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SOUL SEARCHING R

ock & roll changed everything. Not least, the music industry. Before the mid-’50s, sheet music outsold records. But no-one needed a printed score to learn a 12-bar blues, and no score could capture what made Chuck Berry recordings compelling. Record sales overtook those of printed music, and musicians learned songs by ear, not by reading. Then, in the mid-’80s, MIDI took music in another new direction. Sequencing lent itself to repetition and precision rather than more traditional forms of musical expression, while sampled instruments captured one-dimensional snapshots of the real thing. Which brings us to today’s technological revolution. Sound On Sound and Sonarworks have carried out a major survey of musicians’ attitudes to artificial intelligence. We’ve taken a deep dive into the results in this issue, and they make for interesting reading. Unsurprisingly, AI is viewed as a threat to creativity; but there’s also a perception that this threat is genre-specific. It’s widely felt that styles such as jazz and blues, which resist easy MIDI-fication, will also be least susceptible to AI. I’m not so sure. To my ears, the ways in which generative AI currently falls short of the mark mostly relate to audio quality. I don’t hear it as robotic or mechanical, or lacking expressivity. And I think that’s

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because AI is not reductive in the same way that sheet music or MIDI or sampling are. Like rock & roll fans of the ’50s, AI has been trained on music in all its richness, not a dehydrated version of it. Instead of condensing music to its essentials, AI analyses the recordings themselves, identifying patterns on a very deep level. Train it on genres where subtle tempo variations are crucial, and it’ll faithfully replicate that. If a singer has a particular inflection at the end of a long vowel, AI will notice. Consequently, it turns out that AI is actually very good at, for example, playing dirty electric blues. (If you don’t believe me, spend a few minutes with AI music channels on YouTube.) At the same time, it also feels as though AI currently isn’t doing anything more than rehashing existing musical styles. It’s a truism that AI is limited by its training material, and thus can’t be original because it can only draw on what it already ‘knows’. However, when sequencing became mainstream, human musicians turned its limitations into virtues. Precision and repetition became the hallmarks of new genres. Does AI music not have similar attributes that humans can abuse in imaginative ways? Are there no glitches in the matrix that can be chiselled open? If it can’t do more than pastiche rock & roll, maybe AI is not such a threat to creativity after all.

“It’s a truism that AI is limited by its training material, and thus can’t be original.”

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ISSN 1473‑5326 A Member of the SOS Publications Group The contents of this publication are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publisher. Great care is taken to ensure accuracy in the preparation of this publication but neither Sound On Sound Limited nor the Editor can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the Publisher or Editor. The Publisher accepts no responsibility for the return of unsolicited manuscripts, photographs, or artwork. © Copyright 2026 Sound On Sound Limited. Incorporating Music Software magazine, Recording Musician magazine, Sound On Stage magazine, SPL magazine, Sound Pro magazine and Performing Musician magazine. All rights reserved. All prices include VAT unless otherwise stated. SOS recognises all trademarks.

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138 AM/PM LIVE

IN THIS ISSUE

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February 2026 / issue 4 / volume 41

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82 Modular Tobinski’s Danny Tobert talks about building his own system and embracing the creativity of modular.

88 Binaural Recording The final instalment in our series on stereo recording explores the likes of the dummy head, the Jecklin Disc, and more.

94 Vintage Flanging In Your DAW Proper tape flanging has an instantly recognisable sound, but sometimes plug‑ins just don’t cut it.

108 Spotlight: MPE Desktop Synths MIDI Polyphonic Expression (MPE) grows ever more prevalent, and plenty of hardware synths are getting in on the action.

112 SOS Awards

WORTH $2497 PAGE 78

We reveal this year’s winning products, as voted for by you!

134 40 Ideas That Changed Music: Part 3

120 Akai MPC60 Revisited

Our four-part series on the biggest technological breakthroughs in music technology counts backwards from 20...

We travel back in time to see how the story of one of the most important machines in modern music began.

124 AI & Music Tech In 2026

LEWITT & KALI AUDIO BUNDLE

138 AM/PM Live

We asked music professionals how AI is changing their world.

When former Roxy Music members Phil Manzanera and Andy Mackay recorded an experimental improv album, they had no intention of performing it live — let alone in Dolby Atmos!

128 Inside Track: Yungblud

146 Why I Love... Sound On Sound

Producer Matt Schwartz once told Yungblud his songs were stupid. Then he started producing them...

The legendary Ethan Winer on why there’s only one music technology magazine worth reading!

February 2026 / www.soundonsound.com

70 WES AUDIO PANDORA

ON TEST 74 84

1010music Bento

AFK Audio Drumbeam

40 Aguilar Plugin Suite Bass Rig Modelling Software C O V E R

Goodhertz VCME 3

136 Heavyocity Dystropia

MIDI Percussion Pad

56 Akai MPC Live III Sampling Workstation

85 Becos FX CompIQ Yuna VCA Compressor Pedal

80 Black Corporation ISE-NIN 87

Acustica Audio Cadet AEA Nuvo N28

20 IK Multimedia ReSing

52 JBL PRX925 & PRX935

Sample Library

Three-way Monitors

32 McDSP FutzVerb

SPL P8 Eight-channel Microphone Preamplifier

Toontrack The Drum Factory SDX Superior Drummer Expansion

24 Universal Audio

Paradise Guitar Studio

Amp & Effects Modelling Plug-in

28 Kali Audio SM-8 70

Wes Audio Pandora Digitally Controlled Multiband Analogue Compressor

Audio Mangling Plug-in

137 Orchestral Tools Sample Library

Dolby Atmos Production Software

136 Song Athletics Inside-Out Piano

Active PA Speakers

Electro-Harmonix Pico Swello

Console 2

Eurorack Module

Sample Library

James Newton Howard Piano

64 Fiedler Audio Mastering

81 Shakmat Bishop’s Miscellany MkII

Vocal Resynthesis Software

137 Indiginus Electric Blue

Scaler Music Scaler 3 Compositional Software

Sample Library

Eurorack Module

Envelope Shaping Pedal

Portable Vocal Booth

Compressor Plug-in

Active Stereo Ribbon Microphone

GIK Acoustics PIB Pack

Sampling Workstation

Reverb Plug-in

50 Quantum Electric Q24 Valve Capacitor Microphone

36 Rupert Neve Designs OptoFET Dual-stage Compressor

WORKSHOPS 96 SOS For Artists 100 Studio One 102 Cubase 104 Logic 106 Pro Tools www.soundonsound.com / February 2026

ON TE ST

AEA NUVO N28

AEA aim for the middle ground with an intriguing new stereo ribbon mic. SAM INGLIS

t’s hard to describe the sonic character of a microphone without resorting to cliché. In the case of ribbon microphones, those clichés often involve adjectives like ‘smooth’, ‘warm’ or ‘dark’. But are those really fundamental aspects of ribbon mic ‘tone’? Or do they arise because of the way the mics are being used? If you swap out a typical capacitor mic for a ribbon in a close mic placement, the chances are you will notice that the sound is indeed thicker and richer at the low end. But that is partly because nearly all ribbon mics have a native figure-8 polar pattern, and consequently exhibit lots of proximity effect. In terms of true low-end response, a pure velocity mic such as a ribbon struggles to go as low as an omni capacitor mic — so if you carried out the same swap in a distant mic placement, it might well be the capacitor mics that sound deeper. At the same time, most ribbons have less high-frequency extension and presence than their capacitor brethren. Consequently, if a ribbon mic is ‘voiced’ so as to sound balanced in close‑mic placements — perhaps by deliberately attenuating the low end to compensate for proximity effect — it may well sound middly and boxy in distant applications, lacking both air at the top end and weight at the low end. That’s a phenomenon AEA have tried to counter in their well‑established R88, which is described as a “far-field stereo ribbon mic”, and offers greater than usual high- and low-frequency extension. The R88 is available in active and passive versions, and has also spawned a mono derivative called the N8. But now AEA have introduced a second stereo model, with slightly different applications in mind.

Active Service The Nuvo N28 is available only in an active version, and is significantly more compact than the R88. Its two ribbon motors are likewise mounted one above the other in a fixed Blumlein configuration. Below this, a bulge in the body of the mic houses a pair

February 2026 / www.soundonsound.com

AEA Nuvo N28

Active Stereo Ribbon Microphone of Lehle transformers with a 1:89 step-up ratio, and a newly developed buffer/line‑driver circuit that delivers a comfortable sensitivity of 6.1mV/Pa. Finished in satin black, the N28 is very attractive in a self-effacing sort of way, with a helpful white circle identifying the front. It ships with a compact, allegedly shock-absorbing clip and a 15-foot cable that attaches to the mic using a five-pin XLR, with a conventional pair

of three-pin male XLRs at the preamp end. Both of these need to be fed with 48V phantom power. All in all, it’s a classy package, let down only by a cheap and nasty plastic case. Whereas the R88 is designed for distant miking scenarios, the N28 has a different raison d’être. It’s described as being “ideal for medium distances” — which, AEA clarify, means distances from eight inches to three feet. That’s

an interesting and relatively unusual concept for a stereo mic. A Blumlein array has a 90-degree acceptance angle; at three feet from the source, that means the entire stereo field will be filled by an instrument measuring a little over four feet across. Most drum kits are bigger than that, never mind a grand piano or brass section. Conversely, placing the mic eight inches from a source is likely to deliver a stereo picture much wider than the way the same source would ever be perceived in real life. A less rigid interpretation of AEA’s design goal might be that the N28 is a stereo mic intended for capturing individual instruments rather than ensembles. Which, in my experience, makes it pretty much unique. What it means in practice is that the N28 has a “mid-field low-end response tuning” designed to deliver a balanced low end at the kind of intermediate distances AEA are talking about: within the range where proximity effect is a factor, but not close miking as such. AEA also say that the mic has more obvious presence in the upper midrange than the R88. Naturally, the N28 also inherits many characteristics typical of ribbon mics in general. Its two motors both have a very clean and well defined figure-8 polar pattern in the horizontal plane. In the vertical plane, high-frequency response tends to fall away as you move off axis owing to the shadowing effect of the body and the other ribbon motor. The published plot suggests that the frequency response (of, presumably, one N28 motor measured on-axis) is broadly flat up to about 5kHz, and thereafter falls away in a fairly smooth manner, being about 8dB down at 10kHz and 11dB down at 20kHz.

Heavy Metal During the review period, I had the chance to try out the N28

“There’s plenty of heft and weight in the 100-300 Hz zone, so the fundamental resonances of snares and toms really pop out.” on drum overheads as well as on vocals, acoustic guitar, electric guitar amps, mandolin, hand percussion and the like. Lots of engineers I know and respect love ribbon mics as drum overheads, but personally I find it can be a struggle to make them work at the mix. When you’re expecting the body of the drum sound to come from close mics, you can have too much body in the overheads! If that’s your thing, though, the N28 delivers it in abundance. In line with AEA’s description, the very low end is indeed curtailed compared with some other ribbon mics I tried, but we’re really talking only about the 50Hz region here. At all normal distances, there’s plenty of heft and weight in the 100-300 Hz zone, so the fundamental resonances of snares and toms really pop out. At the top end, meanwhile, cymbals come across as polished and, in a subdued sort of way, airy. In the world of compact stereo ribbons, obvious competitors to the N28 might include models like the Royer SF‑12 and Extinct Audio Valkyr Stereo. I didn’t have either available for this review, but I do have a pair of mono Extinct BM9s, which are to all intents and purposes one half of a Valkyr, and have used the SF‑12 a fair bit in the past. Both the BM9s and the SF‑12 are quite forward in the upper midrange in comparison to many ribbon mics, with the SF‑12’s presence lift perhaps tending slightly further up the spectrum than that of the BM9. The same phenomenon is not so apparent with the N28; put it up alongside the BM9 (and, by extension, the Valkyr) and you’ll typically get less 50Hz and 3kHz, and more low midrange. Which is preferable obviously depends both on the source and on your tastes, but I found

myself liking the sound of the N28 on a lot of things. As I said at the start of this review, it’s probably a cliché to describe a ribbon mic as sounding ‘smooth’. Nevertheless, it’s an apt term for the N28. That relatively unhyped midrange means that it doesn’t emphasise gritty hash or harsh resonances in the way that brighter mics sometimes can, yet at the same time, it doesn’t lack definition, and nor does it sound soft. And, as is often the case with coincident stereo mics, the results can be startlingly lifelike. One of the best sounds I got with the N28 was by placing it parallel to the fingerboard of an acoustic guitar. This really did make it sound as though the player was sitting inbetween the speakers, giving a private performance to the listener. Ultimately, it would be a shame if anyone was put off buying the N28 by AEA’s description of its intended role. It’s certainly not a mic that suddenly starts to sound bad when you get an inch further than three feet from the source! Its deliberately controlled low end would only really be an issue if, for example, you were to use it as the only mic on an orchestra or a large brass ensemble. In most real-world applications, that area of the frequency spectrum is likely to be either irrelevant, or covered by other mics. In practice, given the N28’s richness in the ‘upper bass’ and low midrange, any EQ you deploy is far more likely to be cutting than boosting at the bottom end. And with its compact dimensions, stylish yet unobtrusive appearance, healthy sensitivity and ability to drive long cable runs into any preamp, it’s practical as well as euphonic. Although it’s complementary to AEA’s R88 in some ways, there’s a very large area of overlap where both mics will do a fine, fine job.

summary The Nuvo N28 is a high-quality stereo ribbon mic with a wide range of possible applications.

The N28’s clip is decoupled from the stand by a soft plastic loop that adds a degree of shock protection.

$ $2499 W www.aearibbonmics.com www.soundonsound.com / February 2026

ON TE ST

JOHN WALDEN

hether from the core library, or the wide range of add-on SDX expansions that are available, Superior Drummer users really can’t complain about either the quantity or quality of the drums sounds SD3 can put at their disposal. That’s not going to stop Toontrack putting temptation into your path though, and the latest SDX release — The Drum Factory — is a prime example. Created with Sylvia Massy (whose recording credits span artists as diverse as Tool, Prince, Johnny Cash and Jason Isbell), and recorded in four different acoustic spaces at La Fabrique, SW France (a regular stop for the Mix With The Masters seminars), The Drum Foundry promises to be something special.

All The Right Kit Let’s deal with the basics. A total of seven full kits (Sonor, Brady, DW, Tama, Ayotte and two from Trixon), alongside a goodly collection of additional snares, toms and cymbals, were recorded using a comprehensive array of close and room mics. The individual kits were each recorded in one of the studio’s four different spaces; the massive stone-built Dark Room (it has no windows), the wood-lined Library (big but warm), the huge Arcades and the Mill (the studio is built in a converted silk mill and this room is also stone, but with a darker ambience). The SDX is divided into four sub-sections based on these different rooms and, if you pull up any of the presets and focus on the overhead and ambience mics, the different characters are clear, obvious and... well, just plain fabulous. The core drum sound could grace any recording and would work across multiple musical genres.

Sonic Experiments For many potential users, that might be more than enough, but the SDX also gives you a peek inside the creative mind of Sylvia Massy. Sitting alongside the more conventional close/room mic options are a number of more experimental microphone choices. These include a microphone attached to a garden hose, another held within a large plastic (and partly filled) water container, a mic built from an old telephone and a mic integrated with an old children’s cassette recorder. Various combinations of these alternative sound sources are integrated into the preset collection and available within the SD3 mixer to blend with the

February 2026 / www.soundonsound.com

Toontrack The Drum Factory SDX Superior Drummer Expansion

Industry legend Sylvia Massy brings her expertise and creativity to Tootrack’s latest Superior Drummer expansion. more usual mic options. If you want to dial in some lo-fi vibes, or just add some unique character to your drum mix, these options let you do it. It’s weird but wonderful, and it undoubtedly lets you take these beautifully recorded acoustic kits into some more experimental/ alternative musical directions. There is a brilliant video on Toontrack’s website that walks through both the conventional and less-than-conventional mic choices; it’s well worth a watch.

Kind Of A Chameleon As ever, the SDX includes an excellent crop of MIDI grooves to explore and these span a range of tempos, both 4/4 and 6/8 time and straight and swung feels. They could work in a wide range of musical contexts and, as such, they are a perfect foil for the sound library itself. Whether mixing, producing or engineering, Sylvia’s credits span an amazingly diverse range of musical artists and styles.

That diversity is reflected in The Drum Factory SDX and, as a result, the user gets options that could work in almost any musical genre, be that extreme and experimental rock/metal, modern pop or electronic music, lo-fi indie or the most delicate singer/ songwriter styles. Sonically, this is a real chameleon of a drum library and, no matter how many drum options you currently have access to, I’ll hazard a guess they don’t include drums recorded by gaffer‑taping a mic into the end of a garden hose. The Drum Factory sounds outstanding and is well worth auditioning.

summary A brilliant source of high-quality acoustic drum sounds but, thanks to Sylvia Massy’s experimental approach to miking a drum kit, The Drum Factory also offers some truly creative possibilities.

$ $179 W www.toontrack.com

8 channel microphone preamplifier The attention to detail in the design of this preamp shows. No electrolytic capacitors in the signal path, true differential amplifier circuitry, and a transimpedance (current feedback) gain stage result in all the detail of my many microphones coming through. And the pre comes with a 20 year warranty (no joke). I could work with two or three of the m801mk2 units as my only preamps at Jackpot! and I’d be totally happy for the rest of my career. And I could stop worrying about which mic preamp to use...” — Larry Crane, Tape Op Magazine / Jackpot! Recording Studio

www.gracedesign.com

ON TE ST

GIK Acoustics PIB Pack If you need to record dry-sounding vocals in a less than ideal room, GIK’s generously sized portable booth could be just the ticket.

Portable Vocal Booth NEIL ROGERS

s a studio owner with intermediate DIY skills, I’ve built dozens of acoustic panels, and have reached the point where they look quite good, if I do say so myself. To get to that stage, however, has required a fair bit of trial and error and, most importantly, many hours of my

precious time. For those who don’t have the time or inclination, off-the-shelf products may be a better fit, and GIK Acoustics have established themselves as arguably the most prominent alternative source of acoustic treatment to studio professionals and home recordists. As well as taking away the hassle and providing a better-looking result, their products have been objectively tested to measure their performance, so you can ensure you’re using the right tools for the job. This review focuses on the latest addition to GIK’s range of portable vocal booths, which is intended as a complete solution for recordists wanting a professional-sounding recording environment that can be moved or repurposed with ease. The PIB Pack is essentially two of GIK’s Portable Isolation Booths, supplied with sturdy clips and fixings to enable them to be clipped together to form a small rectangular booth, plus a panel that sits on top. Altogether, this forms a fairly solid freestanding structure, the interior of which provides a controlled acoustic space. Who would benefit from such a product? For a studio owner like myself, the PIB offers the ability to create a controlled area within a larger live room for dry-sounding vocal or instrument recording, which can easily be packed away or transported to a different space. For home recordists, voiceover artists, podcasters and others not working in a studio environment, the potential is more dramatic. Whilst the PIB Pack won’t eliminate extraneous noise such as traffic or housemates, you should benefit from a substantially more controlled sound entering your microphone, thanks to the elimination of unwanted room reflections.

Putting It All Together The PIB Pack comes in three large, well-packaged boxes that, whilst cumbersome, ensure your panels will survive the worst that courier services can throw at them. The two bifold screens that make up the booth are a breeze to move if you have a willing assistant, but they are quite weighty if you’re on your own, and I think a sturdy carry handle on each section might have been a good addition. Once you have them in the right vicinity, it’s just a case of unfolding the panels, moving them into position and clicking the fastener in place on the top of the two screens. Once you’ve fine-tuned the placement of the booth, the roof panel adds the last bit of stability and creates a more enclosed acoustic space. My only minor niggle here was that due to the positioning of the clips on the side of the panels, you can’t completely butt the two screens together for a flush connection. This isn’t going to affect the performance, but it felt a little frustrating from an aesthetic perspective. Once unpacked, I was able to assemble the booth for the first time in a little under two minutes.

February 2026 / www.soundonsound.com

There’s no place like home. Your home studio is your sanctuary — a place where your creativity thrives and you can discover your true voice. At Sweetwater, we understand how important it is to have the right gear that complements your space and your needs. That’s why we offer expert guidance, unmatched support, and all the best hardware and software to turn your home studio into an oasis of inspiration.

Photo: Mitch Gallagher’s Home Studio

Sweetwater.com (800) 222-4700

ON TE ST

Goodhertz VCME 3

Compressor Plug-in

It’s billed as a ‘mastering’ compressor, but VCME 3 is way more versatile than that. M AT T H O U G H TO N

nother day, yet another compressor? We all have choice aplenty when it comes to compressor plug-ins, but Goodhertz’s Vulf Compressor Mastering Edition (VCME for short) is usefully different, and if your compression needs go beyond the basic, it’s well worth checking out. As the name implies, some facilities are aimed at mastering, but VCME has plenty of potential when mixing, whether on individual sources, subgroups or the master bus. Now at v3 (this is my first encounter with it), VCME supports the usual Mac and Windows plug-in formats, and channel counts ranging from mono right up to 9.1.6.

Main Features The main screen shows the extended GUI, with a soft clipper limiter/saturation facility in an Advanced Settings pane on the right, which is hidden by default. (You can install this soft clipper as a separate plug-in too.) There are a couple of alternative views in this Advanced panel, which I’ll come back to later. Meanwhile, the left pane hosts something looking more like a conventional compressor, albeit with a distinctive‑looking GUI (you can change the red on white livery to white and red on black, if you prefer). On this compressor pane are the usual threshold,

ratio, attack, release and make‑up gain parameters and, beneath, a wet/dry mix slider. All have stepped settings — something that’s very common on mastering hardware, as it makes channel matching precise and facilitates reliable A/B comparison of settings and precision recall. By default, the steps are very fine (eg. 0.5dB steps for the threshold) but there’s an option for coarser granularity (the threshold’s individual steps become 3dB). Not only is the current amount of compression shown on a virtual needle meter and a numerical readout, but there’s also a history plot displaying overall gain reduction post the dry/ wet mix control. There’s also a neat semi-automatic threshold and make-up gain facility: as you set each manually, you’re always in full control, but next to both parameters’ sliders an arrow can appear to suggest a suitable ballpark setting. For example, if you leave the threshold at 0dBFS, the arrow will indicate roughly where to put the threshold so the changing signal level triggers a broadly sensible amount of gain reduction. You can work the same way with the make‑up gain slider, but this parameter is also joined by a continuously updated numeric readout of the variance between the dry and wet signal levels. Hit Apply, and the make‑up gain is automatically adjusted to compensate for the difference. You

February 2026 / www.soundonsound.com

don’t always need to match the levels before/after compression, of course, but it can be invaluable when you want to A/B compare different processors, or the compressed signal with that when the compressor’s bypassed, not to mention making it easier to adjust the wet/dry blend. Yet another intriguing option is the ability to switch the modelled release behaviour between FET and opto types. That’s not unique, but here you can also morph continuously between those extremes, to access a range of different curves. A cog icon reveals more settings, including window size, GPU acceleration, GUI colour and control granularity. Above, there’s a good range of factory presets, helpfully divided into categories (Mastering, Mix Compression, Heavy Compression, Colourful/Saturation, and Limiter Only), including some ‘name’ presets. Less conventional is a second preset option: the URL button. Hit this and a web page is opened on the Goodhertz site, listing the plug-in’s current settings. You can copy the URL, right-click on the plug-in GUI and paste in those settings. It’s a neat system: it creates a visual record of your settings and might be useful for project archival or, say, students wanting to write up their projects. It also means you can pretty much instantly share settings with anyone, anywhere in the world, working in any DAW. The plug-in offers other conventional features like A and B buttons to compare settings, and undo/redo, of course.

Going Soft The three‑dots icon on the central red stripe pops open the Advanced Settings pane, whose three views are Soft Clip, Side-chain and Linking.

ON TE ST

GOODHERTZ VCME 3

As with the compressor, the soft clipper offers an almost bewildering array of options. I say ‘almost’, because Goodhertz have done such a superb job of keeping the GUI uncluttered, organised and intuitive. There are four saturation options: Chino (which Goodhertz describe as “hi-fi, classic tube sound, notes of Hendrix”), Rødovre (“round, tube-y, smooth”), Detroit (“grungy, warm, notes of Jamerson, Sousaphone-y”) and Melbourn (“punchy and British. Strong transformer saturation”). I sense they’re intended largely for the sort of subtle saturation you might choose to apply during mastering, rather than obvious distortion. You can push it harder if you really want to, though, and the characters on offer are as seductive as their descriptions sound. A Drive control effectively sets the saturation threshold, so you have independent control of the overall loudness and the amount of coloration. Other useful control options include a Mix slider, inverse linking of the input and output faders, and the ability to display the limiter’s gain reduction, loudness range (LRA) or long-term loudness (LUFS-LT). More esoteric options include configuring the limiter as a brickwall type or to ‘follow the mix’, and to change the slew rate. Operating this section is easy. I tend to set Drive to 0 and set up the limiter, before dialling in saturation to taste. For the latter, I find it helps to lightly overcook things to choose the desired character, then bring Drive right down again, before dragging it upwards until I hear just enough of what I want. I particularly enjoyed just a touch of Neve-esque transformer distortion to thicken up a singing acoustic guitarist recording I was working on during this review, but all the options have their place and, for example, I found I preferred Rødovre on a harder-hitting pop mix. Horses for courses. I should note that, even with the useful visual feedback on an input/output plot, it’s incredibly easy to overcook this sort of thing when DIY mastering unless you have reliable monitoring. You have more wiggle room for processing individual instruments, though, and it can be really cool to mix into this section on a drum bus.

morphs into a high‑/low-pass filter as you dial in more assertive attenuation, and there’s a tilt facility at the top to shift the balance of energy in the side-chain towards the high or low end, without losing your main EQ settings. An ear icon lets you audition the side-chain, post-EQ, so you can fine-tune the trigger signal more easily. That’s a heck of a lot of control over the side-chain, with only a handful of controls.

Impressions

With only a handful of controls, there’s ample means to adjust the compressor’s side-chain signal.

Sides relative to the Mid, as well as set the limiter’s stereo link mode to Full Off or Hybrid. That last option can work well: it links the limiter’s left and right channels only during the slower phases of gain reduction, and seems to preserves some of the ‘life’ in the transients. There are similar control sets for the compressor and, if using on surround material (something I haven’t had the opportunity to do) the centre and LFE channels. In the final Advanced panel you can configure the compressor’s side-chain signal. Internal and external side-chains can be used and, intriguingly, with external selected, you can blend between the two. You also get a two-band EQ, but it’s more versatile than that description sounds. A heat map tells you where in the spectrum most of the audio energy resides, and there are nodes for each band and an EQ curve on this. Each band is a cut/boost shelf that

summary A great-sounding compressor and soft-clip saturation/limiter. Laden with parameters that allow you to tweak to the nth degree, VCME 3 also manages to feel quick and easy to use.

