SOUND ON SOUND [UK] May 2025

SOUND ON SOUND THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

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

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Avantis Solo

dLive features for SQ money?

MAY 2025

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Apple’s M4 & M3 Ultra Macs benchmarked Neve 88C

Studio-quality compression — anywhere!

Mixing breakthroughs How the pros got good

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VOLUME 40 • ISSUE 7

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ON TEST: OLLO / RND / SSL / WAVES / RME / VANGUARD / UA / ARTURIA / ROLAND / MELBOURNE /SONICCOUTURE

TECHNIQUE: SPECTRAL BALANCE / CREATIVE DELAYS IN LOGIC

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LE ADER

CORE AUDIO I

n November, this magazine turns 40, and it says a lot about SOS that despite having been part of the team for two-thirds of that time, I’m not even close to being the longest-serving employee.

When I joined the company in 1998, a new world was dawning. Steinberg had introduced Cubase VST, and my G3 Power Mac was the first personal computer capable of natively recording and mixing multitrack audio in real time. Frustratingly, though, for most of the next two decades, it felt as though the progress gained courtesy of Moore’s Law was nullified by the ever-increasing demands of music software. As clock speeds rose, memory expanded and bandwidth was opened up, plug-in developers responded with real-time convolution, sample streaming from disk and other envelope-pushing technologies. It always seemed to be that you needed a cutting-edge computer, even if you couldn’t quite afford one.

Today, at last, it feels as though we’re coming out the other side. As this month’s in-depth look at the current generation of Apple machines confirms, most audio-related technologies no longer tax modern computers all that much. If you’re making music in Atmos, or your template features hundreds of tracks of virtual orchestral instruments, you’ll be all over the new M3 Ultra Mac Studio. But if you’re merely doing what would have been considered an epic mixdown back in 1985, a Mac mini will shoulder the load with barely a shrug.

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The fact that this sort of power is now available for a three-figure sum is incredible, but it’s not just hardware development that has made computer audio better. Back in the day, it felt as though you had to be an elite hacker just to route audio from one program into another, or install a soundcard driver. That began to change back in 2003, when macOS 10.3 quietly introduced a new API for sound handling. Core Audio has since undergone more than 20 years of continuous development, and is now a mature technology. It’s a remarkable and underappreciated achievement, taking care of everything behind the scenes with effortless grace.

And if there’s a key takeaway from Mark Wherry’s benchmarking exercise in this issue, it’s that hardware breakthroughs such as Apple Silicon don’t happen in isolation. The revolution in computer audio is a holistic one that has been enabled as much by vertical integration, OS-level support and long-term planning as by breakthroughs in chip design. It’s been made possible because, whatever else they’ve done, Apple have been pretty constant in their commitment to understanding and meeting the needs of musicians. And when I say that’s one thing this magazine has in common with the world’s richest company, I speak from experience — all 26 years of it...

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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.

Cranborne Audio Brick Lane - a breakthrough design offering six distinct compressor styles in one module. “Stress” control dials in harmonic saturation. Stereo linkable & sidechain HPF. £599

Rupert Neve Designs Newton Channel - fully featured channel strip with Neve mic pre-amp, 3-band EQ and VCA compressor stages. Includes variable Silk for enhanced harmonics. Outstanding value. £1869

Empirical Labs Distressor EL8XS highly versatile compressor delivering a wide range of tones from gentle/transparent to aggressive sonic annihilation. There’s very little this modern classic can’t do. £2974

Tube-Tech CL-1B - this hugely popular valve opto compressor has a rich, smooth and musical sonic footprint, making it the gold standard for vocal compression. Hand made in Denmark. £3120

Neumann U87Ai - setting the standard for over 50 years, U87 remains as popular as ever. Its flattering detail and gentle mid-lift beautifully captures vocal and acoustic instrument performances. £2545

Manley Reference Cardioid - modern valve microphone with a rich, clear tonality and open, present top end. An excellent choice for vocals, guitars and overheads.

Made in the USA. £2649

Flea 47 - meticulous replica of the iconic U47 valve microphone. Every component is made to the highest standard, making this our first recommendation for a “47 type” model. Exceptional. £3679

BeezNeez B67-269 - offers the classic tones of the U67 valve mic with additional voicings for increased versatility. Hand made in-house K6 capsule and Brimar Mullard designated EF86 valve. £1750

Kii Audio Seven - revolutionary active monitor with cardioid dispersion pattern for reduced room reflections. Versatile connectivity for analogue, digital & network audio. Highly impressive. £7785

Barefoot Footprint03 - the classic Barefoot sound delivered in their most compact & affordable package. MEME technology offers multiple voicings from a single cabinet. Bestselling monitor. £2295

ATC SCM25A Pro MkII - a beautifully balanced speaker - excellent HF detail, highly revealing midrange, extended LF and accurate imaging. Made in the UK. One of our favourite monitors. £8749

Audeze MM-500 - specifically designed to deliver mixes that consistently translate well. Durable premium planar magnetic headphones that provide studio quality monitoring anywhere. £1495

Neve Genesys Black - highly specified and versatile hybrid console combining the sonics of a classic Neve analogue desk with comprehensive digital DAW integration.

Immensely powerful.

From £47,940

API ASM164 - 16-channel summing mixer delivering the legendary warm, punchy API sound in compact 2U rack format.

Inserts and pan on every channel with Mix A/B program buses. £3195

Gainlab Audio Lieutenant - advanced studio monitor controller with 4x sources, 3x outputs, cue loop with headphone o/p’s, talkback, pad, mute and integrated power management system. £2599

Universal Audio Apollo x4 Gen2 UA’s class leading range of Apollo interfaces offers an unbeatable combination of high quality AD/DA with UAD plug-in emulations of classic outboard. Bestseller. £2099

For over 20 years, KMR Audio have been supplying the very best recording gear on the planet. We take pride in our unrivalled range of audiophile equipment in stock and on demo. Visit our North London store or contact us by phone/email for unbiased advice to help you choose the best tools for your needs.

Errors & Omissions excepted

KMR

020 8445 2446 • sales@kmraudio.com • www.kmraudio.com • 1375 high road, whetstone, london N20 9LN

30 EVGUENI & SACHA GALPERINE

IN THIS ISSUE

www.soundonsound.com

May 2025 / issue 7 / volume 40

WIN

FEATURES 30 Evgueni & Sacha Galperine The fearless sonic experimentation of composers Evgueni and Sacha Galperine is proving equally at home in mainstream and arthouse cinema — not to mention last year’s surprise hit TV series Baby Reindeer.

48 How I Got That Sound Jürgen Engler explains how a homemade sequencer and EHX Big Muff were at the heart of Die Krupps’ influential 1981 track ‘Wahre Arbeit Wahrer Lohn’.

50 Mixing Breakthroughs:

Moments, Mindsets & More

We invited five top engineers to share some of the key ‘aha’ moments in their mixing journey, and to offer advice for those who’ve recently embarked on their own.

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68 Inside Track: Jordan Fish

PAGE 74

Metalcore presents unique production challenges — and few are better placed to meet them than genre veteran Jordan Fish.

86 Talkback

120 Spotlight: Stereo Mics

Rapper Teether and producer Kuya Neil on the thinking and influences behind their new album YEARN IV.

There’s a huge array of dedicated stereo mics on the market.

We round up some of the best current options.

106 Spectral Balance:

136 Q&A

Tips To Help Your Mix Translate

Subtle issues with the frequency balance can stop mixes from translating well to different listening systems. We offer a couple of suggestions that can help.

Your studio and recording questions answered.

138 Why I Love... Logic’s Sampler SOS’s Paul White tells us of his enduring love for Logic’s Sampler.

56 MELBOURNE INSTRUMENTS ROTO-CONTROL

C O V E R

ON TEST 62

Allen & Heath Avantis Solo Digital Mixing Console

88

Apple M4 & M3 Ultra Macs Desktop & Laptop Computers

102

Arturia KeyLab 88 MkIII Controller Keyboard

82 AudioSmile Little

British Monitor

56 Melbourne Instruments

Roto-Control

Motorised MIDI Controller

118

Modulation Plug-in

22 8

99

Open-back Headphones

98

Sample Library

119 76 40

A-D/D-A Converter

14

Delay & Chorus Effects Pedals

M-Live B.Beat Pro 16 Plus Maono PM500 Cardioid Capacitor Microphone

Rupert Neve Designs RNDI-M Active Transformer DI Box

100

Live Playback System

118

12

RYK Envy Machine Eurorack Module

16

Soniccouture Flowstate Software Instrument

Submarine Pickups SubSix Hexaphonic Guitar Pickup

58 Universal Audio

UAFX Enigmatic ’82

Guitar-amp Emulation Pedal

44

Vanguard V24 Stereo Valve Microphone

20

Waves Sync Vx Pitch & Time Alignment Plug-in

125 Zero-G Ethera Gold Odyssey —

Angelic Voices

Roland V-Stage 76 RME ADI-2/4 Pro SE

Spitfire Audio Tutti Sample Library

Sample Library

Stage Keyboard

Expert Sleepers Disting NT

36 Gamechanger Audio

108

Phantom Sounds Analog Drift Delay Plug-in

Cutlasses Instruments Gloop

Auto Series

125

Overloud TH-U2 Amp-modelling Plug-in

Song Athletics Bones Sample Library

Solo Strings

Sample Library

112

Eurorack Module

124

114 Orchestral Tools Berlin

Iremia Intimate Cello & Intimate Viola

Eurorack Module

OLLO Audio X1

SSL Alpha 8 A‑D/D‑A Converter

Neve 88C Dual-channel Dynamics Processor

Active Monitor Speakers

124 Bunker Samples

Minimal Audio Wave Shifter

26

18

Zynaptiq Punch Transient Shaper Plug-in

WORKSHOPS 126 130 132 134

Studio One Pro Tools Cubase Logic

ON TE ST

OLLO Audio’s eco-friendly headphones certainly stand out from the crowd. Do they sound as good as they look?

SAM INGLIS

T

here are lots of ‘boutique’ manufacturers making high-end outboard, mics and monitor speakers. In part this is because although components such as loudspeaker drivers and microphone capsules are difficult for small outfits to produce in-house, they are readily available as OEM parts. Boutique headphone manufacturers are much more unusual, but OLLO Audio definitely belong in this category. Based in Slovenia, OLLO have a very individual philosophy which is centred around meeting the specific needs of musicians and producers, rather than the wider hi-fi market, in the most sustainable way possible.

Consequently, you’ll search in vain on the OLLO website for Bluetooth or active noise cancelling.

What you will find, however, is a full set of spare parts still available for every model that OLLO have ever produced, right back to the original open-backed S4 and closed-back S4R, reviewed in SOS January 2020 (www.soundonsound.com/reviews/ ollo-audio-hps-s4-s4r). These are now discontinued, and OLLO’s efforts are now all focused on a new model they call the X1. This is available only in an open-backed format.

High Calibre Perhaps the most significant development that has taken place since 2020 is that OLLO have formed a partnership with headphone correction specialists dSoniq. Each pair of X1 headphones is paired with an individual calibration file, created using “the power of AI processing”, and a licence for dSoniq’s USC II. This is a cut-down version of their Realphones software, and the calibration file can also be uploaded into the full version if you have it. Better still, since the calibration file is supplied in the standard XPS format, it can also be loaded into hardware audio interfaces from RME, Neumann, UA, MOTU and so on.

Handbuilt in small batches, the X1 is a more or less conventional moving-coil design, but the drivers are housed in distinctive circular earcups made from wood. As on the S4, these are suspended symmetrically from the headband, meaning that they are not ‘handed’ as such; the right earcup is whichever one you connect the right side of the supplied Y-shaped cable to. And, unfortunately, the indication of left and right on the cable itself is still needlessly hard to discern.

At the same time, the X1 also differs from the S4 in several important respects. The pegboard-style outer surface of the earlier model’s earcups has been replaced by a new design, where sound exits

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OLLO Audio X1 Open-backed Headphones through a cutout shaped like a stylised letter X. The wire headband on the S4 has been abandoned in favour of a metal strip with cutouts in it, presumably with the aim of reducing the unwanted resonances that made the earlier design problematic.

And whereas the S4 models came only with a soft bag, the X1s are supplied with a proper, semi-rigid fitted carry case with handle.

OLLO don’t just talk the talk when it comes to sustainability. They back up their philosophy with

a limited five-year guarantee, and say that all components on the X1 should be replaceable using simple home tools.

The apparent build quality would suggest that their confidence is justified; perhaps because of the new faceplate design, the X1s retain the very individual character of the S4s whilst shedding any vestige of a ‘homemade’ aesthetic.

Drive Time Internally, the X1s employ “custom-made low-impedance drivers”, with a 50mm membrane that has a “composite Pu+PET patented pattern”. It’s not clear whether these are actually manufactured

by OLLO themselves or simply made to their specifications, but either way,

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ON TE ST

OLLO AUDIO X1

they are different from the drivers used in previous OLLO models, and deliver good numbers. THD and second-order harmonic distortion are given as less than 0.05% and 0.02% respectively for a 1kHz sine wave at 94dB SPL. Nominal impedance is 32Ω, and the X1 will output 101dB SPL for a 1kHz tone at 1V RMS. These figures are not so very different from those of the original S4, but that tells us little about how the two models sound. To my ears, the S4 had a rather unusual sound signature, with a prominent ‘upper bass’ and a noticeable suckout in the midrange. Their other good characteristics, such as low distortion, made them promising candidates as mixing headphones, but only with the help of software frequency correction. That wire headband was also troublesome, with even slight touches inducing very audible, undamped ringing.

In comparison, there’s a night and day difference with the X1. Firstly, the new flat headband still transmits mechanical noise, as all such types do, but it’s now effectively damped, so you’re never left wondering whether the resonances you’re hearing are present in the source or not. And secondly, the headphones themselves sound completely different.

In fact, when I first got them out of the box and tried them, I hadn’t yet found out about the bundled calibration options, and my first thought was that they sounded really quite flat and neutral in their raw state. There was none of the aggressive upper-mid splashiness or ‘smile curve’ tonality that you so often

find in even quite high-end headphones, especially models designed for music listening as well as studio work. The only slight departure from neutrality I could readily detect was a hint of additional warmth somewhere in the 300Hz region.

Registering the X1, downloading the XPS file and installing USC II then made a subtle but wholly positive improvement, taming that slight midrange thickness, firming up the upper midrange and bringing up the low bass a shade.

Loading the XPS file into the full version of Realphones revealed that only four EQ bands were active: a 2.3dB cut at 364Hz, a comparable boost at 1600Hz, and equally gentle shelving boosts and cuts at the low and high end respectively. It’s an interesting alternative approach to something like Sonarworks’ SoundID Reference, which generates highly granular EQ curves; whatever the pros and cons, the corrected X1 felt both natural and confidence-inspiring to work with. Even uncorrected, they were closer to my perception of ‘flat’ than many other open-backed designs.

“The corrected X1 felt both natural and confidence-inspiring to work with. Even uncorrected, they were closer to my perception of ‘flat’ than many other open-backed designs.”

USC II also gives you a limited but useful subset of features from the full version of Realphones, most notably a selection of target curves and binaural room simulations that includes a virtual studio control room. Like the full version, it will run either as a plug-in in all the major native formats, or as a ‘systemwide’ standalone version with a virtual audio driver.

Counter Culture As I noted at the start of this review, if you’re hankering after a new mic, a fancy equaliser, a desktop synth or a pair of speakers, there’s plenty of leftfield and niche options out there to complement the established names.

If your quest is for a new pair of studio headphones, however, the field is much narrower. With the X1, OLLO Audio have proved that a start-up with original ideas and an individual philosophy can absolutely compete head-on with the titans of headphone monitoring. These are every bit as good as other models I’ve tried in the same price bracket; they’re built to last and backed with a five-year guarantee; and although they barely need software correction, you get it thrown in for free. Impressive!

summary The X1s are sustainable, good looking, and deliver a mixing-friendly tonality that is impressively neutral.

The icing on the cake is the free software calibration that fine-tunes their sound even further if needed.

The bundled USC II calibration package is a cut-down version of dSoniq’s Realphones.

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May 2025 / www.soundonsound.com

£ £453 including VAT. W www.olloaudio.com

The Ultimate iD Interface

24in | 32out Audio Interface 8 x Class-A Audient Console Mic Preamps | 8 x Switchable Balanced Inserts 2 x Discrete JFET Instrument Inputs | Main & Alt Speaker Outputs | 2 x ADAT Input & Output Ultra-low Latency Software Mixer | Dedicated Alt Speakers & Talkback Controls 2 x Independent Headphone Outputs | Class Leading AD/DA Converters

ON TE ST

Ever wished you could process each of your six strings differently? With the SubSix, you can...

PAUL WHITE

W

e occasionally see hexaphonic pickups on MIDI guitars, or guitars that are set up for use with Roland or Boss guitar modellers, but Submarine’s SubSix has been designed with a different purpose in mind: it provides six separate audio sources from the six strings of a guitar so that each one can either be recorded separately, or passed through different pedals to create effects that just aren’t possible with a regular guitar.

Submarine Pickups Hexaphonic SubSix Guitar Pickup

Overview The SubSix pickup has small sliders on the rear to help you set it correctly for the string spacing of the instrument to which it’s attached (the range is 45.5 to 51.5 mm), and once fitted using the included adhesive strips that come with the comprehensive mounting kit, its height can be adjusted using an Allen key (also provided). The signals exit the pickup on two thin cables, which connect using a pair of TRRS mini‑jacks at the top and bottom of the pickup.

Each one terminates in three colour-coded quarter-inch jacks at the other end. This arrangement allows the cables to be routed either side of the neck and to the rear of the body, where they’re out of the way.

The pickup should be fitted to an electric guitar in between the existing pickups, rather than (as with guitar synth pickups) very close to the bridge. It can also be mounted across the soundhole of an acoustic guitar. With a thickness of

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May 2025 / www.soundonsound.com

just 9.3mm when mounted on its base plate, the pickup should fit beneath the strings of most electric guitars and pretty much all acoustic guitars, and without the base plate, the height under the two E strings is 6.3mm. The pickup impedance is around 2.4kΩ per output, which is a little lower than most conventional pickups, but the powerful neodymium magnets ensure that there’s adequate level available, and there’s a benefit to the lower impedance: when this pickup is fitted to an acoustic guitar, the high frequencies come across in a more natural way than from typical magnetic pickups.

Possibilities So, why exactly might you want separate outputs from all six strings on a regular guitar? There are several possibilities.

For instance, feeding them into a simple mixer allows the strings to be panned across the stereo field, which can sound interesting. And plugging into an audio interface that can accommodate six inputs allows separate plug-ins to be applied to each string. If using a mixer and, like me, you have rather more effects pedals than you actually need, it’s also possible to use different effects pedals on each string.

In the plug-in world, or course, it is no problem to call up six identical processors, for example compressors, if you need them and if excess is your thing, there’s not much to stop you. Actually compressors are a good example to start with: the result from compressing individual strings

is audibly different from that of compressing a mix of strings, generally leaving you with a better-defined note attack when playing chords. The sound from the pickup is clean and neutral, so its also a good starting point for adding effects such as pitch‑shifters, perhaps to drop the pitch of the bottom one or two strings, or modulation effects, with which you might treat only the high strings. While I did notice a little crosstalk between adjacent strings, it wasn’t lots, and it shouldn’t be enough to upset a decent pitch‑shifter.

Hex fuzz is also worth exploring. Having a separate fuzzbox for each string results in much cleaner note definition, and without the intermodulation that you hear when playing chords through a fuzzbox with a standard guitar. Add some global top-cut EQ, a dash of chorus and a sprinkling of reverb and the result can sound almost like a vintage string machine!

The Joy of Six I’ve run through just a few suggestions here, but really the point is that there’s no limit to creative permutations. So the SubSix hexaphonic pickup should be particularly appealing to those involved in experimental and ambient music — which includes me! It works well, it looks good, it’s easy to fit, and the price isn’t bad either. A pickup that could be well worth picking up!

summary The SubSix doesn’t cost any more than many conventional boutique pickups, yet it opens up so many creative possibilities, whether you’re playing live or in the studio.

£ £319 (discounted to £239 when going to press).

Prices include VAT.

E support@submarinepickup.com W www.submarinepickup.com

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ON TE ST

Rupert Neve Designs RNDI-M Active Transformer DI Box Boasting a custom Rupert Neve‑designed transformer, one of the best-sounding DI circuits out there just got more affordable.

BOB THOMAS

I

t’s almost a decade since I reviewed Rupert Neve Designs’ (RND’s) first active DI, the RNDI. Including the newer two- and eight-channel variants that followed, RND have now sold some 35,000 channels (and counting!) of that circuit. But it’s always been priced for professionals...

Now, the company’s RNDI-M Active Transformer Direct Interface puts the same Class-A, FET-based input stage and custom transformer in a more pedalboard-friendly

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May 2025 / www.soundonsound.com

box for a significantly lower price, something made possible by the decision not to cater for power-amp outputs (if that’s important to you, the RNDI remains available).

Outside & In The RNDI-M’s rectangular metal enclosure has a classy black and light grey aesthetic.

Paralleled in and thru quarter-inch isolated jacks and a 48V phantom power indicator LED are on the front. On the rear are a balanced output XLR and a switch to lift the pin 1 chassis ground. Inside, the

phantom-powered circuitry sits on a single PCB carrying the connectors, LED and switch, along with the input stage and the transformer, which isolates the unbalanced input and balances the output.

The input stage accepts levels up to +20.5dBu, so the RNDI-M can handle any electric guitar or acoustic preamp, modeller, effects pedal, modular synth, keyboard, drum machine or (if you gain stage correctly) other line-level source I can think of.

The 2.2MΩ input impedance also ensures that passive guitar/bass pickups see a suitably high load, even if connected in parallel to a typical amp or pedal. The voltage loss inherent in such a configuration is negligible, and is in any case due almost entirely to what you connect to the thru output. In other words, not only does the RNDI-M correctly load passive guitar/bass pickups, but also, all but the tiniest percentage of the voltage produced by the pickup reaches both the RNDI-M and any connected amp or pedal.

The noise floor (-110dBu) and frequency response (28Hz-60kHz ±0.25dB) are also impressive, and ensure that the RNDI-M sounds quiet and transparent. Importantly, that allows the subtle sense of body and warmth that’s created by the transformer to really shine.

With an output impedance of only 40Ω, this transformer is also capable of driving long cables with minimal signal loss, at output levels up to a maximum of +11.5dBu.

guitars and basses and delivering detailed results from my growing collection of synths — all of which I’d expected — it underlined the individuality of my acoustic guitars and the various piezo pickup/preamp combinations I use.

Much as I love the sonic character of Rupert Neve’s vintage transformer-based designs, I believe that much of his best work was done later in the USA, and the R-5052 transformer is a case in point.

As with the original RNDI, the subtle but distinctive sense of warmth, definition and depth is extremely attractive, both sonically and musically, and it becomes more pronounced at increased signal levels as the input stage and transformer increase their contribution.

The RNDI-M’s sonic performance places it amongst the best active DIs that are currently available. My own go-to DI is a similarly priced phantom-powered, FET-based, device, but it’s transformerless and, wonderful though it is, it can sound somewhat ‘clinical’ compared with the RNDI-M. The one DI that I have which can keep up with the RNDI is a valve-plus-transformer 500-series module costing over three times what the RNDI-M does — so the RNDI-M is priced competitively too.

If you want a high-quality, transformer-based active DI box that also offers ‘a certain something’ sonically, the new RNDI-M should definitely be on your shortlist.

Boxing Clever?

Equally at home in the studio or on stage, the RNDI-M delivers a superb sonic performance, conveying detail, clarity and precision, while also lending a sense of warmth and weight.

As I’d expected it would, the RNDI-M performed impressively, conveying the sound of connected instruments with clarity, depth and precise detail. In fact, I couldn’t fault it: as well as accurately reflecting the characters of various electric

summary

£ £199 including VAT. T Rupert Neve Designs +44 (0)208 191 0058

E info@rupertneve.com W https://rupertneve.com

ON TE ST

Soniccouture Software Flowstate Instrument

Transform your samples into soundscapes with Soniccouture’s new software instrument.

MARK GORDON

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lowstate, the latest plug-in from the prolific Soniccouture stable, is an NKS-ready Kontakt Player instrument designed to morph and manipulate any audio clip into a malleable, shifting musical form. Although the plug-in ships with 100 presets and samples, the idea behind it is for you to use your own audio clips to create unique sonic soundscapes that have an almost living, breathing quality.

The main screen is ominously devoid of any parameters, aside from a single, large control knob, a graphic representation of the current waveform, and five mysterious symbols that would look more at home on the side of an alien spaceship. Working with single waveforms (there are no velocity layers or multisamples), Flowstate allows you to load

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any WAV file and, by way of eight tuned formant filters, introduce a pitched element that can be morphed and manipulated in real time using the central control knob. Each sample can have five different states, represented by the five symbols on the main screen, and you can use keyswitching or selection via the mouse to flow between them... hence the name Flowstate.

Flow Motion There is, of course, more to Flowstate than its minimalist initial screen might suggest, and clicking the ‘Editors’ button within any of the five states reveals the Harmonics and Synth editors at the heart of the creative engine.

The Harmonics editor features eight tuned formant filters that can be assigned to a range of harmonic frequencies: 1 is the fundamental, 2 is the second harmonic, 3 is the

third harmonic, and so on.

Negative numbers represent subharmonics. There is a musical methodology behind this, and by isolating a single fader and changing its value you can hear how the waveform is affected, but I found that simply changing values and listening to the resulting interaction between the filters was the best way to achieve a pleasing musical result.

Each state includes an A and B harmonic variation that can be edited independently, and an on-screen slider, mapped to the central control knob (which is, by default, mapped to your keyboard’s mod wheel), allows you to morph between the two configurations. Increasing the level of the Purity fader to the right (which can also be mapped to the control knob) adds a resonance to the formant frequencies, which introduces a unique tonal quality in the audio.

The Synth editor is a little more conventional, offering the more familiar ADSR envelope, filter and LFO parameters, many of which can be mapped to the main control knob.

You can also use the LFO to modulate the formant filters, which can create stunning oscillating textures.

Unlike the Harmonics and Synth configurations, which are unique to each of the five states, the FX editor applies its output across all states simultaneously, which adds consistency and uniformity.

Three simultaneous effects are on offer, including the usual selection of modulation, distortion, delay and reverb options, but, as you might expect from Soniccouture, they’re in no way run-of-the-mill in their application.

Conclusion It’s difficult to describe the type of sounds you can

achieve with Flowstate.

The potential source material is limitless, and the manipulation options are vast but it would be safe to say it lends itself to the more esoteric end of the sound spectrum. It has the ability to create evocative and complex pads at its heart but can also produce searing lead voices and pulsating rhythmic textures.

At first glance, its almost otherworldly design, with ethereal main screen and what look like alien symbols or ancient runes, makes it difficult to work out exactly what this plug-in is all about, and that’s one of the driving forces behind its design.

Creativity in music can sometimes be displaced by too much technical detail, and an air of mystery can stimulate a greater level of creativity, brought about by experimentation.

Flowstate enables you to take anything from a drum loop or vocal phrase to a recording of a mountain stream or traffic noise and create unique, evolving musical soundscapes without really knowing what you’re doing! The ability to morph between the five harmonic states in real time, with additional manipulation via the central control knob, allows for an endless stream of sonic possibilities that are singular and absorbing.

summary Beneath its enigmatic design, Flowstate hides an array of tools that enable you to transform any WAV file into a unique and evolving musical soundscape. Its tuned formant filters, powerful synth engine and extensive real-time control can morph a waveform beyond all recognition, offering a limitless supply of stunning audio textures.

£ £89 including VAT. W www.soniccouture.com

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Apollo X Gen 2 sounds better.

Go behind the scenes with UA Technical Fellow Dave Rossum.

©2025 Universal Audio Inc. All trademarks property of their respective owners.

Apollo Monitor Correction by Sonarworks® Elite-class AD/DA conversion for superior for mixing with confidence audio resolution

Unison preamps for genuine Neve, API, SSL, and Manley tones

Universal Audio Est. 1958

ON TE ST

Zynaptiq Punch Transient Shaper Plug-in

With its many-band approach, Punch is smarter than your average transient shaper.

An adjustable release time allows for the creation of very short and snappy attack transients, while at the other extreme it can deliver smooth dynamic expansion, even helping to reverse the effects of overcompression.

Punch allows for long envelope times without the envelopes contributing to a global increase in loudness, but just in case, the plug-in incorporates a switchable output clipper and gain compensator.

A grid of ready-made frequency contour profiles can be auditioned quickly by holding the mouse pointer over one of them, and there are easy macro adjustments in the form of the Bias and Punch controls. Punch determines the intensity of the processing, while Bias deepens the shape of the frequencies being affected, and can go positive or negative, allowing transients to be exaggerated or tamed. Additional controls for Sub and Air add in up to 12dB of boost to the processed signal before the final wet/dry mix stage. If the presets don’t do what you need, a Custom mode allows users to create their own frequency profiles, using eight vertical sliders that operate much like a graphic EQ, with frequencies arranged in octaves from 85Hz to 11kHz. MIDI parameter control is available, along with preset change using fixed MIDI CC designations (CC116 to CC119).

In Use

PAUL WHITE

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ransient shapers allow you to add impact to the attack of a drum or plucked instrument — or, when necessary, to tame transients to make them sound less aggressive.

Unlike compressors, they’re not level-dependent, so a well-designed one works consistently across a wide range of input levels.

Sometimes, though, the results can still sound heavy-handed, and this is where Zynaptiq’s Punch comes in. Described as an “intelligent dynamics contouring” plug-in, it technically belongs in the same category as other transient shapers, but it’s more sophisticated and gets around that ‘heavy‑handedness’ issue by making the process frequency dependent.

Supporting 64-bit AAX Native, AU, and VST3 plug-in formats for Mac and Windows hosts,

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Punch adds just five samples of latency and supports sample rates up to 192kHz. Authorisation (two authorisations are included) is via iLok either to the computer or a physical iLok 2 (or newer).

Overview Punch can be applied to individual tracks, submixes or even whole mixes, and its operation is underpinned by Zynaptiq’s expertise in source separation, signal statistics, frequency-domain processing and filter design. So you could think of it as a significantly ‘better mousetrap’ — essentially an intelligent transient shaper that can be adjusted to suit the material that’s being processed. There are two algorithms to choose from: Smooth (which operates on 100-plus bands) and Crystal (10 bands). Neither mode seemed overly CPU-hungry to me. For multi-channel applications, up to 16 channels can be set up and controlled from within the plug-in window, and because the processing has a frequency-dependent element, it can be tailored to just about any style of material, from drums and bass guitar to synths and acoustic guitar: if the sound has a transient attack, Punch can deal with it.

Once you’ve decided what you need to achieve, using Punch is very straightforward. Just ask yourself if your transients need to be made stronger or weaker, and whether the problem areas are at the low end, in the midrange or in the highs. In many cases, one of the existing profiles will get the job done. A kick drum may benefit from added impact to the mids or lows, but if you also want to enhance the ‘slap’, it’s easy to create a Custom profile to achieve that. As the Custom faders can be set to negative or positive values, you can make very precise sonic adjustments, such as beefing up the lows of a kick drum but also pulling back some of the 170 to 340 Hz low mids, to avoid a boxiness or boom, and you can still boost between 2.7 and 5.4 kHz to keep the slap sounding tight.

The variable release time means the transients can be kept short and snappy or lengthened, so in practice any instrument that has a sharp attack, such as picked guitar, bass, percussion or piano can be treated. Best of all, Punch manages to adjust the character of the sound being treated in a very natural-sounding way, without the end result ever really sounding as though it has been processed — it avoids the audible artefacts sometimes evident when using a conventional transient shaper.

It may be simple to use, but Punch is both extremely sophisticated and very useful, with a great many potential applications. Highly recommended.

summary This sophisticated but effective and easy-to-use transient shaper adapts to the source material to overcome the problems of traditional processors.

£ $99 W www.zynaptiq.com

ON TE ST

This new ARA2 plug-in aims to make aligning doubles and backing vocal stacks a breeze.

LUKE WOOD

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onceived for use in music projects, Sync Vx has been designed to automatically align the timing and match the pitch of multiple vocal takes, and to tackle everything from double-tracked vocals to large groups of harmonies and ad-libs. The manual mentions ADR as another potential application, though. It’s exclusively an ARA2 plug-in, and if you’ve not used this format before it’s important to understand that placing Sync Vx in an insert slot as you would a typical VST or AU plug-in won’t work. You’ll need to look up how ARA2 plug-ins are handled by your DAW of choice too. Sync Vx is currently officially supported in Pro Tools 2024.10, Logic Pro 11, Cubase 13/14, Nuendo 13, Reaper 7, Studio One Professional 6/7 and Cakewalk Sonar 2024.11.

Waves Sync Vx

Pitch & Time Alignment Plug-in

Getting Started To get started, you load the plug-in on each track requiring alignment. The first time you open it, a setup wizard demonstrates how to define which tracks will be used as a Ref (reference) and which will Sync. It also explains how the key controls work.

Up to 16 mono tracks can be processed at once, and you can designate any four of them as Ref tracks, so the plug-in can be put to work on everything from simple vocal doubles to more complex, multi-part arrangements.

Sync Vx’s GUI resembles a self-contained DAW timeline, each channel of audio getting

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its own dedicated waveform display and per-channel mute, solo and level controls, and there’s independent control over the amount of time-alignment and pitch-correction applied to each Sync. A colour-coding system makes it nice and easy to see which Syncs are assigned to which Refs and, helpfully, the track names all mirror their DAW counterparts.

It’s possible to customise the track layout, thanks to a pair of buttons at the top. Selecting DAW will arrange the tracks in the same order as they appear in your DAW session, while clicking REF will place all Sync tracks below their associated Ref tracks. You can also click and drag tracks on the timeline to put them in any order you wish, and an ‘eye’ icon can hide tracks. I like to keep my sessions well organised, so these are all very welcome features.

Into Alignment Waves say the initial processing time can vary between “a few seconds to a couple of minutes for heavy sessions”, and a loading bar at the top of the plug-in window indicates progress before, eventually, declaring that the tracks are In Sync. By default, the plug-in will apply 100% time-alignment to all of the Sync tracks, but this can be adjusted either globally using a Time Tightness dial at the top of the window, or as mentioned earlier, on a per-track basis from the left-hand side of the timeline. The total amount of Time Tightness processing is calculated based on the values of the global, track and region controls, so if you set the global value to, say, 70%, an individual track set to 50% will end up with a total correction value of 35%. The ideal settings depend on the

material and what sort of sound you’re looking for but, generally speaking, I found that Time Tightness values of around 70‑80% tidies things up nicely, while still leaving the parts with a ‘human’ feel.

There are two time-alignment algorithms to choose from: TimeWarp A and B. The former, the default mode, is intended to be suitable for the majority of sources, while the second is effectively a backstop option, to be used where you find that A results in unwanted artefacts.

I did encounter an issue during my tests, where the phrasing of a particular word on a Ref track was causing some odd-sounding glitches in one or two of the associated Syncs and, sure enough, switching those tracks to TimeWarp B solved the issue instantly.

Another neat feature is that any regions you create on Sync Vx-enabled tracks on your DAW timeline will be automatically created within the plug-in itself, making it possible to apply different processing to different sections of the same track; once a region has been created, a pair of ‘three dot’ icons appear at the edges of the corresponding Sync Vx region, providing access to Time Tightness, TimeWarp algorithm selection, Pitch Tightness, Transpose and Formant settings (we’ll get to those last three in a moment). So, if you have a part with sections that each require

a different approach, you can avoid having to split them across multiple tracks to process them individually.

Tuning & More Although its main purpose is to improve the timing between multiple vocal or dialogue tracks, Sync Vx has a few more tricks up its sleeve. Along with its Time Tightness control, a Pitch Tightness function can be used to ensure that the Sync and Ref tracks remain in tune with each other. Its implementation is similar to that in Synchro Arts’ VocAlign range: the plug-in doesn’t provide full-on chromatic pitch-correction, but rather detects the pitch of the part on the Ref track, and then applies correction to each of the Sync tracks to make them match the Ref. As with the Time Tightness control, Pitch Tightness can be applied globally or to individual tracks/regions, and there are two algorithms to choose from. This time, they’re Absolute, and Absolute Ignore Vibrato — the latter is, as you might expect, intended to remove any artefacts or unnaturalness being caused by matching the vibrato pitch.

Considering that I was working with some very well-performed vocals, I was surprised at quite how much I ended up using the Pitch Tightness control. The choral ‘oohs’ ensemble in particular really benefited from some subtle adjustment in parts. But you don’t have to limit yourself to subtle moves: Sync Vx makes it possible to transpose Sync parts by up to ±12 semitones (the control’s resolution can be set to semitones or cents), and it can apply formant‑shifting too. It’s safe to say that these features didn’t suit the particular track I was using for most of my

tests, but they work well, and I can imagine that those who like to get a little more experimental with their vocals would find a lot to like about them.

The whole point of Sync Vx is to minimise the need for manual edits, but there are a few tools that allow you to give it a helping hand if it doesn’t get things quite right on its own. For example, if an element in the Ref track is throwing off the plug-in’s calculations and resulting in incorrect Sync track adjustment, highlighting the problem area and selecting Ignore will exclude it from the detection process, while leaving the audio itself intact. Similarly, you can make a section of a Sync track immune to any changes using a bypass command.

Once a Sync track has been aligned, it’s also possible to create Sync Markers, a pair of adjustable borders that allow you to define how much audio either side of a sync point is included in the time-compression/expansion process.

Helpfully, all of these commands are tied to keyboard shortcuts, as well as right-click menu options. I found the automatic alignment to be very accurate indeed, and there were only one or two moments where I needed to apply a bit of fine-tuning — the glitches I mentioned earlier were also easily solved by employing the bypass feature on the offending Sync tracks. Aware that my test material was all in pretty good shape to begin with, I decided to up the stakes a bit by nudging some tracks around a little, but the plug-in still did a great job of getting them lined up.

Finally, there’s a helpful DeNoise function, which applies its processing exclusively to the reference side-chain

signal — it does not remove noise from the Sync or Ref tracks themselves, but rather prevents noise from throwing off the correction algorithms that are used to sync the tracks. It can be set to On, Off or Auto. Waves recommend turning it off when there’s no noise present, because it may result in longer processing times.

Again, the main sessions I used to test out Sync Vx were well-recorded with no noise issues, so this particular setting remained in Auto or Off, but it’s certainly a thoughtful feature.

Stacking Up Sync Vx’s main competition comes from Synchro Arts, not least VocAlign.

If you only need the basics of time alignment (no pitch-correction, formant‑shifting, multi-part editing etc), then VocAlign Standard may well tick all the boxes you need for a lower price. But Sync Vx’s features mean it’s more directly comparable with the Pro version, and both Sync Vx and VocAlign Pro are more than capable of delivering the goods, although they vary in their layout and the way they handle processing multiple sources.

For me, Sync Vx really came into its own when I was working across multiple parts on more complex projects. Its multitrack view is really handy, and its DAW-like appearance meant I got to grips with it a little quicker than when switching between and keeping track of Process Groups in VocAlign’s fixed three-lane layout. Being able to define four Ref tracks and see them all in one window with their associated Syncs is a powerful feature, and I preferred Sync Vx’s GUI for working with busier arrangements. The only real down side was that my test session exceeded the 16-track limit, so I needed to render the results and work on the remaining tracks separately. Still, it’s a lot quicker than doing it all manually! There’s plenty to like about Sync Vx, and if your editing and mixing work involves a lot of multi-part vocal arrangements, this latest addition to the Waves family is well worth your attention.

summary A powerful and effective pitch- and time-alignment ARA2 plug-in, with a DAW-like appearance that makes it easy to use.

£ $199. Discounted to $129.99 Sync Vx excels when you’re working with multiple Ref and Sync tracks.

when going to press.

W www.waves.com

www.soundonsound.com / May 2025

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Neve 88C

Dual-channel Dynamics Processor Not only does this portable box grant you access to the 88R console’s channel compressor anywhere you go, but it’s also blessed with several clever features.

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hen it comes to compression, Neve are probably best known for diode-bridge designs such as the 2254. That was released in 1969, and variants of it remain popular today. It was Dave Blackmer of dbx fame who famously developed the use of voltage‑controlled amplifiers (VCAs) as the gain‑reduction element in compressors in the early 1970s, but Neve were fairly early adopters of this technology. The first product of theirs to employ VCAs was probably the 8108 console, released in the late 1970s, which used custom ICs

by Newmarket Transistors in an optional VCA fader system. By the early/mid‑1980s, Neve were working on in-channel compressor designs based on dbx’s 2151 VCA chips, which would first appear in commercial products such as the 5104

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console. They went on to use the modern descendant of that Blackmer VCA chip, THAT Corporation’s THAT2181, in the 88R console, which was introduced in 2001.

It’s the 2181 that takes care of gain reduction in the company’s new standalone compressor, the dual-channel 88C. Like the company’s consoles, the 88C is made in Neve’s factory in Burnley, Lancashire (England), where they go to incredible lengths to get the quality of construction right. For example, Neve scan all the solder joints to highlight any potential defects — they’d like, just as much as you would, for you never to have to return the product to them for repair!

Overview The 88C might look simple at a glance, but there’s some clever stuff going on behind the scenes, and because the various knobs double up as push

switches, there’s also more control at your fingertips than it might first appear.

For most, it will be the form factor that really catches the eye: the various knobs and switches protrude from the front and back of a case that measures 182mm wide, 183mm deep and 76mm high, and is encased in a hard-wearing leatherette finish, just like the company’s 88M audio interface, which we reviewed in SOS September 2022 (https://sosm.ag/neve88m). At 1.32kg (a shade under 3lbs), it’s not unduly heavy, but it’s still sufficiently weighty that its rubber feet kept it firmly in place on my desktop during my review tests, even with a total of seven cables attached to the rear.

In terms of the sound and technical performance of the circuitry, it’s of mixing/mastering quality, so while it obviously isn’t designed to sit in a conventional studio rack, it should be well at home in any studio. But the format makes it more versatile, to allow you access to high-quality analogue compression in pretty much any situation.

I could see freelance engineers taking it

with them to different studios, and it could also serve well in a home studio for music or dialogue, while the robust construction means it could be taken on the road, on stage or on location.

Unusually for a professional-level standalone processor, the 88C is powered using a USB-C cable. There’s no data or digital audio connection, but this means power can come from a USB3-equipped computer (arguably, the biggest advantage of USB3 for audio gear is not its greater bandwidth compared with USB2, but rather that it makes more current available to power attached devices) or from a phone-style USB adaptor that can supply 900mA or more. A suitable USB-C to USB-C cable is supplied. An adaptor isn’t, but given their ubiquity, that’s for the best in my view — none of us need more USB adaptors clogging up our studio storage or heading to landfill!

The provided cable fitted securely in its socket, but there’s no provision for securing it in place. You could fashion something yourself with, for example, a stick-on cable-tie base, but it might have been nice to have something built in.

There is, however, a K-lock socket so you can secure the 88C to a desk. Next to the USB socket is an on/off rocker switch, and with this in the on position and the compressor connected to a suitable power source, ‘boot up’ takes just a couple of seconds.

Also on the rear panel are six TRS jack sockets: a pair each for the main audio input and output, and the other catering for external side-chain inputs that make it possible to key compression independently for each channel. This all sits on a single rear panel, with venting

above the socketry, that’s held in place by four screws.

Features & Control The front panel is divided visually into three sections. The controls and metering for channel one are in the left zone, with an identical set for channel two in the middle, immediately to their right, and a single channel-link button between them. The meters indicate the input level (four LEDs, ranging from signal present to +20dBu) and gain reduction. For the latter there are seven LEDs, ranging from -0.5dB to -50dB, and with the top four LEDs accounting for the first 5dB, to give you greater resolution where you’ll most likely need it. As for the controls, each channel has four push-turn knobs, whose primary functions are to set the threshold level, the amount of make‑up gain, the compression ratio and the release time. There’s no variable attack control other than a F.Atk button, but the way this compressor handles the

Neve 88C £1074 pros • Rugged, portable and very versatile.

• Sounds great.

• Efficient and effective control.

• Clever adaptive attack and auto release.

• Stereo linking and side-chain inputs.

cons • No means of physically securing the power cable.

summary This clever dual-channel VCA compressor has so many potential applications, in the studio, on location and even on stage.

A supremely versatile design, it can even be powered from your USB3 laptop — you really could use it anywhere!

recalling settings. The release pot can be set anywhere from 0.03 to 3 seconds, while make‑up gain runs from 0 to +30 dB. But what of the attack, I hear you ask?

Neve have adopted the same clever Adaptive Attack approach that they use in the 88RS console’s channel compressor, whereby the attack time changes depending on how far the signal exceeds the threshold. In simpler terms, the 88C distinguishes between louder transients that will likely benefit from a faster attack, and the ‘body’ of a sound, for which a slower attack tends to avoid unwanted side effects. It’s not just cleverness for its own sake: it’s very effective in practice, and a big part of what makes the 88C suitable for compressing complex programme material, such as on the stereo drum or mix bus. In subjective terms, assuming you get the other settings right, it makes the compressor sound smooth, natural and effortless. In the default setting the attack times are 1.5ms for signals 4dB or more above the threshold, and 5ms for those that exceed the threshold by less. But press the button to engage Fast Attack mode (the F.Atk LED illuminates blue), and the compressor will react to any signals over 4dB above the threshold with a faster 0.1ms attack.

The release knob also has a secondary function: pushing it engages an Auto Release mode. Auto release features are fairly common on compressors, of course. If you want to understand how they work, our Technical Editor, Hugh Robjohns, did

“Because the various knobs double up as push switches, there’s more control at your fingertips than it might first appear.” attack time is pretty interesting and very effective. More on that arrangement later.

As befits a compressor that’s intended to serve in a variety of different use scenarios, most of the knobs have a generous control range. The threshold knob runs from +18 to -30 dBu, with zero being around the 11 o’clock position. The ratio pot ranges from 1:1 (no compression) to, effectively, limiting, and with 8:1 at the three o’clock position it’s easy to be reasonably precise when matching or

Unusually, the 88C is powered over USB3. Meanwhile, the balanced audio inputs and outputs are on quarter-inch TRS jacks, as are the key inputs.

www.soundonsound.com / May 2025

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ON TE ST NEVE 88C

a cracking job of explaining them back in SOS September 2014 (https://sosm. ag/compressor-auto-release), but it’s conceptually similar to the Adaptive Attack, in that the release time is programme-dependent. The 88C’s Auto Release mode is different from many, though. Usually, there are only two ‘bands’, whereas this design has three, and this again serves to make it suitably smooth-sounding on a wide range of different material. In addition, there’s an always-on, behind-the-scenes feature that prevents the familiar (and almost always undesirable!) ‘breathing’ effect that occurs after a sudden level drop to, in this case, 30dB or more below the threshold. Again, it’s clever stuff that contributes to this compressor’s overall smooth and elegant behaviour, and helps to make it easy to set up.

I said above that the panel was divided into three sections, but thus far I’ve only accounted for two! The third, farthest to the right and above the Neve logo, hosts a pair of seven-LED output level meters, one for each channel.

These indicate ‘signal present’ at -30dBu and thereafter levels up to +20dBu. So you have a total of three meters per channel on this device, which I reckon is pretty impressive given the amount of front-panel real estate that the designers had to play with.

Two final nice touches are the bypass and channel-linking facilities. To indicate the (hard) bypass status for each channel, Neve have chosen not to crowd the front panel with yet more LEDs, but have put the output meter LEDs to good use: normally colourful, each column of LEDs turns white to show that its channel is bypassed, whilst still displaying the signal level.

As for the linking, the user experience is pretty much that you hit the Link button to link the channels. Whichever channel’s control signal is higher will trigger gain reduction on both, as you’d expect.

But unlike with many linked analogue compressors, both sets of channel controls can be used. For example,

ALTERNATIVES There’s an abundance of two-channel VCA compressors out there now, but with most other than the dbx models being based at least loosely on SSL’s stereo bus compressor designs, Neve’s 88C stands out as being a bit different from the crowd.

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Inside, the circuitry comprises mostly surface‑mount devices sitting on a PCB — it’s all manufactured and quality assured to exacting standards in Neve’s factory in Burnley, UK.

whichever threshold or ratio knob is set more assertively will take precedence.

The make‑up gain knob also has a push function, activating a 12dB/octave high-pass filter in the side-chain. With one push for 80Hz, another for 125Hz and another for 300Hz, each indicated by the LED turning a different colour, there’s good scope here to refine the compressor’s sensitivity to energy in the low end and the very low mids. And on the subject of side-chains, pushing a threshold knob activates that channel’s key input. With the channels linked and both key inputs active, you can choose which key input triggers compression by setting the threshold positions, as in normal stereo use. With no jack plugged into an active key input, that channel’s threshold control has no effect.

