic 1970s Neve console, with sampling at 96kHz, and the final playable instruments feature up to eight round robins and six dynamic layers.

Normally, this would be the point at which your reviewer might say something about the Kontakt UI employed but the second rather individual (unusual?) element of Friends is that said UI has absolutely zero controls.

Therefore, for each instrument within the collection, what you see (or rather hear) is exactly what you get. If you want to tweak the sounds, or add some ambience treatments, then you will need to do that within your DAW. However, those carefully crafted 14 instruments don’t half get you off to a good start as they sound beautiful. Warm, natural, and very musical, this is a rather wonderful palette of sounds.

And that is, I suspect, the key point of Friends’ design: to give you a conventional set of instrument sounds, with a very natural sonic ethic — no fuss, just play and compose. If your styles include acoustic pop or singer/songwriter type productions, Friends gives you a compact, easy-to-use collection of high-quality — and very organic — sounds within a single library. It would make for a great production/songwriting soundbox on your laptop for composition sketching on the move. No, it may not be for everyone, and it perhaps won’t be the only sample library you’ll ever need, but Friends does what it does with some considerable class.

John Walden £129 www.songathletics.com Audio examples of this month’s libraries are available at www.soundonsound.com.

www.soundonsound.com / July 2025

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Studio One

TECHNIQUE

We explore the new Cinematic Lights instrument.

ROBIN VINCENT

I

n the 7.1 Studio One update, PreSonus gifted us a new virtual instrument called Cinematic Lights. It’s designed to blend heavily processed strings and brass with synth, modular and environmental elements to offer complex textures, smooth transformations and multiple layers of movie magic.

Cinematic Lights is based on the Presence XT engine and so is essentially a multisampled instrument library. It uses an overhauled front end to give you more intuitive and creative control over how the sounds are layered and modulated.

We find similar ideas in Deep Flight One and Lead Architect, which came with the initial release of version 7. This new hybrid instrument comes with 120 sampled sounds, 120 layer presets, 96 instruments and 50 Musicloops. Let’s dig in.

Layer Cake The front end of Cinematic Lights mirrors the instrument’s intention: it’s moody, interestingly lit and asking for interaction.

When you lay your hands on the keys, the default preset is exactly what you’d think; I half expected the background to spring

The main screen of Cinematic Lights hosts a number of macro controls and a three-way vector pad for blending between the layers.

into life with illuminated flying objects, moodily moving figures and the stings of a few drops of rain. Instead, we have some controls to play with.

A preset is made up of three layers represented in yellow, green and red.

The layers can be seen at the bottom of the GUI and have their own controls for panning, reverb and delay. Clicking the little down arrow opens the instrument browser, where you can select your sound

The detailed layer editor gives you direct access to each layer’s amp, filter and modulation options.

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from seven categories. However, it’s the strange arrangement of triangles in the middle of the panel that really catches your eye. The Triangle Vector Pad is like a three-axis X/Y pad, or a three-edged vector synthesis system. As you drag the little white triangle in the space, the three coloured lines on each side expand and contract to reflect the position. The lines correspond to the volume of the associated layer and so as you move you get a different blend of sound. If you click on the even smaller triangles at the points of the middle triangle, the mix goes 100 percent to that layer, essentially acting as a layer solo. It’s much less complicated than it sounds. All you have to do is hold a chord and move the white triangle and it becomes obvious very quickly.

To the right is an Attack and Release envelope for each of the layers, split into their colours. There’s a full ADSR envelope behind the scenes that we’ll come to in a minute, but for now, PreSonus feel that the journey into the sound and back out again are the most important aspects of the multi-layered sound.

Over on the left is a master filter cutoff.

It’s tied to the filter cutoff in each of the layers and gives you an easy way to control the timbre of the whole instrument.

Next to that is the slightly mysterious Sample Shift. The idea behind Sample Shift is that it changes the speed of the underlying sample without altering the pitch.

This results in some quite interesting tonal

automation and set Studio One in motion, the level movements of all three layers will be recorded on to separate lanes as you manipulate the triangles. Alternatively, you can add waveshapes and write in modulation to the track for the individual parts to mess about with the blend.

Global FX

If you map a MIDI CC to a front-panel control, it will only adjust the first layer.

To map a controller to another layer’s parameter, you need to create a macro in the Channel Editor.

shifts. According to the Presence manual, Sample Shift should be able to go from -36 to +36 semitones, and in the Presence XT instrument, it certainly does. In Cinematic Lights, however, the up/down buttons shift it ±1 octave, with no other values on offer. The only way to access the full range appears to be to map a MIDI control to it. It seems an odd oversight for a control that’s given prominence on the front panel.

Macro Polo Before we move onto the effects and deeper layer editing, it would be a good idea to talk about how best to interact with the front panel. Essentially, it’s a macro interface that works well with your mouse to give you overall control of some choice aspects, but it’s less adept at MIDI control.

Whenever you attempt to map a MIDI control to anything on the front end, it only gets attached to the first layer.

