SOUND ON SOUND [UK] June 2025

SOUND ON SOUND THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

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

S L AT E D I G I TA L

VMS 2 Microphone modelling...

Evolved!

JUNE 2025

Inside Track: Grant Boutin & Tate McRae

Crafting mega-hits in FL Studio

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Which studio desk is right for you?

IK Multimedia iLoud Micro Pro

The smallest pro monitors yet?

Harrison D510r

500-series chassis with Dante

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VOLUME 40 • ISSUE 8

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ON TEST: RME / SOYUZ / ARTURIA / BEYERDYNAMIC / TOONTRACK / TASCAM / ACUSTICA / EASTWEST / AUDIX

TECHNIQUE: FITTING A BOSS GK-5 PICKUP / VOCAL LEVELLING IN CUBASE

June 2025 £7.99

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

Hydrasynth delivers a world of synthesis capabilities to your fingertips with a familiar, organic workflow that unlocks a new world of sound design & performance possibilities.

Available as four distinct models each featuring; 8 or 16 Voice Polyphony, Wavemorphing Oscillators, Polytouch, Multiple Filter Models, lush Stereo Effects, a powerful yet intuitive Modulation Matrix, rugged build quality and so much more.

DELUXE

KEYBOARD

• 73 Note Polytouch Semi Weighted Keyboard • Dual engine with 16 voice polyphony • Dual 8 voice operation with splits and layers • Dedicated stereo outs for each engine (4 outs total) • 5 banks of 128 Multi patches • 8 banks of 128 Single patches

• 49-note Polytouch keyboard • 8 voice polyphony with three oscillators per voice • Four octave ribbon controller • Dedicated coantrollers for Filters & Arp • 8 assignable macro controllers

• Compact form factor

EXPLORER

• 37 note mini keys board with polytouch • Optional battery power • 8 voice polyphony with 3 oscillators per voice • 8 Bankable macro controllers • 8 banks of 128 patches

• 24 configurable RGB backlit Polytouch pads

DESKTOP

• Desktop or 5u rackmount configuration • 8 banks of 128 patches • 8 voice polyphony with 3 oscillators per voice • 8 assignable macro controllers • Multiple CV IO‘s with multiple voltage standards

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

LE ADER

MODEL BEHAVIOUR S

ome people see modelled equipment as a lacklustre substitute for the real thing.

Others view it as an amazing opportunity for modern musicians to access vintage gear without all the maintenance headaches, space issues, cost overheads and functional limitations that come with the real thing. But to my mind, both schools of thought overlook the most important differences, which are more to do with workflow than sound.

For example, the Internet abounds with purists who will tell you that no soft synth can match the warmth of a real Roland Jupiter-8. I’m not going to pretend that I don’t regret selling my own Jupiter-8 many years ago, but that’s not because its sound can’t be recreated digitally. It’s because there’s something about the Jupiter-8 as an object that is inevitably missing in any recreation.

I don’t know whether the patches I lovingly crafted could be precisely replicated using something like Acustica’s THING, as reviewed in this issue, but to me, that is secondary. The point is that I wouldn’t have come up with those patches if I’d been sat behind a screen, mouse and controller keyboard. For better or worse, I’d have made different choices.

The same is true of modelling mics such as the Slate VMS2 that graces this issue’s cover.

I recently did a couple of mic comparisons with

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a singer. I put up all the usual suspects, such as a Neumann U87 and an AKG C414. We decided that his voice didn’t suit capacitor mics at all, and tracked him with a Sennheiser MD441, which seemed to us the perfect fit. On his previous batch of recordings, however, the engineer used a modelling mic, and I’m pretty sure they chose a model that we rejected utterly. Guess what?

Those recordings sound great too.

The lesson here is not that all mics sound the same, or that modelled mics are somehow inaccurate. It’s that choices are always made within a context, and modelling changes the context. With a modelling mic, you can compare endless options without any variation in mic position or distance. But you don’t get to hear how the singer responds to the physical presence of those different mics, or how they interact with your preamps.

And, once again, you make different choices.

You need to spend time with any piece of gear to get the best from it, and modelling gear adds a second layer to this equation, because you need to get to know both the modelling system and the things it’s modelling.

Modelling isn’t better or worse than the real thing. It’s just different — and acknowledging that difference is key to getting the best from any modelling system.

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“Modelling isn’t better or worse than the real thing.

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

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

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

Audioscape Opto Comp - hand built in the USA with premium components, this reproduction of the world’s most beloved optical compressor does not disappoint.

Easy to use with great results. £1349

Manley Reference Gold - flagship high-resolution valve microphone with a rich tonal balance and plenty of HF definition to accurately capture every nuance. Continuously variable polar pattern. £5738

Neumann U67 - painstakingly accurate reissue of the mic that defined the sound of the 1960s. Outstanding vocal microphone with a naturally smooth, warm top end.

A true classic. £5995

BeezNeez B87i - quality reproduction of the original U87. Mid-frequency lift enhances vocal performances. Uses NOS 2N3819 FET and custom wound Mu Toroid transformer. Unbeatable value. £899

Flea M 251 - Meticulous recreation of the legendary 251 based on Tim Campbell CT12 capsule, selected 6072A valve and identical T14/I transformer. A top flight choice for female vocals. £5399

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

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

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

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

API The Box - small format analogue console delivering the signature API sound for smaller DAW-based studios. 8x input and 24x summing channels with one 500 Series slot per channel. £16,995

Universal Audio Apollo x4 Gen2 UA’s class leading range of Apollo interfaces offers a class-leading combination of high quality AD/DA with UAD plug-in emulations of classic outboard. Bestseller. £1610

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

Solid State Logic SSL18 - packed with features, SSL’s 32-bit/192kHz audio interface offers outstanding value for home and project studios, with 8x SSL designed mic preamps and Legacy 4K enhancement. £899

Errors & Omissions excepted.

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76 INSIDE TRACK

IN THIS ISSUE

www.soundonsound.com

June 2025 / issue 8 / volume 40

FEATURES 52 The Oberheim Two Voice Revisited If any synthesizer can be considered to be greater than the sum of its parts, it’s Tom Oberheim’s enduring classic, the Two Voice.

WIN UNIVERSAL AUDIO APOLLO X4 GEN 2 & SPHERE DLX WORTH £3200

64 Fitting A Boss GK-5 Kit Permanently installing a Roland/Boss synth pickup might seem like a job for a guitar tech, but with some basic woodworking skills you can do it yourself. Here’s how.

74 How I Got That Sound Producer and musician Hunter Lea explains how he went above and beyond with a venerable Vox Continental.

76 Inside Track: Grant Boutin Having proved his worth on Tate McRae’s second album, producer and songwriter Grant Boutin played a pivotal role in the third.

PAGE 82

99 Modular Møffenzeef Mødular’s Ross Fish on art therapy and making modules that sound like they’re broken.

120 Spotlight: Workstation Desks

102 Jason & Sarah Dasent:

A dedicated workstation desk can revolutionise the most chaotic studio. We round up some of the best.

Making Music Tech Accessible

A successful musician and producer who happens to be blind, Jason Dasent is helping manufacturers to implement text‑to‑speech and other vital accessibility features.

108 Talkback Engineer and producer Aron Kobayashi Ritch explains why he always brings his own cymbals to sessions.

136 Q&A Your studio and recording questions answered.

138 Why I Love...

Music Technology Marketing

Marketing victim Chris Korff shares his abiding affection for the subtle art of music tech PR.