(Yet) More Options In the Linking tab there’s yet more control over the clipper: you can reduce the amount of saturation applied to the

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As I hinted at the top of this review, I approached VCME 3 with a healthy dose of scepticism. I have plenty of compressor options, both hardware and software, including some excellent ones intended for mastering. Clearly, then, I don’t need VCME 3 — but despite that, it quickly won me over. Not only does it sound good (I include the saturation effects in that statement), but I love that it manages to be both supremely configurable and quick and easy to use, and the more I’ve worked with it, the more I’ve started to appreciate the fast, intuitive workflow it makes possible. Yes, there are other tools that can do what it does, and some that offer different tonalities. But whether you’re working in stereo or immersive audio, on mixing or mastering, this one has plenty to offer and is well worth taking the time to audition.

$ $199 (discounted to $149 when going to press).

The stereo linking options, shown here in Dark mode, include the ability to reduce the saturation applied to the Sides channel.

T Goodhertz + 1 626 720 4815. E support@goodhertz.com W https://goodhertz.com

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JOHN WALDEN

ocal synthesis, whether for sung vocals or spoken-word dialogue, is undoubtedly a process that has made massive leaps forward in recent years, underpinned by advances in AI-based processing. From the perspective of music/ audio production, these products broadly fall into one of two categories: virtual singers (virtual instruments that can generate sung vocals) and voice changers (processors that transform a source voice/vocal into a different voice/vocal). The subject of this review — IK Multimedia’s ReSing — sits within this ‘voice changer’ category. So, if you have a single voice source for sung or spoken vocals, and want to change the character or gender of that voice, or even transform it into a musical instrument, just what can ReSing offer?

IK Multimedia ReSing Vocal Resynthesis Software

Give your vocals a completely new personality with IK’s ReSing.

The Small Print Having already implied that ReSing leans into AI, let’s head off the most obvious (and perfectly legitimate) elephant that might be in your room. IKM make it very clear within their description of ReSing that any training materials used within the product — in this case, the voices of specific singers or voiceover artists — have been used with suitable permission from (and compensation for) those individuals. Assuming you have the rights to use the original source recording you are transforming, any use of ReSing’s voice‑changing technology should not, therefore, get you into any ethical or legal hot water. A second important initial observation is worth making. Sitting within the ‘voice changer’ category, ReSing can transform your sung vocals so that they sound like they were sung by somebody else. As a result, while it can change the character of the vocal delivery in some useful and interesting ways, any dubious pitching within the original source vocal is faithfully recreated in the transformed version. ReSing should not really be considered as a magic button to turn a bad singing performance into a great one.

The Voice ReSing is available as a single-purchase perpetual product (no subscription required) and once installed, all processing is done locally. The standard version includes 10 different voice models, 10 instrument sounds and 10 ‘model generations’ while the Max version ups all three numbers to 25. Yes, voice-to-voice transformations

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aside, ReSing can transform your vocal source material into the sound of selected instruments. Equally, if you want to train your own voice models, ReSing supports that. There is an additional ‘unlimited model generation’ add-on you can purchase if you need to train more voice models beyond the included 10 or 25. Once installed, ReSing itself can run as a standalone application, a conventional VST, AU or AAX plug-in or, DAW permitting, via ARA. The latter provides a more streamlined workflow, but both of the other options are perfectly workable; you just drag and drop the source audio into ReSing and processed versions are simply rendered to your drive system.

Personality Permutations ReSing’s operation is very straightforward. A model browser lets you load up to two models (voices or instruments)

into the Model A and Model B slots. Once your source audio has been imported, three circular control sections — Character, Accent and Dynamic — allow you to adjust how either one of these models (or a blend of both of them using the Fusion sliders) are applied when you hit the Process button. If you make subsequent adjustments to these controls, you have to re-render the modelling. Thankfully, on any recent computer system, processing is actually quite speedy, and much faster than real time. The degree of control offered by the various Character, Accent and Dynamic settings is somewhat source and model dependent and you perhaps shouldn’t expect to transform a whispered original into a raging rock monster by simply choosing the ‘Energetic’ Character flavour, but you can subtly nudge the original performance in the required direction by experimenting with these options. The fourth circular control section provides compact takes on dynamics, EQ, reverb and automatic pitch correction. Applied after the transformation process, these are neat to have, particularly if you are just working in the standalone version of ReSing. However, you can easily bypass them when working within your DAW if you have preferred plug-in options for these tasks. I have to say that the results of the voice transformation process itself are very impressive. For example, when starting with well-recorded sung vocals, even when the source was entirely different from the model (different gender, different vocal range, different

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ON TE ST

I K M U LT I M E D I A R E S I N G

The Max edition ships with 25 voice models spanning a wide range of characters, as well as 25 different instrument models.

The ReSing Modeler app lets you train your own voice models for use within ReSing.

vocal character), more often than not, the transformed vocal could easily be dropped into a mix without its synthetic origins being obvious. It’s worth noting that I did manage to get some decent results from vocals that were originally sung into my phone, and ReSing generally added a bit of a polish to that recording once the fledgeling idea was inside my DAW. That said, the cleaner the original, the less likely you are to experience processing artefacts. Max’s 25 voice models provide a broad palette of very useful vocal styles, from the gruffness of Spencer or Neil to the polished pop-friendly flavours of Clara or Gabriella (the latter with a rather nice hint of a Spanish accent). Equally, there are models that could work very well with rap-style vocals or for voiceover transformation. There are even a few special character options including the cartoonish Emily, Chipmunk and Synth Choir. It’s not difficult to see how sound designers might find a tool like ReSing useful for creating voice-based special effects. The various instrument models are also quite interesting. Yes, they do require a little more careful handling in operation if your intention is to achieve a ‘natural’ result, but if you vocalise an appropriate melodic line in your best impression of the target instrument, the transformed results can be rather good. For example, I had some success with the two saxophone and the trumpet models, transforming short vocal phrases into instrumental parts. And, of course, there are some weird — and sometimes wonderful — creative options to explore here. This AI-based approach to virtual instrument sound creation is perhaps still in its infancy, but the potential is both obvious and very interesting. Watch this space.

take a considerable amount of time depending upon your computer hardware. I did have a stab at this process, albeit with a sub-optimal source data set, and the results were interesting if a long way from the quality of the included models. Again, the potential is clear to see, and I can easily imagine film/TV dialogue replacement suites experimenting with ReSing models trained on lead actors within a project to complement (or even replace?) the traditional studio sessions often required for on-set dialogue replacement. Incidentally, ReSing does allow you to import models created by others in both ReSing and RVC (Retrieval-based Voice Conversion) formats.

Model Construction ReSing also lets you build your own voice models. Done via a separate application, the principle is simple; you just drop audio recordings from your model target into the engine, and it then analyses these to ‘train’ the model. The PDF documentation provides guidance of the type and quality of source material required but if you wanted to do this to the standard of IKM’s included models, I suspect it would require a fairly rigorous approach to compiling the source material, spanning expressive singing covering the full pitch range and energy levels within the singer’s vocal repertoire, sustained vowel sounds, syllable-based scales and examples that emphasise sibilants and fricatives. Bearing in mind the limited model generation counts offered by the standard and Max editions, it’s perhaps something you would be wise to plan carefully. Equally, the analysis of the training data can

February 2026 / www.soundonsound.com

Is ReSing The Thing? In terms of competing products, Sonarworks’ SoundID VoiceAI is the most obvious alternative to ReSing. This is also available as a one-time purchase perpetual licence, and offers both voice and instrument transformation options. It doesn’t provide model training capabilities, but it does include a rather neat ‘unison mode’ for generating double‑tracked voices. Having run both side-by-side, they produce comparable results so a personal choice might need to be based on the broader feature sets and pricing of the two products. I’ll finish with a reminder that ReSing (and, indeed, VoiceAI) is a tool for voice character transformation and not a software solution for rescuing technically poor singing by magically resynthesizing it into technically great singing. That said, it can certainly add some delivery polish to a well sung but scratch recorded vocal and, equally, can provide you with endless voice identity options if your lead vocal, backing vocals or dialogue need to be delivered in a different vocal timbre. Is swapping out the character of one voice for the character of another voice any different from picking a different virtual drum kit or orchestral string preset (things many of us accept doing without question) to better suit the needs of a production? I’ll leave you with your own ponderings on that front but, if it’s a concept you are comfortable embracing, IK Multimedia’s ReSing provides the technology required to do just that.

summary With the caveat that it cannot rescue a poor vocal performance, as a tool for voice transformation — swapping out the original vocal character for that of a different singer or speaker — ReSing provides an impressive technical solution.

$ ReSing $129.99, ReSing Max $199.99. W www.ikmultimedia.com

From Hollywood blockbuster composers and Billboard Hot 100 producers right through to keen beginners, the world of music production trusts in Cubase.

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BOB THOMAS

ull disclosure: I already own two of Universal Audio’s UAFX amplifier emulator pedals, the Dream ’65 and the Enigmatic ’82. And I use them constantly! By sheer coincidence, the very day that I was considering buying their Dream ’65 plug-in, in came an invitation to review the Paradise Guitar Studio plug-in. I’ve been exploring it — playing in Paradise, as it were — obsessively ever since. The usual plug-in formats are supported for macOS and Windows. Note that, to obtain and install the plug-in, you must sign up for free Universal Audio and iLok accounts (if you haven’t got them already) and download the UA connect app.

Overview Paradise Guitar Studio has a single signal path, consisting of five pre-amp effects slots, an amp and miked-cabinet slot, and five post-amp effects slots. Mono, stereo and mono-to-stereo operation are supported, so the plug-in can be used on any source in a stereo DAW session. The pre-effects section can process effects in mono or stereo, as you prefer; opting for stereo means an incoming stereo signal remains in stereo after being processed by a mono effect, but this requires more processing power than the default mono setting. The post-effects section always maintains the stereo signal, even if a specific effect is set to process in mono. A Paradise preset can be configured with up to five pre-amp effects, one amplifier with one miked cabinet, an independently adjustable cabinet room sound, and up to five post-amp effects. In addition, there’s an input noise gate and an output limiter that don’t take up slots. Any effect can be used simultaneously and independently in the pre- and the post-amp effects sections.

Universal Audio Paradise Guitar Studio

Guitar/Bass Amp & Effects Modelling Plug-in Their UAFX pedals have consistently impressed, and now UA have packed those algorithms and more into a single DAW plug-in. Paradise has a clean, uncluttered layout, with various panes and a toolbar above for functions including bypass, MIDI Learn function, and copying and pasting effects to/from the pre-amp and post-amp effects slots. The uppermost pane offers access to a tuner and preset library, and is where you configure the pre-amp effects for mono or stereo operation. Beneath this, a Level & Navigation bar carries a graphical display of each section and its contents. This also hosts controls and meters for the input level, a noise gate, the pre-amp effects section’s output level, the amp and miked cab output level, and the post-amp effects output level. The individual effects and amps have level controls too of course, so there’s plenty of scope to optimise signal levels — you can keep them in the sweet spot throughout the signal chain. A switchable brickwall limiter, placed after the post-amp effects output meter, can ensure the main output level never exceeds 0dBFS. Clicking on a section in the Level & Navigation bar determines the Main panel’s contents. In this panel, you can import, delete, replace and adjust the controls of the effects you’ve placed in the pre- and post-amp effects slots, as well as audition, select, replace and customise Paradise’s six amps — two of which contain multiple

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models, to give you a total of 11 separate amps to play through and customise. The Main panel also provides access to a library of 35 cabinets, with a selection of 1x12, 2x12, 4x10 and 4x12 models featuring a range of different speakers. To inform the modelling, recordings were made with mic setups UA deemed best for the specific cab. The sound and position of the mics can’t be changed, then, but you can choose different cabs, and the Room control allows you to dial in room ambience to taste. The final pane displays the front‑panel controls of the selected amp. The virtual knobs and switches work exactly as you’d expect, and there are buttons for functions that would be footswitched in the real world. On models with a choice of which inputs to plug into (or jumper), you can unplug and replug virtual cables.

Exploring To give you an idea of what Paradise can do, it comes loaded with 300 presets that

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ON TE ST

Kali Audio SM-8 Three-way Monitors

Kali’s new Santa Monica speakers are their most advanced yet. PAUL WHITE

ali Audio’s SM (Santa Monica) series monitors are arguably their most ambitious design so far, boasting advanced features like user-editable DSP that can be controlled through software. This enables the user to adjust delays, levels and eight bands of parametric EQ. Available in five- and eight-inch versions and priced per speaker rather than in pairs, the SM monitors follow an active three-way format but, unusually, the mid driver and tweeter are coaxial, which helps stabilise imaging and also allows for a more compact cabinet. For those who don’t need the control software and who already have high-quality power amplifiers, passive versions are also available to order. This review covers the SM-8, which is driven by a total of 300 Watts of Class-D power and has an LF reach of 37Hz (-10dB), with its -3dB points at 43Hz and 21kHz (-10dB at 35kHz). It can cruise at a level of 94dB SPL at one metre but can handle peaks of up to 119dB. A one-inch metal-domed tweeter is set in the centre of a four-inch, paper-coned midrange driver; the woofer also uses a paper cone. Kali describe both cone drivers as having an ‘optimised’ profile, but they don’t elaborate on that! The woofer crosses over to the

Kali Audio SM-8 $4998 pros • Well-balanced sound with tight bass and clear detail. • Comprehensive onboard EQ that may also be used for room correction in conjunction with suitable room measurement tools.

cons • It would have been useful if a full room correction process had been built in.

summary A highly capable three-way active monitor with useful DSP tuning, allowing you to upload correction curves obtained through third‑party measurement systems.

February 2026 / www.soundonsound.com

mids at 280Hz, while the mid/high crossover point is at 2.8kHz. The THD at 90dB (at one metre) is specified as around 0.5%. The front-ported cabinets weigh 14.5kg each, and measure 48.26 x 25.4 x 30.5cm (HWD). They are compact enough to use in medium-sized studios, though the SM-5 might be a more appropriate choice for the smaller studio. While most studio users working in stereo will mount the speakers on platforms or stands, those setting up immersive sound systems may need to use wall mounts, to which end the speakers have threaded mounting points on both the top and bottom of the cabinet, designed for a 4.25 x 2-inch bolt pattern. On the front of the cabinet is a Kali Audio logo, which engages standby mode when double-tapped. Audio may be fed into the speakers either as traditional analogue via a balanced combi XLR/jack, or as a digital signal encoded as AES3 via a BNC socket. Ethernet (RJ45) and USB-A are available for the DSP control. A set of DIP switches on the rear panel allows for the usual upper and lower shelving EQ settings, as well as compensation for boundary proximity, but there’s also a setting (switch four) for accessing control over the DSP EQ. Other than that, there’s a large recessed rotary control on the rear panel that can be used for setting the input level, with a range of -12 to +6 dB.

Remote Control The Kali Control Panel software runs on macOS and Windows, and gives the user access to eight parametric EQ bands, each of which can be set to peaking, high/low shelving, or high‑/low‑pass filtering (first- or

second‑order). It is possible to delay the output by up to 10ms, as well as applying a level trim. For stereo use, the delays will not be needed, but they can be very useful in a multi-speaker setup such as surround or Atmos. These parameters are handled by the same DSP module that looks after crossover duties, so there’s no additional latency penalty on top of the system’s 3ms processing time. Kali Control Panel is not in itself a room-correction package, but it can load KEQ files or be used in conjunction with the free Room EQ Wizard software to generate the necessary EQ curves. Working this way avoids the need for a separate room-correction product and also avoids the hassle of using a plug-in-based correction system, where operator error can lead to the monitor processing also affecting the mixed song file. It might have been easier for the user if the system could automate the room-correction process using a flat-response microphone, but the DSP-hosted feature set does

_KeyLab Essential mk3 Universal MIDI controller.

Break down all the barriers between you, your DAW, and your perfect sound. Introducing KeyLab Essential mk3, an evolved keyboard controller that elevates DAW integration, software instrument control, and workflow-focused design to the next level your keys to creativity.

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McDSP FutzVerb

Audio Mangling Plug-in

McDSP’s new distortion-cum-reverb promises to put your audio through the wringer.

SAM INGLIS

ome plug-in names capture what the plug-in does so perfectly that it almost seems redundant to write a review. Available in all the usual native formats and authorised using an iLok licence, McDSP’s FutzVerb is a case in point. It adds reverb; and it futzes with your sound. Its core futzing mechanism is a technique, previously used in McDSP’s FutzBox plug-in, which they call Synthetic Impulse Modelling. This, it’s claimed, offers the realism of convolution, but with a much lower CPU overhead and the ability to adjust parameters in real time. The supplied library includes around 100 SIMs, recreating devices such as portable radios, cellphones, electronic toys, metal buckets and vacuum-cleaner pipes. Each of these has a Tune parameter that seems to scale the spectral characteristics of the device up and down the frequency range. Additional futzing power is provided through a filter module, with resonant low- and high-pass bands that can be set to 12 or 24 dB/octave response, and an innocuous-looking module labelled Distortion. As well as the expected choice of saturation types and an Amount dial, this has numerous more obscure controls. Some of these are relatively ‘normal’, like Intensity, which somewhat resembles a tone control for the added harmonics. Others are anything but,

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such as Chop, through which “audio waveform inter-cycle signal level ranges are eliminated (chopped) to create a new kind of distortion,” and Wobble, which uses low-frequency content in the signal to modulate the distortion characteristics. There’s probably some further futz potential in the Pre-verb module, which features simple high and low tone controls, a pre-delay knob and, more unusually, a variable subharmonic generator. The pre-delay can be negative as well as positive, which opens up some interesting sonic possibilities. And finally on the futzing side of things, the SIM module has a switchable lo-fi section that allows you to reduce the word length and sample rate. Both the SIM itself and the lo-fi processor can independently be switched to different positions

in the signal chain, processing the whole signal either pre- or post-reverb, or ‘within’ the reverb. Whilst not technically an agent of futz, moreover, the reverb itself certainly isn’t your run-of-the-mill room simulator. It’s algorithmic rather than convolution-based, and offers a choice of 22 algorithms and 12 “reflection types”. You won’t find comfortingly familiar plates or ambiences among the former list, which instead includes ‘canyon’, ‘abyss’ and multiple caves, waves and silos. It’s paired with a relatively conventional dynamics section offering variable gating or ducking.

Playing In The Dirt I’ve come across reverbs that offer distortion and other unconventional sound-shaping possibilities before, notably Overloud’s Rematrix and HOFA’s IQ-Reverb. However, those plug-ins are primarily reverbs that allow you to dirty up the sound a little if you want. It doesn’t take much experimentation with FutzVerb to see that the emphasis is the other way around here. This is a plug-in that is primarily designed to screw with the sound, but which also allows you to

“If you want to create the effect of a computer announcing ‘Self-destruct in 30 seconds’ over a broken PA system on a burning spaceship, this plug-in makes it almost too easy.” add reverb if you want. So if your goal is to lend a little bit of character to your vocal sound by introducing some valve distortion to a vintage plate emulation, this probably isn’t the reverb for you. If, on the other hand, fearless sonic experimentation is your thing, FutzVerb comes highly recommended. It’s possible, though not always easy, to get a ‘normal’ reverb from it; but its strengths lie far from the normal. Whereas you’d usually use a reverb plug-in as an auxiliary effect and send to it from multiple sources, this is very much the sort of plug-in you’d use on an insert to transform a single source. (I’d still like it to have the ability to lock the wet/dry mix when loading new presets, though, and it’s also a little surprising there’s no mono-to-stereo version.) Only

about half of the presets feature reverb as the most prominent effect, and even there, it’s almost always churned-up, metallic, gritty or downright nasty. Elsewhere, distortion and filtering come to the fore.

Having A Futzball FutzVerb is a plug-in that has at least as many applications in post-production and sound design as it does in music. If you want to create the effect of a computer announcing “Self-destruct in 30 seconds” over a broken PA system on a burning spaceship, this plug-in makes it almost too easy. In a music context, I found it particularly rewarding on background sounds such as pads, where it can turn a cheesy synth drone into something ominous, wobbly and less demanding of

space in the mix. At the other end of the scale, there is also endless fun to be had on drums and drum machines, particularly if you want to stop them from sounding like drums and drum machines! Like all the best plug-ins, FutzVerb packages its enormous potential within a relatively simple user interface. Each of its individual processing modules is a powerful effects generator in its own right, so the sum of the parts is extremely versatile. It takes some self-restraint to acknowledge that you don’t have to use more than one or two of them on a given source, and it helps if your tastes lean towards the characterful rather than the pretty, but in a world where we all have eleventy million distortion and reverb plug-ins already, FutzVerb does offer something new.

summary FutzVerb does exactly what the name suggests!

$ $179 W www.mcdsp.com

“The bottom line is that the Pro Channel III is an incredibly versatile recording channel, and one whose combination of sound and features mean it punches well above its price.” - Matt Houghton, Sound On Sound

artproaudio.com

www.soundonsound.com / February 2026

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AFK Audio Drumbeam MIDI Percussion Pad

AFK Audio’s Drumbeam is a percussion pad with a difference. MARK GORDON

IDI percussion pads come in a whole range of shapes and sizes... except they don’t, really! Almost all follow the tried and trusted layout of nine or more square pads, in a grid format, with a central LCD screen and a selection of rear-panel connections. That’s why your first sight of the Drumbeam, from German start-up company AFK Audio, is genuinely a surprise. At around 21 inches in length and one inch square, this unusual-looking multi-touch MIDI controller is designed to be played primarily with drumsticks and can be configured with up to 12 individual zones along its length. Each zone, when struck, can either send out MIDI notes or Continuous Controller messages.

AFK Audio Drumbeam €429 pros • Unique design. • Rugged and roadworthy. • Good mounting options. • Additional trigger input. • Easy-to-use app. • Endlessly configurable.

cons • A little less dynamic than other drum pads. • Comparatively expensive, as it has no built-in sounds.

summary A well-made, highly configurable performance tool that offers a different slant on the ‘percussion pad’ concept.

February 2026 / www.soundonsound.com

Bucking the trend is often commendable, but are looks deceptive — is this really something new, innovative and, most importantly, usable?

Beam Me Up The Drumbeam is constructed from milled aluminium, giving it a professional, weighty feel that belies its compact form. The top ‘striking’ surface is covered with a thick silicon pad, and a series of 72 individual LEDs run the length of the unit on both the back and front, indicating the individual playing zones. The underside of the device features an integrated mounting bracket that, along with the included multi-mount adaptor, enables you to attach the Drumbeam to a standard M8 threaded cymbal stand or clamp. To the left of the mounting bracket are the connections to the outside world: a USB-C socket that provides both power and MIDI connectivity to your computer, and two 3.5mm TRS sockets. By way of the included TRS-to-MIDI cable, they offer a standard five-pin MIDI in/out connection. A further 3.5mm jack socket provides a trigger input that can be connected to an external drum pad or trigger. To the right of the mounting bracket are two raised buttons that enable you to scroll back and forth through the eight presets that can be stored in the Drumbeam itself.

Make Light Work When you first power up the Drumbeam, the LED strips spark into life with a colourful rainbow animation, before settling to display a default preset, in this

case three distinct yellow zones, each separated by two blank LEDs. Each Drumbeam preset can be configured with up to 12 individual zones along its length, each with its own LED colour. As I mentioned, each zone can send out MIDI notes and Continuous Controller messages. The unique and somewhat minimalist form factor of the Drumbeam precludes the inclusion of a display so any configuration is done within the Drumbeam app. Let’s dive in. Although the Drumbeam connects to a Mac or PC via USB-C for its MIDI connectivity (and power), it utilises a built-in Wi-Fi connection to communicate with its web-based editing app. With your computer, tablet or phone connected to the Drumbeam Wi-Fi hotspot, typing ‘drumbeam.local’ into any browser launches the editing app, which opens in Preset view. Both sets of LEDs, front and rear, can be configured independently, potentially giving the audience a completely different viewing experience during a live performance. The front-facing LEDs can be configured to Highlight, Flash, Sparkle or Beam via the Animation drop-down menu, with the user-facing LEDs simply flashing more brightly when struck. You also have the option of inserting zero, one, two or three blank LEDs of spacing, to help you differentiate between zones. Within the Preset view, you can toggle whether the preset will globally send MIDI notes or MIDI CC information, and activate the trigger input. You can also configure the musical scale and

Octave and Squeeze parameters — more about these options later. Clicking on one of the zones in the app, or striking it on the Drumbeam itself, opens the detailed Segment view. Here you can add and remove zones, modify individual zone size, and set the colour of the LEDs on both the front and rear of the Drumbeam. In addition to each zone having its own MIDI note, MIDI channel and Continuous Controller, you can also limit the dynamic range or controller value range, as well as applying velocity curves.

Hit It With Your Rhythm Stick I quickly configured a preset comprising four zones with MIDI notes assigned to the kick, snare and open and closed hi-hat of a virtual drum instrument running within my DAW, and within a few seconds was happily bashing out a rhythm. Integrating the external trigger input into the preset worked very well and allowed me to connect an electronic bass drum, giving the setup the feel of a kit. While the Drumbeam certainly lends itself to percussive applications, it works equally well in pitch-based scenarios, where zones assigned to notes can play melodies or bass lines. The Musical Scale and Octave parameters I mentioned earlier come into play when creating a pitch‑based preset. You can select a starting note, octave and scale, and the Drumbeam automatically populates the zones you have created with notes within that scale. Also linked to more melodic usage is the Squeeze parameter. As you might gather from its name, this allows you to place your hand around the Drumbeam and literally squeeze the playing surface to send out Continuous Controller, pitch-bend or even note data, similar to the methods employed by the Roli Seaboard keyboards. The LEDs will pulse in unison with the physical interaction and, with careful adjustment of the sensitivity parameter and a little practice, you can add feeling

and expression into a performance, alongside a visual element. The MIDI Continuous Controller data sent from the Drumbeam is based on positional information. This is something I first encountered in the Korg MPS-10, which I reviewed in SOS last year (www. soundonsound.com/ reviews/korg-mps-10). Rather than the value of the controller data being determined by velocity, it is dictated by where the pad is struck within a zone — a far-left strike sends a value of 0 and a far-right one a value of 127. As you strike a zone from left to right you can open a filter, sweep an EQ or affect any parameter of a connected instrument that can respond to a CC command. Up to eight presets can be stored in the Drumbeam itself (and even remotely selected via Program Change commands), but you can configure up to eight banks within the app. Selecting a bank within the app instantly reconfigures the Drumbeam to that combination of eight presets, which can then be stepped through using the buttons on the underside of the unit.

Heavy Hitter Overall, the Drumbeam is a real joy to play, and its design is a breath of fresh air, allowing you to

Bang On: Feel & Dynamics AFK Audio have done extensive research into playing surfaces, and the thick silicon option they have chosen feels great when played with drumsticks. The rebound is natural, and adjustment of the sensitivity setting within the app allows for articulate playing, with fast strokes captured accurately. Dynamics translate well but perhaps are not as nuanced as you might find with a dedicated percussion controller or piano keyboard. Having said that, I don’t think the subtle articulations offered by a mesh drum pad or SPD-SX type of controller are where the Drumbeam’s strengths lie. Regardless of the number of zones you create, there is no crosstalk between them. Even with a full 12 zones and no blank LEDs separating them, each zone plays perfectly although it certainly helps, if only visually, to have some demarcation between zones.

