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Jon Bon Jovi

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Hybrid drumming

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Trinnov Nova

Taking room correction to the next level

ASIO goes native! A giant leap for Windows audio?

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LE ADER

A RIGHTS MESS T

he trouble with writing about AI is twofold. First, the situation changes so fast that the printed word risks being out of date by the time it appears. And second, it’s hard to be positive about a technology that is displacing human creativity whether creative humans like it or not. But there was at least some satisfaction to be gained recently from hearing the tech bros of OpenAI complaining that Chinese chatbot DeepSeek may have infringed copyrights surrounding ChatGPT. Yes, copyright. That’s the same legal concept for which the same tech bros showed the same regard when training their models. There’s a Jeremy Clarkson meme for that. Less satisfying is that, in its wisdom, the UK government is considering creating an exemption that would allow AI companies to use copyright material as training data without explicitly seeking permission first. The idea seems to be that if we want to remain competitive in the AI sector, we need to give it the keys to the archive. Because if we don’t, the AI sector will simply migrate to another territory that cares even less about its creative sector. The government’s consultation document, which you can find at bit.ly/4hEmi5P, makes this argument pretty strongly, and in fairness, it’s hard to dispute that the only real protections that can be afforded to the creative sector are those that

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are also widely implemented elsewhere. So much for Brexit allowing us to take back control. If this consultation is to be believed, the best musicians (and magazines) can hope for is an opt-out clause: an exemption from the exemption, if you will, that would allow us to prevent our work being used for model training unless we grant an explicit licence. This would be accompanied by new rules about transparency, forcing AI companies to disclose what material they’ve used for training, and ensuring that AI-generated content is clearly marked as such. If this can be made to work, it may even be a reasonable solution. But therein lies the rub. Does anyone seriously believe that it can be made to work? How, exactly, are UK authorities planning to audit transparency disclosures made by Google or DeepSeek? How will an exemption that can’t be applied retrospectively help those whose catalogues have already been plundered? How will we block AI tools that don’t comply with the new legislation? How can technical and legal mechanisms be established that are both effective and accessible to individual artists? And is there any chance that this hugely complex endeavour can be completed before irreversible damage is done to the creative industries? Like I say, it’s hard to be positive sometimes.

“It’s hard to be positive about a technology that is displacing human creativity whether creative humans like it or not.”

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ISSN 1473‑5326 A Member of the SOS Publications Group The contents of this publication are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the Publisher. Great care is taken to ensure accuracy in the preparation of this publication but neither Sound On Sound Limited nor the Editor can be held responsible for its contents. The views expressed are those of the contributors and not necessarily those of the Publisher or Editor. The Publisher accepts no responsibility for the return of unsolicited manuscripts, photographs, or artwork. © Copyright 2025 Sound On Sound Limited. Incorporating Music Software magazine, Recording Musician magazine, Sound On Stage magazine, SPL magazine, Sound Pro magazine and Performing Musician magazine. All rights reserved. All prices include VAT unless otherwise stated. SOS recognises all trademarks.

www.soundonsound.com / March 2025

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128 INSIDE TRACK

IN THIS ISSUE

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March 2025 / issue 5 / volume 40

FEATURES 74 Modular Unicorn Ship Explosion’s Sasha Lewis on obsession and the ever‑changing nature of modular.

96 Hybrid Drumming Reinvigorate your rhythms by integrating electronics into an acoustic drum kit.

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100 How To Finish A Mix There comes a point when tinkering with your mixes won’t improve them — it could even damage them. But how do you know when it’s time to let go?

114 Talkback

O V

Tobin Jones is a mixer and engineer who places great importance on how a studio feels and has a strong disregard for reference tracks.

PAGE 84

116 Jon Bon Jovi & John Shanks Jon Bon Jovi has stayed ahead of the pack for more than four decades, with the help of long-term collaborators such as producer and engineer John Shanks.

122 ASIO Goes Native Windows-based musicians have long been asking for better support at the operating system level — and native low-latency ASIO drivers and full support for MIDI 2.0 are finally on their way. We reached out to Microsoft to find out more...

136 How I Got That Sound Roxy Music’s Phil Manzanera explains why his favourite guitar sound can never be repeated.

142 Q&A Your studio and recording questions answered.

128 Inside Track: Ian Fitchuk

146 Why I Love... Teaching Mixing

Working with Kacey Musgraves catapulted Ian Fitchuk from respected producer to one of Nashville’s busiest songwriters.

Can art be taught? SOS reader Gavin Herlihy searches for the answer...

68 TORSO ELECTRONICS S-4

ON TEST 32

Arturia Synthx V

140

Black Lion Revolution 14x16 USB Audio Interface

Schema: Light 76

DJ Swivel HitStrip

Elektron Digitone II

56 58

HLabs CS169 Analogue Channel Strip

iZotope Cascadia Ducked Tape Delay Plug-in

KRK Rokit Generation Five Active Monitors

ME‑Geithain MO‑1 MKII Active Monitors

Morphor Echo Eurorack Module

Patching Panda Blast & Hatz V3 PSPaudioware Wobbler

Speaker & Room Optimisation System

Dynamic Microphones

Shure MV7i Slate Digital Submerge Softube Console 1 Fader MkIII DAW Controller

138

52 Universal Audio SD‑3,

Relab VSR REV6000

Automatic Ducking & Filter Plug-in

Trinnov Nova

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Modulation Plug-in

Stereo Reverb Plug-in

FM Synthesizer & Sequencer

Eurorack Modules

Channel-strip Plug-in

Origin Effects Deluxe55

Therevox ET-5 Analogue Synthesizer

Analogue Amp Emulation Pedal

Eurorack Module

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72 DivKid & Making Sound

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28 Native Instruments

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140 Strezov Sampling

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Software Synthesizer

Native Instruments Claire

Soniccouture Tonal Drums Sample Library

26 Steinberg Cubasis 3.7

& Iconica Sketch Mobile Music Creation System For iOS & Android

Waves Space Rider Dynamic Multi-effects Plug-in

Wolff Audio FREQ 500-series Equaliser

Xvive PX System Personal Monitor Mixer

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WORKSHOPS 106 110 112

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ON TE ST

KRK’s Rokit series just keeps getting better! PAUL WHITE

ith their distinctive yellow woofer cones, KRK’s Rokit monitors are a familiar sight in studios everywhere. The latest Generation Five models have seen some significant upgrades over their predecessors, but remain affordable for the project studio owner. The new series comprises three models based around five‑, seven‑ and eight-inch woofers; for this review we’re looking at the smallest and largest speakers in the range. All three models have common features such as new MDF cabinets with moulded front baffles and rounded cabinet sides, removable protection grilles for both the woofer and tweeter, and an enhanced set of DSP functions accessed from a small screen on the rear panel to aid setup. They feature magnetically fixed protective grille faceplates that can be replaced by alternative, included non-protective faceplates if preferred, and KRK assure us that there’s no sonic difference with or without the grilles. A flared slot-shaped reflex port occupies the bottom of the baffle, and the driver topography provides a slightly wider horizontal dispersion than in the vertical plain.

Hearing Voices The DSP section offers three voicing modes: Mix mode, with a flat frequency and phase response; Create mode, which gives a more flattering sound for writing or producing; and Focus mode, which applies a midrange lift that KRK say can be useful for homing in on vocals or instruments such as electric guitar. My personal choice is always to use speakers in their flat mode, though for those used to the midrange focus of something like Auratones, the Focus mode may be useful. There are also 25 boundary/EQ permutations to help optimise performance for the acoustic environment in which the speakers are to be used. The Rokit 5, as its name implies, is equipped with a five‑inch woofer utilising KRK’s signature woven Kevlar aramid‑fibre cone material, presented in the familiar KRK yellow. This is teamed with a one‑inch, soft-dome silk tweeter, crossing over at 2.27kHz and capable of reproducing frequencies of up to 40kHz. The overall frequency response

March 2025 / www.soundonsound.com

KRK Rokit Generation Five Active Monitors

is specified as 54Hz‑30kHz (±3dB), or 43Hz‑40kHz at the -10dB points. As the speakers are all two-way, bi-amped active systems, separate Class‑D amplifiers (20W HF and 35W LF) are used to power the woofer and tweeter. A maximum SPL of 104dB at one metre is quoted. To aid performance when mounted on stands or shelves, there’s an acoustic foam isolation pad at the base of the cabinet, plus a separate foam wedge for angling the speakers up when they’re placed on a desktop, and there are also mounting points for use with optional KRK brackets should you need a wall-mounting solution. Each cabinet measures 288 x 191 x 242mm and weighs 4.95kg. For the new Rokit 8, the cabinet is scaled up to match the eight‑inch woofer,

which crosses over to the one-inch tweeter at 1.64kHz. The cabinet measures 400 x 266 x 319mm, and weighs more than twice as much as the Rokit 5, at 10.45kg per speaker. As you’d expect, the frequency response shows more low-end extension (42Hz-36kHz at the ±3dB points), and the maximum SPL is also higher, at 111dB at one metre. There’s a fair bit more amplifier power in hand, too, with 68W for the highs and and 135W for the lows. Given all the functionality on offer, the rear panels of the Gen Five monitors are surprisingly uncluttered, with just a power inlet and switch, the XLR/jack combi input, a monochrome LCD screen and a press/turn data-entry knob. To extend screen life, the LCD turns off after five minutes, but springs

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ON TE ST

K R K ROKIT GE N E R ATION F I V E

back to life when you press the to reproduce, because they data knob. occupy the same frequency as When the speakers are first the crossover point, but again powered up, the Home screen there were no problems in this is displayed, showing options area with the Rokit 5s. Given for Level, Voicing Mode and EQ their affordability, the overall Mode. Pressing the data knob performance is impressive, steps through different functions and while a monitor of this while turning the knob selects price can’t be expected to rival options or changes values. expensive, esoteric speakers The display is able to show the for accuracy, these will tell you current EQ curve, and icons are what you need to hear in order used where possible to simplify to create a reliable mix. navigation. In addition to Moving on to the Rockit selecting the voicing mode, there 8s, as expected there’s more are also multiple settings for the headroom and more low-end low and high EQ. extension, making them The rear‑panel adjustments well suited to larger studio are easy enough to make and spaces. The lower crossover there are plenty of options frequency doesn’t seem to to tailor the response to your change the character of the acoustic environment, but as sound in any obvious way, you have to be able to see the either — both the 5s and 8s display to make changes, you have a coherent family sound. have to turn the speakers around, For both models, the sound make the adjustment, then put seemed to be unaffected by the them back again to check if your grilles, so I left these in place changes make things better throughout the review. The voicing modes are selected using the LCD and jog dial on the rear panel, or not. It would not have been In all, then, I’d be happy with options comprising Mix (flat response), Create (LF and HF boost) practical to put the display and to trust mixes made on these and Focus (mid boost) modes. data knob on the front, so this is speakers, with the usual caveat something we have to live with, but in my that it is safest to also check mixes on that would suit long listening sessions case I got the best result with the speakers headphones, just to remove any room without incurring fatigue. That silk-domed set flat and left in Mix mode. If you need issues from the equation. The balance tweeter certainly seemed to be doing to keep switching between Mix and Focus of solid bass, smooth but extended highs a fine job on the high end, while down modes, then accessing the controls could and a capable midrange ticks all the right below the bass was full and solid but still be frustrating in a small studio with no boxes for speakers in this price bracket. tight enough to give good definition to access space behind the speakers. For small rooms I’d recommend dry kick drums. Inevitably there’s a little sticking with the Rokit 5s, as getting an overhang, which is part of the trade-off Rokit Man even bass response in a small space of ported cabinets (the up side being is next to impossible, but if your studio increased LF extension), but not enough I started with the Rokit 5s, which I set up space is the size of a single garage or to compromise performance. on my IsoAcoustics speaker platforms. greater, the Rokit 8s would be a good fit. Midrange sources such as vocals The speakers sounded detailed but Despite being almost twice the price can prove tricky for two-way monitors smooth-sounding, turning in a performance of the Rokit 5s, the Rokit 8s are still very affordable and still turn in an The Rokits are available in three sizes, based on impressive performance:price ratio. In five-, seven and eight-inch woofers. summary, then, the Gen Five models are a worthwhile upgrade on the previous Rokits and they fall into the pricing sweet spot for most home studio operators.

summary As ever, KRK’s Rokits offer impressive value for money, with these new Generation Five versions being packed with more technology than ever.

$ Rokit 5 $398, Rokit 7 $538, Rokit 8 $638. Prices are per pair.

W www.krkmusic.com

March 2025 / www.soundonsound.com

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Sweetwater.com (800) 222-4700

ON TE ST

SAM INGLIS

hen podcasting became big business, Shure were as surprised as anyone to find that their SM7B dynamic mic was suddenly the hottest ticket in town. Even now, it remains hugely popular, and was recently joined by the active SM7dB variant, but Shure have also developed a more affordable range of dynamic mics known as the MV series. And whereas the SM7 mics require an XLR cable and audio interface to cast pods, the MV range includes USB mics alongside the more traditional MV7X. The latest in the MV series features both a USB socket and an XLR connector, but the XLR does not carry the mic’s output signal. Instead, it’s a combi XLR/ jack socket that allows an additional source to be connected. In other words, the MV7i is not only a USB microphone, but an audio interface as well.

Motive Force The MV7i is compatible with macOS, Windows, iOS and Android. Whichever platform you use, you’ll want to install Shure’s free MOTIV Mix (desktop) or MOTIV Audio (mobile) app, and not only because it’s the only way to configure the MV7i’s various built-in DSP options: MOTIV Mix also allows you to aggregate multiple audio sources. You can connect additional MV microphones, in the same way as is possible with Apogee’s Hype Mic, and MOTIV Mix can also accept audio from the built-in mic on your computer, and from other software. In fact, you don’t need to own any Shure devices in order to use MOTIV Mix, and the icing on the cake is that it can make basic recordings, either of all the sources feeding its mixer simultaneously, or of its stereo mix. Out of the box, the MV7i resembles a stubbier SM7, and feels similarly robust, but its foam windshield conceals a much more conventional-looking wire mesh headbasket. At the other end, you’ll find a USB-C socket, the aforementioned combi connector, and a mini-jack headphone output. A multi-colour LED strip halfway along the body provides visual feedback, including slightly vague but funky signal metering, and can be tapped with a finger to mute the mic, whereupon it turns red.

March 2025 / www.soundonsound.com

Shure MV7i USB Microphone & Audio Interface

Shure’s latest USB mic features a built-in audio interface. As you’d expect, the combi socket can be used to connect another microphone, with a 60dB gain range and optional 48V phantom power available, or an instrument such as an electric guitar; but it can also be configured as an unbalanced stereo input for synths, backing tracks and so on. All of this configuration is handled within MOTIV Mix, and there are plenty of hidden depths to explore.

Multiple Mixers The main MOTIV Mix window displays a list of available Sources down the left and a small pane of controls relating to recording along the bottom, with the largest area devoted to the actual mixer. This has stereo output channels for two different devices, labelled Main and Monitor in green, and as many input channels as you choose to drag in devices from the Sources list (which can be hidden once you’ve got everything set up). Every channel in the mixer has a fader plus two large buttons labelled Mute and Monitor. These control whether the source is audible at the Main and Monitor outputs,

respectively: by default, sources are heard at the Main output unless muted, and not heard at the Monitor output until their Monitor button is activated. I was initially puzzled to find that muting the MV7i’s own channels in MOTIV Mix, and even pulling the master fader down, didn’t seem to prevent me from hearing my voice at the headphone output. It turns out that the faders and mute controls within the main MOTIV Mix panel are completely independent of the microphone’s internal low-latency monitor mixer. So if you select the MV7i as the Main output, it’s easy to end up in a situation where the input signal is routed to your headphones both within the mic and within MOTIV Mix. The MV7i’s own mixer and other DSP features are accessed by clicking the cogwheel icon to bring up a pop-up with three tabs labelled Global, Mic and Input. The first of these is where you’ll find the Monitor Mix sliders that set the direct monitoring balance. It’s also where you switch the MV7i between three Audio Output modes. In Mixdown mode, the mic signal is mixed with a mono signal coming from the combi input, in

ON TE ST

HLabs CS169 Analogue Channel Strip

Having studied the sturdy Studer 169 console, French company HLabs decided its classy combination of mic pre and EQ deserved a place in modern studios. M AT T H O U G H TO N

n the world of pro audio hardware, the classic analogue circuits of yesteryear continue to hold appeal, and it seems barely a month goes by where one recreation or another isn’t announced. But while countless products inspired by revered classics have made their way to me for review, I don’t think that I’d ever seen one based on the Studer 169 until HLabs’ latest creation arrive in my studio toward the end of last year. Having used it for a couple of months now, I have to say that I have no idea why that should be the case — because this CS169 is a very appealing and classy sounding device, and one that offers a different character from the various Neve- and API-inspired designs that seem to have dominated this market in recent years.

Overview The CS169 is based on the preamp and EQ section of the 169 console, made by the now-legendary Swiss manufacturers Studer. First made in 1976, the 169 mixer was more compact and portable than the large-format x89 consoles that had come before it, and it was an incredibly robust design. Reportedly, several thousand units were sold, and it saw use right around the world for location, broadcast and post-production sound. In music production circles it soon gained a strong reputation for its open and detailed sound, and smooth, ‘musical’ EQ section, and that meant it found a home in many a small-but-serious studio.

March 2025 / www.soundonsound.com

HLabs’ interpretation of the 169 input channel is housed in a 1U rackmount chassis, with a white-painted front panel providing the backdrop to dark-blue, black and yellow legends and controls. On the rear, you’ll find four XLR sockets, with an input and output each for the preamp and EQ sections, and both those outputs have their own ground-lift button. A short XLR patch cable joins the two sections but can be removed should you wish to access them separately or insert another device between them. The only other feature on the rear is the ubiquitous IEC C14 mains power inlet that accepts 10-240 Volts and 50/60 Hz AC. On the left of the front panel, you’ll find controls for the transformer-balanced preamp section; the input transformer is a Neutrik NTM-4 and the output one a Carnhill VT24499. These controls comprise a large-ish continuous gain control that’s marked with two gain ranges, switched using an adjacent Hi button. The ranges are given as +18 to +42 dB, and +40 to +65 dB, respectively, though HLabs say there’s actually a touch over 67dB available. There’s also a smaller output attenuator knob (-30 to 0 dB) that allows you choose whether you want a cleaner sound (low gain, high output) or to drive the preamp hard in search of transformer coloration, and then back off the levels feeding the EQ or next device in the signal chain (higher gain, lower output). Next up we have buttons for the usual +48V phantom power, polarity inversion and a 33dB pad (making the total gain range a whopping 82dB), before a second-order

(12dB/oct) high-pass filter with an in/ bypass button. Since this filter can be set anywhere from 20 to 315 Hz it could be just as useful for taming proximity effect or pulling some low mids out of the way of another instrument as it can be for filtering out bumps and rumbles. All in all, then, there’s ample gain and both functional and sonic versatility here, whatever mic you’re working with. To the right of all these controls is a high-impedance (>1MΩ) TS jack input, for electric guitars, basses and the like, which feeds the same input transformer. A simple circuit, but one that should make most players happy, given those transformers. Speaking of impedances, the mic preamp input impedance is given as 1.2kΩ (not vast but ample), and the mic preamp output impedance is under 20Ω. In the middle of the front panel is the three-band EQ section, which comprises four knobs and an on/bypass button. Behind the faceplate, this is treated as an entirely separate device and the EQ has a transformerless input and output — having developed the 500-series versions of each circuit, they’ve effectively put these PCBs in the same box with a dedicated PSU and reworked the front panel. It works well, even if I can’t help wondering what it might sound like using the EQ to shape the sound being fed into that Carnhill transformer! The EQ’s outer two bands are shelving filters and the knob controls the gain (marked simply from -4 to +4) and, somewhat counterintuitively, the band on the left is the high shelf and that on the right is the low shelf — it’s an

The preamp and EQ sections each have their own balanced XLR I/O and, as shipped, the preamp’s output is routed to the EQ’s input through a short patch cable.

arrangement that would make sense if the strip were arranged vertically, as on a console, but it initially threw me, given the horizontal orientation. Again, I gather that this is a byproduct of using the same design as their 500-series versions, since this orientation of the boards makes sense internally. The mid band, Presence, has the same markings for the gain knob and is joined by a frequency selector pot that ranges continuously from 100Hz to 7kHz, which seems suitably generous for a general-purpose recording EQ. On the right, we have an output level meter and a large power toggle switch. The moving-coil meter is a horizontal type where the needle moves across the edge of a protruding 3D curve. It’s nicely illuminated, the needle is visible from above, and anything above +4dBu is denoted by a thick red stripe so it’s easy to see if you have a healthy level coming in. Cleverly, the backlighting also turns red just before clipping. I approve of front-panel power switches generally (who wants every device to be always on today?), and this particular switch is flanked by Studer‑esque protective cheeks.

with the result. There was plenty of gain on hand, and it accommodated sensitive mics used on loud snare hits without problem. I found the instrument input a pleasant bonus, again lending a subtle coloration to the signal that’s comparable with that of the better transformer DI boxes I have at my disposal — it usually increased the appeal of the instruments that I played through it, and the front‑panel access could increase its attraction in a project studio setting. On the face of it, the EQ section might not appear particularly remarkable in terms of functionality: there are no transformers, and there are just two shelves and a swept mid here (OK, if you’re using the preamp stage, there’s the high‑pass filter too, which can be used to shape the LF shelf to some extent), and there’s no control over the bandwidth or anything like

at the top end. Other than during my initial confusion over the placement of those low and high bands, I found it nigh impossible to screw things up. There are plenty of other EQs that can do a similar thing in terms of the frequency response, but it does sound like there’s a bit more going on here than that, even when used without the preamp. Enough so that I sometimes found myself deciding to run signals out of the DAW and through it while mixing. In that role, I was marginally frustrated at the lack of post-EQ output control, but that’s not really an issue when using the preamp as intended.

Verdict HLabs’ CS169 is a classy transformer-balanced recording channel, and could be a great option for anyone looking for a single, versatile, high-quality front end, whether for mics or instruments. At this price, it’s competing in a crowded marketplace, with plenty of appealing channel strips available, some with different sounds and others boasting more functionality. But the CS169 is beautifully constructed, offers some thoughtful touches, and while these things are subtle in the grand scheme of things, the tonality is different from that of the many other ‘vintage’ recreations available today. So I’d say it’s well worth auditioning — in fact, if I had to record everything through this strip for the rest of my music-making days, it certainly wouldn’t be the worst thing in the world!

“There was plenty of gain on hand, and it accommodated sensitive mics used on loud snare hits without problem.”

On Test I’ve recorded through a Studer 169 only once before but I recall being impressed by its (for a vintage console) open and detailed sound that also benefited from a pleasing ‘smoothness’, which I mentally attributed at the time to the transformers. That experience was many years ago now, so I can’t claim to have made a direct comparison, but in subjective terms the CS169’s mic preamp and EQ stages certainly chimed with those memories, and I’d use the same subjective terms to described it. I tried the usual sources — various dynamic and capacitor mics (including a ribbon) on vocals, acoustic guitar, miked electric guitar amp and percussion (a kick, a snare and a tambourine) and I was always very happy

that. But really, that’s precisely what you want in a recording channel — something that’s as easy in use as it is easy on the ear, with enough range in the gain controls (those markings on the gain knobs turn out to give you about ±15dB for the shelves and ±11dB for the Presence band) and the mid frequency to let you nudge most sources in whatever tonal direction you want, without the risk of causing carnage. I know it’s fashionable to do EQ’ing after you’ve recorded these days, but I much prefer to get things sounding as close to the way I want on the way in, and if you don’t have a vast mic collection or the time available to audition and reposition mics, a good EQ is not just a perfectly legitimate tool — it can be incredibly useful. The sound of this one is oh-so smooth. ‘Buttery’, if you like, yet for a vintage-inspired design it certainly sounds pretty clean and open, especially

summary Clean and detailed yet smooth, the CS169 packs the sound of Studer’s classic portable mixer into 1U of rack space.

$ $1349 T HLabs +33 (0)6 59 40 62 53 E contact@hlabs.audio W www.hlabs.audio www.soundonsound.com / March 2025

20 YEARS OF ACOUSTIC RESEARCH INTO ONE SOLUTION.

www.trinnov.com

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If controlling your own foldback mix appeals but you’d rather not invest in a digital setup, the PX System might be just what you need. MARK GORDON

t’s great that in-ear monitoring systems are no longer the preserve of professional touring bands — there are now many affordable options that suit the budgets and requirements of semi-pro and amateur musicians. The abundance of digital mixing consoles with Wi-Fi control and plentiful auxiliary outputs has also helped to bring the luxury of ‘hearing yourself on stage’ to many performers. Xvive are a relatively new company in this field, established in 2014 with a line of guitar effect pedals, moving into wireless technology in 2016 and now apparently holding several wireless patents. They manufacture in China, although their designers are based all over the world, including North America, Europe and Asia. Their manufacturing facilities make products for “several top American music gear brands”, as well as Xvive’s own designs, and are described as “highly respected”. Their entry into the wireless market in 2016 was the U2 Guitar System, which paved the way for a range of products focusing primarily on wireless systems and in-ear monitors. The latest addition to their catalogue, the PX System, moves away from wireless technology and uses Cat5 cabling to deliver three discrete audio channels from your mixer to on-stage performers, who can balance each element as required using a small belt-pack mixer. If your budget doesn’t stretch to wireless in-ear monitors and the idea of controlling your own mix from an iPad is too expensive, unappealing or downright complicated, the PX System might be what you’re looking for.

Linked In The PX System comprises three elements: the PX-A headphone mixer belt-pack, the PX-B three-channel snake, and a 25-foot, shielded Cat5 cable to connect the two units together. The PX-A is around an inch deep, an inch wide and four inches long, and looks very similar to the personal headphone amplifiers available from Behringer and SubZero. At one end is a small volume knob that also acts as an on/off switch, and below it is a standard

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Xvive PX System Personal Monitor Mixer 3.5mm stereo headphone socket. A small switch selects either battery power or POE (Power Over Ethernet, which I will explain in a moment), and there’s a battery level LED that glows blue when the unit is on and changes to red when power is low. Three volume knobs, clearly labelled Input 1, 2 and 3, are used to control the levels of the individual audio signals routed to the PX-A. A locking etherCON RJ45 connector allows connection to the PX-B. The PX-A’s battery compartment flips open to reveal space for two AAA batteries, plus three DIP switches that serve as pan controls and allow each of the connected audio signals to be sent to the left, right or centre of the headphone mix. Finally, a sturdy clip is provided to attach the PX-A to a belt. The PX-B is around half the length of the PX-A and, in addition to the etherCON connector at one end, features four 18-inch, braided, trailing cables. Three of the cables terminate in female XLR connectors, which connect to your audio sources, and the fourth can be used for a standard 9V DC PSU (not supplied), providing Power Over Ethernet. This makes it possible to power the PX-A without batteries. Both the PX-A and PX-B feature metal cases that feel well made and solid but not overly heavy or cumbersome, which is particularly helpful if you’re taking advantage of the belt-mounting option while playing.

As I mentioned, the included 25-foot cable uses etherCON locking connectors, which work in a similar way to an XLR and are very secure. Should you need to be further than 25 feet from your audio source, you can use any good-quality shielded Cat5 cable up to 200 feet in length.

Direct Connection The basic idea is that the PX-B’s three trailing XLR cables connect to your audio sources and the PX-A receives the signals over Cat5. The user can then balance each signal independently, using the three

Xvive PX System $129 pros • Extremely well made. • Easy to set up. • EtherCON locking connectors. • Up to 200-foot cable runs. • Flexible in-ear monitor mixing on a budget.

cons • Potentially could occupy three mixer aux sends for one performer. • The XLR adaptors may be too bulky for some mixers.

summary The PX System offers a great solution to anyone looking to control their in-ear mix without investing in complex hardware and software.

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XVIVE PX SYSTEM

volume controls to create their desired mix. On the surface, setting up the PX System couldn’t be simpler but, depending on what you are looking to achieve with your in-ear mix, there are several ways you can configure the system. Most musicians would configure a monitor mix using aux sends on their mixing console, and the PX System can fit in seamlessly with this. Your mixer will need to have at least three aux sends to make the best use of the PX System but, assuming you have those available, you simply connect the three trailing XLR cables to the individual aux outputs of your mixer. Many mixers have quarter-inch jack outputs rather than XLR connections, and Xvive have taken this into account by providing three balanced XLR to unbalanced mono jack adaptors. However, I found that all three of these adaptors together were too bulky to plug into all three aux sends on my Behringer XR2222 desk, as its sockets are located quite close together. This is certainly something to be aware of — you may need some additional cabling or adaptors to resolve the problem. Once the system is connected, the user can then go about the task of creating three independent mixes that will be sent to the inputs of the PX-A. You might, for example, configure aux 1 for vocals, aux 2 as bass and drums, and aux 3 as guitar and keys. How you configure each aux send will depend on what you want to control separately in your own mix. If your mixer doesn’t have the luxury of three aux sends, another option is to use the direct output of a channel or a subgroup

The complete system comprises the PX-A mixer unit, the PX-B breakout snake, and a 25-foot etherCON cable to connect the two.

output as one of your sources. Let’s say you wanted to create the same mix as above, but your mixer only had two aux sends. As with the previous configuration, you could use aux 2 to create a bass and drums mix, aux 3 for a guitar and keys mix, and connect the remaining XLR on the PX-B to the direct output of the vocal channel. Depending on how a given mixer is designed, it may be possible to use the insert point of a channel as a direct output, but you would need to be aware that this can interrupt the signal path to the main outputs on many mixers. If you’re a performer playing to backing tracks, you could take a stereo feed from a laptop or iPad into channels 1 and 2, with the DIP switches in the PX-A set to left and right, to give you true stereo in your headphones. Input 3 could be your vocal, your instrument or, if you’re a drummer, a feed from a click track. The PX System isn’t exclusively for live performance — you could also make it work well in a studio environment, taking

Expanding Universe The most obvious issue with the PX System is that one performer could monopolise up to three aux sends, potentially leaving no monitor outputs for anyone else in the band. This is where the PX-Hub comes in. Although I didn’t have a physical unit to review, it’s certainly worth mentioning, as it does allow you to easily expand the PX System to accommodate up to five additional musicians. This simple half-rack unit resembles an Ethernet hub and, in essence, that’s what it is. You connect the PX-B to your audio sources as normal, and then connect the Cat5

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cable to the input of the PX‑Hub. Each band member can then connect their own PX-A to one of the five available Cat5 outputs and create their own mix. Of course, each mix will be based upon the same three audio sources, but the balance between them is up to each individual band member. It’s a neat solution that saves a lot of cabling, as you only need one PX-B to supply audio and power to five PX-A units.

individual outputs from your audio interface and allowing performers to balance themselves against a backing-track mix.

Wired For Sound The PX System offers a great solution for performers wanting a degree of control over their personal in-ear mix without using Wi-Fi and iPads, which can be over-complicated and confusing for some musicians. As it’s hard‑wired, you achieve excellent quality audio and are not subject to the interference that is the bane of cheaper wireless IEM systems using the congested free frequency bands. Of course, the fact that it is hard‑wired does make it most ideally suited to more static musicians, such as drummers and keyboard players, but a potential 200foot cable length allows some leeway. The hardware is well made and very solid, and all the cables feel very high quality, especially the locking etherCON connections, which add a welcome level of security on stage. Having trailing XLR connectors also gives the system a high-end feel, but in practice I found the XLR-to-jack adaptors too bulky for my particular mixer. To suggest the PX System “lets you hear what you want to hear on stage, in the studio or in rehearsal”, as Xvive do, is perhaps overstating the case a little, as it is limited to three channels, but a bit of lateral thinking with DI boxes and signal splitters — and the way you configure the system — allows a good level of flexibility. To sum up, at around half the price of a reasonable wireless IEM setup, the PX System is easy to use, sounds great and gives you a degree of control over your mix that you don’t get with a stereo system alone. $ PX System $129.99, PX-Hub $79.99. E support@xvive.com W www.xvive.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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With pristine sound and four switchable bands, Paul Wolff’s 500-series EQ design strikes a balance between versatility and intuitive control. M AT T H O U G H TO N

here are lot of 500-series EQs, so a new one had better offer something a bit different from the crowd! That poses an awkward conundrum for the designer: given the space constraints, it’s not easy to add features to an EQ in this format without sacrificing something. But, in his FREQ, industry legend Paul Wolff has attempted to deliver a decent balance of sonics, features and practicality.