16 SOYUZ LAKESIDE

90

Acustica THING

56

Arturia Pigments 6 Software Synthesizer

32

Audix D6X Dynamic Microphone

28

115

48

Modelling Headphone Amp For Guitar & Bass

72

EastWest LO-FI Sample Library

124

Emergence Audio Time Crystals Erica Synths Perkons Voice Eurorack Module

26

GForce Novation Bass Station Software Synthesizer

34

Ginger Audio Sphere v2 Monitor Control Software For macOS

110

Harrison D510r 500-series Chassis

38

Active Monitors

125 Sonora Cinematic

Emma Legato

Sample Library

16

116

94

44 LANG Electronics

P.LANE Type 436

117

Mackie ProFX 10 Go Mixer & Audio Interface

98

Møffenzeef Mødular DitDit Eurorack Module

12

Playfair Audio Dynamic Grading 2 Dynamics Processing Plug-in

20

RME DPS-2 & LNI-2 DC Regulated Power Supplies

TC Electronic Ditto 2 Looper Pedal

60

Toontrack Fusion Software Collection

24

Warm Audio Fen-tone Ribbon Microphone

Mono Valve Compressor

118

Tascam FR-AV2 Portable Audio Recorder

Korneff Audio Shure Level-Loc Compressor Plug-in

Soyuz Lakeside Microphone Preamplifier

Klevgrand Elpiano Sample Library

Slate Digital VMS 2 Microphone Modelling System

Micro Monitor Pro

Sample Library

98

iamReverb Audio iamReverb

84 IK Multimedia iLoud

124

Rode NTH-50 Closed-back Headphones

Convolution Reverb Plug-in

beyerdynamic DT 70 IE Boss Katana:Go

Heritage Audio Synth Buddy Passive Source/ Destination Selector

In-ear Monitors

114

8

DAW Software

Software Synthesizer

68

Harrison Mixbus 11 C O V E R

ON TEST

125

Zero-G Biorhythms Sample Library

WORKSHOPS 126 130 132 134

Studio One Live Cubase Pro Tools

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in May 2022 (www.soundonsound. com/reviews/rode-nth-100), Rode have also introduced a product called the NTH-Mic. This is a short plastic arm with a microphone at one end and a mini-jack at the other. It plugs into whichever of the NTH-100’s dual cable ports you’re not using, and turns them into a comms headset for streaming, online meetings and so on.

When I heard that Rode were about to introduce a second headphone model called the NTH-50, I initially assumed that this would be an even cheaper, budget version of the NTH-100. Not so. In fact, it’s a completely new design, developed with a slightly different user base in mind.

Over Is Out

Rode NTH-50 Closed-back Headphones With their second headphone sync’ed to visuals. And the diversification shows no sign of stopping any time soon. model, Rode are looking to You can now buy Rode video switchers, audio interfaces and boom poles, and shut out the world outside.

SAM INGLIS

R

ode started out making studio microphones, but when YouTube came along, they were quick to diversify into affordable sound-for-picture gear. From simple wireless systems and camera-mounted mics to the superb NT-SF1 Ambisonic microphone, there’s a Rode device to suit every scenario in which sound needs to be captured and

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a couple of years ago, the company introduced their first pair of headphones.

Despite their modest price, the NTH-100s incorporate numerous design innovations that make a real difference in the studio, such as locking headband adjustment, dual cable ports and ear cushions filled with a special gel that keeps your ears cool. And as a bonus, they also sound better than closed-back headphones at the price have any right to. Since I reviewed the NTH-100 back

Most studio headphones, whether open‑ or closed-back, are ‘over-ear’ designs, with relatively large earcups that enclose the entire ear, the cushions resting against the side of the user’s head. However, DJs, location sound recordists and people doing audio for picture often favour ‘on-ear’ models.

These are invariably closed-back, and have smaller earcups that sit right on top of the outer ear. In my experience, on-ear cans are seldom as comfortable as over-ear models, but they have their own strengths. The most important of these is that they typically offer better isolation from external sounds.

When I say that DJs and audio-for-video types favour on-ear headphones, what I mean is that they favour a particular model that happens to be an on-ear design. The Sennheiser HD25-IIs have been around for many years now and are distinguished by their no-nonsense, rugged design, their diminutive, circular earcups, and by the vice-like grip they exert upon your head.

And when you remove the NTH-50 from their packaging, it’s hard not to conclude that Rode have lined up their big guns and pointed them directly at Hanover.

There is, of course, only a limited number of design choices you can make in constructing a pair of on-ear headphones, but the visual resemblance here is surely deliberate. It’s most apparent in the cables that emerge from the top of each earcup, with the right side terminating in a red plug for easy left/right identification. Instead of carrying over their innovative locking headband adjustment system from the NTH-100, Rode have also used what is

Take it anywhere. Create whenever. Trust your instincts.

ableton.com/move

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RODE NTH-50

essentially the same mechanism as on the HD25 (and a million other headphone models), with the earcups mounted on a notched strip, along which they can be clicked up or down.

However, you wouldn’t expect Rode to release something that is just a copy or clone, and that certainly isn’t the case here. For one thing, the NTH-50s are quite a bit larger and heavier than the HD25-IIs, and they include the same dual cable entry as on the NTH-100s.

The supplied cable has alternating curly and straight sections, and terminates in a locking mini-jack that can be plugged in either side without reversing the channels. Sadly, there’s no CoolTech gel in the NTH-50 ear cushions, and they have minimal freedom to rotate either fore and aft or up and down, but in practice I found them considerably more comfortable than the HD25-IIs. I’d still prefer a well-fitting pair of over-ear cans for long sessions, but I could stand to wear these for hours without experiencing serious discomfort. As with the NTH-100s, there’s no proper case, just a soft bag,

and the NTH-50s don’t fold up for portability. (Nor do the HD25s, for that matter.)

Hearing Things The emphasis on ergonomic innovation that marked the NTH-100s has been dialled down here, then, but what’s going on inside those circular earcups is said to be pretty novel. The NTH-50s employ a “custom driver array”, which “features a revolutionary acoustic design with an innovative resonant chamber” and is said to offer unprecedented accuracy with minimal distortion. This driver array has a comfortable nominal impedance of 32Ω and is relatively sensitive, delivering 102dB SPL for a 1mW signal at 500Hz. Rode back up their claims about low distortion with some detailed and impressive

numbers: a 1mW input produces less than 0.03% THD at 500Hz and less than 0.2% at 100Hz, whilst even at 120dB SPL, a 100Hz input produces only 0.5% THD. Interestingly, although good isolation is cited as one of the selling points of this model, the specified 21dBA ambient noise reduction only betters that of the NTH-100 by 1dB. (I haven’t been able to find an isolation specification for the HD25.)

I’ve seen online reviews that describe the sound of the HD25 as being ‘scooped’, with prominent humps at 100Hz and 8kHz. My own pair sound reasonably flat at the low end, but they definitely have a strong emphasis somewhere in the treble region. This, I suspect, is actually part of their appeal.

You wouldn’t want to mix on them, at least not without corrective equalisation, but their brightness goes hand in hand with their good isolation in making it possible to acquire useful information about what you’re hearing even in difficult circumstances. Dodgy edits, hiss, and click tracks all come through with painful clarity, even if they aren’t represented with perfect fidelity.

True to their word, Rode haven’t gone down the same, er, road, and the NTH-50s deliver a much more pleasant, hi-fi listening experience. Like the NTH-100s, they are quite rich in the low midrange — certainly more so than my HD25s — but from perhaps 300Hz up are refreshingly well balanced.

From memory, in fact, the overall sound is not unlike that of the NTH-100, with a characteristic smoothness and gloss that I normally associate with much more expensive headphones. Both physically and sonically, I found them much less fatiguing than the HD25.

What, then, of the relative levels of isolation achievable with the two models? With no sound playing, both models seemed to attenuate ambient noise by roughly the same amount, albeit with slightly different frequency characteristics. With real-world audio, I suspect that the HD25s might appear to have the edge because their additional brightness accentuates the contrast between the programme material and the external noise.

If I was trying to make sense of mission-critical instructions in a helicopter, I’d probably go with the Sennheisers. For making critical decisions about music — and actually enjoying it — the NTH-50s are clear winners. Just like Rode’s debut model, the NTH-50s sound a lot more expensive than they actually are.

summary Rode’s on-ear headphones are perhaps more conventional than their over-ear model, but they nevertheless offer a compelling blend of sound quality, isolation and value for money.

As on Rode’s earlier NTH-100, the cable can be plugged into either earcup, so you never need have it trailing across your lap again!

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£ £99 including VAT. T Source Distribution +44 (0)208 962 5080 E sales@sourcedistribution.co.uk W www.sourcedistribution.co.uk W www.rode.com

U T O P I A M A I N | STUDIO MONITORS OF EXCELLENCE The Utopia Mains range employs the full breadth of Focal’s technical expertise and professional audio industry knowledge to deliver an unparalleled recording, mixing or mastering experience. Comprising a 3-way Main 112 and 3.5-way Main 212 model, each Utopia Main monitor features newly patented drivers paired with Class-H Current Mode amplifiers for ultimate power and sonic precision.