We can get around this by generating our own macro controls in the Channel Editor. You can bring up the Channel Editor window by clicking on the track number or the little hand of cards icon on the console channel strip. Then click on the little knob icon to bring up the macro window. It should have nothing assigned to it, so click on the little spanner icon to get us into the editor.

On the right, you’ll find all the available

parameters in Cinematic Lights listed.

You’ll find the filter cutoff controls under each Part. Simply drag each entry into the middle section and your macro knob will control all three. Now, if you click the ‘Edit mapping’ over on the top right you’ll be able to assign a MIDI knob to control the macro and so control all three layers’ filters at once.

You could do the same for the envelope attack and release.

The Triangle Vector Pad is another matter. You’d need some kind of three-dimensional controller to let you blend and mix three sources with one control. You can use a knob for each layer level or even a single knob that crossfades between two, but that’s as far as we can go. The one route open to you is through automation. If you enable Write

Squeezed between the part selection section and the keyboard is the Global FX section. It offers an overall volume control and a mix knob for the reverb and delay. This is also where we encounter the Arpeggiator and Repeater, which are MIDI note effects rather than audio processors.

Each part layer has a send control for the reverb and delay, where you set the level of signal from that part being routed to them, but it’s here in the Global FX that you set the overall return level of those effects. If you click on the effect label a new panel slides out, giving you a handful of controls over the size of the reverb, and the timing and feedback on the delay.

The Arpeggiator has aspects of a conventional arp, in that it will play whatever notes you hold down in a number of different directions. However, you can also create rhythmic patterns to run the arpeggiator groove, giving it a bit more spice. The editor is on the small side, but you can program up to 32 steps with gate length and note velocity. Simply click on the steps to drag out the gate and velocity values, and enjoy how it complements the regular play directions. There’s a bunch of presets to enjoy from a drop-down menu on the left. Sadly, you can’t save your own.

Alternatively, you can run the Repeater, which retriggers the note you’ve just played up to 32 times. You can set velocity and timing, including whether it’s sync’ed to the tempo or free‑flowing. But perhaps more interestingly, you can scale the velocity so that it gets louder or softer with every repeat, or scale the pitch so it rises or falls from the original note. But wait, there’s

The Triangle Vector Pad can be automated, in which case each axis gets its own automation lane.

www.soundonsound.com / July 2025

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TECHNIQUE

CIN E M ATI C LIGH T S

The Arpeggiator allows you to program your own rhythms, with up to 32 steps and adjustable gate length and velocity.

Like the Arpeggiator, the Repeater is a MIDI effect, and can go beyond simple note retriggering thanks to its built-in basic sequencer.

more. You can also decouple the scaling and set the pitch and velocity individually for each step, effectively turning it into a cool little sequencer. One thing you can’t do is loop the sequence, so if you wanted it to play continuously, you’d have to drop a note into the main Studio One sequencer at the start of each bar.

Interestingly, you can enable the Arpeggiator or Repeater for each layer individually, giving a really nice depth of movement and variation to a preset.

Digging Deep Each layer has a more detailed view, which you can access by clicking on the loaded part. The details will be familiar to anyone who’s used Presence XT. You get two LFOs with multiple waveshapes, rate control, and a useful delay function to hold back the modulation so that it doesn’t kick in until sometime after you’ve played the note.

The rate can also be scaled by the notes being played so the LFO will be faster with higher notes and slower with lower.

The filter is nicely comprehensive, with multiple modes and characterful controls.

You can select some of the modes on the row of buttons, but you’ll find more hidden away under the drop-down menu.

Essentially, you get every sort of ladder filter and every state‑variable filter plus a simple Eco filter that is easier on the CPU. For character variation we have a Soft button for a more mellow tone, Drive for saturation and Punch adds a bit of percussive attack.

The filter can be pushed into self-oscillation if you select the Eco variant and wind up the resonance. The Key control will alter the cutoff frequency as you play up and down the keyboard, and if you set it to maximum you can play an oscillating filter like another oscillator. You still have to have a sound loaded, but it can be quite an effective additional tone.

One tip while you’re trying to find sounds that combine well with a self-oscillating filter

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is that there’s a little padlock next to where you load each layer sound. If you enable the lock, all the editor settings, including the filter, will remain unchanged, and only the samples will be loaded. It’s a great way of auditioning new sounds in a patch without losing all the editing and modulation settings you’ve been working on.

The amp and filter both get an ADSR envelope, although funnily enough, the filter envelope isn’t hardwired to the cutoff.

You have to make that connection in the modulation matrix, which I’ll come to shortly, where it’s called Env2. This envelope also has a delay parameter so you can control when it starts to have an effect.

In the FX A and FX B sections, we get a bunch of layer-specific audio effects.

These include chorus, flanger, phaser, bit‑crusher, distortion (with multiple types), a gater (which acts on the audio, not the MIDI notes like the Repeater does), an EQ and a panning effect.

The dedicated PreSonus Musicloops are a great way to get started with Cinematic Lights.

Lastly, we have the huge 16-slot modulation matrix, which you may struggle to fully populate with parameters. There are actually 18 destinations, so you could certainly fill it, but you might have to double up on some of the 14 modulation sources. You put your main choice of modulator at the top, and then you can add a second modulator that the first one can be modulated by. So you could, for example, shape an LFO with an envelope.