FREQ Scene Describes as a ‘four-band, proportional‑Q, parametric EQ’, the FREQ is a single-channel, single-width 500-series equaliser module. The output is balanced using a custom Cinemag nickel and steel output transformer, which Wolff says was selected for its transparency rather than for obvious coloration, and the amplifiers that drive this transformer are based around OPA2134 ICs, though reportedly the FREQ can be modified to use discrete op-amp modules if desired. What colour this thing does impart is largely third-harmonic distortion at the low end, which I generally perceived as a subtle enrichment of the bass. Higher up the spectrum, it really does sound as clean as a whistle, and there’s lots of headroom too, with clipping not occurring until +28dBu. Each of the four EQ bands can function as a proportional-Q bell type, with the bandwidth narrowing as you apply more boost/cut, making them suitable for gentle tone-shifts or for tackling resonances. The gain control offers ±15dB in all cases. The top (HF) and bottom (LF) bands’ frequency pots range from 2-20 kHz and 25-700 Hz, respectively. The HM (high mid) band ranges from 350Hz to 10kHz and the LM (low mid) from 75Hz to 2.2kHz. So there’s plenty of overlap of adjacent bands. The HF and LF bands each have a blue backlit button labelled Wide that, when lit, turns the band into a shelf. As with all the buttons on this module, they operate sealed relay switches, so should remain clean and crackle-free for many years. The two central bands also each have a a button to change their function. In this case, the

March 2025 / www.soundonsound.com

Wolff Audio FREQ 500-series Equaliser

backlighting is yellow, and the button fixes the bandwidth at 1/3 octave. Being narrower, this essentially makes it possible to perform much more targeted cuts, such as to dip out some harshness in a vocal without too much collateral damage. Between those two bands’ buttons is another, this time green-backlit. This is an on/bypass button for the whole EQ, and is a hard-wired bypass, but it’s also easy enough to bypass individual bands, since their gain knobs have detents at the unity gain position.

In Use I used the FREQ on a range of sources in a mix, and since I had only one module I created a mono bus to see how it fared on drums. It served very well in all cases, being just as versatile as you’d hope a console-style EQ would be. Yes, a true

parametric EQ would allow greater control but it would also leave you with more to do, and in this diminutive format the additional knobs required can increase the risk of fingers catching and shifting adjacent controls. The FREQ’s bell band’s frequency and gain knobs take you a long way very swiftly, while the Width and 1/3 buttons allow you that bit more scope to bend a sound to your will. It all feels like a good, practical balance for a 500-series module, and very instinctive and immediate in use. In terms of sonics, cleanliness is the name of the game here, with very low noise and distortion. I detected just the tiniest hint of flattering colour at the bottom end, but elsewhere my ears were greeted with an appealing clarity. The top shelving band, for example, sounds effortlessly clean and classy. It was trivially easy to tame cymbals or to bring out the breathiness in a ribbon‑miked vocal, and when applying the maximum HF boost, as part of an attempt to rescue a dull-sounding male vocal captured with an inappropriate dynamic mic, there was no trace of ‘grit’ or ‘scratchiness’; all I could hear was the beauty in the source being brought out. There’s headroom for days too, to the extent that I struggled to get a sound directly out of my audio interface that would clip it.

C’est Chic In the 500 series, you can use dedicated modules to put any sort of colour you want into your signal chain at any point, so there’s arguably good sense in opting for elegant simplicity in your main EQ. The Wolff Audio FREQ ticks that box handsomely, while offering enough functionality to make it versatile. Of course, it’s jostling for position in a crowded marketplace, where you’ll find the likes of API, RND, Mäag and Undertone Audio, to name just a handful, but the FREQ is ample evidence that Mr Wolff knows how to make a high-quality, great-sounding circuit. With four switchable bands here, it’s a great all-rounder, and if that’s what you’re after it would be well worth sticking on your audition list.

summary This classy-sounding, versatile, four-band console-style EQ offers just a little low-end transfomer sweetening, along with clear-as-crystal mids and highs.

$ $787. T Wolff Audio +1 805 500 8805. E info@wolffaudio.com W www.wolffaudio.com

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Schema: Light’s UI is stylish and modern but has plenty of creative possibilities to dig into.

Native Instruments

Schema: Light Software Instrument

NI bring some light relief to the Schema concept. JOHN WALDEN

f you like your virtual instruments with a decent dollop of ‘quirky’, Schema: Light from NI might well appeal. This latest release is a follow up to Schema: Dark. It shares the same playback engine but, as the name suggests, provides a somewhat lighter underlying loop‑based sample set to work with. Schema: Light’s curated collection of some 300+ source loops therefore features various bells, plucked sounds and a variety of suitably soft keys, alongside various mallets, noise and percussive elements.

Lighten Up The Schema engine allows to you combine up to four of the supplied loops within asingle preset. These are labelled A to D, and colour coded, including within the central pattern grid display, where the colours indicate which of the four loops has an active slice on each step of the current sequencer pattern. Each loop is chopped

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into 16 slices and the slices can then be sequenced within their own sequencing engine. This provides independent control over the number of steps (up to 16), which slice plays back on a specific step, and step-based control over items such as pitch, filter, and volume settings. Presets offer three switchable step patterns for each loop and, by default, the mod wheel allows you to adjust the blend of the loop layers, giving you real-time control over the degree of complexity in the resulting playback. A set of macro controls (shown at the base of the main display) provide modulation of key parameters, and there is a dual effects engine with various effects including delay and reverb. Amongst an array of other creative options, the playback engine includes scale locking (located top left) so that any pitched slices within the underlying loops will conform to your project’s key/scale requirements. Some 200 Kontakt snapshot presets are supplied, and these demonstrate Schema: Light’s ability to move from

ambient melodics through to more intense electronica. In the main, the ‘light’ nature is retained, but moods can easily span gentle and soothing through to more enchanting, mysterious, or downright creepy. Space precludes a full treatment of the editing features, but you can certainly go deep into the pattern creation, slice selection and modulation sequencing if you wish and, even with a little use, the UI quickly becomes easy to navigate. Usefully, additional levels of preset settings provide presets for pattern steps, filter cutoff, and step volume to speed up sound design. However, in terms of instant inspiration, the very neat randomisation options are particularly cool. These can be applied to any (or all) of the sounds (loops), steps, or slices used on any (or all) of the four layers, so you can go fully random to start from scratch, or just randomise a few elements to create variations from an existing preset.

Conclusion Having used Schema: Dark on a number of recent projects, I found it a great tool for creating low-end rhythms, pulses or bass elements (the kind of thing that might provide a bed for a tension cue in a film/ TV/game context for example). Schema: Light obviously contrasts that with its higher register, and more melody-focused potential. And, providing you don’t get them both running at full tilt (when things might get overly busy), they complement each other very well. No, there is nothing radically new in any of the individual features employed here, but the sum of the parts does make for something unique in Schema: Light’s whole. Music producers creating electronica styles would find plenty to enjoy, but I think the most obvious audience is media composers wanting some additional options for modern scoring work. It’s perhaps a niche product but, like it’s darker sibling, Schema: Light is cool, quirky, and suitably inspiring.

summary Schema: Light provides a powerful pattern‑based sequencing engine for manipulating its library of melody-friendly loops. It’s a great complement to Schema: Dark, bound to appeal to busy composers.

$ $149 W www.native-instruments.com

_AudioFuse 16Rig 2.0 Dive into immersive audio

Discover a versatile audio interface for hardware enthusiasts, offering full customization, seamless connectivity, enhanced creativity, and immersive audio. Meet a workhorse, reliable hub of connections and clever features, with immersive monitoring control and bass management, extensive speaker configuration options, and qualityof-life features requested by pro and creative users alike - be ready for next-gen audio exploration.

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iZotope Cascadia Ducked Tape Delay Plug-in

We check out this budget-friendly delay that aims to stay out of the way! JOHN WALDEN

he first two plug-ins in iZotope’s Catalyst series, which are designed to offer a particularly accessible processing experience, were Plasma (saturation) and Aurora (reverb). The latest addition, reviewed here, is Cascadia. First and foremost, this is a tape-style delay, with conventional delay controls on the left-side of the display. Delay time can be set manually or sync’ed to your host (with all the usual note length options readily available). Ping-pong and reverse delay modes are included and, for use as an insert effect, the large Delay knob controls the level of the delay signal that is blended with the original dry source. When using Cascadia as a send effect, you won’t want any dry sound, and engaging the Solo Delay button will return only the processed signal to your mix. The Feedback and Width controls

also do exactly what you’d expect, while the degree of tape-style modulation can be controlled using the Mod Depth and Mod Rate knobs. Finally, at the base of the large central spectrum display, low‑ and high‑cut filters are available, letting you control the tonality of the delays.

Ducks With Brains This is all very straightforward in use and, as demonstrated by the presets, Cascadia can provide a whole range of conventional delay treatments, from effective vocal thickening to long, ambient delay lines, with all stops in between. Yes, there are delay plug-ins that offer more elaborate feature sets but, in short, these basic controls give you a very usable delay, with a tape contribution can be easily adjusted to suit your needs. However, as with other Catalyst products, there are also some more ‘intelligent’ controls, and in this case they’re focused on ‘unmasking’. When activated, the unmasking process operates dynamically, ducking the delay signal only within those frequency ranges where the dry signal is present, so that the dry signal always remains clear within the mix without the delay seeming to disappear. You can adjust the attack and/ or release of this frequency-based ducking, while the Amount control adjusts its depth. A Sensitivity control sets the threshold of the detection algorithm and is perhaps most useful if you’re using Cascadia as a send effect, and want to drive the unmasking process a little harder. The central display gives informative real-time feedback on the extent of

“ The unmasking process operates dynamically, ducking the delay signal only within those frequency ranges where the dry signal is present.” 30

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ducking in specific frequency ranges but, thanks to the small ‘ear’ button, you can also monitor the delta signal — in other words, you can hear precisely what the unmasking stage is taking away. You can, of course, also bypass the unmasking entirely. I have to say that this unmasking process can be very effective in getting the delay to step out of the way of the dry signal, and keep things uncluttered. Finding the sweet spot does involve some back and forth, listening to the target vocal/ instrument soloed and in the full mix context, but this is straightforward to do, and the control set is easy to master.

Unmasked Ducking delays are not a new concept but, like Recirculate by Newfangled Audio or Sonible’s smart: reverb, Cascadia makes use of smarter algorithms than more traditional approaches. It’s a very useful effect, it sounds good, it’s straightforward to use, the asking price is modest, and it makes handling one of the biggest downsides of delay effects — the potential for adding clutter to your mix that robs more up-front sounds of their impact — incredibly easy. If you want to conduct your own experiments, there’s a free trial, but if the concept appeals I’d suggest checking out iZtotope’s Aurora at the same time, since it brings the same unmasking approach to reverb.

summary Another neat addition to iZotope’s Catalyst plug-in series, this is a decent emulated tape delay with a smart auto‑ducking feature — it’s very effective at preventing the delays from cluttering up your mix.

$ $49. W www.izotope.com

A revolutionary Rig. Revolution 14x16: The truly high-end recording interface with 2 Auteur preamps with Cinemag® transformers Revolution EXP: The truly high-end, AD/DA converter and word clock.

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clocking is based on everything we’ve learned in our 15-year history of modifying audio interfaces for superior conversion. On first listen, you may well notice details in your older mixes that you didn’t hear before you joined the Revolution. The Revolution EXP lets you connect up to eight mono processors, while the Revolution 14x16 features two Cinemag transformer-driven Auteur preamps that add color and weight to your signals. Its onboard headphone mixer lets you create a custom monitor mix for its three (!) headphone outputs. You’ve invested a lot in your studio hardware — choose Revolution to get the most out of it. Learn more at www.blacklionaudio.com. ©2025 Black Lion Audio. All rights reserved.

Revolution EXP: 8x8 AD/DA Converter and Word Clock with ADAT

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ARTURIA SYNTHX V

Arturia Synthx V Software Synthesizer

The latest addition to Arturia’s ever-growing range of vintage instruments is a recreation of the celebrated Elka Synthex. Arturia Synthx V $199 pros • It captures the essence of the original, and can sound excellent. • There’s something very intuitive and satisfying about the Synthex layout. • Its extended features take it way beyond the original. • Velocity, polyphonic aftertouch and MPE make it far more expressive than the original.

cons • There’s no Split facility. • There’s just one effects chain for the whole synth. • You can’t direct MIDI CCs to multiple destinations except through the macros.

summary If you want to extend the range of characters available in your armoury of virtual analogue soft synths, Synthx V is a very good way to do so.

March 2025 / www.soundonsound.com

GORDON REID

espite its good looks and pioneering use of DCOs, the Synthex was an eight-note polysynth with an underlying specification that wasn’t unusual in the early 1980s, albeit with more flexible filtering and some interesting tweaks. It was a massive commercial flop and, despite its adoption by players including Jean-Michel Jarre and Keith Emerson, it came and went almost unnoticed. Yet its reputation has soared over recent decades, turning what was a £400 ugly duckling in 1985 into a £10,000 swan in 2025. So this month’s question is not just whether a soft synth of a failed polysynth can replace the original, but whether it was worth developing in the first place.

The Sound Of The ’80s? You could play the Synthex in what we would nowadays call Whole

mode or using two layers that you could direct to the stereo outputs as separate synthesizers. Each voice was based on two DCOs that offered sync, ring modulation and an unusual cross-PWM capability, a noise generator, a multimode resonant filter and dual ADSR contour generators. Modulation was provided by a single LFO per layer plus a global LFO for vibrato and wow, while a chorus effect helped to create lusher tones than would otherwise be possible. A basic sequencer completed the specification. A very rudimentary MIDI capability was added to later units, and a retrofit was available for earlier ones. At first glance, the top panel of Synthx V looks much like a clone of this but, inevitably, there are differences, some in layout, and more in function. Examples of the former include replacing the oscillators’ octave selection buttons with five-position knobs, and moving the Sync button from the now-defunct Tuning panel to the Oscillator 2 panel. Nevertheless, there are many additions. For example, where there were four filter modes and a contour polarity button on the original, there are now five filter modes and the polarity is taken care of by a bipolar Amount knob. This has allowed Arturia to sneak in a 12dB/ oct low-pass mode, and very welcome it is too. Similarly, there’s an extra

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ARTURIA SYNTHX V

The Multi-Arpeggiator should cover all your arpeggiation needs — and then some.

knob in each of the Envelope panels, and these determine each contour’s velocity sensitivity. By far the greatest number of changes on the front panel relate to modulation. While retaining the underlying structure, each layer’s LFO now offers a random waveform, sync and, most significantly, the inappropriately named Reset function which turns the per-layer LFO into a per-note LFO. There’s also an extra destination (pan) for the Depth 2 modulation path. Elsewhere, the LFO 2 section sees some significant changes. Most obviously, the joystick of the original has been replaced by pitch-bend and modulation wheels, but if you prefer to use a two-dimensional controller, you can still map its CCs to the appropriate destinations. Clock sync has also been added, and PWM, audio gain and pan have been added to the available destinations. Oh yes... and the Performance/LFO 2 settings weren’t saved on the Synthex, but they are in Synthx V, which is a Good Thing. The Synthex’s unusual Glide/ Portamento section has been retained, but Glide can now be applied to the filter cutoff frequency as well as the oscillator pitches. Glide is what was sometimes called Slalom in the 1970s; an upward or downward sweep from a predetermined offset to the pitch of the note played, and this makes it possible to obtain the usual range of hard-sync screams. I find the dedicated contour to be far too linear, but you can use the modulation generators in the Advanced section (which we’ll address in a moment) to produce much nicer results. Although the Synthex offered a Double function that placed two voices under each key, it lacked unison. In contrast,

March 2025 / www.soundonsound.com

Synthx V offers three unison modes and your choice of key priorities when playing a layer as a monosynth. You can now stack five voices per layer with user-defined detune, or retain either four‑ or eight‑voice polyphony on layer B, which is streets ahead of the original. In addition to the Hold button (which has migrated from a now-defunct Envelope Controls panel) the new Master panel includes a Vintage knob, which allows you to inject some uncertainty (or ‘dispersion’) into the oscillator pitch and pulse width; the filter cutoff frequency, resonance, and drive amount; the contour times; and the level and pan of each voice. There are independent controls for each of these, and the Vintage knob scales how much is applied by each, ranging from gentle variations to full-on sonic nonsense. It isn’t true to the original — after all, DCOs eliminate the pitch inconsistencies of VCO-based polysynths — but someone at Arturia thought that it would be valid. To be fair, it increases the available palette, so why complain? Let’s now talk about those layers. There are three modes. Solo plays either layer A or layer B, so you can program independent sounds in each and switch

between them. Duo plays both layers, umm... layered, while Stereo Link copies the current layer to the other and then pans the two hard left and hard right. However (and it’s a big however) there’s no Split mode. I’ve no idea why Arturia precluded this, and it’s a bit naughty. On the positive side, you can now play layered patches with eight-note polyphony and use the new Pan/Volume panel to determine an initial position for each layer and then spread its notes either side of this. With a bit of dispersion dialled in, the results can be lovely. At the end of the signal path, the chorus effect for which the Synthex was famous has been mimicked as closely as possible, emulating the subtle swirl of Chorus 1, the deeper effect of Chorus 2, and the quasi-ensemble of Chorus 3. It’s applied individually to each layer, which is a necessity if you want to program independent sounds in each.

Beyond The Sound Of The ’80s Inevitably, Arturia’s designers couldn’t resist adding more features, and you’ll find most of these behind the word ‘Advanced’ at the top right of the GUI. This opens detailed pages for seven additional sections. The first is the programmer for the Multi-Arpeggiator, which comprises four arpeggiators that can be used to create anything from simple arpeggios to complex, polyrhythmic sequences. Its playback controls lie alongside the wheels on the main screen, but the Advanced pages offer controls for the global rate, gate length and scale, as well as for each arpeggio’s mode, octave range, transposition, start offset, rhythm, length and retrigger mode, its gate time relative to the global gate length, ratcheting, and MIDI velocity offset. Phew! There’s also a randomiser that affects the

You don’t have to create crazy envelope shapes like this with the Function Generator, but you can if you want to.

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Trinnov Nova

Speaker & Room Optimisation System Room-correction pioneers Trinnov were already a step ahead of the competition. Now they’ve taken things further. PHIL WARD

hen reviewing Trinnov’s ST2 Pro monitor and room optimisation hardware in SOS July 2019, I initially felt a little overwhelmed by its combination of extraordinary technical capabilities, somewhat complex configuration and opaque user interface. Despite that, I described it as possibly the “last word” in monitor correction. But time moves on and, in 2023 at the NAMM Show, Trinnov launched a replacement product called the Nova. We didn’t review it at the time but it recently benefited from a significant software update, and this latest iteration is the subject of this review. The Nova takes Trinnov’s immensely powerful room and monitor analysis and optimisation technology, builds it into a compact 1U case (the ST2 Pro was a deep 2U), and adds a Windows/ Mac control and configuration desktop application (and iOS app) blessed with a far more accessible and friendly way

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in. As well as being more compact and approachable than the ST2 Pro, it’s significantly less expensive, and while it costs more than entry-level systems such as Sonarworks SoundID Reference and IK Multimedia’s ARC, it aims to do rather more than them.

Hardware & Licences The Nova’s shrinkage wasn’t achieved by stripping out features or connectivity, but by moving all the processing (from, effectively, a complete Linux PC with separate I/O boards) to an Arm chip on a newly designed, integrated motherboard. Though far more intensive and demanding in terms of R&D and manufacture, this approach is significantly more cost effective and space efficient. The front panel carries a power switch, a USB socket for connection of the optional La Remote desktop controller (see box), a socket for Trinnov’s updated 3D microphone and a headphone socket, which is fed an unprocessed stereo signal. Around the back, the analogue

audio connectivity comprises six line-level inputs (two on XLR sockets and four on TRS jacks) and six XLR outputs. Digital audio connectivity comprises S/PDIF and ADAT inputs and an AES3 output. Finally, a couple of RJ45 sockets cater for network connection and, potentially, Dante AoIP (Audio over IP). The presence of Dante among the Nova’s audio inputs and outputs means that, along with its primary role of optimisation and control, a Nova could be used as an audio interface, rather than ‘just’ a monitor controller and optimiser. With ‘full-fat’ (amplitude and phase) optimisation engaged, the latency is said to be typically around 25ms, but with monitor and room optimisation disabled the latency will be defined primarily by the Dante system — typically well into single figures or even less. For some users, this could be hugely significant. In mastering or mix studios where the need for inputs is modest, it could make a conventional interface superfluous, and the prospect of trading in an interface and monitor controller could make the Nova significantly more affordable. There’s some complexity to explain regarding how the six outputs are

The 1U form factor is much more compact than that of the Nova’s predecessors, thanks to a complete reworking inside — the older units used a traditional Linux-based PC, whereas the Nova was completely reworked to run on an Arm processor.

Photo: Funky Junk Italia

pair of outputs; the other outputs can pass audio, but they can’t simultaneously have optimisation applied. To apply optimisation to 3, 4, 5 or 6 outputs simultaneously, you must buy more licences. (These licences are perpetual: once you buy one, the hardware is permanently ‘unlocked’). It would be possible to create an optimisation profile for different monitor pairs and switch a single stereo licence and profile between them, at the same time as switching outputs, but such shenanigans might soon become tiresome and nudge one towards the purchase of more licences. The stereo-plus-sub case would definitely benefit from an extra licence.

employed and treated in terms of Trinnov optimisation licences, as they can be employed flexibly to suit different installations. Their use as three independent stereo pairs (for example for main monitors, nearfields, and a pair of NS10s), a combined 5.1 system, or for two monitor and subwoofer systems, is all perfectly feasible. With only six outputs, a Nova can’t do Atmos, and that seems to have disappointed some folk, but Trinnov do have Atmos solutions further up the product range. With regard to subwoofers, it’s fine to use just one stereo output to optimise a downstream composite system of stereo monitors and subwoofer, but that will mean the Trinnov optimisation is applied globally, with the subwoofer integration managed conventionally. Using a separate Nova output for the subwoofer (or two for stereo subwoofers) with the Nova’s bass management engaged is likely to result in significantly better results, because the optimisation algorithm will then be aware that the subwoofer is a separate device with a discrete location in the room. Enveloping all of this talk of Nova outputs is the fact that Trinnov offer optimisation licences in pairs. By default, a Nova comes with one licence for one

3D Mic The ST2 Pro’s 3D mic required an internal battery and connected to the hardware using four XLRs, but the Nova mic is phantom powered and connects using a far more user-friendly multi-channel

locking etherCON socket. The mic also now has its calibration data embedded: the Nova automatically identifies the specific mic and handles the calibration (the mic serial number data had to be manually uploaded to the ST2 Pro). The 3D mic looks unlike any conventional room optimisation measurement device, and appears to have rather more in common with a first-order ambisonic mic. This is because it sets out to do rather more than simply measure frequency response. It presents four capsules arranged in a spaced tetrahedral array and, as well as measuring frequency response, is able to triangulate horizontally and vertically the location of the monitors in the room by enabling analysis of signal arrival times at each capsule. But it’s not just the monitor positions that the Trinnov mic can locate: it can also distinguish discrete early reflections from the direct sound, which means Trinnov’s analysis software

Trinnov Nova From $3490 pros • Extraordinarily good optimisation. • Far easier to use than the ST2 Pro. • Great control app.

cons • None. Trinnov’s room and monitor optimisation technology always was a step beyond its competition. The Nova now makes it significantly more accessible too.

Photo: Funky Junk Italia

summary

The 3D mic, which makes it possible for Trinnov’s room-optimisation system to outperform others, has been updated, and connection to the hardware is now much more convenient.

www.soundonsound.com / March 2025

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T RIN NOV NOVA

“It’s as if a layer of audio uncertainty has been removed.”

Diagram 1: The listening-position frequency response of the right‑channel speaker in the author’s studio.

Diagram 2: The same right-channel KH150 listening-position frequency response optimised using the Nova.

has the data it needs to be able to correct monitor phase as well as frequency. It’s genuinely clever technology and, to my mind, is what really sets Trinnov apart. A further element that distinguishes Trinnov from alternative room optimisation systems is that, rather than capturing data using a swept sine wave, it employs an MLS (Maximum Length Sequence) stimulus. Though it sounds much like pink noise, MLS is actually a non-random, periodic wide-band signal with a correlated (ie. time-stamped) input and output. Along with the 3D mic, the non-random and correlated nature of the MLS stimulus enables the Trinnov analysis and optimisation technology to function in both the time and frequency domains.

and channel-selective mute. A talkback mic is also included in La Remote and that can be routed, as required, to monitor or headphone outputs. Switching between inputs, regardless of their format or connection, is similarly simple, and as the Nova is fully Avid EuCon compatible, it can be integrated with Pro Tools workflows and practices.

In Use Installing and using a Nova is easier than I remember being the case with the ST2 Pro. Installation is a case of connecting mains power, inputs and outputs, providing the Nova with wired network access, and connecting the mic. If present, the La Remote desktop remote control connects to the Nova via its

front-panel USB socket, or to a computer running the Trinnov control app. Connecting the input and outputs raises the fundamental question of what use the Nova will be put to. If it’s to work downstream from an interface, effectively as a monitor controller, it’s probably sensible if possible to use the ADAT input, keeping audio in the digital domain and ensuring that the signal path doesn’t pass through two volume control stages (the interface and the Nova volume control). That requires an interface with an ADAT output, of course. Some (eg. my Audient ORIA) don’t have one, and in that scenario, the Nova’s analogue inputs can work perfectly well — though this is perhaps another nudge in the direction of using the Nova as a Dante‑connected AoIP interface. With the Nova connected and fired up for the first time, the desktop app’s configuration wizard walks you through the input/output signal routing, level setting and 3D mic setup stages that are required before you can undertake the first monitor/room analysis and optimisation. Setup is comprehensive, and then some, but there are other options and information available once a Nova is configured and an optimisation profile has been saved. Once I’d first used the Nova and 3D mic to analyse and optimise my Neumann KH150 speakers in my room (a process that took just a couple of minutes), I made some in-room 12th‑octave smoothed frequency response measurements using FuzzMeasure and a conventional measuring mic, as shown in diagrams

Monitor Control I’ve concentrated mostly on the Nova’s room and monitor compensation abilities, but the Nova is also a fully featured monitor controller that offers all the capabilities you’d expect of such a device (especially so when La Remote is used). For example, switching between each pair or trio of outputs can be achieved easily from the desktop or iOS apps, or from La Remote, and reference levels or EQ preferences can be set for each. Mute, solo and dim options are also easily available, as are mono, difference

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Diagram 3: Here, the Nova’s ability to linearise phase is apparent: the red curve is unoptimised, the green curve Neumann MA1-optimised, and the blue curve Nova-optimised.

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Distributed by Music Marketing Inc.

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Black Lion Revolution 14x16 USB Audio Interface

Black Lion take a different approach with their latest audio interface. SAM INGLIS

hat does ‘better’ mean in the world of audio interfaces? For most manufacturers, the answer seems to boil down to measurable numbers. With every new generation, we are promised more dynamic range, lower distortion, less noise, and so on. But is that actually what we want? The thriving market for vintage-style mic preamps would suggest that customers often have a slightly different understanding of ‘better’. Many of us are happy to enjoy a bit of non-linearity and noise if it contributes to the sound we’re looking for, and there are a few interface manufacturers who acknowledge this preference, such as Universal Audio with their Unison modelling preamps and Heritage Audio with their i73 range. Even then, though, any departure from the usual numbers game applies only to the input stage. It’s assumed that although we might want to add colour on the way in, the rest of the circuitry should score as highly as possible when faced with an Audio Precision test kit. Over in Chicago, Black Lion Audio are taking a slightly different approach. As well as offering vintage-style

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preamps, they emphasise the primacy of subjective sound quality over quantifiable performance throughout their designs. Interestingly, though, the sonic benefits they claim for this are actually the same ones you’d usually associate with better specifications; the phrase “crystal clear” appears many times throughout their product literature. Of the product under review here, Black Lion maintain that careful design and component choice reduces “noise contamination” and “is optimised for best signal-to-noise ratio”, but they also say “We found that the best-sounding parts in the output path weren’t necessarily the lowest noise components but they resulted in an optimal tone and signal purity that is instantly apparent to the listener.” So what is the Revolution 14x16, and does its collection of “hard-to-get Vishay, Wima, and Nichicon capacitors and thin film resistors” actually make it sound better than rival products? The first of these questions is much the easier to answer, so let’s start there.

Revolution In The Head The Revolution 14x16 is a USB audio interface that, as the name indicates, offers a total of 14 inputs and 16 outputs at base sample rates. Ten of each are accounted for by a single ADAT Lightpipe input and output and a stereo coaxial S/ PDIF input and output. There is also word clock in and out on BNC connectors,

which employ Black Lion’s proprietary Macro-MMC clocking technology. Connection to the host computer is made through a single USB-C socket. All of this digital I/O is located on the rear panel of the 1U chassis, which is sturdily made of metal and has integrated, non-removable rack ears. Next to the USB-C connector

Black Lion Revolution 14x16 $1299 pros • Switchable inserts on all four analogue inputs, plus duplicate front and rear sockets for the first pair, make for very flexible connectivity. • Transformer-balanced Auteur preamps on inputs 1+2 add a touch of character. • Three headphone outputs. • No software control panel to learn, at least on macOS.

cons • Direct monitoring is limited, and there are minimal ‘master section’ features. • Wall-wart PSU with non-locking connector. • Line-up level of +16dBu is on the low side for interfacing with pro audio gear.

summary Black Lion’s rackmounting audio interface combines vintage-style mic preamps with a unique feature set that sets it well apart from any competition, both in terms of what it will do and what it won’t.

you’ll also find a socket for the ‘wall wart’ DC power supply. Given the emphasis on things like “highly filtered power rails” and “full-circuit power decoupling”, the use of a generic external PSU is perhaps surprising, and although the rear-panel socket has threads to accept a locking connector, the PSU doesn’t have one. Black Lion told me that this is a deliberate choice, as they often encounter stripped or damaged locking connectors on other interfaces sent to them for modding. Things get a lot more interesting, and unusual, when we turn to the Revolution’s complement of analogue I/O. On the output side, a conventional pair of rear-panel monitor outputs on balanced quarter-inch jacks is complemented by an additional pair of balanced line outputs, and no fewer than three front-panel headphone sockets. The arithmetic behind the 14x16 designation would suggest that Black Lion are counting these as a single stereo pair, which is fair enough, since they all deliver the same signal. This maths is reflected in the number of inputs and outputs presented to the DAW, so on the output front, your software ‘sees’ a single stereo headphone output in addition to the monitor and line outputs and digital I/O. It’s on the input side that the Revolution 14x16 gets really generous with the socketry. Whereas inputs 3+4 use the clean preamp design found in BLA’s existing Revolution interfaces, the first pair of inputs feature Black Lion’s Auteur preamp circuit, which is interestingly different from other ‘vintage-style’ designs. For manufacturers like Heritage Audio, the mojo resides in the input

stage, and the key component is the input transformer — which, under normal circumstances, is the very first thing the signal from your mic or instrument ‘sees’. By contrast, the Auteur design features a “modern front end that’s best described as fast and transparent”. Any vintage quality is conferred instead by the output stage, which features a Cinemag CM-18313 transformer. This sort of design choice is the subject of endless debate, but Black Lion’s approach makes sense to me. The sonic contribution of an input transformer is essentially fixed, and determined purely by the characteristics of the source signal. Locating the transformer on the output

and other outboard gear. What’s more, the inserts are switchable from the front panel, so you can leave outboard or synths permanently plugged in and use the front-panel buttons to decide what should be recorded. On top of that, the two Auteur preamps have combi XLR/ jack sockets on both front and back panels. The availability of dedicated insert-returns-cum-line-inputs on all four input channels means that the jack part of each combi socket is given over to high-impedance signals, so you can record up to four electric guitars at the same time. You can also use the Revolution 14x16 as a standalone mic preamp by treating the insert sends as analogue outputs. Consequently, although there are only ever four analogue inputs available for recording and monitoring, the Revolution 14x16 is uniquely flexible in terms of what and how much gear can be connected at once.