The app’s Segment view is where you can edit the pad zones and set the LED colours.

configure it in almost any way you can imagine. The option to assign your own colour scheme to each preset is also a real bonus. Roland, Alesis and Korg have introduced this feature in their latest percussion controllers, and I’ve found it invaluable as a way of identifying which sound is on which pad. Along with the audience-facing animation options, it also makes the Drumbeam look very cool from a live performance perspective. Configuration is easy via the neatly designed app, and the Wi-Fi connection means you don’t need to be physically attached to a computer to perform edits. Although the Drumbeam’s name appears to place it firmly in the percussion controller arena, the ability to assign MIDI notes and play melodies adds a very useful extra dimension. There’s something about using a controller so far removed from a keyboard that makes you play differently, which potentially stimulates creativity. I can see the Drumbeam having a place in the studio but, with its innovative design and the visual appeal of the LEDs, I feel it’s far better suited to live performance. It would work well as an electronic hybrid addition to an acoustic drum kit, and also as a way for DJs or keyboard players to play simple parts, open filters, add samples or trigger loops live. At around £400, it isn’t a cheap option, especially as it generates no sounds of its own, but it is well made, highly configurable and will suit many different types of performer and applications. Its flashing, sparkling or beaming LEDs add the bonus of an eye‑catching light show on a darkened stage. $ €429 W www.afk-audio.de www.soundonsound.com / February 2026

ON TE ST

Rupert Neve Designs OptoFET Dual-stage Compressor

With two separate compressors on board and various ways to deploy them, RND’s new OptoFET is an intriguing prospect. M AT T H O U G H TO N

espite the physical resemblance to its stablemates, the new OptoFET is a pretty bold move for Rupert Neve Designs (RND). Rupert Neve’s classic diode-bridge and VCA compressor designs are well known, of course, but this one is based around two other compressor types: one FET and the other optical. That’s a tried-and-tested a combination in music production — for years, many engineers have partnered an 1176 with an LA-2A, for vocal processing in particular. But the OptoFET isn’t simply a case of nicking that idea. I’m told their R&D team have been toying with different FET compressor designs for years (even if this is the first they’ve released), and they’ve included multiple routing options here, as well as implementing two new ‘character’ options, called Grit and Bloom.

Controls & Construction As you’d hope of gear at this price (very much in ‘professional’ territory, though a keener one than some others in RND’s range), the 1U 19-inch rackmount OptoFET’s build quality is flawless. The case and controls are reassuringly solid, with plenty of finger space around the knobs (all turning 31-position detented pots) and buttons. Visual feedback is

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instantly informative, and the painted finish oozes class. None of which will be a surprise if you’ve used RND gear previously! On the rear, there’s an IEC C14 mains inlet (100-240 Volts AC, 50 or 60 Hz) with an integral power on/off switch. There’s a power indicator LED on the front too, but no switch; ideally, I’d like a second switch there to allow powering off when my studio rack is powered up. Back on the rear, as well as a ground-lift slide switch, you’ll find a single Neutrik Combo XLR to route line-level sources to the electronically balanced input stage, and a male Neutrik XLR to deliver the custom-transformer balanced output. That arrangement is perfectly normal for mono processors like this, but it does mean you can’t simultaneously deploy each compressor on different sources. Such a facility would have added to the unit’s versatility, but the additional I/O would also have increased the asking price, particularly with a second transformer-balanced output. There’s no external side-chain input, and no provision for stereo linking either, which RND say is because matching the optical cells to the precision they’d want is nigh on impossible! On the front, working from the outside in, there’s a ±12dB input gain knob with

signal-present LED on the left (green, turning red when approaching clipping), and a ±12dB master output level control on the right, whose eight-LED output level meter ranges from -10 to +22 dBU. Inside those I/O controls, visually separated by white lines, are the FET compressor controls on the left, those for the opto compressor on the right and, between, them a central section with the routing controls. Those routing options need explaining before we can explore the performance of, and potential applications for, this processor. At the top-left corner of this central section, a backlit hard-wired bypass button takes everything, including the input and output stages, out of the signal path. Alongside this, another button labelled X-Over In determines whether the OptoFET’s compression stages are configured to act in series on the whole signal, or independently on low and high frequency bands. In the latter scenario,

Rupert Neve Designs OptoFET $1999 pros • Great technical performance. • Subjectively appealing compression. • Versatile routing and band splitting. • Grit and Bloom increase the appeal.

cons • Can’t deploy the two compressors on separate sources.

summary Two high-quality compressors in one, with versatile routing and character options to boot. What’s not to like?

a knob immediately below sets the crossover filter frequency (50 to 500 Hz). A third top-row button, labelled Flip, swaps where the compressors are deployed. If in series mode, engaging this puts the opto compressor first in the chain, while in X-Over mode it determines which compressor is assigned to the low band and which to the high. Two quartets of LEDs below, one either side of the frequency knob, indicate the current assignment for each compressor: there’s an LED each for stages A and B in series mode, and for HF and LF in X-Over mode; the current selection lights green. It’s a neat, intuitive system that’s also quick, since there’s no manual patching involved. Each compressor’s control set is almost identical. To the right of their green backlit in/bypass button is a dark-blue threshold knob, though the two have different ranges: -15 to +20 dBu for the FET, with -5dBu at the top position; and -10 to +20 dBu for the opto, with -8dBu at the top. Silver-coloured knobs

govern the attack (0.1 to 50 ms for the FET, 5-50 ms for the opto) and release (50ms to 1.5s for both) times, and each processor has a wet/dry blend knob too. A button toggles between two ratio options (4:1 and 8:1 for the FET; 2:1 and 5:1 for the opto, though the opto’s ratio actually varies somewhat with the signal level). These ones are the only buttons on this device that don’t light — usually, you can see easily enough when these buttons are depressed, but my ‘moody’ studio lighting sometimes made it trickier. A further (green backlit) button switches in a 125Hz, first-order high-pass filter for the side-chain signal, while another (yellow) one engages the Grit function for the FET compressor, or Bloom for the opto. Finally, we have an eight-LED bar meter above each compressor’s controls to indicate gain reduction (1-22 dB).

True Grit So what does this thing sound like? Why have RND put two compressors in one box, and what applications do all those routing and character options open up? To answer such questions, I spent a few weeks auditioning a pair of OptoFETs on several active projects. But first, even before dialling in any compression, I should spell out very clearly that it sounds great: full, clear, and with all the details in the source being conveyed with precision. Patching in just the FET compressor, I began by experimenting with some vocal compression, before turning my attention to bass guitar, acoustic and electric

guitar, and acoustic and electronic drums. This compressor, which has a soft knee, served admirably in the vocal smoothing role. With a 4:1 ratio and a long-ish attack and release, it delivered elegant control of male and female voices, and evened out the dynamics of guitar and bass parts nicely too, without suffocating note onsets. Switching to 8:1, I found more scope to shape the envelope of bass guitar notes and individual drums. It can be really fast too, such that you can clamp right down on transients — a tactic that can be really useful when you bring the mix control into play. While you have only two ratio choices, not the four of the 1176 (never mind its ‘all buttons in’ trick), this isn’t really trying to be an 1176, and there are further factors to consider in any case. First, there’s that mix control, for parallel compression. Second, there’s still a whole other compression stage to play with, should you wish to get more assertive. And third, there’s Grit... Before I looked too deeply into this unit, I’d imagined that Grit might be a dedicated saturation circuit, but it’s more sophisticated than that. When not engaged, there is (for both compressors) some smoothing of the side-chain signal going on behind the scenes. Switch in Grit, though, and this is removed, leading the FET compressor to act faster and more assertively, introducing some upper-mid and high-frequency harmonics in the process. This can lend energy and urgency to proceedings, and usually without making things sound overcooked. But push the compression harder and it becomes more obvious, and again, I particularly enjoyed doing that and using the mix knob to blend this in as an effect.

The central control section: this is where you decide where the two compressors will act on your signal; one before the other, or each on a different frequency band. www.soundonsound.com / February 2026

ON TE ST

Aguilar Plugin Suite

Bass Rig Modelling Software

Screen 1: The main amp/cab home screen.

For this plug‑in, boutique New York bass specialists Aguilar teamed up with modelling experts Korg — and the results are impressive... PHIL WARD

espite being an enthusiastic and busy bass player, I have to admit that when it comes to recording my basses I’ve never really been all that sophisticated or adventurous in my methods. I’ve played with a few amp-and-cab modelling plug-ins but never been entirely convinced or particularly inspired by them. Instead, I’ve stuck to a tried and trusted technique that involves DI via an old hardware valve compressor (configured to provide valve‑based colour more than actual compression), followed by plug-in based EQ, compression, optional dirt, and sometimes a taste of very short reverb to add some air and

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room sound. Being asked to review the Aguilar Plugin Suite, however, had an element of ‘kid in a sweet shop’ about it. Where to start? What will I like? If you wanted to pick a brand to embody the idea of a boutique backline manufacturer, you’d not be too far away with Aguilar. Founded in 1995 in New York, they began as specialist manufacturers of bass amps and cabs, but have since expanded their product range to include bass effects pedals, pickups, and internal guitar preamps (used by a good number of boutique bass luthiers) — and now, obviously, plug-ins. You don’t see many Aguilar backline rigs at, say, metal or indie guitar gigs, but at jazz gigs or down in a musical theatre pit, Aguilar have

secured a significant niche. And while their hardware is definitely not priced at the entry-level point, it’s also not soooo

Aguilar Plugin Suite $200 pros • Easily accessible great bass sounds. • Huge creative possibilities. • Intuitive and easy to use.

cons • None.

summary The Aguilar Plugin Suite offers a wide range of high‑quality and inspiring bass tracking options. Very much recommended.

expensive that it’s completely out of reach. As I write that, I’m glancing at my venerable old Gallien‑Krueger combo, wondering how much an Aguilar amp and cab would cost me... but I digress!

Suite Backline The intent behind the Aguilar Plugin Suite is to model a range of Aguilar amps, cabs and pedals. Despite being billed as a plug‑in, it can actually also be used as a standalone application, and in that mode it offers MIDI for patch selection and, I guess, the possibility to be used in a live environment, assuming you have a sufficiently low-latency computer and audio interface setup. The core modelling technology of the plug-in is licensed from Korg, and it’s used to render 13 different Aguilar cabinets, ranging from a 1x10 to an 8x10-inch and

including various 12- and 15-inch designs. There are also three different Aguilar amps, and nine Aguilar stompbox emulations. The library of pedals includes two compressors, a twin filter envelope, an octaver, a chorus, a phaser and three different overdrive/fuzz options. There’s a wide range of presets, some created by Aguilar-endorsing bass players of note such as Blu DeTiger, Derek Frank, Keithlen Foster and Lex Sadler. Some patches comprise amp and cab combinations, while others offer amps alone for DI modelling. Some also include various chains of virtual effects pedals. Along with being useful in their own right, factory patches also work well as launch pads on which custom patches can be Screens 2 and 3: Pop‑up selector windows allow you to browse the available amps and cabs.

www.soundonsound.com / February 2026

Creativity Amplified

Built on decades of innovation, Fender Studio® brings industry-leading software, meticulously engineered hardware, and legendary tone into a powerful music creation ecosystem for the next wave of music makers.

© 2026 FENDER MUSICAL INSTRUMENTS CORPORATION. FENDER® (STANDARD AND IN STYLIZED FORM) AND THE DISTINCTIVE HEADSTOCK SHAPES COMMONLY FOUND ON THE FENDER® INSTRUMENTS ARE TRADEMARKS OF FENDER MUSICAL INSTRUMENTS CORPORATION AND/OR ITS AFFILIATES, REGISTERED IN THE U.S. AND OTHER COUNTRIES

Empowering artists to take their ideas from from first riff to final release. Fender Studio® Pro Meet Fender Studio Pro, the next evolution of modern music creation. Built on PreSonus®’ award-winning Studio One® platform, Fender Studio Pro combines its celebrated features and trusted performance with a modernized interface, Fender’s renowned amps and effects models, and intuitive new workflows that make creating music faster, more inspiring, and more connected than ever.

Quantum LT Interfaces The Fender Quantum LT Series delivers professional recording results with effortless workflow, combining incredible sound quality, flexible connectivity, and seamless integration with Fender Studio® Pro in three streamlined, affordable interfaces.

Motion Production Controllers Fender Motion Controllers put Fender Studio® Pro at your fingertips, giving instant access to timeline navigation, session controls, and an Add button for instruments and effects—so you can record, edit, and perform without leaving your instrument or breaking focus. Beyond Studio Pro, Motion’s dynamic FATAR® pads, vibrant illuminated encoders, and expressive touch strips elevate any DAW workflow, letting you improvise with intelligent scales, explore iconic chord progressions, and shape sounds in real time. Coming Spring 2026. © 2026 FENDER MUSICAL INSTRUMENTS CORPORATION. FENDER® (STANDARD AND IN STYLIZED FORM) AND THE DISTINCTIVE HEADSTOCK SHAPES COMMONLY FOUND ON THE FENDER® INSTRUMENTS ARE TRADEMARKS OF FENDER MUSICAL INSTRUMENTS CORPORATION AND/OR ITS AFFILIATES, REGISTERED IN THE U.S. AND OTHER COUNTRIES

ON TE ST

SPL P8

Eight-channel Microphone Preamplifier

You don’t always want ‘colour’ in a mic preamp. In fact, there’s a strong argument for making them sound as transparent as possible... PAUL WHITE

he market for microphone preamps is an interesting one. Some designers aim to add character and warmth by deliberately colouring the signal with valves, audio transformers and/or other circuitry in the signal path, while others strive to produce the cleanest, most transparent signal path possible — the proverbial ‘wire with gain’ that adds nothing and takes nothing away from your precious signal. SPL’s new P8 preamp falls into the latter category, utilising a discrete transistor preamp circuit (rather than the more usual op‑amp hybrid topography) to offer a flat and wide frequency response, very low noise and minimal distortion. Its circuitry is designed to provide plenty of headroom, and there’s enough gain range here to allow the preamps to handle anything from hot active microphones to low-output passive ribbons.

Pieces Of Eight The P8 is a 1U rack module containing eight identical microphone preamps. SPL have a long track record of designing high-quality microphone preamps, of course. I owned their excellent GoldMike dual-channel mic pre and Channel One single‑channel unit many years ago, both of which included valve circuitry to add a little flavour, and I rated them highly. Since then, the company have explored a number of different approaches to mic preamp design, including the high-voltage Crescendo 8. With the P8, SPL have taken what might be described as a more audiophile approach, with a design that, while not close to being at what I’d consider the budget end of the market, is

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more affordable than — yet arguably just as highly specified as — lots of single-channel boutique mic preamps. SPL’s fully discrete preamp circuit was designed for wide bandwidth (fast slew rates) and low distortion. Importantly, it maintains a musical character when it’s pushed close to clipping, but essentially the circuit’s function is simply to make the mic signal bigger, rather than to change it. SPL’s designers explained that the preamp consists of a total of eight transistors for signal processing, and five transistors in the current mirror circuit. An FMB2222A dual bipolar NPN transistor package handles the differential input stage, and the thermal coupling helps optimise common mode rejection. BCP56 and BCP53 transistors are employed in the output stage to deliver the wide frequency response necessary to handle fast transients. The rear panel hosts eight Neutrik XLR connectors for the mic inputs, along with two DB25 D-Sub connectors (wired to the AES59/Tascam spec) for the line outputs. There’s also an optional ADAT interface that allows you to use the P8 as an audio interface expander, and this has dual ADAT ports for S/MUX support of higher sample rates. As standard, the DB25 output is linked to a second DB25 connector allowing the signal to be split without the need for an external splitter, but this has to be removed in order to accommodate the ADAT card. The unit sent to me for review model came with the ADAT interface already fitted.

Panel Games On the front, the styling is unfussy and workmanlike, with the eight identical

preamp control sets arrayed across the front panel. Each channel has a gain control, plus four illuminated push buttons and a three-LED level meter. Located toward the right of the panel are a sample-rate selection button with status LEDs, and another LED for word clock sync. The gain knob covers the range +8 to +62 dB, and there’s a switchable -20dB pad to deal with particularly hot signals such as close-miked drums, brass or electric guitar amps. The switchable 80Hz high-pass filter has a gentle first-order (6dB/octave) slope. Phantom power conforms to the standard 48V and is available separately for each channel, as is polarity inversion. Operating the switches doesn’t produce the clicks and pops that you hear from some less sophisticated designs, and in this design transistor circuits are used to turn the phantom on and off. Given the 1U format, large-scale meters would be impractical with so many channels and controls to accommodate, but the three-LED system used here is sufficient to let you know what’s happening, and most of us will have more accurate metering available on the receiving device or in our DAW software. Since there are no insert points to consider, connecting the P8 up is straightforward. The design also deliberately omits DI jack inputs — ideally I would have liked to have had one or more, but it could be argued that fewer connections helps to maintain the cleanest signal path. Some users may also bemoan the fact that the XLR inputs are on the rear panel, since that might require using a patchbay or stagebox to access them in a studio, but really it’s the best way of fitting eight preamps of this quality into a 1U rack space and there are obvious

sync signal is present. There’s also a USB port for handling firmware updates and other maintenance functions (not for use as an audio interface).

Performance

SPL P8 From $2249 pros • Extremely transparent, quiet circuitry with very wide bandwidth. • Optional ADAT card. • Takes only 1U of rack space.

cons • No USB interface capability. • No line/instrument inputs.

summary A superbly clean and detailed eight-channel preamp that delivers transparency rather than adding character.

advantages in keeping the cables out of the way of the controls too. The optional ADC8 ADAT board has dual ADAT outputs plus BNC connectors for word clock in and out. It uses AKM’s AK5538 converter chip, providing a well-specified ADAT optical output capable of up to 192kHz using S/MUX, which makes it possible to carry four channels at 96kHz or two at 192kHz over a single Toslink cable, or twice as many if using dual ADAT connections (so the maximum sample rate for running all eight channels over two cables is 96kHz). Many audio interfaces are S/MUX compatible, so a P8 fitted with the ADC8 board could be a very practical addition to a DAW-based setup, and setting it up as an expander for my PreSonus interface was plug-and-play simple — other than a mains cable, I just needed two optical cables to get set up. Precision quartz oscillators provide a low-jitter master clock but are automatically disabled when an external

The specified frequency response extends from 15Hz to 200kHz, and although obviously nobody needs anything like that range to be able to hear the detail in hi-hats and acoustic guitars, having a wide, flat response ensures phase linearity is preserved across the audible spectrum. Crosstalk is specified as being below -90dB at 1kHz, with a -30dB input signal and 30dB gain. Equivalent input noise is quoted around -124dBu — that’s very close to the theoretical minimum. THD (total harmonic distortion) plus noise is negligible at low to medium gain settings, and distortion doesn’t increase excessively unless the preamps are pushed to the verge of clipping. The quoted figures are THD+N: 0.0025% at 1kHz with a -30dB input signal and 30dB gain. Common mode rejection is quoted as 78 to 80 dB at 1 kHz, which means that balanced mic cables of the lengths used in typical studio installations will be effective in rejecting unwanted interference. As those specifications would suggest, these preamps have an open, transparent character — or rather lack of character! — that allows transient detail to come across cleanly, with a very strong sense of focus. That bodes well not only for drum tracking but also for recording instruments such as acoustic guitar, which can generate very strong and harmonically complex transients. Extended bandwidth and phase integrity are also beneficial when handling arrays, such as Decca Trees. Testing the preamps with a selection of my acoustic stringed and percussive instruments confirmed the sense of transparency — it’s almost like the mic preamp wasn’t there at all! If you’re longing for a preamp that sounds warm and cuddly, then the P8 probably isn’t for you. But my take on preamps is that while you can always

The ADAT card is optional — when not fitted, the P8 instead carries a duplicate set of eight line outs on a DB25 connector, allowing it to function as a splitter.

add ‘dirt’ to a clean signal, you can’t add ‘clean’ to a dirty one. The P8 endeavours to make itself invisible so that whatever comes out of the microphone reaches its destination without being changed, and my experience is that it succeeds in that: it captures the character of the connected microphone accurately, without adding any further colour. And even a relatively quiet sound source subjected to compression didn’t bring up any significant noise; in a typical studio situation, the microphone is likely to contribute rather more noise than this preamp. If you’re looking for preamps to use on location, this unit weighs just over four kilograms, and if using a flight case where both the front and back are removable, the rear-facing inputs will be easily accessible and helpfully out of your way. The optional ADAT card is probably the most straightforward option for mobile recording to a computer/interface setup, so you should probably budget for that. It would have added flexibility if SPL had included audio over USB for recording, since that would allow any existing interface to be bypassed, but using the ADAT output with a compatible interface works perfectly well.

Verdict To summarise, the P8 is not about character or warmth: this is a mic preamp built to last and to deliver optimal transparency, combined with high headroom and very low noise. Yes, it’s a little more costly than some of the eight-channel mic preamps by companies such as Focusrite, PreSonus and Audient, but even though ADAT connectivity is a cost option it’s more keenly priced than SPL’s own Crescendo 8, which runs on 120V power rails, and I think the price is very reasonable given the quality of what’s on offer: you get eight seriously clean, capable mic preamps that can do justice to any type of microphone, including passive ribbons. $ Standard version $2249. With ADAT card $2799.

T SPL Audio +49 (0) 2163 98340 E info@spl.audio W https://spl.audio

www.soundonsound.com / February 2026

ON TE ST

JBL PRX925 & PRX935

Active PA Speakers

MIKE CROFTS

ack in SOS July 2023 I reviewed the first models in JBL’s then‑new PRX900 series of self-powered speakers. I liked very much what I saw and heard and, assuming they would continue to update the range as they had for the PRX800s, was looking forward to trying out the larger models. Well, the PRX925 and PRX935 were recently announced and are now available for purchase, and I have had a pair of the PRX935 units on loan for long enough to put them to good use in a number of different live scenarios. The nice people at JBL’s UK distributors Sound Technology were kind enough to put on an extensive demonstration at their facility, where I also got to hear some of the mighty top-end JBL rigs which were all set up and ready to go (it would have been rude not to, really...).

Family Values The PRX925 and PRX935 are the larger members of the PRX family, and follow the form factor of their predecessors in

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The two big beasts of JBL’s PRX900 family are similarly sized and identically priced — but they’re very different speakers! We put them to the test. the PRX800 line-up, in that the 825/925 model is a two-way speaker with two 15-inch woofers, and the 835/935 is a three-way, with a single 15-inch woofer and a six-inch midrange driver (both sport 1.5-inch compression‑driver tweeters). I know that this family grouping goes back at least to the PRX600 series, as I still own and regularly use a powered system comprising a pair of PRX635 three-way tops over a pair of PRX618XLF subwoofers. I use this combination (or sometimes just the PRX635s on their own) for community theatre shows, larger school

productions, church services and other events where I have lots of acoustic instruments and different voices to work with, as I have always found that the three-way design works really well — and therefore makes my life easier — when clarity and smoothness are needed above all else. As I had a particularly challenging gig on the horizon that would require exactly these qualities, I chose to take away a pair of PRX935s, but not until both of the new models had been given a thorough going-over in the demo room! I covered most of the technical details in my first review of the PRX900 series (you can read that at https://sosm.ag/ jbl-prx900), but it’s worth noting that the power modules fitted to all PRX900-series tops are essentially the same (the subwoofers have their own variant), in that they all provide exactly the same functionality and power amp capability, but tuned to the particular model in which they installed. This suggests an efficiency and cost-saving in design and manufacture, which should benefit users down the line in making repairs more straightforward if they are needed. On that subject, it’s also worth pointing put that JBL have such confidence in the product line that all PRX900-series models come with a seven-year warranty (yes, seven years!).

I for one would be pretty reassured by that, even if I hadn’t personally owned and operated various JBL Eon, JRX, PRX, SRX and VRX speakers over many years without a single failure in all that time.

Working 925 Before taking a pair of PRX935s away to review, I had a bit of a play with the slightly larger PRX925 in the demo room too. I was able to try both models in various configurations with different DSP settings, both with and without paired subwoofers. I listened to the same programme while A/B switching between them and with all the rear‑panel controls set exactly the same. I was expecting them to sound different, but I didn’t think there would be so much difference in sound character between the two models. The PRX925 delivers an astonishing amount of impact for a two-way twin‑15‑inch box, and it’s voiced to provide a big, thumping, solid output that would sit very well in DJ territory and suit your more ‘forward’ rock bands. Of course in a demo room you have the ability to run things quite loud if you feel so inclined and, as I felt so inclined, we gave the speakers a serious hammering with all sorts of test material from dance and rock, to tight thumpy ’70s, some very full-bodied Berlioz, and anything else that came to mind. We ran the PRX935s right up to limiting and then played around some with the limiter settings themselves just to see if we could cause any problems. The man from Harman just waved his hands encouragingly and said ‘do what you like, it’s what we do’, and I was impressed with how clean and clear the output remained well into the zone of discomfort. In their natural state with the DSP at factory settings, you should be able to produce full output from the cabinets, and the onboard protection cleverness

The PRX925 and PRX935 are both 45cm wide and 43cm deep, but the former is 106cm tall and the latter slightly shorter, at 939cm. Despite their similar sizes, they have very different driver configurations, with the 925 being a two‑way speaker with twin 15‑inch woofers, and the 935 a three‑way with a dedicated midrange driver.

will kick in if it needs to but without affecting the material unduly unless any kind of fault condition is detected. As I understand it, the drivers and power stages are both fully monitored and protected against accidental or deliberate mismanagement, and the DSP will handle anything you can throw at it so long as it’s connected up properly to start with.