The specifications are detailed in the manual. There’s little point repeating them here but they read well, with low noise, distortion and crosstalk between channels, and ample headroom.

In Use The first thing to say about the experience of using the 88C is that I enjoyed the look and feel of it ‘in the flesh’ more than I’d imagined I would. It’s a novel form factor and one that I admit I’d been slightly uncertain about before laying hands and eyes on it — but it really does exude a sense of quality and robustness that inspires confidence.

The sound and action of this compressor is also everything Neve say it should be: easy to dial in, and smooth and natural on pretty much any source.

The adaptive attack times make it a joy

to use on vocals and bass guitar, as well as the more complex stereo signals I mentioned above, but it also manages to avoid mangling more ‘delicate’ sounds such as solo piano or fingerpicked acoustic guitar. It’s clean and quiet enough to use on dialogue, and in that context (and working with sung vocals) I really enjoyed feeding the output of one channel into the other, using one as a compressor and the other as a limiter.

Between that, the Fast Attack mode and the side-chain high-pass filter, I felt that I had more than enough scope to control and shape sounds.

Are there down sides? I suppose not everyone likes dual-purpose controls, but the push-turn knobs here are pretty intuitive, and make for efficient use of the front-panel real estate. If you press the wrong thing, an LED or meter colour will make that obvious visually, and it’s a simple matter to return to the previous state. The only other thing to say is that while this is an incredibly versatile compressor, it’s perhaps not the first one I’d turn to for colourful character and pumping effects. You can make it misbehave if you try, but as I see it, that’s the opposite of the 88C’s raison d’être, and there are better tools for that.

The bottom line is that this is an impressive, versatile compressor that should hold wide appeal, and at just a shade over £500 per channel, it’s not disgracefully priced either! £ £1074 including VAT. T AMS-Neve +44 (0)1282 457011 E info@ams-neve.com W www.ams-neve.com

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ON TE ST

SSL Alpha 8 A‑D/D‑A Converter

SSL have chosen not to build mic preamps into their new ADAT expander — but they have included a whole heap of other features.

SAM INGLIS

O

ver the last few years, Solid State Logic’s product designers have displayed two admirable features: a focus on innovation, and joined-up thinking. Recent products of theirs have stood out from the crowd, whilst at the same time fitting neatly into a wider ecosystem. The latter attribute is especially apparent with their UF and UC controllers, and the SSL 360 Plug-in Mixer, but it extends right across their product range, and SSL’s 12 and 18 audio interfaces are no exception. They also have expansion potential, courtesy of their ADAT Lightpipe ports — and now there’s a partner product tailored to exploit this.

The 1U rackmounting Alpha 8 is primarily conceived as an ADAT expander and, as such, can work with any ADAT-equipped audio interface, not just the 12 or 18. But, unusually, it also has a USB-C socket of its own, meaning that it can also be used as an interface in its own right. And, in keeping with SSL’s focus on innovation, it departs from convention in many other ways too.

In the current market, for example, it stands out by virtue of the fact that it doesn’t have any mic preamps. Back when everyone was still working with analogue mixers, this was the norm for ADAT expanders (remember the

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Fostex VC-8?), but it’s more usual today for them to be, in effect, eight-channel preamps. Examples include the Focusrite Clarett+ OctoPre, Audient ASP880 and EVO SP8, Behringer ADA8200, RME Octamic II and so on.

Further differentiating the Alpha 8 from more typical devices is the provision of a headphone output, a dedicated coaxial digital input and output, and the ability to switch its second pair of Toslink connectors to S/PDIF. There’s word clock out as well as in. The analogue I/O is DC-coupled, meaning that it can be used to transmit and receive control voltage signals as well as audio signals. And, to top it off, there’s an unusual degree of flexibility about how all this I/O and routing can be configured.

the Alpha 8 is employed as an audio interface, are useable in tandem with the digital I/O to provide up to 18 ins and outs at base sample rates. When it’s used as an expander, of course, the analogue ins normally feed the digital outs, and vice versa, though as we shall see, there is some flexibility here.

The only socket on the front panel is the headphone amp. This is of the same design as that used in the 12 and 18, meaning that it offers three different gain ranges to suit whatever type of headphones you have available. In some circumstances, it

SSL Alpha 8 £699

Expansive Visions

pros

In terms of industrial design, the Alpha 8 is very much of a piece with the 18 and other recent SSL hardware, meaning that its construction is beyond reproach.

It’s finished in a smart workmanlike grey, and can accept mains power directly on an IEC socket. This is joined on the rear panel by the two pairs of optical connectors, the coaxial S/PDIF and BNC word clock terminals, and the aforementioned USB port, plus two sets of eight balanced jack sockets. These provide eight balanced inputs and the same number of outputs, and when

cons

• Excellent sound quality and audio specifications.

• Alignment adjustable up to +24dBu on a per-channel basis.

• Headphone amp is very useful for troubleshooting.

• Can act as an audio interface in its own right.

• Settings can’t be configured from an attached computer.

summary The Alpha 8 is an unusually versatile line-level ADAT expander that offers superb audio specs and can even double as an 18-in/out audio interface.

could be pressed into service to provide an additional foldback source for a performer, but its primary role is diagnostic, allowing you to ‘sniff’ the signal at any of the Alpha 8’s inputs or outputs.

The rest of the front panel is given over to an unusually comprehensive selection of buttons, through which much of the Alpha 8’s versatility is unlocked.

Buttons labelled 1-8 allow you to select one or more channels from whichever ‘bank’ of I/O is chosen using the Ana, Dig, In and Out buttons. So if, for example, you wanted to select analogue outputs 1 and 2, you’d select Ana and Out, hold down 1 and then select 2 as well.

Selecting I/O in this way has two main purposes. First, it determines what is heard at the headphone port, and second, it allows you to change the alignment level of the analogue paths. For the latter purpose, you can select any combination of the eight channels within the Ana In or Ana Out bank, while the headphone output picks up only the first selected channel or odd/even pair. Thus if you had Ana In 3, 4, 7 and 8 selected, you could adjust the sensitivity of all four of these analogue inputs at once, but the headphone output will present the signal coming in at 3+4 and ignore the other pair.

Lining Up Four separate alignment levels are available for the analogue inputs and outputs, and are cycled using the Level button to the right of the meter section. The options are +9, +18, +20 and +24 dBu. The ability to adjust the alignment in this way might sound like a minor technical detail, but I can’t stress enough how valuable it is in practice. Not only does it allow you to calibrate the entire unit to whatever standard is applicable for your particular region, but it makes it trivially easy to realign individual inputs and outputs to accommodate obscure,

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ON TE ST

SSL ALPHA 8

The Alpha 8’s USB-C port allows it to act as an audio interface.

vintage or semi-pro gear that doesn’t adhere to any recognisable standard.

You need never again be in that position where your drum machine barely tickles the meters, or where you can’t drive an outboard compressor hard enough to do any actual compression.

And if that sounds like a feature you’d expect to be universal on products of this type, a quick glance at the specs of other ‘line level’ ADAT expanders will make clear that it very much isn’t. Arturia’s Audiofuse X8 In and Out offer a default alignment of +24dBu, which can optionally be ‘padded’ down to +12dBu on the input side, while Ferrofish’s Pulse8 AE has a fixed +20dBu aligment, and that of Black Lion’s Revolution EXP is set at +16dBu. You would have to move significantly upmarket, to something like RME’s M-1620 Pro, in order to gain similar per-channel alignment options.

The Level button is part of a group that also includes dedicated buttons for cycling through clock source and, when the Alpha 8 is not externally clocked, sample rate, plus a cogwheel Settings button. This is used to access some of the Alpha 8’s global configuration options. For example, the primary role of the second pair of optical ports is to support S/MUX operation at high sample rates, but if you don’t need them for ADAT Lightpipe, they can be repurposed to carry a stereo S/PDIF signal. And if you want to use the Alpha 8 as an audio interface in its own right, you’ll want to delve into these Settings options to tell it to map the physical I/O onto the USB input and playback channels. The only thing obviously missing from the routing options is the ability to have the analogue outputs mirror the analogue inputs, which might conceivably be useful under some circumstances.

possibilities. If you’re working with an external mixer, line-level I/O may be all you need, and of course its digital I/O could then be used for further expansion. Within a Mac-based system, SSL also envision it being used as part of a Core Audio aggregate device, alongside an 18 or other audio interface.

There’s no reason why this shouldn’t work, although you wouldn’t be able to include any of the Alpha 8’s inputs in a low-latency monitoring scenario.

What did slightly surprise me, given SSL’s laudable emphasis on interconnectedness, is that the Alpha 8 is not integrated into the SSL 360 ecosystem. When used as an expander

rival products don’t even offer similar configuration options, let alone a way to adjust them from an attached computer.

For nearly everyone, then, the Alpha 8 will be a ‘plug and play’ device. And even if you ignore its additional possibilities, it has several advantages over the competition.

Perhaps the biggest of these, to me, is the per-channel variable alignment.

Many other ADAT expanders in this price bracket can’t accommodate +24dBu at all, and none offer the same flexibility to choose other alignments. The onboard headphone amp is also a masterstroke on SSL’s part, making troubleshooting so much simpler. And the Alpha 8 tops the class when it comes to audio specifications, with a dynamic range of 120dB on the inputs and 123dB on the outputs.

Amongst directly comparable products, only the Arturia X8 In and Out come close in terms of audio specs. These also have the advantage that you can add inputs and outputs separately, so you’re not paying for I/O you don’t want; on the down side, metering is almost non-existent, they have word clock in but not out, and each of them requires its own external wall-wart PSU. If you widen the comparison to include mic-level ADAT expanders, which can of course accommodate line-level signals as well as mics, then something like the Focusrite Clarett+ OctoPre offers broadly similar audio performance, but it’s limited to +18dBu on the output side. And none of the other expanders I know of has the Alpha 8’s USB connectivity or headphone output. If you need more I/O, but don’t need or want it to have built-in preamps, the Alpha 8 represents a compelling combination of audio quality, versatility and value for money.

“The ability to adjust the alignment might sound like a minor technical detail, but I can’t stress enough how valuable it is in practice.”

Off The Busses The Alpha 8’s ability to act as an audio interface opens up some interesting

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alongside a 12 or 18, the additional ADAT channels show up as you’d expect, but if you connect the Alpha 8 directly over USB instead, 360 simply reports ‘no device’. I’m not sure if it would be technically feasible to aggregate multiple USB interfaces within 360 rather than Audio MIDI Setup, and in any case the Alpha 8 does not have an internal mixer that could be controlled from SSL 360. However, I would perhaps have expected SSL to have implemented some way of configuring the Alpha 8 from an attached computer.

As it doesn’t have a screen or even a numeric display, visual feedback when choosing settings from the front panel is communicated through the meter LEDs, in a rather limited and cryptic fashion.

A software configuration utility would make it much clearer and simpler, and I hope SSL will add something like this to 360 at some point.

In practice, though, the reason these global configurations are buried in a Settings menu is that you won’t need to change them very often. For nearly all users, I imagine, the default routing will be the most useful; and many

£ £699 including VAT T Headline Audio +44 (0)1908 477142 E sales@headlineaudio.com W www.headlineaudio.com W www.solidstatelogic.com

INTER VIE W

Evgueni & Sacha Galperine Composers

Evgueni (left) and Sacha Galperine are based in Paris.

The fearless sonic experimentation of composers Evgueni and Sacha Galperine is proving equally at home in mainstream and arthouse cinema — not to mention last year’s surprise hit TV series Baby Reindeer.

S O N A L D ’S I LVA

“I

t’s strange, making music for a project that comes from someone’s intimate experience,” says composer Evgueni Galperine, as he reflects on creating the soundtrack for Netflix’s breakout hit Baby Reindeer, which he considers one of his favourite music-making experiences of 2024. The small British mini-series, now a multiple Emmy award-winner, came seemingly out of nowhere and became a hot topic in pop culture discourse, fuelled by the frenzy of fans taking a little too much interest in the show’s drawn-from-reality characters and who their real-life counterparts may or may not have been.

His brother Sacha, with whom he composed the score, adds: “It gives us more responsibility, because we have influence over someone’s intense, real-life experience. It was a very small project initially and we really believed

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in it. We met the wonderful team and [showrunner/lead actor] Richard Gadd who was so excited and vulnerable at the same time — he needed to tell his story.”

When the Galperines were tasked with writing the score, their focus was simply on how best to infuse a disturbing and complicated story with music. Evgueni explains: “Deep wounds, hidden desires, and intense emotions are all great food for music, so it was inspiring, but it was also a challenge because the psychology in the series is so complex. It was our first experience of working with someone who is not only a showrunner, but is also the person who lived this story in real life, so it was very interesting to see how he received the music; he was always comparing his real feelings from the past, and mixing them with his opinion as a showrunner.

This kind of artistic collaboration was very new for us because it was the first time we had done something as personal as Baby

Reindeer. And then there was the rewarding surprise of seeing this niche project become a worldwide phenomenon.”

Marvel Calling The Galperines ended the year with another riveting score, sharing composing duties with Benjamin Wallfisch, on Sony and Marvel’s Kraven The Hunter. Brought in two-and-a-half months before the final mix of the film, the Galperines had their work cut out for them. “Our goal was to try to do something new, sound-wise, but at the same time, because it was a Marvel film, we knew that it was meant for a very large audience. So the challenge was quite exciting: how to keep our style and our tools, while at the same time make it more epic and more direct,” says Evgueni.

The brothers once again went into the project with a vision: “to use instruments like they were the voices of some strange animals”. Evgueni elaborates, “We were looking for instruments, and ways of playing them, which [bring to mind] animal voices and cries, because the hero is very connected to nature and to the animal world.”

Sacha adds: “We used unconventional playing styles for the orchestral instruments.

For instance, there’s an important theme of the father, played on a contrabassoon, with a lot of breath noise. With the cello, we had the harmonics played in a very fierce tremolo, so you don’t really know what you’re listening to, but we used it a lot because it brings a good, heavy sound.”

The electric cello, with plenty of saturation and overdrive, is another sound that features extensively, and for which they called upon musician Ilia Osokin, who, Evgueni tells us, “has a lot of pedals, just like an electric guitar”.

The score also features guitars, virtual synths like FXpansion’s Strobe2, and several Moogs. Evgueni tells us, “We were trying to make the synthesizers sound like organic instruments and the organic instruments sound like synthesizers, to create a hybrid mix.”

The siblings were not worried about introducing such a high level of experimentation to the very mainstream Marvel universe. As Evgueni explains, “We think that even a large audience can totally accept being surprised and not having to listen to the same sounds and the same colours. Most of the time, in Hollywood

films, you have a very common colour — orchestral, with a little bit of synth and percussion — but you can take that from one film and put it in another and nobody will [notice] the change. Coming from European cinema, we feel that by adding epic and direct melodies, we can still keep an unexpected approach and when it’s unexpected, the audience can be moved

and electro-acoustic music at the National Conservatory (CNSM), while the younger Sacha excelled at violin performance at the Conservatory of Versailles before turning to rock and electronic music.

You can hear their eclectic influences in their musical palette: minimalist arrangements, soaring vocal choirs, traditional orchestration, delicate acoustic guitar lines, evolving leitmotifs, gritty distortion, pitch-shifting drops... the Galperines employ an acoustic and electronic toolkit with ease, making sounds from both styles work effortlessly together. It’s all sound, and it’s all fuel for experimentation.

Of their early experiments with sound, Evgueni recalls, “At around 20 years old, I got very interested in electro-acoustic music, which is a kind of electronic music with a little bit more of a classical approach; it’s experimental, but with a lot of dramaturgy and maybe closer to modern classical music than to pop electronic music. This area was created by French composers from the ’60s, like Pierre Schaeffer and Pierre Henry, who were experimenting with the first synthesizers.

I learned how to transform sounds, how to

Evgueni Galperine: “Sometimes it’s interesting when the voice is not very evident but it’s there and you don’t even understand if it’s a voice or an instrument, if it’s a sample or a real singer.” even more. It’s less comfortable, so it surprises you and when you are surprised, you are more vulnerable and more emotionally captive.”

Classical Meets Contemporary The siblings are fluent in the language of music, having spent their early years in Ukraine and Russia, where they began formally studying classical music at a young age. In 1990, they moved to France, where Evgueni studied symphonic composition

Sacha Galperine’s studio is full of interesting musical instruments.

www.soundonsound.com / May 2025

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INTER VIE W

EVGUENI & SACHA GALPERINE

work with synthesizers, and create sounds even with the most primitive synthesizers, like the DX7, or a Roland Jupiter.”

Sacha adds: “I started a little bit later than him, because I’m a little younger, and right away began with the computer and plug-ins. I fell in love with all the possibilities of the electronic effects on the computer and realised that you can really transform a sound. You can just tap your phone, make a little percussive sound, and then, with well-chosen effects, transform it into anything: a drone, a lead instrument, a synth sound, a low sound...”

Among Sacha’s favourites are FabFilter’s Timeless 3 for vintage-sounding tape delay; distortion and filter effect Driver by Native Instruments; XLN Audio’s RC20-Retro Color to add a warm vintage recording vibe; sound reverser Flipper by Soundhack; Decapitator (saturator), Devil Loc (limiter), Crystallizer (Eventide H3000-inspired pitch‑shifting and reversed echo), EchoBoy (delay and echo) and Little AlterBoy (vocal transformation) from Soundtoys; room simulation and reverb effects like Sound City Studios and Lexicon 224 from UAD; AIR Spring Reverb; the Tony Visconti-inspired Tverb from Eventide Audio; Texture by Audiothing and Shuffling by GRM Tools for granular effects; Smooth Operator by Baby Audio for resonance suppression; and the Leslie cabinet rotary simulation MVintageRotary from Melda Production.

Transformations The brothers have used their share of unconventional sources over the years.

For Palme d’Or winner Audrey Diwan’s Emmanuelle, the English-language remake of the famous French film of the same name, “Some of the percussion was created using

The Sherman Filterbank is one of the brothers’ most valuable sound-shaping tools.

just the very quiet sound of the tongue,” recalls Evgueni. “Nobody really understands that it’s a tongue, but it creates a kind of weird feeling when you listen to it; somehow, even unconsciously, you understand that it comes from the human body, so it’s great to create a disturbing effect with something very quiet and very soft.”

Sacha disappears from view and returns with a mallet and a plastic bag. “Just tap on a plastic bag, transform it a little, and you have a great percussion sound.”

‘Martha Suite’, from Baby Reindeer, has a terrifying motif at its centre, employing something that sounds like a woman’s voice but isn’t. Sacha tells us, “It’s actually my whistle, run through effects to make it sound lower and vibrate like a human voice. It’s put through an Octaver, a Tremolator, and then a spring reverb, so in the end, it’s an unreal voice, kind of like a siren.”

On the piece ‘Donny And Teri’, the vocal lurks menacingly in the background, rather

than being the principal element. Evgueni explains: “Sometimes it’s interesting when the voice is not very evident but it’s there and you don’t even understand if it’s a voice or an instrument, if it’s a sample or a real singer. For Baby Reindeer it was great because the character is so confused about what is happening around him, so not understanding even what the colours are, or what the instruments are, is part of the game.”

For Andrey Zvyagintsev’s Cannes Film Festival Jury Prize-winning drama Loveless, the Galperines composed the score without reading the script, as per the director’s wishes. With just the storyline to go by, they transformed their own emotions and reactions, free of on-screen images, into score. The piece ‘11 Cycles Of E’ builds on the sound of a prepared piano and the energy of one single note to convey the desperation of parents searching for their missing son. ‘Trouble’ makes use of the

Sacha’s small but mighty equipment rack is home to (from top) an Apogee Rosetta 200 converter, Chandler Germanium Preamp and Summit Audio TPA-200B valve preamp.

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INTER VIE W

EVGUENI & SACHA GALPERINE

The musical saw used in the brothers’ soundtrack for Radioactive.

sound of a squeaking bicycle wheel to add a layer of discomfort.

Sources Of Inspiration Ideas for compositions can come either from musical motifs or interesting sounds, says Evgueni: “Sometimes when you want to just find the melody, the piano is the best instrument you can dream of because you have harmony and melody and it’s neutral; so while looking for a piece, you can imagine all kinds of different instruments playing different parts because of the neutrality of the piano. A sound we find can also help us to come up with music or a melody, because sometimes the music can be inspired by the sound, or by the instrument.”

“The most important thing,” says Sacha, “is to really understand, first, what the story is about — what message does the creator want to express? — and find the musical tools that make sense very specifically in that direction. It’s like when we were talking about the whistle that sounds like

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a siren in ‘Martha Suite’. It made sense because it was telling the story and was like a metaphor for how she tries to lure the main character into her world.”

Evgueni adds, “Martha is a kind of siren, but [the lead character] is also always listening to the sirens from his past, because his past doesn’t let him go, so it’s also this voice from the past, dragging him back. Sacha’s right, it’s not only about the style, but also about what the music needs to express.

“For Kraven, it was very important to make the mother of the main character exist in the film because his mother — without spoiling the story — is the origin of the motivation of his acts. We don’t really see her in the film, so we wanted to use a lullaby she would sing to him, and it not only appears in the score when he’s thinking about his mother, but also at a very important moment in the film. Sometimes it is very clean, and sometimes it’s mixed with a lot of different instruments.”

Visible amongst the more conventional instruments and equipment at Sacha’s Paris studio is a musical saw, which the brothers used in the score for the 2019 Marie Curie biopic Radioactive. “We used Theremins and Ondes Martenots as well. The film was about a woman who was ahead of her time, so we wanted to use early electronic instruments, instruments that were very progressive and new in the beginning of the 20th Century, that she could maybe even have listened to while she was alive.”

Sacha adds, “It was about the idea: the first instruments of the modern era, even if it wasn’t exactly at the same time.”

Evgueni continues: “The people who created these instruments were also ahead of their time. Two or three crazy guys invented these instruments in the 1920s, 20 or 30 years before the first synthesizer, and what she did as a scientist was... the idea is the same: they were all pioneers. So you find parallels with the story, with the period in which the story is happening; you try to find as many connections as you can and the music becomes part of the whole.”

Resource Management Some of the brothers’ favourite virtual instruments and sample libraries include Symphonic Organ and Cinematic Soft Piano by Spitfire Audio; Piano Colors by Native Instruments; Baby Audio’s physical modelling synth plug-in Atoms; orchestral sample library Landforms by Slate+Ash; GForce’s Mellotron emulation M-Tron Pro IV; Blisko, Nisko, and Helenko by Felt

Instruments; and Salu, Amber, Arbos, and Duplex Saxophones from Orchestral Tools.

Indeed, today’s composers have access to a never-ending supply of libraries and effects, which can lead to paralysis and indecision. Evgueni agrees emphatically: “It’s endless; you can spend days and weeks without any result. The only thing to do is to experiment when you’re not working, when you’re not composing music.

Both me and Sacha sometimes spend days without trying to make music — just listening to sounds, trying them out, and making notes. Spend time getting to know your instrument. You will create connections that will help you to not try every possibility, but only those that you remember and that you connect to what you are looking for in that very moment.”

Sacha adds: “Also, like with any creation, you just have to stop when you decide that it is good enough for you. It’s never totally perfect; it’s never totally achieved. At some point, you have to decide, ‘OK, this works for me. I don’t think I will get any further.’”

Evgueni underscores why this is a good attitude to adopt: “Accept finding something close to what you are looking for, and then be focused on the music and not on the sound, because the most important thing is what you have to say with the music. Once that becomes most important, you accept more easily the fact that maybe you didn’t find exactly the sound you wanted, perhaps because of the lack of time, but also because you want to create music — you don’t want to spend half of your life looking for sounds.”

Fitting Together Although the Galperine brothers work on shared projects, each has their own studio where individual pieces are crafted.

Evgueni’s DAW of choice is Logic Pro, while Sacha prefers Pro Tools. Does this cause problems? Thankfully not, Sacha explains: “We will either compose together in the same room, or compose separately. We don’t often do some instruments here and some instruments in Evgueni’s studio.”

Evgueni elaborates: “Even when we’re working together, we’re working with one station. If we work in my studio, I will be the one who is taking commands and if we’re working in Sacha’s studio, he will be the one taking care of the computer controls. Each time, we discuss music ideas and try things, but depending on which studio it is, one of us becomes a computer operator as well.”

If they do feel the need to transfer elements of a composition to each other,

they can, because the process has become significantly less complicated, says Evgueni. “With audio files, it was always super simple, but now you can also transfer MIDI files from Logic to Pro Tools easily. Earlier, it was totally separate and difficult to transfer something from one to the other so we preferred to just not do it: if you started in Logic, you continued in Logic, and if you wanted to use Pro Tools, you transferred it as an audio file, because otherwise it was complicated.

“Now, the same virtual instruments are available for both software, so when you choose to work with one, it’s mostly a question of doing so because you’re used to it. Of course, each software has its own things that make it work for you, so you choose one or the other, but they’re quite similar. If you are used to one, it’s difficult to change because you’ve worked with it for years; it’s kind of an old friend and it’s not very easy to make new friends.”

Mixing It Up Once the pieces are ready, the brothers move from their home studios to an external studio and focus their attention on the mixing process, in which they are very involved. Evgueni tells us: “We don’t stay in our home studios, because we want to listen with different speakers, in different rooms, rediscover the music a bit, and also give the possibility to our sound engineer to add something. It’s a little bit less personal, in the sense that he didn’t make this music, so he has a more distant and more objective opinion.”

They both emphasise that you can’t mix your way out of a bad arrangement: “It is the arrangement that defines the mix and the role of each instrument,” insists Evgueni. “If you make the decision to place an element lower or louder, it changes the music.”

Sacha adds: “I think the role of the mix is mostly to make the musical ideas and the ideas of the arrangement clear. So the ideas come first, but once we get in the studio with the sound engineer, his job is to make sure that all these ideas are clearly adjusted and that the audience will hear them as they were intended.”

Looking Forward As the duo look to add more projects to their incredibly diverse filmography in 2025, both brothers agree that a common thread runs through their work: collaborating with people who have their own vision. Says Evgueni: “Through this work, we love to discover new worlds, new stories and new ways of storytelling; otherwise it’s boring. Even if we are talking about commercial projects, there is always room to do something fresh and unexpected, because we love art cinema and we love wide‑audience films. It’s not about the style, or if it’s niche or not niche. Take, for example, Mad Max: Fury Road, which we didn’t work on, but it’s a good example of a super‑commercial, wide‑audience film that is, at the same time, an arthouse movie. It doesn’t happen that often, but it does happen, so I think that the common thread is the particularity of the vision of the world of each director, which can also help us to explore ours.

“With each film, it’s great to feel the challenge and to again be able to find a particular key which will open only this door and not others. It’s about the connection between music and cinema and how we can approach it, and the more you work with inspiring people, the more you get inspired as a composer as well, because we work for the film.”

Sacha concludes: “Each new project is also the occasion to be inspired in a new way and create something you didn’t even want to create before because you didn’t think of it. Now the occasion is here, and that gives way to new ideas; it’s like a new adventure each time. We discover a new film, but we also discover a new facet of ourselves.”

Talking Cinema What surprised Evgueni and Sacha Galperine most when they first started working as composers of music for picture was how difficult it can be for people to talk about music. Sacha explains: “It’s not about terminology.

Sometimes music seems very abstract to people who are not musicians, and they can’t really speak about it because it’s just a feeling in the air, but for us it’s very concrete because we analyse it and are used to it.”

Evgueni continues: “We can help with the communication, that’s also our role, but it’s really what Sacha was saying — it’s about the capacity to just understand what the music is doing here. The process is complicated, mostly for young directors on their first features, and it mostly comes from a lack of experience. Some directors have a hard time accepting music in their films, and there’s this kind of paradox of having the strong feeling that they need the music but at the same time when the music arrives, it’s like ‘Oh! It’s taking my movie from me.’ Because even if the music is a part of the larger whole, when it says something and has some kind of energy in it, it’s still another author, another creator who puts his creation in your creation, and sometimes I think it’s difficult to accept.

“When you start, you often know only music — you don’t really know cinema. We sometimes see [a limitation in young composers that comes from] just not having watched enough films. I remember I was on

a jury once, and when we asked the young composers what films they had liked recently, they mostly said Interstellar or The Dark Knight, and things like that. Of course, everybody watches those, but when asked about something a little bit more personal, they were like, ‘I don’t know’.

If you only like The Dark Knight, maybe it’s better to go to Hollywood and try to work with Christopher Nolan, but if you want to work in cinema more generally, it’s so important to know the world you are stepping into.

“So the first advice we give to young composers is to watch a lot of films: from different periods, different countries and genres, and try to understand why the music is there and what it is saying. It’s very interesting to analyse things. For example, when you watch Martin Scorsese films, especially the early ones from the ’70s, like Taxi Driver, the music [by Bernard Herrmann] is always there to say something.

It’s not just an accident that you have this jazz melody with a saxophone.

“It’s always interesting to watch a lot of films to understand the different languages of cinema and music, in order to become capable of finding the right key to the film when they ask you for music, and also to communicate with directors, because, on their side, they are learning film and, sometimes, not learning music so much. So they have the opposite problem, and it’s your role as a composer to find the right examples, references, and words, and to make connections.”

www.soundonsound.com / May 2025

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Gamechanger Audio Auto Series Delay & Chorus Effects Pedals

They might come in guitar pedal format, but the Auto Series effects have much in common with modular synths...

TED MORCALDI

G

amechanger Audio’s Auto Series pedals are a sort of hybrid between traditional effects pedals and synth modules: yes you can still plug a guitar in and stomp on them, but not only can they modulate parameters automatically based on the dynamics and pitch of your playing; they also boast

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an advanced tracking engine and patchbay (a generous array of patch cables is provided) that makes possible control‑voltage automation of any parameter, whether based on the level or pitch of the main input signal or of a secondary one. Such features definitely mark these pedals out as being unique, but as the algorithms available on each one are of really high quality, the sounds can be equally remarkable.

There are currently three pedals in this range: a delay, a reverb and a chorus. For this review, I was sent the delay and chorus, so it’s their features and sounds that I’ll explore below. But the reverb shares the same design concept and form factor, so much of what I write will apply to it too.

Overview These pedals use the same enclosure, with a patchbay protected in a recessed central section, and the main parameter controls on either side. They use 24-bit/44.1kHz conversion and 32-bit processing, and each offers a choice of three algorithms. For the Auto Delay these are Tape, Dig and Analog, and the Auto Chorus offers you Chorus, Ensemble or Flanger. (I’ll explore those below.)

On the rear panel are two power inlets, one accepting a standard 9V 250mA centre-negative supply and the other USB-C, and there are various analogue I/O either side, with two slide‑switches at

the top. One switch sets the input level as EU (+10dB), Line (+5dB) or Inst (+0dB), while the other changes the mode of the Level knob (of which more later).

Two pairs of quarter-inch jack sockets provide the main inputs and outputs, and the lower one of each pair can carry either the left channel of a stereo pair or a mono signal, while the other can carry the right channel or act as a single stereo TRS connection.

There’s also a mini‑jack Track In input, and this can be used in a couple of ways. Gamechanger say that the tracking engine will be most effective when it ‘sees’ the cleanest possible signal so, should you want to use other effects before the main input, you can use the provided splitter box to feed a clean signal directly into this Track In input. Alternatively, in the spirit of experimentation, you could plug an external source into this input, to act as an automation source. A corresponding Track Out mini‑jack carries a copy of the Track In signal or, if nothing’s connected to the Track In, a copy of the main input.

There’s also a clock/MIDI in, again on a mini‑jack, that can accept an analogue clock signal or act as a MIDI input.

On the top is the main control panel, and top left you’ll find the Level knob, and its function is set by the Knob Mode switch I mentioned above. This can be set to In, Wet or Mix, and when In is selected the knob sets the amount of signal that’s sent into the effect engine — the higher the level of this signal, the more extremely the signal will be altered by the effect. In the Wet position, the knob acts in much the same way, except that the dry signal is muted. In the Mix position, it becomes a wet/dry blend control.

Three more of these larger knobs adjust parameters specific to each pedal.

On the chorus, these are Rate, Depth and Feedback, and, with Rate ranging from 0 to 819.16 Hz, you’re able to access anything from very subtle chorusing to abstract ring-modulation. For the delay, there are Time, Repeat and Tone. The time ranges from 40ms to 2 seconds, which is just enough for some short phrase-looping possibilities, and the Tone knob really helps you to get the delay working nicely alongside the dry signal.

I found that this was a great target for my automation experiments too.

Each of these is accompanied by a second, smaller knob that changes how the parameter responds in Automation

Mode, and a three-way Automation Mode switch in the recessed central panel decides their function: the options are Sum, External and Alt. At the top of this panel you’ll find the algorithm selector and, next to it, a switch that’s unique to each pedal. On the delay this is a stereo spread switch called Ping Pong, with options to half or fully pan the delays, or switch this off (it only has an effect on the delay if used in stereo, of course). The same switch on the chorus sets the Mod Shape, and this offers saw, sine and ramp settings.

Algorithms As you’d expect, the Digital delay provides uncoloured repeats of the input signal. This might not sound hugely interesting, but it’s useful and can easily become so when you use the automation features. According to Gamechanger, the Tape delay algorithm emulates the saturation and compression of a vintage tape echo but without imperfections such as wow and flutter, the idea being that you’d use the pedal’s automation features for such effects — automating the delay time, for example, can deliver these much-loved flaws. Because of the way the delay character evolves with

Gamechanger Audio Auto Series £298 Each pros • Great-sounding algorithms.

• Wide range of parameter adjustments.

• Creative, forward-thinking design.

• Great for experimentation.

• Patch cables included.

• Integrates well with modular synth systems.

cons • A significant learning curve before you have precise control!

summary These unique pedals should appeal to experimental guitarists and those who like to get hands-on with modular synths.

long repeats, this algorithm sounds to me like an emulation of a Space Echo. The Analog delay is modelled after a high-end studio bucket-brigade delay unit, and has a rich, warm tone that gets darker with every repeat. Sounding much like a Boss DM-2 to my ears, it’s very nice, and the character really stands out with high feedback levels — the delays smear into that lovely, bubbly ‘analogue delay cloud’ that’s unique to such units.

Alongside the main audio I/O, on quarter-inch jacks, are mini‑jack I/O for the tracking signal — these can optionally be used to feed a clean tracking signal to the unit, to track an external source, or to pass the tracking feed along to other pedals or synths.

www.soundonsound.com / May 2025

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GAMECHANGER AUDIO AUTO SERIES

The chorus’ Chorus algorithm sounds very rich, and before the Depth knob reaches the halfway point it reminds me somewhat of my much-loved Boss CE-2. Turn it further, and you’ll hear more extreme chorusing, and this algorithm also makes it possible to dial in a vibrato, with the Level Knob Mode set to Mix and the Level knob to maximum. The Ensemble algorithm is an enhanced, hi-fi chorusing that, thanks to the blending of the three offset, pitch-shifted voices, is more subtle than Chorus. And the Flange algorithm delivers the classic ‘jet flange’ sound. Unfortunately, I wasn’t able to set its delay long enough to access some of those steel-drum type effects you can create with certain units, but it’s certainly a classic, colourful-sounding flanger.

Patches Explained

Parameter Modulation With the Auto switch set to off, these are straightforward, albeit great-sounding effects, but the Auto section and patchbay mark this series out as being different from anything else available today. The tracking engine follows both the pitch and the level of the input (or Track In) signal, and the user’s first step that will determine how the automation reacts to the input is to store a ‘centre pitch’ or ‘dynamic level’. The Dynamic calibration is done by holding the Auto switch and strumming your guitar (if that’s your input source), to set the dynamic threshold. The tracking engine converts any signals that exceed this threshold into control voltages (CV), and these are used to automate the parameters.

Quite how they do so depends on the patchbay, which consists of four outputs (two each for the pitch and dynamic CV) and four inputs (one for each of the pedal’s parameters). You could, if you wanted, route the CV outputs to external synths, but normally you’d use them to automate the pedal’s own parameters.

As mentioned above, the small knobs below the main parameter knobs are automation controls, and these react differently depending on the chosen Automation Mode. In Sum mode, you can think of the small knobs as being a parameter offset: use the main knob and the automation knob to set the range of the CV automation for that parameter. In Ext mode, an external CV source can be altered by the pedal’s pitch and dynamic trackers: each auto knob can create a ducking or gating

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I thought it might be helpful to run through a couple of the manual’s example patches, to help explain both the pitch and the dynamic trackers. I’ll start with the first, simplest patch example, called Dynamic Send, which can be very useful for delay: loud notes activate the delay, while quieter notes don’t ‘track into it’.

This is achieved by calibrating the Dynamic level with a strum, while holding down the Auto switch. The loud notes activate the delay because the mix mode switch is set to Input (this is different from other patches, in which it’s set to Mix), and when set to Input the Level knobs allow signal above the Dyn Threshold into the delay line.

effect, in response to the pitch or dynamics of any CV source. Alt mode is, to simplify, the inverse of Sum mode.

When the Auto switch is on, the small knob determines the starting point of the parameter and, depending on the amount of CV a patched parameter receives, it will jump towards the larger knob’s setting.

There are then three modes, Fast, Rise, and Gate, that influence how the CV can alter the parameters even further, and three more for the pitch‑tracking engine. In the simplest of terms, Note mode automates parameters based on how far above or below the input signal is from the centre pitch. Tone is the same idea but, instead of pitch, it’s detecting the input signal’s frequency content — think bright sounds versus dark sounds

Now let’s look at the Dry/Wet Tone Follow patch, with the Automation set to Sum and the Level Knob Mode to Mix. We’re also in the Pitch Tone mode, so the CV comes from the input’s frequency spectrum. With a calibrated note/ tone set on the pitch side, tones brighter than the calibrated one increase the CV level, and the Pitch knob sets how much the difference in tone will change the delayed sound. When playing, darker notes lower the Mix setting automatically, essentially turning the delay mix off, whereas brighter notes turn it back up. The notes are sent into the delay differently than in Patch 1 — all notes are sent into the delay, but their frequency (darker/brighter) affects the Mix level.

controlling the automation. Lastly, V/Oct changes the CV strength in each octave, with the centre pitch being 0V/oct.

Lasting Impressions So there’s lots of control, but how are these pedals in use? Coming from a background of playing improvised, experimental music with electronics, I’m always searching for new sounds to create and generally pleased when I stumble upon happy accidents and surprises. Not surprisingly, then, my time spent investigating these pedals was filled with excitement and spontaneity, and I created some wonderful, unique patches. Gamechanger Audio clearly set out to develop a unique range of effects that, dare I say, might ‘change the game’, and they absolutely have done that.

With the delay pedal, I found the range of sounds overall to be somewhat underwhelming and difficult to control with precision, even when they were meant to be automated by dynamics or pitch.

Even dialling in the patch ideas from the manual, I never really found anything that jumped out at me as an exciting sound — not quite the reward I’d hoped for after all that patching. The ducking delay was very nice, though; automating the tone knob was a pleasant discovery; and, digging deeper, I found some very useful sounds when patching in my trusty Korg SQ-1 sequencer. This created some rhythmic movement that I could imagine being popular with the modular synth crowd. Using the SQ-1’s CV out to automate the Tone knob was my favourite trick. I felt that a trigger input for the Time knob could have been really useful for a pedal that lives so much in the modular world, but the MIDI and clock inputs will, of course, be useful for those sync’ing the pedal with other devices.

The chorus really is unique, and I found it much more musically reactive than the delay. The ranges of the parameters allow for such extreme changes in the modulation speed and feedback when they’re automated, and this allows for some truly wild changes in sound. For example, the high notes can

automation section allows you to dive into uncharted territory. Those looking for a good-sounding but basic delay or chorus could well feel overwhelmed — there are modes upon modes here, and given the subtlety of their interaction with the various parameters, there’s plenty of scope for user error to breed confusion and deter creativity.

It may not be the most practical pedal to have on a pedalboard, either, especially if you want easy access to patching and tweaking parameters. But this format should have lots of appeal both for the sort of experimental ‘electronic guitarist’ who may keep some pedals within reach up on a stand while playing, and table-top musicians, especially those with a modular synth setup and who enjoy a hands-on approach.

“Gamechanger Audio clearly set out to develop a unique range of effects that, dare I say, might ‘change the game’, and they absolutely have done that.” have a rich vibrato while the low notes come out as unpitched, ring-modulated sounds. So exploring its sonic possibilities sparked much more creativity in the moment than the delay pedal did for me.

On the whole, then, these Auto pedals form an exciting range that’s geared very much towards players with experimentation at their core. The effects sound excellent on their own, but the

£ £297.86 each including VAT and shipping.

T Gamechanger Audio +1 202 657 4587 E info@gamechangeraudio.com W https://gamechangeraudio.com/auto

www.soundonsound.com / May 2025

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RME ADI-2/4 Pro SE A-D/D-A Converter

Somehow RME have managed to make their world‑class converter even better!

HUGH ROBJOHNS

R

ME’s extensive range of audio interfaces sets a reference standard for audio professionals and keen amateurs alike providing, amongst other things, immaculate sound quality, pragmatic feature sets, and excellent low-latency drivers with legendary longevity.

In recent years, there’s been a growing interest in RME’s interfaces from the hi-fi fraternity too, and RME have responded by optimising some of their smaller products specifically for that market — adding things like handheld IR remote controls, RIAA correction, trigger outputs (for powering external devices on/off), ‘balanced’ headphone connections, and more.

However, don’t let this hi-fi lustre mislead — the mildly refocused functionality doesn’t preclude using these devices within a studio environment at all, and it could be argued that

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some of these features bring valuable benefits not otherwise available on the company’s more conventional products.

The latest model in this ‘enhanced’ range is the ADI-2/4 Pro SE (Special Edition); essentially a development of the ADI-2 Pro I reviewed back in April 2017 (www.soundonsound.com/reviews/ rme-adi-2-pro), which was itself revised as the ADI-2 Pro FS R. Needless to say, the ADI-2/4 Pro SE boasts further improved technical specs and a faster, more powerful DSP than its forebear, as well as extended connectivity and a few other new features.

Overview Like its predecessor, the ADI-2/4 Pro SE can serve many functions. It is a mastering‑quality stereo A‑D/D‑A converter with impeccable specifications, supporting all sample rates up to 768kHz as well as DSD, but it can also serve as the preamp in a hi-fi system. It includes two independent, high-powered

stereo headphone amps (or one super-high-powered stereo ‘balanced’ headphone amp). It features a very versatile USB 2.0 interface configurable for 2x4 or 6x8 connectivity, supporting DSD record and playback formats as well as providing analogue, S/PDIF, AES3 and (stereo) ADAT interfacing. It is fully class-compliant with compatible platforms, and employs RME’s well‑established MADIface drivers on PC platforms for incredibly low latency and rock‑solid stability. Naturally, RME’s SteadyClock technology minimises jitter and its related harms, too.

Comprehensive internal DSP functionality includes stereo sample rate conversion, digital emphasis correction, Mid-Sides processing, mono summing and polarity inversion, headphone crossfeed, adjustable A‑D/D‑A converter filter parameters, loudness contouring, rumble filtering and parametric EQ, spectrum analyser metering — and much more besides. Uniquely for the ADI-2/4 Pro SE it even includes the ability to connect directly with a (moving magnet) turntable cartridge, to perform digital RIAA correction. And it comes with a neat infra-red remote control to access a lot of the unit’s functionality directly, including via several user-programmable buttons.

Physically, the ADI-2/4 Pro SE is an attractive, half-rack‑width unit, with most of the I/O on the rear panel, and

configuration and control functions accessed via menus on the small but crisp front-panel colour display. The unit’s operation and configuration paradigms are the same as other ADI-2 series units, and as described in my review of the ADI-2 Pro, so I’ll not labour that again here. Personally, I don’t find the menu‑diving entirely intuitive on a first outing, but it is logical enough once you’re familiar with it.

The Ins & Outs A supplied 40-Watt ‘line-lump’ power unit has an IEC mains inlet (100-240 V AC) and delivers 12V DC via a secure bayonet-locking coaxial plug (as used on several other RMS products). The negative side of the DC plug is tied to the IEC protective earth via a 1kΩ impedance — so although not directly grounded, neither is it entirely floating, which should avoid both ground-loop issues and static shocks!

A B-type USB 2.0 port allows connection to a computer (a USB cable is included) via a class-compliant (UAC2) mode, although ASIO/WDM drivers are also available for Windows platforms. The unit can be configured either as a simple two‑in/four‑out stereo interface, or in a multi‑channel configuration catering for six inputs and eight outputs (see below).

Most of the physical I/O is the same as the ADI-2 Pro, with stereo digital connections made via a 9-pin D-type socket. A breakout cable is included providing (coaxial) S/PDIF and AES3 in and out. A pair of TOSLink optical ports duplicate the stereo S/PDIF output and allow an optical S/PDIF input to be selected instead of the coaxial connection, if preferred.

Since unsynchronised digital sources are common in hi-fi applications, a stereo sample rate conversion (SRC) facility

is also included and can be applied to either the S/PDIF or AES3 input.

Unusually, but potentially very usefully, this optical input port can also be configured to accept a two-channel input from an ADAT source (only channels 1+2) — and built-in SMUX protocols support sample rates up to 192kHz. Two analogue line inputs connect via a pair of combi XLRs and may be configured for RME’s usual selection of reference levels (0dBFS at +24, +19, +13, +7, or +1 dBu). A digital gain trim menu spans a 6dB range in 0.5dB increments to allow accurate gain matching where necessary.

Uniquely among USB interfaces, these same analogue inputs can also be configured to accept the (unbalanced) signals from a moving-magnet pickup cartridge, presenting a fixed loading of 45kΩ and 150pF (close enough to the IEC 61938 specification of 47kΩ in parallel with 220pF to be perfectly workable). Fine tuning of the HF response ( compensating for the loading capacitance) can be achieved via a five-band parametric equaliser, and input level (compensating for the input resistance and cartridge sensitivity) is controlled by adjusting the input gain (14 to 38 dB). This range corresponds to cartridge sensitivities ranging from 1.25 to 20 mV. This input gain adjustment feature also allows the headroom margin to be managed and, while the published specs assume a 15dB headroom margin, the superbly clear and informative user manual states that 6dB is perfectly sufficient in practice.

RIAA correction is performed digitally, of course, within the unit’s DSP, to give extremely high accuracy of both the corrected frequency response and left/ right channel matching — far beyond

RME ADI-2/4 Pro SE £2174 pros • Top‑level technical performance.

• Versatile configurability with RIAA preamp capability.

• Stereo and multi‑channel USB interfacing modes.

• Impressively capable DSP processing options.

• Balanced headphone mode with Pentaconn socket.

• Trigger out control facility and IR remote control.

• RME’s best‑performing interface to date.

• You get an awful lot of performance and capability for the money!

cons • None that I can think of.

summary An impressively powerful and capable two-channel A‑D/D‑A converter/preamp with dual high‑powered headphone amplifiers, leading-edge technical performance, reliable low-latency USB interfacing, and versatile DSP facilities.

that of most analogue RIAA preamps.

An elliptical filter (mono‑summing the very deep bass) is available to reduce rumble and low‑frequency surface noise, and the parametric equaliser mentioned earlier allows further tweaking of the bass response for personal preferences, if required.

On the output side of things, the ADI-2/4 Pro SE has gained some useful additional features compared with the ADI‑2 Pro, the first being four independent (DC-coupled) analogue outputs. Like the older model the rear panel carries both XLR and TRS connectors, but instead of sharing only channels 1+2, the symmetrically-balanced XLRs output channels 1+2 while impedance-balanced TRS sockets output channels 3+4. And, as you’d expect, the

The front panel is home to a pair of quarter-inch headphone outputs and a 4.4mm Pentaconn socket, something not often spotted in the pages of SOS.

www.soundonsound.com / May 2025

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RME ADI-2/4 PRO SE

ALTERNATIVES RME’s ADI-2 series products set a high benchmark for two-channel A-D/D-A converters that few can match, let alone exceed. The most obvious competition now — as it was back in 2017 — is the Lynx Hi-lo 2 USB, which has broadly similar high-end performance and functionality. I know of nothing else that offers direct MM pickup interfacing and digital RIAA correction in such an easy-to-use product, though.

Round the back things are a bit busier, connection wise, with a 3.5mm trig output, USB B port, TOSLink connectors and a D-type port for digital I/O, XLR and quarter-inch audio outputs and combi XLR audio inputs.

output reference levels are configurable over the same range as the inputs.