Numbers Up As I’ve already mentioned, Black Lion Audio downplay the significance of audio specifications, and in some cases this leads them to be rather terse or obscure on the matter of measured performance. Thankfully that’s not the case with the Revolution 14x16, for which reasonably complete specifications are available. These suggest that the maximum input level for the two Auteur preamps is a meagre +6.5dBu at minimum gain, but Black Lion confirmed that this

“Do Black Lion’s audiophile component choices give it a subjective edge over competing products?” stage instead means that the user has some control over how hard it’s driven, and thus how much it colours the sound. You can deliberately apply more gain than is strictly needed in order to push a transformer-balanced output stage, then compensate for excess level using a fader or other output attenuator if one is available. The Revolution 14x16 doesn’t have built-in output attenuators, but Black Lion have gone one step further by providing balanced insert points on all four analogue inputs. These could be employed to insert an attenuator between preamp and A-D converter, allowing the preamps to be overdriven, but you could also patch in EQs, compressors

measurement ignores the 10dB pad, so in practice there shouldn’t be any problem accommodating hot signals. The variable gain control spans 60dB (albeit with some bunching towards the top of the dial), producing a total gain range of 70dB for inputs 1+2. Inputs 3+4 are less versatile, spanning a 55dB gain range with a maximum input level of +8.1dBu. All four preamps have individually switched +48V phantom power, which can produce a brief but slightly alarming burst of noise if toggled whilst a mic is connected. The Auteur preamps also have front-panel buttons to switch polarity and toggle between front and rear inputs. In other respects, the Revolution 14x16’s specifications are roughly in line www.soundonsound.com / March 2025

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Elektron Digitone II SIMON SHERBOURNE

lektron’s Digitone follows closely behind its twin the Digitakt in receiving a MkII rework. The two Digi units share the same build and sequencing workflow, but while the Digitakt is a sampler, the Digitone is a digital synth. As with the Digitakt refresh, the Digitone II makes considerable gains in terms of sequencing, track count and flexibility. It also branches out into new areas of synthesis.

Raising The Tone The most visible difference on the Digitone II front panel hints at its most

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FM Synthesizer & Sequencer

Elektron’s updated FM groovebox is a powerhouse. significant change. The four colour-coded track selector buttons are gone because there are now 16 synth tracks at your disposal, selected from the main ‘trig’ buttons, as on the Digitakt. Polyphony has been doubled to 16 to facilitate this. This hugely extends the possibilities for the Digitone, giving you more scope to produce layered compositions, and letting you use the Digitone as a drum machine without forcing sounds to share tracks. You can still use the Digitone to sequence external MIDI gear as well. Instead of dedicated tracks, any track can now be designated a synth or MIDI

track. Connectivity remains the same: a full complement of DIN MIDI ports, and a pair of inputs for monitoring one stereo or two mono external sources through the unit and its effects. The redundant track buttons are repurposed for direct access to transposition, the arpeggiator and the new Note Edit mode. As there’s no longer a need for a dedicated MIDI mode, a button is freed up on the left array that now toggles the trig buttons into a track-focused chromatic keyboard layout. Like on the Digitakt II, Bank/ Pattern selection has been consolidated

to a single operation, meaning Song mode now gets its own top‑layer button.

FM Extended The original Digitone was based entirely on a versatile 4-op FM engine. This is still the default synth on the MkII but is joined by three new generators: FM Drum, Wavetone and Swarmer. To recap, the OG FM engine (now called FM Tone) uses an adaptation of classic DX-style FM. Four operators are patched in a choice of eight preset algorithms offering different cross‑modulation, feedback and output routings. Sound design is made friendly by a clever system whereby two of the operators are treated as a pair, and the main frequency controls are quantised to musically harmonic ratios. Fine control is still there, but kept off the main page. You can also adjust the waveforms of the oscillators, tweak feedback and adjust the blend of output sources all from this main view. The actual amount of FM in a patch is dialled in on the second page, which is perhaps the least intuitive part of the layout but makes sense because this is also where you find the envelopes that modulate this FM. My review of the original Digitone (SOS September 2018) goes into a lot more depth about all this, so let’s turn to the new engines. FM Drum is a hybrid engine optimised for creating drums and percussive sounds. This features a 3-op FM ‘body’ side fed through a wavefolder. You then have a transient side that mixes a complex noise source with a PCM element (the same concept employed by

the Korg Drumlogue) selected from a modest collection of classic kick, snare, hat and clap attack phases. I counted something like 24, although it’s not clear as the samples crossfade into each other as you adjust the encoder. The noise gets its own filter, and both sides of the equation have independent hold-decay envelopes. The Wavetone synth engine is a departure from FM. It’s a two‑oscillator setup with linear detune, and ring mod or hard sync cross-modulation options. The oscillators are wavetable based, with two preset tables available. One table morphs through the classic subtractive starting points of sine, triangle, saw and square, while the other contains a range of harmonic blends. Before you ask, no, you can’t change these or load other wavetables. Alongside the main oscillators is a multi-mode noise source

Elektron Digitone II $999 pros • Sixteen tracks. • Multiple synth engines. • All-round workflow improvements.

cons • The Digi units still lack Direct Jump pattern switching. • No polyphonic aftertouch.

summary The MkII rework of the Digitone unlocks tons of potential and versatility from an already great synth workstation.

harmonics. Animation and noise controls provide modulation for some classic anthemic pads and rave stabs.

FM + Subtractive The foundational concept of the Digitone is the mashup of FM sound generation with subtractive filtering. Now, with the Wavetone and Swarmer synth engines this combo gives you a fully conventional architecture for virtual analogue synthesis, which perhaps dilutes the idea but makes the Digitone more versatile. The Digitone II filter section now has multiple modes, again referred to as engines. I covered these in the Digitakt II review, but let’s have a quick rundown. The new standard‑issue module is the Multi-Mode engine, which features a four-pole filter that can morph from low- through band- to high-pass modes. The Lowpass 4 engine gives a warmer, squelchier low-pass, while the original Digitone LP/HP filter is included as Legacy mode. For less invasive shaping

“If you gel with the Elektron way of working and you’re more interested in synthesis than sampling, this is the best groovebox you can buy right now.” with its own envelope and filters. Wavetone’s general outlook then is much like that of classic ’80s polysynths, but it does add phase distortion to the mix, with both main oscillators able to warp their duty cycles to produce PWM results with any waveform. The last synth engine (but as we’ll see, not the last synthesis technique) on offer is Swarmer. This is a supersaw‑style stacked oscillator, although it’s not limited to saws. A simple, one-page UI lets you set the waveforms and octave offset of the fundamental and stacked oscillators, and the width and detuning of the

The rear panel hosts a power switch and PSU inlet, a USB port, a full set of MIDI I/O, stereo line inputs and outputs, and a headphone output.

www.soundonsound.com / March 2025

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ELEKTRON DIGITONE II

there’s a one-band parametric EQ mode. This might sound stingy, but every filter mode also includes a useful second stage of low- and high-pass filtering. Also in the engine list are Comb+ and Comb- filter engines, featuring variable delay and feedback and a low-pass roll‑off. I’m singling these out because they open up another interesting avenue for sound design. Ping these filters with a transient impulse like a noise burst from the oscillators and set the filter to full keyboard tracking and you can achieve Karplus-Strong style physical modelling sounds. There’s a surprisingly convincing acoustic piano patch in the factory library that exemplifies this method.

New & Exclusive

Three new engines explore new synth territory for the Digitone II.

routing for effective side-chain operation, and likewise you can isolate tracks from the incredibly useful Control All mode that’s exclusive to the Digi units. And also in a similar vein you can set which tracks respond to Pattern Transpose operations. Even more ‘new for v2’ features we’ve looked at before are Layers (track linking), Perform Kit mode (pattern/parameter independence) and Page Loop for cycling a range within a Pattern. Exclusive to the Digitone II is Note Edit mode. This new view lets you easily add and edit notes and chords per step. As well as the actual notes, you can edit the timing, velocity and length of each note within a step individually — a big step up from the original implementation of polyphony, and a much more precise way to edit sequences in general.

FM Drum provides a hybrid approach to synthesizing drums and percussive sounds. The core workflow of the Digitone is consistent with all Elektron instruments, so I won’t cover too much old ground. Suffice to say Conclusion that the Digitone is structured like a drum machine, based around The Digitone was already an self-contained patterns comprising eminently usable and useful little multitrack sequences and kits powerhouse. I made more music of synth or MIDI voices. It’s also on the original Digitone during perfectly possible to use the the review period than any other Digitone as a multitimbral sound piece of gear I’ve tested. I got module with each of its 16 tracks Wavetone is a two‑oscillator wavetable engine with phase distortion. into a groove with it, quickly on different MIDI channels. With an generating melodic ideas using external keyboard connected, the the arpeggiator, adding some Digitone II is not only a wonderful simple synth drums using sound synth, it’s a production workstation flipping within one or two tracks, where you can capture sequences then performing a few pattern to whichever track is focused. variations while capturing to the It would be nice if Elektron jumped computer via Overbridge. on the trend of including a MIDI The leap from four to 16 tracks USB host port to simplify this greatly expands what’s possible, use case, and polyphonic aftertouch and makes it feasible to construct wouldn’t go amiss. complete tracks in the box. During Note Edit mode provides precise note entry and editing, In a DAW-based context, using the review the Digitone II became especially when working with chords. Overbridge for audio and plug-in the default drum machine in control over USB makes the Digitone II my studio, running 16 FM Drum engine to freeze results into the main sequencer. a highly powerful ‘plug-out’, to borrow voices. At other times it became the Also present is the spilling out of trigger Roland’s term. Most users will probably polysynth and chord companion to probability, conditionality and fills, which find their own workflow that falls my Eurorack rig. It was also my mobile were previously combined under a single somewhere in between these scenarios. sketchpad for breakout sessions, with option. (I’ll repeat my request here for The MkII upgrade extends and no thought needed to budgeting tracks more inventive chance operations that improves on the original sequencing and between drums and synths. If you gel affect sequences beyond basic gating). pattern creation features in the same with the Elektron way of working and Again consistent with the Digitakt, ways as on the Digitakt II, and adds some you’re more interested in synthesis than all parameters have been reorganised extras. We have the same doubling of sampling, this is the best groovebox you in a much more intuitive layout, the pattern length to eight pages or 128 can buy right now. especially the effects. Speaking of steps. The great Euclidean live sequencer which, there are new master Chorus mode is here with its two rotatable layers and Overdrive effects. Individual tracks $ $999 to keep things interesting, and its ability can be pulled out of the Compressor W www.elektron.se

March 2025 / www.soundonsound.com

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Universal Audio SD‑3, SD‑5 & SD‑7 Dynamic Microphones

UA apply their considerable modelling expertise to a trio of moving-coil microphones. NEIL ROGERS

ew brands have maintained their heritage of classic analogue studio equipment whilst simultaneously pushing forward the world of digital audio as successfully as Universal Audio. This is very much evident in their recent microphone offerings, which include a number of premium capacitor mics made in collaboration with David Bock, as well as the Sphere DLX and LX, advanced modelling microphone systems that resulted from their acquisition of Townsend Labs in 2021. Microphone modelling is not a new phenomenon, of course, but typically it involves modelling high-end capacitor mics — usually models that are financially out of reach for most engineers and home recordists. UA’s new SD-series modelling mics offer something a little different, in that they’re primarily intended

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to offer users emulations of a selection of more common (and affordable) moving-coil dynamic microphones.

Model Behaviour In 2022, UA released their debut dynamic mic, the SD-1. This was intended to cover vocal and broadcast/podcasting applications, and was launched as part of their Standard series, which also included the SP-1 pencil mics. A year later, UA released an affordable large-diaphargm capacitor mic, the SC-1, alongside a new piece of software called Hemisphere, which endows Standard-series mics with the ability to model classic designs. And now, they’ve released a further trio of dynamic mics that can also be used with Hemisphere. For review here we have the new SD-3, SD-5 and SD-7, which are intended for typical tracking applications such as close‑miking drums or guitar cabinets.

All three mics have an impressive, solid feel to them and a nice ’60s sci-fi aesthetic that encourages you — if you’re so inclined — to hold them as mock ray guns for your amusement. Looking at the smallest first, the SD-3 offers models of five very well-known dynamic mics: the ubiquitous Shure SM57, its predecessor the Unidyne 545, the Sennheiser MD 409, the Audix D4 and the Sennheiser e604. The hefty-looking SD-5, as its physical size suggests, is aimed at recreating mics associated with recording kick drums and bass cabinets. Emulations of modern staples like the Shure Beta 52, Audix D6 and AKG D112 are included here, alongside a vintage AKG D12 option. Intriguingly, the SD-5 also presents the option of recreating the ‘speaker in reverse’ trick, in which an NS-10 speaker cone is used to capture a sub-bass tone when being placed outside a kick drum. Lastly we have the SD-7, which features a slightly wider frequency range than its siblings and focuses on classic Sennheiser models such as the MD 421 (both the vintage beige model and the current version), MD 441 and vintage MD 409n. I was also intrigued to see the option for the beyerdynamic M 160 ribbon mic

here, as the M 160 is one of my studio’s most-used microphones.

Enter The Hemisphere The experience of using the modelling software is key to how well these systems work in the real world, and I’m happy to report that it was painless to get up and running with the Hemisphere plug-in. I don’t use UA’s LUNA DAW, but as a user of their plug-ins, I was able to quickly access what I needed via their UA Connect portal. Once installed, it was just a case of popping the Hemisphere plug-in on an insert in Pro Tools and choosing my mic model of choice. Whilst not as comprehensive as the options available with UA’s Sphere plug-in, there are still plenty of controls for adjusting the sound of the mic, and these can of course be tweaked after the fact. Depending on which model you have selected, you may be presented with filter controls which correspond to to real switches found on the mic being modelled. When the mic has fewer than three settings or switches (if the original hardware had no filter options, say), filter settings have been added to provide more flexibility. Also included is a Proximity control, which mimics moving the mic closer or further from its source, allowing quite dramatic changes in tone depending on the model. You can also experiment with an Axis control, which allows you to experiment with the mic’s off-axis response — you can virtually rotate the mic’s capsule as far as 180 degrees. (In practice, I found the effect of this was often of limited use in practice.) Lastly, we have a polarity reversal switch as well as an option for running the Hemisphere plug-in with a reduced CPU load. As you would expect from a company with a long history of producing plug-in emulations, the software performed flawlessly throughout the review period.

the mics fared when used in combination with ‘normal’ mics in my day-to-day recording sessions. If you are already on board with UA’s ecosystem, perhaps using their modelled preamps for tracking with an Apollo interface, you should be able to audition mics very easily and with no added latency, which offers more potential for nailing your sounds with the age-old combination of mic choice and placement. For my own setup, I had to either run sessions with very low buffer settings to avoid any audible latency, or rely on a combination of engineering instinct and listening back to takes when deciding on mic placement, which, whilst not ideal, worked fine.

How Do They Sound? Before throwing these intriguing new mics into real sessions at the studio, I made a point of putting them side to side against a couple of mics I use a lot in my studio — a Shure SM57 and the beyerdynamic M 160. With the mics used in scenarios I know very well indeed, such as close‑miking the top of a snare drum, or positioned just to the edge of the centre of a speaker cone, it was impressive to

Universal Audio SD-3, SD-5 & SD-7 pros • Very affordable mic modelling. • Great selection of dynamic mic emulations to play with. • Rugged build quality. • Hemisphere software works very well. • Probity and filter control offer additional control.

cons • You ideally need to be able to monitor through the modelling software when recording. • Adjusting the off-axis response seemed less effective than other controls.

summary UA’s stylish new SD dynamic modelling range opens up a more affordable route into microphone modelling with emulations of classic dynamic mics used for instrument recording.

hear how close the modelling software compared to the real thing. To my ears, the emulations often had a slightly ‘softer’ edge to them, but tonally they seemed very accurate. I was sufficiently convinced to begin slipping them into recording sessions to provide different options, and

Workflow These mics are probably not intended for an engineer like myself who has a decent collection of mics already, including many of the models covered by Hemisphere. I also work in a traditional way, recording and monitoring through an analogue desk, so having to audition mics in software takes a bit of mental adjustment! In that context, I was a little cautious around slipping these mics into my day-to-day sessions in the studio, but it was also good for my review to see how

Among the models available for the SD-3 are the Shure SM57, and its predecessor the Unidyne 545.

www.soundonsound.com / March 2025

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U NI V E R S A L A U DIO S D -3 , S D -5 & S D - 7

more ‘classy’, mic models and I enjoyed using the Sennheiser MD-441 option for close‑miking drums and guitar cabs — it’s a dynamic mic that offers a more full-range sound than other options. The two 421 options sounded good on toms, and I found myself slightly preferring the older version The SD-5 is clearly aimed at bass and for its more kick-drum duties, offering virtual versions rounded sound. As of the Audix D6, AKG D12 and D112, Shure I mentioned earlier Beta 52, and the Yamaha NS-10 speaker cone wired as a microphone. I’m a big fan of the beyerdynamic M 160, and it’s my go-to mic for micing guitar cabinets. The generally putting them through their paces emulation included with the SD-7 sounded in the real world. pretty close in a direct comparison, Looking at the diminutive SD-3 though if I’m splitting hairs, the software model first, it was interesting to hear the version sounded a little less forward in the differences between the different mic midrange — which could be good or bad options. Perhaps surprisingly, I found depending on the setting. myself liking the older Unidyne model a lot compared to the modelled SM57 Summing Up that I know so well. When close‑miking a snare drum, I also found that playing There’s an interesting debate to be had with the filter and proximity controls in about this new area of mic modelling the plug-in worked nicely for fine-tuning technology due to both the cost of the the modelled mic to blend with the other mics they are emulating and the situations mics used around a drum kit. Auditioning in which these the much larger SD-5 on my kick drum mics are typically was great fun, and I was surprised at how used. With the well some of the more modern emulations large‑diaphragm worked, with the Audix and Shure options capacitor models offering punchy modern kick sounds that that have been would cut through a busy mix. The AKG with us for a few D112 model sounded as underwhelming years, you tend to as I remember the real thing being, but get a selection of the vintage D12 model worked great with truly ‘aspirational’ the proximity exaggerated to bring in a bit mics that you more low end. Looking at the SD-7 last, would never be this is the mic that contains some slightly able to afford otherwise. That type of microphone is also primarily used for I’ve prepared some audio examples vocal sessions, of the SD mics in action on a drum in which mic kit and a guitar cabinet, including some comparisons of the Hemisphere In addition to its models with the original microphones. models of moving-coil To hear how the mics performed, visit mics, the SD-7 can https://sosm.ag/ua-sd-media. emulate beyerdynamic’s classic ribbon, the M160.

Hear For Yourself

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placement can be more predictable than it is when recording drums or guitar amps. Despite being very well priced, for the cost of the three mics I’ve looked at in this review, you could pick up two or three of the cheaper mics that are being modelled and focus on the traditional principle of getting sounds right at the source. The counter side of this argument is that the SD-series mics are affordable, and the mic models found in all three models sound good. With low-latency monitoring you can have an instant mic collection to audition at the click of a mouse, with the potential of using multiple instances to record a drum kit or whole band. You also get the option of changing the mic in the context of the mix of production as it evolves, and I found the controls for changing filter options and fine-tuning proximity effect worked nicely — almost as another layer of control before reaching for EQ and other mix processing. Generally, I was impressed with how these mics performed in my studio and I can see them being a great way of complementing a collection, learning what you like mic-wise, or as a standalone collection of mics for a minimal software-based recording setup. And if you’re already immersed in UA’s modelling ecosystem, they could be an even more attractive proposition. $ SD-3 $129 each or $349 for pack of three. SD-5 $199, SD-7 $149.

W www.uaudio.com

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MINI REVIEWS

Relab VSR REV6000 Stereo Reverb Plug-in I recall being hugely impressed by the original TC System 6000 reverb (which we reviewed way back in SOS April 2003: https://sosm.ag/tc-reverb-6000). That ran on the extremely costly System 6000 hardware platform, which used algorithms rather than IRs, but the original System 6000 reverb incorporated a technology called ‘ray tracing’ to create extremely accurate-sounding early reflections alongside traditional FIR (Finite Impulse Response) filters. In a real space, the early reflection pattern varies depending on the location of the source within the space, and ray tracing proved an effective approach for recreating this element of the reverb character. In terms of the processing, essentially it means that the plug-in feeds multi-tapped delays into EQ and diffusion blocks to create synthetic early reflections. A benefit of this particular algorithmic approach is that you get a wonderfully IR-like realism, but with full parameter adjustment, so the original System 6000 reverb worked really well for recreating believable ‘real’ spaces — in this respect, I’d say there are some parallels with what Bricasti achieved some years later. TC themselves offer their algorithms in plug-in form today, but several other companies have emulated the TC original, and the latest to do so are Relab, with their new VSR REV6000 plug-in. They’ve produced an emulation designed to capture the specific character of the original hardware, which itself was used in numerous film soundtracks and music scores, and it’s essentially a stereo distillation of the surround algorithms that ran on the System 6000. This plug-in supports all the mainstream Mac/Windows formats, including AAX, and authorisation is via an iLok account. The effects available range from small room ambiences and studio rooms to large halls and cathedrals, all of which add a real sense of dimension to the sound being treated. The documentation informs us that while In Advanced and Time/ Freq views, the upper section changes to show different reverb parameters.

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The default Main view.

typical reverb plug-ins allow control over individual parameters, each affecting only one aspect of the sound, the VSR REV6000 takes more of a ‘macro’ approach — a single adjustment might change numerous parameters behind the scenes. This makes the VSR REV6000 easy to navigate while still producing top-tier results. As expected there are numerous factory presets, and they make excellent starting points for your own adjustments, as the sense of space and location that they conjure up is seriously impressive. The GUI is arranged as three tabbed views, the first being Main, which hosts the preset name, tonal colour control over the early reflections and the main reverb parameters: Decay, Size, Pre-delay

and Hi-cut. Metering to the right shows the input and output levels, and there’s additional metering to the left for each of the Dry, ER and Rev levels. Large faders at the bottom of the screen adjust the Dry Level, Early Reflections Level, Reverb Level, Lo Colour, Hi Colour and Decrease, the last of which controls the amount of higher-order reflections. Advanced view keeps the faders at the bottom of the screen but replaces the top half with access to early reflections start and stop time adjustment, reverb type and size, modulation and output format selection, with independent delay time adjustment for the output channels. The third tab, which again retains the faders, allows you to adjust the EQ and damping values, as well as the reverb time (in four frequency bands with user-adjustable crossover points). There are options to process left and right channels differently, and each emulated location incorporates different filtering and reverb settings — a club room, for example, sounds very different from a concert hall, even if you dial in the same settings. To summarise, Relab’s VSR REV6000 excels at emulating real spaces, from the very small to the very large — the sounds it offers really are excellent. Of course, sometimes you might want a less ‘real’, more characterful-sounding reverb for your music project, so it’s worth me mentioning that Relab offer a range of emulations of other algorithmic reverbs too, and all those I’ve tried have sounded great. But if you need to be able to dial in a real-sounding space, and would like more flexibility and control than is possible with impulse responses alone, then VSR REV6000 would be well worth a demo: it does a first-rate job at a very sensible price. Paul White $ $149 W https://relabdevelopment.com

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ME‑Geithain MO-1 MKII

Active Monitors

Designed for outside broadcast trucks, Geithain’s compact monitors could easily find a home in music recording contexts where space is at a premium. PHIL WARD

here’s always been a need for ultra-compact monitors, particularly in the world of professional broadcasting. Perhaps the best‑known example is the BBC‑designed LS3/5A, born in the mid 1970s specifically for the cramped environment of outside broadcast (OB) trucks. Whenever the BBC broadcast a live event, from sports events to classical concerts, there’d be a pair of LS3/5As in the truck. But the LS3/5A was a particularly BBC phenomenon, and national broadcast companies all over the world have had, and do have, their own favourite monitors. In Germany, monitor brand ME-Geithain, a company that can trace their roots back to the 1960s, have long been a supplier to broadcast organisations, and their MO-1 MKII model, the subject of this review, is the archetypal ultra-compact broadcast monitor. While reviewing an overtly broadcast monitor might seem slightly off-piste for the pages of Sound On Sound, the worlds of music recording, composition, production, mix and sound design can also call for compact monitoring, so there often is a role for products of this ilk. You’ve perhaps also noticed that the MO-1 on review here is the MKII, but I was additionally supplied with a pair of MKIs to compare and contrast. I’ll cover some of the differences between the MKI and MKII later on. The MO-1 MKII really is pretty small, with external

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dimensions of 214 x 147 x 192mm (HWD) enclosing an internal volume of just 3.5 litres or so. Its enclosure is equipped on the side panels with tapped holes intended for the attachment of mounting hardware and, along with installation in compact studio spaces, I’d imagine ME-Geithain have in mind compact multi‑channel (Atmos) applications. On the front panel, behind a perforated metal grille, is a nominally

100mm‑diameter bass/mid driver with a concentrically located 19mm dome tweeter. Rather than being mounted at the apex of the bass/mid driver with the diaphragm taking on a secondary role of tweeter waveguide, as is the case with the majority of dual-concentric driver arrays, the MO-1 MKII tweeter is located further forward, approximately on the plane of the bass/mid driver surround. Mounting the tweeter on the bass/mid driver forward plane has the potential advantage of less modulation distortion appearing from the movement of the bass/ mid diaphragm (which in a small speaker like this could be significant), but this style of concentric driver array is nevertheless relatively unusual these days, and is lower-tech and less sophisticated than some other possible concentric driver architectures. For example, tweeters mounted at the bass/mid diaphragm apex are coincident as well as concentric: they are at the same position in space

ME-Geithain MO-1 MKII €2270 pros • Truly compact. • Genuine monitor-worthy performance. • Versatile connection and configuration options. • Dante-compatible.

cons • Slight midrange coloration. • Relatively limited maximum volume.

summary The MO‑1 MKII belies its tiny dimensions to offer some genuine high-performance monitoring quality.

and not just on the same axis, so their time‑domain integration is potentially optimal. Furthermore, a tweeter located in front of the bass/mid driver can constitute a significant diffraction body at midrange frequencies and will potentially result in audible artefacts. But despite the potential disadvantages, the engineers at ME-Geithain have employed this technique for many years and will undoubtedly have developed much technical expertise in optimising its behaviour and performance.

Great & Small Somewhat surprisingly considering its diminutive dimensions, the MO-1 MKII is a closed‑box speaker, and so eschews the potential power handling and maximum level gains of reflex loading. As I’ve described many times in these pages, closed‑box loading typically offers time‑domain benefits over reflex loading, but when it comes to very small monitors where power handling, distortion and maximum volume level are at a premium, the arguments in favour of reflex loading prevail for the vast majority of monitor manufacturers. ME-Geithain may well have concluded that the difficulty of fitting a port of adequate diameter in such a small box swings the argument in favour of keeping the box closed. This is a perfectly reasonable decision, but it does mean that the bass driver (and amplifier) has to work harder. While there’s nothing in ME-Geithain’s description or the technical specification of the MO-1 MKII to suggest that significant measures have been taken in the engineering of the bass/ midrange to provide extra low‑frequency power handling, the specification does list a very healthy maximum SPL of 103‑108 dB at 1m from 150Hz upwards. The range in the specification reflects the fact that

Unusually, the MO-1 MKII features two analogue inputs, which sum internally, allowing you to monitor stereo signals in mono. On the digital side, two Ethernet ports provide Dante connectivity and remote control over the speaker’s DSP. The five EQ settings can be selected using the Preset button.

the MO-1 MKII is equipped with optional low‑frequency EQ profiles, some of which reduce low‑frequency bandwidth and consequently will enable increased maximum level.

Round The Back The rear panel is a relatively crowded place, and this is not only because it doesn’t offer much real estate; it’s also because the monitor is unusually blessed with connection and configuration facilities. Firstly, along with the expected IEC mains input and power switch, the MO-1

MKII unexpectedly offers two balanced XLR analogue inputs. The two inputs can be selected individually using a rear push button, but can also be employed simultaneously to enable a single MO-1 MKII to offer mono monitoring of a stereo signal. Amplification comprises Class‑D amplifier stages for the bass/mid driver and tweeter, each rated at 95 Watts, and the crossover frequency is 3.5kHz. Located between the input sockets is an LED array that indicates signal presence and clipping, and a rotary knob that adjusts input sensitivity by ±6dB. The www.soundonsound.com / March 2025

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Softube Console 1 Fader MkIII DAW Controller Softube’s Console 1 system reaches its third iteration, refining and improving the popular DAW controller. RORY DOW

n our April 2024 issue, Sam Inglis reviewed the new Console 1 MkIII, Softube’s channel strip DAW controller designed for their Console 1 plug-in ecosystem and selected third‑party partners like UAD and FabFilter. Those familiar with the MkI and MkII versions will know that the mixing system comes with two units, the Console 1 Channel and the Console 1 Fader. Unfortunately,

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the MkIII Fader was not ready for Sam’s initial review. The two are meant to work as a unified system where the Fader unit gives basic mixing control over 10 mixer channels, and the Channel unit allows you to get into the details of EQ, compression, overdrive and other effects on the currently selected channel. If you’re interested in the system, start with Sam’s initial Console 1 Channel review (www.soundonsound.com/reviews/ softube-console-1-mkiii) and return here to learn more about the Fader unit.

Pre-Fade The Console 1 Fader uses the same dimensions and case design as the Console 1 Channel. Build quality is exceptional, and together they make a fine-looking pair. Softube even sell a smart-looking wooden stand that will

mount both units together. Alternatively, you can purchase a rackmounting kit or mount them using a VESA stand. There are 10 100mm motorised faders, 10 OLED screens, a single high-precision encoder, and plenty of push buttons for mute, solo, mode switching, channel selection, and more. The encoder is the same type found on the Channel unit and is used for panning the selected track and menu selections when needed. Some people may miss the lack of a dedicated pan encoder for every channel, but I didn’t find it a problem. With a click of the Shift button, the encoder doubles as a stereo width control, as with the MkII. The touch-sensitive motorised faders feel very high quality and include optional haptic feedback, which is new to the MkIII. For example, when you pass through unity, the fader will hit a small

buttons to the left of the faders. Volume and Pan modes control the volume and pan directly in your DAW, and Sends 1-6 allow you to control the first six sends (there is no way to access a seventh send or beyond). Param mode will enable you to control any parameter in the Console 1 plug-in, which is a nice way to access a single parameter across multiple channels. I particularly enjoyed assigning the Character and Drive parameters to faders, which allows you to mix using overdrive instead of volume.

