SOUND ON SOUND [UK] May 2025 by soundonsoundmag

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Avantis Solo

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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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TECHNIQUE: SPECTRAL BALANCE / CREATIVE DELAYS IN LOGIC

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D I G I TA L I S T H E N E W A N A L O G

Hydrasynth delivers a world of synthesis capabilities to your fingertips with a familiar, organic workflow that unlocks a new world of sound design & performance possibilities. Available as four distinct models each featuring; 8 or 16 Voice Polyphony, Wavemorphing Oscillators, Polytouch, Multiple Filter Models, lush Stereo Effects, a powerful yet intuitive Modulation Matrix, rugged build quality and so much more.

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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.

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30 EVGUENI & SACHA GALPERINE

IN THIS ISSUE

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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

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

Open-back Headphones

Sample Library

119 76 40

A-D/D-A Converter

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

RYK Envy Machine Eurorack Module

Soniccouture Flowstate Software Instrument

Submarine Pickups SubSix Hexaphonic Guitar Pickup

58 Universal Audio

UAFX Enigmatic ’82

Guitar-amp Emulation Pedal

Vanguard V24 Stereo Valve Microphone

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

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

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

May 2025 / www.soundonsound.com

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.

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

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

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

ADI-2 Series

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

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ADI-2 Remote & DigiCheck NG

Low Noise Isolating DC Filter, DC Input & output

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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

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

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

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Soniccouture Software Flowstate Instrument

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

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

YMC

Apollo X Gen 2 sounds better. Go behind the scenes with UA Technical Fellow Dave Rossum.

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

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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. M AT T H O U G H TO N

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

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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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

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

May 2025 / www.soundonsound.com

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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www.soundonsound.com / May 2025

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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

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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

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

May 2025 / www.soundonsound.com

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

ON TE ST

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

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

May 2025 / www.soundonsound.com

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!”

May 2025 / www.soundonsound.com

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

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

May 2025 / www.soundonsound.com

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,

www.soundonsound.com / May 2025

ON TE ST

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.

May 2025 / www.soundonsound.com

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

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.” 48

May 2025 / www.soundonsound.com

“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

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

May 2025 / www.soundonsound.com

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.

David Wrench

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

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

May 2025 / www.soundonsound.com

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

ON TE ST

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.

May 2025 / www.soundonsound.com

DAV E LO C K WO O D

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

May 2025 / www.soundonsound.com

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.

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

Allen & Heath Avantis Solo Digital Mixing Console

May 2025 / www.soundonsound.com

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.

www.soundonsound.com / May 2025

ON TE ST

A LLE N & H E AT H AVA N TIS S OLO

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.

May 2025 / www.soundonsound.com

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.

Silver 107.868

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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65 02/04/2025 13:15

INSIDE TRACK

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

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

May 2025 / www.soundonsound.com

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

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

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

May 2025 / www.soundonsound.com

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!

www.soundonsound.com / May 2025

ON TE ST

AudioSmile Little British Monitor Active Monitors

These compact speakers punch well above their weight — and you can build them yourself! PHIL WARD

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

May 2025 / www.soundonsound.com

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.

www.soundonsound.com / May 2025

TALKBACK

Teether & Kuya Neil WILLIAM STOKES

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.

May 2025 / www.soundonsound.com

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

ON TE ST

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.

May 2025 / www.soundonsound.com

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

ON TE ST

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.

May 2025 / www.soundonsound.com

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.

www.soundonsound.com / May 2025

ON TE ST

A P P L E M 4 & M 3 U LT R A M A C S

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

May 2025 / www.soundonsound.com

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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ON TE ST

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

May 2025 / www.soundonsound.com

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

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

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

www.soundonsound.com / May 2025

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Expert Sleepers Disting NT Eurorack Module

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

May 2025 / www.soundonsound.com

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

www.expert-sleepers.co.uk

Cutlasses Instruments Gloop Eurorack Module

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

www.cutlasses.co.uk

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RYK Envy Machine Eurorack Module

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

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

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.

www.soundonsound.com / May 2025

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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

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

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

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

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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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

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

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.

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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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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

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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.

www.soundonsound.com / May 2025

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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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LOGIC’S SAMPLER PAUL WHITE

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.” 138

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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.

THE COMPLETE DRUM PRODUCTION STUDIO. Produce acoustic and electronic drums with control and creative power beyond the imaginable.

NEW EXPANSIONS. Some of the latest releases in our ever-growing collection of SDX expansions.