Three & Easy The PRX935 is not intended simply to be a smaller, more polite version of the PRX925. As well as the obvious three-way speaker configuration, it uses a completely different 15‑inch driver (JBL 275G versus the 925’s dual 915H drivers), which is designed to work more like a sub driver and provide a more rounded, fuller low end, working in conjunction with the dedicated mid driver to give an overall smooth output for contexts where clarity and especially vocal balance and projection are the top priorities. That’s not to say that the PRX935 is in any way lacking in volume, because it still produces a hefty 136.1dB SPL output (the two SPL figures are not that far apart), but maybe it does it with different listeners in mind! One of the PRX900 series’ coolest new features is the adoption of the JBL Pro Connect Bluetooth LE remote control function, which allows remote user control of the full DSP suite built into the speakers, plus muting, grouping and identification. Detailed equalisation is fully controllable using either the back panel LED screen or with a compatible BLE device, and the PRX935 has limiting, delay and

JBL PRX925 & PRX935 $1899 each pros • PRX925 offers a huge amount of impact and ‘thump’, and would suit DJs and rock bands very well. • Three-way PRX935 model excels at smooth midrange and vocal delivery. • Unified JBL Pro Control app makes remotely configuring speakers easy. • dbx Driverack processing on board.

cons • None.

summary The PRX925 and 935 sound surprisingly different but both are excellent performers, with the former offering perhaps more attitude and the latter better suited to live acoustic instruments and vocals. With their extensive DSP and improved remote control functionality, they are both useful updates to their PRX800 series equivalents.

automatic feedback suppression processing taken from the dbx Driverack, all built in and ready to go. After the speaker is paired with the Android/iOS control device (which I found to be a straightforward, quick and bug-free process), running the JBL Pro Connect app can put you in control of up to 10 individual speakers from any compatible JBL range, including Eon One Mk2, PRX One and Eon 700 loudspeakers. In contrast, while the previous PRX800 models also offered remote control, they did so over Wi-Fi and required a PRX‑specific app. Pro Connect provides full access to the whole DSP set, and individual speakers can be muted and given names for easy identification within a system. If you forget which speaker you’re working on, there’s a useful ‘here I am’ function that flashes the LED on that particular speaker. Using the app it’s also possible to store snapshots of speaker settings and group setups. For direct hands-on management, all the usual controls and connections are on the rear panel, and should offer enough flexibility for most setups; there are two combo XLR/TRS inputs, an aux 3.5mm stereo input, two XLR thru outputs, and a balanced XLR mix out. The controls are simple and intuitive to operate, and the bright, clear LED screen displays the parameters under adjustment at any one time.

On The Outside The external styling of the PRX range has stayed fairly similar as new upgraded versions have been introduced, and the www.soundonsound.com / February 2026

ON TE ST

Akai MPC Live III

Sampling Workstation

February 2026 / www.soundonsound.com

The Live III is the most powerful and flexible MPC yet... SIMON SHERBOURNE

kai have reclaimed their position in the sampling workstation game over the last few years, with the portable MPC Live, the classic‑format X, the affordable One and two keyboard models all hits, not to mention a steady stream of feature updates and plug-ins. The competition has struggled to answer these in terms of horsepower (Maschine+) and functionality (Push 3). Now, the Live III appears to pull comfortably ahead of the pack with significantly upgraded internals and connectivity. But it’s not all about hardware. The recent version 3 OS, and now the Live III, bring big changes to the way MPCs work. Could this put off long-time users and make the MPC overly complex, or is it a much needed refresh that propels the platform forward for a new generation?

Live Rundown There’s so much new stuff to cover here, but let’s recap quickly. The modern MPCs are versatile production stations with an open, DAW-like track structure hosting drum and one-shot kits, multisampled instruments, plug-in instruments, effects, MIDI and CV tracks. They are unique in the category in offering open-ended linear audio tracks. You can create or perform with an MPC on its own, but they are also great as hubs for other MIDI, USB, CV and audio devices. You can even connect audio and MIDI interfaces to extend their I/O. Today’s MPCs are capable standalone devices, but they can also work as control interfaces for the MPC app/plug-in.

The MPC has its own terminology and ways of working that connect back to the original units, with Tracks, Programs (kits/instruments) and Sequences (song sections) and lots of pad and screen modes for doing myriad things. These concepts are still part of the MPC, but there have been many developments in the last year. Now, back to the Live III...

Function Over Form The new MPC has sprouted a host of additional buttons, and is slightly wider to accommodate a touch strip like the MPC Studio. I’ve seen some body-shaming around its busy appearance, but would you prefer fewer controls so it looks slicker? More dedicated mode buttons like on the X and One is good news. A 16-button strip adds (among other things) a step sequencer array to the MPC for the first time. Here I’ve read opinions insisting this should be along the bottom like on a traditional drum machine. Think again: an instrument that’s primarily going to be finger-drummed has no business having buttons in front of its pads. The Live has always been chunky. It’s not really backpackable, but you can pick it up, move it about and play with it on your lap. The internal battery makes all the difference, and it’s a great safety net in a live situation. The speaker array is perfectly usable for jamming and there’s now a built-in mic, so you really could show up in many situations and create tracks and demos with just the Live III. Even better, my top wish for the I and II has been granted: a combi input with mic pres and phantom power, so you can plug ‘proper’ mics in as well as line sources.

Powerful Connections MPCs were always well connected, and the Live III pushes this to new heights. Expanding on the previous model’s dual MIDI I/O, six audio outputs, USB host ports and turntable inputs, the MkIII gains two combi inputs with preamps and phantom power, and a USB-C port that can connect 24 channels of audio both ways with your computer or iOS devices. The eight CV channels have been given new powers with the ‘CV Playground’: a set of sequencing and modulation modules available on CV tracks. The only things still unique to the MPCX are dedicated Hi-Z instrument inputs.

Akai MPC Live III $1699 pros • Massive hardware spec boost. • Clip Matrix and step‑sequencing workflows. • Articulations. • Mic inputs. • 24x24 USB I/O.

cons • Matrix mode has some gaps. • The UI is still quite fragmented. • On‑screen note editing still feels clumsy.

summary The MPC Live III is an incredible upgrade in power and functionality; it’s hard to think of any other dedicated, standalone device this capable for end-to-end music production.

Otherwise the panel has the same display, indented data wheel, and four touch‑sensitive encoders. The buttons are the clicky rubberised designs that I like. Most buttons have a primary and secondary function, the latter of which can be reached with the Shift button, or more easily with a double-tap. Internally, the Live has had a phenomenal power-up. It’s now built around an eight-core ARM processor with 8GB RAM, compared to the II’s quad-core and 2GB. Everything is significantly snappier, like you’re working in controller mode with a computer. The horsepower raises the limit on audio tracks from eight to 16, and plug-in instrument tracks from eight to an incredible 32. Total polyphony jumps from 64 to 256 stereo voices. The largest 22‑track demo session hovers just over the 20 percent CPU mark. Internal storage is 128GB compared to the II’s 16GB. You can still install an additional SATA drive in a bay underneath. This seems mostly unnecessary now with so much standard capacity, but I popped my 160GB SSD in mainly as an instant way to bring in my existing samples and expansions.

Corner Kicks The Live III debuts new trigger pads with distinctive, segmented corner lights. These

www.soundonsound.com / February 2026

ON TE ST

AK AI MPC LIVE III

pitch ramps and tons more. Expect to hear these spicing up the hi-hats and snares on beats over the next year.

and simple way to input dynamics. All other per-step settings are tweaked in a dedicated Step Edit mode. Tapping a step here brings up a set of available step parameters with the basic options — velocity, ratchet, probability and timing — set from the encoders. A second set of ‘modifiers’ include envelope adjustment, filter control, and sample slice selection for quickly tweaking sequenced chops. There’s also tuning, which brings a fast way to program melodic sequences in drum programs without converting the channel to a Keygroup track. Maybe unique is the choice over whether step automation affects parameters in absolute or relative fashion. While step edit offers a new way to quickly write a limited set of key parameters, it’s still possible to use the legacy step sequencer view to enable and set automation for any other track or mix parameter. This is one of the examples of how the MPC has layers of overlapping functionality that have built up over generations. Note Seq mode uses the SH-101 convention of advancing the step position as you tap in each note, except it’s polyphonic and ideal for entering chords. A cool feature of this mode is that with Timing Correct off, the exact timing of each note in a chord will be captured, offset from the first played note. I was less enthusiastic about how Step Edit and Note Length modes work for note sequences. It feels a bit unintuitive and non-visual how to find and adjust individual notes. I found using a combination of the step sequence buttons and regular grid mode helpful. It’s easy to see what’s going on in the grid and make adjustments, but selecting notes is still the weakest area of the touch UI, and the buttons help here. I still wish we had a select mode for the wheel as on Maschine, or even Bluetooth mouse support.

Radical Steps

What Is The Matrix?

The other big story of this release is an embracing of drum machine and production workflows not typically associated with the MPC. Step sequencing was available before via a screen/pad mode, but now you can work in classic TR style using the 16‑button strip. Tap a pad to target a sound/ note and drop in gates from the buttons. You can also hold active steps and use the touch strip to set velocity: a fast

The other big lean into modern production and performance workflows is Matrix mode, aka the clip launcher. This represents the final unification of the Force and MPC OSes, with both devices now incorporating the other’s previously contrasting workflows. The Matrix uses Ableton’s clip grid convention to capture, organise and play multiple MIDI and audio clips per track. You can load loops and audio files into clip slots, record and overdub directly into

The Matrix page is central to Live III’s new clip-based workflow.

are dubbed ‘MPCe’, presumably a play on MPE and with the ‘e’ also standing for ‘expressive’. These feel quite different to the previous generation. They have the familiar grippy texture, but are slimmer and more sensitive. The key new feature is that they know where on the pad they are being pushed or struck, an ability that’s been exploited in a number of clever ways in MPC drum programs, but not in any other areas of the workstation so far. You can trigger different samples from the corners of the pad, so you could treat the 4x4 layout like an 8x8. You can also trigger a different sample from the centre of a pad to the corners. And drum programs support eight samples per pad, which you can layer among the zones as you wish. The zones don’t have to be discrete — you can crossfade samples in various ways across the pads. You could set up a varied response that simulates hitting a drum head in different places — like a snare moving to a rim shot. You can also treat the pads like X/Y modulation generators, mapping the coordinates to any parameters in the drum channel as two expression inputs in addition to pressure. A multi-page mod matrix in the program editor lets you configure this in detail. Absolute X/Y positions are just one option, you can also track the distance from the pad centre, or from the point where you hit the pad. Position can scale other mods as well as performing them directly. The potential of the new pads is showcased in the demo patches in the dedicated MPCe Expansion. It is distinct

from MPE as we know it, and it’s not currently possible to access the modulation outside of drum programs. However, Akai plan to extend the functionality to plug-ins and keygroups later on, and the pads have the potential to do MPE if implemented in the future. Another innovation that takes advantage of the new pads, but which is still available on older models, is Articulations. This is just brilliant. Pads, or corners, can be assigned articulations, which trigger different playing techniques or accents like rolls, flams and ratchets. There are scores of these to choose from. Every flavour and timing of note repeat you can think of can be dropped in as well as paradiddles, flams,

“If you want a deep, one-stop-shop for writing, recording and performing it’s right here.”

February 2026 / www.soundonsound.com

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ON TE ST

Fiedler Audio Mastering Console 2 Dolby Atmos Production Software This major update to Fiedler’s immersive audio mastering suite could revolutionise your Atmos workflow. PHIL WARD

y review of Fiedler Mastering Console version 1 was published as recently as SOS December 2024, and the fact that I’m already writing about a version 2 indicates both that Fiedler Audio seem able to move very quickly, and that this new version is a very significant advance. In fact, Fiedler describe it as an entirely new product — and I wouldn’t argue with that. Fiedler Audio Mastering Console (or FMC, as I’ll refer to it from here) is a Dolby Atmos production tool that aims

February 2026 / www.soundonsound.com

to provide an all-in-one fix for a workflow quirk that’s inherent in Atmos signal routing: within, say, a Pro Tools Atmos mix session, it’s not possible to apply global EQ, loudness adjustment, compression or limiting, or anything else, to an entire Atmos mix. This arises because an Atmos session typically comprises beds and objects. Beds are conventionally panned elements spread, typically, over a 7.1.2 channel width, and objects are discrete elements that are panned according to dynamic metadata that defines their intended position in space. Notwithstanding that multi‑channel EQ, compressor or limiter plug‑ins that can be inserted on a 7.1.2 bus are somewhat niche,

Screen 1: The main Timeline window. Note the support for working on multiple files, and for Atmos-capable plug-ins.

such a plug‑in wouldn’t ‘see’ the object channels, because they’re separately routed to the Dolby Atmos Renderer (there can also be up to 128 of them!). This isn’t simply an inconvenience, though; it’s a significant issue because, for example, it makes aiming for a specific mastered loudness level or an overall tonal balance decidedly tricky. It makes fade-ins and fades-outs complicated too because they potentially need to be applied to beds and each object individually. FMC aims to fix all this, and more.

Orientation On launch, FMC 2 presents its Timeline window, which is where most of the fundamental mastering functions live. To begin a mastering session, the first step is to import an Atmos ADM (Audio Definition Model) file, a multi-channel BWF WAV package that carries the Atmos metadata, and is the Atmos format output of the Dolby Atmos Rendererer that hangs off the end of the DAW Atmos mix

Fiedler Audio Mastering Console 2 $399 pros • Provides a comprehensive Atmos and stereo mastering solution. • Huge array of features and functions. • OBAM format brings plug-ins fully into the Atmos landscape.

cons • None.

summary Fiedler Audio Mastering Console 2 is a major advance on version 1, and offers a genuinely gamechanging set of features.

session. Screen 1 shows six ADM files imported, because FMC 2 can master multiple files in one project. Despite the imported Atmos audio shown comprising a 7.1.2 bed with multiple objects, the display in the timeline shows ‘only’ a 5.1 channel width. While this may seem strange, it’s not remotely an issue in practice, and there’s sense in it: the convention is that loudness measurement of Atmos ADM files is determined on a 5.1 downmix. Along with the Atmos ADM shown in Screen 1’s Timeline window, I’ve also imported six stereo files (shown in green) that were rendered and output by the Dolby Atmos renderer at the same time as the ADM files. Mastering for Atmos often means creating stereo companion files and FMC 2 can handle both. This makes sense, as it will help ensure consistency across mastered stereo and ADM exports. There’s an option to consider here, though, because an alternative workflow is to import only the Atmos ADM and then, if stereo companion files are required, have FMC 2 render and export them. That was my preference initially, but I can see that the extra flexibility offered by importing separate stereo files, in particular the possibility of applying different processing (EQ, compression, loudness, limiting, etc) to stereo renders, is often likely to win the day. Still, importing separate ADM and stereo files, as in Screen 1, does enable me to illustrate the trim features and in/out fades that can be applied. The ADMs in the screenshot have been trimmed at both beginning and end, and had in/out fades applied (nonlinear fade curves are available), while the stereo files are untouched. Trims and fades are applied simply by grabbing the handles that appear automatically at the start and end of the timeline once a file has been imported. While I’m writing about the Timeline window, I should mention the functions to which it provides access. Living at the top left are the session import and save functions, and access to a Settings page that I’ll cover below. Next are transport controls and a switch that toggles the playback output between the ADM and stereo files (if one has been imported). Alongside this is a stereo file playback level control. Below the transport controls is a toggle switch for Batch and Album modes, the latter then offering Ripple and Snap options. Some explanation is in order... As I mentioned, FMC 2 is designed to enable multiple files to be imported, so that batches of tracks can be

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ON TE ST

M AT T H O U G H TO N

nalogue multiband compressors remain rare: they’re relatively expensive to implement, and the necessary user controls can take up a lot of space. Unsurprisingly, then, they’re rarer still in the 500-series format, and Wes Audio’s Pandora is the first I’ve seen that delivers anything like ‘conventional’ analogue three-band compression in this format. But it offers way more besides. As well as boasting other processing options that I’ll explore below, it can be remotely controlled by a DAW plug-in to facilitate recall, automation and preset management. Yet, while the plug-in is rightly a headline feature, pretty much everything can still be done using the touch-sensitive front-panel hardware controls too. In fact, you can use the hardware to control the plug-in and record automation as well. Importantly, this arrangement reduces the risk of the Pandora ever being rendered obsolete by OS or DAW developments, and (for whatever reason) a lack of software support in the future.

Hands On So let’s start with the hardware. The Pandora is W I D E! Though a stereo device, it occupies three module slots in the host 500-series rack: it uses only the leftmost pair for power and I/O but physically

February 2026 / www.soundonsound.com

Wes Audio Pandora Digitally Controlled Multiband Analogue Compressor

Already leading the way when it comes to plug-in remote control of analogue hardware, Wes are now bringing their expertise to bear on a wider range of processors. extends across the third, such that slot 3 can’t be used for other modules. There’s a lot of information on the front panel, but it’s all sensibly laid out. While the A and B preset storage/recall buttons (you can also copy settings from one to the other) are arguably harder to use when a USB cable is plugged into the front panel, you’ll likely be using the plug-in’s greater options for that function, and if it really bothers you then Wes’ Titan ng500-series rack can provide the necessary USB communication through the card-edge connector. The main hardware controls are either touch-sensitive digital push-turn encoders or buttons, with various tiny green or red LEDs indicating the status. Nicely visible without being over-bright, these are particularly effective at indicating the position of the rotary encoders: multiple

LEDs of different brightnesses combine to give really precise feedback, and I’ve not seen anyone do this better. Thin black lines group the controls visually, into one of the three frequency bands or for global functions. Each band, running from left to right, has a threshold encoder with a nine-LED gain‑reduction meter above (again, elegantly implemented). The threshold control is a push-turn encoder: turn to set the threshold, or push to bypass compression for that band. This is one of the touch-sensitive ones, and putting your fingers on it will highlight the corresponding controls on the plug-in when in use. The same goes for the output level control, for which a push solos that band. The remaining per-band controls are buttons that step through their different parameters’ settings. One sets the ratio

(10, 4, 2 or 1.5:1 for all bands), and another adds harmonic distortion (High, Med or off), which you can think of as a saturation effect. Further down, the attack (0.1, 0.3, 1, 3, 10 or 30 ms) and release (0.1, 0.3, 0.6, 0.9 or 1.2 seconds, or an automatic release) each have a pair of buttons, one to scroll up through the options, the other down. Below are various global controls. Between each pair of bands is a knob to set the frequency of the crossover filter. While these are also push-turn encoders, there doesn’t appear, at the time of writing, to be any action mapped to the push function. The filters range from 40Hz to 2kHz and 2-15 kHz, respectively, both with gentle 6dB/octave slopes. There are also side-chain high-pass (off, and 60, 90 or 150 Hz) and low-pass (off, and 3, 6 or 9 kHz) filters. A final pair of encoders, at the bottom, set the overall wet/dry mix and the output level, and in between them are the A and B preset save/recall and master bypass buttons.

Remote Possibilities The mini USB socket allows connection to a computer when the Pandora is used in a standard 500-series chassis, and its purpose is to connect with the control plug-in (communication is bidirectional — adjust a hardware control, and that’s reflected on the plug-in; adjust a plug-in parameter and the hardware follows suit). There’s no Ethernet, as on some manufacturers’ devices, and one potential drawback with USB is that (unless using active or optical USB, where cable runs of 10m are feasible), the cables can’t be longer than 2-3 metres. That won’t be an issue for most people, but could be a consideration if, for example, your main studio DAW computer is tucked out of the way in a machine room. Installation of the GControl software and associated plug-in is straightforward. It’s a fairly large download, since it contains plug-ins for all the remote-controllable Wes Wes’ control plug-ins no longer slavishly adhere to the hardware layout, and offer modern features such as node-on-curve adjustments, drag‑handles to set the crossover points, and direct selection of parameters that, on the hardware, you must scroll through to select...

gear, along with firmware updates, but you can choose which plug-ins to install, and the firmware updates are performed automatically for any units that are connected when you install the software. That happened for me, and it was a stress‑free experience. Both GControl and the plug-in can run on Mac or Windows DAWs, and you can install VST2, VST3 or AAX formats. There’s still a part of me that misses Wes’ old skeuomorphic plug-ins, whose GUIs very closely resembled the hardware, making operation completely intuitive. But, on balance, the greater functionality and use of screen real estate of the newer GUIs is winning me over. This particular one has lots to commend it: it’s very slick, and puts me in mind of the FabFilter plug-ins, with a visual representation of the frequency response, and your choice of nodes and drag handles, or floating virtual knobs, to manipulate the band levels, crossover points and compression thresholds. Sensibly, parameter settings that on the hardware are accessed by multiple

Wes Audio Pandora $1359 pros • First and foremost, it sounds good! • Touch-sensitive hardware control. • Very slick remote-control plug-in. • More than a compressor!

cons • Uses up three 500-series rack slots.

summary The _Pandora is a superb three-band compressor and saturation processor that has more potential applications than you might expect.

button presses can here be selected directly, and each is given its own virtual LED (plus a scroll button if you really want one!). Each band also has a vertical gain-reduction meter that’s much easier to read than the hardware meter, as well as dedicated bypass and solo buttons. At the top, you’ll find global controls: not just knobs for the crossovers, the wet/dry mix and the output level, but also a master threshold control that allows you to ‘scale’ the settings of the bands’ individual ones without losing their relationship. Also at the top is a full preset management system, so you can store and recall many more settings than on the hardware, and a useful array of starting points (you’ll obviously need, as a minimum, to change the thresholds to suit the material being processed), as well as the usual plug-in functions such as undo/redo and the global bypass. And at the bottom, as well as details of the connected hardware (which you have to select on first initiating the plug-in, but will be remembered next time you load the project), you’ll find that in place of the hardware A/B compare buttons, you have

...but the control set adapts as you resize the GUI, and if you prefer a more compact hardware-like control set, that’s available too.

www.soundonsound.com / February 2026

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WES AUDIO PANDORA

When you first load the GControl software and connect your Wes modules, any necessary firmware updates are performed automatically — and, mercifully, pretty quickly.

a vast number of snapshot slots, organised in 20 banks of three. There’s still more... While some will love all this control and information, others might prefer a more hardware-like GUI or want to save screen space. Fear not, because not only is the GUI fully resizeable, but as you scale it down some of the elements that no longer fit start to disappear. By the time you get down to about 50 percent of the default size, you’re left with just the essentials: the curve, nodes, drag-handles and so forth disappear, leaving you with a very focused control set. By the way, another nice touch is that while the maximum GUI size is 200 percent, the plug-in detects your screen size and prevents you from choosing a setting that would expand the GUI off-screen. I wish more plug-in developers would do that! It all comes together very nicely indeed.

In Use As with most of the Wes Audio gear that I’ve used in recent years, the Pandora sounds great, and I genuinely can’t overstate that: for all the cleverness of the digital control, what the analogue electronics can do to your sound is all important, and these circuits do indeed sound superb. The Pandora is admirably clean and quiet — the only audible noise I noticed was already in the source signal, and lifted up by compression and make‑up gain. I don’t really feel the need to say much about the compression action itself — it’s everything you’d expect from modern VCA compressors, with anything from fast to slow actions, and very easy to dial in. That there are three bands and you’re able to set the (global) wet/dry mix makes it more versatile than most, but you get the idea.

February 2026 / www.soundonsound.com

Most users will opt to use the plug-in, but the GControl software also offers standalone management of setting storage and recall.

Having said all that, you can make the Pandora sound ‘unclean’ if you wish, as the THD options can inject some attitude. The Pandora being a stereo device, I loved experimenting with it on subgroups and the stereo mix bus, but there’s no reason you couldn’t use it on individual sources. We’ve seen this facility many times before on Wes Audio devices but because of the multiband nature of this processor, its ability to solo bands or bypass their processing and, particularly, the global wet/dry mix control (that facilitates far more subtle mastering treatments), it’s incredibly useful. It’s so easy to focus this distortion on a band to have a very specific effect: using a single band to give the high end a harder edge, say, or add thickness in the bottom end to convey a sense of power on smaller speakers. You can do a bit of both at once, of course, and add saturation to the mid band too if you want, with the High/Med settings set individually for each band. Also worth bearing in mind is that you can use this device purely for this multiband saturation effect, without needing to dial in any compression — to do that, just set the threshold to maximum. And finally, in terms of processing options, let’s not forget that as with any multiband compressor you can use the bands’ make‑up gain controls — again, with or without compression — as a broad-brush tone-shaping stereo equaliser. And if you start those EQ moves with the mix control at the halfway point, you can also use that knob as a means of scaling your EQ curve up/down too. Very neat! The band solo facilities make assessing these sorts of effects and, particularly, judging the optimum crossover

filter frequencies for them, really simple. Another thing that makes all your processing decisions particularly easy to make is, to my mind, one of the key benefits of remote plug-in control, but it’s easy to overlook it... Lots of people will naturally focus on the ability to save and recall settings with your session, to build a library of presets, or even automate parameters (which works very well — no problems with audible clicking between settings). But one of the biggest benefits of any plug-in is that you can make all the adjustments you need while sitting in the listening sweet spot. No need to move your head as you reach for the controls!

Verdict The Pandora is an excellent stereo, three-band VCA compressor in its own right. But it doubles up as a stereo EQ and saturation device too, so it’s by no means limited to a ‘special occasion’ use. Indeed, it could be a real studio workhorse. You could never load the plug-in at all, and all of that would still apply, but the plug-in is not mere icing on the cake: it genuinely opens up different ways of working. Down sides? Well, it does occupy three module slots, which adds a little to the expense. But we’re splitting hairs there, and with six gain‑reduction stages on board, I reckon it’s great value for money. I’ve said it before, but when it comes to plug-in remote-controlled analogue hardware, Wes Audio are currently the ones to beat. $ $1359. T Wes Audio +1 614 732 0777 E info@wesaudio.com W https://wesaudio.com

88R LBC PRECISION. POWER. PURE NEVE DNA. 88RS CONSOLE DYNAMICS IN 500-SERIES FORMAT

www.neve.com

ON TE ST

Sampling Workstation

1010music Bento 1010music’s portable groovebox is the company’s most ambitious product yet.

RORY DOW

1010

music’s Bento aims to deliver a complete sample-based groovebox, with everything you need to create a finished track. They’ve taken elements from their Blackbox, some of the granular and wavetable features from their other products, written a completely new operating system and packaged it in a very compelling portable unit with 16 velocity- and pressure-sensitive pads, soft-click buttons for navigation and transport control, eight endless encoders and a multi-touch colour LCD screen. The competition for sample-based grooveboxes is fierce. Will the Bento be able to compete with mature products

February 2026 / www.soundonsound.com

like the Akai MPC range, Elektron’s many offerings, Polyend’s Tracker and Play and others? Let’s find out.

Bento Box The Bento measures 204 x 218 x 51mm and weighs just under a kilogram. The enclosure is made from rugged plastic; it feels solid and well made. The encoders, buttons, pads and screen are all excellent quality. There is even a handy VESA mount on the rear, which is a nice touch. Power is supplied via USB-C or the internal battery, which should provide approximately three hours of life, with a full recharge taking around an hour. There’s no internal speaker, so you’ll need headphones if you’re out and about. There is no built-in microphone either. Storage is on microSD card, and 1010music claim there’s no limit to the

size of the card you can use. The Bento can stream samples from disk, so there are no RAM limitations, but there is a maximum of 576 samples per project, and samples loaded into the granular engine are truncated to 30 seconds. The colour LCD touchscreen looks good, although there is a large border, so it’s actually smaller than it might appear. This means that the labels on the screen, which show the soft-encoder functions, do not align with their corresponding encoder. The screen is multi-touch, allowing you to use familiar smartphone gestures, such as pinching to zoom and two-finger scrolling. Navigation and transport controls are accessed via soft-touch buttons, with dedicated buttons for various editing modes, like the track mixer, clip launcher, song mode, and effects.

at this time. The pads light up in whatever colour is assigned to the current track.