Fade In

resistance spot. The clever thing about the haptic system is that it is entirely software-controlled, which means it can change positions on the fader or even be disabled altogether. For example, if you have the faders in Pan mode, the haptic feedback will be in the centre of the fader instead of where the 0dB mark would be. There is even a handy option to vibrate the fader when a channel is clipping. The screens above each channel show a variety of information depending on context. In normal fader mode, they show channel name, pan position, volume, VU meter and mute/solo status. The faders can be put into nine different modes using the six Fader Mode

Like the Channel unit, the Fader unit is powered by USB-C. Two recessed USB-C ports allow you to daisy-chain up to four units and a Channel unit to a single computer USB port. The Console 1 system needs custom support from DAW manufacturers. As a result, not all DAWs have the same level of support, and some features may work for some and not others. Softube are constantly working with various manufacturers to improve this, and full or partial support is now available for most of the big-name DAWS on the market. As of this review, Pro Tools suffers from a reduced feature count compared to other DAWs but check before purchasing. Each channel has a Select button. When you click it, this channel becomes the active channel, and if you have a Console 1 Channel unit, that will refresh to show the settings for the current track. If your DAW supports it (not all do), the Select button LED and the OLED screen will adopt the track colour set in the DAW, which helps with navigation. If you have a lot of tracks, you can move backwards and forwards in groups of 10 tracks or one track at a time. Alternatively, if you select a track in your DAW, the Console 1 units will update to show you that track and select it. The Console 1 system depends on having the Console 1 plug-in (free with the purchase of either unit) inserted at the end of the plug-in chain on every track of your DAW project. Tracks that don’t have the plug-in inserted will not

Softube Console 1 Fader MkIII $999 pros • Ten quiet and precise motorised faders with haptic feedback. • Beautiful build quality. • Softube’s software integration ‘just works’.

cons • The price is notably higher than the MkII. • Pro Tools lags behind in support.

summary Softube have taken their Console 1 system to a higher level with the MkIII. Like the Channel unit, the Console 1 Fader MkIII is a beautifully built software controller that keeps improving.

appear on the unit. In a couple of the DAWs I tried, there was a blank channel on the Fader unit for every track that didn’t have the plug-in inserted. I’m not sure if this is common to all DAWs, but it seems a shame to have a wasted fader if you don’t need control over all tracks in the project. For those who don’t want to buy into the Softube plug-in system or have a DAW that isn’t supported by Softube, Console 1 Fader also supports Mackie Control and HUI protocols. MCU and HUI only support up to eight faders, so two faders will be redundant, and the encoder won’t work either. You’ll also be missing out on many of the more integrated features of the Console 1 system. Universal Audio Apollo interface owners will welcome a function mode that allows the MkIII Fader to control the UAD Console software. I didn’t have a UAD interface to test it with, but this could be tremendously useful if you use the Apollo’s low-latency digital mixer a lot. A fifth mode also allows the unit to function as a MIDI CC controller. Three banks of 10 faders and associated mute, solo and select buttons send out unique MIDI CCs, which is extremely handy for MIDI automation of any kind. These modes can all be switched on the fly using the hardware.

The twin USB‑C ports are cleverly recessed so you can keep your cables nice and tidy when daisy-chaining units.

www.soundonsound.com / March 2025

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SOF TUBE CONSOLE 1 FADER MKIII

The Console 1 plug-in itself is remarkable. It is both a plug-in and a plug-in host. It behaves as a channel strip where you can swap out components, for example, an EQ or compressor, for another model. I refer you to Sam’s Console 1 Channel review for the gritty detail, but I will say that despite doing tape emulation, filtering, compression (twice), EQ (twice), overdrive, and all the utility stuff like volume panning and phase control, it is incredibly light on CPU. Of course, if you swap out some of the components for Softube’s more demanding plug-ins or some supported UAD or FabFilter plug-ins, your mileage may vary, but with modern CPUs, this is less of a problem than you might imagine.

Fade Out

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background? In low light, much of the unit’s labelling is unreadable. Some of the screen text also suffers legibility problems, particularly in menus where the font is so tiny that I was forced to squint even when wearing reading glasses. These are minor complaints and should not detract from the MkIII fader being one of the best fader controllers on the market. The motorised faders are precise and quiet, and the whole package feels premium. The software integration was flawless during my testing period. You have to accept that Softube’s Console plug-in will be on every track of your DAW project, but the payoff is a system that feels tightly integrated and reliable. The Fader unit, on its own, will not take full advantage of the software integration, but together with a Channel unit, they make a powerhouse of a control system. As a side note, acknowledgement and praise should go to Softube for their work implementing accessibility features in the Console 1 system. These include a suite of options for screen reader users (Voice Over on macOS and Narrator on Windows). It is good to see companies supporting people with vision impairment. Credit where it’s due. The Softube Console 1 system is 10 years old, and the MkIII Console and Fader combination has matured into a class-leading controller suite. It is evident that Softube care about their product and are listening to customer feedback. Incremental updates and clever hardware revisions have helped the system develop into one of the market’s most mature and premium DAW controllers. There is a price tag to match, but you get what you pay for. Compared to the MkII units, the MkIII have better build quality, nicer faders, rock-solid encoders, and a much deeper integration with software.

“The MkIII Console and Fader combination has matured into a class-leading controller suite.”

Much of the value within the Console 1 system is in the direct software plug-in control and the library of plug-ins that Softube have developed over the years. They have an impressive library of channel strips, compressors, EQs, mastering tools, and more, many of which are compatible with the Console 1. Some big names they have licensed for their emulations include Solid State Logic, Weiss Engineering, Chandler Limited, Empirical Labs and Summit Audio. If you buy the Console 1 Fader or Channel unit, it comes with a ‘Core Mixing Suite’, which gives you a nice selection of EQs, compressors and saturators to use with your hardware, but buying more could be addictive, not to mention costly.

Softube sell optional wooden end-cheeks for mounting the Console and Fader together. Rack ears are also available.

The MkIII Fader is not the main star in controlling channel strip parameters other than volume and panning. That is the role of the MkIII Channel unit. You can use the Fader’s Param mode to access a single parameter on the faders, but attempting to make all your edits this way would be frustrating. Even though the Core Mixing Suite is included with a Fader purchase, it doesn’t make much sense without the Channel unit. The Console plug-in has been designed with hardware control in mind. Controlling it with a mouse felt clunky, and it would be a waste to insert the plug-in on every track simply to give you control over volume and panning.

The sensible purchase path, at least in my eyes, is to get the Channel unit first, and if you gel with that but feel you are missing faders, get the Fader unit. One thing I would love to see added to the Console 1 hardware is transport controls. During mixing sessions, I often had to divert my attention unnecessarily back to the mouse and keyboard to start, stop and navigate a project, which seems a shame given how good Console 1 is at keeping you away from the computer for every other mixing task. Now to a minor gripe: When will manufacturers learn not to put dark-coloured text on a dark-coloured

$ $999 W www.softube.com

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Torso Electronics S-4 Sampler

Torso’s ambitious ‘sculpting sampler’ may be a work in progress, but it promises to be something very special indeed. PAUL NAGLE

orso Electronics are a young Danish company with a preference for user interfaces that minimise the gap between ideas and realisation. Their first product, the T-1 sequencer, has the hands-on directness of a performance instrument and for the second, the S-4, a similar philosophy is applied to audio capture and processing. The S-4 is described, rather mysteriously, as a ‘sculpting sampler’. Our mission today is to discover what that means — and then try to decide if it’s a creation worthy of Michelangelo, or the product of just another bloke with a chisel...

So What Is It? Listing the current functionality feels like a safe enough way to kick off. S-4 workflow

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is based upon four stereo tracks, each of which can act as a virtual tape recorder, a polyphonic sample player or a gateway for incoming audio. The tracks are then subjected to a series of ‘sculpting’ options — currently a granular effect, a morphing resonator, several flavours of distortion and a combined reverb and delay. Additionally, the S-4 can act as a USB audio interface — one that’s capable of expanding its I/O automatically whenever a class-compliant device with extra channels is attached. Lastly, for instant results in any environment, it features a built-in microphone. Onboard storage (4GB) is provided for samples and projects, plus there’s a USB-C connector to enable file transfer from a PC, Mac or peripherals such as external drives. Arguably, the only omission is an SD card slot, or a similar means of boosting the storage unobtrusively, because, if my

experiments are anything to go by, that 4GB will fill up pretty darn quickly. Internally the S-4 sports a — now ubiquitous — Raspberry Pi with a quad‑core 1.5GHz processor. Audio is a clean 24-bit, 48kHz throughout and, while the stereo audio inputs and outputs are on quarter-inch jacks, all the other connections are 3.5mm, including the headphone socket, MIDI in and out, plus the analogue clock I/O. A single adaptor (Type A) is supplied to turn mini‑jack MIDI into the regular five-pin type and the power adaptor ships with a selection of country options; you can pack them away in readiness for those sunlit days when touring is viable once more. As hardware devices go, it’s a petite package measuring a mere 156 x 242mm and standing just 39mm from its rubbery feet to the tips of its encoders. The RGB buttons are chunky-feeling and those

Torso Electronics S-4 $899 pros • An integrated audio environment; its four tracks offer a variety of applications. • The virtual tape recorder is the star of the show. • Includes sample playback and audio processing capabilities. • Its granular processor and morphing resonator are excellent. • The user interface is a pleasure to use. • As is the modulation system.

cons • Some functions are incomplete or not yet implemented. • It’s not been a bug-free journey to date. • The onboard storage isn’t generous — I’ve filled half of it during the review period. • All things considered, not cheap.

summary The S-4 is small but easy to use and powerful without being overstocked. It can mangle audio on one track, perform sample playback on another and still have two left for tape-style looping — the division of labour is entirely your choice. If Torso continue to chip away at its rough edges, the S-4 could prove to be a masterpiece.

encoders map to objects on the 3.5-inch pin-sharp LCD in ways Elektron users will instantly recognise. Admittedly, the encoders don’t handle quite as nicely as those of an Elektron but my main misgiving is their black on black aesthetic — this design choice feels needlessly obtuse, at least in my darkened studio. That said, my fingers eventually learned their way around without troubling my brain too much.

Sculpting Time On power-up you’re greeted by an empty project and so immediately have choices to make — such as what role(s) the S-4 should perform today. Hopefully in some future update there will be an option to start from your own defaults, or from where you left off previously, but for now you must either load a project manually or begin with Torso’s preferred settings. Incidentally, stored projects appear in

a simple list, sorted alphabetically — and it’s not hard to predict this will soon become long and unwieldy. Beneath the encoders are buttons representing the chain of audio devices available. From left to right, these are: Material, Granular, Filter, Color and Space. All can be bypassed if necessary but, at present, Material is the only one to offer multiple modes — Tape or Poly. When Material is bypassed, that track is ready to receive incoming audio. The S-4 boots as a virtual four-track tape machine with an interface that’s so friendly it invites you to begin recording and overdubbing right away. However, you quickly realise there’s no indication of how long you’ve been going — or how much time you’ve got left. Further investigation revealed that a maximum of six minutes is allocated per track, but some kind of time or bar counter visible during recording would be very much appreciated. Recordings are based on a global tempo value — either sourced internally or taken from your selected input (MIDI or analogue). Synchronisation arrived quite late (version 1.1) with further improvements and hotfixes following in 1.2.x. It’s perfectly straightforward to adjust the start and end of each reel on the fly, shift the audio relative to other tracks or chop to precise bar intervals when a perfectly sync’ed loop is required. That said, I’d love to see a zoom function implemented so you could more easily fine-tune those starts and ends when not working to bar/beat divisions. Actually, it’s worth a quick mention of the way units are employed throughout the S-4, because it’s rather neatly done and superior to most audio looping/recording devices I’ve used. For any Tape track, pressing and turning the Tempo encoder presents you with a selection of either Sync, Free or Stretch modes. Sync locks you to the current bpm and all parameters duly reflect this (in beats and bars). Free, on the other hand, acts more like a regular tape recording, with the units for start and recorded length specified in seconds. Finally, Stretch is unique to Tape tracks; it separates pitch and time, instantly lending itself to more granular applications. Unusually, you can override the speed and tempo at track level, but the implementation is not without its peculiarities. For example, and counterintuitively, if you increase the track’s tempo while bpm-sync’ed, the pitch drops (and vice versa). Still, with speed modifications of up to three times (faster

or slower) and options such as scale quantisation, freeze and reverse playback, there are oodles of ways to interact with those virtual reels. A second press of the Material button reveals yet more parameters — Glide, Density, Level and SOS. Glide smooths the transition between tape speed changes, while Level is handy for boosting a recording once it’s been laid down. This is especially important because there’s no gain control provided for the line inputs — hopefully another something to be added when time permits. SOS determines the amount of existing signal retained when overdubbing, so vital for tape loops you want to fade gradually on each pass. And, as you’d expect, dropping out of overdub at any time maintains the current level. When in Stretch mode, Tape reveals its underlying granular nature with the Density parameter controlling both grain size and rate. If you engage Freeze and play around with density, shift and pitch, many happy hours may be lost disrupting the fabric of reality. With four tracks to play with, the Track button is a helpful entry point for selection and for checking which audio devices are currently enabled. It also contains four individual send levels, used to pipe the current track’s output into any of the others, or back into itself. This is a rare level of connectivity and brimming with routing possibilities, some of which we’ll consider later.

Modulation Before moving on, I should point out the Mod button which, if you’re as loose-fingered as I am, you might have already brushed accidentally. Modulation is a simple four‑slot affair, colour-coded on the display for ease of recognition and offering Random, LFO or an ADSR envelope (manually or MIDI-triggered) as potential sources. In use, this is an elegant means of adding movement to parameters on any page; as an extra bonus, the display gives real-time updates of the modulation in action. The main omission is some form of envelope follower to harness the contours of incoming audio but, according to a recently published roadmap, this is on the cards for the next major update (1.3). Each modulator is loaded with numerous options to skew, smooth or distort its basic shape and there’s a choice between sync’ed and unsync’ed rates. On the rare occasions when the sources provided are insufficient, you’ll be pleased www.soundonsound.com / March 2025

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DivKid & Making Sound Machines DivSkip Eurorack Module

ivSkip is a four-channel trigger machine dressed in a dazzling festival of lights. It bombards you with the flashing possibilities and colourful trigger patterns of an all-night rave. The functionality is based around the tilting and balancing of two outcomes: A or B. Through multiple modes and a handful of mathematicians we get to craft fascinating patterns from Bernoulli coin tosses, Turing algorithms and Euclidean rhythms. They are immediately useful as a source of probability-led drum triggers but can also ramp in ratchets, patterns and retriggers in response to stimulation. The workflow is nicely straightforward. Each channel has a trigger input and two resultant trigger outputs, A and B. Between input and output is a bedazzled knob that guards the entrance to the party. The knob can be CV controlled via the slightly confusingly named Probability input, but it makes sense when you use it. There’s a global trigger/gate switch where trigger mode sends a pulse on every event, and gate mode changes from high to low. There are eight modes in total. To choose your mode, you hold down the clickable knob for more than a second, and the halo of 16 LEDs changes to a multi-coloured array where each one represents a selectable mode. The first mode is Bernoulli Gate, which tosses a coin to decide whether output A or B gets the trigger. It’s fabulous for all sorts of things: kicks and snare, open and closed hi-hats, competing filter patterns, or simply

March 2025 / www.soundonsound.com

to add a bit of probability to a stream of triggers. Turning the knob weights the toss towards either outcome. Mode 2 is a useful clock divider and Mode 3 mimics the classic Music Thing Turing Machine’s random generation. Modes 4 and 5 are variations of Euclidean patterns. ‘Euclidean split’ distributes three triggers for both sides in the space set by the knob. With Euclidean Classic, output A gets a choice of 32 classic distributed rhythm patterns, while output B is the inverse. Modes 6 and 7 do interesting things with the incoming trigger rather than seeing them as a clock. In Ramp mode, the DivSkip causes 16 triggers to occur in response to an input trigger. These will either speed up over time, speed up then slow down, or slow down, making wonderful cascades of events reminiscent of twanging a ruler on a desk. Retrigger mode is mild in comparison, giving you a ratchet-style burst of triggers. The last mode turns DivSkip into a pre-programmed four-channel drum machine, where each channel outputs one of 24 patterns designed for either kick/snare, hi-hats or percussion. One very cool feature is that you can mute any channel by pushing down on its knob. There’s also a switch for a second mute mode so you can switch between two different mute configurations: superb in live performance. Each channel is a party of functions that offer myriad ways of handling two outcomes. The patching possibilities that come from this one module are very exciting, and I keep coming up with new ways that one or more channels can drive a patch. At all times, the bright LEDs visualise what’s going on, combining the dance of the halos with the flashing of the A/B indicators. It can initially be bewildering as you take your first skips into the functions, but very quickly DivSkip becomes a solid dance partner. Robin Vincent $

$249.99

www.makingsoundmachines.com

Patching Panda Blast & Hatz V3 Eurorack Modules

ew from Patching Panda are a pair of 10HP percussion modules, each offering a wide range of traditional sounds, but with some serious experimental potential as well. The Blast is apparently the result of two years of research into what makes the perfect kick drum. Luis from PP’s stated mission is that the Blast will give you the tools to precisely design the right kick sound for your track, which by his definition means it will have sufficient punch to cut through a mix without sacrificing body or fullness, and that it will blend well across different genres. Tools are certainly not lacking. The usual Tune and Decay controls are augmented by sliders for Pitch Decay (with adjacent Amount control), Body, Shape (which folds the fundamental sine wave), Compression and Through-zero FM. Modulation inputs control much of the above, but there are also Accent and Amplitude Modulation inputs and outputs for +/- Envelope shape. These last are a nice touch — the negative envelope is perfect for ducking a track around the kick — and invite‑cross patching, say positive envelope to the Shape or TZFM inputs. So there are plenty of tools — perhaps too many for some. If you’re looking for straightforward TR-style sounds with the usual 808/909‑type controls then what the Blast offers may be a bit overwhelming. But if you’re serious about kick drums — and many people are — then there’s an awful lot to explore here, and the Blast sounds good. Really good. It’s capable of a huge range of analogue kick sounds, from a polite ‘phut’ to the kind of noises that will worry structural engineers, with a whole spectrum in between, and all, as Luis intended, precisely controllable. And while useful

for distortion or grit, the TZFM and AM inputs can take it in another direction completely, adding complex harmonics and the possibility of more metallic sounds. In fact you could almost see the Blast as a simple complex oscillator, if that’s not a contradiction in terms. The controls are well judged: the decay reduces from a ringing sustain to an abbreviated click and the interplay between Body and Comp can be subtle and nuanced, while the combination of Pitch Decay + Amount can take you all the way from assertive initial transient to Star Wars laser sounds. All told, it’s clear that a lot of thought has gone into the Blast. If you want to go deep designing kicks it can do that — and much more besides... Alongside the Blast we have the third version of the Hatz. Conceptually, the Hatz is a similar kettle of fish to the Blast — a toolkit for conventional and unconventional hi-hat sounds. It has open and closed hats, each with their own trigger and controls for Decay (hallelujah!), Frequency and envelope decay curve, and shared controls for ‘Metals’ and ‘Texture’, which balance the metallic and noise

Morphor Echo Eurorack Module

id you know that the term ‘bucket brigade’ comes from a style of fire fighting, in which buckets of water would be passed down a line? Me neither. It makes perfect sense, though, when you think of what a bucket brigade device, or BBD, actually does: sequentially ‘filling’ and ‘emptying’ a series of capacitors with a signal. “We love BBD circuits for their imperfection,” say Morphor on their website. ‘Imperfection’ is certainly the word — in essence, imagine those firefighters as simply being rather... sloppy. But needless to say, that’s what we love about them. Now Morphor enter the BBD delay arena with the Echo, a 4096-stage bucket‑brigade-based module in capacious 20HP. With delay times ranging from a super-quick 10ms to

elements inherent in a synthesized hi-hat. There are individual modulation inputs for Frequency and shared inputs for Texture, Decay and Hold alongside an Accent input and a switch that turns choke on and off. Hold is an interesting one; a voltage here will sustain the sound indefinitely, which is good for adding interest to a pattern and equally good for off-piste noise exploration adventures. The sort of thing you never realised you needed before you tried it. Like the Blast, a considerable amount of thought has gone into the Hatz — two previous versions, in fact — and once more the controls and modulation are very well conceived. The decay can be very snappy indeed — more clicks than hats — which is always a good option to have, and the Metal and Texture controls can take you out of hi-hat country completely, especially when combined with Hold. The hats are very ‘electronic’ — more 606 than 808 — but there’s a lot of potential within this sphere (are hi-hats spherical?) and the option to move into experimental noise and texture is always there. It’s easy to achieve a sort of shimmering digital quality that would be at home on a Mark Fell or

a languid 2.4 seconds, the Echo is a proper piece of analogue outboard goodness, with a raft of extra features and options thrown into the mix. The module’s full name, ‘Echo Analogue Quad Tap BBD Stereo Delay’, refers to its impressive array of eight different outputs for various applications of its four taps. Those four taps quarter the set delay time: if that is set to two seconds, for instance, tap 1 will be half a second, tap 2 one second, tap three one and a half seconds and tap 4 two seconds. While the Echo isn’t the first delay module to offer discrete tap outputs (Xaoc Devices’ Sarajewo and Doepfer’s A-188-2 are two examples that spring to mind), it does arrange its outputs in a distinctive way: taps 1, 2, 3 and 4 get discrete mono outputs, but there’s also a stereo mix output for tap 4 and an overall stereo mix output for all taps together.

Gábor Lázár record, and in the unlikely event that Ryoji Ikeda should ask you to recommend some Eurorack percussion, the Hatz would likely be a safe bet. Both of these modules offer useful conventional percussion sounds, especially the versatile Blast, and would be worthy additions to any drum rack, but it’s the additional scope for experimentation that really makes them special. David Glasper $

Blast $239.99, Hatz V3 $199.99.

www.patchingpanda.com

Not only is it therefore possible to send individual stages of the delay line to all manner of destinations; the final two sets of stereo outputs involve the Echo’s wet/ dry mix, stereo spread and panning mode controls. These modes consist of Ping Pong, Traverse and Chaos: Ping Pong alternates repeats across the stereo image, Traverse progressively pans them and Chaos places them at random. The delay itself sounds as smooth and warm as I could hope it to, though fair to say I would perhaps have liked the input to offer a little more gain before things start to clip. As it is, I did need to ‘work’ combinations of the input gain, wet/dry mix and input level on my

mixer a bit to achieve a really healthy gain structure in various instances, but fair to say once in a sweet spot the Echo performs marvellously. One predictable challenge with any BBD-based design is these devices’ inherent tonality and noise. Longer delay times, for instance, will have a duller tonal character, which must be balanced with what can be

www.soundonsound.com / March 2025

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a fairly prominent noise floor. To this end the Echo places a choice of two low-pass filter modes in the delay line: early-tracking or late-tracking. These ostensibly amount to different cutoff settings: early-tracking is less noisy but darker in tone, while late-tracking risks more residual noise but affords a tonally richer signal. The Echo felt most alive and exciting to me with pinging, shorter delays, particularly when sent skittering across the stereo image, and from here it’s not a leap to the obvious point that such things can be used for far more than just a delay effect. A variety of lovely analogue chorus sounds are quick and easy to achieve, from subtle to extreme, and I occasionally found myself

Modular Profile: Sasha Lewis

he more you talk to producer and sound designer Sasha Lewis, the more you learn. Atop his role as one half of the wildly off-kilter Unicorn Ship Explosion along with jazz drummer Rob Pemberton, Lewis has honed his craft working with the likes of Four Tet and Jon Hopkins, and has provided sound design for numerous film companies. And did I mention he’s the founder of one Black Cloud Tea company? “The joy of music is a gut feeling that is beyond your control,” he’s been quoted saying, “and for me, tea is similar.” On his entry into modular Like many people who are into modular, I’d imagine, I’m quite obsessive. And at some point in my sound career and synth obsession I discovered Doepfer. This was back in the early 2000s when the synth market was very different to what it is now, with much less choice and fewer abstract options. Doepfer felt like a raw, exciting and adventurous sonic sandpit. The early Doepfer website added to the vibe that you were buying into a strange and niche weirdo club. In those

in harmonically rich Karplus‑Strong territory. With Sync mode activated, the delay time snaps to different clock divisions when changed or modulated. This not only makes for fun polyrhythms; I also found myself creating some gorgeous, pseudo-quantised melodic patterns from the simplest input signal, since changing the delay time of course also affects pitch. A few other functions add further excitement: there’s a switch to invert the polarity of the wet signal so that sonically it interacts differently with the dry, and also a tempo output, which fires useful clock pulses in time with the delay time setting. More off-kilter is an additional input for breaking the internal delay feedback

days modular seemed an affordable way to explore wild sounds compared to buying ready made synths. I remember thinking the combination of a Doepfer BBD delay paired with a Bit Crusher was the most fun you could have in a recording studio. On his go-to modules I’m not sure if I have a ‘go-to’ module. My interest is always changing with my weak and short attention span. My longest owned module is the Doepfer A-119 [External Input/Envelope Follower], so maybe that’s my most used and needed one. I enjoy interfacing the modular with the real world. Sending electronic sounds through the atmosphere makes the world a better place. It’s such a simple module but I couldn’t not have it, I use it all the time. I also really love my Rossum Assimil8or. The pitch‑shifting on it is pure juicy futuristic quality. But I don’t feel wedded to it like I do with the Doepfer. On Unicorn Ship Explosion Unicorn Ship Explosion might be about entertaining yourself and not knowing what happens next. Risking failure in front of an audience. It’s fun — for us! Every gig we try and work out our relationship to our instruments... at least one of us is a competent musician,

March 2025 / www.soundonsound.com

path with an external signal. This made for some pleasingly wild, multi-input delay chaos, to say the least, and while it wasn’t hugely useful on a fundamental level, it is a welcome addition all the same. The Echo doesn’t give you instant results on a plate, but that’s the nature of the BBD beast. “We love adding features that make people think,” says the developer, and to that end the Echo is an undoubted success, not to mention a worthy addition to the already formidable Morphor range. It’s got all the substance — and idiosyncrasies — of analogue, but it’s reliable everywhere it needs to be. I recommend it. William Stokes $

$429

www.morphor.io

Rob Pemberton. I use an ever-changing modular rig, which keeps it interesting for me. I enjoy making it a puzzle that I need to figure out in the moment. I have tried to use a regular synth for a gig, but I always get bored with the outcome, and I end up returning to the modular. Having said all that, we still haven’t figured out what we’re doing and why we do it, we just have to do it. On what’s next More music, more sounds. I flit between film sound, advertising jobs and music projects. Music for music’s sake is my preferred focus. I’m always working with different people as I love the unknown of a collaborative venture. Mixing your ideas with someone else’s is a wild journey, getting to watch your co-dependent mutation of an idea evolve. Every year I think: ‘This is the year I make something on my own,’ but I predictably prioritise collaborations over indulging in myself. Safety in numbers and all that... On the culture of modular Modular culture is odd; there’s nothing cool about it despite it occasionally trying to be. I think at the core of the culture are obsessives who are just trying to discover new sonic options in search of untapped

potentials. It swings from being ‘a massive waste of time’ to being ‘the forefront of new sonic experiences’. A bipolar instrument that can ruin your life. It’s one quarter train hobbyist culture, one eighth never‑ending consumerism, one fifth brainy electronic geek society, one seventh Panini‑sticker-collect-them-all co-operative and one third renegade sonic adventurers. It’s equally embarrassing and exciting at any given moment. As an instrument it makes the most sense in a Japanese Noise gig, whereas as a studio tool it has become the most expensive rabbit hole in the room. I once received the comment: it seems it’s all about having the right length of cables. How embarrassing. William Stokes

88C DUAL DESKTOP DYNAMICS

Iconic Sound Quality | Versatile Control | Legendary Character

Learn more by scanning the QR code or visit neve.com

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ORIGIN EFFECTS DELUXE55

Origin Effects Deluxe55 Analogue Amp Emulation Pedal

The latest amp-simulation pedal from Origin Effects recreates another much-loved classic tube guitar amp: the iconic Tweed 5E3 Fender Deluxe. DAV E LO C K WO O D

oused in a compact pedal enclosure with top-mounted jacks, and powered from an external 9V DC supply, Origin Effects’ ‘amp recreation’ Deluxe55 uses entirely analogue, solid-state components to replicate the function of every part of the target-amp’s circuitry — that means the preamp, phase inverter, push-pull power amp, output transformer, power supply, and a reactive load with an impedance curve, to make the output stage operate as if it were driving a speaker. Unlike the majority of digital amp modellers on the market, however, the Origin ‘amp’ pedals don’t include an integral speaker-and-mic emulation stage, and that’s because they’re designed to be used as conventional pedals into real guitar amps and speakers, as well as in DI recording situations.

Origin Effects Deluxe55 $339 pros • Sounds and ‘feels’ like a real 5E3 Deluxe. • Works well with real amps and as a DI solution. • All‑analogue circuitry. • Robust build quality. • Good‑quality, click-free buffered bypass.

cons • Replicates the loose bottom end of the real thing!

summary An outstanding, all-analogue, solid-state pedal rendering of a classic 6V6 tube amp circuit.

March 2025 / www.soundonsound.com

In the latter scenario, you’ll need a separate IR host, whether in software or in hardware form, from the likes of Two-Notes or Boss, or a speaker-modelling pedal such as the OX Stomp from UA. If you want to choose the IR route, Origin Effects do offer a very nice, free, downloadable library of impulse responses derived from the actual cabinets used in the design of their amp recreations, including a set for the Deluxe55. Speaking of recording, I created a couple of audio examples that demonstrate what this pedal can do, and you can find them on the SOS website at https://sosm.ag/ origin-effects-deluxe55.

The 5E3 Deluxe The mid-1950s 5E3 Deluxe is a significantly different circuit from the famous ’60s amp that also bore the ‘Deluxe’ part of the name. The latter, with its onboard reverb and vibrato, and a lossy tone stack that results in a mid-dipped voicing, was designed specifically to overcome what was seen within Fender at the time as the ‘shortcomings’ of the earlier model, specifically to make it better at projecting the clean, bright guitar tones of the era. But once distorted tones had become an integral part of the voicing of electric guitars, in the mid-60s and beyond, the limited headroom and midrange-heavy voicing of the older model started to find favour again among discerning players, especially for studio work. At only 12W, with a single 12-inch speaker housed in a small enclosure, a 5E3 will always struggle to keep up with

any band line-up featuring a real drum kit, in my experience, and the amount of loose bottom end and rectifier sag from the power supply will always make it feel underpowered compared with any modern amp. But those same characteristics are what make a Tweed Deluxe a beautiful studio amp, where you can get it into ‘singing’ sustain mode without too much volume, or avail yourself of its highly characterful cleaner tones, as well as everything in between.