One other handy addition for the hi-fi set is the inclusion of a 3.5mm socket which delivers a 12V trigger output signal.

This would typically be connected to a suitable power amplifier, for example, switching it on/off automatically whenever the ADI-2/4 Pro SE is powered up/down.

Moving around to the front panel, the main display and controls follow the same format as previous ADI-2 models, with the only obvious addition being the inclusion of a 4.4mm Pentaconn headphone socket located between two standard quarter-inch headphone sockets. As with previous models the two quarter-inch headphone sockets deliver outputs 1+2 and 3+4, controlled independently and powered via RME’s ‘Extreme Power’ headphone amps.

Alternatively, a ‘balanced’ headphone mode can be instigated whereby each stereo headphone amp is reconfigured into a bridged arrangement to drive just one channel of a ‘balanced headphone’ (via a suitable adaptor cable). In this mode (and slightly confusingly) the right-hand socket drives the left channel, and vice versa... but perhaps more conveniently, the ‘balanced’ headphone signals can be accessed directly via the 4.4mm Pentaconn socket.

RME’s ‘Extreme Power’ headphone amps are well named as they have the potential to deliver up to +21.5dBu — the equivalent of over 2.6 Watts into 32Ω headphones! Very sensibly, configuration menus allow the output amplifiers to be tamed by one of three operating modes: high power mode

(+19dBu); low power mode (+7dBu), or an IEM mode (+1dBu), reflecting typical headphone/in-ear sensitivities.

Naturally, since the balanced mode employs a bridged amplifier arrangement, the potential output power is even higher, reaching a whopping +25dBu in high-power mode, +13dBu in low power, and +7dBu in IEM mode.

However, RME claim to have optimised this configuration to enhance fidelity rather than raw power. Phew!

Testing Testing Technically, the ADI-2/4 Pro SE’s performance improves upon that of the old ADI-2 Pro by a small but worthwhile margin, especially on the D-A side. I use the AES17 test standard to evaluate converter performance

D-A performance also improves on the ADI-2 Pro by 2.4dB, placing it fifth in my personally-measured league table. (My current record holder is the Lynx Hilo 2 at 128.1dB A-weighted).

It’s interesting to note that four of my top-10 measured D-A converters, and three of the top-10 A-D converters, are RME products! The ADI-2/4 Pro SE is currently RME’s best-performing converter in my personally measured league table, sitting very comfortably on the top-table of true mastering-grade converters.

Impressions I described the original ADI-2 Pro as “a beast of a converter” — which I stick by, making the ADI-2/4 Pro SE something of a ‘megabeast’ given its improved technical performance and enhanced features and facilities!

The 114-page manual is crammed full of detailed information explaining clearly and precisely all the impressive things the ADI-2/4 Pro SE can do and how it can be used in a wide variety of both professional and consumer applications.

While the retail price has risen significantly over the last decade, this is still a very impressive and capable device for the money, and it is quite astonishing how large a feature set has been squeezed into such a compact package, and just how well it does what it does!

I simply can’t fault the ADI-2/4 Pro SE in any way: it’s a superb piece of kit.

“I described the original ADI-2 Pro as ‘a beast of a converter’ — which I stick by, making the ADI-2/4 pro SE something of a ‘megabeast’ given its improved technical performance and enhanced features and facilities!”

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as I’ve found it maps reliably to my subjective impressions — and a high score requires very careful attention to detail across all aspects of a converter’s design and construction.

My AES17 bench tests delivered figures of 124.3dB (A-weighted) for the A-D performance and 123.4dB (A-weighted) for the D-A. (Both were measured with the analogue interface set to a +24dBu reference level for 0dBFS).

To put these figures in context, the A-D performance is the best I’ve ever measured, outperforming the previous leader (RME’s ADI-2 Pro) by 0.3dB. The

£ £2174 including VAT. W www.synthax.co.uk W www.rme-audio.de

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

Vanguard V24 Stereo Valve Microphone

Vanguard’s new stereo mic is an exclusive tribute to an even more exclusive original.

SAM INGLIS

T

here are manufacturers who recreate vintage gear to an ever more obsessive degree of accuracy, and there are manufacturers who see classic mics and studio hardware as inspirations rather than templates. Derek Bargaehr of Vanguard Audio Labs is in the latter camp. Most of his designs are obviously informed by historic models from the likes of Neumann and AKG, but they’re not intended as clones or copies. Rather, Derek’s goal is to make the best mic he can within the same broad set of design parameters, whilst keeping the end result affordable.

At first glance, though, the new V24 stereo microphone might seem like a departure from this philosophy. For one thing, it’s recognisably based on a particular

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vintage model. And for another, it would be hard to describe it as ‘affordable’...

Twice Twelve?

Introduced in the late 1950s, the AKG C24 remained in production for nearly two decades, though it’s thought that fewer than 800 were ever made. It is often described as a stereo version of the C12, and used the same capsule, valve and transformers as that mic, but with an alternative circuit topology that gave it a slightly different sound. The C24 was the only production stereo mic to employ the celebrated ‘brass ring’ version of AKG’s CK12 capsule; when it was discontinued it in the late ’70s, it was replaced by the solid-state C422, which used the then-new ‘nylon ring’ CK12. There was one major revision of the C24 during its long career, when for some reason AKG replaced the original diamond-pattern grille mesh with

a reportedly inferior grid-like mesh that is said to cause resonance problems.

Like most single-point stereo mics, the C24 has one fixed capsule and one that can be freely rotated. Turning this upper capsule allows any mutual angle between zero and 180 degrees to be configured. Pattern control is remote, either from a dedicated controller or the power supply depending on the version, with nine switchable patterns running the usual range from omni to figure-8 with cardioid in the middle.

Although the C24 perhaps doesn’t quite have the same legendary status as the C12 or the related Telefunken ELA M 251, its rarity and its similarity to those mics means that it is nevertheless highly sought-after. Yet, having analysed and repaired several C24s, Derek Bargaehr concluded that there was room to improve on the design, both mechanically and electrically. On a C24, there are no

detents or markings to guide you when setting the mutual angle of the capsules, and the mechanism places a Meccano-like strut right in front of each capsule, which presumably affects the sound. Derek found this mechanism unreliable and prone to failure, and consequently, his stereo mics use a new and more robust ‘tank turret’ rotation system with a 120-degree span that is divided into 15-degree steps, and with nothing in front of the capsules.

Derek’s analysis also led him to conclude that the C24 had a problem with crosstalk between the two channels, thanks to their sharing a power rail for the B+ voltage. His design eliminates this, reportedly delivering better stereo imaging as a result. Going further, the Vanguard V24 replaces the Haufe T14/1 transformers used in the C12 and C24 with larger ones, with the aim of extending the original design’s rather constrained low-frequency response. And there are several other ways in which Derek’s design cheerfully departs from the original, the most obvious being the addition of a pair of filter switches on the mics itself. These do not, as you might expect, cut low frequencies, but introduce a very gentle high-frequency attenuation that is intended to make the frequency response more akin to that of a typical ELA M 251. So, although the V24 is more closely inspired by a specific vintage design than other Vanguard models are, it’s still not intended as a straightforward clone of the C24 — as if such a thing could ever be straightforward.

Housed in a smart wooden box (from which it can be annoyingly hard to remove), the V24 ships in a large Peli case along with its shockmount, cable and power supply, and a spare 6072 valve. The PSU is very nicely made and, like the N24S supplied with later C24s, has the pattern controls integrated rather than on a separate remote. Unlike the N24S, it also has a duplicate polarity-inverted output for the top capsule, which is intended to allow Mid/Sides decoding on the fly. The supplied cable appears to be of good quality, albeit rather prone to retaining its coiled shape, and connects at either end using 10-pin Neutrik XLRs. The shockmount is the same model developed for the V44S, and I’m still not enamoured of it. It attaches using a threaded ring that is fiddly to get into place, and because it holds the mic only at its very base, there is considerable potential for droop when used horizontally.

Given the price and the fact that Vanguard

Vanguard V24 £9400 pros • An excellent modern interpretation of the classic AKG C24 design, with sympathetic modifications.

• Handmade using high-quality parts.

• Has the same versatility and sonic strengths as the original.

cons • Scarily expensive.

• Mounting system isn’t great.

summary Inspired by what is probably the greatest stereo valve mic ever made, the V24 is a labour of love that actually improves upon the original in some respects.

thickness, and to the depth of the backplate chamber. These, added to the inevitable variation brought about by handbuilding something so complex, meant that there could be significant sonic differences between CK12s from different eras or even from the same batch.

It’s hard to make useful generalisations, especially when you consider the possible effect of ageing on 60-year-old capsules, but most experts seem to agree that the final variant of the ‘brass ring’ CK12 with red paint around the edge, as used in the C414 EB microphone, was typically brighter than earlier versions.

I raise this point partly because I love nerdy microphone facts, but partly because it relates to my evaluation of the Vanguard V24. I was lucky enough to have an AKG C24 around for comparison, but this particular C24 is a bit of a Franken-mic, and although it has authentic CK12 capsules, they are the abovementioned final version, which are not strictly period-correct for the C24. I’m not sure which iteration of the CK12 BeesNeez have tried to replicate in their copies.

“When you consider what goes into building something like this, using premium parts and handbuilt capsules, it’s obvious that it could never have been a cheap microphone.” have attempted to improve on the basic C24 design in other ways, it’s a bit disappointing that they haven’t come up with a better alternative, especially as the original C24 standmount is a brilliant piece of engineering. The older mic will also fit into something like a Rycote InVision suspension, whereas the V24 is too girthy.

Let’s Twist Again

Capsule Wardrobe

The basic V24 ‘chassis’ and rotation mechanism were actually introduced in Vanguard’s V44S Gen2, which I reviewed back in 2022 (www.soundonsound.com/ reviews/vanguard-audio-v44s-gen2).

Whereas that mic was finished in the company’s trademark ‘pinot noir’ gloss, however, the V24 is stripped back to bare steel and polished, which gives it a purposeful look, and brings it somewhat closer to the C24 in appearance. It is, however, larger than the C24 in all dimensions, and has the same unusually open, single-layer headbasket mesh as the V44S. I do worry that this might not offer enough protection to the capsules under all circumstances, though it was not a problem at all during the review period.

The key issue for anyone seeking to recreate the classic AKG large-diaphragm sound is the availability of suitable capsules. The ‘brass ring’ CK12 was notoriously difficult to manufacture, with a very high rejection rate. AKG themselves gave up on it in 1979, and today, only a handful of boutique manufacturers have the skills and experience to make authentic replicas. Look up the prices of these, and the waiting lists, and you’ll begin to understand why the V24 costs as much as it does. For the V24, Vanguard are using the Australian-made BeesNeez CK12, which seems to be well regarded.

AKG tinkered with the design of the CK12 throughout its run, making numerous changes to diaphragm material and

Side By Side In terms of specifications, the most eye-catching number quoted by Vanguard is a self-noise figure of less than 13dB (A-weighted). That’s very impressive for a valve microphone, and not only improves on the C24 but on quite a few classic solid-state models too.

Other specifications are fairly typical, with sensitivity given at 9mV/Pa, and 0.5% TDH reached at a sound pressure level of 126dB. In comparison to my test C24,

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VA NGU A R D V 24

the V24 was about 3-4 dB more sensitive.

How did it compare sonically?

In terms of imaging, this was a frustratingly difficult question to answer, because it’s hard to persuade the two mics to present the same stereo perspective.

The V24 is a very large mic, and its bulky shockmount makes it awkward to position another mic next to it. It’s also challenging to ensure that two mics of this type are pointing in exactly the same direction. The lack of accurate visual cues makes it hard to precisely identify the ‘front’ of each mic, never mind line the two up perfectly. They can’t easily be swapped out without moving the stand, because each requires its own specific mount. And, though I’m not sure how much difference it makes, the spacing between the two capsules is perhaps a centimetre greater in the V24. The upshot is that although the stereo picture often seemed slightly different for the two mics, it’s hard to know whether this is down to their intrinsic properties or the inevitable differences in placement.

According to Vanguard, the elimination of crosstalk in the V24 design means that the stereo field should have appeared wider for the V24, assuming the same pattern settings on both mics. Because I was expecting this, I listened out for it, but inasmuch as it was present in the recordings I made, it was often overshadowed by the aforementioned other differences. Assuming Derek is correct, the crosstalk in the C24 is as high as -25dB, which sounds alarming — but if you think about it, that’s around the same level as the maximum off-axis rejection that AKG claim for each capsule at 1kHz, and much lower than the typical off-axis rejection at, say, 5kHz. So although this level of electrical crosstalk looks ugly on paper, it’s likely to be lower than the ‘acoustic crosstalk’ under most circumstances, and especially at high frequencies where the detail that helps us locate sources within a stereo image resides.

What matters is that in practice I have no complaints about the stereo imaging of either mic; and, as a lover of coincident stereo techniques, what’s particularly

ALTERNATIVES If this review had gone to press a month earlier, I’d have said there were no direct alternatives to the V24. However, at this year’s NAMM Show, Peluso introduced a stereo valve mic called the P24, which is also based on the AKG C24, and which is said to be shipping now.

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The Vanguard V24 (right) is larger in all dimensions than its inspiration (left). Note the Meccano-like struts in front of the capsules on the C24!

great about both is the availability of those intermediate patterns between cardioid and figure-8. For me, the happy place for roles like drum overheads and piano recording is usually to be found somewhere in the crossed hyper or supercardioid region, and this is equally true of the C24 and the V24.

High Society What, then, about the tone of the V24?

The C24 I used for comparison purposes is the only one I’ve ever heard, so I don’t know how representative it is, but it’s definitely a bright mic, and noticeably more so than the V24. The degree of tonal similarity was quite dependent on the polar pattern, but typically I found that the C24 had more going on above 5kHz, while the V24 was fuller in the low mids and beefier in the 1-3 kHz range. There was enough of a difference that I often had a clear preference on any given source, but not consistently in favour of

one mic or the other. Instruments and voices that sounded perfect on the C24 sometimes came across as a little harder on the V24, while those that suited the V24 could be over-bright on the C24. Many things sounded great on both — and in both cases, you don’t need to be making stereo recordings to benefit from a stereo mic. For vocal recording, for example, there’s a lot to be said for putting both capsules in cardioid, setting the mutual angle to zero and running them through different outboard setups, giving you either a choice of sounds or chains separately optimised for loud and quiet sections.

As I mentioned earlier, the capsules in my test C24 are from the brightest period of CK12 manufacture, so I can well believe that the V24’s slightly more balanced tonality is actually more representative of the true C24 sound, if there is such a thing. Both mics’ on-axis tonality varies with polar pattern; the V24 was clearly preferable in figure-8, where my test C24 got pretty fierce in the upper mids, but I marginally preferred the C24 in omni. And, whether or not you can hear a difference in the stereo presentation, some of Vanguard’s other changes to the design are undeniably improvements, most noticeably the lower noise floor, the detented rotation system and the addition of the filter switches, which do just enough to usefully modify the high-frequency response without totally changing the character of the mic. Most importantly, these changes have been made without undermining the core strengths of the C24: the unique on- and off-axis response of the CK12, and the slight softness and forgiving quality introduced by the use of a valve as the impedance converter.

Although it’s probably less expensive than an original AKG C24, there’s no denying that the V24 is a very costly microphone. But when you consider what goes into building something like this, using premium parts and handbuilt capsules, it’s obvious that it could never have been a cheap microphone. The Vanguard V24 is a serious investment, and if you’re seriously considering it, you should arrange to try one out in your own studio.

There’s no other microphone like it on the market — and only 800 or so mics like it anywhere in the world. £ £9400 including VAT. T Emerging UK +44 (0)118 402 5090 E sales@emerginguk.com W www.emerginguk.com W www.vanguardaudiolabs.com

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Jürgen Engler • Die Krupps ‘Wahre Arbeit Wahrer Lohn’ J O E M AT E R A

W

hen they formed in 1980, German group Die Krupps broke new ground with their electro-industrial metal hybrid. Led by vocalist, guitarist and keyboardist Jürgen Engler, the group’s blending of electronica with heavy guitars went on to influence numerous other bands, including

Depeche Mode and Rammstein. Sonic experimentation has always been key to the Die Krupps sound, and Engler is particularly fond of the groundbreaking synthesizer sound heard on the 12-inch single version of ‘Wahre Arbeit Wahrer Lohn’, which was named Single of the Week by NME magazine on its release in 1981.

In Sequence “Back in the day, because we didn’t have much money, we couldn’t just go to whatever store and pick from 20 different keyboards. We were kind of stuck with what was available, so on this particular track I used a synth that the studio we were recording in — Studio Lambertz Brahm in Düsseldorf, Germany — had, which was a Yamaha CS15. And the owner had built this big sequencer that was the size of a telephone switchboard, and he had it directly hooked up to the Yamaha synth’s interior wiring.

Jürgen Engler: “The sound for the song was created by running the sequenced Yamaha through an Electro-Harmonix Big Muff distortion pedal and an MXR Model 126 Flanger/Doubler.”

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“Basically, the sound for the song was created by running the sequenced Yamaha through an Electro-Harmonix Big Muff distortion pedal and an MXR Model 126 Flanger/Doubler. We plugged the Big Muff into the synth, out of the headphone output and then back into the input of the synth. We then cranked up the Big Muff pedal to give us this really nice distorted and fat analogue sound. Then we heavily sequenced it. Though I had originally written the song on a Casio VL-Tone, when we went into the studio to record it, I basically programmed the big 12-step sequencer by using patch cables for each note.

“Also, we used a real drummer on this track, compared to the rest the material on the album we released earlier that year, Stahlwerksynfonie, which had programmed drums, and were programmed in a similar way to the synth sequence: a Roland CompuRhythm CR-78, which was directly connected to the big sequencer that also functioned as a drum programming machine, so that the individual sounds like snare, bass drum, hi-hats etc could be patched to a rhythm of your needs, overwriting the pre-programmed rhythms of the CR-78.”

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Mixing Breakthroughs Moments, Mindsets & More...

We invited five top engineers to share some of the key ‘aha’ moments in their mixing journey, and to offer advice for those who’ve recently embarked on their own.

NEIL ROGERS

I

always enjoy talking with fellow engineers about mixing, but it’s particularly fascinating when they’re top professionals who offer valuable insights that I can try to apply in my own work. This interview is a case in point.

I started by inviting five very successful mix engineers (more about whom you can read in the ‘Meet The Engineers’ box) to look back at their careers and pinpoint some key ‘breakthroughs’ on their path to mixing professionally — lasting lessons learned on a project, pieces of advice that struck a chord, or perhaps deliberate changes they made in their own approach.

We covered such questions, of course, but, inevitably, we also wandered

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more widely into the skills and mindset that are necessary to become and remain a successful mix engineer.

They all offered lots of advice and insights that every student of mixing should find valuable.

Solid Foundations A common thread was how the engineers’ early grounding in recording studios provided both inspiration and a level of technical understanding that would set them in good stead for their future careers. David Wrench, for example, worked for several years as the house engineer at Bryn Derwen, an independent studio in Wales, and it was this experience that made him want to mix: “I decided I had to really figure out this mixing thing. I would see so many projects come through the studio,

where so much care had been taken during the recording stage and then watch as they ‘overworked’ individual elements or overly mucked around with things at the mixing stage. Respecting and ‘working with’ a production is a huge part of how I mix.”

Barney Barnicott recounted a similar experience. “I was lucky enough to work for eight years in a great independent studio, and I noticed how the most talented engineers made things look so simple and easy. Early on, I realised one of the most important things in mixing is to do as little as possible in terms of processing, and really work hard on balance.”

Marta Salogni, who offered up some great thoughts on the personal and artistic approach to mixing music, was keen to point out that the technical grounding she gained in studios allows her creative freedom today: “I am very thankful for my past as an engineer, as it gave me the tools to be able to not worry

Meet The Engineers Marta Salogni is an Italian-born, Grammy-nominated and Mercury Prize‑winning recording engineer, producer and mixer working from her own Studio Zona, in London. Artists she has worked with include Björk, Depeche Mode, Sampha, MIA, Bon Iver, Black Midi and English Teacher. www.martasalogni.com David Wrench is a Welsh artist, writer, producer and mixer, now based in London. Nominated for multiple Grammy Awards, Wrench has also mixed numerous Mercury-nominated albums, including the 2017 winning album Process by Sampha and the 2021 winner Collapsed In Sunbeams by Arlo Parks. Some of the many other artists who feature on his credits list include David Byrne, the xx, Frank Ocean, Bon Iver, Sampha, the Pretenders, Caribou, Jamie xx, Blur and Yannis & the Yaw. www.davidwrench.co.uk Barney Barnicott has mixed records for the likes of Arctic Monkeys, Sam Fender, Jungle, Coldplay, Kasabian and many more. Barney primarily works from his own Blue Bell Hill

Studios in Kent, in the UK, where he has recently been mentoring engineers and delivering music production retreats. www.barnybarnicott.com www.musicproductionretreat.com Mikko Gordon is a Grammy-nominated mix engineer and producer who was mentored by Radiohead producer Nigel Godrich, and has worked with various bands and artists including Thom Yorke and Jonny Greenwood (Radiohead), Arcade Fire, Idles, Gaz Coombes (Supergrass) and Colouring. Today, he’s based in his own Hackney Road Studios in London, and also lectures in music production at Goldsmiths, University of London. www.mikkogordon.com Jolyon Thomas has won Music Producers Guild awards for Breakthrough Producer and Song Of The Year, and has worked on two Mercury-nominated albums as well as several British and worldwide number one albums. His clients include U2, Kendrick Lamar, Royal Blood, Soft Play and the Horrors. www.jolyonthomas.com

about the technical aspect of a mix any more. The technical side is so assimilated it has become second nature. That is why I feel I can afford to focus purely on feelings, empathy, and the emotional response of my work.”

Mikko Gordon also has an eye very much on creativity, which he attributes in part to working under creative engineers, such as Nigel Godrich, but says: “I was always drawn to mixes and productions that sound ‘interesting’ rather than ‘good’, and I think it’s a great bit of advice to keep in mind. Looking back I can see how much of the music I grew up liking has a distinct sonic identity, which still influences how I work today.”

their clients’ wishes, and David Wrench suggested an imaginative technique for getting into the head of a musician or artist: “The life of a mixer can be a solitary one these days, and you can’t underestimate how different music

David Wrench: “I often try and imagine someone is in the room with me: how would I feel if I was playing this mix with Iggy Pop standing behind me?”

Understanding The Client

Photo: Rachel Lipsitz

In the early parts of our conversation, Jolyon Thomas chose to discuss the importance of interpersonal skills and client relationships. “I learned very early on the importance of finding out what [the client] is feeling, or trying to aim for with a piece of music. You need to find a way of getting that vital information. I once had a call from a fantastic artist I was working with, she said her favourite part of the mix was this vocal effect on one word! If you’re tuned in, these sorts of things will appear with not much effort.”

Most of the other engineers also offered advice about understanding

sounds when you have other people in the room. I often try and imagine someone is in the room with me: how would I feel if I was playing this mix with Iggy Pop standing behind me, for example?”

Marta Salogni observed that “We as people are all different and don’t experience emotions in the same way.

The bridge to those differences to me is an open communication and the desire to be their interpreter. My own ego stays beyond the threshold of my studio door. With my work I am merely asking questions: does this sound like anger to you? Does this feel like love or is it more like longing? Are we on the same plane, looking at the same landscape? If not, I’ll get back to work and plough this mix further, until I hold it up like a mirror and my artists can see themselves in it and say, ‘That’s it, that’s me, that’s what I wanted to convey.’ Nothing brings me more joy than to hear those words.”

Mikko Gordon also suggested an interesting tactic. “In recent years,” he recounted, “there have been a few instances where I’ve felt something was missing in a track I’ve been mixing. In these situations, I’ve mixed the song as it is, but then also made an alternative version with a tape loop or some kind of other atmospheric texture underneath.

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Barney Barnicott: “There was a point in my career where I realised that the mixes I was less worried about turning out well turned out better!”

Barney Barnicott

I’ve then sent both versions to the artists, and nine times out of 10 that’s the version that gets used. So I’ve learned to trust my instincts, but also to make sure I’m respectful to the artist and to include the original version without additions.”

Breakthrough Moments A couple of the engineers shared anecdotes about specific moments when their careers moved forward, or they gained a valuable shot of confidence.

Barney Barnicott said he “was towards the end of mixing a Kasabian album, which I’d also recorded, when the American side of the label told us they were going to get Andy Wallace to mix the record for the American market. I just assumed that was it for me, as there was no way I could compete with Andy’s mixes. When he had finished, we set up the whole album back to back on an A/B button with everyone there in the studio to compare the two mixes. It was extremely daunting, but to my shock, most of his sounded pretty much exactly the same as mine. The band chose half Andy’s and half mine. It really gave me a lot of confidence in what I was doing.”

David Wrench highlighted an experience that occurred on a specific project. “I was mixing the Caribou record Our Love and it was a project where my love of low end began to really cement itself into my mixing style. We were mixing the record at Strongroom Studios

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[in] London and managed to find a way of auditioning the mixes in some nearby clubs (Plastic People and Fabric) during the day. We spent a lot of time fine-tuning the relationship between the kick drum and the bass — changing samples or using EQ or ducking. It was around this time that a clutch of projects I was involved with came out in quick succession and things began to take off for me.”

On that occasion, David had clearly gone the extra mile to ensure those mixes would work in the right environment, and it seems this wasn’t the only occasion.

“Being based in Wales,” he said, “it was relatively easy to get your mixes onto Welsh-speaking radio, which gave me a lot of opportunities to compare my mixes to material being played on national radio — [BBC] 6 Music, Radio 2 etc. Being a primarily self-taught mixer, I was able to use this ultra real-world context to hear how my mixes stood up.”

Learning To Listen Sadly, we can’t all use broadcast radio as a mix reference check! But all the mixers discussed playback systems, as well as the somewhat deeper, more complex subject of developing your listening skills — and understanding precisely what it is that you’re supposed to be listening for.

Mikko Gordon stressed that “aspiring engineers [should] not just think about

mixing in terms of just sitting between a pair of speakers”, and offered a simple tip: “Use a Bluetooth speaker in another area of the room, where you can switch between a kind of micro/macro style of listening. When I mentor engineers,” he continued, “encouraging them to listen and engage the musical/emotional side of their brain can be one of the hardest things to convey. I like to encourage people to work on this side of mixing and think of it as building muscle memory. Ask yourself: how does the music make you feel? In my opinion, music is about having an emotional reaction when listening to it: what is the essence of the song and how is the production supporting that? What can you then do in the mix to emphasise that?”

Similarly, Barney Barnicott told me that “I’m not always in front of the speakers, and spend a lot of time walking around with the music on. It helps distance myself from the technical side of things and it helps me hear issues I need to address.”

Barney also made an observation that really struck me: “There was a point in my career where I realised that the mixes I was less worried about turning out well turned out better! So try not to put pressure on yourself and find ways of not actively engaging your brain and overthinking. You have to train yourself to keep some distance so that you keep that window of critical listening open. Trust your instincts, work instinctively. Oh, and try to not listen in solo!”

This idea of focusing more on the music than the technical aspects came up a lot, and Marta Salogni has clearly thought pretty deeply about how she listens. “The job of a mixing engineer is undeniably technical,” she said, “but the openness to the experience of how something should sound is vital. I always A/B with my eyes closed. Only then can I really start asking myself: is this actually better? I also have a simple rule that if something bothers me, or attracts my attention as needing to be changed, I only act upon it if I notice it twice. Our ears can trick us, and they can fill in the gaps too. Something that might feel like an urge in this instance might just be an itch that gets forgotten in a moment. To truly weigh up how important a change is, is to give it time to dissolve itself.”

Jolyon Thomas told me that he’s “worked in a lot of different studio environments, and can find the pressure of time restraints and making a project

work in a new environment to be quite stimulating. I like to think I can mix with anything... Developing your listening skills is one of the more tricky aspects of mixing to explain. Sometimes, it’s very hard to leave things alone. It’s not unusual to get files from people and it’s just covered in plug-ins, and one starts to meddle. In some instances, the real pro in me can take the plug-ins off, get a really musical balance, do some automation and so on, and send the mix back. Doing this, however, can be scary. The amount of plug-ins doesn’t matter, of course, who cares? Really, it’s about having the confidence in your taste and your ears to be able to do that.”

David Wrench suggested that developing the ability to focus on the emotional impact of a mix is hugely important. “I run intensive five-day seminars for Mix With The Masters [https://mixwiththemasters.com], and one of the things that I double down on is encouraging participants to make this ‘mental shift’ regarding mixing.

Learn to listen to music and approach a mix like a listener. Think about the flow of the music, dynamics, and worry less about the small technical details — [they] can often be picked up or sorted later, anyhow. Find your own way of detaching your active brain. For example, I sometimes take a nap in the afternoon at my studio and will put a track on a loop very quietly in the background. It might sound odd, but when I get myself going again, I seem to instinctively know what a mix needs!”

compass of a clear listen. A break every couple of hours can’t be skipped. Or we might be led down the wrong path, and consequently have to spend a long time retracing our steps.”

Marta Salogni: “If something bothers me, or attracts my attention as needing to be changed, I only act upon it if I notice it twice.”

Mikko Gordon describes how he mixes “as ‘working in waves’, with a gentle reminder to myself that it doesn’t have to be perfect straight away. I try not to spend too much time on one aspect of a mix, and instead look to gradually refine things over multiple passes. Regular breaks and resisting the urge to hit the solo button are key to this approach.”

Barney Barnicott’s approach struck me as being pretty similar. “I only work on a mix

for around four hours,” he told me, “and take lots of breaks. If I can, I like to chip away at a mix over a few days with the gaps in between providing that crucial objectivity that we quickly lose when sitting in front of the speakers for hours.”

David Wrench, on the other hand, told me that he “can work very long hours when mixing, but I always make a point of finishing a mix in the morning when my ears are fresh.”

Gear Talk!

It’s the ear not the gear, as the saying goes, so it’s not surprising that relatively little of our conversations centred on the tools of the trade. Yet, it does seem that there were some points in these engineers’ careers when the gear itself had a notable impact.

“We often crave old analogue gear,” Jolyon Thomas shared, “but it can have serious quirks! I once mixed a record on the EMI Neve at ICP studios [in Belgium] and when listening on headphones, I noticed the mix would go wider and narrower upon each revision, which was a bit odd. I began to notice that James, my engineer, would be walking over to the sofa and tweaking some button that, unbeknownst to me, was

Break Time Taking a break from working might seem like a simple thing, but I think it was telling that nearly all the engineers stressed the importance of breaks, and more generally of working to keep your sense of perspective when mixing. For example, here’s what Marta Salogni told me: “I think breaks are an essential moment of defragmentation for our brain. A mix is an exercise in an endless series of decision making; it can become overwhelming at points, and perspective is essential. That can take the form of a walk outside, a few pages of a book, listening to another record, switching on the radio... Anything that gets my body, brain and ears off this chair. Our ears get tired the more we listen to sounds, so naturally we start to become accustomed to louder volumes, to brighter frequencies, and lose the

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TECHNIQUE

MIXING BREAKTHROUGHS

the air con — basically, if the console was too warm it would affect the stereo image! I gather he knew this but, being a seasoned professional, was covertly doing his job to keep my earphones balanced. We take them for granted, but plug-ins are so good now and provide us with very reliable tools — UAD have really got my back!”

David Wrench returned to his love of low end when talking about equipment, and described incorporating a sub into his monitoring setup as a breakthrough moment. “I can recall sitting in a mastering session with Mandy Parnell,” he explained, “and suddenly being able to hear all this low-end information that I just couldn’t hear on my monitors. I’ve since become a strong advocate for using subs when mixing and even sometimes sit with my feet on mine when mixing — I love to ‘feel’ the bass and kick drum as a physical thing. People often comment on the low end of my mixes and I do think it’s an element of my work that has helped me stand out.”

Barney Barnicott explained that a more recent acquisition has had a positive impact: “I grew up mixing on a Neve, and then an SSL, and I still mix on the SSL now here at Blue Bell Hill Studios. I’ve often struggled with mixing in the box, it Mikko Gordon

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just takes so long and can be a bit boring, but due to some of the projects I get it’s sometimes necessary. I’ve recently bought an SSL UC-1 [DAW control surface] and it’s really helped: I can mix quickly and don’t have to worry about looking for plug-ins, and it actually sounds

Mikko Gordon: “I try not to spend too much time on one aspect of a mix, and instead look to gradually refine things over multiple passes.” exactly like an SSL K Series. It’s been a bit of a gamechanger for me. I just feel comfortable using my hands and I like not having to look at what I’m doing, just listening. It doesn’t replace the sound of the desk, but it’s the next best thing.”

This idea of tools being there to facilitate a way of working as much as for their sound also seems important to

Marta Salogni, who said that “from the very start my aim has been, as well as building my own set of tools, to build an environment and a workflow that allows me to follow my instincts, and to not get stuck in cerebral troubleshooting, which to me breaks the focus of a mix.

Before starting a new project, I make sure that I can dive deep into it without my attention being broken or needed elsewhere. That means that I check my equipment, my desk, my computer, and make sure that there will be no bumps on the road ahead for the day.”

On a similar note, Mikko Gordon told me that he’s “an advocate for using a smaller palette of gear when mixing.

I know I like 1176s and I’m a big fan of spring reverbs, for example, and having a core set of tools that you lean on can help you get to a certain point quicker and without distraction. That being said, I always like to build in a little space for experimentation, and I’m a big fan of using analogue tape for texture or for loops and effects.”

Mixing Tips As we headed towards the end of our conversations, I asked the engineers if they’d like to offer any final practical advice for aspiring mix engineers, so here’s what they said.

Marta Salogni: “I would recommend not letting other people’s methods limit yours. If a session arrives with someone else’s routing, naming, format — remodel it. Don’t get caught in that spider web if it doesn’t make sense to you. All our brains work in different ways. I colour code instruments tracks in the same way in Pro Tools or with different markers on the desk. That way I don’t have to read what they are, I don’t have to engage that part of my brain which is very conscious, and I can instead react quickly to an impulse. I am methodical in my process.

I’ve learned that to be a very valuable tool for my workflow: it allows me to dive from a starting point and retrace my steps if I need to.”

Mikko Gordon: “If you’re new to mixing I think it’s important to learn to finish a mix. We improve by building up our experience over different projects and even if you’re perhaps not 100 percent happy, take the lessons and apply them to the next project.”

Barney Barnicott: “Try focusing on the core elements of a mix first: the drums and main rhythmic elements,

Just Make It Sound Cool?

I could easily have filled this issue with many more of the insights that these talented mix engineers shared during our conversations, but I have to draw the line somewhere, and I’d like to draw this article to a close by sharing just a few more of their thoughts that particularly resonated with me.

Marta Salogni: “I know many people nowadays feel like there is very little sacred space/time in which we can let ourselves feel truly immersed in a creative thought; often I hear loved ones, friends and colleagues telling me that they feel they can only do so at night, or very early in the morning. I try every day to create that feeling for myself, a prolonged night or dusk, for as long as I can. A sort of meditation if you wish.

That’s how mixing feels to me.”

Jolyon Thomas: “Don’t underestimate the importance of fun in all this; try to inject a bit of randomness into your workflow — this might be using some cool guitar pedals or an old synth you have sitting in a corner. Sometimes, I’ll just run something over with absolute disregard for the result, it’s just fun. Even if you

Jolyon Thomas

Photo: Eva Pentel

and vocals. Once happy, begin to bring in the other elements, knowing how much space you have to play with. A nice thing to try is leaving the bass until last, as it takes up so much room — being able to then shape the bass to fit with the other elements can really work well. Also, a good tip is to not start with the drums. It’s very easy to obsess about getting a great drum sound, but it is more effective to shape the drum sound with some context, once everything else is happening. Lastly, I don’t mix with anything on the master bus — this forces me to work much harder to get a great mix.”

David Wrench: “Even if you have been mixing for while, try not to be too set in your ways. I recently mixed an amazing project for Yanis & the Law, featuring the late, great Tony Allen on drums. I got a bit stuck with how to place the drums, so I called the project’s producer to ask about how the drums should sound. He explained how Tony approached his drums sonically, with a focus on the midrange rather than your more traditional drum processing. Once I started down that path everything came together and it was a great lesson for me.”

get one little transition or effect, that can make a huge difference to the mix and leave a little signature.”

Barney Barnicott: “Try to be brave when mixing and take a big-picture approach. Don’t labour over a mix that isn’t working: pull the faders down and

Jolyon Thomas: “Don’t underestimate the importance of fun in all this; try to inject a bit of randomness into your workflow.” start again. At the end of the day, mixing is about taste and instinct — so learn to trust yourself!”

Mikko Gordon: “I think it’s helpful to decide what kind of mixer you aspire to be, or where your strengths are. I have a lot of respect for engineers who can mix in an appropriate style for a certain genre,

but I’m personally drawn to mixes that have a clear sonic identity. It’s a simple thing, but make a point of listening to tracks you like on your monitors or mix headphones — it can often be surprising how bold some of the mix choices are!”

David Wrench: “I remember a conversation I had with a singer from a Welsh band called Topper, many years ago. We were setting up to record vocals and I asked the singer how he wanted his vocals to sound. He said ‘Just make me sound cool!’ The objective is always the same: just make it sound good, make it sound cool — you just need to figure out what that is for each mix!”

I hope that you’ve drawn some inspiration from these conversations — I certainly did — and that they’ll help you make progress on your own mixing journey. And for the more inexperienced engineers out there, who are just embarking on that journey, there’s one recurring theme that I’d like to draw to your attention: although it’s very important that you learn the technical side of mixing, you should focus at least as much on learning to interpret what artists really want their music to feel like, and on developing your own personal approach to mixing.

www.soundonsound.com / May 2025

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Motorised MIDI Controller

Melbourne Instruments Roto-Control Melbourne Instruments turn their attention to hands-on MIDI control...

RORY DOW

M

elbourne Instruments are quickly becoming the masters of motorised controls. William Stokes reviewed their Delia and Nina synths in SOS October 2023 and January 2025, respectively, and was impressed with the motorised encoders, which are perfect for showing preset values correctly. This time, they’ve ditched the oscillators and envelopes in favour of a MIDI controller that offers tight Ableton Live integration and general MIDI programmability.

The Roto-Control is a desktop device with eight motorised knobs, mechanical buttons, LCDs for labelling, and a workflow designed for Ableton Live mixing, plug-in control, and external MIDI gear. It is one of the first compact MIDI controllers to feature motorised rotary encoders, offering instant visual and tactile recall of your settings.

Build & Design The Roto-Control is small for a MIDI controller, measuring 341 x 112 x 44mm.

The chassis is made of sturdy plastic. The buttons are mechanical with a satisfying 1970s computer keyboard clunk.

The main attraction is, of course, the eight motorised encoders. Unlike standard MIDI controllers, where knobs and faders remain static, the Roto-Control’s encoders can move automatically, adjusting to reflect

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the current state of your mix, plug-in or external synth. They feel very satisfying to use and exhibit no wobble. Each knob is touch-sensitive, which is helpful in selecting DAW tracks. One of the most impressive features is their ability to change haptic characteristics. For example, for a pan knob, they will behave like a normal potentiometer, stopping at the left and right boundaries with an indent in the centre. If you assign a parameter with, say, three values, such as a filter type (low-pass, band-pass, high-pass), the encoder will snap to three positions, and if you assign it to something endless, like tempo, it will turn endlessly, as required.

They really are impressive and practically silent, too. You’ll barely notice them move.

Beneath each encoder is a colour display, which displays labels for the encoders and the buttons below. Whether you’re tweaking an EQ band, muting tracks, or balancing mix levels, these displays ensure you always know what each knob is controlling. Sadly, however, they don’t display values. Melbourne Instruments rely on the motorised encoders to ‘show’ the value, which they do — up to a point.

For example, if you’re controlling panning, it’s easy to see the value by looking at the knob position. However, suppose you are adjusting EQ frequency or any control with a non-linear response. In that case, it’s often impossible to tell what the value is by looking at the knob position. I asked

Melbourne Instruments about this, and they acknowledged it could be a useful addition for the LCDs to temporarily show values in place of the labels. They didn’t make any promises, but I hope it’s something we’ll see in a future update.

Below the encoders are eight RGB-lit buttons which can be freely assigned or used for transport controls. The buttons to the left are used for mode switching, MIDI learn and other system functions.

Modes & Functionality The Roto-Control isn’t just a generic MIDI knob box. It comes with Ableton Live integration, which is achieved by installing the Roto-Control app, which will also install the scripts needed for Live functionality.

Roto-Control features three primary modes, each tailored to a specific workflow.

Mix Mode transforms the Roto-Control into a motorised MIDI mixer for Ableton Live, giving you control over track volume, panning, sends, mute, solo, etc. When a Live session is loaded, the track names and colours sync with the built-in displays. The motorised knobs physically move to reflect volume levels, panning or effects sends.

The RGB buttons function as mute, solo and record-arm controls for each track. Unlike traditional MIDI controllers, where you have to ‘catch’ a parameter before adjusting it, the Roto-Control’s encoders and buttons always reflect the current values in your

Melbourne Instruments Roto-Control £389 pros • Awesome motorised encoders that offer tactile recall.

• Tight Ableton Live integration (Bitwig coming soon).

• Motion Recorder adds automation looping.

cons • We’ll have to wait and see if it will integrate with other DAWs. • No on-screen values shown for the encoders.

summary For Ableton Live users (and soon Bitwig), synth tweakers and live electronic musicians, the Roto-Control could be a perfect low-cost MIDI controller with some seriously impressive motorised tech.

project. Hopefully, future firmware updates will expand native support for other DAWs. Melbourne Instruments tell me they are working on Bitwig integration next and are looking into others. But for now, Ableton users will get the best experience.

Plug-in Mode is, unsurprisingly, used to control Ableton devices and third-party plug-ins. Plug-ins are initially unmapped and require some setup. Thankfully, this is an easy process involving hitting the Learn button, selecting a control, and tweaking a control in Ableton. This will create a link between the two controls. Encoder haptics and naming happen automatically, although you can use the Roto-Setup app to edit things. After this, the mapping will be recalled whenever you load up the same plug-in, so it’s a once-only process. I like this approach. Some controllers automatically map every parameter, which means scrolling endlessly through long lists. But Roto-Control’s approach means you get to choose what’s important. If you do want to map all the controls on a particularly large plug-in, it will take some time (you can have many pages of parameters if required), but

I suspect most people will just map the things they need quick access to or the things they automate the most.

MIDI Mode offers MIDI CCs for those who don’t use Ableton Live or want to create mappings outside the DAW. There is support for MIDI control changes and NRPNs, but not SysEx. The standout feature here is the Motion Recorder, a built-in recorder that allows you to record and loop knob movements. You can record up to eight independent automation lanes, loop in forward, reverse, ping-pong or random order, and adjust the loop length and speed in real time. It’s great fun for studio experiments or live performers who want evolving filter sweeps, LFO-style effects, or generative textures. You can even sync the Motion Recorder to MIDI Clock.

Creating a MIDI setup can be done on the unit or via the Roto‑Setup application, which can be downloaded from Melbourne Instruments’ website. It allows you to customise each encoder and button to behave as you like. You can change the MIDI channel, labels and colours, and program the encoder haptics to become either stepped or have indents so they behave like real potentiometers. MIDI is always sent to the MIDI DIN outputs and USB simultaneously. There is no way to target one or the other.

Conclusion The Melbourne Instruments Roto-Control is an intriguing product. I was hoping that it would provide a similar level of Ableton Live integration to the Ableton Push controllers (just without the pads). It doesn’t manage the same level of detail, but it does the basics extremely well. One workflow I particularly enjoyed was Mix Focus mode, where you get the mix settings for the selected track: volume, pan and all sends.

The knobs, of course, update to their proper positions every time you select a new track.

Plug‑in Mode was useful once I accepted its limitations. I planned to map all the most useful controls for, say, EQ Eight and then reach for the controller every time I wanted

The back panel offers two USB-C ports: one for power and the other for MIDI data (the motorised encoders cannot be powered from USB alone). It’s also nice to see a full complement of 5-pin DIN MIDI in, out and thru ports for direct hardware control.

to tweak the EQ on a track. It does work, but the problem I found was not being able to see the values for parameters like frequency. Relying on the knob accurately pointing a certain way isn’t a substitute for knowing your EQ band is set to 340Hz. And so I kept having to look at the computer screen when making edits. Plug‑in Mode is still useful, however. Instead of using it for surgical edits, it makes a great automation controller. Map it to your favourite 303 plug-in’s cutoff, resonance and envelope amount controls, automate to your heart’s content, and then sit back and watch the knobs move with your freshly recorded acid tweaks. Very satisfying.

MIDI Mode is also useful if you accept its limitations. It won’t be a behemoth, all-in editor for your Yamaha DX7. But suppose you’re looking for something small that could integrate nicely into a live rig, or a versatile studio assistant that can add physical controls to any synth, sampler or drum machine. In that case, the Roto-Control will prove extremely handy. The motorised encoders are great in MIDI Mode, too.

And if your equipment sends MIDI control changes, the encoders will move with the incoming MIDI, providing proper two-way communication.

I like the Roto-Control. It’s not perfect.

In some ways it is a work in progress, but that’s fine because it’s a new product from a relatively new company. Creating DAW integration for all DAWs is a considerable development task. Hopefully, the product will be successful enough to warrant support for more DAWs and even bigger controllers in the future.

Don’t dismiss the Roto-Control just because you don’t use Ableton Live.

At this price, the Roto-Control delivers a compact, premium controller with features nobody else can touch. Even if you only use MIDI Mode, it could be a valuable addition to any studio or live setup. I hope Melbourne Instruments will continue to develop the controller side of their business because these motorised knobs are perfect for the job.

£ £389 including VAT. W www.melbourneinstruments.com

www.soundonsound.com / May 2025

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Universal Audio UAFX Enigmatic ’82 Guitar-amp Emulation Pedal

Dumble amps are both revered and rare as hens’ teeth — so it was only a matter of time before UA focused their considerable modelling talents on a stompbox emulation.

Universal Audio UAFX Enigmatic ’82 £380 pros • Unique amplifier voicing.

• Lovely touch-sensitive singing distortion tone.

• Wide variety of ‘personalisation’ options.

• Not excessively noisy at high gain.

• Some of the additional cabs are really nice!

cons • The app doesn’t display the control positions used in saved presets.

summary The Enigmatic ’82 offers up some of the greatest guitar tones of all time.

The recorded sounds of Lukather, Ford and Carlton et al, at various stages of their careers, are era-defining for many.

Those sounds are all in here, but with so much depth and adjustability via the app, there’s a whole lot more here for us to discover our own tonal voice, too.

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DAV E LO C K WO O D

I

n the interests of full disclosure, I have to say that the UAFX Enigmatic ’82 Overdrive Special Amp (to give it its full title) is the first of these UAFX digital guitar-amp emulation pedals that I’ve reviewed where I’ve not actually owned and used ‘the real thing’. I’m well acquainted with a number of clones and recreations, but I’ve never played an actual Dumble. But then neither have 99.9 percent of the guitar‑playing world either, so maybe we are all equal on this one.

We’ve all heard them, of course, unless you haven’t listened to the recorded tones of any of the world’s greatest guitar players over the last four decades. But then we need to ask, which Dumbles have we heard? They are not all the same. Quite apart from the existence of a few different models, even versions of the same model can differ quite significantly. That was the point of the Dumble process: designer

Howard Alexander Dumble would tweak each amp’s circuitry to meet the preferences of an individual player. Such a labour-intensive production process meant that there were only ever a small number of Dumble amps in circulation, perhaps fewer than 300, driving up resale prices to astronomical levels.

Sadly, little of that revenue seems to have made its way to the famously reclusive Mr Dumble, who passed away in 2022, leaving a legacy of some of the greatest electric guitar tones of all time.

Overview The name Enigmatic ’82 may seem to imply that UA have modelled an early ’80s Dumble Overdrive Special, but the software control app actually gives you access to the many circuit mods, revisions and voicings that occurred throughout the ’70s, ’80s and ’90s — so, in effect, you can ‘build’ your own Dumble. The Overdrive Special, in all its guises, is perhaps the amp that most fully epitomises the ‘Dumble tone’ — a kind of singing lead tone that’s bright without being harsh, combined with a kind of lower-mid girth and richness that you

don’t get from, say, a Marshall operating at about the same level of overdrive.