Connectivity The Bento’s slightly recessed rear panel features two USB-C ports, one for MIDI devices and the other for host connections. Either one can be used as a power source. MIDI and audio connections are all 3.5mm mini-jack sockets, but there is a generous number of them: two MIDI inputs and outputs, and three audio inputs and outputs. The Bento is clearly designed to be the centre of a modest desktop setup, where you can connect up to three other devices to use the Bento as a mixer or looper, utilise the extra outputs for hardware effects, and synchronise it all via MIDI. A word of warning, however. If you are using the Bento on stage, make sure you connect your setup in a brightly lit room beforehand. The labels on the rear are almost unreadable. They are etched onto the case in a tiny font, with no colouring; black writing on a black surface. Even in a brightly lit room, I had to squint and angle the unit to find the correct port for any given cable.

Projects

Finally, the 16 velocity‑ and pressure-sensitive pads offer note input. They feel good, maybe not as ‘natural’ as Akai MPC pads, but more than good enough for the job. Pads in this grid-like layout are great for drums, but less suited to melodic input. To help, each project can have a root note and scale setting to which the pads will adhere. I would have also liked a keyboard-style option where the lower eight pads were white keys and the upper pads were black keys, but this isn’t possible

A Bento project can have eight tracks. A track can be either a multisample (for melodic samples), one-shot (for drum kits), loop (for tempo-sync’ed clips), slicer (for chopping), granular (sample-based synthesis), wavetable, or external (for integrating external instruments). You can mix and match track types, except for granular tracks, which are limited to one per project. In the Mixer screen, tracks can be mixed and assigned to one of the three audio outputs. There are also three effects busses (modulation, delay and reverb) plus a master compressor. Each track can have up to eight sequences, and you can launch them from an Ableton-like clip launcher. Scenes can launch a selection of sequences together, and Song mode can chain Scenes together to form a song. Sequences can be any length up to 256 steps, with step lengths ranging from eight bars to 64th notes, which provides ample scope for interesting polymeters and rhythmic experimentation. You can record into a sequence from the pads, or from an external MIDI device, and then edit each note’s pitch, start time, length, velocity and probability afterwards. With the current firmware (version 1.3.4 at the time of this review), it is not possible to record any automation or MIDI data beyond basic note information. Basic copy and paste functions exist for entire sequences, but you cannot copy and paste individual notes or note selections. The sequencer launcher screen is a simple affair, featuring an 8x8 grid that represents eight sequences for each of the eight tracks. You can start and stop sequences individually by tapping on the relevant pad. If there is a particular combination of sequences you like, you can hit the hamburger menu and ‘Snap Scene’ to create a new Scene that will automatically launch all the current clips together. Up to 32 Scenes can be defined this way. Scenes can be edited afterwards. They contain a reference to whichever sequences are selected to

1010music Bento $899 pros • A great new form factor for sample-based fun. • Solid build. • The looping and granular engines are a highlight.

cons • At the time of this review, many areas feel underdeveloped or buggy.

summary 1010music’s new sample-based flagship standalone groovebox represents a bold leap into a world dominated by industry stalwarts like Akai, Elektron and Polyend. All the correct elements are in place: 16 pads, sampling, sequencing, slicing, multisample support, clip launching, internal effects, and wavetable and granular synthesis engines. It may not be as fully developed as the competition, but it’s a promising start.

be played for any given track. Beyond functions to clear, copy and paste scenes, there isn’t much more to it.

Instruments & Sound Design Let’s take a brief tour of the various instrument types that are available for each track, starting with multisample instruments. Multisample instruments, as you might expect, are composed of multiple WAV files, typically used to create melodic instruments. The Bento comes pre-installed with a decent selection of pre-made examples, which is just as well, because creating your own is a laborious task. There aren’t the tools on board to make it feasible. You must meticulously prepare your multisamples on a computer, including root notes and velocity data in the metadata or in the filenames. Then import them into a new multisample instrument. After this, you can edit instrument-wide parameters like pitch, level, filtering, looping, polyphony, envelope and LFO, but these apply to all zones, so there’s no keymap editing baked in. One of the post-launch OS updates introduced a multisample recorder, which can help you create multisample instruments by automatically recording and generating the data for you. With your external MIDI instrument ready, you set the high and low MIDI notes, the number of velocity layers, and the MIDI channel and output to which the instrument is connected, and Bento does the rest. www.soundonsound.com / February 2026

ON TE ST

Black Corporation ISE-NIN Voice Eurorack Module

“If

you’re spending half your life hunting for obscure reasons to decry modern recreations of vintage synths, you’re not spending enough time making music.” Those were the words of Gordon Reid in his SOS review of Black Corporation’s ISE-NIN synth last year, and to that I say: quite right. Gordon was of course writing up Black Corporation’s eight-voice recreation of the Roland Jupiter-8; a polyphonic analogue powerhouse promising nostalgia and exploration in equal measure. Or, perhaps, more of a 70/30 balance. That standalone synth offered a clutch of mod-cons but suffice to say its faithful recreation of the Jupe’s analogue architecture was its raison d’être, and this it did extremely well, as stringently demonstrated by Gordon alongside the real thing. Now comes the ISE-NIN Voice; a single-voice version of the ISE-NIN with every single slider and switch found on its older sibling, presenting a complete Jupiter-style semi-modular

February 2026 / www.soundonsound.com

synth voice in Eurorack form that happily makes up for the deficit in that aforementioned 70/30 balance. Rejoice! I should point out at this stage that the ISE-NIN Voice is a truly excellent recreation of the Jupiter’s architecture and sound. But have I long dreamed of being able to install a Roland Jupiter voice into my Eurorack system? Honestly, no. Am I glad it’s now there? Resoundingly, yes. Indeed, faithfulness to a predecessor for its own sake is — or, at least, should be — just one of many criteria one looks for when it comes to any synth of this type, and if I’m honest for me it often amounts to little more than a footnote unless I’m disappointedly buying it because I genuinely can’t access or afford the real thing. To quote Gordon one last time, “If you were to record the same track four times with contributions from a Jupiter‑8 or Arturia’s Jup‑8 V4 or the appropriate model in a Jupiter‑X or an ISE‑NIN, I doubt that your audience would be able to tell the difference. And even if they could, would it matter?” Again, well said. I should say that before even switching it on, this module is just a lovely thing to behold. Immaculately built and finished, its diminutive sliders are solidly mounted and firm to the touch, its panel is impeccably designed with sections demarcated by trademark Jupiter-esque orange bars on

matte black, and it also has a back panel to protect the PCB, which is always welcome. I won’t waste column space listing functions, which have both already been outlined by Gordon in his ISE-NIN review and well documented as components of the actual Jupiter-8, but it’s worth taking a minute to remind ourselves of the fundamentals. The ISE-NIN Voice constitutes two oscillators with selectable waveforms (triangle, ramp, pulse and square on VCO 1, and the same but with sine and noise in place of triangle and square on VCO 2). The low-pass filter is selectable between 12 and 24 dB/octave slopes and can be modulated by either of the onboard envelopes or by the single onboard dedicated LFO (VCO 2 can dive to sub-audio frequencies if desired). There’s also a high-pass filter. The LFO is pleasantly manipulable, with selectable waveforms (including random) and a Delay control for dialling in some latent movement across a held note. Finally, there’s an input for pitch and gate and a single output, beneath knobs for tuning, glide and volume. In terms of a standalone synth voice, it’s excellent. Its sliders are a pleasure to use, it’s nicely laid out and I honestly can’t really fault its sound. Rock‑solid oscillators, a beautifully responsive, smooth pair of filters, the list goes on. With no jacks for patching to speak of, in this respect it’s not so much

a Eurorack module as it is a standalone synth that happens to be Eurorack mountable. As far as complete analogue voices go, you won’t easily find one as powerful as this in so small a form factor. But as you might have seen coming, with the ISE-NIN Breakout in the picture, the ISE-NIN Voice doesn’t just offer patchability, it offers maximum patchability. Which is to say, voltage control over literally every single control on the panel and outputs for, well, pretty much anything that outputs voltage or audio. Crikey. It’s a no-compromise approach, albeit one that will test your wallet as a $299 investment on top of the $999 price of the main module. But it’s one I happen to like; almost a transverse workflow between two sides of a brain: fairly traditional and linear on the one side and then almost completely open-ended on the other. It’s also highly gratifying blending these two poles. I was not only able to straightforwardly patch an external modulation source into the filter cutoff, I could also compound its signal with that of the internal LFO, and multing the external CV to also modulate the cross-modulation input made for even more dynamic movement. I was also able to do some internal patching to reroute certain aspects of the architecture if I wanted to, for instance sending either envelope any which way, truly stretching the bounds of what an apparent Jupiter clone can really do. Aside from the usual suspects the expander also offers jacks for more complex things, like inputs for the individual stages of both envelopes, outputs for two pitches of noise, and even a headphone output. My favourite fact about Black Corporation is that the director JJ Abrams is a big fan of the company; so much so that he decided to name one of the planets in Star Wars: The Rise Of Skywalker after their Kijimi synth. Apparently the name ISE-NIN could mean many things in Japanese, or not a lot (it’s “just from us being weirdos” according to BC’s Bob Rogue), but to my mind it is synonymous with utterly premium quality, both in build and in sound. Amid an increasingly digital Eurorack industry, The ISE-NIN Voice is a welcome dose of iconic analogue goodness, whose price might make you think twice (particularly alongside the likes of the Rides In The Storm SED-CSM), but whose sound and functionality will serve your system for years to come. William Stokes $

ISE-NIN Voice $999, Breakout $299.

www.black-corporation.com

Shakmat Bishop’s Miscellany MkII Eurorack Module

he Bishop’s Miscellany was Shakmat’s first entry into the Eurorack scene back in 2014. It was a dual-channel stepped CV recorder and looper that had plenty of things to keep a Bishop busy. The MkII builds on the concepts of quick note capture and turns it from an instant riff maker into a sequence-generating, CV-manipulating, pattern-chaining song machine. The original Bishop’s Misc was very instant and playful; you could patch it up, lean on the Record toggle switch to record some notes, loop it and mess with it. The MkII feels a lot more complicated. Your focus is inevitably drawn to the screen and endless encoder; there are twice as many buttons, the knobs are generically labelled, meaning that they are not going to be dedicated to one function, and the module seems busy before you’ve done anything. Once you understand that it’s going to take a bit of learning and a bit of screen/encoder patience, the liveliness that made the original so much fun begins to return, along with a whole load more miscellany.

The basic idea is that the BM MkII can record CV and gate signals into two channels and output them as stepped, scaled notes in up to 64 steps. It has two manipulation engines. One side is generative and regenerative,

www.soundonsound.com / February 2026

FE ATURE

Part 8: Replicating Human Hearing

Concluding our series on advanced stereo mic techniques, we explore binaural recording, disc-shaped baffles and dummy heads, as well as a clever process called ‘Stereo Shuffling’. HUGH ROBJOHNS

he preceding parts of this series took us on a journey from the fundamental concepts of stereo sound recording and reproduction to the theory and practical aspects of a wide range of stereo microphone techniques. We’ve explored Blumlein’s coincident arrays, including Mid-Sides, that focus on capturing and reproducing only amplitude differences between channels; and the wide-spaced omni-mic arrays that capture mainly time-of-arrival differences between channels. We went on to consider a number of popular near-spaced arrays that were designed to combine both amplitude and timing differences between channels — the best of both worlds — before also contemplating some of the more complex stereo mic arrays that combine both spaced omnis and near-spaced arrays, including a brief look at the famous Decca Tree. By this point, then, you’d be forgiven for thinking that there couldn’t possibly be any more stereo mic arrays to examine! But there are — and if you’ve been working through the audio example files associated with this series, you’ll

February 2026 / www.soundonsound.com

already be aware of one example: the Schneider Disc. This, with variations upon it, is the last stereo mic array that I want to discuss in this series, as it represents a final class of stereo microphone techniques that aim to replicate the human hearing physiology — to a greater or lesser extent.

Dummy Head If the idea of stereo recording is to replicate the human experience of listening to performers on a stage, arguably the most obvious approach is to imitate the human listening apparatus. Basically, that means using a head-shaped construction with microphones in place of the ears. This concept, popularly known as the ‘dummy head’, has been explored in depth for decades, and there are several commercial examples, including Neumann’s original KU80 and current KU100, and Sennheiser’s now discontinues MZK2002, amongst various others. These dummy head systems produce a stereo signal in a format generally referred to as ‘binaural’, and intended exclusively for headphone listening. However, the success of these dummy

head recordings, in terms of imaging stability and source location realism, depends very heavily on how closely the listener’s anatomy matches that of the dummy head, particularly in terms of head size and ear spacing (early dummy heads don’t attempt to replicate the folds and curves of the pinnae which, as we know, play an important role in vertical and front/back source location). The physiological parameters of the listener’s head/ears determine what is now known as the ‘Head Related Transfer Function’ or HRTF, which is unique to each individual listener, and it is these parameters that define how we individually perceive sound locations within a binaural signal. As it happens, I own a Sennheiser MZK2002 dummy head. The recordings I’ve made with it certainly sound spacious and are interesting, but I don’t find them consistent, stable or precise. That’s probably in large part because the dummy head is physically smaller than my own. If I attach the same Sennheiser microphones to my own ears for recording, then playing back the audio files gives me remarkably precise and stable imaging information. But it’s important to note

two things... First, if you move your head at all, you change the perspective of the mics. And second, this approach benefits me alone — others who’ve listened to recordings I’ve captured that way usually find the imaging to be imprecise and unstable again. This highlights the critical importance of personal HRTFs for accurate spatial imaging in binaural listening. Today, of course, there’s a huge academic and commercial interest in binaural stereo, not least because of the vast numbers of people who now listen over headphones habitually. And while personalised binaural recordings might not transfer well from one listener to another, binaural reproduction is an ideal format for low-cost immersive-audio playback. Indeed, with the benefit of sophisticated digital signal processing, and the ability to create and implement personal HRTFs for playback over headphones, binaural stereo is making significant headway in popularity, and many manufacturers are now incorporating this technology in smartphones, tablets, laptops and more. One down side of binaural stereo and dummy-head recordings, though, is that the format is specifically intended for use with headphones, and the stereo imaging is usually very poor when auditioned over loudspeakers. That wouldn’t be a problem if the entire audience were known to use headphones, but in most cases a significant proportion of the audience for commercial music is likely to be listening over loudspeakers. So is there an alternative technique that could offer better compatibility between headphone and loudspeaker listening?

pressure (omnidirectional) mic capsules separated by an acoustic baffle. He described the setup in his famous patent, along with sophisticated signal processing to make it work better for loudspeaker listening, which is something I’ll come back to shortly. It wasn’t a line of enquiry Blumlein pursued at the time, though, as he chose instead to focus his work on coincident velocity (directional) microphones, which, to his mind, gave better results with loudspeaker auditioning. But good ideas have a habit of resurfacing sooner or later, and it was the work of Jürg Jecklin in the early 1980s, at the University of Music and the Performing Arts in Vienna, that resurrected the interest and popularity of Blumlein’s idea. He called the technique the Optimal Stereo Signal or OSS — but it became far better known for its physical appearance, and is usually referred to simply as the Jecklin Disc.

Jecklin Disc Jecklin’s earliest design employed a flat, 30cm diameter disc, covered in a thin layer of foam (or fleece) to reduce the baffle’s acoustic reflectivity. A pair of omnidirectional microphones were then mounted on either side of the disc, their capsules being aligned with the centre of the disc, spaced 16.5cm apart (so roughly

8cm above the disc’s surface) and pointed outwards by 20 degrees. In essence, the disc baffle broadly replicates the primary acoustic effects of the human head, with the baffle diameter and mic spacing combining to introduce similar frequency response, time and amplitude differences between the two microphone signals as are experienced at the ears of a listener. Those differences increase as a sound source moves further off axis from the front of the disc, much like it does with human hearing. Most commercially made OSS discs follow Jecklin’s original specifications for the baffle size and mic spacing, but in the mid-1980s Jecklin himself revisited the design, and concluded that it performed better with a larger baffle diameter and greater mic spacing. Consequently, he issued revised specifications, with a disc diameter of 35cm, and mic spacing increased to 36cm (so roughly 17cm above the disc surface). Commercial Jecklin Discs are not inexpensive, but anyone handy at basic DIY can construct their own experimental Jecklin Disc relatively easily, using something like a piece of plywood or Perspex cut into a disc. While the size/ diameter is fairly important, you don’t even need to be able to cut a perfect circle (the human head is broadly ellipsoid

Optimal Stereo Signal (OSS) Take away the physical differences between individuals’ heads and ears, and the main significance of the replica head in dummy head recordings is that it acts as an acoustic baffle, reducing the strength of high frequencies from a source on one side reaching the ear on the opposite side, as well as introducing a time delay to the further ear. So why not just reduce the ‘dummy head’ to its simplest form: a flat baffle, with microphones placed on either side? Needless to say, it’s been done... and as you might already have guessed, the first person to do it was Alan Blumlein, back in the early 1930s. One of his earliest experiments involved a pair of

A range of mics intended to replicate the human listening experience. Clockwise from top right: the Neumann KU100 dummy head; Bruel & Kjaer’s HATS system; the 3DIO FS Pro II; and some ear-worn mics. The last of these could potentially be the most convincing — but only for the person who wore the mics!

www.soundonsound.com / February 2026

TECHNIQUE

Sometimes, nothing beats the unpredictable swoosh and whoosh of the classic tape flanging effect. And you can recreate it in your DAW... CRAIG ANDERTON

ou may recognise ‘flanging’ from plug-ins or guitar pedals, but it was originally created with either two turntables or two tape recorders, each playing the same audio, and their signals then being mixed together. Importantly, one playback device had variable playback speed, and changing that speed caused phase additions and cancellations — it’s these that created the famous ‘jet aeroplane’ sound of the psychedelic ’60s. Over the years, flangers became modulation effects in their own right, but they lost the vintage flanger vibe along the way. Specifically, classic flanging’s climax occurred at the ‘through-zero’ point — that’s where the variable-speed unit was polarity‑inverted, so that when

it was sped up or slowed down to re‑sync up with the other device, the combined sound of the two devices briefly cancelled out. Digital delays can’t go into the future, of course, so unless designers specifically set out to emulate the through-zero effect, their flanger’s sweep almost makes it to the climactic top of the flanging cycle — but always falls short. Also worth noting is that tape flanging didn’t use LFOs: the physical motors had inertia, and that made the flanger sweep somewhat unpredictable. This in turn created excitement: you never knew when the sweep would hit the through-zero point. Today, there are a couple of DAW plug-ins that can emulate the through-zero effect — Apogee’s Clearmountain’s Phases, for instance (reviewed in SOS March 2022: https:// sosm.ag/apogee-clearmountain-phases) — and model some of the other features of vintage flanging, though most plug‑ins don’t do it all. Thankfully, if you have a DAW with built-in parameter modulation, you can replicate the sound and excitement for yourself.

True Confessions: I Burned Out The Hendrix Tape Machine! The sound of true tape flanging was burned into my brain from sessions in 1968 at the Record Plant, which followed the sessions for Hendrix’s Electric Ladyland. The flanging setup was complex, so engineer Tony Bongiovi (who worked night sessions for Hendrix and engineered our album) left it set up for us to use. Yes, of course we used it — and to Record Plant co-founder Gary Kellgren, wherever you are now, I’m still sorry that I burned out the tape recorder’s capstan motor by being too aggressive with the variable speed control!

February 2026 / www.soundonsound.com

The classic ’60s flanging effect employed two tape machines, one with varying speed to pull it ahead and push it behind the other during playback.

Below, I’ll do just that, using Cubase 14’s Modulators and Multi Track Presets, and this time as well as providing an audio example on the SOS site (https://sosm.ag/vintage-flanging) I’ve made available a Cubase Multi Track Preset, so those who use the same DAW as me can dig in a bit deeper.

Basic Track Setup To create the flanger effect, the Multi Track Preset requires two Cubase FX tracks to emulate different playback devices, one of which is variable in speed.

In the first screenshot, you can see that the audio track on the left, with its channel fader pulled right down, feeds the two FX tracks with separate pre-fader sends (note that these must be added manually if you download the Vintage Flanger.trackpreset file). The Pre section’s Phase button sets the signal polarity. You can see that the second FX track has this inverted (orange button, top-right), and that will deliver the ‘hollow-sounding’ negative flanging effect. For positive flanging (a more ‘zinging’ sound), switch that back. For classic, through-zero flanging, we require inverted polarity. The FX track panning is offset somewhat, simply because analogue machines never had perfect cancellation at the through-zero point. This slight panning shift is a simple way to prevent 100 percent cancellation.

The Flangers The main flanger track (FX Track 01A) includes Cubase’s stock Flanger effect, but this is controlled by a Modulator that varies the delay time. The Flanger in Track 01B has identical settings, but this one is not under Modulator control — it has a fixed delay time of 2ms. Note that both flangers are in Manual

My vintage flanger setup in the Cubase MixConsole. If you use Cubase, you can download this setup from the online version of this article as a Multi Track Preset.

“Unless designers emulate the through-zero effect, their flanger’s sweep almost makes it to the climactic top of the flanging cycle — but always falls short.”

mode — so if you’re trying to recreate the settings from the screens manually, the Sync, LFO Rate, Lo Range and Hi Range controls are all irrelevant. Both Mix controls need to be 100 percent (all effect, no dry sound). Increasing the Spatial parameter creates a different type of sound, but this dilutes the flanging effect somewhat. The Modulator is the key to vintage flanging. Here, it sets an initial delay time of 1.5ms, which is ahead of the Flanger with the fixed delay (2ms). The random waveform, and long sweep time, are characteristic of vintage flanging. Setting Shape to 100 produces a smooth, continuous flanging effect. As the Modulator varies the Flanger’s delay time, though, it occasionally overtakes the 2ms fixed delay, then crosses over it again on its way down to a longer time, producing the through-zero flanging effect. Like real flanging, you’re never quite sure when that through-zero is going to happen, which just adds to the sense of drama — which you can hear for yourself in the audio example. Finally, note that there is an inherent 2ms delay in this effect. In most cases, you won’t notice this. But if you find that it’s an issue, simply move the track being processed 2ms earlier on the timeline. And now you have the sound of ‘Itchycoo Park’, ‘Bold as Love’, ‘Life In The Fast Lane’, ‘Head Over Heels’, and so many more! Grab your bell‑bottom pants, Nehru jacket and love beads, and head back in time to when flanging was a mechanical — not an electronic or digital — effect.

The Flanger plug-in settings for the variable-speed flanger inserted in FX Track 01A.

www.soundonsound.com / February 2026

TECHNIQUE

THE SOS TEAM

t’s great that there’s so much to explore in SOS For Artists — but when you first sign up, this can also make it easy to overlook some things. So we’d like to shine the spotlight on some hidden gems, and this month our focus will be on the six Audified plug-ins included in your subscription: U73b Compressor, U78 Saturator, 1A Equalizer, RZ062a Equalizer, RZ062b Equalizer and MixChecker Ultra. Together, they form a useful toolkit that can add something to your DAW’s built-in tools, and you can call on them at pretty much any stage of a mix. In this article, we’ll provide a quick overview of each plug-in, along with some notes on how you might go about getting the most out of them — and we’ve also created a few audio examples (https://sosm-ag/sosfa-workshop-0226), which we’ve based on samples you can find in the SOSFA library. Before we get started, though, note that lots of these plug-ins feature level calibration controls, which can help you ensure that you’re hitting the sweet spot in the modelled processing. There’s no ‘right’ setting — it’s similar to analogue gain staging, but when you’re searching for character through modelled saturation, you should feel free to experiment for the best sound.

U73b: Vintage Compression The U73b Compressor (now on version 3) is a great-sounding emulation of a vintage variable-mu compressor-limiter made by Telefunken. Variable‑mu types are famous for sounding classy — smooth and musical — but the Telefunken device has a reputation for being a bit more colourful, with a slightly forward character in the midrange, and this plug-in does a good job of replicating that. So it’s a useful alternative flavour to the options you’ll typically already have in your DAW. This plug-in offers both compressor and limiter modes, selectable release times, side-chain high-pass filter options (the original had a 100Hz filter built in, but you can disable that) and the ability to link or unlink the left and right channels when used on stereo material. Additional benefits include being able to inversely link the input and output controls, plus a virtual moving-coil meter, with the option to view the amount of gain reduction, or the input or output levels. As with the U78 (more of which below) there’s an oversampling option, which can

February 2026 / www.soundonsound.com

Your SOSFA subscription gives you access to some super plug-ins by Audified. Let’s see what they can do...

Audified’s U73b variable-mu compressor plug-in.

give you higher‑quality results, especially when adding lots of character, but as this comes at the cost of using more of your computer’s CPU power you could, if needed, disable this feature until your final render. For our audio examples, we used U73b in both its limiter

U78: vintage-flavoured saturation.

and compressor modes, on a drum loop and a bass line. For the drums, we switched to the limiter mode and aimed for a combination of loudness and vintage character, just to get the beat driving the track a little more. The release is quite fast, which is typically what you want for drums (so that the compressor recovers before the next hit), and the amount of compression fairly high, with a slight boost given to the output level to compensate for that. Here, we used a loop, but you

and then restored some of the brightness using the Tone control. For the bass track, some of the lower frequencies were again rolled off to clean up the sound, and then we used the Tone control to compensate by darkening the overall tonality.

Pultec-ish

The 1A equaliser is based on a Pultec design, but adds some useful touches.

could equally well apply these sorts of settings to the drum bus when processing live drums. For the bass, we switched to the compressor mode and set the attack to 2.5, again keeping the compressor settings quite high. In this instance we activated the high-pass filter to prevent the really energetic low-end frequencies from triggering overcompression — this really helped to retain a sense of clarity in the bass line; it’s pretty easy to switch it in/out and judge this setting by ear. Finally, note that you can try this compressor wherever you’re looking to add a bit of vintage character, including dropping the plug-in onto your mix bus. The mid-forward character will likely work wonderfully for some material and less well for others, but it’s well worth a try; we’d suggest aiming for only a very subtle amount of gain reduction, with a fairly slow release.