All Under Control The four-control line up of the Deluxe55 — you have just Drive, Tone, Level and Post EQ to play with — reveals that Origin have decided not to replicate the dual-channel setup of a real 5E3, opting instead for just the Inst(rument) channel. It’s one of the quirks of the Deluxe circuit that you can plug into either input and turn up the Volume control for the other channel and get sound. The two Volumes will also interact: turning up the Mic channel whilst using the Inst channel will initially add some more warmth and midrange, up to about halfway, after which it will seem to suck out some midrange. There are some interesting tones to be had there, and

some people may be disappointed to not halfway and you can find everything be able to access them in the usual way. from warm cleans to singing lead under ALTERNATIVES One thing that nobody will miss, however, your fingertips, responding to picking Universal Audio’s digital Woodrow pedal perhaps offers a bit more versatility, is the taper of the Volume pots on the real pressure and volume setting, with the in having the tonal option of channel thing, which give you most of the output lovely natural compression of a 6V6 interaction and integral speaker emulation, as soon as you’ve turned them anywhere output stage working with a tube-rectified but Origin’s Deluxe55 is more adapted above zero! The Drive control on Origin’s power supply and relatively small output to working as a pedal with real amps. For Deluxe55 is sensibly calibrated to give transformer. I put Tone at halfway for recording, be sure to try the Origin ‘Tweed Combo’ IRs for the full effect — I couldn’t you some useful operating range. single coils and three-quarters for beat it with anything else. The single Tone control is not an active humbuckers and never feel the need stage, but simply rolls off treble as you to touch it again. That’s the thing about turn it counterclockwise, and affects the these older amp designs with simple though, and something that you work working of the ‘bright’ cap across the circuitry and no negative feedback, they around in how you play, which guitar volume control as you turn it up beyond instantly offer up so much of the classic you choose and the pickup you select. halfway, adding both treble and gain. guitar-tone ‘vocabulary’ that it simply It’s a common mod to reduce the large Like all bright caps, its effect diminishes never matters that there are hardly any coupling cap values to try to limit the ‘flub’ as the Drive pot is increased. Level is controls to play around with. You can in a real 5E3, but I find that compromises just an output volume control push the front end a little and has no effect on tonality, harder with a pedal for a bit while the Post EQ works with more drive without having the adjacent three-position to set the Deluxe55 itself switch to allow the pedal’s at ‘meltdown’, but with so voicing to be optimised for much midrange in the voicing different scenarios: ‘neutral’ in already, I find it better to the middle position, with the keep away from the ‘usual pot inactive, for DI use or with suspects’ (Tube Screamer, ‘flat’ amplification; or voiced Fulldrive et al), but things to work with either a mid-light that are a bit flatter like a plain Fender-type amp, or a mid-rich boost, or even something a bit Marshall type with the switch scooped like an OCD, can set in either direction. The also tend to accentuate the rotary control can then be loose bottom end. Personally, used to fine-tune the basic I like it best with the guitar response shape set by the connected directly, with mode switch. It works well at nothing in between. around unity gain feeding into When using the real the front end of amps set fairly thing, of course, the speaker, clean, as does the neutral the lightweight, resonant setting into the return input of pine cabinet and its floating an effects loop, where you can baffle, as well as a bit of use a bit more of the plentiful ‘volume-effect’, all play a part output volume available. in the experience, but the The Deluxe55 doesn’t include speaker emulation, but Origin offer a free set of IRs, including some taken using an original 5E3 Deluxe. A small 1x12 open-back cab Origin speaker IR really is with a vintage-style speaker rather good and combined will get you closest to the Tweed Deluxe with a bit of ‘faux room mic’ from a digital the beautiful cleans too much for it to experience, but there is still something reverb makes for a very comfortable DI be worthwhile. Another common mod nice to be had using it with bigger cabs, recording sound. is to substitute a 12AX7 for the 12AY7 too, provided you roll off some of the normally used as the preamp tube — yes, Verdict pedal’s loose bottom end at the amp. the preamp tube: there’s only one! It is replicated here as a switched option Origin’s Deluxe55 is another gem Uncannily Accurate and makes the amp louder and dirtier. from the makers of some of the very In 12AX7 mode with the Drive maxed best all-analogue amp recreations Paired with its matching speaker IR from you can get to full-on Neil Young ‘edge on the market. If you are looking Origin’s library and DI’ed for recording, of explosion’ tone... should you want to. for what a Tweed Deluxe can do I found the Deluxe55 to offer an uncannily Personally, I always prefer the Tweed in a pedal format, you won’t be accurate replication of everything that Deluxe with its 12AY7 tonal palette, for disappointed. If you are not, then be players love about a Tweed Deluxe, and both clean and driven sounds. careful about trying one — you just — perhaps attesting to the authenticity of A real 5E3 interacts beautifully with might find you’ve been converted! the circuit — some of the bits they don’t, different guitar volume-control settings such as the amount of bass ‘flub’ in any and the Deluxe55 replicates that strongly overdriven sounds. To me that’s $ $339. perfectly. Set the Drive somewhere above just part of the sound of a Tweed Deluxe, W https://origineffects.com www.soundonsound.com / March 2025

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BOSS VG-800

The Boss Tone Studio software makes editing presets and creating custom tunings a breeze.

Power comes in on the rear and a Boss 9V power supply is provided (it will also work with a pedalboard PSU that can deliver sufficient current). Here there are also two TRS jack sockets for the GK-5 input and output, another input for the normal guitar signal, send and return jacks for bringing your own pedals into the signal chain, two output jacks for mono/stereo/phones use, and an expander jack for external controls two and three, or an expression pedal. There are further connections on the left panel: mini TRS jacks for MIDI in and out, a USB‑C connector, and an Exp 2 jack for control/ expression four and five. As the GK-5 pickup doesn’t have a volume control on board, most live performance users will probably find it essential to add an expression pedal to control volume.

semi‑acoustic, Rickenbacker and so on. There are also settings for Wide Range pickup, Bright Humbucker and Fretless Bass. Speaking of which, a startup routine configures the VG-800 for use with bass guitars (by default it’s set up for guitars), after which the menu choices change accordingly. Also in the Guitar section are pickup selection options, Volume, Tone, Level, Normal Mix and Normal Level.

for setting up alternative tunings, I’d use the software, where the pitch controls for each string make this incredibly easy. I mentioned the Dual Mode, and this is new for the VG-800. It’s what allows for tricks such as having different instrument model or pickup settings on different strings, or having the change occur above a specified fret. When setting the fret positions, the crossover fret is the same for all of the strings (you can’t split one string on fret 7 and the other on fret 10, for example), but you can set upper and lower limits and this means you could set a region of overlap if you want to. Pressing Effects brings up a signal-path graphic, with editable and bypassable blocks for the effects and amplifiers. The Select knob then moves you through the blocks, with the controls of active blocks shown below. When there are more controls than can be accommodated on the screen, arrow buttons give you access to the additional parameters. Some effects types are fixed, while three slots are freely assignable. The fixed-function effects are Foot Volume, Compressor, Distortion, Amp, EQ, Noise Gate, Chorus, Delay 1 and Delay 2, and for each there are alternative types to choose from. For example, the Distortion block offers you 37 types of distortion, many modelled on classic pedals, running from clean boost via overdrive and distortion right up to fuzz and metal. A typical patch might start with a foot volume control, with settings for range and control curve (as it’s replacing the volume control on a guitar, it comes before the rest of the signal chain). If an effects block is greyed out that means it’s bypassed, in which case pressing Select will bring it to life. You can then use the FX Type encoder to navigate a huge range of effects that goes beyond the usual suspects to include things like Slow Gear, Ring Modulator, Slicer, De-fretter, and Sitar Emulator. Using the Tone Studio software, blocks in the routing chain can be moved and reconfigured simply by click-dragging. The routing supports the creation of two parallel paths, which can host further effects and amp/speaker models. When these paths are recombined, the signal could perhaps be followed by another foot volume and further effects. Routing can be created very easily by dragging blocks to new locations using the Tone Studio editor. The routing options also allow you to mix in some of the sound from the guitar’s

“If you’re left-handed or prefer the cable to come out at the bottom, the pickup direction can be swapped in the VG-800.”

Editing Patches As shipped, the VG-800 has 20 banks of three factory presets, which can be overwritten if you wish. A further 30 banks are available for user patches, though, and these can be created from scratch or modified from duplicates of the factory presets. I’ll run through the main controls and options below, but it’s all pretty intuitive. Selecting Inst takes you to the guitar/ instrument modelling section, and here you’ll find pages for Guitar, Alt Tunings and String/Others. In Guitar mode, the Type encoder goes through all the main guitar-world food groups, from Strats and Teles to Les Pauls (humbucker or P90), semi-acoustic (335-style), large‑body

March 2025 / www.soundonsound.com

The non-electric guitar emulations available in the Instrument section include Acoustic Guitar, Dual Guitar, E Bass and VIO Guitar, which alters the input signal’s envelope and harmonic characteristics to create a bowed type of effect. There’s also Synth, where options include the GR-300, Solo, Filter Bass, Crystal and Organ. They’re generally pretty good, and while the organ might perhaps not provide quite as convincing an emulation as, say, the EHX B9, when used in conjunction with the rotary speaker effect it still does a very effective job. The Alt Tune option can be configured as a virtual capo (ie. pitching the whole guitar up or down in semitone steps) or for custom tunings, and there’s also an auto harmony mode, in which the user sets the key and harmony interval. Meanwhile, the String/ Others options also allow for custom volume and pan settings for each string. Personally,

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Slate Digital Submerge Automatic Ducking & Filter Plug-in

If the classic side-chain pumping effect is your thing, Submerge can do that — and a whole lot more! M AT T H O U G H TO N

e’ve all heard these effects before: side-chaining a compressor or gate to create rhythmic ducking and chopping effects has been a popular technique for a good couple of decades now, particularly in hip‑hop and dance genres. But while you can achieve these effects with pretty much any compressor/gate that supports side-chaining — meaning you probably have the necessary tools in your DAW already — setting up the routing and timing to your satisfaction can take time. There are now a number of plug-ins available that can both remove that hassle and offer more features to help you shape the effect. The latest such plug-in to make it to the top of my review queue is Slate Digital’s Submerge, and I have to say that first impressions have been great. As I find so

March 2025 / www.soundonsound.com

often with Slate Digital’s plug-ins, it provides enough features to makes it useful and different from the competition without overcomplicating matters too much. As you can see from the screenshot, it has an elegant GUI that allows you to achieve most of what you’ll want to do most of the time, using only a few core controls. The expected effects are all present and correct too, but there are also some useful twists that give you access to some really fun, creative effects, particularly if you’re prepared to use a MIDI controller or your DAW’s automation system.

The Sub Way The GUI puts two main panes in between vertical input and output meters and gain/level controls. The first section, on the left, is called Shape, and this is effectively an envelope/LFO source. To its right is an Effect section that determines what action

Submerge with the Sample Rate effect triggered using the Special Shape.

it is that your selected Shape parameters control, and above are various controls that determine how and when the Shape is triggered. It can be tempo-sync’ed, to start on any beat of any subdivision (from 1/4

Slate Digital Submerge $49 pros • Instant side-chain pumping, and many more creative effects. • Streamlined and stylish GUI with some clever controls. • Super easy to get your head around. • Lots of fun!

cons • None.

summary Submerge delivers the classic EDM side-chaining effect, while also taking that idea to new places — not least through its filter and sample rate reduction effects.

to 1/32, with triplets an option), or you can choose to go freeform, setting the cycle in Hertz rather than beats. There’s also a delay slider to offset the cycle start point. In Auto Trigger mode operation is triggered by DAW playback, but you can instead trigger it using MIDI notes (whether from a keyboard or a MIDI track in your DAW), or an audio side-chain input, with a threshold slider determining at what level the plug-in responds. There are three main Shapes to choose from: Natural, which is good for classic ducking effects; Extreme, which is good for gate/tremolo/chopper-style effects; and Special, which defaults to a parabola-like shape, causing some ducking just before the beat as well as after. Click a second time on one of these Shape selection buttons, and you invert the curve of the selected Shape, giving you six starting points. The Morph control can then be used to manipulate the Shape, steepening and/or moving the S-curve for Natural (akin to changing the release time on a compressor); pushing Extreme’s curve forward or backward in time; and narrowing and/or distorting Special’s curve shape. The Depth knob sets the strength of the effect. There are eight different effects to choose from, some conventional, other less so, and a Tweak knob controls one effect-specific parameter. First up is a level effect but with a twist: the Tweak

Using the Extreme Shape with a low Depth setting, sync’ed to 1/32, to inject some movement into a pad with a resonant low-pass filter.

Finally, we have the icing on this plug-in’s cake: Sample Rate. This reduces the signal sample rate for a dirty, crunchy old-school gaming sort of sound. Tweak controls ‘smoothing’; essentially, how much crunch and splatter you hear.

Move & Groove It’s super easy to get started with Submerge. If you want to keep things super simple, all you really need to do is decide what elements to put Submerge on, and explore the presets. The classic EDM quarter-note side-chain pumping effect is

“There’s so much fun to be had when you start manipulating the controls during playback.” control determines how much the Mid and/or Sides components of the stereo signal are affected. With Tweak fully anticlockwise, just the Sides (so any wide, out-of phase ear candy in an EDM tune, for example) are ducked, while at the opposite extreme it’s just the Mid, and in the 12 o’clock position the full stereo signal is ducked. A cool-looking meter above shows clearly what’s going on when. Then we have various filter-based options. Beneath Gain, are high- and low-shelf ducking options, and for each of these the Tweak knob sets the turnover frequency. To the right are four more filters, with Tweak setting the filter resonance for each: high- and low-pass filters, and two combinations of them, one creating a band‑pass filter and the other a sort of broad notch filter or ‘closing curtains’ effect, depending on the Tweak value.

to hand, along with many more options. Used in this way, the one thing you have to be a wee bit careful of is that if what you’re processing has no Sides information (a dual‑mono loop won’t, for example) then it may seem that the effect isn’t working — in that scenario definitely adjust the Tweak control to see what happens. And if your aim is simple, for example to let the kick through loud and clear in a mix, it’s easy to select the Natural Shape and apply it just to the Mid, then use Morph to dial in the ducking just a fraction ahead of the drum. But there’s so much fun to be had when you start manipulating the controls during playback and, for me, this plug-in becomes way more interesting when you investigate what it can do with faster cycle rates of Extreme controlling filters or the Sample Rate effect. For example, using Extreme set to 1/16 to control resonant filter

effects can add some great movement to boring pad sounds, or more predictable spot effects such as filter-swept noise. You can quickly spice up a transition like that. And if you then look at automating the Depth and Tweak controls to increase the complexity of the effect as that whoosh progresses, you can create some truly enticing ear candy. Add another instance of Submerge, with a Sample Rate effect, and the complexity increases... You get the idea. Best of all, you can map a MIDI controller to these controls and have lots of fun ‘performing’ with Submerge, as you write automation in. That really could be a great way to liven up that humdrum house track! The same effects work well when used more subtly in other genres, by the way: I was able to restore a bit of life into some over-distorted rock guitars, for example.

Sublime? I’ve been genuinely impressed with Submerge. It sounds good and it’s super easy to use. On occasion I wondered if more features might be nice — maybe a range limiter for some of the controls, to prevent accidents when tweaking in real time, or an LFO to modulate some of the controls — but on the whole, Slate Digital have got the balance between features and ease of use just right. Submerge is a great route to some really enticing effects, and could be used to liven up pretty much any production. $ Perpetual licence $49. Also available

through Slate Digital’s All Access Pass, from $16.67 per month. W https://slatedigital.com

www.soundonsound.com / March 2025

ON TE ST

DJ Swivel HitStrip Channel-strip Plug-in

With its combination of creative and corrective tools, HitStrip puts all the channel processing you’ll ever need in one intuitive window. DJ Swivel HitStrip $169 pros • Everything you’d want in a modern channel strip focused on speed of use. • Richly featured EQ. • Width and monitoring controls are really useful. • A range of modifier keys. • Good learning/support features.

cons • Nothing to write home about.

summary A clever combination of traditional and modern software processors with a simple GUI that does everything on a single page.

March 2025 / www.soundonsound.com

M AT T H O U G H TO N

hannel strip plug-ins aren’t a new idea, but DJ Swivel’s HitStrip seeks to bring it up to date, combining commonly used elements of different plug-ins — plus a few more special features — in a single, intuitive GUI. This plug-in is available in the usual formats on Mac and Windows and authorisation is by serial number. The download was refreshingly small (a 69.1MB DMG file for my M1 MacBook Pro) and installation painless. On first opening an instance, only the EQ and dynamics sections are activated, but you’ll can see an array of greyed out options in tabs across the top, and I’ll run through all of these facilities below.

Banding About Let’s start with the EQ, which is based around a familiar-looking nodes-on-curve plus frequency-analyser display. There are multiple ways to create nodes, the most intuitive being to click on the yellow line. This adds a bell node and, in FabFilter Pro-Q-esque fashion, the controls for this

A traditional ‘you’d never use these all these settings in real life’ shot to highlight some of HitStrip’s features!

band or whichever you select thereafter float over the display. Other features are borrowed not only from Pro-Q, but some other modern EQs too. For example, not only can each band can be static or dynamic, but it can instead be set to EQ only the transient or tonal portions of the signal, as in Eventide’s SplitEQ (whereby the default range control for the dynamic bands becomes a transient-length control). These modes are selected from a drop-down on the floating control set. Each band is a bell by default, but another drop-down presents different options: low‑pass, high‑pass, band‑pass, notch, low‑shelf, high‑shelf, and two variations on a ‘tilt’ EQ (a straight line or dual shelves). There are two more drop-down menus, the first setting the slope from 6 or 12 dB/octave up to 96 dB/octave in 12dB/octave steps. Such options are a familiar sight for high‑/ low-pass filters, but here they’re available for all filter types, meaning you can do

some very interesting things to the curves, as seen in the main screenshot. But it’s not just about making interesting shapes possible: it also means you can create pretty intricate EQ responses using just a few nodes — and so with less clutter. The final drop-down menu determines which signal(s) the band acts on: stereo, left, right, Mid or Sides. Between the menus, you have the main length/range, gain, frequency and Q knobs, and top right of the floating control set you can engage band solo, and split the band into separate channels (if the band is in stereo, left or right mode, you’ll have separate linkable nodes for left and right, whereas if you start with a Mid or Sides band, you’ll have both Mid and Sides nodes). You can also delete the band. Some of this control is helpfully duplicated in a smaller control set under the node nearest the mouse cursor, but there are some subtle differences: while you can see the settings, for example, you can’t tweak them on this window (you can move the node to change the frequency and gain, of course, and can hold down a modifier key to adjust the Q); the band solo here is a click-hold type, whereas that in the lower window latches on when clicked; and you can’t split nodes, access the dynamic/transient functions or set the slope steepness. You can create anything up to 24 nodes, so there’s plenty of scope to get as surgical as you need to. The display also includes a frequency analyser, and can be set to show a mastering-friendly ±6dB range up to a more radical ±36dB; it defaults to a ±12dB, and the range grows automatically if you boost/cut beyond the display. An EQ wet/dry blend at the top applies to all bands, making it possible to scale back an exaggerated curve, which can be handy if you take things too far. At the bottom, a virtual keyboard can be popped out. As well as giving your EQ moves a musical reference point, you can click on any key to create an instant 3dB bell cut. This makes it trivially easy to nix a fundamental and then zap some harmonics, and since you can lasso-select multiple nodes, you can then change the Q, gain or frequency of them all en masse. Very slick! One little gotcha to watch out for is that if sweeping the frequency of multiple nodes, you can run out of road: if the outermost nodes hit the 20Hz or 20kHz extremes and you keep sweeping, the nodes start to bunch up, and it’s not undone if you then dial things back;

the built-in undo didn’t seem to undo these moves.

In, Out, Squash It All About... Flanking the EQ’s graph display are input and output meters and some interesting level controls. A stereo (L-R) meter on the left ranges from -∞ to 0 dBFS, and a draggable numeric field above provides input gain control (-∞ to +12 dB). You also have a gate, which can be activated in a tab at the top, by turning the attack or release knob under the meters, or popping open the ADV (advanced) panel, which brings up more controls (ratio, threshold, range, knee, low‑ and high‑cut filters, and look‑ahead delay). On the right of the display, it’s a similar setup for the output limiter: there’s

a threshold control on its gain reduction meter, and it features automatic make‑up gain so that as you drag the threshold down the signal gets louder. That’s not how I like to work for every source, as it happens, but it’s easy enough to dial in the desired gain reduction and then pull the signal down post-limiter by dragging/typing the output level at the top. The gate and limiter aren’t the only dynamics processors. Not only can any of the EQ bands be made dynamic, but there’s also a dedicated Dynamics tab with a control set just beneath. As with the other tabbed sections (other than the input and output stages) this can be dragged to the desired place in the signal chain, and it can be a compressor or expander, as you

It’s not just the EQ section that makes use of the main graph display in the centre: tweak the high and low filters of, say, the gate or Transients sections, and their turnover frequencies are displayed there, while the harmonic enhancers appear as a sort of heat map.

www.soundonsound.com / March 2025

ON TE ST

DJ SWIVEL HITSTRIP

prefer. The usual controls are available (threshold, ratio, attack, release, knee and make‑up gain), along with another level meter for the Dynamics section’s input signal, and a large horizontal bar graph for gain reduction; from 0 all the way to -60 dB, with the most useful area in the right half. The make‑up gain can be set manually or automatically and the left and right channels linked or not. This is all handy to have in view, if pretty conventional. But the Dynamics processor’s side-chain support is more unusual: the compressor/expander can react to the internal signal as usual, to an external side-chain sent from another DAW track, or to an internal pulse set to beat subdivisions of your choice (1/1 down to 16th triplets), for sync’ed pumping or tremolo effects. There are dedicated plug-ins that might give you a little more control over this sort of effect, but for the pumping effects you hear in so much EDM and hip‑hop these days, you have exactly what you need, with no fiddly routing or extra plug-ins involved.

On The Tabs The remaining tabs are labelled De-esser, Exciter and Transients. The De-esser takes the form of an additional cut-only dynamic EQ band with a slightly different control set, and it can employ either a cut-only bell or a high-shelf filter. The Exciter tab has controls beneath for a two-band harmonic enhancer (gain and frequency for low and high bands, respectively) and, helpfully, the target area and intensity of process are reflected for each band as a sort of ‘heat map’ on the main EQ display. The Transients processor allows you to set the length of transients and apply gain to them, and to shape them with low‑ and/or high‑pass filters. These filters are, again, displayed temporarily on the main frequency analyser display (the same happens for any process which includes filters to target the processing, by the way). Finally, across the top are a few more global controls and an unassuming-looking master section, whose appearance barely hints at the power hidden within. You can access a neat user guide that gives you a step‑by‑step tour of the GUI and a range of modifier-key shortcuts, which I’ve not listed exhaustively here but which allow you to do things like lock one axis while dragging EQ nodes, make fine adjustments and more — very useful things to learn if you want to work quickly. There are both

The Dynamics section can be a compressor or expander and as well as reacting to the internal or external side-chain, it can be set to react to pulses on beat subdivisions: instant pumping effects!

global and per-effect module presets, and you can organise these in banks on a menu, which is a nice touch. After the ubiquitous undo/redo and A/B compare controls comes the master section. This includes a polarity inverter, a stereo widener and a colour (saturation) control, and the latter two in combination can sound very effective. Importantly, there are some handy listening facilities: you can choose to listen to the full stereo signal, fold it down to mono, or to solo the left,

you have found your feet, you’ll find that it all makes good sense and, more importantly, that this plug-in makes it possible to achieve decent results incredibly quickly. Can you do everything here with plug-ins you already have? Possibly. There are plenty of channel strip plug-ins that cover the basics of gating/ expansion, de-essing, compression, EQ and even width, saturation and harmonic enhancers. Also, your DAW might allow you to arrange plug-ins in a standard window layout. Some of the analogue-inspired virtual racks like Slate’s VMR and PSP’s InfiniStrip provide another approach, albeit without the same visual feedback of more proudly digital plug-ins. You may have a good EQ plug-in that offers dynamic EQ and control over different channels, but not many of them will offer 24 bands, and while something like Fabfilter’s Pro-Q 4 does so (and has plenty of unique features) for a similar price, HitStrip offers other benefits such as transient EQ’ing. Really, though, it’s having an EQ like this at the heart of a full, drag-and-drop re-orderable channel strip, complete with metering, width processing, limiting and monitoring facilities, and with useful visual feedback from different processors all on the one grid, that makes HitStrip different from anything else I’ve used to date. I can see some scope for further features and workflow efficiencies in the future, for example making more options available at a right-click, and I imagine DJ Swivel will already be looking hard at the host of features FabFilter introduced in Pro-Q 4 shortly after HitStrip was launched. But it’s early days for this plug-in, and it already works very well indeed. I for one think Mr Swivel may well be on to something! You could easily make this your go-to channel strip: no more burning of time deliberating which plug-ins to choose, you’d just fire this up and get to work, and then lean on other plug-ins only when you decide you need their specific features. If you’re looking for ways to sharpen the focus of your plug-in collection and to mix and master more quickly, HitStrip combines bread and butter features with some more rarefied tools, all in a neat, well thought-out GUI. It could well deliver the goods for you and is definitely worth a test drive.

“A neat user guide gives you a step‑by‑step tour of the GUI and a range of modifier-key shortcuts.”

March 2025 / www.soundonsound.com

right, Mid (the same result as mono!) or Sides channels. Now, all this could be very handy if you have an instance of HitStrip on the master stereo bus, obviously, but these controls also allow you to check properly for side-effects from your stereo widening and coloration. As with so much in this plug-in, it makes doing what needs to be done quickly very easy.

Stripped Ease? I’ve found HitStrip a genuinely interesting plug-in. DJ Swivel has clearly looked at the software tools he uses in his productions, and set out to combine their features in a way that puts everything he routinely needs in a single plug-in window. He’s gone a step further, though, and usefully included visual feedback from the different processors on the main EQ display. Of course, with so many controls and so much information in that window, it can at first appear busy and take a little time to reach the point where you start reaping the productivity gains. But once

$ $169. W https://djswivel.com

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ON TE ST

Therevox ET-5 The ET-5 may have it roots deep in electronic musical history, but it’s very much a modern instrument. ROBIN BIGWOOD

ontinuous‑pitch electronic instruments like the Theremin and Ondes Martenot have traditionally been horribly difficult to master, repellently expensive, or both. And yet they have a certain exotic appeal, and an evocative musical quality that’s hard to recreate by other means. Not to mention a huge potential for experimental use, and a formidable cultural provenance inextricably tied up with sci-fi soundtracks and psychedelia. The ET series of instruments by Canadian company Therevox might be less well known, but they offer a distinctive take in this field. In a nutshell, a Therevox combines aspects of the Theremin (analogue tone production, continuous pitch) and the Ondes Martenot (pitch control via a mechanical finger ring, pressure‑sensitive buttons), and adds additional, interesting features all of its own.

Phone Home The ET-5 may be an unusual, mould-breaking instrument, but essentially

March 2025 / www.soundonsound.com

Analogue Synthesizer

it’s not hard to describe. The synth signal path, which is fully analogue, is quite conventional: two oscillators into a single low-pass filter. The oscillators have minor differences in footage ranges, and osc 2 has a white‑noise option in lieu of osc 1’s narrow pulse, but otherwise can sound identical. Osc 2 can be offset by ±7 semitones, the square and pulse wave widths of both can be adjusted via patching (of which much more in just a moment), and there’s a switch to sync osc 2 to 1. Next in the signal path is a mono send-return loop intended for effects pedals, with quarter-inch sockets on the rear panel. A nice touch is that standard 9V pedals can also be powered by the ET-5, as long as they don’t exceed 200mA current draw, and conveniently placed on the nearby blank area of the control panel. A front‑panel knob applies gain to the return, and with no pedal connected it behaves as an overdrive/ saturation stage, rather fine-sounding it turns out, with a red LED showing the intensity of the effect. The ET-5’s filter is an 18dB/octave low-pass with resonance, and as you

might expect its character is mid way between squelchy 24dB and silky 12dB. The output feeds a vintage-character internal spring reverb (with maximum decay an opulent 10 seconds or so) that gets its own Blend/mix knob. Finally there’s a volume knob for the single line-level 6.35mm TRS/TS output socket. The power supply, incidentally, is a substantial in-line brick with a 16V 2A AC output, and available in 110V and 230V versions. From here on, things are much less conventional. Let’s start with everything happening on and around what looks like a keyboard, but is what Therevox call a fingerboard. To the left of it, two large wooden blocks are ‘intensity keys’, controlling the amplitude of oscillators 1 and 2 respectively, and essentially acting as real-time, manual envelope generators. The equivalent of what on an Ondes Martenot is called a ‘touche d’intensité’ or a ‘lozenge’, they’re spring-loaded vertically and can be depressed into the casework about 10mm. Initial resistance is light, increasing with depth, and the action is both smooth and quick: they allow for long swells, but striking or tapping gives very fast, almost instantaneous, attacks and decays.

Visually a blend of unusual and familiar synth elements, with some decidedly retro styling, the ET-5 measures 812 x 266 x 88mm and weighs about 5.5kg. Using it in icy wastelands is not compulsory, nor recommended in fact.

The 30mm faders either side of the keys are ‘holds’. They’re not sprung, will stay put where you leave them, and have no predetermined function of their own. They’re variable voltage sources, and do various things when you patch them in the ET-5’s patchbay. With a standard 3.5mm TS Eurorack patch lead patched from Hold 1 to Vol 1, for example, the left hold will take over the role of its adjacent intensity key, breaking an internal normalled connection. Great for setting up drones, but you could just as easily patch the hold to PWM for static control of pulse width, or FX Mix to create a variable send level for the effects loop, which otherwise has no physical control of its own. More about the fingerboard itself now. Conspicuously, it’s marked with a black and white semitone layout reminiscent of a normal keyboard, spanning nearly four octaves. But these aren’t keys, and instead a rigid transparent plastic strip forms the surface, into which evenly spaced dimples have been drilled. So there’s both a visual and tactile guide to where in-tune semitone centres lie. About 2cm above runs a wire, emerging from the casework each side, supporting a metal ring with a rubber

tensioner. The idea then is that you play the Therevox with a right hand finger inserted into the ring, moving left and right across the pitch range to define pitch, and your left hand operating oscillator intensity and the holds. It’s quite like a Theremin in this respect, with the hands performing essentially the same roles, but with the addition of tactile feedback for both aspects of control. Finally, the patchbay, which operates on a 0-5 V standard and with 3.5mm sockets typical of Eurorack. Five short patch cables are included with the ET-5 and they’ll get you quite a long way. I’ve mentioned a few internal patching examples already, but many more are possible. A further infinity of possibility awaits if you have outboard modular gear of course. CV outputs include the intensity keys, holds, ring position (tracking at 1V/octave), the position of a connected expression pedal, and the audio output of oscillator 2 (5V bipolar). There are also ‘gate’ outputs that send a full voltage signal any time an intensity key is operated: they’re ideal for firing external envelope generators. As for CV inputs, most aspects of the architecture can be modulated, and the ET-5 oscillators track at a standard 1V/ octave, so can be controlled by other gear. Patching pitch or volume breaks internal connections from the ring and intensity keys, but most other inputs behave in a cumulative way. So if you have the ET-5’s filter cutoff knob set half open, say, and patch a hold to it, the hold only opens the filter further rather than taking it over completely. Very sensible. There is also some interesting quirkiness. Filter resonance can be increased beyond what its front panel knob can achieve by applying a voltage to its CV in as well, for an excitingly unstable and ringing character. Also, while oscillator Vol and FM inputs allow for intriguing AM and gravelly analogue FM tones, the latter has a DC blocking design that distorts the effect of incoming waveshapes. For example, when using osc 2 as a modulator, in its low frequency mode, with its output patched to FM1, you can achieve sine wave vibrato as expected, but a triangle wave produces a two-note trill effect, and sawtooth and square waves only momentary blips. Nestling between the ins and the outs are two four-way passive mults, for splitting or summing signals. There’s also

a voltage inverter, which can reverse the polarity of a hold fader, for example. With only a single patch lead inserted the inverter becomes a 5V voltage source, which can be useful for some things. Meanwhile, several 3.5mm CV sockets are duplicated on the side of the ET-5 in quarter-inch guise, which could prove handy for some jobs and modular formats other than Eurorack. Especially nifty is an expression pedal (or CV) input that by itself can be switched to control either filter frequency or oscillator 2 pitch. It gets an output socket in the patchbay as well, so pedal movements can be directed to many other parameters.

Slope-style So that’s the way the Therevox is made. What’s it actually like to play and listen to? In short, it’s quite weird, often wonderful, and surprisingly versatile. As with any novel instrument which hasn’t (yet) attracted an established, broadly agreed playing technique, individual players find their own way. With the ET-5 there aren’t that many options, but it’s still open to some variation. For me, I naturally gravitated towards using my left hand middle and index fingers to operate intensity keys, and the right hand index for the ring. At times though I ended up just holding the ring, especially when wanting to do lots of parameter tweaking.