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The controls may look unfamiliar, but there’s good visual feedback and you soon get your bearings.

Playfair Audio Dynamic Grading 2 Dynamics Processing Plug-in There’s more to this fresh take on dynamics processing than a groovy GUI! TREVOR MICHAEL

P

layfair Audio’s Dynamic Grading is a powerful one-stop plug-in for shaping the dynamics of an audio signal — it allows you to access compression, expansion and transient shaping in a rather different way from usual — and those familiar with the previous version will find that v2 constitutes a dynamic (see what I did there?) leap forward. In fact, it’s fair to say that after taking advice from users of v1, Playfair have created something unique.

As well as its completely redesigned graphical user interface, there’s support for multi-channel formats up to 9.1.6, a magic eye spectral display, and some helpful factory presets that can be adjusted and stored as desired. And if

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you’re already a registered user of v1, you get to enjoy all this for free.

Dynamic Grading 2.0 supports 64-bit AU, VST3 and AAX plug-in formats for hosts running on macOS (10.13 or later) or Windows (10 or later), it’s compatible with Apple Silicon and Intel Macs, and authorisation is by serial number. For the purpose of this review, I used an Intel i7 Mac Mini, and installation was pain free.

Overview A quick first glance at the GUI is not met with the familiar controls and labels that you’d expect to find on any dynamics processor — there’s no obvious threshold, ratio, or attack controls, for example. Instead, you have controls labelled Punch, Body and Floor, but don’t worry, as it quickly becomes apparent what’s going on.

The plug-in opens to display two dynamic histograms, giving a real-time visualisation of dynamics at the input (Source) and output (Target). Between

these histograms are those three key controls (Punch, Body and Floor), and these are used to shape the incoming signal to match the desired output, as shown in the Target display. On closer inspection, you’ll realise that these three controls are in fact separate ratios for each of the three parameters — so we’re back in more familiar territory. Along the left-hand side, a scroll bar allows you to zoom in on the histograms for more detailed information. The time slider below the input histogram can be adjusted in bars (the default, following the DAW session tempo) or changed to seconds in the settings menu, offering a longer time scale and a smoother display.

As shown in the second screenshot, when a signal is detected at the input during playback, you’ll see a white area on the display. The outline shows the average loudness of the audio, while the area within the lines lights up to indicate the short-term loudness.

If you’re not accustomed to interpreting histogram graphs, you might initially find yourself wishing for a simple gain reduction meter I suppose, but it’s actually pretty straightforward, and once you get your head around it you’ll find that you can glean lots of useful information very quickly.

The input signal is split into three sections, corresponding to the aforementioned Punch, Body and Floor controls, and the histogram makes it pretty easy to determine where the splits between them will be most useful.

At the bottom of the GUI, though, you’ll find a ‘magic wand’ button that engages a source learn function, and this cleverly sets the markers for you, based on an analysis of the signal. Generally, I found that this worked well, providing sensible starting points. The Body of the sound is the part where the source histogram shows the most energy. The area shown above that, which is what Playfair call the Punch, is typically where the attack is found. And the area below it, they call Floor, which is where you’ll find most of the low-level details, such as reverb tails and background noise.

You can adjust the splits manually (whether or not you use the source learn ‘wand’), by dragging the red and orange source markers: everything above the red marker is Punch, everything below orange

Learn more at seelectronics.com

sE7 sideFire

Small Diaphragm Condenser Microphones

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P L AY FA IR A U DIO DY N A MI C GR A DING 2

“I approached this plug-in with a healthy dollop of scepticism... But once I began to examine Dynamic Grading in more depth, I quickly changed my opinion.”

An instance of Dynamic Grading inserted on a stereo drum bus, with a compression ratio of 4:1 acting on the Body, and a little 1:1.5 expansion ratio for the Punch, giving the effect of pushing up the transients.

is Floor, and the Body lies in between. control. Playfair Audio advise tweaking Once you’ve determined where the the spectrum slider to show more or Body of your sound is, compression or less a horizontal line, so that the audio is expansion of each of the three areas processed evenly across all frequencies: can be adjusted in two ways: you can to do that, you simply drag the spectrum tweak the ratio controls in the centre slider to the left if your source audio is of the display (drag down on the right more treble‑based, or to the right if your side for more compression, drag up material is more bass‑heavy. There are for more expansion); and you can grab also sliders for Response (attack) and and move the threshold markers in the Release, as well as a channel link slider: target histogram. at 100 the plug‑in affects all channels Often, a good place to start is with together; at 0 it operates as independent compression of the Body range, in multi-mono channels across the signal. much the same way you might normally Finally, a curve display at the bottom right apply broadband compression to provides a visual representation of how a signal. Then, if you feel that you’re your adjustments are affecting the level. losing some of the transients as a result, Grade A? you might look to expand the Punch a little. Alternatively, if you feel that the I have to admit that I approached compression is bringing up too much this plug-in with a healthy dollop of background noise, ambience or reverb, scepticism because judging it on first you might expand the Floor. You can appearances it struck me as being little do this subtly, to shape a sound, or you more than a fancy-looking compressor/ can, for example, assertively expand expander. I have plenty of those already, the floor on a drum bus for an instant of course, so I didn’t initially see the gated drum sound. Along with adding point... But once I began to examine a little Punch, this approach could be Dynamic Grading in more depth, I quickly useful for parallel processing drums, changed my opinion and began to to inject a greater sense of detail and appreciate the benefits it offers. clarity in a dense mix — and this is made convenient by an Amount control (at the bottom right of the GUI) that allows blending of the processed and unaffected signals.

Further controls are found along the right-hand side, and first off you’ll find the new Magic Eye spectrum display with a single horizontal slider beneath. A curve represents the frequency spectrum of the signal that’s triggering the level control, and you could think of this broadly as being a side-chain A 7.1.4 multi-channel instance of the plug-in.

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Dynamic Grading is way more detailed than a standard compressor, and neither should it be confused with a multiband compressor. Playfair Audio aren’t reinventing the idea of dynamic shaping exactly, but they are certainly enabling us to think about it in a different way. Being able to control the main body of the sound either without affecting the transients, or while separately enhancing them, is incredibly useful. And at the same time, you’re able to reduce the noise floor, while keeping the body of the sound intact.

It might not be the tool for every dynamics job. For example, this isn’t a colourful-sounding processor, and the dynamic manipulation sounds smooth.

But we have countless options for colour today, so that’s a good thing in my book.

Personally, I’d prefer that some kind of more standard gain‑reduction meter were included as an option, alongside the current histograms. And when used on multi-channel sources, I felt that the individual channel level meters could be made clearer. But that’s to some extent a question of personal preferences, and none of that inhibits the functioning of the plug-in in any way.

Just yesterday, I found it useful on a mix to process a DI’ed plucked acoustic guitar part, taming the transients without affecting the body of the sound, and I reckon this could well become the tool I reach for from now on to breathe new life into the over-compressed audio files I sometimes receive for mixing. Having it available in multi-channel formats up to 9.1.6 is an added bonus for those, like me, who work in immersive audio.

Recommended.

summary A different and clever approach to dynamics processing that allows you more control, not least to mitigate some of the unwanted side-effects of traditional compression and expansion.

£ £141 including VAT.

Upgrade from v1 is free.

W https://playfair-audio.com

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

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NEIL ROGERS

S

oyuz, who started out as a joint American-Russian venture, have been turning heads for several years now with their original, stylish-looking, hand-built microphones that draw on the best traditions of German and Russian designs. The product I’m reviewing here is their first ‘full-blown’ non-mic product: while they’ve previously offered the Launcher range of inline booster preamps, the new Lakeside is a fully featured, single-channel, Class-A rackmountable mic preamp. Why make a preamp? Soyuz explained that after being continually asked by customers for recommendations as to which preamps would work best with their mics, they figured that they should probably build their own!