In the bottom slot, you put the parameter you are trying to control and then the line in the middle sets how deep the modulation goes and whether it’s positive or negative.

It’s a potent modulation system capable of swirling these samples inside out.

Be Inspired If you’re not really sure where to go with Cinematic Lights then I’d recommend making use of the included Musicloops to stir up some ideas. Musicloops are a Studio One file format that can contain audio and MIDI information, including the virtual instrument used and the routing.

You can use them for audio clips or MIDI phrases, but they are best when dragged into space where they can create a track and load up their intended virtual instrument.

You’ll find the Cinematic Lights Musicloops in the Browser under Files / Sound Sets / Cinematic Lights / Musicloops. They are listed with key and tempo information and should shove you down the right road.

TILEYARD LONDON

19TH JULY 2025 MUSIC STUDIO EQUIPMENT SHOW BUY YOUR TICKETS NOW AT GEARFESTUK.COM

Pro Tools

TECHNIQUE

Playlists are the key to managing takes, comping, and much more!

JULIAN RODGERS

I

n Pro Tools, many of the most useful features aren’t recent additions, but quietly solid tools that have been there for years. Playlists are a perfect example.

They’re not new. They’re not exclusive to Pro Tools. But if you record or edit in Pro Tools and you’re not using them properly, you’re almost certainly making your life harder than it needs to be.

Pro Tools playlists allow you to manage multiple takes or alternative versions on a single track, without any unnecessary duplicating. It’s the feature that underpins everything from vocal comping to ADR to guitar solos and multi-mic drum sessions.

Recent updates have added support for MIDI playlists too — a long-requested feature that finally brings parity between audio and MIDI workflows in this area.

In this article, we’ll look at what playlists are, how they’re used, and why they’re still one of the most valuable tools in Pro Tools for producers, engineers and editors.

A playlist in Pro Tools is basically a version of a track’s contents. Each track has a main playlist, which is what you see and hear by default, but you can also create as many alternate playlists as you like. Each one is a full version of the track’s timeline, complete with clips, edits and fades.

Playlists are most commonly used to manage alternate takes, whether captured during recording or created as part of an editing process. Instead of duplicating tracks or muting clips, you can simply create

A guitar solo, comped using playlists. The pink track has been ‘promoted’ to be the main playlist (which you hear when the track plays back), and a section from playlist 04 has been inserted into it.

a new playlist for each take and switch between them as needed.

The beauty of playlists is that they live within the track. There’s no clutter in your session. No duplicated inserts or sends. No reassigning outputs. Just clean, contained organisation that scales well whether you’re working with a single vocal or 20 tracks of layered guitars.

Creating & Managing Playlists One scenario where playlists can be invaluable is when you want to try out an edit to see whether it works. Rather than relying on the Undo feature, or saving an alternate version of your session, you can just duplicate the content of your main playlist to a new playlist and edit it there.

If it turns out to be a bad decision, you can simply revert back to the original playlist.

To create a new playlist, click the Playlist selector in the track header (that’s the drop-down that shows the track name by default) and choose ‘New’. Alternatively, you can use the shortcut Control+\ (Mac) or Start+\ (Windows).

You can duplicate, rename or delete playlists from the track header. And if you’re unsure whether there are alternate playlists on a track, look out for the playlist selector next to the track name. If it’s blue, there are alternate playlists on that track.

Loop Recording

With this option enabled, a new playlist will be created for each take when using the Loop Record feature.

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One of the best ways to use playlists is in combination with Loop Record. Please note that Loop Record is a dedicated

record mode, accessed by right-clicking the Record button. Recording while Loop Playback is engaged is not the same thing.

With Loop Record enabled, Pro Tools can be configured to automatically create a new playlist for each loop pass. You simply let the track roll and every pass is captured in its own clean lane.

To make this work, make sure the preference is enabled: go to Setup / Preferences / Operation, and under Record, enable Automatically Create New Playlists When Loop Recording. Alternatively, you can access this preference from the track header by clicking the Playlist selector.

Once it’s on, a new playlist is created every time the loop cycles. This is ideal for capturing multiple takes quickly without breaking the flow of a session.

Once recording is done, it’s time to switch to...

Playlist View Playlist view is where the power of playlists really becomes apparent. From the track header (where it normally says Waveform or Clips), choose Playlist. This view stacks all the track’s playlists vertically, so you can see every take in one place.

Each lane shows the waveform for a different playlist, with a solo button that lets you audition that playlist instead of the main playlist. You can solo any lane using the Shift+S shortcut, or by clicking the solo button directly.

Once you’ve found the part of a take you want to use, just select it and use the shortcut Option+Shift+up (Mac) or Alt+Shift+up (Windows), or click the upwardpointing arrow in the lane header. That moves the selected audio to the main

Using the shortcut Opt+Cmd+Shift+up/down, you can cycle through different playlist versions of the same clip, to hear and evaluate them in context.

playlist — the one you’ll hear in your mix.