Therevox ET-5 $2950 pros • Playable and approachable from day one, and full of potential for the longer term. • Synth signal path, overdrive stage and spring reverb all sound superb. • Eurorack-style patchbay vastly increases the ET-5’s own tonal breadth while allowing it to be a potent partner for other modular gear. • Many concessions towards practical, serious use on stage or in the studio. • Super hand-built quality, with a solid walnut case.

cons • Currently a 24‑month waiting list for delivery.

summary A unique, highly characterful analogue synth that reimagines the spirit and sound world of the Theremin in a more approachable and controllable form. Modular interfacing and MIDI broaden the scope considerably.

www.soundonsound.com / March 2025

FE ATURE

Tradition Meets Technology What is Hybrid Drumming? And why do you need it? MARK GORDON

ybrid drumming involves the integration of drum pads, triggers and other electronic components into a traditional drum kit setup, enabling you to blend the sounds of acoustic drums with a wide range of electronic sounds and effects. Incorporating electronics into your acoustic kit not only allows the production values of today’s modern music to be reproduced live, it can also introduce a unique level of creativity into your playing that can’t be achieved with acoustic drums alone. While rock drummers like King Crimson’s Bill Bruford and Rush’s Neil Peart were among the earliest hybrid drumming innovators, the trend exploded in the late ’80s with the new romantic bands like Duran Duran, and occasionally more edgy acts like New Order, adding electronic components to their percussive arsenals. But things have moved on a long way since adding a couple of Simmons pads was considered cutting edge, and the technology has become more advanced, more affordable

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and — equally important — more reliable. Modern music demands a wider palette of sounds than ever before, and hybrid drumming is fast becoming something that many drummers are exploring.

Triggering Your Acoustic Kit Augmenting an acoustic kit with triggered sounds is a great way to beef up your drum sounds or add additional elements, such as a tambourine or clap, to a snare. Triggers themselves come in two formats: single and dual. They attach directly to your acoustic drum shells, typically mounting to the top hoop of each drum. Once a trigger is tightly secured to the drum, a small piezoelectric sensor rests on the drum head, detecting the vibrations generated when your drum stick strikes it, and sends that data to a drum module (your source of sounds) via a jack or XLR cable. The single-trigger option only sends a signal when the batter head is struck, but a dual trigger, as its name suggests, includes two transducers that take signals independently from the head and rim of the drum, allowing two different sounds to be played.

The Roland RT-MicS is a trigger with built-in sounds that clips to a drum hoop.

Photo: Sarah Watts Point and Shoot

Designed for hybrid drumming: the Roland TM-6 gives you six trigger inputs and a huge range of sounds to play with.

Roland, Yamaha, 2Box, Ddrum and many other manufacturers all produce triggers spanning a wide range of prices, build quality and style. They all work in pretty much the same way, so my advice for anyone looking to buy would be to choose what suits your budget and the physical requirements of your drums — not every trigger will fit perfectly on every drum hoop. You can buy very simple piezo pickups that stick to the drum head and have a trailing jack socket attached but, in my experience, these are impractical and don’t give great results. I’d avoid those and go for something more substantial.

Yamaha’s EAD10 drum module has kick and snare triggers, and a stereo microphone that allows you to process the rest of the kit.

The Simple Options Keeping it simple, something like the Roland RT-MicS is a great way to start. This self-contained unit clips onto your snare drum (or any other drum in your kit) and has eight built-in sounds that are triggered as you hit the drum, so no external sound module or cabling is required. In addition to the snares, claps and percussion sounds the unit ships with, you can also add two of your own sampled sounds, to customise your sound palette somewhat. Roland also produce a two-channel version (the TM-2), should you want to add a kick and snare trigger, or you could go for the six-input TM‑6 Pro, which (as its name suggests) offers six trigger inputs. I reviewed the last for Sound On Sound in the December 2018 issue. With a fully featured unit like the

TM‑6 Pro, you can add triggers to your kick, snare and toms or incorporate external pads into your setup. We’ll come on to that last bit later. Also worthy of a mention at this point is the Yamaha EAD10, whose initials stand for Electronic Acoustic Drums. This is a slightly different proposition but, having also reviewed it for SOS, I know it would fit very nicely into a hybrid drum setup. In addition to the kick and snare triggers, the EAD10 features a stereo microphone that captures the rest of the kit and allows you to process it through a range of effects. Its compact and uncluttered format means you won’t have

to spend time setting up multiple triggers and mics or dealing with cables trailing all over your kit, but be aware that the microphones will potentially pick up other sounds on stage.

Triggers Or Pads? Using triggers attached to acoustic drums, on the one hand, and adding pads to a kit, on the other, are both methods that have their place, and each works better for different things. Obviously, with a trigger attached you’re still going to hear the drum alongside the triggered sample, so this is ideal for augmenting acoustic sounds. Drum www.soundonsound.com / March 2025

FE ATURE

HYBRID DRUMMING

pads, however, make no sound (certainly in a live scenario) and are ideally suited to triggering samples for which you don’t want an acoustic element. Adding a subtle 808 snare or conga, for example, would be better accomplished via a discrete drum pad than via a trigger attached to a drum. Pads work in exactly the same way as a trigger: a rubber or mesh pad connects to your drum module. If adding one or more drum pads to your acoustic kit is either physically or aesthetically challenging, the Roland BT-1 bar trigger is a great option to consider, as it takes up far less space than a typical drum pad.

Other great options for the hybrid drummer are sample pads such as the Roland SPD range, the Alesis Strike MultiPad, the Yamaha DTX‑Multi 12 and the brand-new Korg MPS‑10, all of which have been reviewed in SOS. These units all provide a compact solution combining pads, trigger inputs and built-in sounds, along with sampling capabilities, allowing your own unique sounds to be incorporated. The SPD, in its various guises, from the original SPD-S to the SPD-SX and new SPD-SX Pro, has been my weapon The Roland BT-1 is a trigger pad that can attach to your drum head or be mounted on its own stand.

of choice in my own hybrid drumming setup for many years. It has an incredible set of functions and features that let me bring a high level of creativity to my acoustic drum performances.

More Than Percussion How about adding more musical elements to your electronic arsenal, such as keyboard sounds and effects? Of course, anything tempo related will need to be played ‘in time’, which opens the tricky subject of playing to a click, but there are many ways you can enhance a performance without needing a cowbell banging in your ear at 150dB. Playing back keyboard pads and ambient soundscapes is a great way to augment songs if you don’t have a keyboard player and want to add something extra. By sampling chords into your drum module and assigning them to individual pads, you can trigger them as required by hitting the appropriate

At The Mercy Of Monitoring As drummers, we’ve been in the enviable position of never having to worry where the power sockets are at a gig. The room itself has always had the most impact on the sound of an acoustic kit, but as a hybrid drummer a few new and important considerations become part of our remit. Previously, the PA was something to enhance the already relatively loud acoustic drums, but now it’s the only way any electronic sounds can be heard. There are two elements to consider here: the front‑of‑house sound (what the audience hears); and monitoring (what you hear). I won’t delve into the front‑of‑house side of things too deeply, as that would be very much an article in itself, but be aware that the PA will dictate how the electronic elements of your kit sound to your audience. If you’re hybrid drumming in a small band gigging in pubs and clubs, the PA will now be required for more than simply the vocals, so you may need to upgrade or add to your existing system. Monitoring also becomes very important — it’s the only way YOU can hear the non-acoustic elements of the kit. If you are using tempo-based loops and sounds you will also need to be able to hear the click clearly. There are a number of ways of creating a good monitor setup for a hybrid drum kit, but a compact mixer is one of the easiest and most robust. The majority of drum modules will allow the click output to be routed independently, so you can bring this into a channel of your mixer alongside the other sources you need to hear. If you have multiple outputs on your drum module, you can send one feed into your mixer and another to the front of house. Finally, you can bring a monitor mix of the rest of the band into the mixer, giving you the ability to balance the band,

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your electronics and (if you are using one) the click. It’s possible to work with a regular monitor wedge, but a good pair of IEMs (in‑ear monitors) is a far better solution, and really the only

option if you’re using a click and don’t want the audience to hear it. Not only will you get a great mix of the band and your electronics, you’ll also be protecting your ears!

Drum pads like the Roland SPD range...

...Korg MPS-10...

pad. The majority of sample pads allow you to create mute groups, which automatically silence one pad when another is struck. This means you can make your samples extra long (memory permitting) and they will simply end when you play the next pad. You could also assign individual notes to pads and, with some level of dexterity, play melodies and phrases. If you’re looking for something very much on the creative side, rather than simply triggering samples, the Nord Drum 3P is a really interesting option, a little bit like a synth for drummers!

...Alesis Strike MultiPad...

...or Nord Drum 3P can make a great all-in-one pads+sounds combination.

If you’re comfortable playing to a click (or have an alarmingly good sense of tempo), you can start to expand into triggering tempo-based elements, which can include keyboard parts, percussion loops and even vocal phrases. In my experience, it’s a good idea to keep phrases as short as possible and retrigger them often, rather than hitting a pad and hoping a 30- or 40-second loop will remain in time. For example, if you want to add a percussion part to a chorus, make the loop a bar long and retrigger it every bar rather than using an eight- or 16-bar phrase. If you trigger

Read More About It! Many of the products covered here have been reviewed in depth in SOS. Here are some handy links... Roland TM-6 Pro www.soundonsound.com/reviews/roland-tm-6-pro Yamaha EAD10 www.soundonsound.com/reviews/yamaha-ead10 Clavia Nord Drum 3P www.soundonsound.com/reviews/clavia-nord-drum-3p Roland SPD-SX www.soundonsound.com/reviews/roland-spdsx Roland SPD-SX Pro www.soundonsound.com/reviews/roland-spd-sx-pro Alesis Strike MultiPad www.soundonsound.com/reviews/alesis-strike-multipad Yamaha DTX‑Multi 12 www.soundonsound.com/reviews/yamaha-dtx-multi-12 Korg MPS‑10 www.soundonsound.com/reviews/korg-mps-10

the loop a fraction early or late, it will be horribly out of time after 16 bars has run its course. I’ve added backing vocals and drum loops live using the SPD-SX. With the benefit of the many clever playback options these units offer, such as mute groups and pad linking (see the reviews for more details), it can be a lot simpler than you might expect to create and control quite complex backing.

Expanding Horizons Hybrid drumming has become a ‘must have’ in a lot of modern professional drumming setups. You only have to watch a few ‘rig rundowns’ on YouTube to see how many sample pads, drum modules and triggers are being used. But with the technology available from manufacturers such as Roland, Yamaha, Alesis and others, it is not difficult to incorporate electronics into any acoustic setup, however modest. Electronics can add an exciting and creative dimension to your playing that has almost endless possibilities. www.soundonsound.com / March 2025

TECHNIQUE

There comes a point when tinkering with your mixes won’t improve them — it could even damage them. But just how do you know when it’s time to let go? NEIL ROGERS

here’s a famous quotation, often attributed to Leonardo da Vinci: “Art is never finished, only abandoned.” In other words, creative projects aren’t like mathematical equations that can be solved. There will always be more that the creator could do, but since there’s no ‘right’ answer they’ll never arrive at a clearly defined end point. So they must learn to live with the fact that they may never be truly satisfied with their creation, and figure out how to recognise when it’s time to let go and move on. It’s a sentiment that applies as much to mixing music as to any creative endeavour... Unfortunately, today’s music-making technology practically invites us to keep on refining our mixes, and strive to inch ever closer to some vaguely defined notion of ‘perfection’. Knowing when a mix is ‘done’ can be even more challenging for those who are still feeling their way into mixing, since they can be tempted to try out lots of new and exciting techniques in a constant chase for results that sound ‘better’ or ‘more professional’. But, at some point, the experimenting must stop, and your focus needs to shift towards ‘signing off’ the mix. That means learning how to wrap things up in a practical sense, but also how to move on emotionally from a mix — and I can’t stress how incredibly important this is in your development as a mix engineer!

Can You Feel It? So, just how do you know when a mix is finished? Well, if you’re mixing for an artist (or any other client), the short and obvious answer is “when they’re happy!” But if you’re to feel at all satisfied and to develop as a mix engineer, it’s important that you’re also happy with the mix before you send it off. And, of course, maybe as well as being the engineer, you’re the artist — in which case, how do you know when you’re content to call it a day? In my experience, there’s rarely one clear moment in which you feel that you’ve suddenly ‘cracked’ the mix, and find yourself punching the air and dishing out

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high fives to unsuspecting strangers in the street. More likely is one of a few different scenarios, and the most gratifying is when you sense that the music you’ve been engrossed in mixing starts to take on a life of its own. It’s when you start to feel more like a listener enjoying the music, and less like an engineer trying to shape it. Maybe the chorus stirs some sense of feeling, perhaps a particular lyric begins to resonate and the hairs on your neck start to tingle. Or it might simply be that you notice you’ve subconsciously started nodding your head or tapping your foot to the beat. I’ve learnt to take such feelings seriously, because at this point mixing can become more about protecting and nourishing this often-indefinable quality in the music. On that note, another scenario might be seen as the flip side of the same coin: you notice a growing sense that you might be ‘overworking’ the mix, and thus making it worse. In that case, maybe it’s time to stop throwing more plug-ins and production tricks at the project, and make a conscious effort to tie things up. More pragmatically, an end point might be driven by client deadlines, or a simple need to get on and get a mix finished so that you get paid. Now, in any of those scenarios, we absolutely still need to feel moved by the music. That’s the whole point! But when there’s a lot to do and we don’t have the time to reflect and experiment, we can,

A good set of studio headphones may well tell you something useful about your mix as it nears the final stages — but you do need to be careful about what you tweak as a result.

hopefully, lean on our experience and some routines to allow us just enough space for creativity while ensuring we deliver consistently good work.

Shifting Gears For clarity, let’s say we’re talking about ‘the last five percent’ of a mix, and how we take the good work we’ve done and move serenely to the point where we’re happy to send it off to the artist, or maybe even invite them into your studio. At this

stage, I find it helpful to make a conscious decision to shift gears, and stop striving/ pushing for ‘something more’. I’m happy to do that because I’ve decided that I like where I’ve got to. I’ve done my experimenting, explored some blind alleys. I feel confident that, at least to my own taste, I’ve arrived at a point where the music has some impact and is making me feel something. Or perhaps I simply feel that I’ve respectfully stayed out of the way of the song while giving it a little polish.

As well as being useful on a practical level, going through your project and organising everything into a ‘macro view’, with tracks in folders and everything in view on the arrange page, can help to shift your mental focus.

www.soundonsound.com / March 2025

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Studio One

Take your music off-grid with Studio One’s powerful tempo tools. ROBIN VINCENT

estled amongst the more familiar Global tracks such as Lyrics, Chord, Arranger and Marker tracks is the often ignored Tempo track. The Tempo track keeps pace with the gear changes in our projects and lets us make sudden or smooth alterations in speed. If your music production is dedicated to the truth of a single bpm, then the Tempo track is probably completely unnecessary. But once things start to drift, timing starts to shift, or when the music breaks free from the grid, the Tempo track can feel quite magical. In this workshop, we’re going to look at how the Tempo track can work miracles on uneven timing and move through different bpms like it’s completely natural. We’ll also check out how Studio One v7’s new Advanced Tempo Detection comes into play.

Tempo Track As with the other Global tracks, the Tempo track can be found at the top of the track list in the arrangement window. To access it, click on the drop-down menu that looks like a hamburger with a notch taken out of the filling; it sits alongside the Inspector, Spanner (options) and Automation icons. The menu reveals the eight Global tracks that apply their skills over the whole project. The bottom one is the Tempo track.

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When you open this track, you usually see a flat line representing the current project tempo. You can mess with this line in the Tempo track just as you would any other automation: adding automation points, drawing in lines, and making the project speed up and slow down by following the line. You can set a minimum and maximum bpm range so that you can focus on the sorts of changes you want to make. The result on playback is that Studio One works overtime to change the speed of the MIDI and audio in line with whatever changes you write in there. Let’s demonstrate this with the ‘DJ Jazzy Jeff - Get Some Air’ demo project that you can download from PreSonus. On the Tempo track, you’ll see that the tempo is set to a flat 100bpm. If you hover the mouse over the line, it will turn into a finger. Click at the beginning of the verse to set an automation point at 100bpm, and then click at the end of the verse to set another and drag it up to about 150. You should now have a nice straight line running from 100 to 150 bpm across the verse. Press Play and DJ Jazzy Jeff will stay in perfect sync with the increasing speed. It’s seamless and brilliantly impressive. As with other automation, if you hover your mouse between the two automation points, another little spot turns up that you can drag to add some curvature to the line.

The ‘Get Some Air’ demo project is very amenable to tempo changes: you can draw in all sorts of sudden, subtle or extreme changes and it will still sound perfect. If you want to be more precise with your changes you can right-click on the point in the project where you want to make the change and enter the tempo value in the pop-up box that appears. You can also use the Option key on a Mac or Alt on Windows with the arrow keys to adjust the position of the automation point. Holding Shift at the same time makes it even more precise (this works with the mouse as well). It’s not always obvious, but the currently selected automation point is whiter and slightly larger than the rest. You may also notice that the timeline changes as you change the tempo. It appears to shrink and expand, depending on whether you are speeding up or slowing down. This is purely a visual effect as Studio One attempts to represent the timeline in an appropriate way. You can force it to maintain the same ratio between bar lines if you rightclick the time ruler and select Beat-Linear under Timebase instead of Time-Linear.

Stretching Time Changing the tempo of MIDI tracks is very easy, but keeping audio tracks in sync is a slightly different matter. The audio in the Jazzy Jeff project is already prepared for

The Tempo track, shown here at the top, lets you make adjustments to your project’s tempo just as you would apply any other type of automation.

tempo changes, which is why it works so well, but what does that entail? There are two main elements: Timestretch, and knowing the original tempo of the audio. When you bring a piece of audio into Studio One, its tempo will either be defined in the file name, or easily detected by a quick bit of analysis that occurs when you import it. And so usually all you need to do is enable Timestretch on the audio track, and you’re good to go. You’ll find these two attributes in the Inspector for the audio event. The top section of the Inspector is the Track Inspector, where you’ll find Tempo and Timestretch options for the track, and the bottom section is the Event Inspector, where you can set the tempo, transposition and other processes for that specific selected event. If the file tempo is not defined or detected, then Studio One has no frame of reference for time‑stretching, and so the event will not follow any tempo changes in the tempo track. However, if the file has a tempo, then all you need to do is enable Timestretch in the Tempo section of the Track Inspector and it will use an algorithm

to stretch the audio to fit the changing tempo. You can improve the process by selecting an appropriate Timestetch mode in the next drop-down menu. Now, what happens if you’ve recorded the audio directly into Studio One without any click or reference to tempo? Well, first of all, you’re making things difficult for yourself, and perhaps if you want to apply postproduction tempo changes, then maybe you should consider recording to a defined tempo in the first place. But let’s assume we’re too free and creative for those sorts of constraints, and we’ve captured something beautiful that we want to work with. This is precisely where the new Advanced Tempo Detection comes in; we’ll get to that in a moment, but let’s continue with our undefined tempo-free piece of audio. We can tell Studio One to try to detect the tempo, just like if we’d imported it. You can find Detect Tempo under the audio menu or by right-clicking the event and selecting Audio / Audio Bend / Detect Tempo. It may or may not be successful, depending on how steady your performance is. Another, more interactive

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THE TEMPO TRACK

Studio One

option is to use the largely unheard-of tap tempo function. Set your audio track playing and click repeatedly on the word Tempo on the transport bar in time with the music to set the tempo based on your click pattern. You could then use this as the file tempo for your audio event. However, none of that sounds like it will work flawlessly, so we need a different approach, which is where the Advanced Tempo Detection comes in.

Advanced Tempo Detection Now, instead of forcing audio to fit to a fixed or varying tempo, we’re going to extract a varying tempo from a piece of freeform audio and apply that to everything else around it. You may have a live drum performance that has an unbelievable groove that you want to keep; it may be that you want to add sympathetic production to an acoustically recorded guitar and vocal, or you simply want to shape your production around something that didn’t stick to the grid. Whatever the reason or scenario, Studio One can help you work with it. The process is pretty straightforward. The first thing you’ll want to do is get Studio One to detect the tempo of the audio event. It may have already assigned it a File Tempo in the Event Inspector to match the tempo of the project. However, you have a freeform recording as you are not following the project tempo or playing to a click. So, right-click the event, select Audio and then Detect Tempo. If you look down at the File Tempo, it will have changed to Map, which means it can’t give you a single exact tempo, but it can give you a map of tempo changes. We have a couple of choices for getting that map into our Tempo track. We can rightclick on the audio event, go to Audio and select Extract to Tempo Track. Alternatively, dragging the audio event onto the Tempo track will achieve the same thing. Studio One does a bit of processing, and then a landscape of tempo bumps and ridges will appear on the Tempo track. In my example, I’ve tried to be as obvious as I can with my waywardly expressive guitar track, particularly with the slowing down I did at the end. You can test out whether the tempo changes have worked by turning on the click and listening back to it. The tracking of the tempo should be pretty much spot on, especially if you use a slightly less extreme audio event than my example. If nothing seems to happen when you try to extract the tempo, then make sure

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If you want an audio event to conform to the Tempo track, it must first have Timestretch enabled in the Track Inspector, and a tempo defined in the Event Inspector.

you’ve detected it properly first. Check the File Tempo value in the Event Inspector; it shouldn’t have a value. If it does, then you need to clear that, detect it again and then try Extract to Tempo Track again. With the tempo map in place, all your MIDI parts and drum patterns will follow those changes. Even if you are adding more audio tracks, the click is going to keep you to the right time. And if anything seems off, you can edit it by hand or add some curves or more gradual changes to smooth things out.

It’s perhaps not a great idea to rely on your guitar playing as the source of tempo. It could also be argued that these features encourage sloppy playing, and that it no longer matters whether you’re playing in time or not. But really, the Advanced Tempo Detection algorithm is giving you a way to step off the grid in post-production and find a way into a groove without compromising the integrity of the performance. At worst, it helps keep everything together, and at best, it will save you a huge amount of time trying to follow a groove with intense editing.

Studio One can detect a varying tempo in a freeform recording and create a tempo map from it, which you can then use as the basis for your Tempo track.

www.soundonsound.com / March 2025

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TECHNIQUE

Get your vocal levels just right with these handy tips. JULIAN RODGERS

ocal production has changed enormously since the start of the DAW era. While not every production requires the big, bright, and ultra-processed sound common in pop productions, even the most natural styles today exhibit a level of polish that simply wasn’t there a couple of decades ago. If the vocal isn’t right, its failings will let the whole production down. So, what’s the difference between an ‘OK’ and a ‘great’ vocal sound, and how can Pro Tools help users get there? This article won’t focus on tuning and timing nudges, which are often integral to modern vocal production. Instead, we’ll focus on levelling vocals, because if this isn’t handled correctly, no amount of further processing can hide that fact. We’ll look at four contrasting and complementary approaches. While these techniques can be performed in any DAW, we’ll highlight specific ways to achieve them in Pro Tools using stock features and plug‑ins, as well as third-party tools where appropriate.

Next Level Vocal levelling is the process of managing a vocal track’s volume to reduce inconsistencies in level over time. This might sound like dynamic range compression, and indeed, compression is a part of it. However, there’s a distinction between managing the differences between words and phrases, which can often be done manually, and the faster, syllable-level adjustments that require compression or detailed automation. While it might seem tempting to rely solely on a compressor to control amplitude variations, compressors always leave their

Clip Gain is an excellent first port of call when it comes to achieving a consistent vocal level.

sonic fingerprint to some degree. Even so-called ‘transparent’ compressors alter the sound, and overuse can leave a vocal sounding squashed. This is why combining different levelling techniques is essential for a polished and professional result.

Clip Gain Clip Gain can help you to manage your vocal levels in the first instance, before you move onto further processing (in fact it’s best to perform initial Clip Gain adjustments without any compression, as the compressor will affect how your Clip Gain adjustments sound). In Pro Tools, this feature allows you to adjust the level of audio clips at the pre-mixing stage, much like setting channel gain on a console. However, Clip Gain offers much more precision by allowing dynamic adjustments within a clip, such as addressing loud breaths or inconsistent phrases. To use Clip Gain efficiently, make an Edit Selection across the area you wish to adjust, then use Control+Shift (or Start+Shift on Windows) with your scroll wheel to trim the level up or down. Unlike the Pro Tools mixer’s fader, which has only 12dB of boost available, Clip Gain has up to 36dB, so large changes up as well as down can be accommodated. For more precision, display the Clip Gain line using Control+Shift +minus (Start+Shift+minus on Windows) and draw in your changes directly. These adjustments can be used to handle issues such as mouth clicks, loud consonants, or even basic de-essing.

While Clip Gaining entire phrases or words can be effective, don’t overlook the ability to create ramps for smoother transitions. This approach is especially useful for levelling dynamic performances. Don’t obsess over exact decibel values during this stage; focus on creating a natural-sounding vocal by auditioning changes as you work. Later tweaks can refine these adjustments.

Compression Compression is the quickest tool for controlling vocal dynamics, but it’s important not to expect a single compressor to do all the work. Over-compression can rob a vocal of its natural dynamics, making it sound overly processed. Consider using multiple stages of light compression for a more transparent result. For gentle, natural levelling, start with an optical compressor like the LA-2A. Avid’s included BF-2A emulation is a great option, offering a smooth response that’s easy to dial in. Set the threshold to achieve a few dB of gain reduction and let the compressor handle broader level inconsistencies. For an alternative and very natural sound, the 304C (an updated version of the Joe Meek compressor plug‑in) has a helpful behaviour where beyond a certain amount of gain reduction it stops compressing, which can help keep the results sounding natural. Compressors like the Pro Compressor or Avid Channel Strip provide advanced features for more precise compression. Use a soft knee and slower attack and release times as a starting point for subtle levelling. If your aim is to shape the sound’s envelope, add a second, faster compressor for additional control. This combination of slow and fast compression has been a trusted technique for decades. When it comes to compression, the BF-2A optical compressor is great for natural, smooth vocal level control. Another good choice is the 304C, which only applies gain redutcion up to a point, helping to avoid over-compression.

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If you’re finding it tricky to perform automation in real time, try setting playback to Half-Speed mode.

If tonal balance is an issue, multiband compression can help. For example, Pro Multiband Dynamics allows you to compress the high frequencies without affecting the lower frequencies. This is particularly useful when dealing with sibilance or overly bright performances. A quick tip: hold Shift while adjusting a parameter to temporarily link all bands, making global tweaks faster. Compression is a powerful tool, but use it wisely. Think of it as just one part of the levelling process, complementing manual techniques like Clip Gain and volume automation.

Volume Automation Volume automation remains the most precise way to get a vocal to ‘sit’ perfectly in the mix. Despite all the software solutions available, automation still offers unparalleled control over vocal dynamics. Top mixers rely on detailed automation rather than heavy-handed compression to maintain a vocal’s natural energy. In Pro Tools, automation can be written in real time by riding a fader, or drawn manually in the automation lane. For detailed control, focus on words or syllables that need emphasis, ensuring the vocal remains consistent yet expressive. While drawing automation is great for precision, using a physical fader on something like an Avid S1 (or Avid Control on a tablet) provides a better way to judge decisions in context, because you can react immediately to what you’re hearing. If a move is too tricky to perform in real time, writing moves in Half-Speed Playback (Shift+space bar) allows for better accuracy. Pro Tools Studio and Ultimate users have advanced automation features that make this process even easier. If you need to test a specific adjustment without overwriting your existing automation, try using Preview Mode, which lets you experiment freely before committing changes. Once you’re satisfied, Trim Automation can be used to make subtle

global adjustments without rewriting your original automation, providing as it does a second layer of automation which can, if desired, be coalesced with the existing automation, freeing up the Trim Automation for further tweaks. Automation may seem time-consuming, but it gives you complete control over the vocal’s dynamics. Rather than relying on compression alone, automation enables a vocal to respond to the song’s emotional flow. A good tip is to use Back and Play (you can set up a custom keystroke for this in the Keyboard Shortcuts menu) to quickly rehearse automation moves and refine them until they’re just right.

Specialist Tools For those looking to speed up their vocal levelling, there are several third-party tools that can complement or even replace traditional methods. One popular option is Waves Vocal Rider, which automates volume levelling in real time. You set a target level and adjustment range, and it adjusts the vocal level dynamically to maintain consistency while retaining its natural feel. Automation created by Vocal Rider can be fine-tuned afterwards so no control is lost. The same idea was pioneered by Quiet Arts’ WaveRider TG, which offers a similar approach to automating the creation of automation tracks. Another excellent tool is Sound Radix POWAIR, which separates levelling from compression by employing a loudness-based leveller

and compressor that can be controlled independently. This makes it ideal for situations where subtle levelling is needed without affecting the vocal’s tone. For more detailed adjustments, Melodyne Studio’s Levelling Macro offers a way to graphically balance dynamics. While Pro Tools includes a basic version of Melodyne, upgrading to the full version unlocks these more advanced features. As Pro Tools continues to evolve, content-aware AI-based tools, such as NoiseWorks’ DynAssist, which Avid previewed working under ARA in Pro Tools at the 2025 NAMM Show, promise even greater efficiency in vocal levelling. It represents an excellent way of streamlining vocal workflows, though as with any AI assistance, careful checking of the results will always be required and even with these tools, critical listening and manual tweaks are essential to achieving the best results.

Which Approach Is Best? Unsurprisingly, there’s no single ‘best’ approach. These techniques are complementary and should be combined as needed. For instance, Clip Gain is excellent for correcting large inconsistencies, compression helps control dynamics and add character, and automation ensures the vocal works musically in a mix. Pro Tools has everything you need to get the job done, but exploring third-party tools can offer valuable speed and flexibility. Whichever methods you choose, always prioritise the vocal’s role in the song and let that guide you. Experiment with these methods to find a workflow that suits your style and project needs. Remember, great vocal production is more to do with careful listening than it is about the tools you use.

NoiseWorks’ new AI-powered vocal levelling tool DynAssist was recently demo’ed running in Pro Tools using ARA, so expect full compatibility soon.

www.soundonsound.com / March 2025

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Pro Tools Cubase

TECHNIQUE

Cubase’s Drum Machine features some good kits, but also lets you create original sounds. JOHN WALDEN

ubase 14 brought some very interesting new additions to all three of its main editions but, for Pro and Artist users, the new Drum Machine instrument is a real treat. Supplied with its own dedicated track type, it comes packed with a broad range of ready-to-go preset kits. While these are all aimed very much at electronic music styles, the engine provides plenty of scope for designing your own sounds too, so in this workshop I’ll explore some of those possibilities by targeting some drum-meets-percussive sounds that would be suitable for a more cinematic context.

Show Some Backbone Drum Machine can, in some ways, be thought of as a cross between Steinberg’s Groove Agent and Backbone instruments. It offers the multiple 4x4 grids of virtual drum pads that will be familiar to GA users, and on these pads you can lay individual drum sounds (or create playable pitched sounds). You also get a ‘lite’ version of the layered drum synthesis/ sample-based sound‑design tools that lie at the heart of Backbone. Each sound in Drum Machine can be built from up to four layers. The layers can be based on any combination of the instrument’s various synthesis engines or audio samples. The former are divided into categories based on drum type (Kicks, Snares, Hihats, Toms, Claps, Cymbals and Percussion), but each category also

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Each of Drum Machine’s sound engines offers something unique. Shown here is the Kick 4 engine, with its FM-based oscillator section.

offers multiple options in each of the synthesis engines. For example, each of the four Kicks options has a distinct feature set, going from a simple oscillator combination through to a more versatile FM-based oscillator. Samples can simply be dropped into a layer from the MediaBay or your OS file browser. With additional filters, distortion and envelope controls at the layer level, plus both per-pad effects and global delay and reverb with individual sends from each pad, Drum Machine manages to pack a lot of sound-design potential into a compact, easily navigated GUI. Indeed, there are enough options here that there are multiple ways to approach any specific sound‑design target. With that in mind, please take the examples that follow for what they are: launch pads for your own experiments.