I’ve had the pleasure of reviewing a couple of Soyuz mics in recent years, and they’ve always impressed from an aesthetic point of view, with a pleasing amount of care and attention given to the way their products are presented. More importantly, they always sound great. And it’s this combination of factors that has seen Soyuz gain a firm foothold in the high-end microphone market. The Lakeside certainly stands out visually, but with so many new and established preamp options on the market, I was genuinely intrigued to hear what, if anything, the Lakeside might offer from a sonic perspective to justify what is a substantial outlay for a non-vintage, single-channel device.

Features The Soyuz Lakeside is an all-discrete, single-channel, Class-A preamp, in which large (and very heavy!) custom transformers balance the audio input and output.

Although intended primarily to bring out the best in their own mics, it’s not intended to sound clinical, and Soyuz claim that the Lakeside will add a lush, pleasing character and dimension to any microphone, instrument or source.

Control-wise, the Lakeside features a stepped input control that offers up to 60dB of gain in 5dB increments, as well as a variable output trim control that allows the user to add more level going out of the device, or to pull the level down when driving the input stage harder — which is a key intention of the design. We also have the typical controls you’d expect to find on most mic preamps, including polarity inversion and 48V phantom power, and there’s a very welcome DI instrument

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Soyuz Lakeside Microphone Preamplifier

Soyuz’s first rackmount preamp evokes the sound of vintage designs. input on the front of the unit. In the spirit of high-end vintage preamps, there’s absolutely no metering whatsoever on the Lakeside: it is very much intended as a device whose sound should be set up and judged using just a few luxurious-feeling controls and your listening skills.

On the back, as is so often the case on vintage-style preamps, there’s little to see: just an IEC power inlet that can be switched for different territories’ mains voltages, and two Neutrik XLRs to cater for the audio input and output.

What I haven’t mentioned yet is Soyuz’s decision to put a variable high-pass filter front and centre in their design — it’s the knob with the bright yellow cap — and this gives you a clue as to how important they consider it to be. Soyuz explain that it’s a “unique, inductor based high-pass filter” that’s designed to be a creative addition, as well as allowing you to perform the more mundane tasks for which a switchable filter is so often included on preamps, such as removing unwanted low-end rumble

or correcting any proximity effect bass boost. High-quality analogue filters can be creative musical tools not just because of their turnover frequency, but also due to the resonant ‘bumps’ that often sit just above where you set the filter. As with the metering, the markings are deliberately vague so as to encourage you to listen: the 5dB steps are only briefly marked between a range that extends right up to 330Hz. I’ve always been an advocate of using filters when tracking, and I was excited to see if this section lived up to its billing.

In Use My first encounter with this retro-looking unit was on a hectic tracking session, with a band who were on a mission to get a track down sharpish. For the live drum-kit recording, I used the Lakeside to amplify a Shure SM57 close-snare mic. Typically, I use an API preamp for this job, and I was immediately struck by a sense that the Lakeside was delivering a little more ‘presence’. Also well worth noting is that,

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SOYUZ L AKESIDE

using the Lakeside on such a loud source as this, I noticed straight away that this is a preamp that will add a nice, saturated character when required — it’s clearly not intended to be a linear, reference device!

A quick peek at the waveforms of the recorded snare drum revealed just how firmly the high transients are controlled by the Lakeside’s sizeable transformers.

Exploring this feature in search of more subtle effects, I could discern a lovely sense of saturation that added a touch of richness and excitement, but if you desire more obvious distortion you can easily head into that territory too. In more subjective terms, I’d say that to my ears it sounded much more akin to vintage Neve-style distortion than the op-amp‑influenced character of API’s classic designs.

With the drum tracking done, the bassist was keen to work on his bass parts — so I tried plugging him straight into the instrument input on the front panel, and he got straight to work. Again, I could hear that midrange presence that I’d noticed on the snare, and later, when mixing, I noticed that I wasn’t looking to boost around 1-2 kHz to make the bass more audible, as I so often need to. These bass recordings also led me for the first time to notice that the bottom end had a nicely ‘contained’ feel that, on certain sources such as vocals, I found could tip the sound into feeling a bit harsh if the input was a little too hot. This was just one of the many nuances that I noticed when balancing the Lakeside’s gain and output controls, and I remained intrigued as I tried to learn and predict the subtle but noticeable way in which the input level affected the tone.

Recording electric and acoustic guitars provided a good opportunity to get more of a feel for the inductor-based filter section, and it felt smooth and natural to

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Inside, you can see the chunky input and output transformers and the high-pass filter’s inductor, which combine with the gain cotnrol to make this preamp so tonally versatile.

experiment with just how far you could roll the filter up before a guitar sound began to lose its body. On a very quiet acoustic part, it allowed me to get very, very close with a mic and then filter out the substantial proximity effect. Apart from the more typical workmanlike uses for a HP filter, I could hear something interesting happening when used on guitars and vocals between the 55Hz and 120Hz marks. It was never dramatic, but I would often find the sweet spot where a source would suddenly feel that little but fuller, even as I removed the low end.

There’s plenty of nuance here to explore, then, but with a little more time spent with the Lakeside, I began to get a much better feel for how the input and output control can be combined with the filter to access different sonic options that shape the tone, and how to dial in just the amount of saturation you might dare to capture when tracking. Of course, you can also use these tools when mixing, for more obvious effects and experimentation: it offers more than enough character once you’ve spent a bit of time getting to know it.

Summing Up Today, for a very modest outlay, you can buy very capable preamps that will do a great job of cleanly amplifying the output from your microphones. Reviewing preamps can sometimes be a bit tricky: there’s a healthy and ongoing debate amongst audiophiles about just how much difference a preamp can (or should) make to a recording. The Lakeside, though, is clearly intended to offer more than amplification.

In the course of testing the Lakeside for this review, I made a few audio examples so I could compare what the Lakeside offered

with the results from the onboard preamps in my studio’s Audient mixing desk, and you can find these on the SOS website (https://sosm.ag/soyuz-lakeside-preamp).

In more sedate settings, the differences are minimal — except that I often had a sense that the Lakeside offered a subtle boost to the low mids. Things always get rather more interesting with high-end transformer-based preamps, and the sonic differences between them more pronounced, when you ‘drive’ them with hotter input levels — and it was when doing this that I found the Lakeside really began to come alive. As I’ve mentioned, you can dial in saturation or head more into distortion territory, but either way I nearly always enjoyed what I heard — even if I sometimes decided the effect was a bit much for the current job.

In summary, the Lakeside is a beautifully crafted piece of equipment that’s a great new high-end preamp option.

Hand-built, artisan equipment of this quality inevitably comes with a hefty price tag, but this box can do more than simply amplify your mic, and I reckon it’s a good thing that we have great options available to suit different budgets.

summary Conjuring up the spirit of high-end analogue designs of old, the Lakeside is a premium mic preamp that encourages you to use your ears to explore the nuances that only a high-quality transformer-based preamp can bring to recording sessions.

£ £1399 including VAT. T Funky Junk +44 (0)207 281 4478 E sales@funky-junk.com W https://www.funky-junk.com W https://soyuzmicrophones.com

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Need a better power supply? Then these boxes are worth checking out.

HUGH ROBJOHNS

A

poor power supply — that is, a noisy or poorly regulated one — can substantially degrade a product’s technical and audible performance. Thankfully, professional devices usually incorporate satisfactory power supplies (often supplementing low-noise switch-mode types with effective voltage regulation and noise filtering) and most op-amps routinely used in audio circuitry are remarkably good at rejecting power line noise. But in parts of the world where the mains supply isn’t as clean or stable as we’d like, or for applications in which the lowest possible noise floor is essential, an ‘enhanced’ power supply system can be beneficial. RME now offer two interesting ‘enhanced’ external power supply devices to address this potential need. Though primarily intended as options for the company’s ADI-2 series interfaces, they can be used to power other manufacturers’ devices too.

DPS-2 Marketed as a “linear, super‑low‑noise, dual audio power supply with µFilter and Sensor technology”, the quarter-rack‑width DPS-2 comes in steel housing and contains a beefy transformer, so it’s quite heavy (1.89kg). It accepts a mains AC input (factory set to 230 or 115 V ) via the usual IEC inlet, and generates a nominal 12V DC output with a 2.5A capability, via one of two output ports, of which more in a moment.