You can also drag clips from any playlist lane into the main track if you prefer working with the mouse. Playlists support crossfades, clip gain, clip effects, and all the normal editing tools, so once your comp is built, you can fine-tune transitions, timing and dynamics exactly as you would on any other track.

Keyboard Shortcuts Comping together the best bits from your alternative playlists is faster and more intuitive once you know a few key shortcuts: • Control+\ (Mac) or Start+\ (Windows): Creates a new playlist. Fantastic for quickly capturing another take without the singer having to wait for you.

• Shift+S: Solos the selected playlist lane, great for auditioning takes. You can rate them out of 5 from the right-click menu.

• Opt+Cmd+Shift+up/down: Cycles the clip selection through available playlists, making it easy to audition alternative takes in context.

Automation Playlists?

Automation for mix and plug‑in parameters is stored on separate playlists, but unlike audio or MIDI clips, you can only have one playlist per automation parameter on each track. You can’t, for example, have multiple alternative takes of volume automation.

This means that you can’t use this playlist‑based approach to experimenting with alternative automation. There is a way to compare alternative automation passes if you use Pro Tools Ultimate or Studio, though.

To do this you need to use Preview mode, which is part of the advanced automation features. That’s beyond the scope of this article but well worth knowing about.

Once these are second nature, playlist comping becomes a quick and fluid part of your workflow — not something that slows you down or interrupts your creative decisions.

MIDI Playlists Until recently, playlists were exclusive to audio tracks. That changed with the introduction of MIDI Playlists in Pro Tools 2024.10, which finally brought the same comping workflow to instrument and MIDI tracks.

The concept is exactly the same.

Record multiple takes (with or without Loop Record), and each pass can be stored in its own playlist. Switch to Playlist view and you can solo, audition and promote MIDI clips just as you would with audio.

This is a big deal for people working with performance-based MIDI parts, and it’s especially useful for anything involving multiple passes. It also means that users who work heavily with MIDI no longer need to adopt a different approach just because they’re not recording audio.

Like the audio version, MIDI playlists support all the editing tools you’d expect, including quantisation, clip gain, and Real-Time Properties.

simultaneously, keeping different takes organised and avoiding chaos.

Each new playlist is automatically named with a numbered suffix (.01, .02, and so on), which Pro Tools handles intelligently. If you’re working with grouped tracks (for example, a multi-mic drum setup), Pro Tools will apply consistent numbering across all tracks. Even better, if you add a new track later, its first new playlist will pick up in sync with the others. So, for example, if you create playlists on a group of five tracks, then later add a sixth track to the group and continue creating playlists, it picks up the numbering right where the others left off. You won’t end up with track six starting at 01 when the other tracks are on 04, for example.

This kind of consistency makes session management easier, especially when comping across multiple mics or instruments. You’re never stuck trying to remember which version of ‘Take 3’ is the one that was actually used.

You can rename playlists if needed, of course, but for most users, the default behaviour is more than enough. It’s just another way Pro Tools lets you get on with the work rather than spending time on admin.

Why Playlists Still Matter Playlists have been around in Pro Tools for a long time, and like many of the best features, they help with the unexciting details and free you up to do the creative stuff. They let you manage multiple takes cleanly and non-destructively — all within the track you’re already working on. For vocal producers, they’re invaluable. For anyone recording multiple passes, they’re the right way to keep things tidy.

Now that MIDI playlists are supported too, playlists have become even more central to everyday Pro Tools work — whether you’re recording vocals, layering synths, or managing multiple takes in post.

Your Number’s Up Pro Tools’ playlist naming and numbering system is particularly useful when recording multiple instruments

Pro Tools will automatically number your playlists in order — and will maintain the correct naming scheme even if you add tracks to a group that already has playlists.

www.soundonsound.com / July 2025

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Pro Tools Cubase

TECHNIQUE

The new DAWproject file format promises simple project transfers between Cubasis and Cubase.

JOHN WALDEN

T

he act of moving a music project between DAWs isn’t always straightforward, and even though most DAWs offer one or more project export formats the transition can still be somewhat cumbersome and frustrating — especially when it comes to configuring any virtual instruments or effects plug-ins you want included in the process. In late 2023, Bitwig and PreSonus introduced the DAWproject file format, a new project ‘container’ designed to transfer all of the most important elements (audio tracks, MIDI tracks, mixer and plug-in configurations included) between any DAWs that support the file format.

While warmly welcomed by many users, relatively few developers have added support — but, happily, Steinberg are now one of them. DAWproject import/ export was added to Cubase Pro and Artist in v14.0.20 and, wonderfully, Steinberg

added the same support to Cubasis 3.7.5.

So if you like the idea of moving projects between your mobile and desktop working environments, DAWproject now has the potential to make that easy. In use, it’s a remarkably straightforward process — but, understandably, there are still some ‘gotchas’ to be aware of, so below I’ll take you through the pros and more obvious cons.

To There...