Although it has footswitchable clean and overdrive ‘channels’, a Dumble ODS isn’t a genuine two-channel amp: the lead channel’s overdrive is achieved by cascading the clean channel into an additional preamp stage. Dumble began by modifying Fender designs that generally have their tone stack located right at the beginning of the circuit, which usually puts it before the distortion‑generating stage in a cascaded-channel amp. This can be a problem when it comes to trying to find tone-stack settings that work for both the clean and distorted sounds. Dumble addressed this issue in the ODS by adding an additional, internal bass, mid and treble EQ stage, specifically for voicing the distorted tones. Further options include Jazz and Rock tone presets, as well as significantly different tone stack designs that became available during the life of the model. Dumble would use different bright capacitor values according to where the player generally liked to set their input volume control, and there was also an option to switch in a solid-state FET high-impedance buffer stage before the first tube. If anyone is reminded of the Fetron used in some early Mark-series Mesa amps, the comparison is not quite on point, as Mesa used it instead of a V1 tube, but I do have a 1981 Mark IIB that mirrors the ODS basic topology quite closely, and can get you into some of the same tonal territory, albeit perhaps with a little less finesse.

All of these choices would be made

by Mr Dumble swapping out components and making adjustments, listening to the guitarist playing and responding to what they were hearing, but once set, that would be the amp — they weren’t generally user-adjustable parameters. Of course, none of us will have access to that kind of ‘real-time design and refine’ process, but we’ve got the next best thing in the UAFX Control app for the Enigmatic ’82, because all of these parameters appear there for us to set and design our own Dumble. Let’s get stuck in...

Volume sets basic amp gain. Generally, you’ll want this as high as possible without compromising the clean sound, as it also sets how much signal is available in the overdrive stage. In Alt mode, accessed via the centre mini switch, the Volume knob sets the amount of added Room ambience, lifting the sound out of its close-miked dryness. Overdrive does what you’d expect, with a range from off (fully counter-clockwise) to ‘more than most people can use’. In Alt mode, the enigmatically named Ratio parameter determines the clean-to-overdrive volume change as you switch the cascade in/out.

The Bass, Middle and Treble controls are self-explanatory, but the secondary functions are important: the Bass control turned up anywhere above halfway rolls off a bit of high-end (Deep mode) when using the Classic tone stack type, or boosts midrange when using the Skyline tone stack, each function replicating a hardware switch that would have appeared on the original amps.

The Alt function for the Middle knob gives you a Presence control, whilst the Treble control offers up a choice of four bright cap settings: Off (no bright cap), 150pF, 196pF and 300pF. If you think that’s getting a bit picky, it really isn’t: the bright capacitor (a single passive component across the active lugs of the input volume) is fundamental to the voicing of many amps and the effectiveness of some of the controls that follow. The bright cap is progressively shorted out as you turn

up the volume, so the value you want will rather depend on where you tend to leave that set: the lower the volume, the more you’ll be hearing the capacitor seeming to boost the highs at different points in the spectrum depending on the value chosen.

And There’s More...

You also have a choice of four fundamental amp types, from a 50W model with a high plate voltage on the output tubes, for a higher gain but more compressed sound, to a 100W model with a “very stiff power supply” and lower plate voltage, for something faster and more responsive, plus a couple of intermediate settings in between. There’s a Preamp Boost mode that partially bypasses the tone stack.

There’s the famous HRM (either ‘Hot Rubber Monkey’, or ‘Hot Rodded Marshall’, depending on whose version you believe) ‘internal’ overdrive-EQ stage. And a Master Gain parameter sets how hard the phase-inverter and power‑amp stage is being driven, which has a significant effect on the amount of ‘bloom’ and harmonic saturation generated.

Then there are the miked cabinet emulations: three by default and a further six (three more than any other UAFX amp pedal) when you register the pedal with UA via their UA Connect app. The default set offers a 1x12 25W Celestion greenback, miked with a ribbon 160; a ‘boutique’ ported 2x12 with 65W speakers, miked with a 57 dynamic mic and a ribbon 121 mic; and a 1990s partial open-back

All Under Control The Enigmatic ’82 follows the standard form factor of this series of UAFX amp pedals, with stereo in and out jacks, a 9V DC power (400mA) inlet, and the six rotary controls all having a secondary function except for the Output level.

Most of the physical controls have useful secondary functions.

www.soundonsound.com / May 2025

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U NI V E R S A L AU DIO UA F X E NIGM ATI C ’ 82

The associated app allows you to choose and tweak both amp and cabinet models, including features such as the bright capacitor value and three-band EQ.

Dumble-design 2x12 with EVM12L speakers, miked with a U67 capacitor model, “custom made for an artist”.

‘Bonus’ cabs include a 4x12 with Vintage 30s; a 2x12 with JBLs; a 1x12 with an EVM12L; a stripped 4x12 with greenbacks, double-miked with a ribbon (160) and a 57 dynamic; a custom-made Dumble 1x12 with a vintage JBL G-125, miked with a 421 dynamic; and a 1970s Dumble oversized, closed-back, front-ported 2x12 with EVM12L speakers, miked with a 414 capacitor. Some of these bonus cabs are pretty special and well worth registering for!

what you want. For this model, more than any of the others in the UAFX range, it really is a shame that the UAFX Control app isn’t able to display all the control settings used by the presets. There would be a lot of useful learning to be had there.

And while we are talking about the app, of which I have been fairly critical in the past,

as well. I can say with great certainty that I know what a miked Deluxe Reverb or a Marshall Super Lead sounds and feels like coming back at me on studio monitors with no processing. That’s the basis on which I assert my opinion that these UA pedals offer uncannily accurate renditions of the amps, speakers and mics they model. I don’t have the same experience with a Dumble, but what I can say is that Enigmatic ’82 is capable of some truly stellar sounds that don’t sound exactly like anything else I’ve experienced.

And that was, to some extent, what made Dumbles so desirable: they didn’t just sound like a hot-rodded Fender, or a Mesa or a Marshall — they had something unique, and so does this pedal.

It took me a while to get it dialled in, but when you do, there is something special there that I don’t hear in anything else.

This is another fabulous piece of work from the UA modelling team!

“Dumbles... didn’t just sound like a hot-rodded Fender, or a Mesa or a Marshall — they had something unique, and so does this pedal.”

Design-a-Dumble There’s a lot a to play with in designing your own Dumble, and some magical sounds to be found. And, it has to be said, quite a lot that are not so magical — this is not a ‘plug in and go’ instant gratification amp. I’d definitely recommend starting by auditioning the presets, so at least you’ll know what it can sound like before you start trying to set it up for yourself and find you aren’t quite getting

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I have to say that I am no longer having any connection issues, having updated all the pedals and the app to the latest versions, so do give that a try if you are still having connection issues.

UA’s Enigmatic ’82 is capable of some truly ‘golden’ tones, the like of which most of us will only ever have heard in recordings, from crystal cleans to singing distortion voicings. Bear in mind, of course, we won’t just have been hearing a Dumble and a miked speaker in those recordings, but often a whole chain of recording and mixing processors

£ £380 including VAT. W www.uaudio.com

Introducing new Black Lion Audio PBR LIT Patchbays.

Eliminate confusion when routing your racks of precious gear with the PBR-LIT patchbays from Black Lion Audio. Built to proven Black Lion standards of build quality and signal conductivity, the PBR LIT patchbays — available in TRS and TT/ DB25 configurations — feature customizable, brightly-lit LCD displays for every input and output. Simply connect a USB keyboard directly to your LIT to customize the displays, or use the included PatchCAD LIT software to create custom text to export directly to your patchbay using the same port.

With the patent-pending PBR LIT patchbays, you’ll have an easier time routing your signals while adding instant vibe to your studio. Save the labels and markers for your next yard sale.

Visit www.blacklionaudio.com to get your rack LIT.

©2025 Black Lion Audio. All rights reserved.

PBR TT LIT: 96-point TT/dB25 patchbay with LCD label system, audiophile-grade jacks, and 3 normalization modes.

PBR TRS3 LIT: 48-point TRS patchbay with LCD label system, audiophile-grade jacks, and 3 normalization modes.

ON TE ST

The newest member of the Avantis family is a portable powerhouse.

MIKE CROFTS

A

Allen & Heath Avantis Solo Digital Mixing Console

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llen & Heath are an established and highly respected brand in the world of professional audio, especially in the area of digital mixing consoles. The current A&H digital console line-up comprises four main ranges, namely the Qu, SQ, Avantis and dLive series. The Avantis models sit second in the hierarchy, in terms of pricing and features, below the flagship dLive series and above the very popular SQ mixers that you’ll find almost everywhere, either installed in a live venue or as part of a portable on-the-road setup.

The original ‘full-size’ Avantis offered a stepping stone for those aspiring to the full dLive experience but who didn’t have the budget for the dLive, and the most recent Avantis model, the Avantis Solo, packs all of the mixing performance of the larger version into a very compact format that is aimed at smaller venues and portable applications. Allen & Heath have not compromised on audio performance or flexibility, so the only differences between the Solo and the original Avantis are that the Solo has one (very large) touchscreen instead of two, 12 faders on the top surface rather than 24, and a smaller local I/O count on the rear of the console (see ‘Ins & Outs’ box).

The Avantis range uses exactly the same XCVI FPGA-based mix engine that powers both the dLive and SQ mixers, running at the same 96kHz sample rate.

The preamps are taken directly from the dLive S Class. The Solo, despite having limited rear‑panel I/O, has the same 64-input, 42‑bus mixing capability as the dual-screen Avantis, and provides six customisable fader layers for control over up to 72 channels/busses. Each layer of 12 faders is fully configurable to suit the task at hand, however your personal workflow requires it. There are 16 freely assignable DCA groups available, and the bus architecture is fully configurable.

Main Features The Avantis Solo seems aimed mainly at owner-operators, smaller and medium-sized venues, and corporate AV users. The trademark Avantis features are all there, accessed via

Allen & Heath Avantis Solo £7740 pros • Powerful, high-end mixing in a portable unit.

• Excellent touchscreen-led workflow with a useful amount of physical controls.

• Extensive DSP capabilities.

• Highly expandable with optional cards.

• Strong, all-metal casing should withstand road rigours.

cons • Limited local I/O mean a remote stagebox will be required in most scenarios.

• Relatively high price point, but reflects its high-end capability.

summary This compact console builds on the existing Avantis tech, and offers all the same processing power and digital connectivity, while taking up considerably less space!

a touchscreen-driven workflow, in a unit that’s small and light enough to be carried in the boot of a small car. Like the full-size dual-screen Avantis, the

Solo features two 96kHz card slots that can be populated with any of the dLive digital I/O cards, adding up to 128 I/O per slot in Dante, MADI, Waves and gigaACE digital formats, among others. For venues that have already invested in Avantis hardware, show files can be directly loaded between the two models, so there are no compatibility or continuity issues.

Likewise, the SLink Ethernet port is fully compatible with all of Allen & Heath’s digital stageboxes, as well as their ME personal monitor mixing system, making the Avantis Solo a drop-in replacement for an existing A&H-equipped venue.

Especially interesting in a theatre or AV context is that the Avantis Solo can work seamlessly with RF mic systems from Shure and Sennheiser, including the ability to show battery life data and RF receiver levels right there in the channel strip. Other areas of wireless integration include display of RF channel ident, control of audio levels and muting.

The Avantis Solo supports A&H’s range of IP controllers — the IP1, IP6, and IP8 — via the Ethernet ports on the rear, allowing for artist-controlled monitor

mixes, say, or simply to add extra faders alongside the console.

Despite its smaller size (about half the footprint of the twin-screen version), the Solo retains the full onboard processing power of the Avantis, and matches its very low latency figure of 0.7ms. It is equipped with 12 ‘RackExtra’ effects slots for reverb, delay and the like, features an Automatic Mic Mixing (AMM) mode, and can be upgraded with the dPack expansion ($1599), which adds a number of additional processors including analogue emulations, specialised dynamics processors, studio-style effects, a valve preamp emulation and more. These can replace the ‘default’ processing modules in a channel, or be loaded up into one of the RackExtra slots.

The mixer can be remotely controlled in a number of different ways: via the Avantis Director software, the Avantis MixPad app for iPad, or the OneMix personal monitor mixing app for iOS. See ‘Remote Control’ box for more info.

First Impressions The Avantis Solo arrived in a rather large box, too big to pick up and carry by myself, so I opened it up straight away to extract the mixer; it was suspended in large sections of shaped packing foam, and the unit itself was easy to handle once freed from the shipping materials. The console measures only 546 x 627 x 269mm (WDH) and weighs 17.5kg, so it’s definitely what you’d call portable. The Solo has a substantial look and feel to it, and it’s equipped with lifting bars that are positioned low down on the sides. There are smaller but similar-looking bars on either side nearer the top surface, but don’t try and pick it up using these, as they are not handles but mounts for attaching peripherals such as an iPad (there’s a neat accessory available from A&H designed for exactly this purpose).

Boot-up time was around 40 seconds (to be expected as there’s a lot of The Avantis Solo is based around a 15.6‑inch colour touchscreen, and sports 12 motorised faders.

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processing going on inside), and the Solo runs very quietly. I began by downloading and installing the latest firmware, and this process was extremely straightforward — Allen & Heath have made everything to do with software as automatic as possible, and the mixer recognised the new firmware version and installed it without any fuss.

Of course, the main attraction is the large touchscreen that’s used to control and inform you about every aspect of the mixing and processing functions.

It’s bigger than my first telly and a couple of diagonal inches bigger than my current laptop. It’s also sharp as a new pin, and the capacitive touch response makes it a joy to use. Sitting in front of the Avantis Solo definitely brings that ‘pro league’ mixer feel, and with the flexible approach to just about every aspect of the mix architecture, it’s much closer to the dLive than the SQ in terms of what you can see and do.

The mechanical side of things is very nicely designed and made, with super-smooth (and fast) motorised faders, and every button and rotary control has a quality feel. The press-buttons are all of the illuminated rubbery type, which I personally like as they have a positive action. Some are on the small side, but they all have clear legends printed on

overall black surface theme. One point about everything being configurable is, of course, that knobs and buttons can have several different functions, and so can’t be permanently labelled. Because the touchscreen takes up a fair amount of width, the 12 faders are well spaced, and there’s plenty of finger room around them and all the other surface controls.

As already mentioned, the large touchscreen is very easy on the eye and fingertips, and is really at the heart of almost everything that happens within the machine. There are usually at least two ways of achieving the same control function, and users will be able to set up the Solo according to how they need to use it — moving up from a ‘fixed architecture’ desk to a fully flexible ‘put it all just where you want it’ system opens up so many possibilities for developing and streamlining the mixing workflow.

Worthy of mention here, and typical of the ergonomic design

“The large touchscreen is very easy on the eye and fingertips and is really at the heart of almost everything that happens within the machine.” them. The rotary knobs, on the other hand, are quite chunky and have a sort of ‘matte smooth’ finish with decent-sized grip mouldings — I much prefer these to either the shiny chrome-effect knobs or, worse still, those with a horrible rubberised coating that retains much of whatever the previous user had for lunch. The knobs are all plain black, with integrated LED indicators for the ones below the screen, and fit in well with the

There are six fader banks available, giving control over up to 72 channels/busses using the 12 faders. On the far side of the console are 12 completely assignable soft buttons.

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Ag

detail, is the surface illumination built into the Avantis Solo. We all know that all live gigs have to take place either in complete darkness or blazing direct sunlight, and the touchscreen has plenty of brightness to help with the latter, while its sheer size means that on-screen characters and symbols are large enough to see easily. For mixing in darkened rooms, there’s a neat LED strip that shines upward from underneath the lower handrest bar and illuminates the fader area quite nicely. You can alter the light balance from deepish blue to almost white, and although it does show up bits of detritus, I much prefer it to the top-mounted light strip on my own SQ7, which I find a bit distracting as it points directly towards me when I’m sitting at the desk.

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Show Time To get some hands-on time with the Avantis Solo, I took it along to a couple of band rehearsals. My colleague Patrick Woodington (a long-time A&H user) also used it for a musical production along with one of my AR24 stageboxes, and so managed to get in some quality familiarisation time too.

We both had very positive experiences of the Solo, and although I didn’t explore all of its extensive feature set, I really liked using it and found it easy to configure and quick to get around the basics. The large, bright touchscreen is simply first-class, and makes going back to my SQ7 seem like I need new glasses! Everything seems to have been very well thought through, and implemented in a clear and logical way.

The Avantis Solo has a very compact footprint for such a powerful piece of kit, and the build quality is excellent. The whole thing appears to have been

Ins & Outs Round the back, the Solo has just six analogue inputs and outputs, all on XLR connectors (the full-size Avantis doubles that to 12 analogue ins and outs). There’s also a seventh XLR in line with the others, but this is a digital AES3 port; just something to bear in mind when groping round the back to plug a talkback mic in, or quickly hook up a last-minute feed for the video guy... The limited number of local inputs means that, in the majority of applications, the Avantis Solo will need to be hooked up to at least one remote stagebox via the SLink Ethercon port.

Whilst this effectively bumps up the starting price of an Avantis setup, the good news is that in an upgrade scenario it will work with any of the range of extenders used with existing A&H mixers, so it

could be a simple plug-andplay upgrade to an existing theatre installation. Integration and operation with any supported remote units is automatic and seamless as far as the user is concerned (as with the SQ), and you don’t have to worry about firmware version compatibility either up or down. The local I/O on the rear can still be very useful for things like talkback, local walk-in, sound effects and so on, and it does mean that the Solo can be used completely standalone if all you need is portable AV in a campus or corporate setting.

The important point is that the Solo is completely flexible, anything can be patched to anywhere, and it will handle the full complement of 64 inputs and 42 busses however you set it up.

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Local analogue connectivity is limited to six XLR inputs and outputs. A further pair of XLRs offers AES3 digital connectivity, the SLink Ethercon port allows connection to Allen & Heath’s range of digital stageboxes, and the two I/O Port slots can accommodate a range of digital expansion cards.

both well-designed and very well put together — an impression that’s reinforced by the use of Neutrik XLR and Ethercon connectors throughout.

The desk is light enough to be portable but reassuringly solid and stable at the same time (cables aren’t going to drag this across a table!), and the handles underneath are well placed.

A number of thoughtful design elements stood out for me, such as the fact that there are USB‑A ports (for stereo recording/playback, saving show files and updating firmware) on both the front and rear panels, so you can use whichever is most convenient.

Similarly, the headphone output is duplicated on both ‘full-size’ 6.5mm jack and a 3.5mm mini-jack (how many times have you lost a mini‑jack adaptor?).

In terms of operation, the UI is easy to navigate and should be easy for guest engineers to pick up. As on any mixing desk, the layout takes a bit of getting used to, but there are lots of user-assignable keys over to the right of the faders so you can customise the workflow to suit your needs. There’s not a huge amount of encoders to work with, but the touchscreen easily makes up for this. Likewise, there’s no dedicated metering on the console’s surface,

but the channel overview at bottom of the display shows levels in detail, so it wasn’t something I missed.

It can feel like there is a lack of fader banks — there are only six, and these will soon fill up once you’ve got additional layers for effects sends/returns and bus masters alongside the inputs. That said, the DCAs can help you manage multiple channels/busses, and assigning them is a breeze. You can also lock a channel in place so that it stays put regardless of what layer you’re on, which is very useful indeed!

One slight niggle is that the console has to be rebooted every time you

Remote Control The two network ports on the Avantis Solo’s rear panel allow it to be connected to a computer network, which in turn means you can control the desk from any computer (Mac or PC) on the network running Allen & Heath’s Avantis Director software. This mirrors the screen on the desk and allows complete control over the console. It can also be used offline — great for preparing for a show, as a basis for tutorials, or for simply learning to use the desk.

With a wireless router or access point on the network, the Avantis can also be controlled over WiFi from a tablet running the Avantis MixPad app.

This, again, allows for complete mixing control, as well as portability, so the engineer can control the console from anywhere in the venue. You can control the desk locally and remotely at the same time, and multiple iPads can be connected to the desk simultaneously.

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Two further apps are designed for controlling monitor mixes. Avantis 4You is a personal monitor mixing app, intended for use by performers, and runs on iOS and Android devices. It can be configured to allow full control over an aux mix, including EQ and compression, or it can present a simple four-fader UI to keep things simple. Avantis OneMix, meanwhile, offers all the functionality of the MixPad app, but is locked to a single aux output to prevent any changes to the main mix.

Finally, Allen & Heath’s IP range of hardware controllers includes simple wall-mounting button and encoder panels, designed for installations, as well as the IP6 rotary controller and IP8 fader controller, which both sport assignable soft keys and can be powered over Ethernet. These can be used to expand on the console surface’s controls, or set up remotely to control different mixes or venue zones.

“The Avantis Solo is an impressive little mixer, offering top-quality performance with a high-end feature set.” change network settings, and with a 40-second boot-up time, that could become annoying (though it’s not something you’re likely to have to do often). Still, a dedicated reboot option would have been useful, instead of having to shut down and then manually power it up again.

Another feature that some users might miss is unbalanced RCA inputs for local line‑level devices, which could potentially have been very useful for AV work.

In terms of processing, the Free Pack that you receive on registering the console includes three different graphic EQs and the Tube Stage Preamp module, and these all work very well. However, the Avantis dPack represents a significant upgrade, and includes a very nice 12‑band parametric EQ as an alternative to the graphic EQs on the mix outputs.

The other plug-ins in the bundle are all low‑latency, easy to use and sound really nice.

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Avant Garde The Avantis Solo currently lists around £7500 (as of March 2025, at least two major retailers are around this price). This is significantly lower than you’d pay to get into dLive territory, and as the Avantis seems to be much closer to the dLive than the SQ in terms of its flexibility and features, this doesn’t seem like a hefty price for something that’s going to be pretty rider-friendly and offer a high degree of control with 64 inputs, 72 control channels and 42 fully configurable and assignable busses. For most applications a remote input expander of some sort will be necessary, so unless you already have an Allen & Heath system that includes one or more of these, this will have to be taken into account when budgeting.

All this considered, the Avantis Solo is an impressive little mixer, offering top-quality performance with a high-end feature set. The Avantis Solo benefits from established, advanced Allen & Heath processing and an excellent touchscreen-led user interface, while its flexibility and expandability make it a serious contender for any application where true professional capability in a compact package is required.

As always, though, there’s no substitute for trying something out yourself, so if you’re considering a purchase you should definitely dive into the Allen & Heath website and read through the extensive information available — and then try to arrange a hands-on demo through your favourite dealer. £ £7740 including VAT. T Audio‑Technica UK +44 (0)1132 771 441 W https://distribution.audio-technica.eu W www.allen-heath.com

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INSIDE TRACK

Metalcore presents unique production challenges — and few are better placed to meet them than genre veteran Jordan Fish.

PAUL TINGEN

A

s co-songwriter and co-producer with Bring Me The Horizon, Jordan Fish has had a defining role in their success. Late in 2023, he left the band to pursue a production career, since when he’s made successful records with Poppy, House Of Protection, Dutch singer S10 and metalcore band Architects.

Fish co-wrote and produced almost the whole of Architects’ 11th album, The Sky, The Earth & All Between, which went to number two on the UK album charts.

Jordan Fish • Architects

Play It Safe

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Photos: Ed Mason

Jordan Fish at Middle Farm Studios.

Although he works in genres where virtuosity is highly valued, Jordan Fish cheerfully admits that his own instrumental skills are limited. He chuckles that his impressive BMTH live keyboard rig had “a lot of buttons for not much talent! I can’t play keyboards to save my life. My first instrument is probably guitar, and I used to play trumpet when I was younger, which also gave me some music theory knowledge.

Playing a little bit of guitar is helpful for rock music, and I have an understanding of how a drummer plays, which is helpful for programming.

“It’s not that I have a problem with virtuoso players. I would love to be one, but I’m not. I respect anyone who is amazing at their instrument. I often tell myself that one day I’m going to really learn how to play the piano properly, but it’s probably not going to happen because I’m working all the time. My musical mind is my strength, more than my ability to play. Any time I should have dedicated to learning to master an instrument, I spent trying to make things. Producing and writing is a natural fit for me, so I would say that my first instrument is the laptop.

“As a teenager I started messing around with production on a computer using Nuendo, and then I went to the School of Audio Engineering in London for a year, where I learned Pro Tools. After that I worked in a studio called Phoenix Sound, at Pinewood Studios in London, where they were also using Pro Tools.

“I worked there for a couple of years, starting in reception, making tea, and earning the minimum wage and all that stuff. They let me use the studio in the evenings and at night, and eventually they let me run sessions. So I was very much on the trajectory of being a recording studio engineer. Over time, I was working with bands and I would make small suggestions or I would add stuff. That moved me into production. Because I was into engineering, I was into the more techy side of things, and I guess that’s why I ended up being a synth person, but it’s not really where I came from. I was more an engineer going into production.”

Soft Start Fish spent almost 12 years with Bring Me The Horizon, taking the band’s sound more into electronic music territory, and to some degree bluffing his parts live with a live setup based around Apple’s MainStage software and Arturia’s KeyLab MkII and Beatstep hardware. At the end of 2023, while still in BMTH, Fish connected with Architects, initially in an informal way.

“The first song I had an involvement in was ‘Seeing Red’. Dan [Searle, drummer] and Sam [Carter, singer] from the band sent me the song and I gave some feedback

The Sky, The Earth & All Between is Architects’ 11th album.

via text, and they took it on board. We also went back and forth a little bit over the vocal melody, just over voice memo with me singing some vocal parts, and they then sent me back another version. I never opened my laptop. It was just a mate giving his opinion, but technically speaking it was to some extent production and writing.

“The song came out really well, and in the beginning of 2024, Sam, Dan and I spent a few days together for writing sessions at the studio of the band Royal Blood in Brighton. They had a bunch of demos, and I had some ideas. Each song

was different. Sometimes we’d start from nothing. Sometimes Dan had a riff that I’d pull into my computer. I’d say, ‘Give me half an hour, and I’ll try and find a way to make it a song.’ Maybe I changed the tempo, or added some sounds, or I was tweaking the riff to see if I could make it better. I was just trying to catch a vibe to be able to say, ‘OK, here’s what I’m hearing for this.’

“If we all liked where it was going, we’d work out whether what we had was a post-chorus riff or a chorus, and then build on it and create some other sections.

Usually the first few hours were focused on trying to get some music that we were all excited by, and then melodies came pretty quickly. I would get onto top-lining pretty early on. I top-line quite a lot, if it’s something that the artist or band wants.

Dan and Sam were open to it, so it was a matter of passing around the mic, and if anyone had an idea, they’d sing it, and then someone else might develop that.

The melody writing was very collaborative.

At this point the musical parts were all programmed, even though I sometimes added some guitars.”

One DAW Opens...

While Fish works in Pro Tools, Sam Carter and Dan Searle were in Logic. “I don’t mind

Like Jordan Fish’s own band Bring Me The Horizon, Architects have painstakingly built a huge worldwide fanbase over a lengthy career.

www.soundonsound.com / May 2025

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JORDAN FISH • ARCHITECTS

Architects drummer Dan Searle goes for a take in the Wood Room at Real World Studios.

the different DAWs situation,” comments Fish. “We swapped files using AirDrop and hard drives. It feels a bit messy, but I actually like getting a bunch of audio files.

It makes working in Pro Tools like having a collage, which is more interesting for me.

I’m ripping through the audio files and find the things that work. Sometimes I change the sounds, or add programming, synths or atmospheres. I’d be doing this in front of Dan and Sam, and they’d be guiding.

“I know people say that Pro Tools is less suited for MIDI, but it’s all I’ve ever known. For me it’s always been OK. I’m so familiar with Pro Tools, it feels like my safe place. The way I work in Pro Tools is completely wrong on every front, in any case. For example, I don’t know any shortcuts. I’ve made full albums on my laptop with a trackpad. When I first went to LA, my friend [mixer] Zakk Cervini said ‘You’re going to have to stop working on the trackpad. You need a mouse and a keyboard because it’s insane.’ Switching has sped up my process a little bit, but other than that I dare not change my process much because for some reason it works.

“A lot of people use samplers and stuff, and they manipulate audio within plug-ins.

I don’t really do that. I manipulate audio on the timeline. My timeline looks like an

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absolute mess, because I constantly chop and paste things. Let’s say I’m doing some pitched vocal type thing. A lot of people would map it to MIDI and they would draw in MIDI notes. But I will take the audio and grab [AudioSuite plug-in] Pitch Shift and do everything on the timeline. It’s a messy way of working. The same with beats. A lot of people use [NI] Battery, or some sampler type thing. But I will paste one shots in the timeline. I do use MIDI, but prefer manipulating audio. I just feel a bit more at home there. So my sessions consist of manipulated audio combined with MIDI.”

Making Notes The MIDI data that Fish does generate is rarely played in live. “I have a very small MIDI keyboard, but honestly, I rarely use it. I’m usually drawing notes in with a mouse. I’ve always been a soft synths guy, and keyboards are for live. Before I had the Arturia live setup, I had Nord and Akai keyboards. There used to be quite a bit of piano in the earlier stuff I did with Bring Me, for which I used the Nord. But it was really complicated. The software was primitive. It used to take forever to sample things in there. So eventually one of my techs suggested we run everything from a laptop, to which we connected the keyboards via MIDI.

“In the studio, my sound sources are pretty basic. I use Xfer Serum often. I’ve saved many good presets and sounds over the years, so my Serum’s pretty loaded up with stuff I like for starting points. I usually start with Serum unless I am going for something a bit more sample-based. I also use Spectrasonics Omnisphere a lot, actually more than Native Instruments, because I like not having to make decisions about the sounds up front.

When I open Kontakt, I immediately have to make a decision as to what library to load. Whereas when I open Omnisphere, all the libraries are there. Usually I start from a preset, because I like to get an idea down quickly, and also, I enjoy having the random element of not really knowing what I’m going to get sonically.

“I still use a lot of pianos, particularly Noire, which is a Kontakt piano. It’s unusual because it has noise and stuff built into it, like weird hammer and vinyl sounds.

It’s a bit more characterful than a straight piano, which I quite enjoy. When I’m programming real drums, I use GetGood Drums. I also have plug-ins for bass and guitar, when I don’t have access to a real guitar. If I’m trying to write a riff on a plane, I can just use MIDI. That’s basically it.

Nothing crazy. I have a lot of software that I forget about for a few months and then I’ll

INSIDE TRACK

JORDAN FISH • ARCHITECTS

come back to it. I go through cycles. There are just too many options. Realistically, I could probably do everything with just Serum and Omnisphere.”

On The Farm

Final guitar and bass tracks for the album were laid down at Middle Farm.

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After the sessions at the Royal Blood studio, work on The Sky, The Earth & All Between continued at several other studios. “We wrote pretty much the entire album during the Royal Blood sessions. We also demo’ed vocals there, using a Shure SM7, without being bothered about the fidelity, and we ended up redoing the vocals again later. The band went to Real World studios to record the drums. I wasn’t there because I was doing another record.

The MIDI drums were all written, and Dan replaced them with real drums.

“After that we spent a spent a couple of weeks at Middle Farm Studios in Devon, where we mostly finished the album. We also wrote one more song there, called ‘Blackhole’, which was the last piece of the record that we felt we needed, and that ended up being one of the singles.

We split into two groups at Middle Farm.

Engineer Chris Clancy worked with Adam [Christianson] and Alex [Dean] to record all the guitar and bass parts, and I went in the studio B with Sam to retrack all the vocals.

Sometimes we’d go over to check how they were getting on with the guitars and bass. It was like a boot camp situation to finish their record, which I like.

“The bass and guitar followed the blueprints of the demos. We kept some of the rough guitars from the demos, but Adam replayed most. There were some creative elements in there, but at that point the songs were written and the arrangements mapped out. It was mostly a matter of how to play the parts and with what tone, and how to layer them. I left it to Adam and the engineer to work out which guitar to use. The guitars were recorded using a Neural Amp Modeler, and they would choose their amps. The bass was tracked using a SamsAmp Pedal DI. They gave me the guitar and bass parts frozen, as audio. I could have unfrozen them and dug out the DI recordings if I wanted to, but I never did.

“The engineer at Middle Farm, Pete Miles, set up me up with a vocal chain, and Sam and I took it from there. Pete was the first person I recorded with when I was 16. I trust him completely. The signal chain would have been a Neve pre, and a [Empirical Labs] Distressor on the way in, just like at my home studio, where my

signal chain is a Neve 1073, a Distressor, and a Warm Audio WA-800, which is a Sony C800 clone. I can’t recall what mic we used at Middle Farm, but it will have been a condenser.”

Taking Inspiration During the entire process of writing, arranging and recording, Fish kept a keen ear on the overall musical vision, the performances and the sonics, constantly rough mixing as the project progressed. In so doing he was drawing on his previous production experiences, and also his biggest inspiration, Nigel Godrich’s productions of Radiohead and Thom Yorke. “It’s a completely different genre, but Radiohead is a reference that comes up for me every time I make a song. I think of something they did at some point, all the time. Nigel Godrich’s productions are super, super influential. The first time I heard Kid A and Amnesiac, I loved the almost surreal, super-clean, dry sound. It didn’t sound like it was in a room, it existed just in your brain. I really love that sound.

I also adore Thom Yorke’s solo album, The Eraser. Obviously most of my work is in much heavier music.

Radiohead is like going to the theatre, whereas heavy music is like going to a theme park. You’re going to be blasted out of the cannon and fly!

The emotions are a lot less subtle, but it’s no less fun.

“When working on the Architects album, I would rough mix at the end of every day. We had roughs right from day one. You can call the demos we worked on together rough mixes, and they just grew organically as the whole thing came together. They were always evolving and changing. I tried to keep them sounding as good as possible. The roughs sound pretty decent. I’m tweaking as I go, but unless we get to the end, I’m not really using any automation, because there may still be structural changes to the arrangements.”

Hot & Dry “I am not an audiophile and am normally not too worried about fidelity,” insists Jordan Fish.

“Whatever works, works. But I hate the sound of dynamic mics for vocals.

“I like condensers for the high-end detail, especially when the mic has been massively saturated. I like the really bright, fizzy, white noise element that you can get out of a vocal.

I use the same mic for clean vocals and for the screaming vocals, but we tend to record those during separate sessions. Generally, we go for the cleaner stuff first, because the screaming is fatiguing, for everyone in different ways. And yes, it’s full volume screaming. Sam [Carter] screams as loud has he possibly can.

singing, there are hurdles: how do you get from an angry, intense place to a section with clean singing in a way that doesn’t feel jarring? How do you make all emotions feel coherent when they’re so different? Usually with this music, if the choruses are clean, they end up being quite epic. There’s so much aggression in the louder bits, that the only other feeling that can match this

“It depends on the mic, but I usually don’t use the pad. Honestly, a little bit of preamp distortion is OK, as long as it’s not crazy. The screaming is treated like you would expect, with lots of compression and saturation. In the box I’ll probably use a UAD 1176, and two more compressors. I don’t tend do use reverbs. I prefer delays. I’m not a fan of a super wet vocal, unless I’m going for an effect or specifically a wet section. I’ll certainly keep all my screams completely dry. Reverb gives me the ick. I don’t mind reverb when there’s loads of stuff going on and it’s just sitting in. But when it’s an open verse and you can hear a medium length reverb, it feels synthetic to me.”

home studio during his mixes, giving my two pence of opinions to help the project get over the line.

“Zakk used my roughs as a reference.

He usually does not rip everything I have done out, but he’s also not chasing it. He’s worked with Architects before, so he knows what they are going for. Because he also works in Pro Tools, it’s easy for him to pick up where I left off, and if he likes what I have done he keeps it, but he’s also not afraid to change things. Zakk always laughs at what I have done, because I do terrible things, like using a [stock Pro Tools 1176 emulation] BF76 on the mix bus. I’m like, ‘It sounds good!’ but he always replaces it. I like the BF76. It sounds nothing like an 1176, which is quite present and aggressive, whereas the BF76 is like cling film. In general I’m accustomed to the Pro Tools plug-ins, and also use the Pro Tools EQ a lot.

“Zakk has a great ear, and I trust what he does. Sometimes when mixing, perhaps in one in 10 cases, things gets lost for some reason, or a section’s just not hitting the same, or not doing the same thing, and you have to try and work out why. For example, there was a section where there was a loud and resonant fuzzy guitar, which was technically wrong, but once it got fixed and mixed in, it was no longer doing the same thing. We’d lost something. So there was a bit of that. But I enjoy leaving some room for someone else to do a better job than me.

I don’t want to fight to make my stuff sound amazing, knowing that I’m going to get the extra 20 percent from someone mixing who actually knows what they’re doing.”

Jordan Fish: “I am not an audiophile and am normally not too worried about fidelity. Whatever works, works. But I hate the sound of dynamic mics for vocals.”

Switch It Up A characteristic feature of metalcore is extreme changes between gentle, almost pastoral instrumental sections and full-on metal, accompanied by a switch from singing to screaming. Were these dramatic shifts a challenge to handle?

“Definitely in terms of threading together really heavy elements and clean

is intense emotion. ‘Blackhole’ is a good example of a song that’s very aggressive, while the [cleanly sung] chorus is very emotional and desperate.

“Also, with a metal band, with the guitars and the drums being a real drum kit, there’s so much midrange energy that something has to give. In the pursuit of loudness, the real low sub usually gets compromised somewhat. It’s a matter of space. It’s just a different style of music.

With electronic music, a lot of the energy is super low. But with rock and metal, the energy generally is in the midrange, and mostly from the guitars.

“We wrapped up the album in June of 2024, at Middle Farm. I also did some work at my studio, where I have PMC6 monitors with a Focal sub, and a Universal Audio Apollo x4 interface. Once we had finished tracking and I had finished my work, I delivered the album to Zakk Cervini, who mixes most of what I produce. I actually had to be in LA, so I sat in with him at his

www.soundonsound.com / May 2025

73

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May 2025 / www.soundonsound.com

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www.soundonsound.com /May 2025

75

ON TE ST

Roland V-Stage 76

Stage Keyboard

How does Roland’s V-Stage fare against the other big red stage keyboards in town?

GORDON REID

T

his isn’t the first time that Roland have dipped their toes into the warm waters of piano-plus-organ-plus-synth keyboards.

But this large, red and black slab is clearly aimed at a specific market dominated by another large, red and black slab of stage keyboard. So... how will it fare?

Overview There are two V-Stage models, and I was offered my choice of the smaller version with a 76-key semi-weighted waterfall keybed, or the larger one featuring an 88-key piano action with escapement.

On this occasion, I opted for the smaller model because the pre-release blurb revealed that it offers channel aftertouch, which the larger doesn’t. Unfortunately, as I’ll explain later, this didn’t turn out quite as I had expected.

The layout is attractive, but the pitch-bend and modulation wheels plus the Leslie fast/slow and brake controls (all of which can be assigned additional functions) are positioned behind the keyboard. I’ve stated my dislike for this

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arrangement before, and I’ll do so again.

How would you feel if the whammy bar of one of your guitars was moved to save space or to satisfy some other design purpose? You’d not be happy.

There are four instrument sections revealed on the panel: organ, acoustic pianos, electric pianos and dual synthesizers. Ignoring the organ for a moment, you can play the instruments in each of these sections across the whole width of the keyboard or either side of its own split point, choosing the octave transposition for each so that sounds sit comfortably in their chosen ranges. In contrast, you can create three ranges in the organ section using two user-defined split points. These correspond to the upper and lower manuals plus bass pedals of a typical organ and, in addition to playing them on the V-Stage itself, you can play them using an additional MIDI keyboard and MIDI bass pedals.

Three buttons in the organ section then allow you to select which registration the drawbars are controlling at any given moment. Unfortunately, offering splits rather than ranges means that, while you can layer sections, you can’t have, say,

an organ in the lowest three octaves, a piano and pad layered in the middle two, and a solo synth at the top. While I’m not proposing workstation-like facilities, it’s a shame that Roland have limited the V-Stage in this fashion. On a more positive note, you can choose whether each section transmits to an external sound engine, either in parallel with or instead of the internal sound generation.

The sounds generated by each section are created from building blocks called Tones. To construct a sound, you choose the Tone you want, adjust the available parameters and then — with the exception of the acoustic pianos, which has none — apply the dedicated effects units within each section. Their outputs then pass to three Total Effects — an MFX (multi-effects unit) followed by a delay/chorus and a reverb. There are two routes through

these. The first mixes the outputs from all of the sound generators and then passes the signal to the MFX, the delay and the reverb in series. The second applies the MFX to the output of just one sound generator, and then passes the output from this plus user-defined amounts of each of the other sections through the delay and reverb in parallel or in series, or a bit of both. Roland have used variations of this architecture many times before, and it’s one of those occasions when it’s harder to describe something than it is to use it.

Finally, there’s a Master section comprising a global EQ and a compressor, the combination of which can be useful when dealing with dodgy PAs and inexperienced sound engineers. But if you want any of the sound generators to bypass the Total and Master effects, you also have the choice of sending their sounds to the ‘dry’ sub outputs.

The resulting setup (with the exception of the Master effects, which are saved as system settings) is called a Scene. There are 512 Scene memories, and you can create up to eight chains of these, each containing up to 128 Scenes. There’s also a facility called EX Scene, which offers eight slots. These auto-save the Scenes within them as you edit, which could save your bacon (or your sanity) if you’re fine tuning your sounds and something untoward happens.

But hang on a moment. Where do you save your edited Tones? I could edit sounds and rename them, but nowhere was there an option to save them. So

I saved a modified Tone within a spare Scene, and that seemed to work. But there was still no mention of it anywhere within the Tone lists. What’s more, when I moved away from that Scene and later returned to it, the new name had disappeared to be replaced by the original. The modified sound had survived, but there was nothing to tell me where and what it was. Once I had saved a few of these, it became almost impossible to locate them again.

So I contacted Roland, whose product specialist told me, “We wanted the V-Stage to be a quicker user experience, and we have had feedback in the past that our usual method of saving can be a bit time‑consuming because the user has to save all edited Tones individually, then the Scene. So to simplify things on the V-Stage, everything is saved within the Scene; when you write a Scene, all the Tone information is saved along with all of the Scene data.” I understand this, but I think that this system is — to put it kindly — crap. Is it really so mind-bending to save a Tone in a Tone list so that you can later include it in any Scenes you wish?

In my view, Roland’s engineers have tried to fix a problem that doesn’t exist and, in doing so, introduced the problem that they claimed to have eliminated.

The Organs The V-Stage seeks to emulate four Categories of organ: Hammond tonewheel, Vox Continental, Farfisa Compact, and pipe. Within each category, there are then four Types that provide

Roland V-Stage 76 £2999 pros • It’s attractive and well built.

• The e-pianos are fab.

• Many players will enjoy the V-Piano-based acoustic pianos and the organ emulations.

• It offers numerous effects and flexible routing.

• The underlying sound quality is first-class.

cons • It uses Roland’s horrible Scene-based patch structure, which means that you can’t save your sounds except as elements within a Scene (and even then there are problems).

• The synth section is very limiting and you have to buy other products to edit synth Tones.

• Its aftertouch capabilities are severely limited.

• Its MIDI implementation renders it almost useless as a controller.

• You can’t apply effects to external audio.

• The documentation can be woeful.

summary The V-Stage should be a top contender for your stage keyboard cash, but it’s marred by several significant limitations. If Roland sort it out, it could be excellent. If Roland don’t, it will be a huge missed opportunity.

the building blocks of the sound, and no fewer than 23 pages of parameters to help you sculpt it. These include a flexible overdrive plus a dedicated delay and reverb. There’s also Roland’s latest rotary speaker effect, but despite being able to obtain some pleasing Leslie

The Rear Panel The V-Stage isn’t short of connections. The group of sockets on the right (as you look at the rear panel) includes balanced main outputs in both quarter-inch TRS and XLR forms, plus a headphones socket that echoes them and which, as always, would be better positioned on the front of the instrument. Next to these, you’ll find sub outputs that bypass the Total Effects and Master sections. Then there are stereo line inputs and an XLR mic input suitable

for dynamic microphones. Signals presented here can be directed to either the main or sub outputs but do not go through the effects.

Four control pedal inputs are provided. These accept a damper pedal and three expression pedals or switches, the action of which can be defined within the V-Stage.

You can also use three of them to accept Roland’s RPU-3, which offers the same pedalling facilities as a traditional grand piano.

Digital I/O is provided by MIDI in, out and out/thru, plus four USB sockets: a dedicated USB-A for memory sticks, two further USB-As for connecting external devices such as MIDI controller keyboards and pedals, and a class-compliant USB-C interface for connecting the V-Stage to your computer. In addition to carrying MIDI, the USB-C can act as both an output and an input for stereo audio. With

Roland’s dedicated drivers installed (Windows 10 onward and macOS 13 onward) each of the sound engines, the main and sub mixes, and any audio received at the mic and line inputs can be transmitted separately over a total of 18 channels. In addition, up to four audio channels can now be received and directed to the main and sub outputs.

The final hole is for an IEC mains lead. Thankfully, there’s no wall‑wart here!

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ROL A N D V-S TAGE 76

emulations from this, I couldn’t persuade it to produce the deep ‘whoompf’ that you obtain from the real thing.

As you would expect, the organ panel echoes that of a Hammond organ, and it controls the latest version of the Virtual Tonewheel sound generator found in the Fantom series. With four underlying tonewheel Types called Vintage 1, Vintage 2, Solid and Clean (although nowhere does the documentation tell you what these represent), the organ engine can be tweaked to sound very good, although sometimes a little pristine for my taste.

In contrast, no attempt has been made to imitate a Vox Continental or a Farfisa Compact. So, for example, the tabs of the Farfisa are missing and Boost can only be accessed through the menus.

Similarly, there are no mixtures when you select the Vox sounds, and you can only control the contributions of the waveforms using the menus.

To be fair, I could coax some usable Farfisa emulations from the V-Stage, but these were only a small subset of what Compacts can produce. As for the Vox sounds, I would file these under ‘Vox-ish’. The same is true for the pipe organ sounds; no attempt has been made to imitate the structure or controls of a genuine pipe organ. Instead, the V-Stage allows you to select one of four pipe-y waveforms and then build registrations of this single type of pipe using the drawbars. The results can be quite satisfying, but with none of the flexibility or authenticity of a dedicated pipe organ synthesizer.

by increasing many of the resonances and noises a tad, and using the dedicated EQ to shape the results. I then found that judicious use of the Total Effects helped me to obtain the best from this section.

Perhaps most surprisingly, a touch of compression could work wonders on some sounds, not just controlling the level, but altering the tone of each note in pleasing ways. Now the V-Stage was starting to exhibit the character of an acoustic piano as I experience it.

In contrast, I loved many of the e-pianos from the moment that I tried them. Very little editing is available from the panel so you have to dive into the menus to get the best from some of them, but many are fab just as they are. This didn’t surprise me — I’ve been a fan of the SuperNatural electric pianos since I first played them using an ARX-02 expansion board installed in a Fantom G7.

of research with artists using this kind of stage keyboard and they told us that, with this type of keyboard, they tend to use the synth as more of a supporting role, complementing the other sound engines in layers or splits rather than using it for extensive sound design. With this in mind, we focused on choosing a selection of preset Tones in the synth section that were ‘gig ready’ and easily deployed so that users could quickly get what they needed and be off and running rather than doing extensive programming. This is perhaps where we make our distinction between a V-Stage customer and a Fantom customer. V-Stage is all about quick access and control over essential sounds for live performance.

Should players require more detailed sound design, or assignable modulation, or production features, we see them as Fantom customers, because that’s more closely aligned to their needs.”

Despite their quality and range, the preset synth Tones are unlikely to satisfy everybody.

However, the V-Stage is a member of the Zen/ABM ecosystem so, if you create a free Roland Cloud account, you can access additional sounds compatible with it, including four Model Expansions: the Jupiter‑8, Juno‑106, JX‑8P and SH‑101. Valued at $149 each, these are free of charge for V-Stage owners and greatly increase the selection of sounds available including, where applicable, recreations of the original factory patches. Unfortunately, the V-Stage has no greater ability to edit these than it has the Tones with which it’s shipped. Should you want to tweak them, you’ll have to buy the Roland soft synths from which they’re derived, either paying for a Cloud subscription or purchasing lifetime keys. Alternatively, you can edit or create new Zen-Core ‘SDZ’ Tones on a compatible Zen-Core device and import these into the V-Stage. You can also purchase additional sound packs that range in price from $5 to $20.

Obviously, I wanted to review the V-Stage in its best light, so I attempted to install the Expansions. This caused me to waste a great deal of time before I discovered that, as shipped, the review unit was incapable of loading them! It needed a firmware upgrade first, so I downloaded the latest version and tried to install it. This was when I found that there’s no USB communication for

“Sometimes I’ve felt warm and fuzzy toward it while, at others, I’ve wanted to throw it into the canal at the bottom of the garden.”

Pianos For its acoustic pianos, the V-Stage uses a derivative of Roland’s V-Piano physical modelling technology. Its Piano Designer pages allow you to adjust the tuning, volume and ‘character’ of each note as well as global effects such as the string, cabinet and soundboard resonances, but some other V-Piano parameters — including the Unison Detune that helped to capture the imperfections of a genuine piano — have gone walkies.