Essentially, the DPS-2 replaces the standard switch-mode line-lump power unit supplied with the ADI-2 range and provides a more stable and lower-noise DC supply. Its front panel carries a pair of LEDs (one for each output) that illuminate for the first 30s after power-on to confirm operation. They also illuminate if a fault is detected, while a permanent display mode is accessed by pressing a button between the LEDs. On the back are a mains on/off switch, fuse holder and mains inlet, and a slide-switch provides a ground lift that separates the mains protective earth from the power supply output’s ground. There’s also a separate grounding terminal, to tie the output ground to a technical earth or other appropriate reference, and two Kycon KPP-4P locking four-pin sockets for the DC

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RME DPS-2 & LNI-2 DC Regulated Power Supplies outputs. A pair of one-metre output cables is included, both terminating in 5.5 x 2.1mm (centre-positive) coaxial plugs, but one features RME’s usual bayonet‑locking coax plug, while the other has a standard plug for use with other manufacturers’ equipment.

Although both DC outputs provide a 12V DC power feed, they do so through different technologies, labelled Linear and µFilter.

One or the other can power an attached device, according to preference, but you can’t simultaneously power two devices from different outputs; when a connection is detected at one output the other is disabled and its LED is extinguished.

Internally, the DPS-2 features a traditional linear power supply but it’s built on a massive scale, with careful attention to detail throughout. For example, it employs a generously over-specified 60VA toroidal mains transformer, with additional capacitive and magnetic shielding, and the mains input side employs an HF filter (to reduce mains noise above 3kHz) and a DC filter (to reduce asymmetry on the mains AC voltage). The transformer’s output feeds a bridge rectifier and then a reservoir with a capacitance of 20,000µF, with an intermediate choke inductor and a further 10,000µF reservoir, to create a raw rectified and smoothed DC

voltage. This is further processed by the two output sections to reduce ripple and noise.

The Linear output routes the raw DC voltage through a conventional hybrid voltage regulator circuit to achieve a noise floor of less than 50µV, with load regulation of 3.63% and an output impedance of 0.22Ω. That’s a very low-noise, well-regulated power-supply source — far better than any conventional line-lump unit!

The µFilter output employs an even more sophisticated voltage regulation system, in the form of a switch-mode regulator, followed by RME’s µFilter (Micron filter) technology. In essence, this is a discrete, ultra-low-noise voltage regulator that reduces power supply noise to less than 2µV, with load regulation of 0.2% and an output impedance of 0.012Ω. That is exceptionally quiet and tightly regulated.

Astonishingly so, in fact.

The 24-page user manual explains the technologies and concepts behind this power supply design, and includes a technical measurements and graphs.

Clearly, RME’s designers are very proud of their achievements, and rightly so!

Both the Linear and µFilter outputs are rated for a nominal 2.5A, but the latter can manage 3.5A, while the Linear output can

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RME DPS-2 & LNI-2 DC

deliver up to 4A should that be needed. Full protection against overload, short-circuit, and overheating is included on both outputs, as well as under-voltage detection.

Given that the output cables terminate in a two-pole coaxial connector, you might wonder why the hardware is fitted with four-pin output sockets, and there’s a good engineering reason. All cables have some resistance and incur some losses that increase with current flow, so it’s common practice in laboratory test gear to use ‘remote sensing’, with additional wires in the cable allowing the device to monitor the actual voltage being delivered at the end connector. This ‘sensing’ arrangement neatly circumvents the potential losses by adjusting the power supply’s output to achieve an accurate voltage at the destination’s input. RME have adopted this solution (only on the µFilter output) to ensure that a perfect 12V DC supply reaches the connected device.

LNI-2 DC The second power supply option, the LNI-2, is described as a “super‑low‑noise DC power filter and stabiliser”, with galvanic isolation between the input and output.

Essentially, this is the µFilter regulator section from the DPS-2, repackaged into a much smaller standalone unit ,measuring just 80 x 30 x 123mm and weighing only 605g. The case is milled from aluminium, with two LEDs on the top indicating the

presence of raw DC input and active, regulated DC output. Like the DNS-2, the µFilter output voltage is presented on a four-pin socket with the same remote sensing feature, and two 1m output cables are included, with locking and non-locking coaxial plugs.

A (not supplied) 9-18 V DC input source with a 2A capability (nominally 32W) is required, and connects via a coaxial plug (5.5 x 2.1mm, centre-positive). RME’s bayonet or standard coax plugs can be used. The LNI-2’s output is a rock-solid 12V DC, rated at 2A but capable of 3A, with a noise level below 2µV, an output impedance of 0.012Ω, and a load regulation within 0.2% — in other words, it’s the same impressively low noise and consistent DC supply as the DPS-2’s µFilter output.

Like the DPS-2, a ground terminal allows the output ground to be connected to a protective or technical system earth.

This is an important feature, as the switch-mode regulator employed in the LNI-2 uses a high-frequency transformer as part of its DC-DC conversion system, and that transformer provides the galvanic isolation between the external power supply source and the unit’s filtered DC output. Such isolation can be very useful for reducing ground-loop problems and noise current leakage issues, but sometimes it’s helpful to be able to tie the DC supply to a reference ground, and the terminal provides that option. Again, a manual (17

pages) explains the theory of operation in useful but not overwhelming detail.

Impressions Both the DPS-2 and LNI-2 power conditioners are superbly engineered products that achieve exactly what they are designed to do: provide the cleanest possible, solidly consistent DC power supplies for sensitive audio equipment.

The technical specifications attest to that, and in use, while both units can get quite warm to the touch, they’re silent.

Of course, whether such power supply conditioning is genuinely required in practice is a more ambiguous matter, and one that’s heavily dependent on the detailed design of the equipment being powered, as well as the state of the local mains supply, and the grounding efficiency of the entire installation.

To test for any benefit in my studio, I compared the AES17 dynamic range performance of the ADI-2/4 SE’s A-D converter stage, substituting the standard line-lump power supply for the DNS-2. This product’s A-D converter achieved the best AES17 score I’ve ever measured at 124.3dB (A‑weighted), outperforming its sibling (the ADI-2 Pro) by a small but significant 0.3dB. To have scored so highly on its standard power supply indicates a very well-designed converter with incredibly low-noise internal power arrangements anyway, and for that reason I wasn’t entirely surprised that switching between the two power supplies didn’t reveal consistently reliable differences in the dynamic range figures.

However, in mitigation, the mains supply in my studio is unusually clean, symmetrical and with very low noise, coming from a substation that’s only a few hundred metres away, and feeds only domestic premises. In a location with a more contaminated mains supply, the DNS-2 or LNI-2 might well make a measurable or audible improvement, especially with audio equipment designed to a lower technical standard than RME’s ADI-2 series products.

summary These regulated DC power supplies are astonishingly well designed.

£ DPS-2 £1208, LNI-2 DC £601.

Prices include VAT.

The DPS-2 accepts a mains AC input, while the LNI-2 DC can run off 9-18 V DC supplies.

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T Synthax Audio UK +44 (0) 1727 821 870 E info@synthax.co.uk W www.synthax.co.uk W https://rme-audio.de

Intel Plus Ultra, Further Beyond Expectations Faster, Cooler, Quieter with the new Intel Ultra 200 Series

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scan.co.uk/proaudio • 01204 47 47 47

ON TE ST

SAM INGLIS

D

espite its small size, the Bang & Olufsen BM3 ribbon microphone casts a long shadow. Decades after the Danish manufacturer gave up making mics to focus on hi-fi and lifestyle products, it remains the ultimate inspiration for current products from manufacturers such as Royer and Extinct Audio.

In the USA, Bang & Olufsen mics were imported by a company called Fenton, and the BM3 was sold under the Fen-tone moniker. This brand name has now been revived globally by Warm Audio for a microphone very obviously inspired by the BM3.

Warm Audio Fen-tone Ribbon Microphone

Is there substance as well as style to Warm Audio’s Danish-inspired ribbon mic?