A common use case is transferring an idea started on Cubasis on an iPad, while working away from your studio, to ‘full-fat’ Cubase for development or completion in the studio. As you’ll see in a moment, the transfer process is very straightforward. But Cubasis is now a pretty feature-rich recording environment in its own right — so, just how much detail from your Cubasis project might arrive intact in Cubase via the DAWproject container?

Let’s assume we’ve created a typical project in Cubasis for a new musical idea.

With my demo ready to be moved to Cubase, Cubasis’ Media page now offers the DAWproject format for sharing projects.

It consists of a number of virtual instrument tracks (drums, bass, piano and synths, for example) and a few audio tracks recorded on a compact, mobile-friendly audio interface. Perhaps a guitar part or three and a vocal demo. Within the Cubasis mixer, the tracks have been routed to Group Channels (busses for drums, bass, keyboards, guitars) to keep things organised, and to make it easier to create a static mix balance from the various instrument groups. Equally, some routine insert effect processing such as EQ and compression has been applied to some tracks, and both reverb and delay send effects have been configured.

When opened in Cubase, the Cubasis project opens pretty much fully formed, albeit with some notable limitations as noted in the main text.

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We now want to move this idea over to Cubase for further work...

As shown in the first screen, Cubasis’ Media / Share feature options now include DAWproject. Select this and you get a choice of how to move the actual file. In my example, with my iPad sat alongside my desktop macOS system, I could simply used AirDrop to transfer the DAWproject file over Wi-Fi. A few seconds later, the file was available on my desktop’s drive system.

In Cubase, you then create an empty project and, from the File menu, choose Import / DAWproject, and browse the file transferred from Cubasis. One click, and, as shown in the next screen, Cubase will pop open the session. Remarkably, you might already be able to continue working. All your audio, MIDI and Group Channel (bus) tracks will appear, along with FX Channels for any send effects used in Cubasis. The routing to the Group Channels is retained too, along with fader levels for all track types. Panning and send settings are also imported. Usefully, from a navigation perspective, track colours are also transferred. And if you’ve used the Tempo or Signature tracks in your Cubasis project, their data is also carried across to Cubase on the desktop — although I did experience an issue with tempo data being displaced along the project timeline in some cases.

Know Your Limits Of course, given that this is an early iteration of the DAWproject feature in Cubase and Cubasis, and that mobile and desktop music software exists in somewhat different ecosystems, there are some limitations to the transfer process. In the case we’re considering here — an idea sketched out in Cubasis being moved to the desktop for further development — the most important considerations probably centre on the exchange of plug-in instruments and effects data.

For example, Cubasis can’t export data connected with AU, IAA or CLAP plug-ins. So, if you wish to ensure optimum compatibility with Cubase, your Cubasis project really needs to be constructed using only the effects and virtual instruments bundled with Cubasis itself. Still, it’s not all bad news: if your Cubasis project includes some of Steinberg’s IAP (in-app purchase) instruments (for example Halion Sonic Selection, FM Classics, Neo FM, LoFi Piano, Iconica Sketch and Micrologue), the Cubase desktop version of the project

will find suitable presets in HALion Sonic and Retrologue.

I found that the situation with effects was a little more hit-and-miss. For example, RoomWorks SE seemed to map OK, including any settings I’d made. However, while StereoDelay did appear in the imported desktop project, the settings used in Cubasis didn’t. Equally, despite confining myself to the most basic selection of compressor, limiter and EQ choices in Cubasis, sometimes the plug-ins appeared in Cubase and, at other times, they did not.

And while static fader and pan positions translate perfectly, another significant limitation is that any automation data you created in Cubasis does not. Of lesser importance, but I think still worth noting, is that after importing projects into Cubase, I had to reactivate the display of some MixConsole sub-panels.

And Back...

There may well be occasions when you wish to move a project in the other direction, perhaps taking a sketch created in Cubase with you as a Cubasis project while you work away from home or to play with on stage. The same sorts of qualifiers apply, of course, but provided that I tailored my virtual instrument and effects choices broadly to those mentioned above, I found that, if anything, the transfer seemed to work more consistently going in this direction.

This seems to be particularly the case for effects plug-ins: Cubasis was more successful in translating plug-in selections (although not all of their parameter settings) on importing a project from Cubase.

Interestingly, tempo data seemed to move in this direction without any issues.

And, if I did happen to include a third-party plug-in in my Cubase project before exporting it via the DAWproject container, Cubasis simply ignored it very politely — without throwing a tantrum!

third-party plug-in data should transfer intact via DAWproject, provided that the plug-ins concerned are available on the respective host computers (if you are moving projects between two DAWs on the same system, that would obviously be the case).

In that broader context, support for the DAWproject format by Cubase Pro and Artist is obviously a very positive thing. However, I think it’s really neat that Steinberg have looked to include it in Cubasis too. While it has its limitations (some being inevitable due to the very different plug-in architectures of the desktop and mobile worlds), it does provide a very easy means of moving projects between Steinberg’s desktop and mobile music-production systems, and when it comes to transferring initial ideas well before you get into the details of the mix, it’s undoubtedly very useful. So I’ll happily take a glass-half-full attitude to this additional option!