Think of it as ‘V-Piano Light’.

There are four piano Types — two grands and two uprights — and, while many of their Tones are usable, the results could be a little precise for my taste. So I tweaked, and then I tweaked again, and then I tweaked some more. The most immediate improvements were obtained

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With Rhodes and Wurlitzer emulations, Clavinets and a range of miscellaneous digital pianos that include the excellent E.Pianos from Roland’s early RD-series, there’s lots to enjoy. There are also three dedicated effects in this section: an MFX, a tremolo with Rhodes, Wurli and Roland modes, and an amplifier simulator offering Rhodes, Wurli and guitar amp modes. Magic happens here. Just dial in a suitable vibrato, a bit of delay, and a touch of drive from one of the amps and... oh damn, there goes another evening. My only disappointment here was the lack of a damper for the Clavis, but I think that we can overlook this given all the other goodies.

The Synthesizer In addition to basic features such as level, pan, pitch, keyboard response, vibrato and EQ, the Zen-Core Synthesizer section in the V-Stage offers just five synthesis parameters — amplifier attack and release plus low-pass filter cutoff frequency on the panel, and amplifier decay and filter resonance in the menus. So I asked the chaps at Roland why they had limited it so severely. They told me, “We did a lot

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ON TE ST

ROL A N D V-S TAGE 76

The V-Stage 76 measures 1181 x 342 x 109mm and weighs in at 15.2kg. The larger 88-note version measures 1331 x 353 x 143mm and weighs 21.8kg.

updates or sounds, and that I needed to format a memory stick on the V-Stage itself. The first I tried failed. So I found another, formatted it and used it to install the update. I was then (finally!) able to install the new sounds and Scene libraries. By the time I had finished, I wasn’t a happy bunny, but I have to admit that the Expansions can sound excellent. To be honest, I don’t know why Roland doesn’t pre-load them before shipping the V-Stage. They’re free, they take up minimal memory, and they make it a better instrument.

In Use Playing the V-Stage as a preset instrument works well, and many of my difficulties were caused by trying to push beyond this. Consider its aftertouch. While this can affect the sound of some synth Tones, I couldn’t find a way to direct it to other destinations, despite spending considerable time trying to do so. So I again wrote to the chaps at Roland to ask what I was missing. As it happened, I wasn’t missing anything. They told me, “After many discussions with professional musicians and artists, aftertouch was not listed as a high priority; their most important hardware needs were playability and portability.

To achieve this in the 88-key model, we used an existing Roland weighted action that is lightweight and compact, but does not have aftertouch. To achieve the same playability with the 76-key model, we developed a new key

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action which is capable of generating aftertouch but, since the V-Stage 76 and 88 share the same firmware, it isn’t deeply implemented, and Roland aren’t promoting it as an important feature.

So aftertouch can only be used with some of the preset Zen-Core tones, and cannot be reassigned to other functions.” What a waste!

Now let’s consider its MIDI implementation. On the surface, it appears to be very flexible, with a manual running to no fewer than 118 pages. But this is deceptive. While changing Scenes transmits MIDI CCs and SysEx, and each section’s Level knob sends a MIDI CC7 (Volume) command, every other message that you generate by sliding, turning or pressing something on the control panel is output as SysEx. This is crazy!

Although you can play external sound engines and soft synths using the keys and the wheels, you can’t control any of their parameters, severely curtailing the V-Stage’s use as a MIDI controller.

Roland Japan told me, “Ultimately, the reason was to try and keep V-Stage as focused on the most important user‑requested features as possible, and not overcomplicate it with less important ones at the expense of simplicity”.

You may feel that that’s a reasonable argument, but I don’t. Either way, I suspect that the MIDI implementation will persuade many potential buyers to shy away from the V-Stage.

Then there’s the issue of the mic and line inputs. The documentation shows

these being mixed into the audio path before the Total Effects and Master sections, so I spent considerable time trying to apply effects to external signals.

Eventually, I gave up and again contacted Roland UK. They responded, “We just double-checked this with the team in Japan and they have confirmed that this is a mistake. The line in and mic in do not go to the MFX as indicated, they just go direct to the master out. Sorry for the confusion, they will correct the version of the manual on the website. Thanks for drawing our attention to it!” Now, imagine that you were a solo artist who had just handed over a few thousand quid for a V-Stage on the basis of the documentation and ecstatic reviews by far-from-independent ‘influencers’, planning to use it to treat your vocals while singing along to your playing... then got it home and found that you couldn’t. I doubt that profanities would suffice. Let’s hope that the manuals have been updated before you read this.

This wasn’t my only problem with the documentation. The paper manual is superficial and the online documents are not always clear. Furthermore, you need to have multiple documents on screen at any given time — for example, the descriptions of the Total delay and reverb effects are contained within the Reference manual, while those for the MFX are to be found in the MFX Parameter List manual.

Consequently, it took me longer to get to grips with the ‘simple’ V-Stage than it did to master many more complex instruments in the past.

But despite all of this, there’s still a lot to like about the V-Stage. It looks good, the built quality feels great, the colour screen is larger and clearer than found on many keyboards of this type, and the feel of the knobs and switches is reassuring. It’s also flexible in terms of allocating sounds to effects and outputs.

Then there’s its polyphony, which is full (ie. every key) for the acoustic pianos and organs. Furthermore, while no numbers are quoted, I didn’t encounter any note stealing when using the e-pianos and synths. You’ll also like ‘Scene Remain’, which means that — whichever sound engines you’re using — any existing notes (including all their effects) decay correctly after you’ve selected and started playing a new Scene. None of this should be overlooked.

the Tones as presets, and you may even be surprised by the flexibility that can be wrung out of the limited synthesis parameters and dedicated MFX. But, inevitably, this lacks the flexibility of the virtual-analogue polysynths found on some stage keyboards, and the powerful ability to create your own sample-based sounds on others.

So, while I’ve tried to reach a clear conclusion about the V-Stage, I must admit that I’ve failed. Sometimes I’ve felt warm and fuzzy toward it while, at others, I’ve wanted to throw it into the canal at the bottom of the garden. It’s the keyboard equivalent of a football match that ended 3-3 with the same team scoring all six goals — three against the opposition, and three into their own net.

If I were a prospective purchaser I would yearn for Roland to sort out the issues because, if it did, the V-Stage could be a fine instrument. In the meantime, I would ask myself whether the different compromises embodied in other stage keyboards would serve me better. Is it worth paying more for a Nord Stage 4 73?

What about cheaper alternatives such as the Hammond SK Pro 73, the Korg

Grandstage X and the Yamaha YC88?

And how about Roland asking you to spend more to be able to edit or create your own synth sounds? Or would I be better served by considering something along the lines of a Fantom 7 or a Kronos 73, either of which can do everything that a stage keyboard can, and myriad other things besides?

Ultimately, if you think that the factory sounds and Expansions are the best you’ve heard and plan to do no more than use these, the V-Stage might be the right choice for you. But if you want to do more, even within the bounds of the user-friendly ‘stage keyboard’ genre, you should consider looking elsewhere to see whether any of the alternatives offer the combination of facilities that you need at the price that you want to pay.

I would like to extend my thanks to Andrew Pimblott at Roland UK, without whose patience, knowledge and advice the V-Stage may well have ended up in the aforementioned canal. £ £2999 including VAT. W www.roland.com

Conclusions?

Despite some serious flaws, the V-Stage is a quality instrument. But can it compete at the very highest level? I think that that’s a tricky question. The Hammond emulation can be excellent, but is it as impressive as the latest Hammond SKs? Well... not for me, although I wouldn’t be offended if you felt otherwise. Similarly, some of the acoustic pianos can sound great but, given the choice, I would still lean toward the Bösendorfer Grand Imperial XL in the Nord library. In contrast, the Electric Piano section really shines.

If you’re not inspired by this, it’s time to become an accountant, or a guitarist... or something. Finally, there’s the dual synth section. You won’t go far wrong if you download the Expansions and use

www.soundonsound.com / May 2025

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AudioSmile Little British Monitor Active Monitors

These compact speakers punch well above their weight — and you can build them yourself!

PHIL WARD

B

ack in the deep, dim and distant past, loudspeaker kits were a big(ish) thing. British hi-fi speaker brand KEF (Kent Engineering & Foundry, if you ever wondered) in particular did pretty good business with their KEFKIT 1, 2 and 3 and Constructor Series. The kits included drivers, passive crossover assemblies, internal foam, cables and connectors, and all the various specialist hardware items required. All the customer had to do was build a couple of cabinets and they’d have a pair of high‑performance KEF speakers for somewhat less than the equivalent finished retail price. But as time moved on, intense competition pushed the price

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of high‑performance speakers inexorably downwards, and speaker kits faded away.

So, when I came across the Little British Monitor (LBM for short), a compact active monitor that’s available in both finished and kit form, I couldn’t resist having a go at the latter.

The LBM comes courtesy of a company called AudioSmile, founded by audio designer Simon Ashton. Simon has a track record in a variety of audio engineering contexts and the LBM is by no means his first loudspeaker design. Simon’s experience with speakers is clearly apparent from the genuinely unusual and interesting electro-acoustic engineering that can be found within the extremely compact dimensions of the LBM.

Although Simon describes the LBM as ‘little’, ‘tiny’ might have been more appropriate, because the LBM truly is minuscule. This will mean that while the LBM offers great benefits should you be looking for something easily portable or to fit on a small desk, its compact dimensions and very small bass driver will also have significant implications for the low‑frequency bandwidth and maximum volume levels achievable.

My decision to build and review the kit version of the LBM added some complexity to the review process. In particular, it was important that the finished speaker I listened to and tested was built to the appropriate standards, so along with submitting a kit for review Simon also kindly supplied a finished pair of monitors. This arrangement also meant I had to build just one monitor, and while generally it’s the case that building one LBM makes building a second one more straightforward, pairs of LBMs are configured as an active primary (in which

all the electronics and amplification are installed) and a passive secondary.

These are connected by a 2m cable that terminates in a locking multiway connector.

And rather than clutter the guts of the review with stories of speaker building, I’ve covered that in a separate box.

Little Boxes An LBM is 208 x 123 x 137mm (HWD), which puts its internal volume somewhere around 2 litres. By way of comparison, a Neumann KH120 MkII nearfield monitor (which I’d consider compact) has an internal volume of around 6 litres. The LBM cabinet is predominantly constructed of 12mm thick, CNC‑machined birch plywood, and it incorporates some steel bracing bars that both tie the two side panels together and provide some extra rigidity in the bass/mid driver attachment. The finished cabinets feel very solidly put together. The pre-made version of the LBM is finished in a black textured metallic paint with exposed plywood edges on its front panel and bass/mid driver rebate. It looks pretty smart, in a slightly homespun sort of way.

The drivers employed in the LBM are a nominally 75mm (3‑inch) diameter bass/mid unit and a nominally 25mm waveguide‑loaded tweeter. The bass/mid driver is an OEM component from US company Dayton, manufactured in China.

It’s a genuinely interesting driver featuring

a neodymium‑iron‑boron magnet motor system with, very unusually, an underhung voice coil and copper shorting ring.

The driver diaphragm is aluminium.

For its compact dimensions the driver offers an impressive ±4.6mm linear diaphragm excursion.

The tweeter is one of the “unusual and interesting” elements I mentioned earlier.

This is because, while the world of OEM drivers isn’t remotely short of cost‑effective high‑frequency driver solutions, Simon chose not to go down that route. Instead, for the LBM, he engineered a composite tweeter using an OEM full‑range inverted‑dome driver of the kind that might be found in a computer or ultra-compact desktop speaker, and combined it with a custom waveguide and ‘lens’. The reason for doing so was primarily to create a driver able to operate with a high-pass crossover filter around an octave lower than the 3kHz typically necessary with a conventional low‑cost tweeter. To operate down to 1.5kHz, as the LBM tweeter does, both a very low fundamental resonance frequency and extended diaphragm excursion are necessary, and those characteristics simply aren’t available from any compact and inexpensive OEM tweeter.

Simon’s need for the LBM tweeter to have extended bandwidth and a low crossover frequency arises from his desire for the system to offer well‑controlled

Whether bought pre-made or as a kit, the LBM comes in pairs: the ‘primary’ speaker houses all the electronics, and feeds the secondary speaker via a locking connector.

AudioSmile Little British Monitor £419/£635 pros • Compact monitoring in kit form.

• Good fundamental performance.

• Great dispersion characteristics.

• Well engineered reflex port.

cons • Modest volume levels only.

• Restricted low-frequency bandwidth.

• Kit needs time and patience.

summary The Little British Monitor is a unique and quirky product that combines highly capable basic performance with a hugely satisfying option of self-build.

vertical dispersion through the crossover region. At higher crossover frequencies, off-axis path length differences from the two drivers to the listener (or indeed a measuring microphone) constitute a greater proportion of the radiated wavelength, so phase errors will increase, resulting in destructive interference between the drivers and a ‘hole’ in the frequency response. A lower crossover frequency will reduce the effect of path‑length differences because the radiated wavelength is longer. When I get to presenting some acoustic measurements I’ll take a look at the LBM’s vertical dispersion.

Pass The Port A second interesting design element of the LBM is its reflex port. With such tiny dimensions and so little bass driver diaphragm area available, reflex loading was pretty much the only viable option, and Simon has clearly worked hard to optimise the reflex port and overcome some of the potential pitfalls of the technique. The LBM port vents from a slot in the top panel, and is created by two full-width panels spaced generally around a centimetre apart that extend down into the cabinet. The two panels are, however, gently curved over their length, and more tightly curved at either end to create generous port entrance and exit flaring. The combination of curving and flaring not only helps ensure airflow remains laminar to higher volume levels but also helps suppress the likelihood of port organ-pipe resonances. The port panels are engineered from laminated materials that offer a degree of constrained layer damping to suppress resonance still further. There is an interesting technical paper describing the design processes and principles of the LBM port on the AudioSmile website.

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AUDIOSMILE LIT TLE BRITISH MONITOR

Diagram 1: The on-axis response of the LBM, measured at a distance of 2m (playing at 75dB SPL at 1m).

The green and blue traces show second and third-harmonic distortion, respectively.

Diagram 2: Comparing the on-axis response (red trace) with measurements taken ±20 degrees (blue traces) and ±40 degrees (green traces) vertically off-axis.

Somewhat unexpectedly, the amplification topology of the LBM is traditional Class A/B rather than contemporary Class D. The reason for this is that Simon believes Class A/B generally offers a subjectively kinder distortion profile.

Similarly surprising is its generous amplifier power rating of 50 Watts for each driver, so 200 Watts in total. It’s no surprise, then, that the primary monitor has a not insubstantial aluminium heatsink attached to its rear panel. There’s also a power supply input (the LBM has an external PSU), unbalanced analogue 3.5mm and RCA phono inputs (the LBM has no balanced inputs), a mono

RCA phono subwoofer output, a four-way locking connector for connection of the secondary speaker, a rotary volume control knob, a rotary boundary compensation EQ knob and an input selection knob. The input selection switch includes a Bluetooth option, and it can play audio wirelessly from Bluetooth-enabled transmitters. It is compatible with the latest Bluetooth aptX HD standard.

Measuring Up I took the pre-built LBM to my usual measurement venue (a large church space) and performed a few basic acoustic

Build It Yourself Building a pair of LBMs actually turned out to be somewhat trickier than I initially imagined, but with some planning and patience, it is not beyond the capabilities of the majority of those people who may be interested. The only potentially new skill required is soldering, so if you’ve never done that before, a bit of research and practice is perhaps advisable. Those new to soldering might also need to budget for a soldering iron.

While I’m on the subject of buying things, the LBM kit doesn’t include the adhesives and sealant required, so be prepared for a trip to your

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local DIY store. And also be prepared then to factor in a not insignificant sum at the checkout.

I spent over £40 on adhesives and sealant.

The construction documentation provided

by AudioSmile is a 50‑page A4 PDF that, for the most part, works well. It majors on explanatory photos rather than text, and leaves little doubt over what’s intended at each stage. At this point I’ll admit that it wasn’t actually me who did the LBM assembly. My wife, sound designer Helen Skiera (www.helenskiera.com), volunteered, and I was very happy for her to take the glory (and get covered in glue).

measurements. Diagram 1 shows the axial frequency response measured at 2m at a level (at 1m) of 75dB SPL (it’s a very small speaker, remember, so it’d be unfair to drive it too hard). The overall frequency response is impressively flat, but the limited low‑frequency bandwidth that’s pretty much unavoidable with just a 75mm bass driver and a 2L enclosure volume is obviously apparent. Taking into account the sub-100Hz approximation that is unavoidable with my measuring technique, I’d put the -3dB point at around 70Hz, which ties in well with the specified -6dB at 58Hz. Also shown on Diagram 1 are the second and third-harmonic distortion. The distortion levels aren’t quite as low as might be hoped for a truly high‑performance monitor, especially in the upper bass/ low midrange (100Hz to 300Hz), but the situation is better than it looks because the predominant distortion products in that frequency range are second-order, so relatively benign. There is no escaping the fact that small driver diaphragms, like that fitted to the LBM, have to move significant distances to generate volume at bass and low‑midrange frequencies, and in the absence of extremely sophisticated and expensive driver engineering, some distortion is unavoidable.

Diagram 2 investigates the vertical dispersion to see if Simon’s efforts to minimise vertical off-axis response flaws has paid off. And they very definitely have. The diagram shows frequency response data comprising the axial response overlaid with those measured at ±20 and ±40 degrees vertically off axis. The LBM exhibits just about the tightest such group of response curves I’ve ever seen from a speaker with separate, spaced drivers. There’s little sign of the usual sharp suck-out in the frequency response either side of the crossover.

There’s two things going on here: firstly the LBM’s composite tweeter and the low crossover frequency it enables definitely make a difference, but secondly, the small diameter of the bass/midrange driver means that physical space between it and the tweeter is minimal. This also has the effect of reducing path length differences away from the perpendicular axis from the two drivers to the measuring microphone.

The result of such well‑controlled vertical dispersion ought to be a monitor that’s relatively insensitive, in subjective balance terms, to vertical listening position. It’s potentially a very useful characteristic for a monitor that’s likely to be located on a desk at a position below head height.

Diagram 3 illustrates a measurement made with the microphone positioned close to the exit of the LBM port. It provides an approximate indication of a 70Hz port tuning frequency but more significantly, in terms of Simon’s aims with the LBM port design, it shows that he’s been very successful in terms of minimising port organ-pipe resonance. The port output above its 70Hz peak is very clean down into the noise floor (the noise being predominantly ‘crosstalk’ from the bass/midrange driver). Bearing in mind the number of ‘big name’ monitors I’ve found to have a port organ-pipe issue, this is genuinely impressive.

Finally, Diagram 4 illustrates the effect of the LBM Boundary Compensation EQ. The Boundary Compensation EQ knob is a variable control and I measured it at its maximum and minimum positions plus a couple of locations in between.

However, the maximum attenuation position of the Boundary Compensation knob actually engages a 100Hz high-pass filter (with mid‑level boundary compensation) that enables the LBM to be used with a subwoofer. The Boundary Compensation EQ generally is a sensibly configured and useful facility — again, especially considering that the monitor is likely to be located on a desk and close to room boundaries.

All The Small Things So, the LBM appears to be a very competently designed and engineered little monitor. But how does it sound? I set up the pair of pre-built LBMs both on a secondary desk in our studio room and on our DAW workstation monitor shelf, and in both locations it was pretty obvious from the off that the measurements don’t lie. The LBM doesn’t just measure like a well‑engineered monitor, it sounds like one too. Its fundamental tonal balance is reasonably neutral, if, to my ears, a little restrained in the upper midband, and the LBM demonstrates no major flaws in terms of significant coloration. Having identified the slight midband reticence, I found that along with compensating for room position, the EQ was very useful for tweaking the general

ALTERNATIVES In terms of self‑build active monitors, the LBM has the market pretty much to itself.

However, for roughly the same price as the LBM kit, monitors such as the ADAM Audio D3V, IK Multimedia iLoud Micro, Kali Audio LP-UNF and PreSonus Eris E5 come into play.

Diagram 3: A close-mic measurement of the port output, showing the 70Hz tuning frequency... and not much else!

Diagram 4: The Boundary EQ set to minimum (red trace), maximum (green) and in the middle (orange).

The blue trace shows the effect of the 100Hz high-pass filter.

tonal balance more to my liking. A dB or two of EQ did wonders for the overall balance.

LBM bass is capable and usable as far as it goes, but unfortunately the pesky laws of physics mean it doesn’t go very far in terms of either bandwidth or volume level.

The stereo imaging quality is strong, with good focus to central images, and the LBM’s well‑managed vertical dispersion is clearly audible, in that its tonal balance changes little with listening head height.

I suspect the well‑integrated drivers also help with imaging quality. I also really admired the character of the LBM’s unique composite tweeter — it reveals good levels of high‑frequency detail but does so with a satisfyingly natural and unforced style.

However, no monitor as small as the LBM can disguise the limitations that come with such compact dimensions, and, to my ears, the genuinely good performance of the LBM is only really available at relatively modest volume levels — the kind of levels suitable for restrained solo monitoring at very nearfield listening distances: at a temporary desk, or in a hotel room or Airbnb, that kind of thing. Push the LBM a little vigorously and the performance begins to degrade noticeably as the bass/mid driver diaphragm is required to work harder. In particular, the bass character takes on a degree of

dynamic softness and pitch uncertainty, and the lower midrange becomes thicker and less revealing of the source quality. You can still dance and sing along with the LBMs playing loud, but you probably wouldn’t want to make important mix decisions.

But then, the LBM is far from alone in this respect. Pretty much all really compact monitors out there do their best work at modest volume levels. Keep them working in their comfort zone and they’re perfectly viable, portable professional monitors.

As discussed in the box, building the LBM is not an insignificant undertaking.

There’s a surprisingly large number of components involved, and assembling them into a pair of working monitors isn’t entirely straightforward. It’s occasionally a little fiddly and you need patience, a good uncluttered workspace and some decent tools. But if you do choose to go the LBM kit route, you’ll end up with a pair of genuinely capable compact monitors along with the satisfaction of knowing you built them with your own hands. £ Pre-made £634.80, kit version £418.80.

Prices are per pair, including VAT.

T AudioSmile +44 (0)7952 478 193 E simonashtonuk@gmail.com W www.audiosmile.com

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85

TALKBACK

Teether & Kuya Neil WILLIAM STOKES

A

lready movers and shakers in Melbourne/Naarm’s musical underground, producer Kuya Neil and rapper Teether began collaborating in late 2020. Describing themselves as “raised by the Internet” and their sound as “future-focused rap”, the duo draw on a combination of experimental sound art and various modern club styles, not to mention the influence of African electronic subgenre gqom. Out on May 2nd, new album YEARN IV pushes the envelope even further, also incorporating elements of thrash metal and lo-fi.

At the moment I can’t stop listening to Teether: I’ve been in the time-chamber, making a lot of music. So I’ve predominantly been walking around South London with demos on loop as new ideas slowly come to mind. To balance that out, it’s been quite a varied soundtrack. To name a few: Susumu Yokota, Actress, Space Afrika and Erik Wøllo’s records have been on a lot...

Timewave Zero by Blood Incantation has remained in my rotation since it came out.

The new releases by Tony Bontana, Pretty V and Oklou, Penance by Sidney Phillip, REVELATOR by E L U C I D... 40 by Jawnino may have been my 2024 album of the year.

We’ve been bumping Alice In Chains, Voivod and Metallica in the house heaps too.

Kuya Neil: Treasure by Cocteau Twins.

I love the guitar tones, colour and production on this record. The ’80s seemed like an exciting time — solid‑state amps, early digital recording, chorus on everything! ‘Persephone’ is my favourite one on Treasure. The vocal melodies, the guitar tone and that snare sound! I am definitely taking cues aesthetically from this band with my guitar sound for Teether & Kuya Neil live shows. Other than that, I obsessively listen to music I’ve made before I release it, as I know once it drops I’ll never want to listen to it again.

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The artist I’d most like to collaborate with Teether: Sade! I’m not sure if it’s immediately apparent, but she’s inspired a lot of artistic choices I’ve made, whether it be melodically, tonally or visually. I think we could make a banger track.

Kuya Neil: Maybe Steve Vai, but I would only want to do it if it came about naturally and made sense. There are so many artists whose music I love, but whom I don’t have any interest in collaborating with. I am happy to remain a fan. I’m also very content collaborating with the people I know, who are some of my closest friends and favourite artists. (Cue the “awwww” sound effect!)

The first thing I look for in a studio Teether: Whatever is in there, we will utilise.

I’m a fan of working with limitations as we have for so many years. I think being forced to make a decision within certain parameters can really make you rack your brain and think outside the box. That being said, I love having gear at our disposal and being able to really indulge in exploring sounds. I always get real hype about a new synth or guitar or vocal effect that I’ve never used before.

Kuya Neil: The WiFi password. But in terms of gear, guitar multi-effects pedals! I love those things so much, particularly the old ones. I want something I can pick up and dial in a sound on. The early digital stuff has so much character, and takes me back to my formative years making music as a teenager. I loved the Korg ToneWWorks AX1000G and I still use the Boss ME-50 to this day! So many great sounds, and the ‘resonance’ effect on the pedal is so good.

The new digital modelling technology has come such a long way, but it’s become so good that it’s almost boring. I want it thin, digital and trashy.

The person I would consider my mentor Teether: I have to give credit to Mat Creedon here. This man really started this whole thing for me. I was always

a self-taught guitarist and bassist until the end of high school, when I was encouraged to take lessons for exams and that. My initial plan was to study jazz bass at university and become the next John Myung, but my attention span couldn’t handle music theory like that. I took lessons with Mat for a couple of years in his home studio and he recognised my interest in the studio side of things, so he helped me record my first band’s demo and taught me the basics of recording and mixing. He told me what I’d need to have a basic setup at home and it snowballed from there. He really opened up a significant door for me. I need to reach out and thank him.

Kuya Neil: I don’t have one at the moment.

There are definitely artists I look up to, but I feel too shy to ask them to mentor me, and it hasn’t always worked out in the past when I have been in a dynamic like that.

I am pretty sensitive to feedback so I think I would struggle with proper mentorship, to be honest!

My go-to reference track or album Teether: I’ll be real: I hate to reference. I get so stubborn on a new project sounding only like itself that I refuse to A/B compare my stuff while working on mixes. Obviously, some things have sounded wild as a result in the past but I trust my ears and fate enough at this point.

Seems to work itself out, you know!

Kuya Neil: Not a specific album, but anything by Kush Jones or DJ SWISHA, especially when making club or footwork-y stuff. I think they hit a sweet spot of their productions feeling raw, but never sacrificing clarity. They can make everything hit loudly and cleanly with colour, dynamics and plenty of groove. Damn, did I really just say groove unironically?

My secret weapon in the studio is Teether: Waves GTR3, and Ableton saturator on everything! And obsessively applying subtractive EQ.

Photo: Phillip Dixon

I think in an interview he said he learnt how to make beats on the PS1 game Music 2000!

Seeing him sequence everything in that game would have been amazing to witness.

The producer I’d most like to work with Teether: Actress! I’m such a fan for real.

I actually first heard of Actress through my friend and producer Deanjames, when I sent him some instrumentals I was working on and he said it reminded him of Actress’ production. This guy is definitely operating on a wavelength I can understand.

Kuya Neil: DJ Love! I’m so inspired by Budots and the homegrown Filipino dance music coming from the Visayas. I think the roots of this music and the spirit behind it are important things to cultivate and carry over into how we make music here. I also need to learn how to use FL Studio properly.

I think a lot of the sauce comes from that sequencer and its built-in soft-clipper.

Kuya Neil: MAIM. It’s a free VST plug-in that emulates old MP3 compression, and has some really interesting algorithms for sound design and bit-rate reduction. Outside of gear, the secret is to take the time talking to people, making them feel comfortable and finding ways to relate to each other.

Being curious and asking questions about their lives, sharing music they’re listening to or having a YouTube sharing session is standard for me before any session.

The studio session I wish I’d witnessed Teether: Barter 6 era Young Thug would have been wild to see. Some of the things he does on that record blow my mind, still.

Witnessing those ideas pop up organically would have been amazing. It’s such a cohesive record, even though you can tell so much of it is pure instinct and gut feelings.

Kuya Neil: The sessions for Dizzee Rascal’s Boy In Da Corner. I can’t believe how young he was when that classic record was made.

Kuya Neil: “The new digital modelling technology has come such a long way, but it’s become so good that it’s almost boring. I want it thin, digital and trashy.”

The studio experience that taught me the most Teether: We’ve been in all sorts of environments over the years, I couldn’t pin down one in particular. I think working in larger studios has shown me I prefer to work in a smaller, more intimate space. All the gear in the world is cool, but what we really need is some nice lamps and a rug and a little window you can blow smoke out of and/or ponder.

Kuya Neil: Recording [acclaimed Kiwi artist] Coco Solid in 2017 at an Airbnb home studio in Melbourne, for her mixtape Cokes. This was my first experience producing and recording a rapper. I was younger, super nervous and on the first day I was overthinking the technical aspects a lot. I remember Coco gave me some really good advice to focus on enjoying the process, and that in the end, the listener doesn’t care about many of these technical things that young engineers get hung up on. This was hugely important advice at the time and set the tone for how I work in the studio to this day.

The advice I’d give myself of 10 years ago Teether: You may have many of the same problems in 10 years, but overall life is really worth it and your music will just get better and better, exponentially. Never stop creating, never stop being a fan of music, and never let your ego get in the way of making a good song. Trust your gut and your ears!

Kuya Neil: Trust yourself, and trust your own taste.

www.soundonsound.com / May 2025

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Apple M4 & M3 Ultra Macs Desktop & Laptop Computers

Apple’s M-series Macs continue to power ahead.

We benchmark the latest Mac mini, MacBook Pro and Mac Studio models for audio performance.

Apple M4 & M3 Ultra Macs

MARK WHERRY

From £599

ith one exception, Apple have upgraded or reintroduced new models of every single Macintosh in the last few months. Last October, the company unveiled updates to their iMac and MacBook Pro families — in addition to a brand-new Mac mini — with March bringing an updated MacBook Air and Mac Studio, which takes the crown as “the most powerful Mac ever”. Most of these are based around their latest M4 processor. The M4 debuted in the iPad Pro back in May 2024, but it was only a matter of time before Apple brought this latest silicon to the Mac, and, in doing so, continued the established pattern of augmenting the standard chip configuration with more powerful Pro and Max variants.

pros • The new Mac minis are much smaller and lighter than their predecessors, offer better performance, and are more affordable.

• Thunderbolt 5 on many models for lightning-fast external storage.

• The MacBook Pro offers a potent combination of power and efficiency plus the excellent nano-texture screen.

• New Mac Studio models offer huge performance gains, especially with the new M3 Ultra chip.

cons • As usual, the cost rises quickly with specification level.

• The base 256GB M4 Mac mini has slower storage than the other models.

summary The base M4 Mac mini models are capable computers representing great value, while the M4 Pro models offer plenty of power for music and audio.

The new Mac Studio and MacBook Pro can be costly, but the former is the most powerful Mac ever, and by a large margin — and the latter is unquestionably the best laptop available for musicians and producers.

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W

What’s It All M4?

The M4 represents a significant leap over the M1 and M2 architectures already familiar to musicians and audio engineers, building on the M3 chips released a year earlier. The M3 family were the first personal computer chips to be fabricated on a 3nm process node using TSMC’s N3 technology. This was

A new M in charge: will the M3 Ultra achieve pro audio domination?

significant because, simply put, moving to a smaller process allows for tinier transistors and greater transistor density, which can afford improvements in both performance and power efficiency. Prior M-series chips were fabricated on a 5nm process, so even if Apple had just manufactured the existing M2 chip with the newer N3 technology, it would theoretically have resulted in a more energy-efficient chip with better performance.

However, the M3 was more than just the sum of its process. Where the M2 had been an evolution and refinement of the M1, offering around a 20 percent increase in real-time audio performance, the M3 represented a much bigger leap. Thanks to significant advances in both manufacturing and architecture, the M3 Max delivered 58 to 75 percent more audio processing compared to the M2 Max, depending on the host application. So key were the M3’s gains that the M3 Ultra can outperform an M4 Max — much as an M1 Ultra could surpass an M2 Max. Consequently, whereas the base model in the new Mac Studio line features an M4 Max CPU, the top-end model is based around the new M3 Ultra.

Core Performance As you may already be aware, M-series chips employ a heterogeneous architecture for general-purpose computing tasks. This means that while every CPU core supports

the same basic instruction set capable of executing the same code, there are two types of cores, with different performance and efficiency characteristics. Performance (P) CPU cores are finely tuned for high performance, whereas Efficiency (E) cores are optimised for handling less demanding tasks using the least amount of energy.

Music applications typically rely on P cores for real-time audio, leaving the E cores to handle other, non-real-time tasks such as lookahead processing.

However, applications can’t directly assign specific tasks to certain cores.

Instead, macOS 11 introduced the Audio Workgroups API, enabling applications or plug-ins to flag certain tasks as being responsible for handling real-time audio. This, in turn, allows macOS to optimise the usage of performance and efficiency cores to achieve glitch-free playback. And although such a scheduling system isn’t perfect, it generally provides the best possible overall performance. Because whilst one application or plug-in can’t know the specific processing requirements of another, macOS knows everything — at least in terms of an application’s needs.

The M3 was a notable advancement over the M2, introducing a next-generation GPU architecture alongside many key advancements to the design of the CPU cores, such as improved branch prediction. Simply put, branches are points in a program where a condition needs to be evaluated to figure out what happens next. A naïve example might be to check if a channel is muted on a mixer: if the channel is muted, the program can skip over that channel and proceed to the next one, but if the channel is not muted, the program will need to process the playback required from that channel.

To speed up the execution of our modest mixer, a branch predictor would make an educated guess as to whether a given channel is muted or not based on its previous state. If the branch predictor

guesses correctly, the CPU will have already started processing the correct instructions ahead of time, allowing the program to run smoothly. However, if the prediction is wrong, the CPU must discard the instructions that were speculatively processed and reprocess the correct ones, which can adversely affect performance.

In practice, branch prediction works at a much lower level, enhancing DSP algorithms where decision making is

instructions to be processed simultaneously, while a wider execution engine enables more of those instructions to be executed in parallel. Such improvements benefit both traditional DSP algorithms — especially computationally intensive effects like convolution reverb — and modern machine‑learning-based audio effects for restoration and assisted mixing.

Apple claimed the M3’s performance cores were up to 30 and 15 percent faster compared to the M1 and M2 respectively, with the efficiency cores achieving a greater increase of up to 50 and 30 percent. Perhaps most impressively, though, and thanks in no small part to the 3nm process, the M3’s architecture could deliver the same multi-threaded CPU performance as the M1 using half the power.

“As we will see, the real-world difference in application performance between the M4 Max and M3 Ultra is, frankly, chasm-like!” inherently part of the process. For example, a limiter needs to decide whether incoming audio samples are above or below the threshold. If the current sample is above the threshold, it’s very likely this condition will be true for the next sample. Therefore, the CPU can predict the outcome for the next cycle.

The P cores also feature wider decode and execution engines, both of which enhance audio processing performance.

A wider decode engine allows more

From London To South Wales The M4 builds on these architectural advancements, with the performance core offering further improvements in branch prediction, the ability to decode 10 instructions simultaneously, a 40 percent larger reorder buffer, and what Apple refer to as “Next Generation ML [Machine

New (top) and old (centre) Mac minis atop a Mac Studio.

The latest version really does justice to the ‘mini’ name!

Test Spec Apple Mac mini with M4 (four Performance cores and six Efficiency cores), 16GB memory and 512GB storage.

Apple Mac mini with M4 Pro (10 P cores, four E cores), 48GB memory and 1TB storage.

Apple MacBook Pro with M4 Max (12 P cores, four E cores), 40-core GPU, 128GB memory and 4TB storage.

Apple Mac Studio with M3 Ultra (24 P cores, eight E cores), 256GB memory and 4TB storage.

All systems running macOS 15.3.1 (Sequoia).

www.soundonsound.com / May 2025

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A P P L E M 4 & M 3 U LT R A M A C S

Figure 1: This chart shows the results of Geekbench 6’s single‑ and multi-core CPU tests for the Macs featured in this review, along with results for older systems and the M4-based iPad Pro. (The number of CPU cores are indicated in brackets for systems available in different configurations.)

Learning] Accelerators” (see box). The larger reorder buffer helps to improve general compute performance, but has fewer direct benefits when running DSP code, for reasons that are a little beyond the scope of this article. However, it can still improve the performance of audio applications and plug-ins, for example, by speeding up the execution of user interface handling and so leaving more resources available for executing DSP code.

Differ-M-tiation The Pro and Max variants of the M1 and M2 families shared many similarities, particularly in terms of CPU core configurations. While the standard chips each featured four performance cores and four efficiency cores, the Pro and Max versions were equipped with eight P cores, with a more affordable Pro option offering six P cores. The M1 Pro and Max chips had two E cores, while the M2 Pro and Max chips offered four.

For the ultimate performance, Apple then introduced an Ultra variant. The M1 Ultra debuted with the Mac Studio, while the M2 Ultra arrived alongside the first Mac Pro based on Apple Silicon. However, unlike other M-series chips, the Ultra combines two Max dies interconnected via an Apple-designed high-speed fabric known as UltraFusion. This technology allows the system to treat the package as a single unified chip with double the resources. As a result, the Ultra delivers not only twice the number of CPU and GPU cores, but also dual Neural and Media Engines.

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The Pro and Max variants of the M1 and M2-series chips were both available in identical 10‑ or 12-core configurations, and consequently, systems containing these chips achieved similar levels of performance. However, the Pro and Max variants were still distinct in other ways such as the amounts of physical memory supported and the number of GPU cores and video encode and decode engines.

With the release of the M3 family, Apple kept the same eight-core composition for the standard M3 chip, but decided to more significantly differentiate the number and distribution of cores in the Pro and Max configurations. The M3 Pro was available with either 11 or 12 CPU cores, comprising six E cores and either five or six P cores, whereas the M3 Max became further maximised by offering 14 or 16 CPU cores, featuring four E cores with either 10 or 12 P cores. The belated M3 Ultra combines two M3 Max dies, but unlike previous generations, is available in two variations, depending on which variant of the M3 Max it uses. Both offer eight E cores, and you can choose either 20 or 24 P cores.

With the M4 series, Apple have once again decided to rebalance the CPU core configurations. The standard M4 in the Mac is a 10-core chip with six P cores and four E cores. The M4 Pro is available in 12‑ or 14-core versions featuring four E cores and either eight or 10 P cores, while the M4 Max configurations have also been upgraded to comprise 12 or 14 cores, each featuring the same four E cores, adding either eight or 10 P cores. Another significant change with the

standard M4 chip is the inclusion of 16GB unified memory as standard — arguably a long-overdue decision. Previously, 8GB had been the minimum for base models.

Figure 1 shows that the Geekbench results for Apple’s new chips are mostly as one would expect, aside from one noticeable anomaly: the multi-core score for the new M3 Ultra chip. According to this result, the Mac Studio with M3 Ultra appears to be less than six percent more powerful than the M4 Max-based model — an increase that would hardly justify the price gap, even considering other differences in the specifications. However, I think this result is indicative of a flaw in Geekbench’s benchmarking rather than being reflective of the M3 Ultra’s raw horsepower. As we will see, the real-world difference in application performance between the M4 Max and M3 Ultra is, frankly, chasm-like!

The Mini Studio The design of the new Mac mini with M4 clearly owes much to the Mac Studio; indeed, a more apropos name might have been the Mac Studio mini. Apple have shrunk the 7.7 x 3.7-inch dimensions of the Mac Studio to a 5 x 2-inch cuboid.

An honourable mention must be given to the team that engineers Apple’s power supplies. It’s remarkable when you consider

Figure 2: Blackmagic’s Disk Speed Test offers a glimpse at the read and write speeds of which the internal storage supplied with various Apple Silicon Macs are capable.

that, in addition to housing such a powerful system within a compact aluminium enclosure, Apple also manage to integrate a 155W power supply within the Mac mini’s tiny form factor. The Mac mini by itself could easily be mistaken for the power brick of a competing product!

Considering its size, the new Mac mini offers a surprising array of connectivity.

The 3.5mm headphone jack has finally been moved to the front of the device, accompanied by two USB-C ports with support for USB 3 (up to 10Gb/s). Sadly, however, this means there is no space for any USB-A ports.

The power button, previously located at the rear of the Mac mini, has also been relocated. However, rather than moving this seemingly vital control to, say, the front, or maybe the side, Apple’s engineers have instead decided to place it underneath. This makes it hard for the button to be pressed accidentally, but, for the same reason, it also makes it hard for the button to be pressed intentionally. In practice, this is only really inconvenient during Touch ID setup, where a physical button press is required to confirm secure intent and establish a direct connection to the Secure Enclave.

A more important consideration is that the base model Mac mini with M4 and a 256GB SSD delivers lower data transfer rates when compared with the more expensive, higher‑capacity alternatives (as you can see from the Blackmagic Disk Speed Test results). These speeds don’t make the machine unusable for recording and playing back audio tracks

Despite its size, the Mac mini is surprisingly capable!

or running sample-based instruments, but it’s something to bear in mind. The Mac mini with M4 is also available with 512GB storage — which is a more useful minimum — and you can configure a model with a 1TB or 2TB SSD if required. In terms of unified memory, the standard Mac mini starts with 16GB and can also be ordered with either 24 or 32 GB.

To Pro Or Not To Pro? Like its M2-based predecessor, the new Mac mini is again available in two models featuring either the M4 chip or the new M4 Pro variant. However, unlike its predecessors, these two models are externally indistinguishable from each other,

since they both have the same number of connectors at the rear of the device. Aside from the figure-8 power socket, there are three Thunderbolt (USB-C) ports, HDMI and Gigabit Ethernet, which is configurable to 10Gb Ethernet.

This means the entry-level Mac mini now offers an additional Thunderbolt port compared with the previous M2-equipped model, whereas the M4 Pro model has one fewer than its M2 Pro-based predecessor. However, while the Mac mini with M4 supports Thunderbolt 4 and USB4 with a maximum speed of 40Gb/s, the M4 Pro model features a newer Thunderbolt 5 controller with USB4 v2.0 offering three times the maximum

Thunderbolt 5 The adoption of Thunderbolt 5 for Macs powered by either the M4 Pro, M4 Max or M3 Ultra chips is going to be a big deal for musicians, particularly those who work with large sample-based instruments. The internal storage on all but the most basic Apple Silicon Macs has always been capable of fast read and write speeds, but until now, only the Mac Pro has afforded the option to add extra storage capable of similar performance.

For example, using a 1TB WD Black SN850X M.2 SSD attached to a Mac Pro via OWC’s ‘Accelsior 1M2 M.2 SSD to PCIe 4.0’ card, Blackmagic’s Disk Speed Test reported read and write speeds of 5702 and 5603 MB/s respectively. Which isn’t too shabby for an SSD that’s quoted as having sequential read and write rates of 7300 and 6300 MB/s. However, when connecting this same SSD externally via Thunderbolt 4 with the fastest enclosure I could find that supports 40Gb/s — OWC’s Express 1M2, which connects using USB4 — I achieved read and write speeds of 3454 and 3393 MB/s.

Although respectable, and more than fast enough if you’re primarily working with audio tracks, this speed is still about 40 percent less than the performance offered by connecting the same SSD to an internal PCIe 4.0 slot. And this translates into noticeably longer load times when working with large sample-based instruments located on external storage. Changing patches is less snappy, and opening a project from scratch can become the moment you decide it’s time to put the kettle on.

Thunderbolt 5 will change the game for those wishing to attach fast external storage.

Unlike Thunderbolt 4, which implemented PCIe 3.0, required full compatibility with USB4, and offered transfer speeds of 40Gb/s, Thunderbolt 5 supports both PCIe 4.0 and 5.0, including USB4 2.0, and achieves transfer speeds up to 120Gb/s — with one slight caveat. This maximum speed can only be achieved unidirectionally with what’s referred to as ‘Bandwidth Booth’ mode, meaning

that either reading or writing data is prioritised, reducing the amount of bandwidth available in the other direction to less than 40Gb/s.

This is great if you’re working with large sample libraries, since you’d want to prioritise reading when working with an instrument and then writing only when downloading or copying sample data to that drive. In normal usage, though, where read and write bandwidth is more or less equal, such as when recording and playing back audio tracks simultaneously, Thunderbolt 5’s default mode enables bidirectional transfers of 80Gb/s in each direction. Therefore, when used with a suitable M.2 NVMe SSD with support for PCIe 4.0 or 5.0, real‑world bandwidth could be in the range of 8000 to 15,000 MB/s, depending on the mode being used. Unfortunately, though, although many Thunderbolt 5-capable docks and storage peripherals have been announced, none seem to be available to buy at the time of writing.

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bandwidth at up to 120Gb/s for compatible devices (see box).

The Mac mini with M4 Pro includes 24GB unified memory as standard and is configurable with 48GB or even 64GB if needed — the previous model with M2 Pro could accommodate a maximum of 32GB. A 512GB SSD is also standard, with 1, 2, 4 and 8 TB options also available.

The Ultra Studio The first thing you’ll notice about the new Mac Studio is, well, nothing, since it’s outwardly identical to the previous models. So, whilst it’s understandable that the new Mac mini features USB-C connectors exclusively, I’m still delighted that the Mac Studio retains its pair of USB-A ports at the rear, supporting USB 3.1 (up to 5Gb/s).

In addition, both the M4 Max and M3 Ultra-based Mac Studio models feature the same four USB-C connectors at the back of the enclosure, supporting Thunderbolt and USB4. And, just like their predecessors, there’s an additional pair of USB-C ports at the front offering USB 3.1 (up to 10Gb/s) or Thunderbolt on the models with M4 Max and M3 Ultra respectively.

As with the M4 Pro-based Mac mini, the new Max and Ultra Mac Studio models both support Thunderbolt 5 and USB4 2.0, which highlights an interesting point.

When the M3 series was announced at the end of 2023, the M3 Max — upon which the M3 Ultra is based — implemented a Thunderbolt 4 controller and didn’t support version 5 (because it wasn’t yet released). In implementing the M3 Ultra, Apple’s engineers were able to retroactively add support for Thunderbolt 5, once again underscoring the advantage of controlling your own silicon destiny.

new M4-series MacBook Pro models from the previous generation, except that Space Black has now vanquished Space Gray altogether, and every model of MacBook Pro is now available in either the foreboding Space Black or the traditional Silver finishes. Space Black employs a “breakthrough chemistry” intended to reduce the fingerprint marks that so often besmirch the appearance of such a dark finish, and, in practice, works rather well.

A particularly welcome addition to the latest MacBook Pro range is the option to add a nano-texture coating to the display to minimise reflectivity.

Unlike traditional matte finishes, which diffuse light through internal scattering, Apple’s nano-textured glass diffuses light at the surface courtesy of an etching process that creates nanostructures directly on the glass. This reduces perceived glare whilst preserving image sharpness, colour vividness and contrast. Nano-texture displays work surprisingly well in studio control rooms, which are often dimly lit and windowless, and sometimes have bright spotlights above the console.

Internally, the MacBook Pro models offer the same improvements as the new Mac mini and updated Mac Studio models, which is no surprise since the entire system is essentially contained within an M-series package. The USB-C ports on the MacBook Pros with either M4 Pro or M4 Max offer Thunderbolt 5 connectivity. As always, the only obvious improvement would be to incorporate an Ethernet port into the power supply, as Apple do with the iMac. The MacBook Pro with M4 is available with either 16, 24 or 32 GB of unified memory. The M4 Pro-based model can be ordered with 24 or 48 GB, but not the

Figure 3: Maximum track counts achievable with Logic Pro. Note the outstanding performance of the M3 Ultra-equipped Mac Studio.

The base Mac Studio with M4 Max, which features a 14-core CPU and 32-core GPU, includes 36GB unified memory. The M4 Max-based model, with a 16-core CPU and 40-core GPU, comes with 48GB as standard, but can also be ordered with 64 or 128 GB unified memory. For those who need more, both M3-based Mac Studio models include 96GB unified memory as standard, with a 256GB option also available; and should this still not be enough, a 512GB configuration is also available if you’re purchasing the M3 Ultra featuring a 32-core CPU and 80-core GPU. In terms of storage, the M4 Max and M3 Ultra-based Mac Studio models include either a 512GB or 1TB SSD respectively, with 2, 4 or 8 TB options also available.