Shine On The Warm Audio Fen-tone closely mimics the shape of the original, to the point of including the ‘collar’ around the base even though this has no obvious function in their version. Internally, however, Warm have incorporated modern improvements, and as well as offering much greater sensitivity in passive mode, the Fen-tone has a built-in, fixed-gain preamp. So, in place of the BM3’s three-way switch, which turned it on and off and introduced a high-pass filter, the Fen-tone has a two-way toggle that transforms it into an active, phantom-powered ribbon mic.

The Fen-tone is available individually or in stereo sets; the latter were not yet available at the time of this review, so I can’t comment on Warm Audio’s pair matching. It is supplied in a tough and practical Peli-style plastic case, with a nice branded sock to protect it from wind blasts when not in use. However, you don’t get any sort of a shockmount, just a cheap ‘crocodile clip’ sprung holder made of black plastic, which doesn’t exactly enhance the stylish vintage aesthetic. Though well built, the Fen-tone is noticeably smaller and lighter than either the BM3 or other modern interpretations such as the Royer R‑121 and Extinct Audio BM9, in part because the main housing is made from thinner metal than is used in those mics. The other obvious visual difference between the Fen-tone and the other mics is that it’s a lot more shiny!

Band Politics In a ribbon mic, the transducer is a thin strip of conductive metal suspended in a magnetic field. For this to work effectively, the ribbon needs to be corrugated. This increases its effective

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The Warm Audio Fen-tone (left) is both smaller and shinier than its inspiration (right) but very much recreates the general aesthetic.

length, and allows the resonant frequency to be fine-tuned by varying its tension. Bang & Olufsen exclusively used ‘pistonic’ ribbons, corrugated longitudinally for most of their length, with just a couple of horizontal pleats at either end. This has some theoretical advantages over the simpler ‘zig-zag’ approach, although whether these translate into measurable or audible differences is debatable. However, none of the modern BM3-derived mics that I know of have pistonic ribbons, and the Fen-tone is no exception. Sensibly enough, Warm Audio have used pure aluminium as the ribbon material rather than the alloy in the originals, which is prone to losing its elasticity over time.

At the same time, Warm Audio have also steered clear of reproducing some of the innovations that other manufacturers have brought to the basic BM3 design. For example, the ribbon in the Royer R‑121 is slightly offset from the centre of the frame, which gives the front and back sides of the mic somewhat different tonalities. Warm Audio haven’t done this, so the Fen-tone is a true figure-8 mic and can be used as part of a Blumlein array, or as the Sides mic in an M-S setup. They have also kept faith with the straightforward magnet geometry of early BM3s, rather than the more involved arrangements found in later versions and some derivatives.

In practice, though, the Fen-tone is both more sensitive than many passive ribbon mics, and not at all noisy in everyday use. And if passive mode is too quiet, the built-in ‘Warm Lifter’ delivers the promised 26dB gain boost. Unlike the active stages found in some cheap ribbon mics, it does so without swamping the signal in additional hiss.

A natural point of comparison for the Fen-tone was my own Extinct Audio BM9 ribbon, another modern take on the BM3 design. Subjectively, the two sounded very similar through the midrange, and on sources such as male vocals, you’d be hard put to tell them apart. I sometimes thought I could detect a slightly gritty quality to the upper midrange on the Fen-tone that wasn’t present on the BM9; depending on the voice, this either gave it a welcome edge that helped it cut through, or slightly negated the appealing smoothness of the ribbon tone. Either way, it was marginal, and you could easily drop in on a vocal track recorded with the other mic without anyone noticing.

successfully matched the sound, whilst offering the sensitivity we have come to expect from modern mics and the convenience of active circuitry. And in a sense, that’s exactly what you’d expect, because Warm Audio’s specialism is modernising classic designs and making them more widely available.

However, we’ve also come to expect Warm Audio’s recreations to be more affordable than the alternatives, and that’s less obviously true of the Fen-tone. For example, the aforementioned Extinct Audio BM9 is only a little more expensive, and whereas the Fen-tone is made in China to a good standard, the BM9 is built in the UK to an exceptional standard.

Across the pond, the Royer R‑10, Alder H44S and Samar AL95 are high-quality ribbon mics made in the US, and all are less costly than the Fen-tone, albeit that none has the same B&O-derived visual aesthetic. (If you love that look, you could even buy an original B&O BM3 and have it upgraded with modern magnets for around the same price.)

The main thing that all of these alternatives lack compared with the Fen-tone is the switchable active preamp. This in itself is quite an unusual feature — most ribbon mics are either active or passive, but not both — and does its job very well. A good-quality inline preamp such as Warm’s own Warm Lifter will typically set you back at least £100, so this definitely adds value. In my tests, I found that the sensitivity in passive mode was healthy enough that I rarely needed or wanted to use active mode, but if you don’t have good high-gain preamps, or you want to use long cable runs, it could make all the difference.

The Warm Audio price advantage is less clear-cut with the Fen-tone than with most of their products, then, but it undoubtedly represents another neat exercise in vintage cloning from the company who lead the way in this field.

“Warm Audio haven’t just replicated the look and feel of the Bang & Olufsen BM3. They’ve also successfully matched the sound.”

Numbers Up

Photo: Samuel Fitzgerald Kay

Warm Audio quote fairly detailed specifications for the Fen-tone, including a frequency response that is ±3dB from 30Hz to 15kHz; within this, their plot suggests a slight presence lift around 2kHz and a smooth roll-off above 10kHz or so. They also specify a signal-to-noise figure relating to use in passive mode.

However, since the only source of noise in passive mics is resistive or thermal noise, the signal-to-noise ratio is a function of sensitivity and output impedance; and Warm’s quoted figures of -47dBV ref 1V/Pa and 300Ω respectively don’t quite match their claimed 87dB SNR. Moreover, the output impedance of the review mic in passive mode actually measured 600Ω rather than the specified 300Ω. This is probably because the construction uses longer wires between the ribbon motor and the transformer than are really necessary.

However, clear differences emerged when I tested the two mics as overheads on a drum kit. The BM9 was audibly more extended above 10kHz, and the bass drum sounded less weighty on the Fen‑tone.

EQ matching suggested that the Warm Audio mic was at least 5dB down at 50Hz compared with the BM9. More scientific measurements later confirmed this, and also revealed that the Royer R‑121 and the original B&O BM3 both fall somewhere inbetween the two in terms of bass response. Impressively, the Fen‑tone’s measured response matched that of the BM3 extremely closely between about 100Hz and 10kHz — a band that covers most of the frequencies that matter for most sources! The slightly shy bass response could be partly down to the tension of the ribbon, but it could also be that the relatively high output impedance in passive mode loads the mic preamp, as it improved a little in active mode.

Warm Or Cold?

With the Fen-tone, then, Warm Audio haven’t just replicated the look and feel of the Bang & Olufsen BM3. They’ve also

summary The Fen-tone closely matches the look, feel and sound of the classic B&O BM3, with the useful option of an active preamp built in, but faces some strong competition.

£ £639 including VAT. T Source Distribution +44 (0)208 962 5080 E sales@sourcedistribution.co.uk W www.sourcedistribution.co.uk W www.warmaudio.com

www.soundonsound.com /June 2025

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GForce Novation Bass Station

Software Synthesizer

A second page reveals a cunning Macro sequencer where you can program per-step values of the two macro knobs.

The macros can be mapped and scaled to as many controls as you like. They are fantastic for pulling off radical changes with a single flick of your mouse or MIDI controller. I’d love to see the mapped knobs move in response, but apparently that’s not a thing. The macro knobs are also present and helpfully labelled in the preset browser, which is handy for auditioning sounds.

Oddly, the labels are not carried through to the main page.

Extras

Now a classic in its own right, Novation’s Bass Station gets the GForce treatment.

ROBIN VINCENT

B

ack in the early ’00s, Novation’s ‘Virtual BassStation’ was considered by Sound On Sound to be “A stable, fine-sounding software emulation”. However, all memory of that has been purged to make space for GForce’s definitive take on the Bass Station.

Station To Station So, 22 years on, what’s changed? Well, it looks great. It’s a lot less distorted than the earlier one and feels much more comfortable in its skin. Compared to the hardware, they’ve added an extra octave to the keyboard, but that balances out once you add the extra modulation and effects section above the main controls to form the familiar Bass Station shape. There’s a very funky oscilloscope, and overall you get lots of lovely nostalgic Bass Station vibes.