Finally, it’s worth noting that Steinberg seem to view DAWproject support as a work in progress, and have already published a list of features they’d eventually like added.

This includes support for automation data, time-stretch data and for multiple outputs.

Fingers crossed that there’s enough impetus to continue the required development work. That may well depend upon the likes of Apple and Avid backing the format, but it would likely also bring further benefits to Steinberg-only users moving between Cubasis and Cubase.

Glass Half Full When Bitwig and PreSonus introduced the DAWproject concept, their focus would naturally have been on moving projects between different desktop DAWs, but as I’m not a user of Bigwig Studio or Studio One I’ve not had a need to explore this desktop-to-desktop process yet (I do collaborate with Logic and Pro Tools users on a regular basis, though, so I’d love to see Apple and Avid get on board!).

When moving between different DAWs on a desktop system, my understanding is that

Going from Cubase to Cubasis is also possible via the DAWproject format.

www.soundonsound.com / July 2025

133

Pro Tools Live

TECHNIQUE

Live’s Drum Rack remains an extremely creative feature, with depths that may be hidden...

OLI FREKE

D

rum Racks have been part of Ableton Live since version 7, and in the fast-moving world of DAW updates, that can make them feel like a legacy feature. But don’t let their age fool you. Drum Racks remain one of the most versatile, flexible and downright powerful tools in Live’s instrument line-up. Combined with the possibilities of the new Drum Sampler introduced in Live 12.1, Drum Racks have plenty of life left in them.

In this article, we’ll look at what makes Drum Racks still worth mastering in 2025, from core concepts to deeper techniques that will help unlock their creative potential.

Getting The Basics Right At their heart, Drum Racks are containers: multiple 16-pad grids that can hold anything from single samples to layered Instrument Racks and effects chains. Each pad acts as a note-triggered lane designed for drum sounds — but by no means limited to them.

Each pad can house one of the sampler devices (Drum Sampler, Simpler, Sampler),

a Drum Synth or even another Instrument Rack. Each represents a miniature production chain that can support its own effects and internal routing options.

For realism on instruments like hi-hats, choke groups are essential. Assigned in the I/O panel, you also have quick access to a pad’s note assignment, volume, pan, and mute/solo buttons.

Sample hot-swap is supported at pad level, but best done from within the sample device to keep all your envelope, filter, and effects settings intact while just switching out the sample.

The eight Drum Synth devices (DS Kick, DS Snare, etc) can be fired up together in one pre-built Rack by choosing the ‘DS Rack’ option under each of the individual DS devices. Handy for an all-analogue percussion work-out!

Next-level Layering This is all more than enough to get a drum part up and running, but layering can open up more tone‑shaping options, performance dynamics and neat production shortcuts.

A Drum Rack with internal effects routing and chain colouring.

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Drum Synth Rack — A Drum Rack with all eight Drum Synths pre-loaded.

Creating a layered sound on a pad is easy — simply drag a second (or third, or fourth) sample onto the same pad by holding down the Cmd (Mac) or Ctrl (PC) key as you do so. To avoid rigid‑sounding layers, try offsetting a layer by a few milliseconds using a Delay device. Go crazy and control it with an LFO for an organically varied sound.

Creative Grouping & Routing Whilst each pad can support its own effects chain, you may wish to apply the same effect to multiple pads — perhaps a subgroup of toms or metallic percussion.

This is quickly accomplished by selecting multiple instruments or samples from the Chain List and right-clicking ‘Group’. Voila!

A nested Drum Rack has appeared, which can now have a subgroup effect applied.

(This grouping can also be accomplished from the mixer channel strip by unfolding the Rack.)

However, nested groups don’t allow for per-pad send levels. For more flexible routing, use Drum Rack’s built-in Return chains. Click the R(eturn) and S(end) buttons at the bottom left of the Rack to reveal the audio effects slot and send level

Sampler’s Zone view showing the Chain Selector and Chain Fade technique.

controls. Drop an effect in the Rack’s Effect panel and a Send amount column appears in the Chain view.

This internal routing keeps everything tidy inside the Drum Rack. But you can still send a Chain to Live’s global return channels using the Audio Out option if you need to keep any effects independent of the Rack.

Randomisation & Variation Variation keeps listeners’ interest.

The Sampler device’s round-robin and randomisation features offer plenty of tools to introduce subtle — or not so subtitle — variations.

Load multiple samples into Sampler’s Zone view and click RR for Round Robin.

You can then select the method by which the different samples are iterated through — Forward, Backward, Random or Other (i.e won’t immediately repeat the same sample).

An alternative round robin method can be achieved by loading a shaker or hat loop into Simpler: choose Slice mode and put the Random MIDI device in front of it and each new note will select a slice at random.

Sampler’s modulation section is also useful for subtle variation: assign the Random LFO shape to control pitch, pan, filter frequency or envelope parameters, for change that doesn’t require multiple samples. Panning randomisation is built into Simpler (Ran / Pan in Controls) and is effective for subtle stereo movement.

Performance & Expressive Control Drum Racks aren’t just for programming beats, they can support live performance and advanced expression control too.