Additionally, the Mac Studio with M3 Ultra can be ordered with a 16TB drive.

Black ’Books As with the Mac Studio, there’s little to distinguish the external appearance of

Accelerated Learning Given that Apple’s M-series silicon has always incorporated a Neural Engine for accelerating neural‑network-based computation, you might be wondering why the CPU cores would also need to provide accelerators for machine learning. The simple reason is that while the Neural Engine excels at duties such as object detection, voice recognition, classification and image processing, it’s not equally suited to every ML task. This is because the Neural Engine is designed for efficient inference, which is to say running pre-trained models rather than training new ones from scratch. It works best with structured data and models that can leverage lower numerical precision. In fact, given these prerequisites, the Neural Engine can handle an insanely high

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throughput, measured in trillions of operations per second (TOPS), using far less energy than a CPU or GPU. When it comes to audio processing, the Neural Engine can be useful for some tasks, such as noise reduction, classification, voice isolation, or detecting so-called wake words — such as “Siri” — but it’s less useful for reverbs, delays and complex filter structures, oscillator-based synthesis, or higher‑precision 32- or 64-bit floating-point arithmetic. So, audio software might use the Neural Engine to recognise what type of drum sound a sample contains (such as kick, snare, hi-hat, and so on), but rely on the CPU’s ML accelerators or the GPU for separating the drum sounds from a mixed audio track.

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A P P L E M 4 & M 3 U LT R A M A C S

The new M4 MacBook Pro range features the options of a nano-texture screen and Space Black finish.

64GB configuration that is available with the Mac mini. Like the base Mac Studio model, the MacBook Pro with M4 Max (featuring a 14-core CPU and 32-core GPU) includes 36GB unified memory, while the M4 Max model with a 16-core CPU and 40-core GPU comes with 48GB and can also be ordered with 64GB or 128GB.

Testing M1, 2, 3, 4 To benchmark the performance of Apple’s M4 family of custom silicon and the new M3 Ultra chip when running professional music and audio applications, I returned to the methodologies used when benchmarking the M2 and M2 Pro back in the July 2023 issue (www.soundonsound.com/reviews/ apple-m2-macbook-pro-mac-mini-part-2).

However, major new versions of both macOS and my test DAWs having appeared since those tests were performed, so I decided to upgrade and retest certain Macs from the earlier generations of Apple Silicon to put the latest chips in perspective. And it’s interesting to note how the older M1 Max and M2 Pro-based systems actually perform better in almost every test with the latest macOS and application version than they did two years ago, which is definitely progress of a kind.

All applications were tested using default settings (unless specified otherwise) using the built-in audio hardware with a buffer size of 128 samples at a 44.1kHz sample rate.

Logic Pro Beginning with Apple’s own Logic Pro, real-time performance was once again evaluated with a project consisting of stereo audio tracks, each with two instances of Amp Designer — employing, with the unerring instinct of an English musician, the Britannica Crunch preset — inserted on the corresponding channels. As I’ve

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noted before, this test is loosely based on information gleaned from Apple’s website some years ago when the company last detailed Logic Pro-based comparisons on a Mac’s product page.

By default, the M4 and M4 Pro-based Mac minis could play back 170 and 468 tracks respectively (and thus 340 and 968 Amp Designer instances) at the same time, while the MacBook Pro with M4 Max managed 572 tracks (using 1144 plug-in instances) simultaneously. And although this is rather impressive, it gets better.

Logic Pro allows the number of active processing threads to be adjusted in the Audio page of the Settings window, making it possible to configure the number of cores used for audio processing. Apple recommend leaving the number of threads set to Automatic, which in most cases means only the P cores are deployed for

audio processing, leaving the E cores free for handling other application and system tasks. (The track counts quoted reflect this Automatic setting.)

However, it’s possible to push your Mac even further by setting Processing Threads to the maximum number of CPU cores available. This means both the performance and efficiency clusters of cores will be used for audio processing, though it’s worth pointing out this option doesn’t always yield better performance in every situation.

Because the E cores are, by definition, less performant than the P cores, it’s also possible to achieve less performance by employing them for audio processing, depending on the routing of a project. In this context, however, setting Processing Threads to the highest option with the test project described made it possible to play back 214, 507 and 622 tracks with M4, M4 Pro and M4 Max-based Macs respectively, representing improvements of 26, 8 and 9 percent.

The reason for the large difference in the results between the M4 and M4 Pro (170 to 248, or 214 to 507) is due to the fact the M4 has four P cores where the M4 Pro on test has 10. This also explains why enabling the four E cores makes a bigger difference on the M4 than the Pro or Max variants, since switching from the Automatic setting to ‘8 Processing Threads’ represents a bigger increase in the number of cores available for audio processing.

Moving along to the Mac Studio with M3 Ultra, if you were expecting an improvement over the M4 Max to correspond the Geekbench multi-core scores, you’re in for a shock, since this system effectively broke my test project. I hadn’t anticipated reaching the 1000-audio-track limit in Logic Pro, which happened with Processing Threads set to either Automatic or the maximum available.

Figure 4: Offline bounce times in Logic Pro — shorter is better.

Figure 5: Real-time track counts in Cubase, with the M3 Ultra once again showing its power.

To compensate for this, I began adding extra pairs of Amp Designer instances to the tracks in order to synthesise a result that should have been available if more audio tracks were available. So, the results ended up being approximately 1013 and 1054 tracks with the Automatic and maximum settings, respectively.

To measure offline performance, I turned to the ‘Montero Spatial Audio’ demo project from Logic Pro 10.8, which I’d previously used when testing M2 performance. With Processing Threads set to Automatic, a surround bounce took 54, 24 and 20 seconds with M4, M4 Pro and M4 Max-based systems respectively. The Mac Studio with M3 Ultra was slightly quicker, taking 18 seconds.

With Processing Threads set to the maximum, these rendering times for the M4, Pro and Max-based systems were reduced to 43, 22 and 19 seconds respectively, while the M3 Ultra system shaves just a further second off the bounce time. Once again, the M4 receives the biggest bump, taking 20 percent less time to bounce when compared to using the Automatic setting.

the release of version 14, Steinberg have added a new Dropouts section to Audio Performance Monitor that greatly assists in the benchmarking ritual. This section features a timeline to show Real-Time and — if it’s enabled — ASIO Guard performance curves during playback, indicating any dropouts that occurred with a vertical red line at the appropriate timeline position.

In an ideal scenario, the Audio Performance Monitor would show no dropouts during playback. Therefore, in testing Cubase Pro, I set out to discover how far each system could be pushed before any dropouts were reported.

However, since occasional dropouts can occur without being audible, or frequent enough to prevent you from working, I also

Cubase Steinberg’s Cubase Pro has one of the most interesting native audio engine implementations on Apple Silicon thanks to its ASIO Guard feature, which was discussed in detail in the second part of our extensive M2 review back in the July 2023 issue. And, with

Figure 6: Offline bounce performance in Cubase doesn’t follow the pattern set in Logic.

decided to see how far playback could be pushed before it became impossible to continue. When ASIO Guard is enabled, this ‘peak’ is easy to find, since Cubase essentially acts out the DSP equivalent of throwing its hands up in exasperation.

Using a project not dissimilar to the one created for Logic, I substituted the two instances of Amp Designer with instances of Steinberg’s bundled VST Amp Rack plug-in, loaded with the Big Clean Reverb preset.

With ASIO Guard disabled, the M4, M4 Pro, M4 Max and M3 Ultra-based systems could comfortably play back 32, 99, 104 and 135 tracks simultaneously. Pushed into the red, the same machines reached track counts of 36, 100, 100 and 136 before the output became audibly unusable.

These small differences between the normal and peak track counts become more pronounced once ASIO Guard was enabled. With ASIO Guard set to Normal, the M4, Pro, Max and M3 Ultra-based Macs played back 64, 118, 149 and 222 tracks without showing any dropouts, and 74, 128, 157 and 229 tracks until the systems waved the white flag. (Further increased track counts can be achieved by setting ASIO Guard to High, but the down side is that the latency of tracks being processed ahead of time using ASIO Guard can become annoyingly noticeable, especially when mixing.)

If these track counts don’t seem astonishingly high compared with Logic Pro, that is probably down to the relative demands of the respective

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plug-ins. Ultimately, it’s the relative results between different Macs that demonstrate the real value of such tests.

For offline performance, I decided to export the project Steinberg supply to demonstrate the Iconica Sketch orchestral library, which employs 30 instances of Halion Sonic. Unlike Logic Pro, where the fastest system achieves the quickest result, the export times in Cubase don’t seem to follow a similar pattern.

One explanation for this is that Logic’s bounce command is closely tied to the real-time playback of a project, which is presumably why the settings of the audio engine affect the speed at which a project is bounced. This is obviously not the case in Cubase. When I performed the bounce on the Mac Studio M3 Ultra with Activity Monitor’s CPU History window open, it was clear that the system is being underutilised, with numerous cores barely showing up for work. More investigation is needed to fully understand what’s happening, although, to be fair, I’m quibbling about a few seconds here and there.

Pro Tools Of the three applications tested, Pro Tools is by far the simplest to benchmark. There aren’t any settings to configure the native

Figure 7: Real-time track counts in Pro Tools.

performance of the Avid Audio Engine for Apple Silicon, which is reassuringly straightforward. And you can observe from Pro Tools’ System Usage window that all 32 cores are available and being utilised, which can be confirmed by looking at Activity Monitor’s CPU Usage and History windows.

I should note that the latest version of Pro Tools (2024.10) is not officially supported on M4-series Macs at the time of writing; full support is planned for the second quarter of 2025. The latest Mac Studio isn’t listed on Avid’s page detailing Supported Apple Computers, either,

The Windows Comparison Many people will want to know how the M4-series Macs and the M3 Ultra-based Mac Studio discussed in this article fare against alternative systems from Windows-based competitors. However, this is an extremely difficult comparison to make. For one thing, Apple have set the bar so high in terms of the amount of energy required to achieve a given performance level, it’s hard for other manufacturers to come close. And this is in no small part due to Apple’s vertical integration, meaning the company’s engineers control every aspect of the system: from the power supply and silicon to the enclosure and the operating system.

One direct consequence of this is fan noise — or the lack thereof. A quiet system might be a luxury to some users, but can be very important to musicians and audio engineers who don’t necessarily have access to a machine room.

However, if one were to make comparisons, let’s first consider the Mac Studio. If imitation is indeed the sincerest form of flattery, then you don’t need to look much further than Lenovo’s ThinkCentre Neo Ultra. A system with the fastest processor option (a 14th-generation Intel Core i9-14900 processor), 32GB memory and 512GB storage comes to £2031.49, which is as close to the Mac Studio base model with M4 Max I could find. In this case, you’re getting 4GB less memory, but with six USB-A ports, no Thunderbolt 5, and one USB-C connector offering USB up to 20Gb/s.

Although Geekbench’s results aren’t always an accurate indicator of real-world performance, with single-core and multi-core scores of 2889 and 17479 respectively, this system is not only outperformed by the M4 Max, but by the Mac mini with M4 Pro as well. And for not much more, you can get such a system with an M4 Pro featuring a 14-core CPU, a 20-core GPU, faster connectivity options, 48GB memory, and 1TB storage.

HP’s Z-range of workstations are a solid choice when it comes to pre-built Windows systems ideal for music and audio, especially since Avid officially support Pro Tools on most models and Steinberg have recommended them in the past. The cost of a midrange Z2 G9 Tower Workstation with an Intel

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Core i9-14900K processor, 64GB memory and a 512GB SSD is $2525.82, at the time of writing. (HP don’t offer the same configurations in the UK.) According to the Geekbench Browser, the 14900K scores 3193 and 21540 in the single- and multi-core tests, thanks to being capable of a higher clock speed than the standard 14900, which is a level of performance to the M4 Pro. A Mac mini with M4 Pro and the same 64GB memory and 512GB SSD comes out at $2199. The HP Z2 G9 offers internal PCIe slots, but doesn’t feature Thunderbolt connectivity.

While the MacBook Pro undoubtedly carries a high-end price tag, it’s not unreasonably expensive when you consider the performance. For example, a 16-inch MacBook Pro with M4 Max fully loaded with 128GB memory and 8TB storage will set you back £7349. However, a similarly specified Mac Studio with an M3 Ultra (with 128GB and without the higher number of GPU cores) with roughly the same CPU performance is priced at £7999. When you consider the MacBook Pro is portable and includes a 3456 x 2234, nano-textured display, it doesn’t seem like such a bad deal.

If we turn our attention to a comparable Windows laptop, let’s consider the venerable Lenovo ThinkPad P1 Gen 7 mobile workstation. Featuring an Intel Core Ultra 9 185H processor with 64GB memory (since no 128GB option is available), two 4TB M.2 2280 drives (since there’s no single 8TB) option, and a 16-inch, 3840 x 2400, anti-reflective OLED display with touch, this model is available for £4475.99 (the online Lenovo‑discounted price at the time of writing). A saving of £2873 over the MacBook!

However, the Core Ultra 9 185H processor scores 2250 and 12033 in Geekbench’s single and multi-core tests — 44 and 54 percent lower than the MacBook Pro with M4 Max. And if we reconfigure the MacBook Pro with 64GB memory and 4TB of storage, the price drops dramatically to £5349, making it merely £1409 more expensive than the ThinkPad P1 with a single 4TB drive (£3940). But, given that the Core Ultra 9 chip scores lower than the M4 in terms in Geekbench’s CPU tests, that price differential represents more than twice the performance.

although one could presume that the M3 Ultra model should be fine.

To benchmark performance, I recreated a similar Session to those used for Logic Pro and Cubase, this time employing one instance of Eleven Mk II with the default preset on a stereo audio track. The M4, Pro and Max-based systems could play back 69, 105 and 125 tracks respectively, once again showing performance relationships similar to those observed in the other applications.

And, as anticipated, the M3 Ultra result also reveals a similar processing improvement of 74 percent over the M4 Max, with the ability to play back 218 tracks simultaneously — a 68 percent increase over the M2 Ultra.

Given that Pro Tools can also employ dedicated DSP chips for running plug-ins (or a combination of both DSP and native CPU resources in Hybrid mode), I also explored the performance of an equivalently configured session on a Mac mini with M2 Pro running a two-card HDX system via an expansion chassis. Using only the DSP resources available (36 chips), it was possible to run just 17 tracks simultaneously.

However, with Hybrid mode enabled, the track count increased to 73 — 13 more tracks than running natively on the M2 Pro system without HDX hardware.

Eleven Mk II requires two DSP chips (each consuming 96 percent) to run two mono instances when used as a multi-mono instance on a stereo track, with two DSP chips required for Mixer instances (or just one instance in Hybrid mode). Since Pro Tools HDX is now over 13 years old, though, it’s not all that surprising the raw

The M4 and M3 Ultra Mac Studios are outwardly identical to their predecessors.

DSP resources of HDX can be outdone by modern Apple Silicon chips. After all, there are still situations where HDX systems are useful in terms of providing a large number of audio inputs and outputs at low latency, especially when tracking.

To evaluate offline performance, I bounced the ‘Final Mix’ session for Sol Levante, which is made available under Netflix’s Open Content program (https:// opencontent.netflix.com), using Pro Tools’ internal Dolby Atmos renderer. Given that offline bouncing in Pro Tools is always processed natively, there was no need to include HDX systems for these tests.

As you see from the chart, the M4 Max is again quicker (by six seconds) than the M3 Ultra, offering about the same performance as the M4 Pro. Again, as with Cubase Pro, this is presumably because bouncing in Pro Tools doesn’t efficiently scale to use as many CPU cores as are available. It could equally be that this particular session can’t be bounced any quicker than with the 16 cores provided by the M4 Max, and that trying to deploy additional cores simply adds scheduling overhead.

Conclusion

Figure 8: As in Cubase, offline bounce times in Pro Tools don’t straightforwardly mirror the raw processing power of the system.

Apple’s engineers have really outdone themselves with the company’s latest Macs and M-series silicon. The Mac mini is the best it’s ever been, and the company’s engineers have redefined

what ‘mini’ means. It’s also something of a bargain; when the M1 model with 8GB memory launched in November 2020, it cost $699 (around $860 adjusted for inflation). Now, released just four years later, you can buy a Mac mini with M4 and 16GB memory for $599.

But it’s not just at the low end Apple have been busy. While the Mac Studio might not be powered by the M4 Ultra you were looking for, the M3 Ultra delivers the performance even an insomniac has always dreamt about. However, as desirable as the Mac Studio with M3 Ultra undoubtedly is, it’s also likely to offer more performance than many of us actually need — unless, of course, you’re a composer working with big arrangements, even bigger track counts and sample-based instruments so dense they could turn any SSD into a black hole.

The M4 Pro and M4 Max-based Macs, whether mini, Book or Studio, represent an extraordinarily capable middle ground, especially since the M4 Max offers comparable performance to Apple’s previous high-end chip, the M2 Ultra — which still powers the current Mac Pro. And the 14- or 16-inch MacBook Pros, powered by either the M4 Pro or M4 Max chips, remain simply the best notebook computers for musicians and audio engineers. When you consider the performance of the M4 chips, particularly the M4 Max, the near-silent operation and long battery life, the speed of the internal storage, the inclusion of Thunderbolt 5 and pretty much everything else, it’s hard to reach any other conclusion. £ Mac mini from £599; MacBook Pro

from £1599; Mac Studio from £2099.

Prices include VAT. W www.apple.com

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T

hose familiar with my reviews might have sussed by now that, when describing multifunctional modules, I do my level best to avoid that most lazy term, ‘Swiss Army Knife’. But when it comes to anything as multifaceted as the Expert Sleepers Disting NT, I think I might need to make an exception.

The original Disting offered 16 different algorithms, each delivering its own CV and audio process. It doesn’t take a genius to work out the name ‘Disting’ is ostensibly a bastardisation of ‘this thing’ because, well, what else would you call it? Now, five models on, comes the Disting NT. Devotees will be pleased to know that Os has pledged to migrate all the functionality of previous models into the NT, but considering what it’s capable of, even that feels like it won’t come close to plumbing the potential of this thing (see what I did there?). As for the ‘NT’ in the name, I’m informed it doesn’t officially stand for anything. So I’m going to here declare — what with it representing such a monumental new chapter of the Disting story — that it can only stand for ‘New Testament’; a turning of the page.

A step into a wholly new era of Disting.

The New Testament Disting’s immediate distinguishing factor is of course its interface, with a panel expanding upon that of its predecessor by almost a factor of three.

The Disting EX boasted a small OLED screen; here that more or less doubles in size to occupy a sizeable portion of the NT’s faceplate. It’s accompanied by controls constituting three knobs, two encoders and four buttons.

A USB-C port has appeared on the left side of the panel, and there is copious I/O constituting 12 inputs and six outputs. The module also ships with an expander for incorporating TRS MIDI and a brace of extra outputs.

Whew! All controls being assignable, these are not labelled (save the numbered jacks), and as such the Disting NT has a sleek, minimal look to it.

Expert Sleepers’ ambition and precocity with the NT is evidenced by the fact that by the time you read this

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review, the firmware I’m currently working with will almost certainly be out of date. Already numerous revisions have been released, with more in beta, and things show no sign of slowing down. If it wasn’t clear, Disting firmware updates saliently consist of new function algorithms, the list of which is already too long for me to exhaust here. But before getting to those in and of themselves, it’s important to note that, whether with five or 50 algorithms in the picture, the foundation of the NT is its workflow, and this is clearly an area into which much R&D energy has been poured.

The NT’s ‘home’ view renders its algorithms in a list, not unlike plug-ins in a DAW channel strip.

With signal flow displayed as

moving from top to bottom, you simply choose an algorithm and instantiate it. Once more than one is in the picture, you can reorder, duplicate or delete algorithms any which way. Since these are so varied — some are instruments, some are functions, some are effects, and so on — it’s for you to configure each, often down to a fine level of detail.

Each algorithm can either ‘replace’ its input signal (that is, like an insert) or add to it (like a send), and of course there are myriad ways to route signals, either internally or via the NT’s jacks. You could have three independent signal paths running from three different inputs to three different outputs.

You could also use the NT’s

“Previous Distings have been magnificently multifunctional, but with its rebooted interface, extensive I/O and plug-in-style workflow, the NT takes things to a whole new level.”

physical I/O to insert another module into any stage of the signal chain, not just at the beginning or end. You could reserve one or more outputs for a complete voice entirely of the Disting’s own making, with its respective signal routing all happening internally. Once you’ve got a configuration of algorithms to your liking, you can save that as a preset. With so very many options, it would only be limiting to try and map algorithms’ parameters into knob-per-function configurations, as it would also be to limit how an algorithm can be patched — finding patches that a developer never dreamed of is one of the joys of modular, after all.

In this way and more, the Disting NT’s innards do feel somewhat like they are on display — no Teenage Engineering-style pretty pictures here — you can almost see the lines of code churning away underneath. This might be because the screen’s display resembles the startup sequence of a PC in the ’90s, or perhaps it’s the fact that each active algorithm’s live CPU usage flickers on the right‑hand side of the screen. It might also be the fact that in many instances, more than one panel control can change a given parameter or access a particular screen, which I found occasionally makes it difficult to settle into a fluid workflow and leads to some slightly haphazard navigation at times. Since a module like the Disting NT will invariably be menu-heavy — to say the least — knowing exactly where you are at any point is key, and the NT’s menu-diving can quickly feel a little ‘thickety’. This said, there is visual help available for navigation, and a method of sorts does eventually emerge: essentially it’s best to default to the two encoders for navigating menus, and to the three knobs for parameter changes.

As for the algorithms themselves, as I say, I could

never exhaust the list with this modest column space. Suffice to say they range from the utility (such as tuning, clocking, mixing and MIDI interfacing), through the conventional (various oscillators, noise generation, sample & hold, LFOs and the like), all the way to the generative and wildly creative. There’s a slew of spatial effects; from simple reverb to the Expert Sleepers classic Augustus Loop, a deep tape-inspired stereo delay with huge scope for syncopated sonic obliteration.

There’s also a reverb harvested from Mutable Instruments’ Clouds — in fact, there are full iterations of both the Mutable Plaits and Rings in the offing, because why not? The USB-based algorithms mean, among many other things, that the NT can be rendered a 12-channel audio interface if desired. There’s a sequencer, a fixed filter bank, a very well-equipped looper, and several types of sampler. One of my favourite algorithms is the Kirbinator, designed in collaboration with synthesist Simon Kirby, whose very cool functionality happily belies its distinctly dorky name.

Described as a “stochastic audio processor”, the Kirbinator chops, stutters, reverses, repitches and slices audio according to a set of deployed markers and probability values.

“What is missing from the NT arsenal from your perspective?”

Os asked me in a recent email.

Good question. Argh, I may as well go for it: the Disting New Testament is nothing less than the epitome of Swiss Army Knife modules. Swisser than a wheel of cheese on skis. Previous Distings have been magnificently multifunctional, but with its rebooted interface, extensive I/O and plug-in-style workflow, the NT takes things to a whole new level. It’s priced at the premium end, but don’t let that fool you: considering the legions inside, it’s nothing short of a steal.

William Stokes £

£599

W

www.expert-sleepers.co.uk

Cutlasses Instruments Gloop Eurorack Module

T

he Gloop, the debut module from Cutlasses Instruments, is a tape-style looper based around an ingeniously simple idea — instead of having one playhead, the Gloop has four. You record a loop using the big red recording button, and then use the four clickable encoders to change the start point and loop length of the four different channels. You can also change the playback speed (either as a ratio or in semitones), the playback direction and ‘jitter’ (how faithfully — or not — the loop will adhere to its start point) for each channel. The four knobs beneath the encoders give you volume control over each channel. There. Simple.

Except it’s not. Well, the idea is simple — rather brilliantly so — but the results it produces are not. Record a simple 16-step sequence, say, shuttle through the four channels by pressing the first encoder, changing loop lengths, pitch, etc, and very soon your simple sequence is nothing of the sort.

For example, imagine channel 1 is reversed and running at half speed (Hainbach style), channel 2 is looping around three notes, channel 3 is pitched up by two octaves and channel 4 is showing re-runs of Property Ladder, sorry, I mean is playing half the loop length, but the position of the loop is being controlled by CV. Controlled by CV? That’s right. There are four CV inputs and a healthy list of things you can control with them,

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either for all channels or individually. This includes, but is not limited to, playback speed, loop start point, loop length, volume and pan. Pan? Yep. The Gloop is stereo, at least on the way out.

While all this is going on you can use the four level knobs to create a performance, fading the double-speed version subtly in and out and taking full advantage of the built-in effects. Effects?

Oh yes. Pressing the small white button takes you through pages for delay and reverb, bit‑crush and bit-depth, tape-style effects (wow and flutter, saturation, hiss and age) and Degrade.

Degrade deserves its own paragraph as it’s essentially a very good implementation of Frippertronics. In the Gloop version you record a loop, stop recording and then press record again.

The recorded loop will now fade as you record more parts over it. Turning the Degrade knob affects the speed at which the loop fades — the further the quicker. The Gloop can do two things that tape-based Frippertronics cannot, though. Firstly, you can turn the Degrade control down at any point to arrest the loop at its current level, and secondly you can record with reverse enabled so that you’ll hear what you play in forwards as it’s recorded, but then backwards at every subsequent playback. A mixture of forwards and backwards channels and adjusting levels as you go is a lot of fun.

Speaking of fun, The Gloop’s Trigger input will start recording when it receives a trigger, and stop recording when it receives another, and there’s also a grid mode. Combining these things means you can record in-time loops and then mess with them. And yes, you should try it with the Amen break. And lots of other things beside.

So, not really so simple after all, but perhaps the best thing about The Gloop is the balance between immediacy and depth. It’s very easy to grasp the idea behind it and get interesting results straight away, but there’s a lot more to explore when you want to. And the process of exploration seems to happen without you noticing it. Yes there are menus to navigate, but they’re mostly self-explanatory and don’t detract from the experience of using The Gloop. And like all the best modules, the more you use it, the more ideas you’ll have for how to use it. David Glasper £

£325

W

www.cutlasses.co.uk

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RYK Envy Machine Eurorack Module

A

nother swish-looking module from RYK Modular. Gorgeous in black with those pin-prick LEDs and mysterious-looking endless encoders. The Envy Machine is a four-channel, five-function modulation generator that fools you into thinking it’s intuitive when you might need to concentrate and learn some ropes before the enjoyment begins to flow.

The layout appears simple enough.

You have a column of three patch points for each channel, and above them is a rather lovely LED strip crowned with an encoder. On the surface is a scattering of words, letters and symbols that seem straightforward, but nothing is exactly as it seems. Each channel can be an ADSR envelope, an AD envelope, an LFO, a random source or a recordable channel of custom CV. The four neat columns are actually an illusion and each channel steals the whole interface for itself. To know what you’re looking at you have to learn what the LED colour and layout means. It’s not hard as such; it just makes a casual approach more difficult.

However, once you get into it, the functionality becomes sublime. You use the blue button to swap between the channels, denoted by a red LED lighting above the corresponding patch sockets.

You can then hold the blue button and tap the Opt button to cycle through the modes, which are then represented in the four LED strips. The text means nothing at this stage. Envelope modes are red, with the ADSR lighting all four strips and using all four knobs for the four stages. AD mode lights the first two strips and takes the first two knobs for attack and decay. LFO mode is slightly orange, and you get three LED strips that give rise and fall controls along with an offset.

The Random mode remains orange but gets the first and third channels lit for frequency and offset. The last mode is yellow for CV recording, and you get a single strip for that.

The text comes into play when you switch to the green LEDs of Options mode. Pressing the Opt button lets you assign a destination for the selected channel modulation input and set an amount for that. You can set the shape of the LFO modes and set a bipolar or unipolar output level. It all

comes together nicely once you know what you’re looking at.

The assignable CV input is a really nice feature and well implemented without having to go into levels of menus.

The recordable CV channel is particularly satisfying and I can see that getting a lot of use. It also highlights what feels like a missed opportunity. When you record the knob movement, the LED strip animates up and down showing the modulation: it’s visually useful and superb.

If all four channels are set to record CV, then you will see all channels animated at once. But this doesn’t happen in any other mode, and that’s a shame. The single LED above the patch sockets gamely tries to indicate what’s going on with each channel, but the strip of LEDs would be a far better place to represent the movement. It feels like it needs an overview mode so that they can all dance as brilliantly as the recorded CV. Otherwise, the Envy Machine is another thoughtful and cerebral module from RKY Modular that offers a lot of modulation versatility in a beautiful package. Robin Vincent £

£205

W

www.ryk-modular.com

88C DUAL DESKTOP DYNAMICS

Iconic Sound Quality | Versatile Control | Legendary Character

Learn more by scanning the QR code or visit neve.com

ON TE ST

Is Arturia’s new KeyLab 88 MkIII the best 88-note controller keyboard in its class?

ROBIN BIGWOOD

T

he KeyLab 88 MkIII is the latest iteration of Arturia’s fully-fledged MIDI controller keyboard, and rounds out a range that already includes 49- and 61-note weighted‑key models. All the models in the range have almost identical scope and capability as controllers, offering deep integration with Arturia’s software instruments (running on your computer) and various DAW software, and more generic control of virtual and hardware MIDI devices alongside. All of what Robin Vincent discovered in his SOS December 2024 review of the KeyLab 49 and 61 holds true here, so I won’t cover the same ground. But there is uniqueness here in the form of the much larger keyboard, a few other points of design, and how they play out in the user experience generally. So let’s fire up the Bunsen burner and get stuck in.

Arturia KeyLab 88 MkIII Controller keyboard

Bed-making Where previous KeyLab 88s used a Fatar TP/100LR keybed, the MkIII gets something more recent, a TP/110. This model is still not documented on Fatar’s website as I write this, but it offers velocity and channel aftertouch sensitivity, and from word go I found it to be both controllable and musically satisfying, with minimum and maximum velocity and aftertouch values easy to generate from a normal range of touch and force. Three different response curves are provided for both velocity and aftertouch, as well as a fixed value option and a programmable custom curve.

Physically, white keys measure a full-size 150mm front to rear, with a standard 165mm octave span. Notably, though, key dip is just 9mm or so, which is less than most acoustic pianos and many stage keyboards. It seems quite appropriate, given the likelihood that the 88 MkIII will be used at least as much for synth and sample duties, and doesn’t meaningfully compromise the weighted, swinging piano-key feel. Aftertouch is sensed in a further 3mm or so of firm resistance for white keys; less for black.

The TP/110 action is relatively quiet: there’s a damped, wooden-sounding knock at the bottom of the keystroke, and a much quieter sound on key release. You will hear

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of players criticising their TP/100 actions as being (or becoming) too bouncy, notchy, or having too much lateral key movement.

None of those things was evident here, apart from a minuscule bounce, less than 1mm, observable with staccato playing.

Indeed, the overall impression I got was one of smoothness and precision. The KeyLab generates release velocity data, incidentally, but even exaggeratedly fast releases only produce values in the 30s (or 40s, for black keys) so it’s not terribly useful.

It’s fair to say that the older TP/100 (in its various different guises) divides opinion — and that’s being diplomatic. A full love-hate spectrum can be experienced on many an Internet forum. I’d personally judge the TP/110 here a fine, middle-tier, quite ‘neutral’ action on a par in quality with Roland’s PHA-4 or Yamaha’s GHS, but with the

distinct advantage (for many jobs) of very good aftertouch. Much better for ambitious playing, I think, than that TP/100, and rather smoother than the Fatar TP/40 (in a Nord Piano 3) I have in my studio. So top marks, Arturia. This is a notably fine action for the money, and helps the KeyLab 88 MkIII carve out a distinct niche in the market.

If there’s an elephant in the room still, it’s the expectation some pundits had that this flagship-level controller would offer polyphonic aftertouch, especially given it’s in Arturia’s recent PolyBrute 12 and Native Instruments’ Kontrol S88 MkIII controller (which uses a specially developed Fatar TP/100 variant). It’s a fair point, as poly aftertouch and MPE compatibility becomes more and more common. Perhaps, though, in the toss-up between offering that or an improved piano touch (with boring old

Arturia KeyLab 88 MkIII

Les Accoutrements Carrying on a tradition established with the KeyLab 88 MkI model, the MkIII comes as standard with two accessories. The first is a shelf, and it’s great: metal with a non-slip rubber surface, aligning with rear-panel metal tabs to create a spacious area to the right of the faders and encoders. It sturdily supports a laptop or module up to 380mm wide and 250mm deep (so, a 17-inch or even bigger laptop).

Then there’s a metal and perspex music desk, which is useful too, and thoughtfully has an open slot at its centre to allow for a portrait format tablet to placed with a USB plug connected. The desk sits left of centre of the keyboard though, which is visually a bit odd, and at about 190mm tall it’s not big enough for really large, heavy show-type scores, or some Real books. Still, it’ll be useful for some jobs.

channel aftertouch), it was the more mainstream use consideration that won out. I have some sympathy with that choice. Whilst I enjoy exploring the possibilities offered by MPE, it’s arguably better suited to controllers with lighter synth‑action keys and really ‘bendy’ aftertouch, or indeed a purpose-built key action, like that of an Osmose or Seaboard. Also, the ability of a controller to generate MPE or poly aftertouch doesn’t immediately imbue every instrument you control with a way to respond to it: far from it, in fact. Arguably, purpose‑built instruments or controllers (Osmose, Seaboard, Hydrasynth, PolyBrute) are the best bet.

Pedals, Pads & PAs If you’re into pedals, you’ll like the KeyLab. There are four rear-panel pedal sockets, marked Sustain, Expression, Aux In 1 and Aux In 2, but the names don’t mean a great deal as all are identically, equally capable. Plug in one or several switch- or continuous-type sustain or normal expression pedals and you can derive all sorts of useful control from them, with configuration and calibration options available on board. In Arturia’s Analog Lab software, for the Arturia integration mode, pedal assignment is currently only available for the KeyLab’s Sustain and Expression inputs, but perhaps dedicated Aux pedal support will come in future.

As for the drum pads, which can be usefully employed to generate MIDI CC and Program Change messages as well as notes, they’re more of a qualified success. At about 29mm square they’re compact, with translucent edges that allow their multi-colour backlights to shine through. I found velocity values pleasantly easy to control, above a value of about 20: beneath that, triggering is unreliable. A long pad press can produce poly pressure messages. However, a down side is that both velocity and

pressure sensitivity is hugely reduced at the edges and corners of the pads, so you need to be accurate. Oh, and MPC diehards will have already spotted that there are 12 pads, not 16, which might compromise mappings for a few drum-based software instruments.

Further playing-oriented features include pairs of dedicated buttons for semitone transposition (always useful) and octave transposition (less useful with an 88-note keyboard) along with the Scale, Chord and Arpeggiator features described in the previous review. Those are good, far from an afterthought, and wonderfully easy to control via the LED display and soft buttons.

SpaceLab Expectations for a big MkIII KeyLab were great in some quarters, and users who

£839 pros • The Fatar TP/110 keyboard action works very well for both piano and general playing duties.

• A really well-designed operating system that’s dead easy to use.

• Encoders and faders are touch sensitive, coming into their own with Arturia software integration.

• Good pedal connectivity, and some handy accessories included.

cons • Some physical control locations are suboptimal.

• Pads are small, uneven in response across their area, and there’s only 12 of them.

• No poly aftertouch (if that is what you were looking for in a hammer‑action controller).

summary A proper, grown-up 88‑note MIDI controller that feels great to play and is really versatile. Good value for money too.

were holding out for a poly aftertouch and CV-equipped monster might be disappointed. However, looking at what we have (rather than what might have been), I can’t help but think the

Analog Lab V, part of the KeyLab’s bundled software suite, gives a slick and reliable hardware control experience. The software knobs and faders mirror the control layout on the KeyLab, and operating one of the physical touch-sensitive faders or encoders shows parameter values in meaningful units on the KeyLab’s display.

Bidirectional communication is handled completely behind the scenes, with no setup required, even when running Analog Lab in your DAW, which is really quite impressive. One notable limitation of the system is that whilst patches can be filtered by multiple criteria in software (eg. to drill down into Strings, on the Jup-8 synth, in an ‘Atmospheric’ style), only one filter at a time (along with ‘Liked’ status) can be applied from the KeyLab’s controls.

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ARTURIA KEYL AB 88 MKIII

I found the KeyLab operating system delightful and quick for making MIDI control assignments.

Holding the Settings button and touching a control instantly brings up a really clear, informative configuration screen in the LED display. For some jobs though, Arturia’s freely available MIDI Control Center application (or Windows/macOS) will make ambitious setups even quicker to achieve. Here, fader 6 is being configured to generate a MIDI CC message, in a user memory. Six user slots are available in the hardware, but any number can be offloaded to or stored from the app.

KeyLab 88 MkIII is very fine: a grown-up controller keyboard, marrying quality with versatility, and musical potential with excellent ease of use.

Things I was less keen on were few.

The positions of the pads, DAW controls and faders/encoders could desirably have been swapped, in this form-factor, for a more noodle-friendly setup (for right-handed players at least). Also the location way out west of the often-used Prog and Settings buttons (along with the pitch and mod wheels for that matter) felt a little awkward at times. The pads leave something to be desired: they’re usable as they are, and are great as a bank of momentary/latching buttons, but having 12 instead of 16 will be a bitter pill to swallow for finger-drumming MPC-schooled users, and they simply do not trigger well at the edges.

Against this, though, I found masses to enjoy in the KeyLab 88 MkIII. First and foremost, that new keybed, which should work well for pianists and one-finger synth widdlers alike, offering precision and fluidity alongside easily explored aftertouch. The programming system, with that very legible and bright LED screen, is the clearest and quickest of any MIDI controller I’ve ever used, by miles. I loved it, and never once felt the need to consult a user guide, even for really specific programming tasks.

The way the three main modes focus

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“I’d say we have something that strikes a really great compromise, between multi-purpose playability, software-specific integration, tremendous ease of use, value for money, and general all-round usefulness.

It’s one of the best 88‑note options on the market, right now.”

operation for different tasks makes for great clarity too. Arturia software integration could certainly be further improved, but it’s already good as it is.

It’ll make many virtual synth aficionados very happy, and I never saw the slightest flakiness or lag in the connection.

Ultimately, any MIDI controller keyboard is a compromise, one way or another, but in the KeyLab 88 MkIII I’d say we have something that strikes a really great compromise, between multi-purpose playability, software-specific integration, tremendous ease of use, value for money, and general all-round usefulness. It’s one of the best 88-note options on the market, right now. £ £839 including VAT. W www.sourcedistribution.co.uk W www.www.arturia.com

ALTERNATIVES There’s a surprising lack of choice of feature-rich 88-note hammer‑action controllers, particularly at the more ‘premium’ end of the market. Sitting just above the KeyLab MkIII in price is Native Instruments’ Kontrol S88 MkIII. It offers a bigger LED screen, sophisticated NI Komplete Kontrol integration, and poly aftertouch alongside the NI Light Guide system. At the same time it employs a variant of the older TP/100-based keybed, and has

just a single set of eight encoders for real-time control. Roland’s A-88 MkII, meanwhile, offers MIDI 2.0 velocity resolution but has no aftertouch, just eight small control knobs and buttons, and an arcane programming system without a screen. Interestingly, StudioLogic’s significantly cheaper SL88 Studio MkIII has the same TP/110 keybed as the KeyLab, and MIDI 2.0, but real-time control is limited to two joysticks and a handful of knobs and encoders.

PODCASTS ELECTRONIC

RECORDING

MUSIC

& MIXING Using Tube And Tape Plug-Ins

Robert Aiki Aubrey Lowe

In this episode, sound-designer Eddie Bazil looks at ways of enhancing a variety of individual tracks and full mixes using plug-ins that are designed to achieve the retro warmth and cohesion of tapes and valves.

Musician Robert Aiki Aubrey Lowe talks to Caro C about his preferred sound modules and working methodologies, plus shares details some of the film and TV projects that he has recently been collaborating on.

Six Creative Examples

Composer and Sound Designer

ELECTRONIC

PEOPLE &

MUSIC

MUSIC INDUSTRY

Martin Kohlstedt

Malcolm Toft

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.

Malcolm Toft gives a fascinating insight into his career, from building his first mixing desk at age 17, being part of the team at the launch of the iconic Trident Studios, where he recorded Elton John, Queen and David Bowie, to founding Trident Audio Developments to manufacture his mixing consoles.

Piano & Electronica

Trident Mixers Founder

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

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TECHNIQUE

Spectral Balance Tips To Help Your Mix Translate

Subtle issues with the frequency balance can stop mixes from translating well to different listening systems.

We offer a couple of suggestions that can help.

SAM INGLIS

M

ixing was once called ‘balancing’, and there are at least two senses in which a mix needs to be balanced. The obvious one concerns the level of each individual source or instrument. We want the vocal to be clearly audible. We want the drums to be powerful but not overwhelming. We want the guitars to drive the song along without overshadowing the vocals. And so on.

Less obviously, a mix also needs to contain a satisfying balance across the frequency spectrum. Some low bass is often desirable, but too much low bass causes all sorts of problems. We want the midrange to be full and punchy, but we want to avoid pushing it so far that the mix sounds boxy or tinny. We want to retain detail and excitement at the top end without taking the listener’s head off.

The two goals are, to some extent, interconnected; fading up the bass guitar or the kick drum will mean more low bass in the mix, and so on. But it’s still very possible to end up with a mix that is ‘balanced’ in the first sense yet not in the second. In fact, I’d go as far as to say that this is the single biggest problem I hear in mixes not done by professional engineers. It’s also one of the challenges I find hardest to overcome in my own work. Familiar techniques such as A/B’ing with level-matched reference tracks can be a big help, but in this article, I thought I’d share a couple of less obvious strategies I’ve developed.

Light & Dark

Linearising your headphones with tools like dSoniq’s Realphones or Sonarworks’ SoundID can be valuable in its own right, but what’s often even more useful is the instant change in perspective you get from toggling correction on and off.

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A word you’ll hear a lot in articles about mixing is ‘translation’. One of the mix engineer’s key skills is to make the song sound good not only on their own playback system, but to anticipate how it will work on other systems, including consumer devices.

To this end, many engineers employ secondary monitoring systems, including

‘grotboxes’ deliberately chosen for their low fidelity. However, in my experience, these are more useful for fine-tuning level balances than frequency balances.

A two-inch cube speaker can be very helpful for highlighting how well the vocal is sitting in the mix, but it won’t tell you whether you have too much going on at 50Hz. However, switching from speakers to headphones can also give you a useful change in perspective, and my first tip for checking frequency balance builds on this idea.

Over the years, I’ve reviewed lots of different headphones, as well as several software correction tools designed to linearise their frequency response. It’s not hard to hear the difference these tools make, or to notice that one pair of headphones has a ‘scooped’ sound and another a midrange focus. But what has surprised me in practice is that if I’m forced to mix on headphones, linearising them doesn’t help as much as you might expect. I don’t consistently do better mixes on uber-flat cans costing £2k than I do on closed-back models costing £150, and I think that’s because there are other, more important factors that make it hard to mix on headphones.

Despite this, I’ve come to rely very heavily on headphone correction as a tool for checking mixes, and specifically for judging frequency balance. The way I use it is as follows.

I’ll carry out a mix at my studio, listening mainly on my Genelec main monitors and occasionally switching to my NS10s.

When I think the mix is getting close, I’ll bounce it out, bring it home and fire up my laptop running either Sonarworks SoundID or dSoniq Realphones. (Other headphone correction tools are available too, including Acustica’s Sienna and Toneboosters’ Morphit.) I’ll then listen to the mix on my trusty Oppo PM-3s: headphones that I know well, which have a full frequency response, but which, like almost all closed-back headphones, aren’t completely flat in their raw state.

In the specific case of the PM-3, which are planar magnetic headphones, the correction curves apply a broad high-frequency boost, a dip around 1kHz, and another dip in the low midrange; with typical moving-coil headphones it’s more likely that the correction will tame an upper-mid boost instead, but it doesn’t

really matter. What’s important is that the raw and the corrected sound be noticeably different, but both within the realm of normal for headphone playback. In my experience, toggling correction on and off during playback then does a uniquely good job of highlighting frequency anomalies in the mix.

Very often, for example, I’ve played back one of my mixes with headphone Oeksound’s Bloom will ‘push’ the spectral balance of your mix towards an correction disabled, ideal target. The changes it makes in doing so can be useful clues as to what’s thought it sounded wrong with your mix in the first place! great, then winced when SoundID reveals that the cymbals a diagnostic tool. Put Bloom in the last are sounding vicious at 10kHz. Other insert slot on your mix bus, crank up the times I’ll be feeling pleased with myself Amount dial and listen to the results. when the ‘flat’ response sounds perfect, Does the frequency balance of your mix change significantly?

If so, is it improved, and which regions are most affected? If Bloom suddenly makes the lower midrange clearer, chances are you have some mud in there that needs sorting out. If it radically brightens up your mix, you probably need only to discover that it sounds horribly to revisit your EQ choices in the upper honky or muddy on the uncorrected midrange. The same applies right across ’phones. And, in general, the less the the spectrum. spectral balance appears to change when Like most plug-ins, Bloom is available I switch, the closer it is to being right, and for free for a trial period, so you can see to translating well across other systems. whether it works for you. Alternatively, you could try other automatic Blooming Marvellous tone-shaping plug-ins such as ProAudio DSP’s Dynamic Spectrum Mapper, or the Having found this technique surprisingly Stabiliser module in iZotope’s Ozone 11. effective, and often brutally revealing of The same approach can also be made my limitations as a mixer, I’ve also evolved to work with a ‘match EQ’ plug-in and a related habit which helps to forestall too a suitable set of reference tracks — just many nasty surprises at the translation be aware that the genre, crest factor stage. Last year I gave a glowing review and even the key of those tracks can to Oeksound’s Bloom, a plug-in that make a big difference. Software is no dynamically rebalances the frequency more infallible than the human ear, but response of your audio to push it towards like headphone correction, it can give an idealised target. As with most such you that fresh perspective on your mix, tools, it’s very easy to overdo things with making you aware of issues you wouldn’t Bloom, but you don’t have to use it as otherwise have spotted. a corrective tool: it’s also invaluable as

“What’s important is that the raw and the corrected sound be noticeably different, but both within the realm of normal for headphone playback.”

www.soundonsound.com / May 2025

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M-Live B.Beat Pro 16 Plus From £1062 pros

M-Live B.Beat Pro 16 Plus

Live Playback System

The B.Beat Pro 16 is much more than just a dedicated hardware backing track player...

MARK GORDON

W

hether or not you agree with the principle, using pre-recorded audio backing tracks has become an essential part of many live performances. From named acts doing arena shows through to function bands and solo artists playing at a local level, many performers now rely on backing tracks to support what they do on stage.

There’s an almost infinite number of ways for artists to incorporate backing tracks into their performances, from something as simple as an iPad playing back mono or stereo files through dedicated multitrack hard-disk recorders to a laptop, DAW software and an audio interface. A complex computer-based system allows for the most flexibility, in that you can send individual stems to FOH, change the playback order of songs, use MIDI files to control patch changes, and even incorporate video into your performance. However, it does come at a cost, both financially and logistically, particularly if you’re working

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at a semi-professional level. There is also the school of thought that suggests computer-based systems are unreliable and will crash when used live, which I think in this day and age is unfounded, although there is arguably a certain fragility when compared to a dedicated hardware solution. So what are the alternatives, if any, for a system that doesn’t break the bank or test your technical ability?

One option may be the B.Beat Pro 16, from Italian company M-Live. It’s a 2U rackmounted unit that aims to offer almost all the flexibility and functionality of a laptop-based playback system, in a robust hardware format. Featuring 16 analogue outputs, 12 inputs and up to 1TB of internal storage, the B.Beat Pro 16’s ability to play back audio is only part of the story. It also includes two HDMI outputs, one to transmit synchronised video and a second to display lyrics to an on-stage prompter. It also features a MIDI output, dedicated headphone output, and the ability to display PDF chord charts or scores via iPads and tablets. Add to this an

• Easy integration with Mac or PC via B.Beat Manager.

• Excellent monitoring options.

• Flexible playlist creation and editing.

• Video, MIDI file and PDF support.

• Synchronised Lyric display via HDMI or app.

cons • External PSU.

summary The B.Beat Pro 16 is an excellent alternative to using a laptop-based system for live playback, offering a huge amount of functionality and flexibility in a rugged and easy-to-use format.

iOS and Android app that allows full remote control and the B.Beat Pro 16 has the potential to run your whole show. In fact, you might only need to turn up to press play!

Audio, MIDI files and video prepared in your DAW can be easily reproduced in a live environment using the B.Beat Pro 16, with tracks sent to individual outputs, images to projectors or video screens and songs organised into playlists for seamless performances.