It’s no surprise, given GForce’s track record, that it sounds marvellous, although perhaps a lot more epic than it should. It might have been nice to have a ‘Pure Bass Station’ preset category that gives you the original sound unfettered by all the fabulous modern updates. For that I guess we can start with an ‘Init Patch’.

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The basic patch gives you a slightly detuned dual sawtooth, and from there you can explore everything the original Bass Station had to offer. This is made easier because all the doubled-up controls, like oscillator modulation and envelopes, have been expanded to fill their own space. Without all the extras, it initially feels underwhelming in its dry simplicity, but as you tweak and pull on the filter, it starts to open up into the great-sounding little synth that it is.

Sequencing A switch above the keyboard replaces the keys with a detailed sequencer panel and arpeggiator. The arp is super simple with a handful of modes, octave range, swing, rate and a useful probability knob. With the sequencer, you have 16 monophonic steps with accent, slide probability and velocity for each. The pitch is shown as a grid of 12 notes combined with an octave setting. So your sequence always appears within those 12 notes, but the notes themselves could be octaves apart. The octave selection on the first note seems to transpose the entire sequence and is upside down for reasons I can’t explain.

However, it’s brilliantly easy to program, has cool randomised generation of notes, length and velocity, scales for instant melodies, and you can step-record.

It’s the perfect way to get your acid on.

Along the top, we have a row of modulation and effects. The XLFO and XADSR let you map modulation to any selected parameter. You have multiple LFO waveforms, and control over the delay, sample & hold, smoothness and phase. There’s a little display showing the modulation output, and you can copy modulations from one parameter to another. You can modulate everything independently, giving the Bass Station enormous movement and possibility.

While there are useful visual clues to what’s being modulated, I would really appreciate a bit of knob animation.

The effects offer a really decent level of excitement, depth, change and placement.

The distortion lands just right for those ripping bass lines, the chorus fattens things up nicely, and the delay and reverb throw the sound out into the wilds with the right amount of presence. There’s even a high-pass filter for those people who don’t think low‑pass filters are all you need.

Polyphony is not really what you look for when approaching a Bass Station but by golly does it add a whole new dimension. It becomes more Juno-like if you turn up the Vintage knob and more Peak-like as you lean into the effects. This is where all the modulation possibilities are given space to breathe, and it’s a lovely place to spend some time.

summary The Bass Station is a decent little dual-oscillator synthesizer that GForce have taken to stratospheric heights. It has all the authenticity you could want, a fun sequencer and the ability to open doors to sumptuous and interesting polyphonic sounds while keeping its feet firmly in their acid roots.

£ £99.99 including VAT. W www.gforcesoftware.com

The sound that defined an era

SILVERR 47 ®

ALL TUBE DUAL MIC PREAMP

The sound of yesterday, built for tomorrow.

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P.LANE TYPE 436

ALL TUBE VARIMU COMPRESSOR

ON TE ST

beyerdynamic DT 70 IE In-ear Monitors

beyerdynamic’s first ever in-ear range comes in multiple flavours, including an explicitly studio-focused model.

SAM INGLIS

L

ike full-sized headphones, in-ear monitors can be designed using several different technologies, all of which have their own strengths and weaknesses. Balanced-armature drivers are now very popular, and can achieve impressive technical specifications when multiple drivers are used to handle different regions of the frequency spectrum. The flip side of this is that crossover artefacts can introduce a slightly unnatural quality to the sound. Elsewhere, manufacturers such as Audeze have experimented with planar magnetic drivers, to good effect.

Given beyerdynamic’s heritage as one of the world’s oldest manufacturers of dynamic microphones and headphones, it’s no surprise to see that they have kept faith with the more traditional moving-coil driver in their first-ever range of IEMs. But that’s not to say that they aren’t innovating.

The DT 70s use a new TESLA.11 driver that achieves a total harmonic distortion figure of less than 0.2% at 1kHz, which is impressive,

and beyerdynamic are also offering what is outwardly the same model in four variants, with different ‘voicings’ intended for different applications. The DT 71 IEs are tagged ‘drum & bass’, the DT 72 IEs ‘guitar & voice’ and the DT 73 IEs ‘classic instruments & keys’, whilst the DT 70 IEs are said to be intended for studio uses such as mixing and critical listening. Appropriately enough, the DT 70s were supplied for review.

Moderation In Everything

The Y cable for the DT 70s incorporates earhooks that can be moulded to achieve a secure fit around the user’s ears.

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If you’re curious as to what differentiates these four variants, beyerdynamic make available a handy PDF document which attempts to characterise them in terms of four frequency bands labelled ‘bass’, ‘fundamental tones’, ‘midrange’ and ‘treble’.

The DT 71 IEs, for example, are ‘very

strong’ in the bass and treble, but ‘soft’ in the midrange, whilst the DT 73 IEs are ‘moderate’ in all bands except the treble, where they are ‘strong’. Perhaps surprisingly, the studio-focused DT 70 IEs are portrayed as having quite a scooped response according to this scale, with a ‘strong’ response in the bass falling through the ‘fundamental tones’ region to a ‘soft’ midrange before shooting back up to ‘strong’ again in the treble region.

The DT 70s are supplied in a robust and practical soft-shell case. They are very compact, and connect to a removable Y-shaped cable using MMCX connectors.

At the business end, both branches of the cable are reinforced by a semi-rigid sheath that retains its shape. The user can thus bend this to form appropriately shaped earhooks to hold the DT 70s in place. As the DT 70s are ‘universal’ IEMs, there’s no option for custom moulds. Comfort, sound quality and isolation are all therefore dependent on your choice of tips. Unless your ears are particularly unusual, you should be able to find something suitable among the included complement of five

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BEYERDYNAMIC DT 70 IE

Like all of beyerdynamic’s high-end products, the DT 70 is manufactured in their factory at Heilbronn, Germany.

silicone and three Comply foam tips in various sizes. I had no difficulty achieving a setup that stayed in place confidently.

Driving The DTs With a quoted 16Ω nominal impedance and a sensitivity of 113dB SPL for a 1mW signal at 500Hz, the DT 70s should be as easy to drive as any modern pair of passive headphones, and I had no problem getting them up to ‘ouch’ levels on a laptop headphone socket.

The specifications claim a frequency response of 5Hz to 40kHz, but as ever, this is pretty meaningless in the absence of any tolerances.

Having seen beyerdynamic’s own characterisation of their tonality, I was a little apprehensive about how the DT 70 IE would sound. I’m not generally a fan of scooped playback systems, and the thought of having ‘strong’ bass and treble frequencies blasted directly into my ears didn’t appeal.

In practice, however, I found it hard to relate this description to the actual sound, which struck me as mostly well balanced.

There is an audible low emphasis, true; but to my ears, this is relatively restrained, and occupies the 100-200 Hz range rather than the true bass region. Below 100Hz the response seemed to fall away somewhat, but this is the region most affected by the

fit of the IEMs, so I suspect other people’s experience might be different.

Isolation is also directly affected by the fit, and the specifications claim that the DT 70s offer “up to 39dB” of noise attenuation. This, inevitably, is quite frequency-dependent, but they seem to do a decent job of keeping out external noise.

Subjectively, I’d say they offer a fair bit more isolation than a typical pair of passive

they are as flat as any other in-ear models I’ve heard in the same price bracket.

Given that the obvious use cases for in-ear monitors mostly concern live music, it’s interesting that beyerdynamic’s first ever range includes a model specifically targeted at studio applications. Would I choose to mix on them if I had the choice of using either studio monitors, or a highend pair of open-backed headphones such as beyerdynamic’s DT 1990 PRO MkII? Possibly not, but that’s hardly the point. IEMs will go where conventional monitoring systems won’t, and the DT 70 IEs will let you get useful editing and mixing work done in circumstances where nothing else will. And, of course, although they’re intended for mixing and critical listening, there’s nothing to say you can’t use them on stage!

“Given that the obvious use cases for in-ear monitors mostly concern live music, it’s interesting that beyerdynamic’s first ever range includes a model specifically targeted at studio applications.”

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closed-back headphones, but less than you can achieve with active noise cancelling.

This would certainly be a boon in those circumstances where, for example, you have to get sounds on location whilst setting up in the same room as the musicians.