U78: Saturation The U78 draws on the sonic character of U73b to give you a dedicated saturator, with tools to help you focus the resulting energy where it’s needed. There are high- and low-pass filters, a Tone control that operates rather like a tilt EQ, and a wet/dry mix

control that means you can be as heavy‑handed or subtle as you like with the resulting colour. An Auto Output facility ensures the overall level remains broadly consistent. To demonstrate the sort of thing you can do with this plug-in, we applied it to synth and fretless bass parts, but you can try it on any source that you think will benefit from greater harmonic richness, warmth, grit or presence, be that individual tracks, groups or even, with a suitably subtle mix setting, the main mix bus. On the first demo, we’ve used U78 to apply some valve brightness to a synth, to give the part just a little more bite and edge. To do this, we rolled off some of the lower frequencies, cut some high‑frequency harshness,

As you can tell immediately from the GUI, the 1A equaliser emulates the classic Pultec EQP-1A, a valve-amplified passive EQ famous for the unusual curves created when boosting and cutting around the same frequency. That’s possible here, and it can be especially useful for shaping the low end — often effective when you need to rework kick drums and

“1A’s M-S option means you have more scope than on many emulations for sculpting stereo signals, such as synths...” bass instruments. Two other standout features are the Calibration knob — again, this allows you to dial in analogue-ish saturation to taste — and a Mid-Sides mode. Pultec-style EQs can be useful on any number of instrument tracks, and vocals, but 1A’s M-S option means you have more scope than on many emulations for sculpting stereo signals, such as synths, drum and mix busses or even in mastering. An output level control lets you compensate for any overall level change resulting from your EQ moves. For our demos, we applied 1A to both guitar and choral vocal tracks. For the

The RZ062 suite, comprising ‘a’ and ‘b’ versions of a plug-in that emulates a valve EQ developed for film production in the 1950s.

www.soundonsound.com / February 2026

TECHNIQUE

EXPLORING THE AUDIFIED PLUG-INS

Mix Checker is a really handy tool for checking how your mix will translate on other playback systems. There are lots of options, and there’s some handy metering thrown in for good measure.

guitar, the lower frequencies were pulled down, while the higher frequencies were boosted to give a sense of air, resulting in a much cleaner, clearer sound. On the vocals, we set 1A to its Mid-Sides mode, to ensure depth and solidity at the centre and on mono playback, while emphasising the width in higher frequencies for a more dramatic stereo presentation.

RZ062 Suite The RZ062 suite, comprising RZ062a and RZ062b, offers you emulations of two different vintage EQs, this time rare Siemens valve EQ modules that were originally used in the Klangfilm system in European film studios in the 1950s. Although these were designed and built to the highest standards of the day, they have since become known for imparting an engaging warmth and density and have naturally found favour in music production circles. The ‘a’ version features a tilt-style slope with a centre frequency of 650Hz, and a central knob moving the slope up or down, while also imparting an enticing valve character — you might call it ‘warmth’! This makes it an ideal tool for gently enhancing vocals, strings, pianos or the overall mix. There are separate control sets for the left and right channels and, again, there’s a Mid-Sides option, which makes it particularly versatile. We used this version of the plug-in on string and piano samples downloaded from the SOSFA library. For the string sample, we opted for the Mid-Sides setting, and unlinked the two channels before giving the lower and upper frequencies slight boosts. The input levels were raised to give a little more saturation, and the output levels dropped to compensate — and note that you have the option of driving the two channels into different amounts of saturation, by setting their input/output levels differently. For the piano, our aim was to add some warmth and presence to the part. This was once again achieved by switching to Mid-Sides mode and

February 2026 / www.soundonsound.com

de-linking the two channels, before boosting the upper and lower frequencies on each channel to taste, until we felt we had a nice tonal balance and stereo image. RZ062b has a similar heritage, but in place of the simple tilt point the middle EQ control offers four selectable frequencies for adding presence, with six boost settings for each. These controls make it a great option for use on guitars, snares and other instruments with a lot of information in the mids, as well as any tracks that you feel might benefit from a boost in presence or attitude. For our audio examples, we applied the plug-in to a drum loop and guitar. In the first, the plug-in was left on linked mode and the snare enhanced by boosting the high frequencies and adding extra presence in the 2kHz area. Again, the input level was boosted for a little extra colour, and this was counterbalanced by pulling down the output level. For the guitar, we increased the presence and unlinked the left and right channels so we could boost different frequencies on each side (2kHz on one

side, 4kHz on the other), to widen the stereo image, making the resulting sound more dramatic.

MixChecker Ultra Finally, we come to MixChecker Ultra. This one isn’t intended as a processor, but it can be used as one, as well as being a handy mix referencing tool. The idea is that it mimics how your track will sound on a range of real-world playback devices in different listening environments. With over 90 device simulations (phones, laptops, car systems, small bluetooth speakers, club PA, etc), plus impulse-response support, you can test how your mix will translate outside your studio. The built-in meters help you to compare loudness, phase, dynamics and more. You can insert MixChecker Ultra either as the final plug-in on your master bus (in which case remember to turn it off before you bounce your mix!) or, if your DAW allows it, a dedicated monitoring or control-room output. Then it’s easy to toggle through different device simulations to check for critical mix issues. It’s perhaps not a complete replacement for checking your final mix in a different room or in your car, but the metering is useful, and it can be a very effective way to discover if the bass end or dialogue come across OK on smaller speakers. Of course, you could choose to print the processed sound as an effect — for example, for an authentic voice-on-phone sound in video post production or as a spot effect in a track. In our audio demo, you can hear the track dry, followed by the desktop monitors, 22-inch TV, White Cone Vintage 7-inch (based on Yamaha NS-10s) and Classic Pods emulations.

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Studio One

TECHNIQUE

ROBIN VINCENT

Spice up your MIDI data with Studio One’s Note FX.

here comes a time in every musician’s life when they get sick and tired of what they’re playing. The creative juices are simply not flowing, and you’re stuck in a loop of the same worn-out chords and hackneyed melodies. Or, maybe, you’re not that accomplished and need just a little something to elevate your one-finger meanderings to another level. Perhaps what you need is a little pick-me-up: a sweet line of possibilities that will make your head spin and return your sense of self to its rightful place at the top of the heap. What you need is Note FX. Note FX have always been there, and in many ways, they’re Studio One’s secret weapons against banality. In this workshop, we’re going to remind ourselves of their awesomeness and start making the most of their largely forgotten set of very particular skills.

Take Note Note FX are real-time effects processors that act on MIDI notes, so not in an audio effect way of messing with signals, but in a way that reinterprets incoming note data from your MIDI controller keyboard or instrument track. All the action happens before the notes reach your virtual or external MIDI instrument, which makes them recordable, manipulable and reworkable. Studio One 7.2 currently has four Note FX: Arpeggiator, Chorder, Repeater and Input Filter. Each one brings something interesting to the table, and they can also have a lot of fun working together. You’ll find them towards the top of the Instrument Browser, where you can drag them directly onto a track. Alternatively, you’ll find they have their own section in the Inspector where we can load them and have some control over how they work. Let’s dig in.

Arpeggiator Everyone loves a good arpeggiator, and this is a great one that starts off very familiar and then gets increasingly funky as you get into it. The idea is that it takes any number of held notes and turns them into a run or pattern of single notes. So you could hold a chord and have the arpeggiator play the notes in series, either running up, running down, up and then down, down and then up, in the order your played them, or completely randomly. When you load it up, the default preset is precisely as you’d expect. It’s set to 16th

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The Arpeggiator and Chorder Note FX are powerful on their own, but can also be put to work together to create arpeggiations from the automatically generated chords.

notes over one octave, running up and down. The basic controls are on the left. You can push it up to four octaves and change the rate with the large knob, to anything from a 64th triplet to a bar and a half per note. You can introduce a bit of swing to give the arpeggiations some lilt, and change the gate length from short and sharp to sustained and overlapping. Along the top, you have the choice of note flow modes previously mentioned. The last one is a chord mode, which plays the notes you’re holding as a chord in the rhythm of the arpeggiator. If you’ve set a larger octave range, it will play the same chord in different octaves. The Hold button is very useful. It lets you take your hands off the keyboard by holding the notes for you so the arpeggiations keep playing. The Velocity knob enables you to set a velocity for the notes.

It’s set to Input by default so the notes are arpeggiated at the velocity you played them, but if you switch it to Fix you can have them all played the same. Those are the basics, and it’s pretty fun just like that. But there’s another side to the arpeggiator called Pattern. Click the button at the top right and see what happens. The middle display lights up with bars of orange. If you run your mouse over them, you’ll notice that you can resize the height and the width of the bars. The bars set the velocity and gate length of each note along a set pattern duration, turning the arpeggiator into something approaching a pattern sequencer. I find this particularly effective in Play Order mode, where I put it in Hold and grab two handfuls of my keyboard in a chord of around eight notes. I tend to move the Pattern Length up to at least 16; you can switch to another page and up to 32 notes if you wish, but I prefer to focus on a single page. And then just experiment with the velocity and gate on each note. You can even turn notes off completely by clicking their number. If you have a look in the preset drop-down menu, you’ll find some interesting examples of pattern sequences. You might also notice that you can pull the gate across other notes to create overlaps of up to 200 percent if you want, which gives it an interestingly polyphonic feel. Don’t forget to save your own presets of patterns that you make. A quick tip: many of the basic controls are available as simple menus in the Inspector, so you don’t even

The Repeater can work like a simple ratchet effect... or you can use the Scale controls to generate mutating repeats.

have to open the GUI. Another tip is that a quick way to bring up the GUI is to click the tiny little icon, like two piano-roll notes, that’s appeared next to the track name, before the Instrument Editor button.

Chorder The basic idea behind the Chorder is that you ask it to memorise one chord, and then you can play that same chord with one finger, transposed to whatever key you press. If you load up the Chorder, that’s exactly what the default preset does. If you’d like to change the chord, click on Learn Mode, then click on Clear All so that it has no chords in its memory. Play your chord, and you’ll see it appear in the top half of the keyboard. You can add more notes, either with key presses or with your mouse. When you’re happy, turn off Learn Mode, and you can play your chord on every note. However, playing the same chord transposed on different notes only has so many uses, and the Chorder would like to be more useful than that. If you turn off the Auto Fill button, you’ll find that you can only play your chord on the original note it was set to. What this means is that if you go back into Learn Mode, you can actually set a different chord on every single note. Chorder becomes a super-powerful place for chord progressions. If you dig around in the presets, you’ll find all sorts of chord progressions for you to play with, and you can set your own up and save them as presets for later use. Other things of note are that you can set a key range in which the chords are active. With the Filter Outside button off you can play single regular notes outside the range; with it on, nothing plays outside the range. The great thing about that is that you can play one-finger chord progressions with one hand and then a bass or lead part outside the range with the other. The Arpeggiator and the Chorder work brilliantly together. You can build your chords in the Chorder and then turn them into arpeggiations or sequences in the Arpeggiator. You can load as many instances of any of the Note FX plug‑ins as you like onto the same track, although I’m not sure how helpful that would be. However, if you put the Arpeggiator under the Chorder you have quite a fabulous combination going on.

but one that feels somehow unnatural because the repeats don’t fade away. Well, we can fix that, and in fact, Repeater is much more interesting than an echo effect. You have a large knob to set the resolution or timing of the repeats, and next to that you’ll find a Sync button. With Sync enabled, the knob sets time divisions based on the tempo of your project. With sync disabled, you can set any timing you want based on frequency. The Steps knob can increase the number of repeats up to 32. Then we get to the interesting stuff. There are Scale knobs for gate, velocity and pitch. If you turn these knobs, you’ll see the display scaling the steps in interesting ways. The gate’s Scale knob gets that more natural fade going on. Pitch is slightly different because it throws in pitch changes to each step,

Input Filter This one isn’t quite as impressive as the others, but it’s a useful utility. It lets you set a keyboard and velocity range for any incoming MIDI. Yes, that’s it.

which quickly turns it into anything other than a simple echo. The pitch is set to semitones, and if you click the Pitch button at the top right next to Individual you can set the pitch change of every step by itself. The Repeater becomes a fascinating sequence of events that’s triggered by a single note press. It could be melody or bass lines that transpose with the notes you play, or it could be something very experimental. Check out the presets for some great examples.

Recording Input Or Output? One question remains when making music with the Note FX. Should you record the notes you are playing, or the ones generated by the effect? It depends largely on whether you’ve finished fiddling with the Note FX parameters, or if you’d prefer to start editing the result. By default, when you go into Record and start playing, Studio One will record the notes from your keyboard. These are then routed to the Note FX for processing, and you hear the effect. When you play back the track the same thing happens, and you can change the Note FX settings to your heart’s content. However, there’s a tiny little icon in the Inspector alongside the loaded Note FX that sets the input mode. If you enable it, the Note FX are rerouted before the track rather than after it. Now, when you start recording the track, it captures the output of the Note FX. The Note FX are a lot of fun and can be used in all sorts of surprisingly creative ways beyond what you’d normally expect from arps, auto‑chorders or ratchets.

Repeater The Repeater, also known as a ratchet, simply repeats the note played for a set number of times and at a set timing and resolution. The default is eight times in 16ths. It can be quite a weird effect, a bit like a delay or echo,

By default, Studio One will record your MIDI notes as played, so that when you play back a recording, the Note FX will process your notes again in real time. However, you can record the output from the Note FX by changing the input mode.

www.soundonsound.com / February 2026

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Logic Logic

TECHNIQUE

S I M O N PAT E R S O N

Logic’s MIDI plug‑ins are your key to instant inspiration!

hen Logic Pro 11 was released back in May 2024, it introduced a wealth of new features, many of which have already been covered in these pages. However, some of the more subtle, but no less powerful additions may have flown under the radar. Among these are significant improvements to MIDI routing, which could open up creative possibilities for producers. In this workshop, we’ll explore some examples of routing and how to extract editable MIDI data from plug-ins like Arpeggiator. We’ll discuss why this is useful, and how these new tools can complement existing functionality to add depth and colour to your production palette.

MIDI En Place Let’s start by creating a synth sound to be triggered by Arpeggiator. For a warm, analogue-style pad, I chose Retro Synth in Analog mode and loaded the Dark Swell Pad preset. It’s rich and lush, but to make it more suitable for arpeggiation, I made a few tweaks. First, I switched the filter to a low‑pass 12dB/octave ‘Creamy’ style for added brightness without sacrificing low-end warmth. Next, I increased the filter envelope attack speed whilst simultaneously emphasising it by raising the Env parameter in the filter section by a good notch. After that, I shortened the attack in the amp envelope to allow 16th-note arpeggiations to pop through more clearly. Then, whilst playing about, I started tweaking the Glide parameter, and as I did so, enjoyed the effect of the extra movement laced with a little unpredictability. This inspired me to automate this portamento-style parameter using the Modulator plug-in.

Routing For Change Modulator is a gamechanger when it comes to expanding the functionality of any instrument and/or subsequent processor, as Paul White explained in his December 2019 workshop. In this case, I wanted to use it very simply to route a low‑frequency oscillator to subtly modulate the amount of glide in Retro Synth. First, I inserted an instance of Modulator in the first MIDI Effect slot, just above the Retro Synth plug-in on the Software Channel Strip in the Inspector. The left side of Modulator is dedicated to the LFO and, towards the bottom of the GUI, you can

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Here, Logic’s Arpeggiator plug‑in is triggering notes in Retro Synth, while the Modulator MIDI effect is modulating the amount of Glide.

click-hold on the ‘To’ pop-up menu to patch the LFO towards its destination. I selected Learn Plug-in Parameter, which allows you to simply click on any automatable parameter of any plug-in inserted below on the same channel strip. With both Modulator and Retro Synth open, I clicked on the Glide parameter. They were now connected but, because the LFO was oscillating far too wildly, I had to dramatically tame it to ensure that the Glide parameter was modulating within the range and speed I had in mind. To do this I first changed the shape of the LFO to a sine wave, then slowed it right down to 0.18Hz. I had to click on the Sync button (the quaver) to unsync it from Logic’s tempo. Finally, I tamed the range by tweaking the Output Level and Offset to taste.

Get Your Steps In! Next, I recorded the chords I wanted to arpeggiate. As much as I treasure a loose feel, this situation can benefit from quantising and making the note lengths uniform. Whilst Edit / Trim / Note End to Following Notes (previously Force Legato) works well on

monophonic lines, in this case I found the Function / Convert Sustain Pedal to Note Length worked a treat to prepare my chords for arpeggiation. I then turned to the Arpeggiator plug-in. Again, this plug-in has been covered in depth (see SOS December 2013), but for this example I set up a 16th-note arpeggiator with an ‘outside-in’ note order, a slightly expanded octave range and, in the grid, altered the velocity and gate time of each semiquaver to create some interesting dynamic movement. I also enjoyed the sound of Arpeggiator periodically spelling out the entire chord, an effect achieved by clicking on some of the Chord on/off buttons in the pattern.

Re-render Me Well Just because Logic now gives us the ability to record all the MIDI data generated by MIDI plug-ins, does it mean we should? I, for one, certainly think so, and will argue that there are multiple reasons to turn the otherwise ephemeral musical output of such MIDI Effects as Arpeggiator, Randomizer, Chord Memorizer and Note Repeater into concrete, tangible MIDI regions. Having the ability to process actual real MIDI data will open up options for arranging your productions in a much more sophisticated manner that can be achieved by just letting ideas loop ad infinitum. In fact, there’s almost infinite potential for tweaking, editing, rearranging and transforming. For example, just in the Piano Roll editor you can manually alter or randomise pitches and note lengths, humanise notes, subject the MIDI data to invert or retrograde using the MIDI Transform functions, or even use Scale Quantise to spice up your sequence with a different mode or scale. Sometimes though, it’s just the little things that matter, the attention to detail: try muting individual notes, or small groups of notes, to create pockets of space, dynamics and other changes of texture.

The Record MIDI to Track Here option lets you capture the output of any MIDI note‑generating plug‑ins. Select it and hit Record, and Logic will create a new region filled with editable MIDI notes.

Here’s how to do it. First, click-hold in a MIDI Effect slot and select Record MIDI to Track Here in the pop-up menu. The two small yellow triangles that appear will indicate you can record all the MIDI generated by any plug-ins that come before it. You simply press Record and watch as the MIDI plug-in spits out incredible quasi-fractal patterns directly onto your tracks. This all works very nicely when you’re playing a MIDI keyboard in real time, or using the Latch function in Arpeggiator. However, it won’t work when you already have MIDI regions on the track, as in my example, where I have already recorded simple block chords that I want to be arpeggiated. So, to avoid a MIDI feedback situation, you must first create a new, temporary MIDI or software instrument track, then activate the new ‘Internal MIDI in’ setting. You’ll find this in the Track Inspector. Use the pop-up menu to select which Logic or third-party software instrument (or external MIDI) track you’d like to record MIDI from. Then, simply record in real time. What you’ll end up with is the

unmodified original track, plus a new one with the arpeggiated MIDI region. You can now replace the old region with the new one, and of course, you may now turn off the arpeggiator. Top tip: rather than deleting the original region (after all, you may want to re-tweak it at a later date), mute the original regions (select them, then Control+M) and create a New Track with the Same Instrument (Control+Shift+Return). This hugely useful function simply creates a new ‘lane’ for the same instrument track, rather than a whole new set of CPU-draining, RAM-eating but duplicate plug-ins.

Layers Layering sounds is a very common production technique. You can easily layer your arpeggiated sounds by having an

The Step Sequencer is a handy alternative to Arpeggiator.

exact copy of the MIDI data on separate tracks. I found that by copying the MIDI region onto a new software instrument track loaded with the Studio Piano, and sending that on a bus to a stereo delay, I was able to add a very effective percussive element to the mix, whilst also thickening the sound’s texture and dynamically filling the stereo spectrum. Another tip: whilst Option+drag will copy the region, don’t forget that Option+Shift+Drag creates an ‘alias’, a virtual copy whose contents change whenever you edit the original region. Perfect for layering sounds before they’re fully edited!

Patterns In Repeat Finally, a word about the Step Sequencer (née Hyper Editor) in Logic, which provides an alternative yet very intuitive method of entering and manipulating notes in a variety of creative ways. To move between the Piano Roll and the Step Sequencer, Control‑click on any MIDI region, choose Convert in the pop‑up menu, then Convert to Pattern Region. This could be especially useful in our example, because it makes it very easy to click on a grid to switch individual notes on and off. But it also allows a great deal more: clicking on Velocity/Value adds the ability to change the length (gate time) and velocity of notes, add ties or slurs between notes, subdivide notes to create triplets or tuplets (up to 16 per semiquaver if you dare), as well as introducing elements of chance. There’s much more besides, and again I refer you back to a previous Logic column, this time in SOS December 2020, for a more detailed look. www.soundonsound.com / February 2026

105

SPOTLIGHT

LUKE WOOD

n the September 2025 issue, we rounded up some of the currently available options in the MPE controller market. They range from keyboard-based units like the Expressive E Osmose and ROLI Seaboard 2 to more experimental offerings that can be considered instruments in their own right — Roger Linn Design’s LinnStrument and the Embodme Erae 2, for example. Whichever approach you decide to take, your controller of choice will need to be paired up with an MPE-compatible instrument to fulfil its potential. As well as an abundance of software options, there’s now a healthy selection of hardware synths that boast MPE support, some of which have been built from the ground up to make the most of the protocol, and others which were already well-established instruments that have gained new expressive capabilities through firmware updates.

Sequential Trigon-6 Desktop Module

ASM Hydrasynth Desktop

Packing in the same wavetable engine as its keyboard-equipped counterpart, ASM’s Hydrasynth Desktop offers all the power of the standard instrument in a compact, rackmountable desktop enclosure. Its three oscillators offer a choice of 219 waveforms — two of them feed the company’s renowned Mutant processors — and are joined by a pair of filters (one multi-mode, one SEM-style state-variable) and no fewer than five LFOs and five envelopes. The instrument’s modulation matrix comprises 35 sources and 191 destinations, and its effects section houses a versatile collection of processors that range from EQs and distortion to reverbs and delays. A set of assignable macro controls make it possible to control up to eight parameters from a single encoder or button. As of firmware version 1.3 the Hydrasynth Desktop boasts full

ASM Hydrasynth Desktop

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MPE-Capable Desktop Synths

MPE is becoming ever-more prevalent, and plenty of hardware synths are getting in on the action. MPE support, allowing for individual pitch-bend, timbre and pressure control over each of its voices. As the instrument already supported polyphonic aftertouch, switching to MPE mode will result in MPE pressure automatically being mapped to aftertouch, while a selection of MPE options (MPE-X, MPE-Y absolute, MPE-Y relative) have been added to the instrument’s Modulation Sources menu. $ $779 W www.ashunsoundmachines.com

Black Corporation

There are a few MPE-capable options to choose from in the Black Corporation range. First up is the Kijimi Mk2, an eight-voice instrument which the company say was designed to be one of the most powerful polysynths ever built! Its oscillators continually morph through sine, triangle, sawtooth and pulse-width waveforms, while its two envelopes offer some interesting features including a cycling trapezoidal mode similar to

Black Corporation Kijimi Mk2

Dreadbox Artemis

that of the EMS Synthi. The ISE-NIN then offers the company’s take on the Roland Jupiter-8, while the Deckard’s Dream draws inspiration from the much-loved Yamaha CS‑80. Finally, there’s the Xerxes Mk2, an eight-voice polysynth that aims to combine the iconic sound of an Elka Synthex with modern expressive control. There are two digitally controlled analogue oscillators, a multi-mode filter, white and pink noise generators, and an analogue BBD chorus effect with three operating modes. The Xerxes MkII boasts a comprehensive set of envelope and modulation controls, and also supports micro-tuning. $ From $3499. W www.black-corporation.com

Dreadbox Artemis / Nymphes

The Artemis is the most recent addition to Dreadbox’s range, and combines the company’s signature analogue sound with a collection of digital stereo effects designed by DSP experts Sinevibes. It’s a six-voice design that features a pair of waveshaping oscillators with hard sync and through-zero FM, along with a resonant low-/high-pass filter, two ADSR envelopes, a pair of LFOs and an onboard sequencer and arpeggiator. As you might have guessed from its inclusion here, the Artemis offers MPE support straight out of the box. There’s also the Nymphes, another compact six-voice polysynth that combines a single oscillator with low- and high-pass filters, a pair of ADSR envelopes, one LFO per voice and

Groove Synthesis 3rd Wave Desktop

Erica Synths Steampipe

a built-in reverb. Despite its apparent simplicity, the Nymphes fully supports MPE and many of its parameters feature a polyphonic response. $ Artemis $1499, Nymphes $549. W www.dreadbox-fx.com

Erica Synths Steampipe

Designed in collaboration with Dutch plug-in developers 112dB, Erica Synths’ Steampipe brings something a little different to the table. It’s a physical modelling synthesizer that has no oscillators, opting instead for an envelope and noise generator that have been designed to mimic the behaviour of wind and string instruments. It’s probably no surprise that it’s been optimised for use with wind-based MIDI controllers, but it will also happily pair up with any MPE (or standard MIDI) controller. Precise control of the instrument’s physical modelling parameters means that it’s capable of producing organic emulations of acoustic instruments, but it goes well beyond their conventional pitch and dynamic ranges and, of course, can be pushed into far more experimental territory. $ $1289 W www.ericasynths.lv

Groove Synthesis 3rd Wave Desktop / 8M Groove Synthesis were met with significant praise when they launched their PPG Wave-inspired synth, and since its initial release its been treated to a series of updates and gained some powerful new features. It’s also

now available in the form of the 3rd Wave Desktop, a module version that packs in all the power of the flagship synth, and with the exception of a handful of top-panel controls, delivers exactly the same programming experience. Each of the instrument’s three oscillators can function as a wavetable (classic PPG and modern modes are provided) or an analogue-modelled waveform, while filter duties fall to a Dave Rossum-designed 2140 analogue low-pass and an SEM-style state-variable option. As well as offering a generous 24 voices, the synth is four-part multitimbral, with each of the four layers treated to their own independent modulation matrix. The more recent 3rd Wave 8M then offers a similar experience, but comes in at a lower cost thanks to a reduction in voice count (eight instead of 24), and a simplified front‑panel design that offers fewer physical controls. Much like their keyboard-equipped sibling, both modules now fully support MPE. $ 3rd Wave Desktop $3495, 3rd Wave 8M $1999.