Velocity layering is, of course, the longstanding method for providing more realism and dynamic variation of otherwise static samples. It is accomplished in Drum Racks by loading the variants into Sampler’s Zone panel and assigning velocity ranges.

You can also build a crossfading pad that lets you morph between two (or more) samples in real time using a macro or automation. Create an Instrument Rack in a pad and load two

samples into separate chains. In the Zone view, click ‘Sel’ and expand both blue zone ranges fully. Next, apply opposing fade ranges by click-dragging the white strip at the top of the blue bar. Finally, map the Sample Select Ruler (the pale blue vertical line) to a macro. Now you’ve got a live crossfader between textures, and that macro can be automated or mapped to a controller.

Save snapshots of macro settings for quick recall by clicking New in the Macro Variations panel. Click Rand to randomise all macro settings for the element of surprise (right-click ‘Exclude Macro from Randomization’ on a setting if needed).

Don’t forget there’s a macro ‘Map’ button which shows a list of all mappings, which is handy as things can get complicated!

Want more chaos? Use an Arpeggiator MIDI effect in front of a Drum Rack sound for auto-rolls, flams or glitched textures. Set Distance to zero — unless you want pitch shifting... and why not, eh? Using Velocity & Decay can create very interesting — and fun — glitchy, timed effects.

Pad triggering is not limited to the MIDI clip on the Drum Rack’s channel either — Drum Rack can accept MIDI from other MIDI channels by selecting the Drum Rack option in its ‘MIDI to’ drop-down menu.

There are some Selector kits built into Live — multiple sample options for a single type of drum that can be seen as a kind of preset hot-swap. These are listed under Instrument Racks, not Drum Racks.

Potentially confusing but it makes sense when you consider that Drum Racks are just a special category of Instrument Rack that pre-filters by MIDI note. These Selector kits also allow drum sounds to be played across the whole keyboard — good for those tuned-808 bass lines! Multiple samples are loaded in and a number of samples can be demo’ed in context by assigning the pale blue Chain Selector to a macro.

Save Smarter, Work Faster When dropping a sample onto a pad a default instrument can be chosen

to house it. For example, if you would prefer Drum Sampler instead of the default Simpler, you can choose it by going to the User Library in Live’s browser and following the path: Live Places / Defaults / Dropping Samples.

Drag your preferred device from the Rack onto ‘On Drum Racks’.

Once you’ve created a Drum Rack, complete with routing, macros, effects and sample organisation, save it as a Preset. Better yet, right-click and ‘Save as Default Preset’ to make it load up every time you fire up a Drum Rack.

Colouring Chains is also possible, and a handy way to see what’s what at a glance; especially useful in marking out subgroups or particular drum types. When it comes to mixdown, you may choose to ‘Extract Chain’, which gives that chain a new channel in the mixer. This can help simplify things if there’s a lot of internal complexity.

Final tip: don’t forget the ‘Sort Chains’ option. If you rearrange pads by clicking and dragging, the Chain List doesn’t update automatically unless you right‑click and select this.

Still Evolving Drum Racks might not be the shiny new toy in Live 12, but they’re still one of the most powerful and adaptable tools in the entire DAW. Whether you’re building dynamic kits, experimenting with sound design, or refining your mix workflow, there’s always another trick to discover inside those humble pads.

If you’ve just been using Drum Racks to load up a few kicks and snares and then calling it a day, it might be time to dig a little deeper.

It can easily be configured to support any level of expressive and creative beatmaking you might want, and combined with the new Drum Sampler device, it still has the power to generate creative surprises, speed up workflow and to empower your music.

Using Simpler’s Slice mode with Random MIDI effect for variation on each hit.

www.soundonsound.com / July 2025

135

Q

Are passive speakers better than active ones?

I was asking for advice on a well-known hi-fi forum and encountered a widely prevailing view that passive speakers are superior to active systems — in some opinions, vastly superior. I think the belief is partly about the benefit of a separate power amp, which will be ‘cleaner’ or do a better job than an amp integrated into the speaker itself. Is this true — are passive systems better than active ones, and does this apply to studio monitors as well as the hi-fi world?

SOS Forum post Hugh Robjohns, Technical Editor Yes and no! The problem in answering this thorny question is that it all depends on the quality of the speaker design and, to a large degree, its price. Ultimately, a well-designed active speaker is capable of achieving a far higher standard of accuracy and fidelity than any passive system, and that’s why most professional recording studios and high-end mastering suites now employ active full-range monitor systems. However, I doubt the correspondents on that hi-fi forum have ever heard such systems, which typically will cost well north of £25k (circa $35k).

More likely, their views are based on comparing very good passive speakers driven by well-made amps with budget active monitors, and at the lower end of the studio active-monitor market the active idea is employed for convenience rather than fidelity. A good-quality, standalone, discrete amplifier will always outperform the ‘chip amps’ that are often built into budget active studio monitors, but high-end active systems use the same calibre of amplifier designs favoured in the hi-fi world.