Hard Times Available in two formats, the B.Beat Pro 16 comes fitted with either a 128GB internal hard drive or a 1TB hard drive, the latter adding the ‘Plus’ suffix. The 1TB version, which I had for this review, also includes built-in Wi-Fi connectivity, whereas the 128GB version requires the optional M-Pen 2 USB aerial to add Wi-Fi functionality.

Housed in a functional black metal casing, the B.Beat Pro 16 strikes an industrial pose. The rear panel sports 16 balanced audio outputs and 12 unbalanced inputs on TRS

quarter-inch jack connectors, along with a standard 5-pin MIDI output port, two HDMI connectors and a quarter-inch footswitch socket which functions as remote start/stop. The on/off switch is located at the rear above a 12V PSU socket. I found this a little disappointing for a product intended for use in a professional environment; to me, an IEC mains connector always feels more substantial and robust.

The front panel offers a plethora of buttons, knobs and sockets, surrounding a relatively small but functional 128x64 monochrome OLED display. Beneath the display are three function buttons allowing direct access to menu items displayed on the lower portion of the screen.

A large data-entry wheel and up/down buttons enable further navigation around the display, and dedicated buttons are provided to access the library, playlists, shows, built‑in metronome and system settings.

Playback is controlled by a two large, brightly backlit Stop and Play buttons, alongside a small Record button that enables the B.Beat to record audio directly to its internal drive via the 12 analogue inputs.

To the right of this is the monitoring section, which is particularly well thought out, with a quarter-inch headphone socket sitting beneath three volume knobs that independently control the click, inputs 1+2 and backing track volumes. As a drummer who plays to live backing tracks, having quick access to individual volume levels is a big plus point for me. The inclusion of a discrete level control for inputs 1+2 is also a stroke of genius, as it allows you to use these inputs as a feed from your monitor mix

and balance the band level, backing tracks and click track independently.

Get Appy Although the front panel of the B.Beat offers many ways to control playback, import and organise songs, the initial configuration of the audio, MIDI and video source material is more easily done via the B.Beat Manager, an application available for Mac and Windows, which can be downloaded from the M‑Live website. The B.Beat Manager defaults to displaying the Audio Projects screen, where three demo/test Projects are displayed.

combination: WAV 44.1kHz, 48kHz and 96kHz at 16/24‑bit; MP3 fixed bit rate (64-320 kbps); MP3 VBR; OGG fixed bit rate (64-320 kbps); AAC; and STEMS, a relatively new open multitrack audio format. Supported video formats include MP4 and MOV, and you can also import and display static images in JPEG or PNG format.

Set Up A Song Once everything is imported, it’s time to configure the Project (song). Opening an imported Project by double-clicking on its name brings up a list of the associated audio files, alongside a graphical display of each waveform. Transport controls at the top of the screen allow you to audition the Project, and you can also drag the playback wiper freely within the waveforms to locate to specific sections. At the bottom of this screen is the ‘Extra’ content — the video, MIDI and PDF files, along with a Lyric option that I’ll come to later.

Within each Project, you can allocate audio outputs to each track, set a volume level, and decide if that track will be sent to the headphones in addition to its own output. Audio can be sent to all outputs in any combination, with outputs 15+16 mirroring what is sent to the headphones.

The B.Beat Manager doesn’t allow for any additional editing of the MIDI and video files in the Project. These simply play ‘in sync’ with the audio, so will need to be pre-prepared in your DAW, prior to loading onto the B.Beat. If the video file is shorter than the audio, the final frame will remain static on screen until the song is finished, but you can enable a loop option that will continuously cycle the video for the duration of the song.

JPEG and PNG images will also display

“The ease with which projects created on a computer can be transferred makes constructing sets and shows an absolute pleasure.”

Transferring your own tracks to the B.Beat couldn’t be easier. A large ‘+’ button on the main screen gives the option to ‘Create a new Audio Project’, and it really is as simple as dragging files from your computer to the B.Beat Manager. Once they’re imported, hitting the Transfer button sends the files to the B.Beat. Transfer is fast and seamless, and a Project containing seven stereo audio files (around 400MB in total) took 40 seconds to transfer via Ethernet and 120 seconds via Wi-Fi. Each Project can contain 24 audio tracks, one video file, one MIDI file and (as far as I can tell) a limitless number of PDF files. All audio is converted into uncompressed 24-bit/48kHz PCM format during the transfer, but you can import almost any audio format in any

The back of the unit is home to a quarter-inch pedal input, two HDMI ports, a MIDI out socket, 16 quarter-inch audio outputs and 12 quarter-inch audio inputs.

www.soundonsound.com / May 2025

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M - L I V E B . B E AT P R O 16 P L U S

The final element is Scores or, more accurately, PDFs (so as not to put off the non-readers among us). Any number of PDF files can be attached to a Project and, via a similar method to that employed by the Lyric mode, individual pages can be synchronised to a point on the song’s timeline.

At this point, you could perform any of the songs contained in the B.Beat.

However, the B.Beat offers still more, specifically the ability to build Playlists, Shows and Scenes.

Get Sets

Imported projects displayed in the B.Beat Manager library show audio waveforms, output routing and family groups, along with any associated video, PDF and MIDI files.

as a static image, which is a handy way to display your band logo.

A MIDI file could potentially contain Program Changes, Continuous Controller data and even note information, to control any connected MIDI devices such as effects units, and potentially lights, for complete automation of your live performance.

Any included video is routed out via HDMI 1, while the HDMI 2 port is dedicated to Lyric mode, a feature that allows you to import lyrics, sync their playback to the track and display them

in what I would describe as karaoke style. This is a fantastic feature that works extremely well and is easy to configure. Copy and paste your song lyrics into the edit window and, as you play back the song in real time, simply hit Enter or press the ‘Sync Lyrics Row’ button to time-stamp the lyrics. The left side of the window provides a preview, highlighting each line as you press enter. Once this is completed, you can use the preview screen to check everything is in time and, if necessary, edit any sync points.

The Prompter App The B.Beat Pro 16 happily functions as a standalone unit, but M-Live have created Prompter, a free app for iOS and Android that offers full control of the B.Beat via a phone or tablet. The Prompter app can also display PDF scores and chord sheets or lyrics associated with each song, enabling each member of the band to see the relevant charts and notation. The app is based around three main screens: • Remote is a virtual representation of the B.Beat front panel, giving access to every control found on the B.Beat. Functions respond instantaneously, with no perceptible delay when pressing Play or Stop, or accessing any of the functions.

• Scores: This screen will display any PDF files associated with the currently selected Project and scroll through each page in sync with the song. From a drop-down menu at the top of the display, you can select any Instrument Group that has been created in the B.Beat Manager and only view the PDF files associated with that group. This can be a useful aid if you’re working with a large function or show band, where each player selects their instrument group and is only presented with the charts they need. As the song plays, the pages ‘turn’ automatically, as configured in the B.Beat Manager app.

• Lyric: This screen displays, as you might expect, any lyrics that have been created using Lyric mode in the B.Beat Manager, and mirrors the output of the HDMI 2 port, using the same ‘teleprompter’ style display, which scrolls in sync with the song.

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The Prompter app for iOS and Android devices allows full remote control of the B.Beat Pro 16.

Playlists, Shows and Scenes are ways of collating songs into groups or sets and are, again, most easily created within the B.Beat Manager software.

Playlist is the simplest format. You can add any song that exists in your Audio Projects library and drag them into the preferred order. Once created, each Playlist displays the included songs plus a total running time, which is very useful when creating performances that are time-critical.

Press Play and the B.Beat will immediately play the first song in your selected Playlist, with the LCD displaying the name, elapsed time and which song is next in the list. A Skip function lets you easily skip a song in the Playlist, should you need to, and you can also manually insert a song ‘on the fly’ via the Library button. Very handy if you realise you’ve included ‘Suspicious Minds’ in your wedding Playlist and need to replace it with something more suitable! This is a temporary change and doesn’t permanently affect the Playlist, but you are able to create and properly edit Playlists from the front panel should you need to.

The default state for the B.Beat is to stop after each song and wait for the Play button to be pressed again to resume playback, but you can change this within Settings, to ‘Autoplay’ each song. This can be useful if you want to press Play at the start of your show and not interact with the B.Beat until the current Playlist is completed. Autoplay is a handy option, but the Shows and Scenes features allow for even more advanced ways to construct a performance.

You can think of a B.Beat Scene as a Playlist on steroids, a list of Projects but with more options. If you create a new Scene and add Projects from your library in your preferred playback order, you can

select each one to Autoplay or to wait for a press of the Play button on a per-song basis. You can also add an auto-start delay, so that the next song will wait for a predefined number of seconds before playing. It’s also possible to insert just a metronome click for a pre-defined time period, which could perhaps accompany a ‘live’ section of the performance.

Once you’ve created several Scenes, you can either select and play them back individually via the Scenes button on the B.Beat, or move to the next level in the B.Beat Manager software, and create a Show. A Show is a way of ordering Scenes and deciding whether they should Autoplay, stop or loop at each transition point. If you were creating a typical rock or pop band performance with two sets, the Show option might not offer many advantages, but if you’re creating a more complex production for a theatre show or musical, making a Show that contains several Scenes might be a useful option.

There’s a fair amount of overlap between what Playlists, Scenes and Shows offer, but each one adds an extra degree of flexibility or functionality that may allow you to tailor a performance in the best way for its context.

B Impressed As a gigging drummer using backing tracks on a regular basis, and being the band member who creates and constructs the backing tracks, I was intrigued by what the B.Beat Pro 16 might offer... and I haven’t been disappointed. The ease with which projects created on a computer can be transferred makes constructing sets and shows an absolute pleasure,

B.Beat’s Little Brother If you don’t need all the features of the B.Beat Pro 16, M-Live offer a compact alternative simply named the B.Beat. Available in 32GB, 64GB and 128GB versions, the B.Beat has a standalone ‘desktop’ form factor but remains compact, measuring 130 x 220 x 50mm.

In operational terms, the B.Beat offers almost identical functionality to its bigger brother and uses the same B.Beat Manager software to transfer files and configure playlists. The primary differences between the units are physical: the LCD and many of the direct access buttons remain, but outputs are limited to six balanced quarter-inch jack sockets. The input count is similarly reduced, to two, but these inputs can still be used as monitor returns, routed to the headphone output and balanced with backing tracks and clicks. Only a single HDMI socket is provided for video output, and the MIDI output connection is via a 3.5mm jack that will require an adaptor to convert it to the more recognisable 5-pin DIN format.

At around half the price, and with only minimal limitations compared to the B.Beat Pro 16, the B.Beat could be a great option for many gigging musicians looking to incorporate backing tracks into their performances.

with all files automatically converted to the correct format and placed in the right location. Assigning outputs and levels in the B.Beat Manager app is intuitive, and the inclusion of additional features such as Families and Instrument groups adds a useful level of additional functionality.

The inclusion of MIDI and video playback sets the B.Beat apart from many other hard-disk‑based solutions and the Lyric and PDF score features are incredibly well thought out and implemented.

Scenes and Shows allow the B.Beat to

Any number of PDF files can be attached to a project. These can include scores and chord charts that can be viewed by each musician via a dedicated app, and scroll in sync with audio playback.

work well in more theatrical environments and offer a lot of flexibility in how it can be used. You even have the option of sync’ing multiple units together to create a redundant system, where a second unit can take over playback without affecting the performance, should part of the system fail.

Physically, the B.Beat appears rugged enough to handle life on the road and requires little or no setup compared to a computer-based playback system, which is a big attraction for any live performer. The large, backlit buttons make it straightforward to use on a dark stage, and simple but important things like being able to skip or add songs and edit output routing on the fly are very well thought out and implemented. The Prompter app (see box) is a great addition for both controlling the B.Beat remotely and for displaying charts and lyrics for musicians.

I’m finding it hard to find anything I didn’t like about the B.Beat, and the more I used it, the more I could see how it could fit into my own live performances, where it would not only replace my existing setup but also enhance it. £ Pro 16 £1062, Pro 16 Plus £1285.

Prices include VAT.

W www.m-live.com

www.soundonsound.com / May 2025

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Overloud TH-U2

Amp-modelling Plug-in

Overloud’s amp emulation has been treated to new features and a facelift.

PAUL WHITE

S

ome modelled amp plug-ins can sound lacking in low-end punch, but to me Overloud’s models come over sounding like a real amp miked in a real studio, and I’ve always held their TH-U amp and effects modelling packages to be up there with the best. The company employ some interesting technologies, the most intriguing of which is their Fluid IR approach to amp and cab modelling.

With well-modelled amps, the tone controls and gain structure can respond in a way that parallels the real hardware. It can be argued, though, that their weakness lies in their ability to accurately recreate the dynamic interactions of more complex components such as valves, transformers and loudspeakers. Amp ‘capturing’, on the other hand, is a different technology that delivers a very accurate and responsive version of an amp and speaker. Its own down side is that it’s really only accurate for the control settings at the time of capture, so you have less control than with modellers.

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Overloud’s answer to this conundrum is to combine elements of both technologies in their Fluid IRs: the idea is that the user gets an authentic amp emulation that responds correctly to control changes and to picking intensity.

Take Two In this new version of TH-U, Overloud have made significant changes to the user interface. This includes better visual representations of the kit that’s being emulated. That may not be a gamechanger in itself, but it does make the package more attractive, and Overloud have also made navigation even easier. For example, they’ve introduced multi-row model arrangements that reduce the need for horizontal scrolling when viewing the elements in a complex chain of pedals and amps. Parallel chains of amps and effects are easy to set up too, with drag-and-drop used to good effect. Overloud’s Unified Model Panel now integrates modelling, capturing and IR processing — with it all in one location, it’s easier to access what you need. Rigs and

IRs can be added directly to a sound chain from a library listing. Also under the heading of improving navigation, the SuperCabinet has been redesigned with the addition of a smart list and search function.

In terms of new features, there’s a new, really natural-sounding, MIDI-controllable, polyphonic transposer in the Master panel — this could be thought of as a bidirectional

Overloud TH-U2 €299 pros • Top‑tier amp and speaker emulation.

• Huge range of categorised presets.

• Improved layout and graphics.

• Clean polyphonic pitch transposer.

• Improved MIDI support.

cons • Only that I’d like to see a few more experimental effects.

summary TH-U was already good, but TH-U2 boasts lots of small improvements that add up to make it a very impressive amp and cabinet emulation package.

virtual capo, with a range from -5 to +3 semitones, in semitone steps. There have been stability improvements too, which will benefit those relying on TH-U2 for live performance. MIDI support has also been updated, for more extensive parameter control in both studio and live situations, with support for MIDI Bank Select messages in addition to Program Change messages for patch selection. Overloud have also introduced a number of new optional Signature Artist Packs, based on captures of the actual kit used by the artist.

With such a huge number of permutations of amps and effects to experiment with, it was no surprise to find an impressive library of presets to explore (there are over 1000, arranged into logical sections for ease of access), and apparently more free ones are offered every month. You favourite presets can now be tagged (for models, rigs, and IRs), and it’s possible to search for specific types by entering a name (such as Marshall) to see all the relevant presets. If the standard library still isn’t enough, you can buy optional add-ons, as well as the Signature packs.

We’re told that the available add-ons span over 32,000 captured rigs, and in excess of 42,000 Fluid IRs!

SuperCabinet For those unfamiliar with SuperCabinet, which is the Fluid IR engine used here, this can also be purchased separately. It leans on Overloud’s Fluid technology to enable

SuperCabinet, which is also available separately, provides access to a huge number of miked cabinets.

The impressive new transposer can be thought of as a virtual, bi-directional capo.

cabinet emulations to respond according to the level and dynamics of the input, which not only sounds more realistic than a static impulse response but, importantly, feels more natural to the player. TH-U2 comes with an impressively large library of IRs, as well as the ability for users to combine the characteristics of multiple IRs to create their own hybrid speaker cabs and stereo speaker setups. The editing tools include a multiband mode and the ability to export a cabinet response as a conventional WAV format IR to load into third-party modelling hardware or software.

To put some numbers on what you get in this latest version, there are 248 so-called Premium Models, based on Fluid modelling technology. These comprise 89 guitar amplifiers, four bass amps, 50 guitar cabinets and two bass cabs. In the effects department, there are 85 pedal and rack effects. When setting up cabs, there are also 18 mic models, with placement options.

SuperCabinet is backed up by over 1728 included Fluid IRs ,while Rig Player has 20 captured rigs in which the Fluid IR process has been used to capture the whole signal

chain as a single entity. The sound quality, whether clean, hairy or dirty, is excellent, with good touch responsiveness, and there’s a practical range of familiar effect types, both pedal and rack.

A mobile device version based on the same processing framework as the computer version offers 248 premium models that still include the fastidiously modelled Centaur overdrive found in the computer plug-in. There are plenty of effects and officially licensed amp models from Randall, THD, DV Mark and Brunetti.

Lasting Impression There’s a lot here for new users to get their teeth into, then. The combination of modelling and Fluid IRs works well, and there’s definitely enough choice that there’s something to suit everyone. If I have any criticism at all, it’s that there’s relatively little by way of adventurous ambient effects, other than the usual reverse delays and shimmers — I’d welcome the addition of a few granular delays, for example. Still, in your DAW, there’s nothing to stop you adding your own third-party plug-ins to the chain, and if you needed to put them midway through an effects chain, you could always open up two instances of TH-U2, with half the chain on one and the other half on the other with your own choice of plug-in between the two.

For existing TH-U users, most of the updates in TH-U2 relate to the visuals and workflow, but they’re worthwhile improvements, and there are some worthy new features too. There are a number of new presets, for example, and the new polyphonic transposer sounds really smooth and natural, whether dropping down to baritone or capo-ing up. The improved MIDI support is also welcome, and all the smaller tweaks add up to an overall improvement in the user experience. And existing users can upgrade for only $29 — the polyphonic transposer has to be worth more than that alone! £ €299. Discounted to €179 when going

to press. Upgrade from TH-U €29.

Prices include VAT. W www.overloud.com

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Orchestral Tools Berlin Solo Strings Sample Library

The latest addition to Orchestral Tools’ Berlin series shines the spotlight on the soloists.

S O N A L D ’S I LVA

S

tring sample library veterans Orchestral Tools add to their flagship Berlin Series with Berlin Solo Strings (BSS for short). Comprising four soloists (cello, viola, and two violins), BSS was recorded, as with the other instruments in the series, at the renowned Teldex Scoring Stage in Berlin.

The question a newbie might immediately ask is: ‘Isn’t Berlin Strings First Chairs also a solo instruments library?’

Yes, it is. So how is this different? The short answer: greater number of articulations; no double bass; and the instruments in BSS are designed to perform solo parts, while First Chairs is meant primarily for layering with larger ensembles, although it also works as a string quintet.

The other question that comes to mind, for those of us that like having a manual to refer to while exploring a new library, is: ‘Where is it?’ The bad news is that there doesn’t appear to be one; the good news is that you can cobble one together from the Helpdesk section of the OT website. Download the PDFs that explain each section of OT’s Sine player and you have everything you need to get started. Speaking of Sine Player, BSS runs in the latest version, either standalone or in a DAW, in VST, VST3, AU and AAX formats, with 24-bit/48kHz patches and a total file size of 168GB of samples (68GB compressed SINEarc format).

Overview The patches for each instrument come in three styles: Expressive, Cinematic and Sul Tasto, each designed for a specific mood, ranging from in-the-spotlight leads to in-the-background accompaniments.

Articulations are grouped into two categories: Longs and Shorts (sustains, marcatos, staccato, spiccato, pizzicato); and Dynamics and Ornaments (swells, repetitions, tremolo, trills). For added texture we have harmonics and sul ponticello stylings.

With one exception — only Violin I has marcato harmonics — all the instruments contain the same articulations, in the same order. This not only makes things

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The legatos lie in different ‘speed zones’ and are triggered automatically by speed of playing.

well organised for keyswitching, but also lets you quickly audition parts on different instruments. For instance, if you want to hear what it’s like to have the cello play the top line instead of a violin, it’s an easy switch (and back, if you don’t like it).

Legato & Sustains BSS puts a special focus on legato and equips each of the three styles listed above with its own legato approach. As a result, the slurs, runs and portamento for Expressive, Cinematic, and Sul Tasto blend with the tone of the sample being played.

During my early experiments with the library, I found that some transitions sounded louder than others unless I focused on hitting the key with exactly the right amount of pressure. This was fixed by tweaking the legato volume slider, which smoothed the transitions across the board.

Legato is available for most articulations where, to quote OT, “it makes sense” (so don’t go looking for it in the spiccato patch). It is automatically active for the most part, but in some cases you have to manually turn it on, like with the Expressive Sustains Evolving Vib patch in Violin I, whose legato remains inactive until the option is selected in the menu on the bottom right. The sustains come in the following variations: immediate, accented or soft attack, and evolving vibrato, all of which are distinct enough that you can easily hear the difference.

range to make use of the four recorded dynamic layers (pp, mp, f, ff) and you can have fun playing expressive, sharp, stabbing Hermannesque chords.

Swells (especially the Mid and Short), lend themselves to some lovely minor and diminished chords, and by ‘lovely’ I mean appropriately tense and foreboding.

Combine a Cm or Cdim on the Violin II Swells Short with effects like Logic’s Creeping Filter in Delay Designer, for an impactful horror film trailer moment.

Repetitions, with the three reps variation, is especially dramatic and brings to mind a chugging string section à la the finale of ‘November Rain’ after everything is swept away in a flood and even the roses don’t make it. I can almost hear Slash’s wailing guitar solo over the part I hurriedly put together. Top notch.

When I first explored the staccato patches, I jabbed away at the keyboard enthusiastically, but it didn’t seem to make a difference to the dynamics — all the notes sounded at the same intensity. They responded to key velocity when I made adjustments to the volume range slider and turned it all the way up for full dynamic range. I don’t know if this is the case for all composers, but, especially with staccato parts, it feels like their intensity needs to mirror how hard or softly you hit the keys, or else you might as well program them in, instead of playing the part.

Dynamics & Vibrato A dedicated Dynamics tab for each patch contains the following controls: Niente, Soft Low Layer (makes volume changes to the

Orchestral Tools Berlin Solo Strings €399 pros • Shorts patches perform energetically across all instruments.

• Viola and cello sound rich and warm.

• Clear and easy-to-use GUI. • Compatible with larger Berlin Series ensembles.

cons • No manual.

• Some audible artefacts.

summary Berlin Solo Strings brings four versatile soloists into your DAW, up to 27 articulations per instrument, three distinct playing styles and seven mic positions (plus one extra for cello), giving you more than enough control to make them shine on their own, or work together as a solid string quartet.

lowest dynamic layer first before fading in the next layer), and Cut Off Filter. When Niente is enabled, “the lowest dynamic setting (usually CC1 or key velocity at 0) will cause the instrument to be completely silent. Use this setting when using your instruments with a breath or wind controller”. It worked well on a patch like FX Sustain Harmonics, where you turn Niente on and use CC1 to gradually dial in the note, effectively heightening the suspense and letting the tension of the harmonic slowly drip into your piece.

Vibrato has an outsized effect on expression: too much and it sounds overwrought, too little and it sounds

Other Articulations While several BSS demos showcase soaring lyrical lines and mellow melodic phrases, I found the library particularly well suited to playing energetic, rhythmic pieces — using more of the Shorts articulations, in effect.

Drama, thriller, horror and other genres that draw from the darker side of the musical palette will be well served by BSS. Take the Violin Marcato Longs, for example: turn the volume slider up to full

You can work with multiple articulations per instrument in Poly mode, switching between them either

by MIDI CC or keyboard velocity.

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flat. While BSS does offer variations in vibrato, there is no way to control it via CC. According to OT: “The idea is that you decide how much vibrato (or ‘passione’) you want, and then use the appropriate mood/voice. ie. Expressive, Cinematic, and Sul Tasto. It’s that way by design with the overall goal of delivering the most realistic sound possible.” With a virtual instrument, it isn’t even about realism at this point, it’s about whether the existing vibrato matches the feeling you’re going for. I decided to build my pieces around the vibrato available. No use micromanaging; might as well work with what you have and what you have with BSS is a lot.

Mic Positions Each instrument offers seven mic positions to choose from, ranging from the spot mics for the driest and most direct sound; the standard Tree array; to Immersive F (front overhead speakers in a 7.1.4 setup) and M (back overhead speakers for 7.1.4). Most of the instruments were recorded with two spot mics: Spot 1 has the open sound of a large‑diaphragm capacitor mic, while Spot 2 brings the warmth of a ribbon mic. The cello was recorded with a third spot mic, “placed in front of the bridge to get some extra bite”, according to the literature.

Here’s where personal preference really kicks in. In a library featuring soloists, you do want detail; you want the parts to stand out, so a certain brightness to the sound should be acceptable. Some people might be okay with the tone of the library as is, but to my ears, it was a little too bright. Could it intentionally be this way to help it cut through a mix? That is definitely a possibility.

The spot mic samples on their own were direct and clear as promised, with the down side being that playing noises and, in some cases, artefacts are fairly audible, becoming even more pronounced when you turn up the volume on quieter patches like Sul Tasto. I stuck with a mix of the spot and the Tree to soften things up.

Once you dial in the levels on your selected mic positions and achieve a sound you like, you can use the Mic Merge feature to merge the mix down to a single, new stereo position, thereby reducing the load on system resources.

The Instruments Violins I & II The sound of the violins is a matter of taste, and right out of the box, the higher registers are bright bordering

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Mic positions can be downloaded individually — great for disk space, and since the mic positions are independent, deleting one after the fact doesn’t affect the others.

on shrill for me. However, it’s nothing that some mic selection and EQ can’t fix.

Add a hint of reverb and the sharpness disappears entirely.

Like with the other instruments, the Expressive patches are significantly louder than the Cinematic and Sul Tasto.

I would have liked a more normalised sound across patches, but I wonder if, again, that’s a matter of personal preference. Sul Tasto, by its very nature, is meant to be quiet and mellow, but the difference was a little startling the first time I heard it.

There are some inconsistencies.

In the Violin I Spiccato patch, some notes are crunchier than others. The sound — like a scratchy click — might be a playing noise, and is most heard on B2, C3 and C#3. Because of the round robin, it doesn’t appear on every instance of the note if you repeat it, but it is definitely present. I consider these inconsistencies to be quirks of the library, and choose to lean into them to make the performance more human.

The style that stands out among the violin patches is Cinematic Sustains Evolving Vibrato. The performance you can put together is so moving, even mediocre keyboard skills can’t get in the way of the pure emotion it can generate. As a test, play something like the melody of the old folk song ‘Red River Valley’. Goosebumps!

Viola The viola has a beautiful rich tone and does well in lyrical, emotional passages as well as tense, jittery ones. The Cinematic

Sustains patch is particularly beautiful and I can imagine it playing some wonderfully strong, supportive middle lines.

Cello The cello does a stellar job of adding gravitas and warmth to the proceedings and it really shines when the melodies need an extra hit of emotion. The spot mic on the cello sounds the best of all the spots in the library; leaving it in the mix allows you to hear how the body of the specific instrument used shapes the tone of the sound. There are some oddities though, like the Sul Tasto Sustains Soft patch, which contains almost brush-like breathy attacks on several notes.

Conclusion BSS is an ambitious library, aiming to provide an arsenal of solo string playing styles for composers to write with. I have come to think of working with recorded sample libraries as a form of remixing — you’re working with samples, not a live player, so get ready to roll up your sleeves and do some editing after laying down your composition. As long as the recordings are good quality, and the tone of the instruments is evocative (and mostly close to the real thing), I prefer to treat timing issues, minor glitches and performance inconsistencies as things to be fixed in post. BSS isn’t perfect, but it provides you with tools that span a vast emotional range — from tenderness to terror. You can do a lot with that. £ €399 including VAT. W www.orchestraltools.com

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ON TE ST

Maono PM500 Cardioid Capacitor Microphone Taking their name from the Swahili word for ‘vision’, Maono describe themselves as a “global bestselling Internet microphone brand”, and offer a range of keenly priced mics and accessories. Unsurprisingly, many of these are USB models aimed at podcasters and vloggers, but they also manufacture mics that are studio-friendly as well as wallet-friendly.

Their flagship studio model is the PM500, a fixed-cardioid capacitor mic that retails for the princely sum of $149.99 and comes with an impressive range of accessories.

The PM500 has a businesslike rectangular body that recalls some of the Lewitt models. Build quality seems very decent given the low cost: the headbasket is robustly put together and reasonably well damped.

The aforementioned accessories include a pretty substantial shockmount, which does a good job of holding the mic at your chosen angle but is only moderately effective at isolating it from stand-borne

Minimal Audio Wave Shifter Modulation plug-in Offering modulation options that are rather different from the norm, and available in all the common macOS/ Windows plug-in formats, Minimal Audio’s Wave Shifter employs frequency shifting, ring modulation, and amp modulation. The last of those is described as being similar to ring modulation but also including the original source audio frequencies. These allow treating the input signal in both musical and inharmonic ways. The amount of

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noise. A smart clip-on pop shield attaches to the front of this, and you also get a 2.2m mic cable and a chunky desktop stand with a round metal base.

The specifications don’t make clear whether the PM500 is an externally polarised ‘true’ capacitor mic or an electret but, either way, it needs 48V phantom power to operate. It uses a single-sided, one-inch capsule of a type I’ve not seen before, with a black plastic surround, and is presumably transformerless, though this is again not stated. Sensitivity is a comfortable 18mV/Pa or thereabouts, and a self-noise figure is given as 20 microvolts RMS; this is not directly comparable with the conventional measurement in dBA, but the mic seemed quiet enough to me.

When affordable Chinese-made capacitor mics began to appear in the ’90s, many of them shared a characteristically over-bright, edgy quality, typically because a K67-type capsule was used without any de-emphasis in the electronics. I was half expecting the PM500 to have a similar sonic footprint,

shift in frequency shift mode is applied to the whole audio spectrum, while ring modulation produces only sum and difference frequencies.

The frequency shifting process shares common ground with FM synthesis, with controls relating to frequency modulation, ratio and soft sync controlled by a modulation section that can be free‑running or sync’ed to tempo. The modulation LFO can be set to sine, triangle, ramp or square shapes, while a Randomise control modifies the modulation and an Offset control adjusts the relative phase of the left and right outputs. The Shape control morphs between the LFO wave shapes, unless Snap is enabled, in which case they change abruptly as the Shape knob is turned.

“It’s actually a surprisingly soft-sounding, smooth mic... If you’re afflicted with a strident, harsh or sibilant voice, it could be just the ticket” but I couldn’t have been more wrong.

It’s actually a surprisingly soft-sounding, smooth mic, and in comparison with staples like the Neumann U87 and AKG C414B-ULS, has a noticeably less forward upper midrange.

Whether this is a good thing depends on what you’re recording with it. It wasn’t a good match for my own voice, which benefits from a bit of additional ‘bite’ somewhere in the not-many kilohertz region, but if you’re afflicted with a strident, harsh or sibilant voice, it could be just the ticket. Equally, it might be a good choice for honky guitar amps and the like, though in practice I preferred my U87 in a side-by-side comparison on my Fender Vibrochamp. The PM500 also responds pretty well to EQ, so all in all, it makes for a versatile and very inexpensive addition to any setup.

Sam Inglis £ $149.99 W www.maono.com

The FM section offers a choice of Sine, Even, Dirty, Noise, Input, Output or External as modulation sources, and rotary controls for Ratio and Amount.

The large knob in the centre sets the oscillator frequency for the mode (frequency shifting, ring modulator or amp) that’s set using the Style box in

the top right‑hand corner of the window.

In frequency shifting mode, there are clear positive or negative shifts and the frequency can be switched to Fast, Tuned, Slow or Sync modes. In Tuned mode the frequency can be set via MIDI notes. When soft sync is on, the oscillator reacts to the input signal polarity, which adds an interesting harmonic timbre.

To my ears, this produces a smoother sound at low modulation frequencies but it can be more obvious at higher modulation rates. Spread introduces a difference in the relative frequencies of the internal oscillators, to create stereo shifting effects.

As you might expect, the smoothest modulation effects are created when the Frequency control is set to Slow, with a rate of only a few Hz. Note that the Frequency control can introduce both positive and negative frequency shifts. High rates create screeching FM‑type sounds that can work well on rhythmic sounds when controlled via the LFO section. The difference between frequency shift and FM is subjectively very noticeable so it comes down to picking the mode that works best for the current material.

Feedback Amount adjusts the level of the shifted output signal that’s fed back into the input, and at high values self‑oscillation can occur in some instances. The feedback can also be sync’ed to the host tempo, and a variable band‑pass filter shapes the signal being fed back. There’s also the expected wet/ dry balance control, input gain and the option to apply input gain only to the affected signal. The output gain control is accompanied by a limiter with soft clip or gentle limit/saturation options.

Having experimented with the plug-in, my own view is that the more subtle modulations available here can work well on bass and pad sounds, while the more assertive treatments, especially if using a sync’ed modulation waveform, are a good fit for adding impact to rhythm loops or for turning static synth/guitar sounds into rhythmic elements. As with any FM or frequency shifting device, there’s plenty of scope for producing inharmonic chaos, so experimentation is the key. There’s a lot to like about Wave Shifter, especially if your music has an experimental edge. Paul White £ $49 (discounted to $34 when

going to press).

W www.minimal.audio

Phantom Sounds Analog Drift Delay Plug-in Yet another delay plug-in — just what the world needs, right? While you’d be forgiven for thinking that, a few minutes’ experimentation with Analog Drift might well change your mind, and that’s something you can do using the 14-day free trial. Supporting AU, VST3 and AAX platforms and macOS or Windows hosts, Analog Drift is designed with a clear layout and a very manageable number of controls. It offers three echo modes: Glass Delay is the most conventional, being based on a clean digital delay; the delays can be set to Ping Pong mode; and there’s also a Mid‑Sides mode that applies different delays to the Mid and Sides signals when instantiated as a stereo plug-in.

Once you get beyond the familiar delay time and feedback controls, which are independent for each channel and have tempo sync and linking options, further stages can be used to sculpt the delay in a number of ways. The sliding high- and low-cut filters operate on the delayed signal in the usual way, but we then get to a Tape section, with separate controls for wow and flutter plus a Wear control that blurs the sound to emulate a worn tape. A further control, Dust, adds what sounds like vinyl crackle and, though that’s not something you’d expect to hear from a tape machine, it is certainly a useful addition for creating lo-fi effects. Finally, a Drive section has two controls: Bias and Density. Bias degrades the clarity of the delayed signal (the highest settings are the cleanest).

Density adjusts the harmonic content and goes from a gentle warmth to the sound of overdriven tape. A Reverb knob gives the delays a denser, more reverb-like character, while a Ducker allows the delays to swell during pauses between sounds but drops their level when the dry sound is present. There are separate level controls for the in and out levels, plus the usual wet/dry Mix control.

So far so good, but that’s still nothing too far out of the ordinary. However, switching to Prism mode takes you closer to special effects territory. I’m not

entirely sure what happens behind the scenes in Prism mode (it sounds like the feedback increases, dub-style, combined with modulated low-cut filtering), but the manual describes it as “a dynamic, shimmering delay that brings vibrant and ethereal soundscapes to life with subtle movement”. I tried this with a few widely spaced guitar chords and, when setting the Ducker around half way, was greeted by what I can only describe as a granular shimmer that swells in level between sounds. The timbre of the effect can be fine-tuned using the Feedback, Filter and Wear controls as well as the Ducker, and with even modest Feedback settings I found the shimmer could continue indefinitely during pauses. By the way, the sustained delay tail is also affected by the Tape controls.

Finally there’s the Vaporwave Bliss mode, which comes complete with another poetic description: “Dreamy, lo-fi echoes infused with vintage warmth and gentle pitch modulation for a nostalgic vibe.” This mode does pretty much what it says on the tin, offering a warmer, somewhat more vintage character compared with the Glass Delay mode, and turning up the Reverb knob makes it even dreamier.

Describing something as sounding ‘musical’ is a somewhat overused adjective, but in this case it’s absolutely apt. Analog Drift is easy to use, and can handle all the usual delay treatments, but it adds a few surprises of its own that add greatly to its creative potential and have already earned it a place in my ‘favourites’ plug-in folder.

Paul White £ €79 (discounted to €59 when going to press).

W https://phantom-sounds.com

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Sontronics Apollo 2

Stereo Mics There’s a huge array of dedicated stereo mics on the market. We round up some of the best current options.

LUKE WOOD

A

t first thought, stereo microphones might seem like a solution looking for a problem. Setting up a pair of ‘normal’ mics for a stereo recording isn’t a particularly difficult task, after all. However, an all-in-one stereo mic actually offers a number of significant benefits. Firstly, two capsules housed in a single assembly can be placed closer together than is often possible with a pair of individual mics, minimising time-of-arrival differences and resulting in more accurate stereo imaging.

There are practical benefits, too: rigging a single mic on one stand is quicker and easier than configuring a stereo bar or wrestling with a pair of stands, for example.

It’s also easy to move a single mic around in search of the best possible placement, something which would be very awkward to attempt with a pair of large mics set up on two stands! There are plenty of stereo mics on the market, and although lots of them are boom- or camera-mounted types aimed at production sound duties, there’s a healthy selection of studio-focused models, too. So, let’s take a look at what’s on offer.

AEA Nuvo N28, R88 MkII & R88A AEA’s first offering comes in the form of the Nuvo N28, a phantom-powered mic that houses a pair of matched ribbon transducers in a fixed AEA Nuvo N28

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Blumlein configuration. With a mid-forward frequency response, precise stereo imaging and ‘tamed’ proximity effect, the mic is said to excel at everything from close-miking strings to drum overheads or room-mic duties, while its compact design makes it easier to position. Next up is the larger R88 MkII, which is said to be a stereo version of AEA’s N8. Featuring two of the AEA R88 MkII company’s Big Ribbon transducers in a Blumlein configuration, the mic has been designed with far-field applications in mind, and will retain a natural-sounding frequency response even when used at a distance. Like its mono counterpart, the R88 MkII boasts the widest frequency response of any AEA mic, and promises to effortlessly capture complex and powerful sources such as drums, strings and pianos. There’s also the R88A, a phantom-powered version that offers an additional 12dB at the output. £ Nuvo N28 £2499; R88 MkII £2199;

a precise polar pattern across the mic’s full frequency range, while transformerless circuitry virtually eliminates low-frequency distortion. A three-way selector switch offers a choice of three operation modes: Stereo 90, Stereo 127 and M‑S. The first two modes offer a traditional left/right output courtesy of an onboard Mid-Sides matrix, with the option of selecting either 90- or 127-degree stereo acceptance angles; setting the switch to M‑S mode will deliver discrete Mid and Sides signals at the mic’s outputs, making it possible to adjust the stereo image in post-production. The AT4050ST also features a -10dB pad and an 80Hz high-pass filter. £ £1319 including VAT. W www.soundonsound.com/reviews/ audio-technica-at4050-st

W www.audio-technica.com

Avantone CK-40

The Avantone CK-40 employs a pair of 35mm pressure-gradient transducers that promise to deliver R88A £2659. Prices include VAT. accurate stereo W aearibbonmics.com imaging without suffering from proximity-based phase issues. Both Audio-Technica’s AT4050ST features capsules can be independent cardioid and figure-8 independently capsules configured in a Mid-Sides Avantone CK-40 switched between arrangement, and mounted in cardioid, omni and figure-8 polar a robust housing that has been patterns, and there’s an 80Hz high-pass designed to minimise internal reflections.

The company say that the Audio-Technica filter along with a -10dB pad. The upper capsule and grille assembly dual-diaphragm capsules maintain AT4050ST

Audio-Technica AT4050ST

can be rotated by up to 90 degrees, which, combined with the per-capsule pattern switching, means the mic can be quickly and easily configured in X-Y, Mid-Sides and Blumlein arrangements. The CK-40 features quiet Class-A FET circuitry, and boasts a full-range response that remains within ±3dB between 20Hz and 20kHz. £ £749 including VAT. W www.soundonsound.com/reviews/

Extinct Audio Valkyr Stereo

avantone-ck40

W www.avantonepro.com

Brauner VM1S

Brauner are well known for the quality of their high-end mics, and the stereo VM1S is no exception. It essentially places two of the company’s renowned VM1 mics in a single housing, with precisely matched capsules promising to deliver everything you could wish for in a stereo microphone! Both capsules boast a continuously variable control that allows users to move seamlessly from Brauner VM1S omni, through cardioid to figure-8 polar patterns, while a moving grille assembly allows the top capsule to be rotated through up to 90 degrees to accomodate a range of coincident stereo techniques. Like its mono sibling, the VM1S delivers a natural sound that’s capable of tackling everything from soloists to ensembles and choirs. £ £9695 including VAT. W brauner-microphones.de

Coles 4050

The Coles 4050 is a bit of an outlier in that it’s technically a plain old mono ribbon mic.

However, thanks to a clever magnetic mounting system, it’s possible to join two 4050s together to form what’s essentially a single stereo mic.

Once combined, the mount allows the mics Coles 4050 to be rotated through 360 degrees, meaning that any mutual angle can be accommodated.

The mics offer the ‘smooth’ sound for which Coles are renowned, boasting a frequency response that is “exceptionally flat” between 30Hz and 20kHz, and a polar pattern that

remains constant in both the horizontal and vertical planes. £ £1859 including VAT. W coleselectroacoustics.com

Extinct Audio Valkyr Stereo

Offering a stereo alternative to Extinct Audio’s BM9 ribbon mic, the Valkyr Stereo is said to be the perfect one-point solution for capturing ensemble, orchestral and choral performances, as well as being a great choice for drum overhead or room-miking tasks. Its ribbon transducers are closely matched for frequency response, resulting in an accurate, detailed stereo image that boasts impressive depth and positional accuracy. A fixed 90-degree angle between the two capsules allows the mic to offer Blumlein and Mid-Sides arrangements, and a wide stereo acceptance angle means that the Valkyr Stereo is easily capable of capturing a large band or ensemble. £ £1799 including VAT. W www.soundonsound.com/reviews/ extinct-audio-bmx2-black-ops

W www.extinctaudio.co.uk

Golden Age Project R1 ST

The R1 ST from Golden Age Project has been designed to offer the smooth and natural sound associated with vintage ribbon mics, and promises to capture transient details without any harshness. Stereo capture is provided by a pair of 50mm ribbons in a fixed

Blumlein configuration, while a transformer-balanced output stage helps to deliver a warm and punchy sound. Golden Age Project say that the R1 ST is well suited to a wide range of recording applications, and will happily tackle everything from vocals and solo acoustic instruments through to large ensembles and choral groups. £ £529 including VAT. W www.soundonsound. com/reviews/ golden-age-project-r1st W www.goldenageaudio.com

Golden Age Project R1 ST

Josephson C700S

All of the mics in Josephson’s Series Seven line-up feature multiple capsules with independent outputs, meaning users are able to alter their polar pattern after recording.

The stereo C700S takes things a step further, adding a side-facing figure-8 capsule to the forward-facing omni and figure-8 elements found in the C700A; by recording each signal individually, users are

Josephson C700S

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provided with a huge range of control over the stereo image of their source material at the mixing stage. As you’d expect from Josephson, the C700S is built to exacting standards, with each capsule feeding its own Class-A output circuitry and internal shockmounting negating the need for any additional mounting hardware. £ £7999 including VAT. W www.josephson.com

specifications. And much like the U87 Ai, its suggested range of applications is vast, with Neumann recommending it for everything from vocals and solo instruments to large ensembles, drum overheads, and just about anything else in between! £ £5460 including VAT. W www.neumann.com

a pair of long ribbons in a single body, with the top element capable of rotating through 90 degrees. In terms of sound, it draws its inspiration from early RCA mics, delivering a smooth, natural sound that responds well to EQ and offers an alternative ‘flavour’ to capacitor-based models. £ P‑24 £3999, SR14 £1279.

Pearl DS 70

W www.pelusomicrophonelab.com

Mesanovic Model 2S

Building on the capabilities of the DS 60, Pearl’s DS 70 benefits from an upgraded body designed to minimise internal reflections, improved wiring, and an updated capsule arrangement that improves the phase relationship between the mic’s two capsules. It places a pair of the company’s 2800 dual-membrane capsules at a 90-degree angle, and thanks to individual outputs for both sides of each capsule, the Pearl DS 70 mic is capable of forming Mid-Sides, X-Y and Blumlein arrays. Yet more options can be accessed by recording each output separately, which makes it possible to adjust the DS 70’s polar patterns and stereo configuration in post-production. £ $4481 W www.milabmic.com

The Mesanovic Model 2S essentially places two of the company’s Model 2 ribbon mics into a single body, housing a pair of matched ribbons in a Blumlein configuration. The sound character remains the same as its mono sibling, combining the smooth midrange response associated with ribbon mics with an extended, detailed high end and a fast transient response. Mesanovic say that the mic maintains a uniform frequency response both on- and off-axis, and will deliver lifelike stereo imaging that results Mesanovic in recordings that make you Model 2S feel like you’re in the room with the performers. As well as being recommended for orchestral and choral sessions, the Model 2S promises great results in a wide range of applications from drum overheads and room miking to solo acoustic instruments. £ £2099 including VAT. W www.mesanovicmicrophones.com

Neumann USM 69 i

The USM 69 i has been designed as an ‘all-rounder’ microphone that’s capable of covering a range of coincident stereo miking techniques. It features two of the large-diaphragm capsules used in Neumann’s legendary U87 Ai, and thanks to individual polar pattern selection and a 270-degree-rotatable body surrounding the top capsule, can be configured to provide X-Y, Mid-Sides and Blumlein arrays.

As you’d expect from Neumann, the USM 69 i is a high-quality affair that employs premium components throughout and boasts excellent technical

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Peluso P‑24 & SR14

There are two stereo offerings in the Peluso range: the P‑24 and the SR14. The former is a valve capacitor mic inspired by AKG’s C24, and employs a pair of Peluso’s P-CK-12 elements — capsules which the company say are the most accurate recreation of AKG’s CK12 in production today — along with custom output transformers and a 6072AM triode valve.

Both capsules have nine independently selectable polar patterns, and the upper grille assembly can be rotated through 180 degrees, allowing the mic to cover X-Y, Mid-Sides and Blumlein techniques. The SR14 then offers a stereo version of the R14, housing Peluso P‑24

Prices include VAT.

Rode NT4

All of the options we’ve looked at so far are either large-diaphragm capacitor or ribbon designs, but Rode’s NT4 takes the small-diaphragm capacitor route. It features a matched pair of half-inch cardioid capsules fixed at a 90-degree angle, offering an all-in-one solution for X-Y stereo recording. It’s said to be an ideal choice for a wide range of applications including drum overheads and percussion, choir vocals, environmental recording and sound design, either in the Rode NT4 studio or out in the field. For those looking to record on location, the NT4 can be powered from a 9V battery instead of phantom power, and comes with both XLR and 3.5mm mini-jack cables for compatibility with portable recorders and cameras. £ £479 including VAT.

W www.rode.com

Peluso SR14

Royer Labs SF-12, SF-24 & SF-24V

Royer are a popular name in the world of ribbon mics, and their current range includes three stereo models. The first, the SF-12, places a matched pair of ribbons in a Blumlein configuration, with their controlled frequency response and polar patterns helping to minimise coloration of off-axis sources. The mic is said to be a great choice for everything Royer Labs from drum overhead SF-12 duties to capturing whole ensembles — the Royer website features a demo recording of an entire orchestra captured with a single SF-12! The SF-24 is a phantom-powered version of the SF-12, with a 14dB increase in sensitivity helping the mic to provide an output that’s more in line with a capacitor-based mic.

Finally, there’s the flagship SF-24V, a version of the SF-24 that’s equipped with the same valve-based circuitry found in Royer’s award-winning R-122V. £ SF-12 £2730, SF-24 Royer Labs SF-24 £4075, SF-24V £5125.

Prices include VAT.

W www.soundonsound.com/reviews/ royer-labs-sf24v

W www.royerlabs.com

Samar Audio AL959, VL373 & VL373A

Designed by Samar Audio founder Dr Mark Fouxman, the AL959 promises to couple high-end performance with a price tag that’s unheard of for a US-made stereo ribbon mic.