W www.groovesynthesis.com

GS Music Bree6 / e7

The Bree6 is the second synth to be launched by Argentina-based GS Music, and manages to pack some serious functionality into a sleek, compact enclosure. It’s a six-voice analogue polysynth, and features a single oscillator that includes sawtooth and pulse waveforms along with a modulatable pulse-width modulation control; there’s also a square-wave sub-oscillator. Its filter employs a resonant four-pole low-pass design, and a pair of ADSR envelopes

GS Music Bree6

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MPE-C APABLE DESKTOP SYNTHS

Oberheim OB-X8 Desktop

Melbourne Instruments Nina

are joined by a single LFO, while some stereo effects are provided in the form of chorus and delay. Importantly, the Bree6 supports the use of MPE controllers, and GS Music say they have carefully tested its implementation with the Expressive E Osmose, Roger Linn Designs LinnStrument and moForte’s GeoShred software. MPE support is also present in the company’s debut release, the e7 — a larger seven-voice polysynth that features a ladder-type low-pass filter. $ Bree6 $999, e7 $1899. W www.gsmusic.com

Melbourne Instruments Nina

Melbourne Instruments’ Nina combines a trio of oscillators (two analogue and one digital wavetable) with a Moog-inspired ladder filter, allowing it to cover a wide variety of classic sounds as well as stepping into more modern territory. It’s a 12-voice instrument that boasts four-part multitimbrality, and includes an innovative Morph function that makes it possible to blend seamlessly between entire patches. There are two ADSR envelopes and a powerful modulation matrix, and thanks to the motorised pots, the instrument offers instant recall of settings and is capable of displaying automation and modulation directly on the top-panel controls in real time. The Nina has always been equipped with a powerful sequencer, but thanks to a series of firmware updates it now offers four independent, polyrhythmic, polyphonic tracks that benefit from per-step automation. There’s a dedicated mode that offers full MPE control across the instrument’s multitimbral layers, and also adds a decay envelope for expressive sound control on MPE aftertouch and CC74. What’s more, Melbourne Instruments have teamed up with Expressive E and sound designer James Terris to create an exclusive MPE preset pack specifically for users pairing the Nina with an Osmose. $ $3599 W www.melbourneinstruments.com

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Oberheim OB-X8 Desktop / OB-6

Introduced in 2022, the OB-X8 aims to deliver the ultimate Oberheim experience by bringing together three of the legendary designer’s renowned designs: the OB-X, OB-Xa and OB-8. Key elements of each design, including the all-important filters, were faithfully reproduced in order to capture the sound of the original instruments, and the new flagship synth also makes it possible to ‘mix and match’ elements to create sounds that weren’t obtainable on the originals. The newer OB-X8 Desktop features an identical engine to its keyboard-equipped counterpart, with its discrete VCOs, VCAs and filters complemented by knob-per-function panel design — despite the lack of a keyboard, the module retains the arpeggiator and keyboard split functions, keeping its abilities consistent with the OB‑X8 when connected to the user’s choice of controller. Although there was no support for MPE when the instrument launched, it has since been added with the free OS v2.0 update, making the OB-X8 Desktop fully compatible with MPE controllers. The OB-6 module, which was developed in a collaboration between Tom Oberheim and Dave Smith — and originally released under the DSI/ Sequential brands — is now officially a part of the current Oberheim line-up, too. The module features a broadly similar architecture and layout to the Sequential Prophet-6 (more of which later), but employs a two‑pole state‑variable resonant filter based on the Oberheim SEM. $ OB-X8 $3499.99,

OB-6 $2399.99. W www.oberheim.com

Sequential Prophet-6 / Trigon‑6 Desktop Modules

There’s plenty to choose from in the Sequential line-up, with the desktop module versions of the Prophet-6 and more recent Trigon-6 both boasting MPE support. As their names suggest, both are six-voice instruments, with the former offering a modern take on the company’s vintage designs and the latter delivering the last synth designed by founder Dave Smith. The Prophet-6 combines a pair of discrete VCOs with a two-pole resonant high-pass filter and a Prophet‑5-inspired four-pole resonant low-pass filter. The Trigon-6 then boasts a three-oscillator design — with triangle, sawtooth, reverse sawtooth and variable-width pulse waves — and features a switchable two‑/ four-pole resonant low-pass ladder filter. If you’re after more voices, the Trigon-6 is equipped with a Poly Chain feature that allows two units to behave as a single 12-voice instrument. Both modules will happily respond to MPE controllers. $ Prophet-6 $2299.99, Trigon-6 $2499.99. W www.sequential.com

Supercritical Redshift 6

The Supercritical Redshift 6 has been treated to an array of new features since it launched in 2024 — with yet more already planned for introduction in 2026 — and as of October 2025 it supports MPE. There are no fewer than 16 analogue oscillators per voice, and thanks

Sequential Prophet-6 Desktop Module

to a set of Character Profiles, the tuning, gain staging, filter, amp and envelope behaviour of each of the instrument’s six voices can be adjusted to deliver everything from clean, modern tones to “distorted misbehaviour and beyond”. The four-pole, state-variable analogue filter is equally versatile, and is said to be capable of emulating just about any classic vintage design. There are four envelopes and four LFOs, and a generous selection of DSP-based reverb, delay and modulation effects. $ $1349 W www.supercriticalsynthesizers.com

Supercritical Redshift 6 Twisted Electrons TWISTfm

Twisted Electrons TWISTfm

Aimed at those looking to “push the boundaries of sonic creativity”, Twisted Electrons’ TWISTfm offers a combination of FM synthesis with analogue filtering and modern control. The instrument employs a pair of Yamaha OPL-3 YMF262 FM chips, delivering eight voices that are each complemented by their own multi-mode filter. A total of 15 filter modes cover low-, high- and band-pass behaviours Synclavier say along with phasing, that the Regen notch and more, brings together while Liquid and all of the sounds MS20 resonance from their rich modes provide both legacy, as well as subtle and aggressive introducing an array filter characters. There’s of modern synthesis a built-in 128-step sequencer with types and expressive Synclavier Regen MIDI sync and looping options, control. Its DSP engine offers and as well as supporting MPE, the additive, subtractive and FM synthesis instrument is equipped with an onboard as well as sampling, all of which can be ribbon controller. mixed together (a patch can contain up to 12 partials) with the ability to assign $ $1199 W www.twistedelectrons.com crossfades to the likes of velocity, pressure, mod wheel and keyboard inputs. MPE and polyphonic aftertouch are both fully supported, and thanks to The module version of UDO Audio’s a comprehensive Modulators menu, it’s Super 6 packs all of the features found possible to adjust the response curves of in the keyboard version into a compact the parameters that are linked to control enclosure that can be used as a desktop inputs. There’s an onboard arpeggiator, instrument, rackmounted with the and the instrument also comes loaded optional ear attachments, or held in with a wealth of effects including chorus, place just about anywhere else thanks to reverb, vibrato, tremolo and bit-crushing. a built-in VESA mount. Despite its name, the instrument actually has 12 voices, but $ $2499 W www.synclavier.com features a binaural mode that combines them into six ‘Super’ voices — left/ right pairs of voices that are routed and processed separately by the synth’s true stereo signal path, allowing for everything from subtle stereo movement to more extreme creative effects. There are two FPGA-based oscillators, a four-pole low-pass filter and simple high-pass filter (which can be used together to mimic a band-pass), two envelopes and two LFOs; there’s a powerful onboard UDO Audio Super 6 Desktop

Synclavier Regen

UDO Audio Super 6 Desktop

sequencer and arpeggiator, too, along with built-in high-quality stereo chorus and delay effects. And of course, its inclusion here means that the Super 6 Desktop is capable of responding to MPE. $ $2385 W www.udo-audio.com

Waldorf Iridium Desktop

Described as a “direct descendant of the PPG Wave”, the Waldorf Iridium Desktop makes the engine that powers their renowned Quantum synth available in a more compact form. Its three oscillators offer a choice of five modes: wavetable, virtual analogue, particle (sampling and granular), resonator and kernel, the last of which allows each oscillator to be divided into six oscillators. There are three filters per voice (12 or 24 dB/octave low-, high- and band-pass), six envelopes, six LFOs and a 40-slot modulation matrix. Despite its complex nature, the synth promises an intuitive programming experience, with a wealth of physical controls joined by a powerful touchscreen system. MPE is supported via both DIN and USB MIDI inputs, and by default, a controller’s X, Y and Z axes are mapped to the pitch-bend, modulation source and channel aftertouch parameters in the Iridium’s engine. Waldorf also offer the Iridium Core, a scaled-back version of the instrument that features the same sound engine but with 12 voices and fewer physical controls. $ Iridium Desktop $3499, Iridium Core $2199.

W www.waldorfmusic.com

Waldorf Iridium Desktop

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Chosen By The Readers Of Sound On Sound The SOS Awards celebrate innovation and contemporary design, with nominations being limited to products launched in the 12 months prior to the voting period. These results celebrate the ‘best in class’ products launched or reviewed in the 12 months from September 2024 through to the end of August 2025. Congratulations to all the winners, as well as the very worthy close contenders in each category who have been awarded our Highly Commended accolade.

WINNER

BEST AUDIO INTERFACE

UNIVERSAL AUDIO APOLLO X GEN 2 www.uaudio.com

WINNER

BEST EFFECTS & PROCESSING HARDWARE

BEST DAW

www.warmaudio.com

www.apple.com

WARM AUDIO WA76 SERIES 112

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APPLE LOGIC PRO

WINNER

BEST GUITAR & BASS TECHNOLOGY

UNIVERSAL AUDIO UAFX ENIGMATIC ’82 www.uaudio.com

WINNER

BEST SOFTWARE PLUG-IN

FABFILTER PRO-Q 4 www.fabfilter.com

WINNER

BEST MUSIC SOFTWARE

STEINBERG SPECTRALAYERS PRO 12 www.steinberg.net

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SOS AWARDS 2026

WINNER

BEST PERFORMANCE CONTROLLER

BEST KEYBOARD/SYNTH

www.arturia.com

www.moogmusic.com

MOOG MESSENGER

ARTURIA KEYLAB MKIII

WINNER

BEST STUDIO MONITOR

ADAM AUDIO D3V www.adam-audio.com

WINNER

BEST MIXING CONTROLLER

SSL UC1

www.solidstatelogic.com

WINNER

BEST MICROPHONE

SHURE SM4 www.shure.com

WINNER

BEST DRUM MACHINE

ROLAND P-6 www.roland.com

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FE ATURE

SOS AWARDS 2026

WINNER

BEST MICROPHONE PREAMP

SSL REVIVAL 4000 www.solidstatelogic.com

WINNER

BEST HARDWARE AUDIO RECORDER

ZOOM H5STUDIO www.zoomcorp.com

WINNER

BEST SOFTWARE INSTRUMENT

NATIVE INSTRUMENTS KONTAKT 8 www.native-instruments.com

WINNER

BEST HEADPHONES

BEYERDYNAMIC DT 1990 PRO MKII www.beyerdynamic.com

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WINNER

BEST LIVE SOUND PRODUCT

ALLEN & HEATH QU SERIES www.allen-heath.com

To find a dealer visit www.musicmarketing.ca

FE ATURE

SOS AWARDS 2026

AUDIO INTERFACE

SSL 18 www.solidstatelogic.com

HIGHLY COMMENDED SOS AWARDS 2026

EFFECTS & PROCESSING HARDWARE

NEVE 88C www.ams-neve.com

GUITAR & BASS TECHNOLOGY

NEURAL DSP NANO CORTEX www.neuraldsp.com

DAW

SOFTWARE PLUG-IN

AVID PRO TOOLS

NATIVE INSTRUMENTS KOMPLETE 15

www.avid.com

www.native-instruments.com

PERFORMANCE CONTROLLER

NOVATION LAUNCH CONTROL XL 3 www.novationmusic.com

MUSIC SOFTWARE

STEINBERG DORICO 6 www.steinberg.com

KEYBOARD/SYNTH

ROLAND V‑STAGE www.roland.com

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DRUM MACHINE

TEENAGE ENGINEERING OP‑XY

MIXER & MIXING CONTROLLER

teenage.engineering

SSL ORACLE www.solidstatelogic.com

MICROPHONE

SENNHEISER MD 421 KOMPAKT www.sennheiser.com

SOFTWARE INSTRUMENT

ARTURIA V COLLECTION 11 www.arturia.com

STUDIO MONITOR

IK MULTIMEDIA ILOUD MICRO MONITOR PRO www.ikmultimedia.com

HARDWARE AUDIO RECORDER

ZOOM LIVETRAK L6 www.zoomcorp.com

LIVE SOUND PRODUCT

UNIVERSAL AUDIO APOLLO E2M www.uaudio.com

HEADPHONES

BEYERDYNAMIC DT 70 SERIES www.beyerdynamic.com

MICROPHONE PREAMP

HARRISON AUDIO 32CLASSIC MS www.harrisonconsoles.com www.soundonsound.com / February 2026

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Akai MPC60

Join us as we travel back in time to see how the story of one of the most important machines in modern music began. ALEX BALL

ost of us would be happy enough to have one famous musical invention to our name, but some people have produced so many that they almost seem run-of-the-mill. Roger Linn is a case in point. Products he’s created have helped define the sounds of genres and, arguably, even entire periods in popular music. And none has been more influential than the original grid-based sampling beatbox, the Akai MPC60.

The LM-1 Roger Linn was born in California in the 1950s. Unlike many of his later peers, Linn didn’t have any formal training with electronics or engineering, and instead was a musician who taught

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himself what he needed to know in order to satisfy his sonic curiosities. In the early days, he’d open guitar pedals and tweak and modify them to produce different flavours of effect. A keen instrumentalist and songwriter, at only 21 Linn landed a gig as a touring guitarist for Leon Russell. Linn subsequently managed to persuade Russell to use the fruits of his career to buy a synthesizer for use in the studio, and Russell wound up purchasing a modular system from a fledgling company called E-mu Systems. Experimenting with this new technology, the ability to use the E-mu’s sequencer to modulate its filter as external audio ran through it was particularly compelling to Linn. Whilst working with Russell, Linn also first experienced a musician using a rhythm machine to lay down a beat that

the band would record to. This was done partly for timing, but also because it meant that if Russell then wanted to re-record the drums or something else rhythmic, there would be a rock-solid timing reference to work from. This ability to create drums artificially was interesting to Linn as it was an ingredient he struggled to produce for his own demo recordings, having never learned to play them himself. However, there were two problems: firstly, the machines were all based around preset, factory rhythms and couldn’t be programmed, (with a couple of exceptions — see the ‘Outliers’ box), and secondly, the analogue drum emulations of the day weren’t known for their realistic sound. Setting out to solve the first problem, in 1977 Linn became competent with computing and built a programmable

drum machine and sequencer using a COMPAL-80 and a Frankensteined sound board from a Roland rhythm unit. The sequencer could then control a synthesizer via CV and gate to boot. This custom machine was used on Leon Russell’s album Life & Love, which Linn co-produced, engineered and provided guitar for. Next came the second problem. Analogue drum machines that employed oscillators, noise generators and envelopes sounded like alien interpretations of acoustic drums, rather than the real thing. Whilst the sounds are steeped in nostalgia, charm and history nowadays, at the time manufacturers were looking for a more realistic solution. Fatefully, it was Linn who got there first because, fortuitously, around this time, it was becoming possible to record digital audio at a price that might just be within the realms of practicality from a commercial perspective. Whilst RAM was highly expensive, drum sounds were inherently short and were therefore a perfect, skinny contender for a product that was loaded with digital recordings of real instruments (although longer cymbal sounds would have to wait on the sidelines for the time being). Linn booked session drummer Art Wood into a studio and captured the sounds he’d need for his new machine. At 28kHz and 8-bit, the resulting lo-fi recordings were raw and crunchy, but, crucially, they were a huge breakthrough for the time. Nestled into a sleek chassis with hands-on controls and extremely useful new features such as tap recording, auto-correct, adjustable swing, pattern chaining and more, the resulting LM-1 was an instant hit, despite having a $5000 price tag upon its release in 1979. Once the LM-1 was out, it’s not an exaggeration to say that the drum palette of popular music was changed forever, and despite there only being around 500 units produced, the LM-1 was heard on the hits of the Human League, Fleetwood Mac, Michael Jackson, Queen, Prince and Vangelis amongst many, many others, and remains a prized possession to those who own them. With the success of the LM-1 in the bag competitors were working hard to try to catch up. With the technology improving and coming down in price, it was clear there needed to be a successor, and that came in 1982. The first instrument had used the initials ‘LM’ because the company

had been founded by Roger Linn and Alex Moffet. However, Moffet departed in the early ’80s and so the successor was simply titled the LinnDrum, rather than LM2. With a smaller profile, better fidelity, and more features and sounds, the LinnDrum went on to become one of the most famous drum machines of all time. With the LinnDrum outselling the LM-1 at almost 10:1, if a song was in the charts in the ’80s, the chances are that there was a LinnDrum on it. It would take so long to list all the artists that used the LinnDrum that it would dominate this article, so probably the best way to sum it up is by mentioning the fact that there is an archive on the Internet that lists songs that contain confirmed use of the LinnDrum, and at the time of writing, it has nearly 1900 entries!

Near Misses Following the accomplishments of the LM-1 and LinnDrum, you’d be forgiven for thinking that Linn Electronics rode the success happily into the sunset but, alas, history is littered with cruel twists of fate for inventors, and Linn’s story is no exception. Had it not gone wrong, however, we probably would have never wound up with the star of our story, the Akai MPC60. But before we get there, we’ve got to roll back to 1984 when Linn unveiled their third drum machine, the Linn-9000. The 9000 was a bold attempt to move forward and significantly, the unit was able to sample, had programmable hi-hat decay, doubled as a MIDI sequencer and contained a grid of square, touch-sensitive rubber pads and a liquid crystal display. All seemingly quite familiar... Whilst it was a powerful box that, like its predecessors, appeared in hit records, the 9000 was riddled with technical problems. As well as costly power supply issues on early units, the firmware was buggy and prone to crashes. In fact, on a personal note, I once had a conversation with the late Bob Heatlie who recalled his unit losing an album’s worth of work for a Neoton Família album he was producing the day before he flew to Hungary for the sessions. His Linn-9000 found itself in landfill! There was also a sister MIDI sequencer called the LinnSequencer that was unfortunately endowed with the same buggy operating system of the Linn-9000, and the two products, combined with fierce competition from other manufacturers, made the outlook

rather grim for Linn Electronics as 1985 drew to a close. In 1986, Linn arrived at the Winter NAMM show with a prototype of a new product, the LinnDrum Midistudio, but this never saw the light of day as the company went out of business only a few weeks later. The photos of this never-released product show a clear evolution from the Linn-9000 with sampling, MIDI sequencing and a hi-hat decay slider on the left of the rubber pads, now arranged in a 4x4 grid...

Seeing Red The demise of Linn Electronics proved to be something of a blessing disguise, because it opened the door for a new opportunity; Japanese company Akai (originally founded back in the 1920s) had established themselves with reel-to-reels and hi-fi equipment in the preceding 20 years and had branched out and founded a music division, Akai Professional, in 1984. By 1986, they’d tested the waters with a few synthesizers, such as the AX80 and AX60, and had begun developing samplers like the S612 and VX7000. Whilst they were good at the production, sales, finances and logistics, Linn had a track record with innovation and product design and his availability was an opportunity too good to miss. With Akai now backing him, Linn picked up where he’d left off at his own company and began working on a more refined and complete version of the Linn-9000, Sequencer and Midistudio that would ultimately hit the stores in 1988. The team at Akai who worked with Linn included Dave Cockerell of EMS fame, who had already created the S612 sampler and who would later work on the classic S1000 with two other famous engineers, Tim Orr and Chris Huggett. Initially the new product was titled the ADR-15, with the letters being an abbreviation of Akai DRum machine, but this title belied what was to be the full functionality of this new device. The concept was to make a multitrack digital sequencer and, indeed, the final production name MPC stands for ‘MIDI Production Center’. However, the MPC was still a drum machine if the user wished, as it was able to play back its own sounds too. These could be loaded from floppy disks, but crucially, the MPC had an input on the rear and 13.1 seconds of mono audio (or 26.2 seconds with the memory expansion) could be sampled into the unit and then be edited and saved onto floppy to be recalled later. This feature was one www.soundonsound.com / February 2026

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AI & Music Tech In 2026 We asked music professionals how AI is changing their world. SAM INGLIS

o-called artificial intelligence is the hottest topic in music tech. Generative AI services such as Suno and Udio allow users to create entire songs from simple prompts. Will they make conventional music production redundant? Or will producers take advantage of a wave of innovative, assistive AI tools to do their work faster, better and smarter?

Sonarworks are well known to SOS readers for their SoundID speaker and headphone correction tools. They too have recently branched out into AI, with the launch of their VoiceAI virtual vocalist. With new AI products seemingly appearing every week, Sonarworks decided it was a good time to take stock. Are musicians and producers embracing these tools? How are they using them, and what for? What opportunities and challenges does the rise of AI open up? To answer these questions and more, Sonarworks commissioned Sound On Sound to carry out a comprehensive survey. Almost 1200 music creators responded, and we can be confident that the results are representative: over 70 percent are working professionals with more than a decade’s experience in

The respondent base for our survey represents a true cross-section of the music production world.

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production and engineering. This is one of the first comprehensive surveys of attitudes towards AI in music production, and as such, provides a unique snapshot of the current situation. You can read Sonarworks’ full analysis on their website; in this article, I’ll try to draw out some of the most important conclusions.

The Only Way Is Ethics In any technological revolution, there will be early adopters, and a larger body of people who are more cautious. That pattern is clearly evident in this survey. Around one in five respondents self-describe as early adopters of new tech in general, and the same proportion claim to be regular users of AI products or services. By far the largest group, around half of the total, describe their use of AI as

Our data shows that one in five producers is a regular AI user already, while almost half have used it experimentally.

being in the occasional or experimental stage, whilst less than 20 percent express no interest in learning or using AI tools. When producers were asked what factors limit their use of AI, the most widespread concern that emerges is around creativity. More than a third of respondents worry that relying on AI tools would compromise their creative intent, or undermine the individuality of their work. Nearly as many express unease about the ethics of artificial intelligence. With many issues surrounding copyright and training still unresolved, some feel it’s too risky to depend on tools that have an uncertain legal basis, and that the use of AI tools is exploitative of the other musicians on whose work they may have been trained. More than a quarter of respondents also report that the AI tools they’ve tried simply aren’t yet of sufficient quality to be useful in a professional context. In many cases, this links back to the concern about creativity, with the output of AI often described as generic or bland. Whether or not participants are actually limiting their use of AI on ethical grounds,

Loss of creativity and ethical doubts top respondents’ list of concerns about AI.

it is almost universally felt that attention to the ethical basis of AI products is non-negotiable. Asked to evaluate the importance of ‘ethical sourcing’, two-thirds of respondents rate it very important, and fewer than one in 10 rate it unimportant or relatively unimportant. Survey respondents are also alive to the ethical implications of their own use of AI, which principally centre around disclosure and transparency. A large majority

advocate either full transparency over the use of AI, or ‘situational disclosure’, meaning they would acknowledge their use of AI if asked. Only 6 percent say they would prefer to deliberately conceal their use of AI.

Levelling Up Concerns about creativity and the ethics of AI are reflected in assessments of its most fruitful use cases. Just under a third

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AI AND MUSIC TECH IN 2026

see it as a source of inspiration, while a similar proportion see it as enabling new forms of collaboration, or a route to saving money. However, almost three-fifths of the survey participants recognise the potential for AI to speed up or automate tedious tasks such as vocal tuning, drum editing and file management. In other words, the applications to which respondents are most open are the most utilitarian ones. Producers are most willing to use AI for tasks they see as technical rather than creative. This attitude ties in with the answers to wider questions about the skill sets that modern professionals need. There is a widespread perception that once-essential technical skills such as notating and transcribing musical scores, manual audio editing and noise reduction are increasingly becoming redundant, and that this process is being accelerated by AI. Interestingly, a significant number of respondents also feel that basic ‘bread and butter’ mixing skills such as balancing, corrective EQ and dynamic control are becoming optional, with AI tools picking up the slack. By contrast, the skills that are perceived as essential for modern producers and musicians are primarily the creative ones — musicality, instrumental ability, understanding of music theory, critical listening and compositional ability — alongside the ‘soft skills’ that foster collaboration and enable producers to get the most from people. It is also widely felt that engineers who specialise in old-school analogue recording will continue to flourish, and that their skills will prove resistant to AI replacement. Half of participants agree that the most important differentiator for human producers in the AI era is the ability to find and pursue an original creative direction. Respondents are also alive to the need to develop new skills, with around two-fifths including the need to master AI tools as one example. Almost the same proportion point to the increasing importance of networking and marketing, and the same again to the need to look beyond music and acquire expertise in multimedia, video, virtual reality and other intersecting technologies. Perhaps most remarkably, almost a third of participants view coding and scripting as important for music production. As Sonarworks point out, these results may have implications for

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Efficiency is seen as the most likely benefit of AI, but many also perceive it as being a valuable creative partner.

The most popular AI tool categories are currently those with relatively utilitarian functions.

Jazz/blues and classical music are the genres seen as most resistant to AI.

educational institutions offering music tech programmes. Potential students will seek out courses offering what they perceive to be the skills needed to succeed in today’s landscape — and they may not be the ones that are favoured on existing curriculums.

Across The Board Participants were asked to name up to three main genres in which they

work. Around 45 percent specify rock and pop, with electronic music not far behind at 38 percent. The next most popular genres are ambient/new age and film or game scores, both polling around 24 percent. Only 13 percent cite metal as a primary genre, perhaps reflecting its longstanding status as a niche subculture. It is widely anticipated that the impact of AI will vary with musical genre.

“AI is here to stay. Although a significant number feel it’s over-hyped, only 3.6 percent believe it to be a passing fad, and nearly a third see it as revolutionising the music industry.” Two-thirds of respondents feel that EDM and mainstream pop will prove amenable to AI generation and replication, whilst ambient and ‘functional’ music are also expected to do so by half of the people answering the question. Interestingly, both game/film scores and hip-hop/rap are seen as much less vulnerable to AI, for rather different reasons. Bespoke soundtracks are relatively long-form and require a clearly identifiable compositional voice, whilst rap centres the human performer in a way that’s perceived as hard to replace. At the other end of the scale, the genres viewed as most resistant to AI generation are jazz, blues and classical music. It is felt that the subtlety and nuance of human interaction in these genres will prove hard to replicate.

help them to realise their vision faster and more effectively. A further 21 percent selected “major automation with human oversight” as the most likely future for AI in music, with only 9 percent expecting “full automation” and human producers becoming largely redundant.

In short, the attitude of a majority of participants towards AI can probably be summed up as grudging acceptance of the inevitable. Consequently, developers of AI tools still have a lot of work to do in order to win hearts and minds in the music production community. AI products are most likely to gain acceptance if they can demonstrate their ethical credentials, and offer genuine assistance with jobs that human producers and engineers find tedious. Above all, they need to ensure that their users feel in control of the creative direction at all times. Artificial intelligence is both a novel challenge and a unique opportunity — but if it ends up sidelining human creativity, there will be no winners.