The main limitation of a passive loudspeaker is that the crossover circuitry that splits the audio spectrum

Q

Passive speakers are so called because of the passive crossovers that are employed to split and route the incoming high-level (amplified) signal so it can move the different drivers.

to feed the various loudspeakers has to operate at very high power levels and low impedances, with entirely passive components which are large, expensive and tend to interact in unhelpful ways.

This all restricts what can be done in terms of phase and amplitude response accuracy. An active system moves the crossover ahead of the power amplifiers so it works at line level with active electronics, and this allows far greater phase and amplitude accuracy, as well as steeper slopes and far better matching between drive units. It also allows for time alignment, and bespoke protection tailored precisely for each individual drive unit. Some recent systems also now use DSP to compensate for inherent foibles in drive units, and even to correct for room acoustic anomalies, which can never be achieved with passive crossovers.

Active systems also ‘win’ because connecting a separate power amp directly to each drive unit affords much tighter control of the drive unit’s movements and avoids any possibility of intermodulation, which can be a significant issue in

What should I use to clean the pots on my gear?

I need to clean the volume and tone controls on a Quad 33 preamp to try and cure ‘scratchiness’. I have a can of Servisol for contact cleaning, but would Deoxit do a better job?

SOS Forum post Hugh Robjohns Caig’s Deoxit D5, which you’ll often see recommended, is designed to deal with the tarnishing that builds up over time on

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passive designs. (It’s probably worth noting that not all speakers with built-in amps are genuinely ‘active’: some are simply passive speakers with a power amp built into the cabinet, and while these have the ‘all-in-one’ convenience of an active design, they offer none of that format’s other benefits.)

So, I’d say that for the ultimate loudspeaker system, an active arrangement will achieve the very best possible performance. But only if it employs very high-quality amplifiers and a well-designed active crossover — so it will inherently be very expensive. At the budget end of the scale, a good passive speaker with a decent amp may well perform better than a cheap active monitor.

It might not be as convenient, compact or robust, which could be important in a home-studio setting, but there may be other practical benefits. For example, if the amp were to fail, you could probably plumb another one in more affordably and with less disruption than repairing/ replacing active speakers.

the metal contact surfaces of connectors and inside switches.

It does that very effectively but it lacks the lubricant that’s essential to the smooth operation of rotary potentiometers and faders, so I’d recommend sticking with Servisol Super 10 or Kontakt Super 10 (the new name for the same thing) for treating ‘scratchy’ pots. Caig’s equivalent product, F5 Faderlube, should be equally effective. Bear in mind, though, that there’s only so much cleaning an old pot can take — eventually, the track will wear out and the pot will need replacing.

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137

TINY GEAR LAWRENCE JOSEPH

M

y early musical experiences included a university course in electronic music as well as playing guitar in various garage bands.

I studied modular synthesis in a room with an expansive wall of original Moog modules, and I lugged around a heavy Stratocaster flight case with an even heavier tube amp. Live processing a guitar through this massive synth struck me as a neat way to expand one’s sound palette, but the logistical problems seemed insurmountable. Aside from the obvious transport hassles, live tweaking of knobs on more complex Moog patches required coordinating several sets of hands, and keeping track of a layout that spread over 25 square feet in area was discombobulating.

I dreamed of a more portable and compact, yet equally versatile, setup.

Over time, technology advanced and equipment downsized. I discovered a handcrafted miniature guitar, and companies like Quilter Labs started producing tube-like amps that could fit into a jacket pocket. The synth side of things remained problematic.

Powerful systems remained too large, and smaller systems lacked the one-knob-per-function convenience for improvisational live patch tweaking.

Then two articles in Sound On Sound caught my attention. First, the March 2022 issue reviewed the Empress Effects Zoia, a comprehensive 90‑module modular synth in a 5.7 x 3.75‑inch guitar pedal format, weighing only 1.5lbs. Both CV and MIDI-enabled, the Zoia contained not only all the synth building blocks like oscillators and filters, but also effects, from distortion to granular synthesis, pitch detection and envelope followers, delays, loopers and reverbs, sequencers, samplers and a usable keyboard!

The Zoia is infinitely flexible, but only a partial solution, since there is only a single knob to control all patch parameters. The missing link came in the form of Intech Studio’s Grids, reviewed in the February 2024 issue.

At four inches square and 0.38lbs, each

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Grid module is a programmable MIDI controller containing up to 48 buttons, pots, faders and encoders. Combining these tiny units with the Zoia provides a modular synthesis system more powerful than a wall of Moog, with individual tactile control over every parameter of every module. One Zoia, three Grid modules and a tiny amp fits into a kid’s lunchbox, and the entire rig including the guitar with carrying case weighs only 6lbs!

Advantages of tiny gear include not only easier transport, but also aesthetic and other practical concerns. No one

can deny the inherent cuteness of tiny gear, but equally important is enabling one to focus attention on a small space with minimal fussing about. Manipulating a multi-module modular synth rack can take deep concentration, but mental strain eases when all control knobs are collected into one tiny space with no patch cables in the way. Yes, the setup requires some organisation and preparation, as the Zoia and Grid modules both need to be programmed in advance to partner well with each other, but that is also part of the fun. My dream system is finally a reality... Tiny gear rules!

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