It features a pair of the company’s ‘piston’ corrugated ribbons, which are said to offer lower noise, a higher output, higher SPL handling and a fuller, more controlled bass response than their traditionally made counterparts. The ribbons are mounted in a Blumlein configuration, and the company say that the mic’s modern design and linear response make it a great choice for everything from orchestral and choir recordings to smaller pop, rock and jazz Samar ensembles. The VL373 then Audio AL959

takes the motor structure and tuning of the company’s flagship MF65, and features a rotating top assembly that allows the angle between the two ribbons to be set between 60 and 120 degrees to facilitate a greater choice of stereo configurations. There’s also the VL373A, a phantom-powered version that boasts a hotter output and slightly lower self-noise. £ AL959 $1299, VL373 $2199, VL373A $2399 W www.samaraudiodesign.com

boasts a phantom-powered design that delivers capacitor-like sensitivity and low self-noise figures. Its frequency response extends from 20Hz to 15kHz, and Sontronics say that its accurate, expansive stereo image makes it a great choice for capturing a huge range of instrument and vocal ensembles. £ £2249 including VAT. W www.sontronics.com

Vanguard V44S Gen2 & V24 Samar Audio VL373

Schoeps MSTC 74

Schoeps MSTC 74

For obvious reasons, the vast majority of stereo mics tend to have their capsules arranged in coincident configurations.

The sole stereo offering in the Schoeps line-up, though, takes a slightly different approach, offering an all-in-one solution for ORTF recordings. The rather unusual-looking MSTC 74 employs a pair of the company’s MK4 caridoid capsules with the 170mm spacing and 110-degree angle required by the near-coincident technique. The fixed design makes it possible to rig a perfect ORTF array quickly using a single stand, with no additional mounting hardware required, while Schoeps’ renowned capsules help the MSTC 74 deliver a natural sound with great sound localisation and spatial qualities. £ £3189 including VAT. W www.schoeps.de

Sontronics Apollo 2

The latest iteration of Sontronics’ stereo ribbon mic has been designed to strike a balance between the warm, smooth characteristics of vintage ribbons with the clarity and detail that’s expected of modern mics. The Apollo 2 employs a matched pair of the company’s hand-tuned ribbon elementsina fixedBlumlein configuration, and Sontronics Apollo 2

The second-generation version of Vanguard’s FET-based V44S builds on the success of the original, introducing a redesigned capsule assembly that rotates through 120 degrees with detents at every 15 degrees.

Thanks to independently selectable omni, cardioid and figure-8 patterns for each capsule, the V44S Gen2 allows for a variety of configurations including X-Y, Mid-Sides and Blumlein — the included splitter box offers an additional polarity-reversed output from the top capsule, making it simpler to capture Mid-Sides recordings.

A newer addition to the range is the flagship V24, a vintage-style valve mic built around a pair of BeesNeez CK12 capsules that each offer a choice Vanguard of nine polar patterns.

V44S Gen2 If you’re keen to find out more, then you’re in luck, as the V24 is reviewed in this very issue! £ V44S £1478, V24 £9400.

Prices include VAT.

W www.soundonsound.

com/reviews/ vanguard-audio-v44s-gen2 W www.soundonsound. com/reviews/ vanguard-audio-v24 W www.vanguardaudiolabs. com

Regrettably, in last month’s Spotlight we neglected to mention Yorkville, whose portable PA range Vanguard V24 includes no fewer than nine battery-powered models! We’ve updated the article on the SOS website, so do take a look if you’re in the market for a mains-free gigging system.

www.soundonsound.com / May 2025

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Bunker Samples Iremia Intimate Cello & Intimate Viola Kontakt Instrument

HHHH With the Iremia series, Bunker Samples get quiet, intimate, and very close to a solo viola and cello. Running in the full version of Kontakt 6.7.1 or later, and each being about 8.5GB, the instruments come with matching articulations presented in five patches. Within the main patches are nine sustain articulations: normale, muted (CS) and super muted (recorded using a big rubber practice mute), each with ppp, pp and p dynamics. These sustains have some lovely, delicate movement to them, which is great when holding long chords, and the loudest articulations, while still quiet, can actually sound very intense, especially with the close mics.

The most exciting part of the libraries are the additional textures — these sounds are an inspiration engine. The bow and vibrato pulses sound amazing layered with normal sustains, I got great results adding them to another library’s string ensemble patch to add some movement. They feel really vivid and alive. I also had great fun using Logic’s articulation sets to trigger different sounds via velocity. Improvising with these varied textures is an amazing shortcut into slow, emotional string writing. The random rebow, tremolo bursts and fluctuating trem are full of drama. These are best used sparingly, and there are some human imperfections which make these articulations feel beautifully real.

These same samples are also presented in additional X-fade patches. While in the main patch you’re able to mould simulated dynamics within the ppp sustain, for example, in the dynamic sustains patch you can move between the ppp, pp and p layers.

The crossfading is sometimes noticeable as these are solo instruments, however, use in a non-exposed way they would be fine. Similarly, the CS to normale patches allow crossfading between the muted and standard sound. This sounds particularly lovely on the viola vibrato pulses, but my favourite sounds here are the super CS to CS sustains, especially on the ppp. Being able to move with this subtlety between the different levels of muting allows for some wonderfully emotional dynamic movement.

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The Swells patches offer 1, 2, 3 and 4 second swells, in normale, CS and tremolo CS variants. These are easy to get lost in, and the time machine versions allow you to compress/expand the speed, though I would have liked a tempo sync option here as well.

The different microphones give a wide range of perspective, but the far mic is still pretty dry, which is good for flexibility.

The inside mic gives a lovely warmth to the cello, and each of the mics can be independently panned, stereo collapsed (where applicable) and sent to the built‑in delay and reverb, which is useful.

With their vivid, human and intimate textures, the Intimate Viola and Cello are certain to provide inspiration. Thomas Field $99 each. www.bunkersamples.com

Song Athletics Bones WAV Audio

HHHHH Bones, from British creative studio Song Athletics, is a ‘super sample pack’ containing over 400 loops, 800+ one‑shot samples and more than 100 drum stems, performed by professional musicians and recorded at studio-grade quality. The royalty-free collection (5.73GB uncompressed) is provided as WAV files and can be used easily in a variety of software.

There are variations on a large number of parameters: dynamics (soft, hard, medium); tempo (loops ranging from 45 to 155 bpm); chords (major and minor), textures (clean, distortion, pads, noise), styles (classic to quirky), instruments (acoustic, electric, synth, vocal), and performance (clean pluck, muted, arpeggios, stabs).

‘Loops’ consists of two offerings: Drums and Melodies. The melodic loops include ambient pads, melancholic piano lines, electric guitar phrases with lush reverb, and beautifully mixed ensemble pieces. Drum loops come in two forms: as

mixdowns or stems, which is useful when you have to deliver alternate mixes with parts of the drum kit muted. It also offers the creative option of dropping a specific sound mid-loop, like a kick, for example, and bringing it back later for dramatic effect. Both drums and melodies are broken down further into their individual elements, as single‑hit samples in the one‑shots category.

In terms of styles of music, there’s an interesting mix here: the drums stay solid and classic (with the exception of their edgy friend ‘Dark Trap’), while the melodies push the boundaries of experimentation, with sounds ranging from gorgeous acoustic guitar chords to granular effects gone wild.

You could have a lot of fun seating the two next to each other, like at a spontaneous house party that yields unexpected and wonderful friendships.

All the samples are meticulously recorded; you would have to work hard to make them sound bad in your mix. A special shoutout for the close‑miked drums for clarity and incredible imaging, even when listening on basic headphones. The samples hold up well to manipulation — reverse the audio, time‑stretch, pitch‑shift, and it all works fine.

Bones is a solid and generous collection of starting points: whether you’re a newbie whose ability to play currently lags behind your ability to compose; a veteran struggling with writer’s block; or there’s a deadline looming and you need a new drum part but can’t record one today.

Being a collection of WAV files, the library is easy to get started with. Good news: no learning yet another product-specific GUI. Bad news: there’s still hard work to be done.

Not to be a spoilsport, but to truly make the most of Bones, you have to study it and get familiar with the variety it offers so that the next time you’re deep in the middle of a composition, your brain already knows what is available within the library and recalls it right when you need it. This is good practice for any new sample library and the pros always advise that you get familiar

with your tools. The hardest work you will do with Bones is keeping track of all that it offers because it offers a lot and it would be a shame to not make the most of it.

Sonal D’Silva £100 www.songathletics.com

Zero-G Ethera Gold Odyssey — Angelic Voices Kontakt Instrument

HHHHH If your music composition involves genres that span the fantasy, medieval, Celtic or otherworldly, then you will undoubtedly be familiar with Zero-G’s various Ethera and Element series of Kontakt-based sample libraries. The latest addition to this collection is Ethera Gold Odyssey, a further collaboration between Zero-G, Xfonic and Stefano Maccarelli.

While all of the current titles are designed to work within this musical world, each has its own theme and the tagline for Odyssey — Angelic Voices — provides a pretty accurate description of what this latest Ethera title provides.

The vocals for this stage of the Ethera’s journey are provided courtesy of soprano Sarah Claudia Mueller and, if the extensive sampling conducted for Odyssey is any indication (the library runs to just over 8GB), then she has a truly wonderful voice. On the technical front, the full version of Kontakt (6.8 or later) is required. Once installed, the 27 multisampled instruments fall into four main groups; true legato, vocal phrases, polyphonic syllables builder and texture designer.

In terms of the sampling technology (and, I suspect, the bulk of the sample content), the true stars of the show are the two true legato instruments; Angelic True Legato and Odyssey True Legato.

The Angelic option provides a lighter (yup, angelic!) vocal style with just a hint of breathiness to the voice. On the surface, this is a very straightforward instrument featuring two articulations based upon ‘ah’ and ‘uh’ vowel sounds, with keyswitching available between them. However, it only takes about half a dozen key presses to realise just how beautiful these samples can sound; this is pretty heavenly stuff.

As with other Ethera titles, Odyssey provides an impressive set of vocal phrases.

In this case, they are housed in a single

preset, with keyswitches to move between groups of phrases and an additional key range that contains vocal drones. Again, the scripting provides some useful performance options including sync to host tempo and a legato option that allows you to effectively join parts of separate phrases together. I’m not sure what the total phrase count is but it must run into the hundreds.

They are instantly gratifying and sound fabulous.

If you are an existing Ethera user, you will have a pretty good idea of what to expect from Odyssey. Yes, this is a sample library series with a very particular musical theme, but for any composer looking to add some atmospheric vocal parts to their fantasy-inspired film, TV or video game scores, all the Ethera titles are a great source of inspiration. Actual angels who do vocal session work are hard to come by but Sara Mueller — via Ethera Gold Odyssey — can get you wonderfully close. John Walden £69 www.zero-g.co.uk

Spitfire Audio Tutti Kontakt Instrument

HHHHH Mindful of the fact that many media composers lack specialist orchestration skills, Spitfire have created an all-in-one library that brings together a broad range of orchestral textures in one easy-to-use package. The original source recordings are taken from Spitfire Symphony Orchestra and were recorded from three mic positions at AIR Studios’ Lyndhurst Hall (for details, see my SOS August 2024 review).

The man entrusted with distilling SSO’s multifarious contents into usable blended sections is composer Andy Blaney.

A longtime orchestral sample user and Spitfire associate, Mr Blaney programmed new performance legato patches for the 2024 SSO relaunch and composed a brilliant online demo for Tutti. Here follows a rundown of what you’ll find in the library, a collection of classic orchestral sound combinations which can be woven into compositions of great power and subtlety.

A combo liable to fire composers’ imaginations is strings and woodwinds, a blend of SSO’s 60 string players with the library’s 11 solo and four ‘a2’ duo woodwinds. Silvery traces of flute and piccolo add a nice breathy timbre to the violins in the high register, while below Middle C bassoons and bass clarinet can be heard amidst the cellos, culminating in a rousing contrabassoon and double basses blast on bottom C1.

Mapped over a C1-E6 range, the strings and brass ensemble is a great sonority for soaring melodies and grand orchestral themes. Trumpets and violins are evenly matched in the high and middle register, but the brass becomes more dominant when the French horns kick in around Middle C. Adding to the pomp and majesty, bass trombones and tubas provide a massive foundation at the low end.

The big symphonic tutti (Italian for ‘all’ or ‘together’) effect reaches maximum intensity in the ‘orchestra’ patch: this blends strings, brass and woodwinds with a charming quiet percussive layer of xylophone and piano played in octaves. While the latter adds colour, I wish there was an option to turn it off — there is a standalone woodwinds, brass and strings patch, but the blend of those three sections sounds somehow less powerful than in the full tutti patch.

Articulations are restricted to high and low legato, long, short, and long with short attack, the latter a very welcome layering that gives the strong attack needed for rhythmic work.

There are no separate strings, woodwinds or brass ensembles (I guess the moral is ‘buy SSO’), but you can load the delicate xylo/piano, tuned percussion (marimba and celeste, methinks), snares/congas and low drum/ cymbals as four separate instruments.

Tutti is a great-sounding, inspirational sketching tool with the potential to end up in your master mixes. Hopefully it won’t break the bank, and its modest 3.92GB size shouldn’t overburden your hard drive either.

The library runs on the free Kontakt Player and also on the full Kontakt version 7.5.2 and up. Dave Stewart £99 www.spitfireaudio.com Audio examples of this month’s libraries are available at www.soundonsound.com.

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Studio One

Speed up your session navigation with these pro-user tips. select Return to Start on Stop.

Now, it instantly leaps back to wherever playback started tudio One has from, ready to play again. In a reputation for speed different scenarios, this can be and agility. It has the equally annoying, so maybe it’s sort of interface that feels fluid, worth committing the shortcut integrated and easy to navigate.

Alt+Numpad 0 to memory to But that only really starts to gel toggle between the two. once you’ve mastered a number Instant looping: While we’re of essential key commands and right-clicking on the transport shortcuts. Otherwise, working bar, there’s another feature your way through menus can there that solves, for me at be as dull as it is in any other least, another ancient gripe.

DAW environment.

I’ve always found setting So, here are my tips for loops in the timeline to be getting to where you need to go unnecessarily laborious, which and doing what you need to do is a bit ridiculous because you in the fastest possible time. only have to select something Avoiding The Timeline and press P to loop around that selection. Still, for me, that’s one Clicking in the timeline to key too many. If you right-click change the play head cursor’s the transport bar and select position or create loop points Loop Follows Selection, the loop can be annoying. The timeline is markers will be instantly placed narrow, and if you happen to be around anything you select — working on a track all the way that’s such a relief. at the bottom of your massive The other thing you need monitor screen, it can require to know is that the shortcut for an enormous amount of mouse turning looped playback on energy to get yourself back to and off is Numpad /. It’s not the top. So, here are some ways an entirely satisfactory choice, to minimise messing around in but L for loop is already taken the timeline. to locate the playhead to the start of your current selection, Clicking in empty space: This and C for cycle turns the click one is a hidden gem. Click on the spanner icon at the top of The ‘Locate when clicked in empty space’ option lets you move the play head on and off. without you having to drag your mouse up to the timeline each time.

Cursor sticks to edits: Lastly, for the Inspector, and it will bring up the timeline enhancements, we an Options menu. One of those can remove the need for the L shortcut Timeline reset on stop: One minor options is ‘Locate when clicked in empty that brings the playhead to our selection. annoyance that I’ve been living with for space’. Tick this box, and whenever you There’s an option on the toolbar to make years is that I’m always having to drag click outside of a clip in the arrangement that happen automatically. It’s a tool the song pointer back to the beginning window, the play head will leap to that located next to Autoscroll, it looks like to where I want to play from. Play and point or the nearest quantised position. a vertical line with left/right arrows going Stop, by default, works very much like Saves many long throws of the mouse. out from it, and is called Cursor Follows a ‘pause’ on playback. So whether you This can also work within events Edit Position. are pressing the space bar to toggle if you have the Smart tool engaged. playback or clicking on the transport This is selected on the toolbar by A Zoom With A View controls, the song pointer/play head clicking the left-hand icon that looks cursor pauses and then starts again like a left square bracket. The Smart There are numerous ways to zoom in from where it stopped. Commonly, tool gives combined tools so that if Studio One. To zoom horizontally, you can you’re going over the same part of your mouse is on the top half of a clip, click and drag in the timeline, spin the your song again and again, and while it uses Range, and on the bottom half, scroll wheel while your mouse hovers, or looping the section can be useful, it uses the Arrow tool. Another tool can scroll anywhere while holding Ctrl+Shift it would save you time if you could be swapped in by holding Ctrl (Windows) (Windows) or Cmd+Shift (Mac). To scroll restart playback from the same place or Cmd (Mac). Anyway, with the Smart tool vertically, you can hold Ctrl (Cmd) and without having to think about it. selected, the Range tool can be used to scroll pretty much anywhere. There are The way to change the behaviour is locate the play head by clicking on the also two little levers on the bottom edge to right-click on the transport bar and top half of the event. of the arrangement window that you can ROBIN VINCENT

S

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The shortcut Alt+Numpad 0 toggles Return to Start on Stop behaviour: when enabled, hitting Play after stopping playback will play back from where you first played, rather than the point at which playback stopped.

play with to get the zoom just right. The vertical one is hidden under the track headers, and you can choose a size from Minimal to X-Large; the horizontal one is in the bottom‑right corner.

You can also use keyboard shortcuts to zoom in and out in predefined increments. There’s never just one way to do things in Studio One! However, all these methods zoom in on the entire project and are not particularly focused on anything. There’s a better zoom option called Zoom to Selection, which does exactly what you think. Click on an event, press Shift+S and the selected event becomes the main character, filling your screen, ready

for editing. Press the same combination again to return to the previous view.

If you want to zoom out to a full overview, then we can get into some Auto Zoom. The Auto Zoom commands don’t have any keyboard shortcut assignments by default, so we’ll have to set those up.

Under the Studio One menu, you’ll find Keyboard Shortcuts. Search for ‘Auto Zoom’. There are four options here, but the most useful one for me is Toggle Auto Zoom Full. This pulls back and puts the whole project into the frame, which is perfect for scuttling around your project. I’ve assigned it to the backslash key, which appears to be otherwise idle and unproductive. You’ll also find these

options on the toolbar if you feel that mouse mileage is more your thing.

Another zooming option is to dig out the old magnifying glass. You probably haven’t used one of those since version 6 of Photoshop, but it can still be useful.

Hold Shift+Alt (Shift+Option), and your cursor turns into a magnifying glass. Click and drag to select a range within the arrangement, and you will zoom directly into it. You can go back out by holding the same keys and clicking the mouse in empty space.

Data Zoom There’s a different sort of zoom that’s not about the arrangement or size of events, but is actually about the size of the data within the event. There’s a tiny control, down in the bottom right-hand corner next to the horizontal zoom lever, that’s called‘Data Zoom. If you click on that and slide it up, all your audio events will bloom into maximisation. It looks like you are clipping the heck out of your audio, but this is just a visual zoom on the waveform data, so the audio level is not being changed. (If level change is

Toggle Auto Zoom Full is a good one to assign a shortcut to: it automatically zooms out to show your whole project, from where you can zoom in to wherever you need to.

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S E S SION N AV IGATION

Studio One

your goal, then you can adjust the gain envelope within the clip itself.) Data Zoom leaves all the levels as they were, even if the two things have the same look. There’s plenty of room for confusion here, but if you’d like fuller, more pouty waveforms without having to record hotter or normalise, then Data Zoom is for you.

Launcher & Mixer The Launcher is the newest aspect of Studio One, and comes with a whole bunch of potential keyboard shortcuts.

I say potential because most of The Data Zoom slider lets you perform a vertical zoom on your audio clips (no gain is added: it’s purely a visual aid). your keyboard is already mapped out ‘optional’ things will be cleared away. names. There’s a very cool way to make to other stuff. By default, there are only Track Filter: Talking of clutter, there’s them all the same as the track name. three shortcuts given to the Launcher. a shortcut that will help you filter out Double-click the track name as if you are B opens the Launcher, ; (semicolon) a bunch of tracks so you can focus on going to rename it and hit Shift+Enter: all launches cells and ’ (apostrophe) the ones you are currently working on. the events will be magically renamed to stops cells. There are dozens of other It doesn’t have an assigned key, so match the track. possibilities that you could map shortcuts you first have to open the Keyboard Return to default: You can return many to if you wish to focus in on your Shortcuts window and search for ‘Filter’ controls to their default settings by Launcher workflow. in order to set one. There are two simply holding Ctrl (Cmd) and clicking Assorted Shortcuts options: filter channels for decluttering on them. Fabulous for throwing faders the mixer or filter tracks for sorting out back to unity or pan controls back to the I’ll leave you with an assortment of the arrangement. Filtering the tracks centre, or undoing all of your terrible other shortcuts that can help speed seems more important to me. Assign it plug‑in fiddling. up session navigation. an unused key if you can Show/hide inserts: Pressing find one — full stop appears F3 to bring up the mixing to be free. Now, when you console is, I’m sure, old hat by press your assigned key, now. But did you know that if the track list appears on you press Shift+F3, you can the left with the filter box toggle the insert/send section ready with a flashing cursor. on and off? It can give you Simply type what you want a lot more space to mix and to see, and only tracks with arrange at the same time. You that text will be displayed can also toggle the width, in the arrangement. This but no keyboard shortcut is works particularly well with assigned to that by default. folderised tracks and groups.

Event names: Events on Finally, a top tip you wish you’d known Optional views: If you want a quick way a track can often come from different at the start of this workshop: you can to get out of a messy and cluttered view, places, different takes and different assign a keyboard command to open say with the Browser and Mixer getting in sources. As such, they can end up with the Keyboard Shortcuts window! the way, then hit Shift+F12 and all these incremental or completely unrelated

“I’ve always found setting loops in the timeline to be unnecessarily laborious...

If you right-click the transport bar and select Loop Follows Selection, the loop markers will be instantly placed around anything you select.”

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Pro Tools

TECHNIQUE

Multiband Splitter opens up a whole new world of processing options.

JULIAN RODGERS

A

mazingly, the Avid Pro Series of plug‑ins that come bundled with Pro Tools is over 10 years old.

They still feel fresh, and despite having some top-flight alternatives, I use them in every session. But there are some which I’m sure get overlooked by most people. The AudioSuite Pro Limiter Loudness Analyser, for example, makes an excellent partner with Pro Limiter and offers offline loudness analysis, while Pro Multiband Dynamics also has a close relative, Multiband Splitter. This simple plug‑in presents a set of crossover filters, which split the full frequency input into up to four bands at customisable crossover frequencies. These can then be routed via dedicated busses out to other destinations in the Pro Tools mixer.

Perhaps because it presents a blank canvas, it would benefit from some suggested uses. After all, plug‑ins have to solve real‑world problems!

When you insert Multiband Splitter onto a track, Pro Tools will automatically create a new bus for each frequency range.

The manual doesn’t provide specific details about the filters used in Splitter, but Pro Multiband Dynamics employs eighth-order Linkwitz-Riley crossovers, so it’s safe to assume that Splitter uses the same. These steep 48dB/octave filters have great band separation, and both Multiband Dynamics and Splitter report zero latency, confirming that they are not phase-linear. The Pro Multiband Dynamics plug‑in actually offers the same facility for outputting each band via automatically created busses, so you could use Pro Multiband Dynamics to split and route audio without engaging the dynamics processing, and the results would be identical.

When The Split Goes Down To set up Splitter, simply instantiate it on the track you want to split. In this case I’ll use it across a whole mix, though it can be useful on groups or individual tracks. Once it’s loaded you can access the automatically

created busses, which can output each band separately, from the Plug‑in category in the input selector of a track of the corresponding width. To create a four‑band split across the whole of a mix you’ll need to be routing to a mix bus which is then routed to a Master Fader (in the way most people do as a matter of course). Load Multiband Splitter on the mix bus aux input and create four new aux inputs next to it; you could create these in a Basic Folder to help with session organisation. Next, select the four new aux inputs and set the inputs for these to the Multiband Splitter Plug‑in busses. You can set them all at once using the Cascade feature: select the four aux inputs, hold Command+Option+Shift (Control+Alt+Shift on Windows) and select the Low Band bus in the first selected aux.

The inputs will be set to the next available bus for each consecutive track.

I don’t recommend this splitter for parallel processing, so I’d always route the mix bus on which Splitter is instantiated to a dead‑end output, so its contribution can’t be heard as well. With this all set up, it’s a good idea to save it as a Track Preset for use in other sessions.

Splitting Image

Combine Multiband Splitter with a short delay on the high-frequency band, with one side polarity-inverted, for some mono-friendly stereo widening.

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In this example I’m using all four bands, but in most real‑world cases I find I’ll often use a few as two bands depending on the application. Bands can be disabled using the four buttons in the lower left of the plug-in UI. The three crossover points are freely adjustable between 40Hz and 10kHz, and with these two sets of controls, it’s easy to set up whatever you need.

So once you have your splitter set up in your session, what can it help you to achieve? The most obvious application is multiband dynamics, but that is already comprehensively covered by the Pro

Multiband Dynamics plug‑in, so unless you want to experiment with different compression styles on different bands that’s probably beyond the scope of this article.

One of the things you can achieve very easily is manipulating the width of the stereo image across the frequency spectrum. On stereo tracks Pro Tools uses stereo pan controls as opposed to balance controls, so it is possible to narrow, widen and even invert the stereo field. For example, it’s very straightforward to narrow the width at the bottom end using these controls, though because of the steep filters, I’d avoid dramatic moves from hard right and left to mono; this could sound peculiar if a part moves across this crossover band during the performance.

Pan controls only allow the stereo image to be narrowed. If you want to widen the top end, as so often is desired, you can use your usual Mid‑Sides techniques on the high band. An alternative way of achieving this is to duplicate the top band and insert Mod Delay on the copy. Set delay to a 100‑percent wet, short (under 30ms) single repeat (no feedback). Follow this with a Multi-Mono Trim plug-in with the master Link button unlinked and one of the two channels polarity inverted. This introduces a slightly delayed, out‑of‑phase version of the top band, which adds width — very neat!

results; for example, driving the low band moderately hard creates a different sound from low-pass filtering a saturated full-range band in parallel with a clean version.

Individual Tracks Subtle widening or harmonic enhancement is probably as far as you’d want to go when processing the entire mix, but in single‑instrument applications the creative possibilities increase. Instead of using LoFi to add a little colour, try the venerable PSA-1 Sansamp plug‑in for some multiband sonic mangling. Adding four instances across the Splitter outputs can be achieved

is an ideal candidate for the multiband treatment. A single band of chorus on the most appropriate area might be all you need, but if you want to experiment with multiple choruses with individually tweaked settings for each band, you absolutely can.

Ambience In a typical mix you might use two or three reverbs and maybe a couple of delays shared across the session using sends and returns, but there will be occasions where you want a special effect on a single instrument too, from a specific snare reverb to a shimmer reverb on a guitar. One area that rewards experimentation is of course delay, and I’ve had great success splitting delay treatments across a multiband split. Be aware, though, that using different delay times on different bands can get confusing. As an alternative to filtered repeats, try using the same delay time on all bands while varying the feedback. For example, you might have more feedback on the high bands than on the lower bands. This creates an effect that is similar in nature to having a high‑pass filter in the delay’s feedback path, but is a useful alternative, as it offers more control and the option for different processing in each band.

However you use Multiband Splitter, it’s an intriguing plug‑in and you probably already have it. Since time spent setting up a complex effects chain can be recalled as a Track Preset or from an old session using Import Session Data, it is never time wasted.

“Using filters to split the input signal before introducing independent saturation processes on each band is a powerful tone‑shaping tool...”

Saturation One great use for Pro Multiband Splitter is saturation effects. Saturation and distortion introduce additional harmonic content to the input signal, and the specific blend of new harmonics created by the process is responsible for all the different timbres available. Using filters to split the input signal before introducing independent saturation processes on each band is a powerful tone‑shaping tool, and considering I first experimented with it back in Cubase VST with Steinberg’s Quadrafuzz plug‑in, I’m surprised this technique isn’t more common. FabFilter’s excellent Saturn plug‑in uses this technique to great effect, but you can create some very useable effects by splitting the signal, placing a LoFi plug‑in on each band and experimenting.

My preferred option is to use the distortion control in very small amounts (less than 1.0) and then tweak the amount for each band — and, importantly, to adjust the crossover frequency to best suit the material. It’s possible to produce unique

in a single operation using Option+Shift (Alt+Shift on Windows) to instantiate a plug‑in on all selected tracks. Use Option+Shift+Command (Alt+Shift+Control on Windows) when clicking one of the plug‑ins to bypass them across all of the selected splits to make sure you haven’t gone too far.

I’ve never been much of a fan of chorus as I find it can easily make things sound too unstable, but if the effect is applied only to the overtones of a pitched sound, leaving the fundamental unaffected, then the rather queasy feeling which can result from too much chorus can be easily avoided. This

Chorus effects can be a bit much across the full bandwidth of a signal, but applied only to a select frequency band, the results can be much subtler.

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Pro Tools Cubase

TECHNIQUE

The Pattern Editor makes a great starting point for generating arpeggio-style patterns.

Need a better arpeggiator? Cubase’s Pattern Editor can deliver the goods!

JOHN WALDEN

T

he new Drum Machine and Pattern Editor features in Pro and Artist versions of Cubase 14 are obviously designed to complement each other.

However, the Pattern Editor feature set also provides some interesting possibilities for designing synth arpeggio patterns. So, if you are fond of the occasional arp, pick yourself a pluck sound, pluck up your courage, and let’s see what I’m ’arping on about...

Making Notes Cubase has long had a number of tools for creating arpeggio patterns, including the Arpache MIDI plug-ins and the arpeggio engine built into Retrologue. However, the Pattern Editor provides a very neat alternative because it offers some creative options.

Once dropped onto the timeline, you can easily change the pattern number or convert the patterns into standard MIDI parts.

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For this example (for which I’ve created accompanying audio examples on the SOS website: https://sosm.ag/cubase-0525), I’ll use an instance of Retrologue with a short ‘pluck’ style synth preset. The first step, though, is in the Add Track dialogue box: select the Pattern Editor option as the Event Type. When you then select this new track in the Project window (as shown in the first screenshot), the Editor tab in the Lower Zone will show you the Pattern Editor rather than the standard MIDI Editor.

When used with Drum Machine, the Pattern Editor automatically configures lanes for each of the available drum sounds linked to the correct MIDI note. That’s great for obvious reasons, but for a melodic instrument you’ll need to reconfigure this — to make pattern creation easier it makes good sense to set up lanes only for any notes that you know you’ll need.

For example, in the screenshot, I’ve just set up lanes for a single octave (C3 to C4), using only notes in the C major scale. You could pick more exotic key/scale/note combinations if you wish, and use a span of notes over multiple octaves too; the choice is yours, depending on how complex you like your arpeggio patterns to get. Also, observe that I’ve assigned the notes in pitch order (lowest pitch at the bottom of the lane

list). This isn’t how the Pattern Editor would do things by default, and I don’t think you can simply drag/drop lanes to reorder them — plan ahead, and you won’t have to do this twice!

Built By Hand To keep things simple for this demonstration, I opted for a conventional 16-step, 16th‑note grid on which to build my arpeggios, but you can adjust these settings globally (in the Pattern Editor’s topmost menu bar) or on a per-lane basis (although, there are some quirks to how lanes with different step counts behave if more unusual rhythmic patterns are your thing).

Creating your first pattern manually (by clicking on step combinations when playback is active) lets you quickly experiment with the basic structure of your arpeggio’s starting point. And although we’re using it as an arpeggiator, note that this is really a step sequencer, and that means you can add multiple notes on an individual step if you want to trigger a chord here or there as well as a melodic line.

Once you have an initial pattern (named Pattern 1 by default), hit the Duplicate button in the toolbar. This will preserve your original pattern should you need to go back, while also creating a copy (named... yup, Pattern 2)

that you can begin to get a little more creative with. Patterns can be renamed, of course, and that can make it easier to keep track of multiple versions of your arpeggio as you work.

Variations On A Theme The Pattern Editor is a super-simple environment for hand-crafting an arpeggio, but it really comes into its own for adding musical interest by generating variations.

This is possible because of the various randomisation, probability and modulation elements in the Pattern Editor’s features set.

Randomisation options are available both globally (in the toolbar) or on a per-lane basis.

For our use case, the most appropriate options are available in the Parameter Lanes, located at the base of the Pattern Editor panel. These operate on a per-lane/ per-step basis and allow you to set, for example, per-step velocity for the notes in your arpeggio pattern. However, providing your synth sound is velocity sensitive, you can also add Velocity Variance and playback Probability (as in the main screenshot) to any active notes in the currently selected lane. And, as both of these elements (Velocity Variance and Probability) add their randomisation on each pass through the pattern, you can introduce some performance variation on playback.

Subtlety generally works best here, but feel free to experiment.

These randomisation options work best if you avoid notes that hold the core of the pattern (such as the whole-beat steps), because when these go ‘missing’ the rhythmic sense of the arpeggio can get lost.

Applied to notes towards the end of the bar, though, such as steps 13 to 16 in our 16-step example, they can work really well to add melodic interest/variation.

Three further things are worth noting here. First, you can add step-based modulation of your synth’s parameters in the Parameter Lane section. Simply tap the ‘+’ icon and you can select a target parameter to add to the existing selection.

Second, if you want to reset the values in any parameter lane to their original starting points, hold the Ctrl/Cmd button while clicking (or dragging) in the lane. Third, to start with at least, just build your initial patterns starting on the same root note.

We’ll get on to adapting them to the chord sequence in your project shortly.

Arp Arranger Once you have created a few pattern variations, the next step is to arrange them on your project timeline. This aspect of the Pattern Editor’s design is rather neat: you can drag and drop patterns to your track using the button next to the pattern name box. Whichever pattern you currently have selected is placed on the timeline.

Like a MIDI clip, you can then drag the middle-right handle of the dropped pattern to repeat it (for example, to create four bars of Pattern 1). You can, therefore, easily lay out your various patterns into a longer sequence. Each pattern shows its pattern number top-right, and clicking on this opens a drop-down menu — great if you need to change the pattern number for a particular instance. This can be seen in the topmost track of the first screen.

With your sequence of patterns in place, you can get those patterns to adapt to the chords used in your project. There are a number of ways to approach this, but (as in the screenshot) the simplest is to use Chord events placed on the Chord Track. In this first Pattern Editor iteration, pattern events don’t automatically follow

the Chord Track, but if you select all the required pattern events and open the pattern number selection drop‑down menu again, you can choose Convert Pattern Event to MIDI Part. This does exactly what it says on the tin: it generates a standard MIDI clip. But, rather wonderfully, these ‘rendered’ MIDI clips still include the randomisation and probability variations we configured in the Pattern Editor.

If you add a Chord Track, you can sequence your project’s chord changes using Chord Events placed on that track in the usual way. With that done, in the Chords panel in the Inspector you simply set your newly converted Pattern Editor MIDI track to Follow Chord Track. There are a number of configuration options here, but the most basic Chords & Scale or Chords options should be fine as a starting point. The notes in your various patterns will then be adjusted to match the chord changes in the Chord Track. The second track in the main screenshot has been treated in this way.

This may well be all you need, but if you want to add a little extra interest then, as on the third track in the first screenshot, simply select one of the MIDI clips and, in the MIDI Editor, select the notes in that clip. You can then dip into the Chord Editing panel and start experimenting. You can actually change the chords themselves here, but simple options such as the Inversions and Drop Notes buttons can be very effective — these make it easy to voice the notes in a specific chord in different ways, including over more than a single octave.

Finally, the advantage of using the Chord Track as part of this process is that you can get multiple Pattern Editor‑inspired tracks to synchronise to the same chord changes. This is exactly what I’ve done with both an arpeggio synth and bass synth in the last of the audio examples that you can audition on the SOS website.

More Mods?

Once your pattern-based MIDI is following the Chord Track, the Chord Editing panel provides additional options for voicing of the chord notes.

Finally, we mustn’t forget that Cubase’s new Pattern Editor is great for its primary design intention — inspiring new drum beats. But as I’ve hopefully demonstrated above, it’s also brimming with unplanned creative potential, so I’m now crossing my fingers and hoping that Steinberg continue to evolve the possibilities it offers. For example, both a pattern preset system and the option for patterns to follow the Chord Track would be great (though I’ve no idea of the technicalities of making this latter option work!). Meanwhile, playtime with patterns can be a lot of fun. Enjoy!

www.soundonsound.com / May 2025

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Logic Logic

TECHNIQUE

We explore some creative delay treatments using only Logic’s bundled plug-ins.

Combine Tape Delay with Space Designer and some unconventional impulse responses for some intriguing Objeq Delay-style sounds.

PAUL WHITE

W

hen looking for effects that are a bit outside the mainstream, it is tempting to search out third-party plug-ins. While many of those produce excellent results, with a little creative thought it is possible to conjure up some

inspiring and unusual effects using just Logic’s own plug-ins. This month I’m going to take a look at delay effects.

As a quick example of what is possible, you can get a really good retro, lo-fi sound from Logic’s Tape Delay (set up in a bus as a send effect), simply by setting the frequency sliders to 130Hz/1200Hz and then adding around 45 percent of LFO modulation, at 0.7 seconds or thereabouts to add a bit of wobble. If you’d like the fi to be even lower, put the delay through Logic’s Overdrive plug-in, set to a modest amount of drive and with its filter knob set just to take the edge off the sound. Alternatively, put the Rotor Cabinet plug-in set to Fast after the delay, and your piano or electric piano repeats become sonar pings! You might also want to add some basic reverb to the source track so that the dry part of the sound doesn’t sound too dry.

Objeq Permanence

Placing Step FX before a delay can yield some interesting rhythmic filter results.

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Those are some very basic examples, but there are more adventurous combinations of Delay and other Logic Pro plug-ins to try.

My inspiration for this next idea was Objeq Delay from Applied Acoustic Systems, which is one of my favourite delay plug-ins for ambient music creation, especially for adding interest to rhythmic parts. While it isn’t possible to exactly recreate what Objeq Delay does using Logic’s plug-ins, you can conjure up effects with a similar character.

Objeq Delay treats the repeats with convolution, using impulse responses (IRs)

based on drum heads, metal sheets and so on, the outcome of which is that the delays take on the character of the IR rather than being straight repeats. It lets you modulate the pitch and formants of the filtering, but it’s the use of impulse responses that got me experimenting.

Logic Pro’s Space Designer reverb comes with a host of useful non-reverb IRs in its Warped Effects / Drum Transformers section. You can also drag in any short WAV file, such as a recording of a ringing circular saw blade or a metal barrel being struck, and use that to process your delays.

The setup is straightforward enough: you put the instrument or audio file to be treated on a track with a post-fade aux send feeding a bus with Space Designer inserted on it, which is then followed by your delay of choice. I favour Logic’s Tape Delay here as you can also add pitch modulation and filtering, but any of Logic’s delays will work, and Modulation Delay also has useful modulation features. When using convolution, something you need to be aware of is that the level of the processed signal can sometimes be quieter than you might wish due to the way the modulation process essentially multiplies the levels of both sounds, so putting a compressor between Space Designer and the delay is a good way of getting the levels back under control. I find the strongest results come about when the IR has plenty of frequencies that overlap with those of the source sound.

Both Space Designer and the delay would normally be set to ‘wet only’ when used as send effects (because the dry sound comes

either after the delay or between Bitcrusher and Delay, and then play a loop of piano or percussion while skipping through the Bitcrusher presets — you’ll hear lots of usable options! You can also try switching out Spectral Gate for EVOC Filter Bank and you’ll be rewarded by even more useful treatments, many with a sense of movement or sync’ed modulation.

Design For Life Bitcrusher is great for adding colour to your delays.

from the original track). However, if the effect of the IR in Space Designer seems a bit too much, you can turn up the dry level in Space Designer until the effect sounds right. As with most creative endeavours, the only rule is that you have to like the sound!

The kinds of source sounds that show up these flavoured delays most strongly tend to be percussive rather than sustained pads, so things like pianos, guitar plucks, synth stabs and percussion tend to work well. Logic Library IRs that give good results on a sampled piano include Distant Air, Fragmented Clip and Glowing Strike.

If you want to get a little more adventurous, try instantiating both Space Designer and Tape Delay as dual‑mono rather than stereo plug-ins, so that you can apply completely different settings to the left and right channels. For example, you might use the IR of an anvil strike with a quarter‑note delay on the left and something like a bottle smash IR running into a dotted eighth‑note delay on the right.

Another option that can yield interesting results is to drag an Apple Loop directly into Space Designer. You can then use the IR offset control and the envelope curves to home in on just the part of the sound you want to use. The Car Crash sound effect worked surprisingly well in this way, again using a piano as the test subject.

I’ve also had good results using a sparse rhythm loop as an IR, which can add some interesting movement to the delays, in addition to tonal coloration.

that can be achieved. While you can use Logic’s AutoFilter to add moving filters, Step FX makes it easier to arrive at more dramatic results as it can add stepped chorus, distortion, reverb and even delays of its own. The rhythmic nature of the effect can also add interest to sparse drum or percussion loops.

Logic’s Bitcrusher is another useful tool for colouring delays. Try setting the Fold mode to 10x Downsampling, 8-bit Resolution and Drive to 10dB. If the delay filter is set wide open, you get a nice metallic jangle when using a piano as an input, and by closing the filter, the high end of the jangle can be tamed if it feels excessive. This same setting also works well on percussion loops.

Taking this effect further, try placing Logic’s Spectral Gate (try Crunchy Bandpass and then tweak the centre frequency by ear)

Logic’s Delay Designer plug-in allows for unusual delay patterns, and also pitch‑shifting of the delay repeats, so can conjure some pretty unusual sounds on its own. But if you fancy a bit of enharmonic chaos, try putting Ring Shifter before it. Beat Breaker can also be put too good use to add rhythms to your delayed sounds — the Vox Cut preset is really rather good in this application, and works well either before or after the delay.

Once you find a combination that gives you a useful result, it is worth saving the channel strip setting so that you can call it up again when needed. If you need to pass on your user channel strip settings to a friend or copy them to another computer, you’ll find them in Application Support / Logic / Channel Strip Setting. Sub folders hold the settings for Bus, Instrument, Master, Output and Track channel strips, so the setups discussed here will be in the Bus folder. The files are very small, so you can send the whole Bus folder as an email attachment.

Step On Another creative way to mutate delays is to use Logic’s Step FX plug‑in before the delay, again set up in a bus to be used as a send effect. This can add rhythm and resonant filtering to the delays. The Brittle Spike and Deep Dub presets provide a good illustration of the type of effect

Beat Breaker’s Vox Cut preset works well for mangling your delays, and can be placed either before or after the delay plug-in.

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Q

Why do most XLR and jack cables have soldered connections, rather than crimped ones?

I saw some crimped XLR connectors online, and I wondered what the advantages are of soldered connections over crimped ones.

I’ve seen crimping used in lots of other industries and other audio connectors like EDAC looms, terminals on passive loudspeakers and RJ45 connectors. I’ve seen solderless jacks used for guitar pedalboards. Having soldered a few DB25 looms, I can see why crimping might be more convenient! But I’ve very rarely encountered XLRs that are constructed this way, or premade cables for that matter. Is there an advantage of using soldered joints (or a disadvantage with crimped ones)?

If so, does that apply to all signals, or perhaps only to quieter ones such as mic level, where XLRs are routinely used?

Ian Gorant, via email Hugh Robjohns, Technical Editor First off, I don’t believe signal level is relevant here: a good connection is a good connection, and that’s all that’s required whether we’re talking mic level or line level.

I’ve never seen a crimp-style jack connector, but Neutrik currently offer a crimp version of the XLR and, in the past, I recall that they also offered an IDC version (more on this type below) for a while too. However, I’ve never knowingly used XLR cables made with either type of connection; they’ve all been standard ‘solder bucket’ types.

Soldering wires into XLRs, or jack plugs for that matter, is a familiar, well-established, and relatively easy-to-learn technique that uses commonly available and low-cost tools (ie. a soldering iron and solder).

Importantly, that methodology has decades of proven reliability in the field, and can be repaired easily in the field with standard tools and skills.

The two common industrial alternatives to soldering are insulation displacement connections (IDC) and crimping. Both techniques essentially ‘cold weld’ the wire to the contact to make a gas-tight connection, which is necessary to impede corrosion of the joint. Both can theoretically be used in factories or in the field. Crimping and IDC are designed to

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There are XLRs with crimp connections, available from the likes of Neutrik, but they’re arguably better suited to large-scale factory manufacturing techniques than DIY.

be faster to implement than soldering, and crimping in particular can be very beneficial in very high-density connectors. However, in both cases, the connector has to be designed appropriately for crimping or IDC, and special (meaning expensive) dedicated assembly tools are normally required.

For IDC, an axial tool (working at 90 degrees to the wire) is usually required to force an insulated wire between the terminal connector blades, although ‘no-tool’ IDC connectors are increasingly common (where a bladed cover is tightened down forcing the wires into the contact terminals). IDC systems have the advantage that they don’t require the wire insulation to be removed prior to insertion as the blade cuts straight through it and into the wire itself, speeding up the assembly process. However, IDC only works well with solid-cored wire, and while that’s fine for installed cabling systems, it’s not so good where cables need to be flexible and robust as that usually requires multi‑stranded wires. Also, the gas-tight connection area between the wire and blades is inherently very small indeed, potentially affecting reliability.

In the case of crimping, a bespoke radial tool (working around the wire) is required to crimp the contact terminal onto the (insulation-stripped) wire, which means the connector has to be designed to facilitate that technique. Unfortunately, the multi-pinned XLR doesn’t naturally lend itself to that approach, but a common solution for complex connectors like this is to crimp the contacts onto the wires away

from the connector, and then insert them all into the connector body afterwards.

Key benefits of crimping include a much larger gas-tight joint area than IDC, and the technique works reliably with stranded wire.

Crimping suits factory manufacturing very well, with cables being prepared at one station, connectors crimped on at another, and then insertion into the connector body at a third, followed by testing. However, a bespoke crimping tool is generally required, designed for the specific contact terminal design. And inserting the contacts into the connector body often requires another special tool.

In most cases, a third dedicated tool is also required to release the contacts when necessary for repair/replacement.

Dedicated crimping tools aren’t cheap — that for the Neutrik Crimp-XLR costs around £400 (about $500), for example — and they’re not something you’re likely to find in most audio repair shops, let alone the field, so repairing crimped connectors can be a serious problem.

For a manufacturer producing a very large number of cables, crimping can potentially save enough time and money to justify investing in the specialist equipment, especially for complex, high-density connectors. But, in the case of standard XLRs, which are easy to solder, it seems few think there is a cost-effective benefit. There are also significant disadvantages including the initial bespoke equipment cost and the difficulty of repair because the relevant tools are likely to be unavailable outside of the factory.

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E‑mu keyboard, we look back at the history of a true classic.

LOGIC’S SAMPLER PAUL WHITE

T

here are lots of elaborate samplers on the market, with Native Instrument’s Kontakt in particular providing a powerful platform for third‑party libraries in addition to the extensive NI portfolio of sounds. Kontakt has much to commend it, like its support for custom user interfaces, and it has some very capable onboard effects too, but it can sometimes seem overkill for what you are actually trying to do. There are basic sample players around too, with relatively few controls, but sometimes those are just too basic! So what’s needed is something that falls into the Goldilocks zone somewhere in between.

For me, as a Logic Pro user, it has to be Logic’s own Sampler (previously called EXS24). It doesn’t have effects built in, though Logic has no shortage of those if you need to process whatever samples you are using. But one big plus is that I can see what I’m doing with it (albeit with reading glasses these days), as the legend is nice and clear, plus you can resize the window if necessary. Sampler also hosts all the features you’d expect from a subtractive synth: a choice of filters, comprehensive envelope generators, a flexible but non-intimidating modulation matrix, practical LFOs and so on. You can also set the number of voices and whether or not to use legato mode.

There are relatively few third-party sample libraries for Sampler — in fact the only one I can think of off hand comes from sub51.co.uk — but that isn’t too much of a problem, as Logic comes with a huge library of samples covering all genres. And if that isn’t enough, Logic also makes it easy to create your own sample sets. You can drag and drop audio directly from Logic’s main window; you’ll see various options for turning those samples into a usable multisampled instrument, including automatic pitch recognition, which works fine as long as the sounds aren’t too weird.

Logic Pro also comes with Autosampler, which can automate the creation of

“Kontakt has much to commend it... but it can sometimes seem overkill for what you are actually trying to do.

What’s needed is something that falls into the Goldilocks zone somewhere in between.”

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multisampled, multi-velocity instruments from either software instruments (or instrument stacks) running within Logic Pro, or from externally connected hardware synths.

Autosampler offers various options including one-shot playback and a choice of automatic looping strategies so the user doesn’t need to do much more than check that the levels are OK and then hit go. The finished Sampler instrument is then stashed away in Logic’s library for when you next need it.

Other benefits to working with Logic’s own samplers include the ability to save any used samples as part of a Logic song file, so when you send a copy of a project to a friend to continue working on, all the necessary samples are already there. What it boils down to is that I like things to be as uncomplicated as possible — and Logic Pro’s Sampler fits the bill nicely.

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