Boom & Drop My first cinematic target was a sort of ‘boom with a pitch drop’ sound. I deliberately confined myself to a sound based on a single layer, and the first screenshot shows the settings I arrived at. This is based upon the Kick 4 synth engine and, aside from adding a dollop of Drum Machine’s own reverb, I didn’t touch any of the pad-level effects. Kick 4 uses an FM oscillator and offers some interesting ways to change the character of the sound. The key settings to note are that I set the Tune control to -12 semitones, for a little extra low‑frequency energy, and both the FM Env and FM Dec were maxed out. While you don’t get full control over all the aspects of the amplitude and modulation envelopes (Backbone gives you more on this front), the latter extends the FM modulation envelope decay time so,

when triggered and sustained, the tonal character of my ‘boom’ changes over time. The other settings to note are the Filter LP Freq (set to 100Hz to focus the overall sound in the lower frequency range) and the relatively long Pitch Decay and Amp Env Decay settings. Alongside the also maxed-out Pitch Env Depth, the first of these generates an extended pitch drop as the sound sustains. The second lets you influence the overall length of the sound once triggered. The end result can be heard in the audio examples on the SOS website (https://sosm.ag/cubase-0325): a nice beefy boom, with a little growl and pitch drop appearing after the initial hit, and which responds to MIDI velocity. It’s pretty effective even just using a single layer like this, but you could easily take things further by blending in additional layers to target both the initial transient/hit and the sustain portion of the sound.

Take My Pulse My second target was a ‘pulse’ sound of the kind that might be used to underscore a tension/drama cue to give it a sense of pace. The second screenshot shows a composite of my Drum Machine settings. I used the very simple Kick 1 synth engine but hardly touched the controls, as the main sound-shaping came thanks to the Filter and Equaliser modules in the PadFX section. A low-pass filter was set to 150Hz (and given a little resonance), while both Low Cut (100Hz) and High Cut (1000Hz) were used to shape the low and high end, respectively. Add in just a touch of reverb and the sound – heartbeat-like, but with a little punch to it — was good to go. Again, you could add more layers to refine the sound but it could also be interesting to experiment with

Creating my ‘pulse’ sound was as much about filtering and EQ as it was using the Kick 1 engine itself.

the combination of the oscillator’s Click settings and the filter cutoff. If you add some ‘click’ and don’t set the cutoff too low, you can create an interesting attack dimension to the overall sound that could be used to give it more emphasis. Alternatively, dial back the oscillator’s Tone control for a somewhat softer end result.

Tick Tock My third target was a ‘ticker’ sound, something that’s often used as a high-frequency complement to a pulse. Drum Machine offers a number of potential starting points for this, and I opted for the Closed HH 3 engine. In this case, I made two variants of the sound: a ‘tick’ (mapped to G1) and a ‘tock’ (slightly lower in pitch, and mapped to A1). You could impersonate a ticking clock with this combination, of course, but used in a more percussive

role, the two variations simply provide some performance variety. The final screenshot shows the settings used for the ‘tick’ version. A little reverb aside, there are only three settings to note. First, the Amp Env Decay was made short, so the overall sound is shorter and less obviously cymbal-like. For the ‘tick’, the Oscillator Tune control was slightly raised and the filter cutoff (which affects both the oscillator and the noise components of the sound) was set to about 11 o’clock. For the ‘tock’ version, the Tune control was adjusted to lower the pitch and the cutoff set to the eight o’clock position. Both these changes emphasised the lower‑frequency component of the sound. There are plenty of other options to explore. For example, if you wished to make the sound more assertive or aggressive, the Closed HH 3 engine’s Dist section or the more flexible PadFX Distortion module could be explored.

Stuck In Reverse As mentioned earlier, Drum Machine lets you base a layer on an audio sample. I’ve not exploited that possibility in the

examples above, but there are plenty of processing options, with low‑/high‑pass filtering, pitch adjustment, a full filter section and adjustment of the sample start/ end points and the amplitude envelope. You can also reverse samples but, unlike Backbone, it’s not possible to control the amplitude envelope in a way that syncs to the project tempo — so, if you create a riser effect (for example), triggering it to reach its peak right on your target beat/bar position can be a bit hit-and-miss. Eagle-eyed Pro users may have also noted that, in its first iteration at least, Drum Machine doesn’t offer a pop-open upper panel to access Cubase Pro 14’s new Modulators. Happily, you can configure Modulators as automation sources for Drum Machine’s parameters and those can be sync’ed to the project tempo, so you could pretty easily configure a pad’s Level control to gradually increase over the course of (for example) a bar using something like the Step Modulator.

Full Hit Kit So, Drum Machine might have been designed primarily with electronic drum sounds in mind, but it also offers enough sound-design features that it can be useful in other musical contexts, and if you’re currently using Elements, Drum Machine might be just another item to add to your ‘reasons to upgrade’ list. With eight banks of 16 pads in a single Drum Machine instance, you could easily build quite a comprehensive single preset containing different types and styles of cinematic hits and sound effects. Yes, there are dedicated third-party virtual instruments that offer this kind of sound palette too, but rolling your own isn’t hard, it’s fun, and it means the sounds you use will be unique to you.

Set with very short Amp Env Decay, the Closed HH engines all provide plenty of options for ‘ticker’ style sound design. www.soundonsound.com / March 2025

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INTER VIE W

Jon Bon Jovi & John Shanks Jon Bon Jovi has stayed ahead of the pack for more than four decades, with the help of long-term collaborators such as producer and engineer John Shanks.

Jon Bon Jovi and John Shanks have been working together now for more than 20 years.

“One of the important lessons I learned in those early years was that there was nothing to fear about the red button,” Jon Bon Jovi recalls today. “It’s there for you to experiment, so that you are able to try another take and another idea. When I started in the studio, I didn’t know what a comped vocal was. I remember saying to an engineer in 1980, ‘I’m sorry if you want me to sing it four or five times,’ and he was like, ‘No, that’s how you do it.’ And I was like, ‘Oh! I thought everything was off the floor?’ So, I couldn’t have

been so green, and though I never became an actual engineer, I did learn and understand what life was like making a record and how you went about it. Like, how an arrangement worked, how getting tones worked, and that it all mattered when it came to making a record. “Having been in the room with the masters and their assistants too, like Bob Clearmountain, Neil Dorfsman, Scott Litt and Jason Corsaro, who were all there at the start of the Power Station, benefited me greatly too. And I had a band, and

J O E M AT E R A

n 1980, the young Jon Bon Jovi began his journey to becoming one of the world’s biggest rock stars by running errands at New York’s Power Station recording studio, operated by his cousin Tony Bongiovi. Observing stars ranging from Queen to Chic at close quarters gave Bon Jovi an education and insight into the world of recording that would hold him in good stead in the years that followed.

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the band needed to be recorded, and the assistants and the lower‑ranking engineers also needed time behind the board. All these guys were on staff at the Power Station and were making a flat $250 a week to record whoever was there, so none of them were getting points or making big bucks working for the rock star band. So, I had the benefit of going in late at nights and on weekends while they were learning their craft, and I was learning mine.” Bon Jovi also credits those early years learning the craft in the confines of the recording studio with his approach to production, something which came to the fore when he shared production duties with Canadian rocker Aldo Nova for his 1991 album Blood On The Bricks, on which Bon Jovi played a major role in helping shape Nova’s musical vision. “When it comes to production, it’s about how and what I am able to add to that equation and to the songs,” he says. “If I’m able to add anything to the song and it is still genuine for the artist, then that is a great gift of a producer, because it’s about helping the band create their sound, and not making the record the producer’s sound or vanity project or whatever descriptive word you want to use. So, for me, at the end of the day, a great producer is a collaborator.”

record-making, and Bruce facilitated that. He captured what we could do live on record that wasn’t being captured properly on the first two records.” The experimental spirit that Bon Jovi had gleaned from his early years at the Power Station was another factor in the album’s success, particularly on the anthemic ‘Livin’ On A Prayer’, which featured a talk box, a guitar effect first popularised in the ’70s. “We were kids of the ’70s, so that was really easy and evident, but it wasn’t as prevalent as it was by the time 1986 came around,” says Bon Jovi. “Richie [Sambora] was very cognisant of Joe Walsh’s using it, and Peter Frampton of course, but it hadn’t been used or spotlighted in that way for a number of years. So, when the opportunity came up, we both were like, ‘Yeah, good idea!’” New Jersey, which arrived in 1988, continued Bon Jovi’s upward trajectory. The grunge-soaked ’90s were greeted with 1992’s Keep The Faith and 1995’s

These Days, and the new millennium with 2000’s Crush and 2002’s Bounce. Each successive release showcased the group’s knack for writing stadium-sized anthems whilst refining and evolving their sound.

John & Jon In 2005, John Shanks, a producer, guitarist, and songwriter with a background in both pop and rock as well as country, came on board to assist Bon Jovi with production on the group’s ninth studio outing Have A Nice Day. Twelve years later, Shanks’ role would also expand to touring rhythm guitarist for the group. “Working with John was a huge shift for the band, because by that point, we were ready for a change,” recalls Bon Jovi. “I think back to the Crush album and the passing of Bruce Fairbairn, who was going to get back together once again with Bob Rock and do Crush, and I remember coming home from Europe

For initial demos, John Shanks will track scratch guitars to Logic on his ‘road rig’.

Making The Difference

Photo: Lise Pedersen

Bon Jovi’s debut single ‘Runaway’ was recorded at The Power Station in 1982, with session musicians who frequented the studio. The single later won a radio competition run by a local New York radio station in search of the best unsigned group and quickly became popular on the station’s playlist, prompting the singer to assemble a group. With a line-up secured, Bon Jovi re-entered the Power Station to record their 1984 self-titled debut album. Its follow-up, 7800° Fahrenheit, appeared the following year and was their first album to be certified gold, before 1986’s Slippery When Wet brought commercial success on a massive scale. Jon Bon Jovi credits Slippery When Wet producer Bruce Fairbairn as the catalyst in finally capturing the potential of the band on record. “The big difference between our first two records and the third one was that in our early years on those first two records, Tony Bongiovi and Lance Quinn and us were still learning the ropes,” recalls Bon Jovi. “But by the third record, we had really figured out www.soundonsound.com / March 2025

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INTER VIE W

JON BON JOVI & JOHN SHANKS

John Shanks’ full-fat guitar tracking rig is a no-holds-barred affair, with multiple amp heads, remote cabinet switching and some seriously tasty outboard.

where I was doing a film and literally walking in my home, with my coat still on, and the phone was ringing. I grabbed the phone before I took my coat off to hear the news that Bruce had passed away. We had been exchanging all kinds of notes on the pre-production for what was going to be the start of that album. We ended up doing that record myself, Richie, and a guy named Luke Ebbin, and then because of the success of ‘It’s My Life’ and that album, we used Luke one more time on Bounce and then post 9/11 we thought, ‘OK, time to move on from that.’ “And when we met John, it was a shift in everything from a production standpoint, as he brought a collaborative new sound. He was the hot kid too: producer of the year and producing all these young Michelle Branches of the era, working with Sheryl Crow and a lot of big hit records. And he was a guitar player, too, which was also really different as we hadn’t worked

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with a guitar player since Lance Quinn. Bruce Fairbairn was a trumpet player, so he couldn’t play guitar. Bob Rock was a guitar player too when we were making Keep The Faith, and Luke Ebbin was a bit of a keyboard player. So, when we went in the room with John, he was not only a great guitar player, but also able to collaborate in the songwriting. So, this was a first with a producer co-writing a song together, because John could actually add something to the process, which I really loved.”

Home Cooking “When Jon invited me to work with him and Richie [Sambora] on ‘Have A Nice Day’, that was pretty much the first song we wrote, the three of us together,” remembers Shanks today. “And it was great because Jon kind of comes from where I come from, which is kind of oldschool most of the time. I mean most of the time when we’re writing it’s just a legal

pad and acoustic guitar and no bullshit, because you can’t hide behind the track or the beat or the production. And, I’m always the one in the room that’s like ‘Now, let’s go put it down.’ But Jon’s like, ‘Not yet, not ’til I cross the ‘t’s and dot the ‘i’s.’ “So, when we wrote ‘Have A Nice Day’, what was beautiful about that particular day is that once we wrote it in his kitchen, the three of us walked across the lawn and into his studio in New Jersey and spent the day recording it. We put it down just enough for us, some acoustic guitars and maybe a beat or something, just enough for Jon to sing it so then the three of us could hone the melodies and really craft the song as songwriters, making sure the key was right, and the tempo was right. And that vocal that appears on the final recorded version is from the vocal we recorded on that day when we wrote the song! “Obviously on the final version, we later overdubbed some ‘oohs’ and ‘aahs’

and the harmonies, and tweaked a word or two, but the lightning in the bottle occurred that day, which also happened on ‘Who Says You Can’t Go Home’. When Jon gets into the zone, everyone’s emotionally connected to that moment, you’re tracking it and everyone is zoned in on that song emotionally, spiritually, whatever. He connects to the vibe of the room and so he knows what his job is and he goes out there and will just sit in the pocket, so to speak.” “We’ve also done it every other way too,” says Bon Jovi. “Where some tracks that we have worked on prior to recording have been an idea that John might have suggested from a tracking standpoint where I’d come up with a chorus for it, others have come from out of my notebook and cassette tapes that I’ve had.”

From The Top When it comes to making a new Bon Jovi album, much time and energy is given to pre-production, with Shanks demo’ing the bulk of the material in his

studio. “Prior to recording an album, I will put together a demo,” Shanks explains. “A Logic template is my starting point, as it’s my go-to writing software. There’s literally 74 things that are in that template. Everything from kick samples, drum snares, hi-hats, closed hi-hats, and 707 Roland drum kits. There are also bass synths, Minimoogs, synth basses, different augmented strings, weird piano sounds, winding vinyl, winding piano, analogue strings, different types of Junos... there’s just so much stuff so it allows me to start stacking everything. “For the guitar session, once I have the structure down, I will usually check what tempo and key feels right for the song and I’ll probably do some double-tracking using an amp or several amps from my vast amp collection that I have, from Marshall Plexis and Dumbles to Fenders and others. And it all goes into my Universal Audio rig, which has a 4x12 cabinet set up underneath my house and which is miked up with a [Shure] SM57.” At the demo stage, Shanks also contributes scratch drum tracks. “Once

The SERIES SlatFusor

I have sketched it out to a certain point, I’ll bring the session into Pro Tools, where I’ll then add real drums. On one of the drum kits I use Royers, and the other I use AKG C414s as my overheads on my kit upstairs. I have two different drum kits in the studio. There’s one that’s set up in the living room that’s more of a big rock kit and another kit downstairs in the real studio. I will use Coles 4038 ribbon mics for the overheads, a Sennheiser 421 on the toms, and an AKG D19C for the kick. I will also add a mono mic on the drums, using a D19C which is kind of what the Beatles used, and will position that either above the kit or between the kick and the snare and then compress it with a [Empirical Labs] Distressor. For the room mics, I will use Gefell UM57s along with an EMI compressor. Then I’ll Beat Detective the drums, put them on the grid and add real bass. Once that’s all done, I’ll take it to the band where it gets recorded properly.” The downstairs setup in the ‘real studio’ is used to record keeper guitar takes once tracking has started in earnest. “When I go downstairs I dump

Modern Elegance Superior Acous�c Panels & Bass Traps

www.soundonsound.com / March 2025

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FE ATURE

ASIO GOES NATIVE! Microsoft Announce ASIO Drivers For Windows PETE GARDNER

t the Qualcomm Snapdragon conference in Hawaii in October, Microsoft surprised us all by announcing a new version of Windows for Arm processors with built-in ASIO drivers for USB 2 audio interfaces. I say surprise, because it marks a big shift from the way Windows has handled audio in the past. The flexibility of Apple’s Core Audio has always been an attractive feature, not least due to its native latency levels, and it’s a feature that lots of Windows-based musicians have been requesting for many years. At present, this new feature is exclusive to this version of Windows for Arm processors but, importantly, Microsoft say they plan soon to embed USB ASIO support in the more familiar desktop versions of their OS — so it’s potentially very big and promising news for the music-making community.

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At the Snapdragon conference, Steinberg’s demonstrated Cubase running on Windows for Arm, using the native ASIO drivers for a low-latency electric guitar performance.

Windows-based musicians have long been asking for better support at the operating system level — and it seems to be very much on its way, with native low-latency ASIO drivers and full support for MIDI 2.0. We reached out to Microsoft to find out more.

The new driver was developed in partnership with Yamaha, the parent company of Steinberg (who developed the ASIO driver protocol) as well as a manufacturer of various USB audio devices and musical instruments, and at the conference Steinberg demonstrated Cubase running on a Snapdragon laptop. They used the native USB ASIO drivers with one of their interfaces to process a live guitar signal with great low-latency performance. Seeing this sort of performance (it’s on YouTube if you’re interested: https://youtu. be/6SALIz428JM?si=c_gAiTYkLWReDaRq) on a native hardware driver in a Windows environment was certainly refreshing. Several notable developments then followed. Shortly after the presentation, a supporting blog post appeared from Microsoft Principal Software Engineer Pete Brown, in which he discussed the

announcements and expanded on the new audio developments, which included not only the native ASIO drivers, but also support for MIDI 2.0. Notably, interface manufacturers Focusrite announced their backing for native Arm drivers. Then, a preview build of Cubase for Arm was released a few days after the event, CockOS launched a preview build of Reaper, and Reason announced that a preview of their DAW would be forthcoming. But what drove this development? Why did Microsoft focus first on Arm processors, and what might it all eventually mean for those using Windows for music and audio? To help me answer such questions, I caught up with Pete Brown, and our interview forms the second half of this article. First, though, I thought it would be helpful to offer some background on the Arm chips supported by this new OS, and to explain why we might be about to

From Little Acorns... Most of us probably know that the M-series CPUs now used throughout Apple’s product range are based on the chip technology of Arm Holdings PLC, usually known simply as Arm. Apple’s new processors have an enviable reputation for both power and efficiency, but the Arm name (ARM until a 2017 rebranding) and their processors actually go back several decades. The underlying ‘silicon’ concept dates back to the 1960s, but the acronym ARM was first used in 1983 by Acorn Computers, when it stood for Acorn RISC Machine. Conceived to handle simplified code that can be ‘pipelined’ in a relatively efficient manner, RISC (Reduced Instruction Set Computer) chips are a competing technology to that used in the Intel and AMD CPUs (known as CISC, or Complex Instruction Set Computer chips) that we’ve become accustomed to using in laptop and desktop computers. Acorn’s first RISC chip appeared in 1987 in their Archimedes desktop computer, which some readers might remember as a more powerful replacement for the BBC Micro in UK schools. But in 1990, Acorn, Apple and VLSI Technology (now NXP Semiconductors) teamed up in a joint

venture called Advanced RISC Machines (ARM), which was later listed on the stock exchange and became the Arm we know today. Along with VLSI, Acorn developed the early ARM cores, but it was Apple who first used one of these chips in a generalmarket product: their Newton PDA announced in 1992 featured an ARM6 chip. The Newton struggled to establish itself in the face of competition from the Palm Pilot, and its commercial failure led ARM to embrace a different business model: rather than create end-user products, they’d license their IP to production partners, who would make their own chips based on ARM’s RISC technology. Starting with the Nokia 6110 GSM phone, which appeared in 1997, their chips found favour in the mobile devices sector,

Photo: PaulVernon1974 — Own work (Creative Commons)

see more of them appearing in ‘serious’ music-making Windows machines.

The Acorn Archimedes: the first widely available commercial computer to feature an ARM processor.

RT OS for ARM-based mobile devices. This was used for the Surface tablets and a small number of third-party devices, all built around Nvidia Tegra 32-bit ARM processors. Ultimately, the platform’s inability to run popular PC software meant that these devices were largely passed over for more traditional x86-based laptops. Then, in 2016, Microsoft launched a 64-bit ARM version of Windows 10 for Qualcomm’s Snapdragon devices. Hardware shipped over the following years from Microsoft, ASUS and HP, amongst others, but the platform failed to capture users’ attention from the more powerful PC laptops and desktops already on the market and, again, it wasn’t regarded as a commercial success. Meanwhile, Apple had been successfully using their own custom ARM chips in various devices, including iPads and iPhones, since 2009. But in June 2020 they signalled that they’d be moving from Intel to Arm-based processors for their more powerful desktop and laptop machines. Their M1 chip first appeared in a product at the end of 2020, and the M-series chips are now found throughout their product range. Importantly, these Apple chips have enjoyed a good reputation for both power and efficiency, and the products have proved commercially successful.

“Microsoft plan to embed USB ASIO support in the more familiar desktop versions of their OS.” and today they are reportedly found in 99 percent of smartphones. Of course, we’re more interested in their potential for ‘serious’ machines that might be used for music making, so it’s perhaps of greater interest that, in 2012, alongside Windows 8, Microsoft brought out the 32-bit Windows

Right Here, Right Now

A number of companies announced that they’d be developing their popular software and hardware audio products for Arm-based Windows platforms — and more are sure to follow suit.

This brings us to the present day. As you’ve probably gleaned, in the past it was often software incompatibility that limited the user appeal of older Windows-based ARM devices. But the emulation layer has matured over the years, and there’s also been a steady increase in PC RISC chip performance. Having the flexibility to run existing applications through the emulation layer should help the platform to expand, www.soundonsound.com / March 2025

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Photo: Binarysequence (Creative Commons)

Apple’s move from Intel to Arm-based CPUs may seem relatively recent, but along with Acorn they were one of ARM’s founding companies and began selling their first ARM-powered device, the Newton, way back in 1993.

and with greater uptake the incentive for developers to compile software specifically for Arm should increase too. With the barriers to wider adoption falling away, various parties (including silicon behemoths AMD and Nvidia) have announced their entry into the Arm computer market, and new hardware is on the way. Small wonder, then, that Microsoft seem keen not only to support the newest Snapdragon X Elite-equipped laptop models, but also to deliver the sort of accessible, more streamlined experience that should make the platform attractive for typical computer users.

Pete Brown & The ASIO Team That’s not the briefest of technical backgrounds for a music-focused mag, I know, but I think it’s important context for what follows. Through several roles since joining Microsoft in 2009, Pete Brown has often been a public face for the company,

with official titles like Program Manager, Evangelist and Developer Advocate. However, he has always remained first and foremost an engineer, and he spends much more time writing code than any of those official role titles may suggest. When I caught up with Pete, I started by inviting him to explain how he’d been involved in this latest project, as well as in the music/ audio side of things more generally. “My day-to-day [work] at Microsoft,” Pete explained, “involves working with several partner teams like the Audio team, Surface engineering and others, to help make Windows a great place for musicians to be productive and happy. I co-engineer with different teams on things like the new MIDI stack in Windows, where I wrote the app SDK, tools, loopback and app-to-app transports, all of the MIDI 2.0 protocol management and the upcoming Network MIDI 2.0 and Bluetooth MIDI 1.0 transports. “Alongside this, as Chair of the Executive Board of the MIDI Association [the standards body and trade association for MIDI], and as someone participating in the working groups there, I also do my best to help ensure that any new standards will work great on Windows, and that Windows-based musicians are represented in the discussions. “In addition, I run the Windows

Developer MVP [Most Valued Professional] program, which recognises advocates and leaders in the developer community, and also tend to be heavily involved with our Build developer event, run every Spring.” One of the more recent ventures keeping Pete’s team busy has been the Windows for Arm project discussed at the top of this article. This included offering support to software developers who are looking to move to the new platform and develop native solutions. I asked Pete to tell us a bit more about the team working alongside him: “The all-up lead for Arm64 on our team is Marcus Perryman. He deserves a shout-out for all the great work he’s done to help unblock the vast ecosystem of applications and middleware for Windows on Arm. I also work with Jason Cook here, who does a lot of partner management, specifically on creative apps. I’ve been providing guidance to partners on which compilation targets to use for things like VST plug-ins, hosts, and also to help unblock them when they run into third-party components that don’t yet have Arm64 support. “I’ve also been working closely with Qualcomm and other silicon partners to support creators and specifically musicians on Arm64 PCs. PJ and Rami at Qualcomm have been huge advocates for musicians on the platform, and together we worked to create the musician theme at Snapdragon Summit... complete with musicians performing live on Cubase on Snapdragon PCs, and even a skit in the Jimmy Kimmel show. It’s been fantastic working with them, and especially with PJ, who is a huge fan of this area. I knew I found the right partner there when I saw his EVH guitar hanging on the wall behind him on our first call.”

Why Now? Why Arm? Next, I wanted to tackle what I consider ‘the big question’. Right since the creation of ASIO, having a low-latency driver option built into the OS has been a very popular request, and it’s something that could really simplify the setup process for many people recording and processing audio. Just how and why, after all these years, did support for ASIO become a focus for Microsoft? Pete started by explaining some of the practical difficulties they’ve faced over the years. “Windows is a general-purpose operating system that has a lot of different types of users. On the PC,

With ample power and superb battery life, Snapdragon X Elite-equipped laptops now have the potential to be great portable music-making machines.

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you see everything from casual media consumption to high‑FPS gaming, to visual and audio art. Each has different requirements for audio. For example, for media consumption like watching a video, you can have large audio buffers which minimise the number of times rendering and output threads have to wake up to service the buffers, which results in much better battery life. For gaming, FPS is king. Audio is important, but secondary. “For live music performance, of course, we want audio to be as near-real-time as possible, without glitching, with all other concerns being secondary. Add to that that musicians have very different I/O needs, and don’t necessarily want an additional codec or on-chip processing to, say, compress the audio so it comes out of laptop speakers better. Finally, musicians typically want to lock a sample rate and not have any up/downsampling or bit-rate translation, whereas normal Windows app expectations are that they supply the data in the format they have, and Windows will resample and mix as necessary to match the hardware.” He then explained how he and colleagues started to make progress, and why they started with Windows for Arm. “For years, I’ve been working with the audio team, kicking around ideas for how we can make low-latency audio a reality on Windows, without breaking existing applications, harming battery life or the average user experience, while adding things like device aggregation. Back in 2015, we introduced a low-latency WASAPI mode, which didn’t require exclusive mode but still allowed lower latencies than the usual approaches. Some applications, like Cakewalk Sonar, tried that out but most didn’t, because the ASIO API was just so much easier for them to work with for pro audio tasks. “Meanwhile, Arm64 devices ran the risk of receiving limited pro audio support because many hardware companies were not creating Arm64 native ASIO drivers for in-market products, with even fewer considering creating Arm64 drivers for older devices (which are no longer sold but may still be popular). It’s important to note that while apps can run under emulation, kernel drivers cannot, so they must be developed/ recompiled specifically for Arm64. “So I proposed that we ‘meet them where they are’, and officially adopt ASIO as an interim measure on Windows. The intent is to unblock music creation on Windows on Arm devices for all USB Audio Class 2 devices, whether they are current in-market [audio] devices or older models that are no longer sold. This means app developers wouldn’t need to adopt a new API, and audio devices will ‘just work’ on Arm64 PCs.” But he also stressed that it isn’t just about low-latency audio. “Although we often refer to it as ‘the ASIO driver’, the ASIO component is just one piece. The new kernel driver coming out of the project will replace our current in-box USB Audio Class 2 driver, which has higher latency and doesn’t support the required number of I/O ports. Besides being a great low-latency driver, the new driver also incorporates best practices for handling power management events without audio glitching, and uses the newer ACX (Audio Class eXtensions).”

Yamaha I was curious to understand how this all lead to Yamaha’s direct involvement as a partner, helping to implement the native ASIO support. “We’re really excited to have www.soundonsound.com / March 2025

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Ian Fitchuk Working with Kacey Musgraves catapulted Ian Fitchuk from respected producer to one of Nashville’s busiest songwriters. PAUL TINGEN

“F

Ian Fitchuk at Blackbird Studios in Nashville.

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or about 10 years I was very much outside the mainstream,” recalls Ian Fitchuk. “It allowed me to work as a producer and session musician with very different types of bands and artists. I had a respectable arc of work and a lot to be grateful for. At one point I was like, ‘Wow, I can support my family and don’t need to get another job, and while I don’t live in a mansion and have a car that needs work done all the time, life is pretty good.’ “Then, in 2016, everything changed. I signed my first publishing deal, with Razor & Tie, which turned into Concord. I had played on Kacey Musgraves’ second album, Pageant Material [2015], and one day Kacey’s publisher reached out about co-writing. I invited my friend Daniel Tashian, a songwriter and multi-instrumentalist I admire, because I thought that things would click between the three of us. My guess was accurate. “The three of us co-wrote and co‑produced Kacey’s fourth album, Golden Hour [2018], which became a big success. It changed my life. Golden Hour was not defined as only part of the country genre. I remember chatter about the record changing country music, and I’ve since had a lot more interest from artists outside the country world. This works for me, because I love working on all kinds of music.

“I’m interested in being challenged, and I’m grateful that the first thing I became known for on a big level was something that I had been able to put a lot of myself in, musically and aesthetically. Some people call it organic, timeless music. Music with heart became my calling card. When people reach out to me now as a co-writer and producer, they’re not necessarily looking to make a country record. They’re looking for my ability to tap into whatever their essence is. It means that I am learning new things, and starting from square one, every day. “I’m working with artists who are not only interested in creating songs, but also in the concept of an album, the presentation, the aesthetic and the artwork. They’re thinking about how they’re going to perform these songs. This alleviates me from feeling that I have to come up with some sort of magic potion for them. All I have to do is hear the things they’re saying and the cues they’re giving me musically, and then develop ideas. Once you have that rapport with somebody, you don’t need to go to a million different places or have dozens of songwriters.”

A true multi-instrumentalist, Ian Fitchuk often plays nearly all the instruments on tracks that he produces.

Keeping It Real As well as going to number one on the US and UK country charts and becoming a Top 10 hit on the album charts, Golden Hour also earned Musgrave, Tashian and Fitchuk two Grammy Awards, for Album of the Year and Best Country Album. The trio have since collaborated on Musgraves’ equally successful albums star-crossed (2021) and Deeper Well (2024). Fitchuk also received Grammy nominations in 2021 (Best Country Album for Little Big Town’s Nightfall), and 2022 (Best Country Song for Musgraves’ song ‘camera roll’). Deeper Well is this year nominated for Best Country Album, and Fitchuk for Producer of the Year, Non-Classical. Other achievements listed for this nomination are co‑writing and co‑producing albums by Stephen Sanchez (Angel Face) and Maggie Rogers (Don’t Forget Me), and songs by Beyoncé (‘Amen’), Still Woozy (‘Lemon’), Role Model (‘Oh Gemini’) and Leon Bridges (‘Peaceful Place’), and producing Musgraves’ version of ‘Three Little Birds’ and Leon Bridges’ rendition of ‘Redemption Song’ for the Grammy-nominated soundtrack for Bob Marley: One Love — Music Inspired By The Film.

Despite very much not pigeonholing himself as working exclusively in country music, 25 years of living and working in Nashville have naturally informed

musicians willing to share their experience and their knowledge. All you have to do is pay attention and listen and not think that you have it all figured out, and be brave enough to try things that are outside of your comfort zone. There have been plenty of times I’ve been called to write a song or to go play on something where I feel completely inadequate or that’s out of my wheelhouse. But I show up, and by nature of putting yourself in uncomfortable, stretching situations, you widen your net of experience.” Also in keeping with Nashville’s modus operandi is Fitchuk’s preference for physical rather than virtual. However, instead of the band-in-the-room session approach common in the town, Fitchuk likes to play many of the instruments himself.