Not All Doom & Gloom Overall, participants express predominantly sceptical or negative judgements about AI in music technology. Only a fifth describe their feelings about AI as positive or very positive, whilst the remaining 80 percent are divided about equally between neutral, and negative or very negative. However, given the emphasis placed by most on creativity, it may be that this negativity relates less to assistive AI tools targeted at music production, and more to generative AI services that threaten to bypass the role of the music producer altogether. At the same time, the large majority of survey participants acknowledge that AI is here to stay. Although a significant number feel it’s over-hyped, only 3.6 percent believe it to be a passing fad, and nearly a third see it as revolutionising the music industry. Going deeper, a small majority of respondents (58 percent) see the eventual role of AI in music production as primarily a supportive one. In this vision, human producers and musicians will retain firm control of the creative reins, but assistive AI tools will www.soundonsound.com / February 2026

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INSIDE TRACK

Matt Schwartz • Yungblud Yungblud capped a triumphant 2025 with a hit collaboration with Aerosmith. Long-term producer Matt Schwartz was behind the desk. PAUL TINGEN

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Photo: Kyle Hoffmann

Matt Schwartz at NRG Studios, one of many studios used for the making of One More Time.

roducer Matt Schwartz vividly recalls his first meeting with a teenage Dominic Harrison. “I met Dom in 2016, when he was 19. Labels sometimes send me artists for a chat to see if they’re the real deal or not, because I’m pretty harsh. I dig into someone’s soul. So my manager and friend Tommas [Arnby] asked me to check out this kid he’d found. Dom came in, and played me some songs, and I stopped him, and said, ‘What is this? It sounds like bad Ed Sheeran. We already have Ed Sheeran. And do you think teenagers are stupid? If not, why are you writing stupid lyrics? If you want to work with me, I want another Bowie or Freddie Mercury. If you want to be an artist, figure out what you want to say, and what culture you want to build.’ “Eight months later, Tommas calls me again, and says, ‘You got to meet the kid again.’ I’m like, ‘Why?’ And he says, ‘Because you don’t realise what that meeting has done to him. He locked himself in a bedroom for eight months and figured out who he wants to be as an artist.’ So Dom came in, and had this crazy look in his eyes, and his whole mannerism had changed. The first song he played me was ‘King Charles’, and I was blown away. We recorded that song there and then, in four hours, including mixing and mastering. Yungblud was born. We also did the next five records really quickly. I was thinking to myself, ‘He’s so young, and to write those lyrics with that vocabulary is amazing.’ Over time I also became a much better writer myself through working with him. His ability has pushed me to places I never thought I could go.” ‘King Charles’ was released as Yungblud’s first single in 2017. The artist’s first album, 21st Century Liability, produced by Schwartz, Martin Terefe, Sluggo and Patrick Russel, was released in 2018. Aided by hit singles like ‘11 Minutes’ (2018, with Halsey and Travis Barker), which Schwartz

your crazy ideas. Starting with a backing track limits what you can do with the song.”

Going Slow

Matt Schwartz (at the console) with Aerosmith and Yungblud at Baysound Studios, Sarasota.

Matt Schwartz: “If you’re an artist, never let anyone else start your ideas for you. Ever. If you work with somebody else, you need to start the idea.” co-wrote and produced, and ‘I Think I’m Okay’ (2019, with Machine Gun Kelly and Travis Barker), Yungblud’s star continued to rise, culminating in 2025 in the number one album Idols and the EP One More Time (with Aerosmith) and his momentous performance at the Black Sabbath Back To The Beginning concert.

Idol Moments “Generally, it’s just Dom and I, in an enclosed environment, without outside interference. I’ll usually sit at the computer and record. He plays everything, I play everything, we’re good. We’re not virtuosos, but with the computer we’re the best. We’ll grab guitars, or are on a piano, or sometimes a bass line triggers something, and we’ll sing some ideas. Once we have a rough idea, we’ll jump on a microphone and develop it. Sometimes we start with

a concept or a lyric. A melody inspires you to do another melody or another lyric, and suddenly it morphs into a story. It’s an amazing process. “The one thing that is crucial is that I insist that Dom starts the idea, every time. If you’re an artist, never let anyone else start your ideas for you. Ever. If you work with somebody else, you need to start the idea. They need to be inspired by your ideas, not you by their ideas. That’s the rule. That’s what real artists who write their own songs do. You cannot be in the shackles of a record label. You cannot be in the shackles of a producer or another writer, or whatever. The greatest artists always inspire the people around them. Whether it’s in music, whether it’s in style and fashion, whether it’s in ideas of how to promote things and marketing — you have to inspire, you have to get everybody to go with all

The opener of Idols and also the album’s first single, a nine-minute epic called ‘Hello Heaven, Hello’, was co-written by the two of them, plus guitarist Adam Warrington, and composer and producer Bob Bradley, known for his work with Jeff Beck. “We went up to Bob’s studio in north England, and Dom had the idea for the first section, which was good. He went to bed and I said to Adam, ‘Let’s try something.’ We came up with the basis for what became the middle section, and I thought, ‘This is effing sick, let’s connect it to the first section. Dom must have heard what we had done while asleep, because when he woke up, he thought it was part of a dream. He said, ‘I have some ideas, let me go on the mic,’ and I was like, ‘Go go go.’ He went in and immediately came up with what’s on the record. It was insane. Dom found a real and natural persona on that track that really inspired the rest of what we were doing. “Then I was like, ‘You know what? It’s missing a third part.’ Everybody thought I was crazy, but we got this great other idea. I started orchestrating it, and tried to fit the three sections together. However, the problem was that the new section was in a different bpm. How do we go from 83 bpm to 94, or the other way round? I decided to automate the tempo change in Logic, but because of a slip of the mouse, or whatever, the tempo dropped to 5bpm. It went all the way to 5bpm and then back up to 94, and it sounded incredible. Everybody on the sofa looked at me with their mouths wide open. “When we went to live performance, the guy who was doing FOH was using Ableton, and I sent him the parts he had to play from his computer, with a MIDI file, so he could do the tempo change. However, it was not working in Ableton, though it worked fine in Pro Tools and Logic. We found that Ableton only goes down to 20 bpm. I contacted Ableton, and told them. They were like ‘What? We’ve never heard anybody go to 5bpm. Why?’ I’m like, ‘Well, there’s always one, isn’t there?’ I hope they’re sorting it out.”

Time Travel Although the arrival of One More Time half a year after the release of Idols and Yungblud’s Birmingham performance appears like a natural progression, the collaborative EP was a long time in the www.soundonsound.com / February 2026

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Our countdown of the most influential music tech breakthroughs reaches the Top 20. THE SOS TEAM

o mark the 40th anniversary of your favourite music recording magazine, we’ve put together our own league table of the 40 most important technical achievements in the history of recorded music. We begin the third instalment in the dark days of World War II... 20: AC bias Early in the history of magnetic recording, it was discovered that impressing a strong, fixed bias signal onto the medium ensured that it operated within its linear zone, reducing noise and distortion. In early wire recorders and tape machines, a simple DC signal was used, but in the late ’30s and early ’40s, various researchers more or less independently discovered that employing

an AC signal instead resulted in dramatic performance improvements. The effect was first exploited commercially by AEG during WWII, reducing noise to the point where it was no longer possible for the Allies to discern whether a radio broadcast was live or pre-recorded. After the war, two of these AC-biased Magnetophon machines were taken back to the USA, where they inspired the creation of the first Ampex tape recorders.

frequencies. Studio engineers were quick to realise that these filters could be used to ‘improve’ signals as well as to restore them, but they remained relatively inflexible until the invention of the parametric equaliser. The term seems to have been coined by George Massenburg, working with Burgess Macneal and Bob Meushaw, but Daniel Flickinger patented the first practical implementation in 1971. Suddenly, it was possible not only to sweep an equaliser through the frequency range to locate the region of interest, but to determine how focused the cut or boost needed to be. Parametric EQ quickly became a standard tool in the audio engineer’s armoury.

19: Parametric equalisation In telephony and other early audio technologies, the frequency content of a signal could become skewed in unwanted ways, for example by high-frequency loss in long cable runs. The earliest equalisers were simple filters that could correct this loss to restore an ‘equal’ balance of Photo: George Shuklin

As well as inspiring the tape recording revolution in the USA, AEG Magnetophon recorders also ended up behind the Iron Curtain, like this one from a Russian military museum.

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18: Lossy data compression In the ’80s, file sizes and bandwidth limits presented serious obstacles to the distribution of digital audio. Yet it was widely understood that many of the ones and zeroes in a digital bitstream represented content that was imperceptible. Could the size of the bitstream be reduced by, in effect, including only the information that was audible? Indeed it could, and numerous different technical solutions had been developed by the end of the decade. The direct effect on music recording and production was minimal, but the indirect effect was devastating. When the MPEG Audio Layer 3 standard was adopted in the early ’90s, it was envisaged that end users would only ever access MP3 decoders; encoding would remain a tightly controlled, proprietary process. Instead, both encoders and decoders became ubiquitous and free, and file-sharing services such as Napster wiped out music industry revenues almost overnight. 17: The Portastudio In 1979, Tascam kicked off a revolution in recording when they released the first Portastudio, a compact (for then!) four-track cassette recorder with a simple built-in mixer

that made low-cost multitrack recording accessible to the bedroom musician, and let professionals make demos pretty much anywhere — and, as in the famous case of Bruce Springsteen’s Nebraska, even more than just demos! As well as the machines themselves being more affordable than the open-reel tape recorders that had previously been a necessity, so too was the medium, and cassettes were readily available on every high street. Various models followed from Tascam and others before, in 1995, Fostex kicked things up a notch with the DMT8, the first digital eight‑track hard‑disk recorder/mixer, which brought us innovations like basic cut and paste editing and instant locate functionality. 16: The click track The single project responsible for the greatest number of innovations in the history of recorded music is surely Disney’s Fantasia. Released in 1940, this animated masterpiece is credited with the first use of surround sound, the first simultaneous multitrack recording, the invention of the pan pot, and possibly the first use of overdubbing. It also pioneered an all-analogue form of mix automation. Leopold Stokowski’s orchestra was tracked using more than 30 mics at once, with eight optical recorders capturing the audio and a ninth yet another first: a track containing regular pulses that could later be used to time overdubs and synchronise the animators’ work. Needless to say, it would be many years before the click track began to shape the sound of rock and pop music. 15: The MIDI+Audio DAW Whilst Digidesign’s 1989 Pro Tools software is often credited as the ‘first computer DAW’ in that it offered four-track digital audio recording/editing running on a Mac, it was Studio Vision from Opcode Systems, launched in the following year, that was the first program to offer integrated audio and MIDI recording to a common timeline, pioneering the model for all today’s software DAWs. Being first doesn’t always get you the win, however, and the limited power of the computers of the day, as well as limited hard‑drive capacity and expensive RAM, meant that Studio Vision failed to expand its

user base beyond the professional arena. A subsequent, ill-fated acquisition by the Gibson guitar company led to Opcode’s rapid demise, and they never capitalised on their pioneering work. A number of the key developers at Opcode went on to work with Digidesign/Avid (Pro Tools), Apple (Logic, and Core Audio), MOTU (Digital Performer) and other prominent DAW companies. 14: Artificial reverb As Jason O’Bryan explained in last September’s SOS, the use of reverberation as an effect has its origins outside the music industry. Large radio broadcasters had their own echo chambers as far back as the early ’20s, at a time when records were made entirely acoustically. Music studios followed suit in the 1940s, and the Harmonicats’ ‘Peg O’ My Heart’, recorded by Bill Putnam Sr and released in 1947, was hailed for its innovative sound. As is so often the case, however, Hollywood was years ahead. The first commercially released movie soundtrack album, Songs From Disney’s Snow White And The Seven Dwarfs, was released in January 1938, and makes clever use of artificial reverb on ‘I’m Wishing’.

12: The drum machine Musicians have been plotting to get rid of drummers for a surprisingly long time. Léon Theremin, inventor of the instrument that bears his name, produced a device called the Rhythmicon way back in 1932. From the ’50s onwards, drum machines that could play back a selection of preset rhythms became common add-ons to home organs, and by the late ’60s some pop artists were experimenting with them on record. However, it was the advent of the microprocessor-controlled, programmable drum machine towards the end of the ’70s that moved synthesized rhythms out of the realms of novelty. The Roland CR-78 and TR-808, which generated their drum sounds using analogue circuitry, vied with sample-based units like the Linn LM-1 and Oberheim DMX; and music was the winner.

13: Remixing The dancehalls of Jamaica and the nightclubs of New York provided record producers with very direct feedback. A three-minute single might be over too soon, or might contain one short section that drove the crowd wild. Producers like King Tubby and Lee Scratch Perry began creating alternative interpretations of the reggae records they were making, de-emphasising the vocals and incorporating varied effects. Soon, dub mixing became an artform in its own right. Then, in ’70s New York, DJs began to work with tape, chopping up and looping segments of disco records. Artists and producers started to be involved in remixing their own work, with the 12-inch single becoming the format of choice. By the time house music took off, remixing was a specialist career for some, and in the hands of producers like Aphex Twin, would eventually reach the point where a ‘remix’ might be a completely different track in all but name.

11: Stage monitoring One of the reasons why the Beatles retired from live performance in 1966 was that sound reinforcement technology couldn’t keep up with their popularity. The PA systems of the time were no match for stadiums full of screaming teenagers. Over the next 10 years, live sound would undergo a complete revolution, from simple column speakers designed only to amplify the vocal microphone to complex arrays through which the entire band would be presented, sometimes in stereo or even quadraphonic sound. Perhaps the most revolutionary step in this process was the introduction of additional speakers on stage to direct these amplified signals back at the artists. Stage monitors first appeared around 1969 and were likely developed independently by several different FOH engineers, among them Bob Pridden with the Who.

The Roland CR-78 was the first programmable drum machine that could store and replay user-created rhythms — as long as you also had the super-rare WS-1 ‘write switch’.

www.soundonsound.com / February 2026

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AM/PM

LIVE

Phil Manzanera: Performing In Atmos When Phil Manzanera and Andy Mackay recorded an instrumental improv album in 2023, they had no intention of performing it live — let alone in a cinema, through an immersive Dolby Atmos sound system... J O E M AT E R A

n late 1971, nascent art-rock group Roxy Music placed an ad in Melody Maker seeking “the perfect guitarist for avant-rock group: original, creative, adaptable, melodic, fast, slow, elegant, witty, scary, stable, tricky. Quality musicians only.” When a young Phil Manzanera applied, he fitted the description so well that, half a century later, the same qualities still drive his creative endeavours. “What we’re doing as AM/PM really does describe the kind of music that Roxy was trying to do,” affirms Manzanera today. “It goes back to that original mission statement for Roxy, which was to do interesting music and perform it in a visually entertaining way.” AM/PM is the instrumental and improvisational outfit Manzanera formed with Roxy Music saxophonist Andy Mackay, the first fruit of which was a self-titled album, issued in 2023. March 2024 saw AM/PM, along with another Roxy stalwart, drummer Paul Thompson, perform three sold-out shows at All Is

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Joy, a venue located in the old Warner Brothers’ De Lane Lea building in Soho, London. “How we ended up doing those shows all happened by chance,” explains Manzanera. “The AM/PM album was mainly done by a series of improvisations, which I did in my studio, and then sent to Andy asking him if he could put something on top of it. I had no thought of ever playing it live. I thought there’s no way I could learn what I did because it was so random, it would take me forever to learn, so there was no chance of doing any live performances. Then Andy suggested doing these gigs, and I thought that it would be a terrible idea, because how the hell could we do it because a lot of it was meant to be sort of improvised and just on the spur of the moment. I was concerned it was going to be a disaster, but Andy insisted.”

Just Like In The Movies The venue’s intimacy and sound system allowed the pair to perform concerts that were totally immersive in every sense of the word. “So, we decided to

do it in a very small space, a 75-seater screening theatre in Soho, which had a 7.1 Dolby Atmos speaker system, and a huge screen,” says Manzanera. “I took advantage of that, making special videos that were going to be projected onto us. And then the whole sort of thing started developing into what Andy used to do at the end of the ’60s in the arts labs with Brian Eno and stuff before he joined Roxy, and before they became Roxy. So, it was a bit like going back to that time period, but with modern tech.” The performances were also to be recorded, and later mixed in Dolby Atmos. But with the original AM/PM album very much an improvised affair, showcasing the interaction between Manzanera and Mackay, one of the early challenges the project faced was finding a strategy that would keep true to the album’s improvisational spirit and inherent electronic elements without becoming a static backing track. “To solve this, we made small loops of elements from the multitracks that were then loaded into Ableton, using an Akai APC Mini,” explains Mike Boddy,

AM/PM performing live at the All Is Joy screening theatre in Soho, London.

AM/PM engineer, mixer and co-producer. “I was then able to trigger loops live and create evolving electronic arrangements for the show. Rehearsals started at Phil’s Gallery Studio with Phil, Andy, Lucas, [vocalist] Sonia Bernardo and myself ironing out technical issues and getting a feel for the set. Paul Thompson didn’t join any of the rehearsals until the day before the first live show, due to him living over three hours away in Newcastle, so realistically he probably had only two hours of rehearsals before having to perform. This really highlighted what an incredibly talented drummer he is. Despite the lack of rehearsing and having to play to a click sent to him from the Ableton rig, along with having improvised electronic elements going on around him and no structure to follow, he never missed a beat.” Any initial concerns Manzanera may have had quickly dissipated upon completion of the first song on the inaugural night’s performance. “After we played the first number, we looked to each other and went, ‘This is pretty good!’, exclaims Manzanera. “I then told

the audience I had no idea what we were doing, but hope that it was going to work. And it seemed to work, and they seemed to enjoy it. Luckily, we recorded it because the thing with live performances, especially if you’re improvising material, is that it could all go horribly wrong as it depends on the moment, the atmosphere, and the audience. “The shows were a very collaborative effort with Mike Boddy as part of the band up at the mixing desk with his Ableton, and moving stuff around the immersive speaker systems, and [guitar tech/guitarist] Lucas Polo fulfilling the sort of role Eno used to do for me in Roxy Music, by treating my guitar using my pedalboard. So, the combination of that, along with Mike treating it again, and sending it around, influenced the way I played and the way I left spaces, in turn leading to Andy and I being able to have a musical conversation with sound.”

On Stage Boddy was tasked with engineering the sound for the live performances and recording each night’s set. “As the

original album had a very electronic nature to it, it was helpful to keep the stage quieter,” explains Boddy. “Paul used a Roland V‑Drum setup for the kick, snare and toms but had a real hi-hat and some real cymbals. Keeping the stage quieter massively helped with Sonia’s vocals for the recordings and meant we could get away with a Shure SM7 on the sax. We had AKG C414s on the drum overheads, AKG C451 on the hat. Phil had his amp in a box, also, helping keep the stage nice and quiet; inside the box was an SM57. For the recordings the audience were captured with a pair of KM184s and a pair of Audio‑Technica AT4051s. “Rehearsals were in stereo, but as the venue had a d&b Soundscape Dolby Atmos system, along with a cinema screen, the plan was always to try and make the experience as immersive as possible for the audience. Phil made videos/visual treatments for each song, which for the shows were projected onto the stage, musicians and screen. For the shows themselves, these were loaded into [AV playback software] QLab and triggered via a Stream Deck from there. www.soundonsound.com / February 2026

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A M / PM LI V E IN ATMOS

The Atmos Pedalboard Guitar tech Lucas Polo discusses Phil Manzanera’s guitar effects rig: “The pedalboard itself was originally created for Phil for the Roxy Music 50th anniversary reunion tour. Because they hadn’t gigged together for over 10 years, a new rig was needed. As his guitar tech, it was my role to design something that was suitable for the tour. That took a lot of investigating, and trial and error. Thankfully, I had lots of help from Javier Weyler and Jake Elsegood from 5dB, and also from Daniel Steinhardt at GigRig, who were integral in putting it all together. “On the Roxy tour, I was on the wings of the stage as his guitar tech, and because the setup was fairly complicated, the stage pedalboard’s GigRig unit was connected via MIDI to another GigRig controller, placed in a tray on the guitar rack. So not only I was making the guitar changes, I was also changing the pedals from a distance so Phil could focus on playing. “Phil is a master of tone and texture, and his work with Roxy is so varied. We knew lots of different tones would have to be accessed very quickly, but didn’t want to sacrifice the quality found in analogue circuitry. That’s how we arrived at the idea of having a MIDI switcher operated remotely, masterfully executed by Dan and the GigRig team. When it got to AM/PM shows, the setup had to shrink a little bit because of space. With Roxy, they were in big arenas, so we had a lot of room, but we just didn’t have that luxury at AM/PM live as it was an intimate venue. So, we decided that instead of me controlling the pedalboard remotely, I would just do it on stage. “Instead of the traditional thing where the pedalboard is in front of the guitarist, the pedalboard was in front of me on a table at a height where I could use my hands to treat the sound in real time. Originally when I designed the pedalboard, they gave me access to the 1970s multitrack recordings from Roxy Music so I could solo all of Phil’s guitars and all of Eno’s treatments and figure out a way to do it in the 21st Century with current technology. “When we did AM/PM, it was also a tip of the hat to Eno, who used to be on stage treating Phil’s guitars live. Back in the ’70s, Eno would use his VCS3 synth and a Revox A77 tape machine to achieve all the crazy effects you hear on Phil’s guitar [as seen on the ‘Ladytron’ performance on The Old Grey Whistle Test]. To replicate that, I used a combination of the DigiTech Whammy,

Eventide H9 and Strymon Timeline pedals, as well as a Jam Pedals Wahcko wah‑wah as a filter, always tweaking the pitch and delay time with my hands to mimic the behaviour of tape and an analogue synth. “When making the original recordings of the AM/PM album, they quickly realised it was very immersive and quite an expansive sound. When they decided to perform it live, Andy’s idea was to stage it in a cinema because it had surround sound. And the Atmos mix was a way to replicate that live experience, as the original AM/PM album was only mixed to stereo. “As the live performances were never going to be the same each night, given that a lot of it was improvised, it was one way of letting those sounds travel around the venue, because the cinema had surround speakers. Mike [Boddy] was at front of house and he had matrices on the console which allowed him to pan all the effects across the room to envelop the audience. So Andy and Phil would have this musical interplay, this conversation, and I would treat Phil’s guitar in real time, and then Mike would fly it across the room to sort of envelop the audience. You had this thing where not only was I reacting to what Phil was playing, but Mike too, as he also had a little pedalboard at front of house that he used for Andy’s sax and oboe sounds. And those two sounds would then be flown across the room. And we recorded the three nights, and later it was a very happy coincidence that Phil met up with Guy Fletcher from AirSound as they were getting ready to mix the album, and they were like, ‘Let’s do it in Dolby Atmos!’ so the listener at home can recreate the experience of having been at the gig. “I think it’s worth mentioning the fact too that we had an almost silent stage, meaning we had electronic drums, both my guitar and bass were DI’ed, and Phil’s guitar was going from the guitar to the pedalboard and then to a head placed on stage behind me and on top of a Grossman Fat Box isolation cabinet fitted with a Celestion G12M speaker, and miked with an SM57. It meant that we could have the guitar really loud and yet completely silent on stage. And it sounded so clean at front of house because the microphone is inside the box, so the microphone is unaffected by the sound on stage. It wasn’t getting all the noise of the drums, the bass, the rumble. So, when I was using those effects, the effects were only affecting Phil’s guitar, and it was a very clean sound, which allowed us to make a much cleaner mix.”

Guitarist and guitar tech Lucas Polo designed Phil’s pedalboard, and operated it from on stage.

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NEXT MONTH IN Better Vocals In The Home Studio

Photo: Brett Sayles

The vocal is the most important element in any track — yet so many tracks are let down by weak vocal recordings. From acoustic treatment to noise reduction to virtual vocalists, we survey the options that help you get professional results outside professional studios.

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SOUND ON SOUND ETHAN WINER

’ve been a professional audio engineer and musician since the early 1970s, and I’ve read pretty much every recording magazine ever published. Back then the main magazines for recording engineers and amateur enthusiasts were Recording Engineer/ Producer and db. Both of those magazines featured technical articles to explain theory and explore best practices. Of course, new product announcements and product reviews have always been a staple too, but the content was often technical, and always accurate. Walt Jung contributed many articles to db with schematics for DIY projects, and I wrote similarly technical articles for R E/P. Audio magic and ‘new physics’ were noticeably absent in those days, and most product advertisements included full details if not complete performance specs. In the 1970s, if you wanted a professional quality vocal microphone you had to spend a fair amount of money for a Neumann U87 or AKG C414. I remember paying $7000 for an EMT 140 plate reverb because the only alternative was the crappy spring units found in guitar amplifiers. Magazines back then catered to recording professionals who had both the funds to

buy the necessary gear, and the technical knowledge to align analogue tape machine transports and their electronics. But then things started to change. In 1972 TEAC introduced a four-channel tape recorder at a price that was (just barely) affordable for consumers. Now aspiring home recordists could record and bounce basic tracks, and add overdubs, just like the pros using Ampex and Scully machines. When Tascam came out with their cassette-based Portastudio in 1979 the price of admission dropped even further, though so did the fidelity! But still, with care and practice, decent-quality recordings were possible. In 1992 the Alesis ADAT solved the quality problem, then digital recording finally took hold with the release of Windows 95. Even with those early 16-bit soundcards, the fidelity of the recording medium was no longer the limiting factor. As more and more amateurs eagerly bought affordable recording equipment, a new generation of recording magazines sprung up to serve this fast-growing new audience. But this audience was less sophisticated, basically self-taught by what they read. Worse, the writers and editors of this new generation of home recording magazine were just as ignorant as their

“It’s the only magazine I trust to give me useful facts...” 146

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readers. Phase and polarity were (and still are) regularly confused, and magical solutions to muddy mixes were routinely suggested via analogue summing boxes, ultra-high sample rates, and of course famous revered tube microphones and equalisers. Rarely was the obvious problem of typical bedroom acoustics acknowledged. Further, equipment reviews became increasingly dumbed down. A loudspeaker’s woofer size might be stated, but not its low‑frequency cutoff, which obviously is what really matters. Worse, equipment reviews started to read more like ad copy than critical assessments, with wording such as “incredible smoothness” and “colour that goes on for days”. But — and you knew I’d get here eventually — one magazine has stood out among all the others. Only one magazine has a technical editor with the depth of knowledge we see in Hugh Robjohns. Hugh’s reviews, articles, and answers to reader questions are always factual, accurate, and complete. Only one magazine has a loudspeaker expert as capable as Phil Ward. Phil’s detailed reviews include in-depth explanations about speaker driver and enclosure theory, crossovers, and more. The Leader section in every issue, previously by Paul White and now Sam Inglis, invites readers to consider varied and interesting aspects

of recording and music production. For many years Hugh and Paul visited studio owners in person to help them with their acoustics — a hugely important part of home recording that other recording magazines often ignore in favour of yet more gear praise. Every Sound On Sound product review lists both pros and cons, and the past articles comparing mic preamps and external clocks set new standards for audio journal honesty: advertisers be damned. In a recent product review, Matt Houghton shared more generally useful information about using compression than the dedicated ‘tutorials’ I see in other magazines. Then a fabulous article by Neil Rogers explored techniques for quickly getting a good sound from toms. John Walden’s review of the Komplete 15 soft synth was thorough and highly informative, much more so than we typically see in the US magazines. Likewise for Mark Wherry’s in-depth review of Cubase. And please don’t let me forget to mention Dave Stewart’s ongoing series of sample library reviews that are always informative, way beyond explaining the features in the current product du jour. I won’t mention the other great writers for this fine publication, but know that they are all appreciated. Long live Sound On Sound magazine! It’s the only magazine I trust to give me useful facts — real facts — instead of bullshit.

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SOUND ON SOUND [USA] February 2026