Ian Fitchuk: “A lot of the time, things that I think are just demos for the sake of getting the song written end up standing the test of time and being released.” Fitchuk’s approach. It shows, for example, in his preferences for working in a room with others. “One of my favourite things about making music is the way that it connects me to other people. I’m not prolific on my own. I’ve always been most productive when I’m working with other people. So I met a lot of good role models, and

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Kenny Greenberg and Matt Rollings, and I recognised Matt’s name from liner notes of CDs that I had. Béla Fleck was also on the album. I was impressed, so I quit school to play keyboards and tour the world with them, opening for Béla Fleck, John Mayer and others. “I enjoy travelling and music, but after one and a half years of touring I realised that the life of a touring musician was not for me. So I left the band. When in high school bands I had played keyboards and drums, and also had taught myself enough bass and guitar to make songs. So from about 2002 to about 2010 I did a lot of work in Nashville as a session player. Certain people would know me as a drummer, certain people knew me as a keyboard player, others knew me only as a bass player. “During this session work I was on the floor with three or four other musicians, and I learned about whatever instrument I was playing, and I learned about what the other musicians were doing. So the next time I was playing that instrument, I was like, ‘OK, I noticed he used this bass,’ or ‘I noticed that the drummer avoided using cymbals the entire session, which really brought out the tonality of the drums,’ or ‘Wow, he played really light,’ or ‘This or that chord voicing on guitar.’ I had a front seat for really long. “While working as a session player I also got to know some of the big producers in Nashville, like Jacquire King and Jason Lehning, and I learned about production and working in the studio. Also, a friend called Josh Moore was producing majorlabel records in nice studios in Nashville, and I spent a lot of time watching what he was doing. I liked the way that everything came together in the studio, the pairing of musicians, the communication with artists, and so on, and went into production myself. “I made a bedroom recording with an artist named Griffin House, using the first version of the Pro Tools HD on a rig a friend of mine had. We also had a Blue Kiwi microphone and a Focusrite pre, and that was about it. We made an EP with Griffin House that got picked up by Nettwerk Records, and this brought me out to LA and introduced me to more of the music business side of things. That’s when I realised that I could enjoy producing and putting albums together, even as I was only successful in the margins.”

Expert Help At this point, Fitchuk’s story has arrived where this article started, at the end of

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Konrad Snyder.

his 10 years working as a producer and session musician outside the mainstream. Before Golden Hour, he recalls, he “wasn’t doing much songwriting, or part of the Nashville songwriting community”, but he has since contributed to many recordings as co-songwriter, multi-instrumentalist, and co-producer. It’s a remarkable number of hats to wear, made easier by the fact that he leaves the engineering and mixing largely up to others. For this reason he does not have his own studio. “As I mentioned, when I first started producing, I was working with a friend who had a Pro Tools HD rig, and was very capable as an engineer and handled the computer. I have learned Pro Tools and Logic enough to be able to get ideas down, but other than that I’ve kept my distance from the computer. I’ve always been around so many people who have amazing studios

that I felt no need to have my own. I have thought about it, but I like the change of pace of going to different studios. If I had my own, I might get sick of it pretty quickly. “I try to leave the tech details to the experts. I have worked with many engineers who day in and day out are obsessing over the nuances of the innumerable ways to make great recordings. I’m sure that some of the records that I produced in the early-mid-2000s could be ripped apart sonically, but there was still something that musically resonated with people that goes beyond the fidelity. We’re making music, right? There’s gonna be bleed. There’s gonna be imperfections. There’s going to be buzzing guitar lines. As long as that’s coming from a place of inspiration and not just lack of care, I think it’s OK. “I know when something doesn’t sound good to me. I might even be able

to tell you that it’s not the right mic, or to identify that it sounds over-compressed. But with the engineers that I work with I don’t have to use language like that to explain what I’m hearing, or to translate what the artist wants to hear. I can be the middle man between musical language and more technical speak. I can describe things, like ‘We’re battling a high‑mid frequency here,’ or ‘She really doesn’t like the way that mic enhances the esses.’ I can facilitate those conversations and make people hear each other. I have my plate full already, thinking about the story that the song is telling, or managing the expectations of the A&R, or of management, or consoling the artists with whatever existential thing they’re going through. It’s all hands on deck. So when I find engineers who are really good at what they do, I let them do that.”

idea. Kacey takes recording as seriously as she takes writing, so it’s pretty common that we get a final vocal the day that we’ve written the song. We may come back to and touch it up later, but in many cases her final vocals were recorded with a guide guitar with maybe some rhythmic element or a few textures, like piano or something else, that demonstrate the feeling and the aesthetic of the song. “When we have one song that feels good, it gets the ball rolling, and then there’s the next, and so on and so forth. With Kacey, we do the backing track recordings later on. Most of the time we end up in commercial studios, whether Sound Emporium or Blackbird in Nashville, or Electric Lady in New York. Deeper Well

was mostly recorded during multiple trips to Electric Lady.”

Unforgettable Almost the whole of Maggie Rogers’ acclaimed album Don’t Forget Me was co-written and co-produced by Rogers and Fitchuk. The latter plays the vast majority of the instruments, and the album sounds like experimental, arrangement-driven indie pop. “Deeper Well and Maggie’s album were mostly recorded in the same top‑floor room at Electric Lady. While we did some writing with Kacey in New York, almost all of Maggie’s album was written there during very rapid-fire sessions. Maggie and I wrote the majority of the songs in a two‑ or three-day span and I recorded

Plum Job Songwriting sessions with Kacey Musgrave usually take place at Daniel Tashian’s studio. “Daniel has a studio called Royal Plum in his backyard. It’s a converted garage, which has everything you need: a drum kit, lots of guitars, lots of keyboards, a handful of great microphones, and a very cosy living room feeling. Daniel and I are very similar in that we both feel pretty confident on most rhythm instruments: piano, guitar, bass and drums. We also sometimes work at our friend Konrad Snyder’s studio, an engineer and mixer who I work with a lot. “Kacey, Daniel and I sit with one or two guitars and/or a piano. She’ll come in with a handful of titles, and we write towards them. This is a common songwriting strategy in Nashville. It’s not only important to her to have a title that is interesting or sounds good, but also, how do you write towards it in a way that makes sense, and has an interesting journey? So we’re spinning around a concept or a title until we get a clear lyrical idea that we can latch onto, and from there it’s a matter of finding what matches that feeling musically. So we’re oldschool, where the song gets written around a piano or a guitar or both. It’s very rare that we write to a pre-existing track. “So the first step is writing songs, and we initially record them as little voice memos on the iPhone, making sure that we remember what we think is good. Typically towards the end of the day, we record a guitar/vocal. It’ll be really quick, just to make sure that we’ve preserved the

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months later and tried to replace and improve on what we had recorded. But the minute we tried that, the demos started to lose some of their identity. Every step forward felt like a step back. “With many artists it’s common to be in the studio for the day with them and explore song ideas, and the moment something starts to catch, I’ll be putting things down as musical placeholders. By the time the song has been written and we later try to re-record or perfect the demo, we realise that a lot what was meant to be replaced has a feeling that is hard to recreate afterwards. So a lot of the time, things that I think are just demos for the sake of getting the song written end up standing the test of time and being released.”

to let go in the end, again happy to leave things to the experts. “With regards to vocals, I’d like to say that I like to capture the vocals while we’re going. I don’t like that moment when you have all the basic tracks recorded to scratch vocals, and then there’s this big pressured time when you need to get the final vocals. Instead I like to record the vocals while the arrangement is being fleshed out and the production created. The chains that we use are very interchangeable, but usually some beautiful mic like a Telefunken 251, a Neumann 47, or an AKG C12 or C24, but there are artists who love to hold a mic, which always leads to a Shure SM7. “Most of the artists I work with are performance-based, and don’t need a lot of comping and editing later on. They also like to be involved in that final comping process. I don’t mind comping vocals, but I prefer for it to be a collaborative process. We do tune sometimes. I don’t like it when you can hear it, and it sounds like you’ve tried to make a not very good singer palatable. But when there’s an amazing performance that has a few notes that need massaging, I don’t see that as different to having a great guitar part and punching in a mistake or out-of-tune note. I prefer to do a punch in with a singer, but when we are using tuning, it’s mostly [Celemony] Melodyne, and maybe a little bit of Antares Auto-Tune once in a while. I try to make it as musical as possible. I challenge anybody to point out sloppy tuning on the records that I make. “As for adding reverb, I actually like the natural reverbs that both Sound Emporium and Blackbird have. There are a million great plug-ins. All the UAD and Waves and Valhalla stuff is awesome, but for me it’s not as good as natural reverb. At Konrad Snyder’s studio, where we’ve recorded a lot, he has a kitchen outside of the control room with a great ambience. I love it when you can achieve dimension and space with the actual sound of the room you’re recording in. It’s also why I like recording in different studios. I’ve been at Echo Mountain Recording in Asheville, and Muscle Shoals, and EastWest. With all these different studios, and even the regular houses that I’ve recorded in, using the room you record in as much as possible as an instrument is more exciting to me than a one-size-fits-all plug-in.”

Keep It Natural

Letting Go

Ian Fitchuk played most of the instruments on Maggie Rogers’ album Don’t Forget Me.

most of what’s on the record during those first couple days. “We had a creative exercise where she’d play a track from a legendary album, like Rumours or Thriller, or whatever. I would quickly play something that she would react to that immediately generated melodies and ideas. As soon as we had a chord structure that felt like a verse or a chorus, I would put it down on guitar. She would then go into another room to write her parts, while I fleshed the arrangement out, still not having heard the whole arc of the melody. Thirty or 45 minutes later, she’d get on the mic and sing. Two hours later, we’d be onto another song. “At that point we were thinking, ‘Wow, these songs are coming really fast. Let’s come back in a few months to record them properly.’ We even discussed putting a band together to track the album in the old-school way. We booked more time at Electric Lady a couple of

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Fitchuk gets quite involved in post-production processes, whether vocal production or rough mixing, but is happy

Many of the tracks and albums (co-)produced by Fitchuk are mixed by Shawn Everett or Konrad Snyder.

Konrad Snyder: Nashville Tracking

Ian Fitchuk regularly works with Nashville engineer, mixer and producer Konrad Snyder. Snyder has an engineering credit on all but one track on Deeper Well, and mixed 11 of the 21 songs of the extended version. He elaborates on some tech details for two of the album’s singles, ‘Cardinal’ and ‘The Architect’. “On ‘The Architect’. Todd Lombardo engineered his own guitars. He used a combo of a Neumann U67 mic and a Coles mic. In the mix we ended up favoring the 67, and we compressed with the [UREI] LA-3A. Kacey’s vocals were also recorded with a 67, through an older Brent Averill 1084 and then through an Undertone Audio Unfairchild. The reverb of choice was the Bricasti. The piano in the song is a Yamaha upright recorded with the new Neumann M49 reissues, going through the Rupert Neve Shelford, and again the Unfairchild. The banjo was recorded through a Neumann M49, the Shelford, an 1176 and the Echofix EFX3 chorus echo. “On ‘Cardinal’ I used an AKG D12 on the kick, a [Shure] Unidyne 545 on the snare, an SM57 on the snare bottom, the EV RE11 for the

“I never hand things off to a mixer scared that they’re going to destroy it,” Fitchuk comments. “I’m always excited to hear where they take it and to hear risks, to hear musical decisions that go beyond just trying to preserve what’s already there. That to me is a big part of post-production. Sometimes you want a mixer that’s just going to lightly touch what’s there. But I like taking a plunge

toms, Neumann KM56 for overheads, and Coles 4038 for the room sound. All mics went through the Inward Connections MP820 mic pres, bussed into an 1178. For the bridge we sent the drums into an Otari 5050 and shifted it to half speed to take them down an octave. “The hammered dulcimer was recorded with a Neumann KM56, going through the MP820 and a [UREI] 1176 Rev D. The acoustic guitars went through a AKG D19, going into the MP820, 1176 Rev D, and [Studer] J37 tape. Blackbird actually got a J37 a week before we came in. And it was a one-inch two-track. It sounded unbelievable. We took some time while we were there and ran a lot of what we recorded at other places through their gear. The bass went directly into the MP820, and I also used a vintage [Teletronix] LA‑2A. Kacey’s vocals were recorded with an AKG C24, Neve 1084, Heritage Audio Motorcity EQ, and the Unfairchild. The reverbs were a mixture of an EMT 240 plate and a Bricasti M7.”

into the deep end and hearing things done by a mixer that take the music that you’ve been kind of obsessing over to a different place. It’s nice to have somebody shake that up a little bit and challenge what you thought you were doing and blast it out and exaggerate it in some ways. “I love a lot of modern music, and also pop music that has been scrubbed

and tightened and tweaked to every little detail. From time to time I will employ some of those techniques in my own way. I don’t like things to be messy or dirty just for the sake of it. I think things can feel good if they’re really tight and really organised. I also think things can feel good if they’re a little dishevelled. It’s really about what emotion you’re trying to convey.”

THE FUTURE OF GLOBAL RF

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Photo: Joe Matera

Phil Manzanera • Roxy Music ‘Amazona’ J O E M AT E R A

ell known as guitarist with glam art-rock pioneers Roxy Music, Phil Manzanera has also etched out parallel careers as solo artist, session musician and producer. Over the course of his 50-year career Manzanera has contributed to over 80 albums, and his production credits include albums by Split Enz, John Cale and Pink Floyd guitarist David Gilmour. Asked to dissect a favourite sound, he chooses the guitar tone he created for Roxy Music’s ‘Amazona’. “The guitar sound on that track is totally unique, even though I will say it myself. It’s one which I’ve never been able to repeat. When we started recording the album, I was faced with a dilemma because Brian Eno had left the band, and on the previous two albums, I had used Eno to treat my guitar through his VCS synthesizer, and the

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combination of my direct sound and the treated sound created a unique sound. “You have to remember that in those days you could only get really a fuzzbox and wah-wah pedal and some kind of echo unit, and so to create unique sounds was very difficult. And I thought that in order for me to be able to treat my guitar myself, I needed to have some specially made guitar modules by the same people who had made Brian Eno‘s VCS3 synthesizer.

Phil Manzanera: “What’s on the record is a unique recording, and I defy anybody in the last 52 years to come up with a similar sound.” “So, I rang them up and they actually made me two modules which had the envelope shaper and a modulator. But because they had knobs on them, I had to work out how to control it using a different method, which was by the use of DeArmond volume pedals, which went up and down sideways to control the filter

and the modulation. I also had a special unit made with a pin board on it so that I could send signals to different areas of the VCS’s three modules. I could then control this from the volume pedals. So, it came out of this and into a Revox tape recorder which had been modified with sel-sync and vari-pitch, which I also controlled with another DeArmond volume pedal.”

One Time Only “In theory this should’ve been quite a stable unit but in practice, it was incredibly unstable, so when I played the guitar part on this track, it worked brilliantly, but only once. I was never able to repeat it, so what’s on the record is a unique recording, and I defy anybody in the last 52 years to come up with a similar sound. Of course, having the fantastic producer Chris Thomas work on our album — a gentleman who had worked with the Beatles and George Martin — meant that the mic placement on my Hiwatt amp and WEM speaker cabinet and also the Fender Twin Reverb was immaculate, and helped create that unique sound that’s on the record too. “I have to mention too, that it was recorded at AIR Studios in London, the studio that George Martin had set up after he decided to leave Abbey Road. And as an extra note, Ice-T sampled it, to create a hip-hop track, ‘That’s How I’m Livin’ in 1993.”

www.soundonsound.com January 2025 Period : 01-31 January 2025 [source: Google Analytics 4] Stats updated 02 February 2025

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ON TE ST

SAMPLE LIBRARIES

Zero-G Elements Kepler Kontakt Instrument

HHHHH SOS regulars will undoubtedly be familiar with Zero-G and Stefano Maccarelli’s collaborations through the popular Ethera titles. Their latest joint venture is Kepler, part of the Elements series, and very much aimed at media composers. The Elements series is an interesting concept, with each title intended to provide composers with a targeted set of virtual instruments suitable for a specific type of, or ‘element’ of, film, TV or video game scoring. In this case, Kepler is inspired by the sounds and moods of film scores such as Dune, Tenet and Oppenheimer, and the aim is to give you a complete scoring toolkit within a single library. That’s a suitably ambitious target, especially in a product at this very budget-friendly price. So, just what does Kepler have in its toolbox? Kepler provides over 30Gb of 24-bit/48kHz samples for the full version of Kontakt (v6.8 or above) and these form the basis of some 20 Kontakt instruments. These can be divided into three groups; the Core Designer, the Texture Designer and a series of additional individual instruments that span various drums, percussion, cinematic hits, whooshes and sound effects, various atmospheric vocals, and a bass instrument. In use, the Core Designer and Texture Designer provide a UI that allows the user to blend three sound sources in various ways. They are, essentially, multiple instruments in themselves as both have underlying sample collections that can be loaded into their three‑slot UI, and each is supplied with well-stocked collections of Snapshots (Kontakt presets) to get you started. The Core Designer presets cover an impressive sonic range, including some very cinematic drum/percussion-based rhythms and designed sound loops. All the rhythmic elements will sync to your host tempo. In contrast, the Texture Designer is focused of playable instruments and contains a huge selection of epic pads, soundscapes, vocal and string-based textures and a whole lot more. The straightforward but functional front ends provide plenty of tools to tweak the sounds, and a simple but effective arpeggiator. The additional individual instruments provide some suitably atmospheric vocal

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tools but also some very useful hits, whooshes/risers, percussion and sound effects that would provide plenty of options in a modern drama or epic sci-fi context. They round out the collection very nicely. Of course, there are plenty of sample libraries aimed at modern hybrid film or video game scoring so, while Kepler perhaps isn’t unique in aiming at that sonic target, it certainly isn’t out of place alongside some of the obvious competition. However, it does hit that target in a very accessible form; put together an instance of the Core Designer and Texture Designer, accent it with sounds from the individual instrument group, and you could easily have the basis for a complete musical cue. Repeat for 90 minutes, and you could have yourself a coherent film score. Elements Kepler delivers excellent value for money and composers at all experience levels could find something to enjoy here. John Walden $109.95 www.zero-g.co.uk

Soniccouture Tonal Drums Kontakt Instrument

HHHHH Soniccouture are fast becoming one of my favourite software developers. Their unique and innovative approach to virtual instrument design, along with an incredible attention to detail, really set them apart. The latest addition to their ever-expanding catalogue, Tonal Drums, maintains their usual high standards. Tonal Drums is an NKS-ready Kontakt Player instrument that focuses on the pitch and tuning of a meticulously sampled Canopus Yaiba drum kit, featuring a 22-inch kick drum; 14-inch snare; 10-, 12-, 14- and 16-inch toms; 15-inch hi-hats; an 18-inch crash; and 21- and 22-inch ride cymbals. Recorded at Rockfield studios, using both sticks and brushes and with up to six mic channels per drum, each kit piece has been sampled across a range of pitches, so you can tune the drum kit precisely to the key of the song you’re working on.

The main drum screen offers a clean and simple layout, with all drums visible in a single window. Virtual knobs select the level of dampening, switch between sticks and brushes, and enable you to select the specific pitch of each drum. Tunings range from G to F# on the snare, C to A on the kick drum and A to G across the toms. The edit screen features a mixer, individual mic level controls, filters, EQ, transient control and compression, providing a huge degree of power over each drum’s sound. As I mentioned, Tonal Drums was recorded at Rockfield, and Room and Chamber ambiences from this iconic studio can be introduced to further enhance the characteristics of each drum. A second page in the edit window reveals further options, including Attack, Hold and Decay parameters, as well as providing the ability to modify the velocity curve of each drum. Chosen pitches can be further adjusted in the edit window by ±12 semitones, and via a fine-tuning knob, allowing any note value to be easily achieved. Although the cymbals and hi-hat cannot be tuned to specific pitches, their tonal character can also be altered over ±12 semitones. Tonal Drums also includes Soniccouture’s Beat Tools, a comprehensive set of beat creation and editing windows (Beat Shifter, Euclidean Beats, Poly Beats and Blocks) that features in most, if not all, their percussion-based virtual instruments and enables you to create beats in a unique, fun and extremely creative way. Finally, the extensive MIDI mapping page deserves an honourable mention. A set of preset MIDI maps is included for popular VIs such as Toontracks’ EZdrummer and Addictive Drums, alongside maps for a range of electronic drums from Alesis, Roland and Yamaha. You can, of course, create and save your own maps, all of which makes integrating Tonal Drums into an existing setup very easy. Working with drums expertly tuned to specific pitches is a subtle but rewarding experience. A kick drum tuned to a low C with plenty of decay can greatly

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ON TE ST

SAMPLE LIBRARIES

enhance a bass line, and toms tuned to the key of the song suddenly become something you can’t do without. As with all Soniccouture products, everything has been beautifully recorded, and the level of sonic manipulation possible is very impressive. This results in a virtual instrument that can fit seamlessly into a wide range of styles and genres, all the time adding a unique and enjoyable twist. Mark Gordon £169 www.soniccouture.com

Native Instruments Claire

Kontakt Instrument

HHHHH Native Instruments are back with another gorgeous piano library, and this time, it’s Claire — a Fazioli-based Kontakt instrument that delivers realism and expressive depth. Following in the footsteps of the acclaimed Noire, Claire offers a rich and versatile sound that feels as great in delicate pop ballads as it does in cinematic scores. At the heart of Claire is the Fazioli F308, a stunning 10-foot Italian concert grand, recorded in detail at Galaxy Studios in Belgium. With a download size of 29.2GB, it’s packed with everything you’d expect from a high-end virtual piano: crystal-clear tones, beautifully detailed dynamics, and enough flexibility to shape it into whatever your music calls for. Claire sounds amazing. Its tone is pure and vibrant, leaning into the bright, forward sound Fazioli pianos are famous for. Whether you’re scoring an emotional film scene or layering chords for a modern pop production, Claire’s clarity will cut through the mix without ever feeling sterile. That said, if you’re after darker, moodier vibes, you might still gravitate toward libraries like Noire. One of the things that sets Claire apart is its intuitive interface. You can effortlessly adjust microphone positions (close, mid, and room) to fine-tune your tone, and there’s lots of options for customizing the piano’s nuances: mechanical noise, pedal noise, reverb, delay... the controls feel natural and responsive. A standout feature is the inclusion of the fourth pedal, exclusive to the Fazioli F308. This unique dynamic control allows you to soften the sound without altering its timbre, giving you an extra layer of expressiveness, especially in quieter passages. It’s a small

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detail, but it makes a big difference when you’re after that extra touch of realism. And let’s not forget the Particles Engine, a creative tool carried over from Noire. This granular synthesis tool can transform Claire into something entirely unexpected. Whether you’re crafting ethereal textures or subtle rhythmic effects, the Particles Engine is a playground for composers looking to push boundaries and add a cinematic flair to their projects. Claire feels like a natural evolution of Native Instruments’ piano libraries. At $149, it’s a serious contender for anyone who wants a piano that’s both timeless and forward-thinking. Whether you’re drawn to its vivid tonal clarity or intrigued by its creative possibilities, this is an instrument that will quickly become a go-to in your library. If you’ve been waiting for a virtual piano that combines technical precision with musical soul, Claire is definitely worth checking out. Benjamin Boukris $149 www.native-instruments.com

Strezov Sampling String Contours Kontakt Instrument

HHHH With their latest release, titled String Contours, Bulgarian company Strezov Sampling bring the world of sample libraries one step closer to the twin goals of expressiveness and realism, by focusing their efforts on the musical possibilities within a string swell. The Kontakt-compatible library features patches of basses, celli, violas, violins, solo violin and solo cello called ‘Contours’, which are defined as recorded performances “with a fixed length and dynamic shape”. At the risk of stating the obvious: sample libraries built using recorded performances of instruments have phrasing and dynamics baked into the patches and you, as a composer, have to work around them. String Contours makes this limitation a little less limiting by allowing you to “play these performances and seamlessly change notes [within a contour] without triggering a new performance”. For each string section/instrument, there are seven articulations, such as Normale, Tremolo, Sul Tasto and Loure; three tempos,

with the DAW’s tempo automatically detected by the library and the appropriate patch chosen; and four dynamic shapes: 1 bar crescendo-1 bar decrescendo, 2 bar crescendo-2 bar decrescendo, 2-1 and 1-2. Without any MIDI CC programming, you can ‘perform’ a piece using the combinations of articulations, dynamic shapes and tempo-adjustable patches on offer. If you do want to tweak the arc of each swell even more, you can use the mod wheel to further shape dynamics. Towards the end of the demo walkthrough video are words that bring sweet relief to most overworked composers: “there is barely anything that you need to master or learn with this library.” Mostly true, but don’t relax quite yet; there’s still work to be done and it involves being aware of the articulation you’ve selected and the phrase you’re playing within it. For example, a phrase you can play when you choose the 1-1 dynamic contour is different from one you can play with a 2-1 contour in terms of the timing of the swells and room for note changes. You do also have to time note changes so that the legato feels natural. This can be edited in the piano roll later, of course. If you close your eyes and just go with it, be aware that you need to be in sync with the ebb and flow of the patch, giving the proceedings a colour-within-the-lines feeling. You can always end in the middle of a contour, thanks to the adaptive release feature, which ensures a smooth tail for your note, made even smoother if you have reverb turned on. There is a plug-and-play element to the library which is helpful when putting ideas down, even if it does require a bit of practice to get the hang of how to play within the Contours. It also provides a kind of failsafe for beginners: as long as you adhere to the basic principles of good orchestral arrangement, your lack of experience won’t show in the way your string parts sound. Once you’ve taken the performance aspect out of the mix, your composition lives and dies on the strength of your musical idea, and String Contours is a worthy addition to your toolkit for bringing those ideas to life. Sonal D’Silva $249 www.strezov-sampling.com Audio examples of this month’s libraries are available at www.soundonsound.com.

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Can I mimic NS-10s and Auratones with EQ?

I was just watching an online tutorial by iZotope, and they discussed NS-10s and Auratone cubes being used to focus on the midrange. The question occurred to me whether the same effect could be achieved with simply simultaneously high-pass and low-pass filtering the signal at the appropriate frequencies. Is there more to it? SOS Forum post Hugh Robjohns, Technical Editor The benefits of sealed-cabinet speakers go a long way beyond just emphasising the midrange. In particular,

the ‘time-domain’ response (effectively, how quickly the speaker starts and stops making a sound) is important. If you’d like to know more about that, I’d recommend checking out Phil Ward’s excellent explanation in SOS September 2008 of what made the NS-10 so popular: https://sosm.ag/ the-ns10-story. In the case of the Auratone, as well as being a sealed‑cabinet design, it’s also a single-driver speaker, which means there are no crossover filters involved and the sound emerges from a single point. These factors are all due to the physical characteristics of these speakers, and they can’t really be replicated accurately on a different type of speaker. You can find lots more about the different considerations in speaker design in another

“The benefits of sealed-cabinet speakers go a long way beyond just emphasising the midrange.”

What’s the best way to record whispering?

of Phil’s articles from SOS June 2019, called Innovation In Loudspeaker Technology: https://sosm.ag/innovationstudio-loudspeaker-technology. Of course, it could be that all you want to do is to

focus just on the midrange — for example, you may want to check what your mix could sound like on smaller-speaker playback systems — and filtering out the lows (and possibly the highs) will do that.

Yamaha’s NS-10 speaker has a reputation for its midrange focus — but while its frequency response may be mimicked using EQ, there was way more to this speaker’s popularity with engineers.

times or other configurations would be appreciated. The whisperer has acting experience but I wouldn’t think they would know how to work the mic. SOS Forum post

I have to record some whispering. The room is not great so I will put the whisperer close to the mic. Hugh Robjohns The mic will be a Rode NT2A I’d suggest putting your or the new Earthworks compression questions SR117, both using a pop on the back burner for screen. I also have a Royer now, and focusing on the ribbon mic, but would it mic technique. There’s an be dangerous to work so inherent expectation that close to a ribbon? I’m very a whisperer must be close paranoid about blasts of air to the listener’s ear, so the When close miking someone close-miking approach (I’m even paranoid about whispering, an omni polar sliding it in and out of the you suggest is appropriate. pattern can be a good option. cloth protection sleeve it However, talking directly into came with!). The mic will go through the mic on-axis tends to over-emphasise the UA 6176 pre/compressor then into the talking mechanics — things like lip the box. I’m thinking in the box I might smacks, tongue noises and so forth. So need a limiter and maybe another compressor. Any suggestions on attack

I’d suggest talking across the mic instead of into it. In other words, place the mic near the cheek, pointing across the front of the mouth a couple of inches away. A pop shield probably won’t be needed in that position, but it’s better to use one if possible; the size of the pop shield is likely to be the limiting factor here. Working that close, you can probably get away with an omni pattern (you could switch your NT2A to that) because room sound will be strongly subdued. That would prevent there being big proximity effect bass level variations if the talker moves in and out. Ribbons don’t like strong blasts directly into the ribbon, but talking across it, or at an angle to it, is fine. Though do note that there will be a mighty bass lift if you’re close-working a normal ribbon. You can compensate for a static bass boost with EQ but, again, if the talker moves in relation to the mic it will vary, and that could cause you a headache!

“I’d suggest talking across the mic instead of into it.”

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145

TEACHING MIXING G AV I N H E R L I H Y

hat’s the biggest insight I’ve learned from working as a mix engineer and a music production lecturer? You can’t really teach people mixing. You can show them techniques and tricks. You can explain the science of acoustics or sound. You can demonstrate how you solved problems and crafted mixes over the years by revealing your song projects to your students. And lest we not forget, this is exactly what Sound On Sound has been doing for you for more than 39 years. But you can’t really. Really. Teach people to mix. The unfortunate news is that they have to learn to mix themselves. The hard way. By doing it over and over. For at least the requisite 10,000 hours. And yes, that magic number really does apply to the craft of mixing too. Perhaps it’s maybe even a lowball? You never stop

learning to mix. There are always new insights to gain. And music and technology are always changing and requiring adaptation from you, the engineer. So what can you do? Well you can show people the tricks, the techniques, the mixes you’ve worked on. You can explain the concepts. You can lecture until you’ve exhausted every morsel of information you have to give. But the real benefit that you can give to people learning to mix is to listen and give feedback. To lean over a shoulder, pop a set of headphones on and analyse their work. To lend an alternative viewpoint. To suggest a dB reduction here or a lower threshold there. These are the milestones that make all the difference, because people learn best by doing the thing. They need to understand when something has been pushed too far or not far enough. They need

to make mistakes. And then discuss those mistakes with a trusted pair of ears. And this is something that cannot be gained from YouTube, as great as YouTube is for learning mixing or any other technical process. I did not study production at University (I studied a degree in Journalism), but I earned my music production masters over my almost 20 years of tunnelling through a professional career as a producer and mixer. And while YouTube really helped and continues to help progress my learning along the way, the real progress came from making mistakes. Running into problems. Countless hours of dissatisfaction and starting again. Taking feedback on the chin. Going back to the drawing board. Feeling elated when things did go to plan. And this process is thankfully now a lot more streamlined, but the learning continues. And my learning about how people learn, continues in parallel.

“What’s the biggest insight I’ve learned from working as a mix engineer and a music production lecturer? You can’t really teach people mixing.” 146

March 2025 / www.soundonsound.com

This column began as a social media prank. I thought it might be funny to post some written content that proposed to reveal my number one fastest method for learning to mix like a pro. As a mix engineer I personally cringe every time I see this idea. The audience would then swipe to find that the next post exclaimed: “Spend 10,000 hours mixing records!” And, sorry to say, it’s true! But I thought it a little cruel to post. So I hit delete and decided to long-form the idea with this piece instead. So what have we learned? To do the thing, you have to do the thing. But a trusted teacher peering over your shoulders, to build your confidence in your work, to point you in the direction you need to go, is the fastest way to accelerate the process. The prize is being able to mix with confidence. To know it will sound great wherever it is listened to. To level the odds, so that the mix won’t hold back a great song again. And to allow you, the producer, the peace of mind to spend the bulk of your creative time on the job that is of far greater importance: Making art that matters.

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SOUND ON SOUND [USA] March 2025 Issue