Most importantly, despite their alleged strength in the treble region, I didn’t find the DT 70 IEs at all over-bright or fatiguing. In fact, compared to models that use multiple balanced-armature drivers, the upper midrange is quite smooth, and the ‘air’ frequencies a little subdued. If anything, the DT 70 IE’s most prominent characteristic is a slight emphasis somewhere in the 1kHz region that can occasionally exaggerate the ‘knock’ of snare drums and the like. All in all,

summary beyerdynamic’s entry into the IEM market is typically classy, and the DT 70 IEs make a powerful case for the idea that in-ears can be a valuable tool in the studio.

£ £429 including VAT. T Polar Distribution +44 (0)1444 258 258 E sales@polar.uk.com W www.polar.uk.com W www.beyerdynamic.com

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Audix D6X Dynamic Microphone

Audix’s new drum-focused dynamic is three mics in one.

SAM INGLIS

A

t the time of writing, Audix’s vast catalogue includes no fewer than 168 wired microphones. Whether you’re looking for a discreet boundary mic or a specialist clip-on model for an obscure wind instrument, the American manufacturers are sure to have something to offer. And out of all of these models, perhaps the closest thing they have to an industry standard is the D6. Developed for use with bass drums and other low-frequency, high-SPL sources, this stubby dynamic mic is a familiar sight on stages and in studios all over the world.

Rival Schools There are two schools of thought relating to bass drum mics. In one corner are the purists, who feel that the bass drum is an instrument like any other and should be captured in the most neutral fashion

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possible. They are likely to gravitate towards something like the DPA 4055, or perhaps to a general-purpose mic like the Neumann U47 FET or Electro-Voice RE20.

On the opposite side of the room we find the realists, who anticipate that tons of equalisation is going to be needed to make the average bass drum fit into the mix, and feel that this might as well happen in the mic, to save them a job further down the line.

The Audix D6 belongs firmly in the second category, along with models such as the AKG D112 and Shure Beta 91A. It’s deliberately designed to have low sensitivity, so that it won’t overload preamps and converters even when placed inside a loud kick. And it has a ferociously ‘scooped’ frequency response that is intended to emphasise the click of the beater and the whoomph of the low end, at the expense of all that nasty midrange you were going to dial

out at the mix anyway. To some, the D6 delivers the perfect ‘mix ready’ bass drum sound. But to others, its relatively extreme frequency response takes things a bit too far, and it certainly limits the alternative applications for such a mic.

Consequently, Audix have now followed manufacturers such as AKG and Shure in offering a kick drum mic with multiple, switchable tonalities.

Switch It Up That mic is the D6X, and it’s largely identical to the D6 apart from a recessed three-position switch located about halfway along the shell. I like the way Audix have implemented this switch: you can adjust it with your finger or thumbnail, but it is immune from accidental operation, and there’s never any doubt as to which position is selected. In every other respect, the D6X is just like the D6, meaning it’s a compact mic that is easy to manouvre into place and lightweight enough to stay put

on its supplied swivel clip. The robust aluminium shell and gloss black finish are unchanged apart from the large X that has been carved out of the metalwork.

The D6X is described as being a cardioid mic. Audix’s graphs suggest the pattern errs towards the supercardioid at most frequencies and is not as well controlled as you’d get from something like a small-diaphragm capacitor mic, but given the applications, that’s hardly likely to be an issue in practice. Sensitivity is listed as 0.8mV/Pa, but for a 75Hz signal rather than the more usual 1kHz tone; again, this reflects the intended role of the mic.

In its first position, the switch introduces exactly the same filtering as is found on the D6, making the two mics to all intents and purposes identical. In the centre or Balanced position, the ‘smiley face’ frequency response is dimished but still present, while the third position is described as Natural. The frequency response plots suggest that this is a somewhat relative interpretation of ‘natural’, with a broad scoop between 100Hz and 2kHz and a gentle presence emphasis between 3 and 7 kHz. The same plot shows that Balanced and D6 modes both have extended high‑ and low-frequency responses compared with the Natural setting, with D6 mode going a dB or so further in both respects. The mid scoop is another 3dB or so more pronounced in Balanced mode and much more so in D6 mode, bottoming out at 800Hz some 16dB below the nominal zero.

Compared with something like AKG’s D12VR, the tonal variations on offer are much more radical here, at least on paper.

The D6X’s three-position switch is recessed to prevent accidental operation, but can still be adjusted without a tool.

extent. I can’t deny that the D6 is one of those mics that has sometimes produced the latter reaction: perfect for metal drumming and for getting a usable sound fast when time is of the essence, perhaps not something

want to make on a general-purpose mic such as a beyerdynamic M88.

I also found Balanced mode a lot more useful than full‑fat D6 mode on kick-drum-adjacent sources such as floor tom. Finally, Natural mode is something of a revelation. It still has plenty to offer on percussive sources, and if it’s not sculpted enough to produce your kick drum sound of choice, it’s very much worth trying out on snare and rack toms. It even turns the D6X into a surprisingly effective vocal mic, as long as you don’t mind its low sensitivity. If the Swiss Army were to issue recruits with an instrument microphone, I imagine they’d come up with something very much like the Audix D6X!

“In the centre or Balanced position, the ‘smiley face’ frequency response is dimished but still present, while the third position is described as Natural.”

Centre Parting When it comes to drum miking, I have often found myself standing uncomfortably in the middle of the room occupied by the purists and the realists.

The times I’ve tried using something like an RE20 for miking a bass drum, I’ve ended up with some pretty extreme EQ settings at the mix. But when I’ve used a mic that’s deliberately tailored for the role, I’ve sometimes wound up with the sense that it’s making every kick drum sound the same, or emphasising the beater attack to an uncomfortable

you’d reach for on a jazz session.

If all the click and all the sub is what you want, the DX6 absolutely delivers the goods with the switch in the D6 setting. I couldn’t detect any difference between the two mics, not that it’s easy to position them close together inside a bass drum.

But if that’s not what you want, the other two settings really do greatly increase the mic’s usefulness.

I am not sure that Balanced mode sounds balanced on anything other than a bass drum, but there, it often seems to replicate the EQ moves I’d

summary A simple three-position switch transforms Audix’s D6X into much more than a bass drum microphone.

£ £249 including VAT. T SCV +44 (0)330 122 2500 E sales@scvdistribution.co.uk W www.scvdistribution.co.uk W www.audixusa.com

www.soundonsound.com /June 2025

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Screen 1: Sphere 2’s main screen

PHIL WARD

T

he term ‘monitor controller’ used to refer to a hardware box that sat on your desk and provided volume control, output selection, some channel matrix options, a headphone socket or two, and maybe a talkback mic. But although hardware monitor controllers are still very much ‘a thing’, monitor control has also moved into the software realm. When paired with a suitable audio interface, software can offer all the same functionality — including, if you add an external controller, a big volume knob.

Ginger Audio’s Sphere 2, though, offers much more besides. In this review I’ll be focusing on the Immersive edition, but Stereo (2.0, 2.1) and Surround editions are available for a lower price, offering most of the same features but being constrained in terms of channel formats. All three editions are available via subscription, or you can pay once for a perpetual licence.

Sphere is a standalone macOS app that sits between the computer’s audio apps (a DAW or Apple Music, for example) and the audio interface’s outputs. When describing the full scope of its features and functionality, it’s tempting to resort to the cliché that it would be easier to list the

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Ginger Audio Sphere v2 Monitor Control Software For macOS

Whether you’re mixing or mastering in stereo, surround or immersive formats, this powerful virtual monitor controller could soon become indispensable. things it doesn’t do. That’s because almost the first thing to strike you on launching Sphere is just how much it brings to the monitor control function. So steel yourself — there’s a lot to take in if we’re to explore Sphere in detail!

Key Functions The first thing to note is that Sphere Immersive offers not only 12 multi-channel virtual outputs (that can be fed to an audio interface’s outputs) but also eight virtual multi-channel inputs. You can (as I did) have a Dolby Atmos input from Apple Music

routed to one Sphere 2 input and another routed from Pro Tools, or the Dolby Atmos Renderer app. These inputs comprised 7.1.4 Atmos streams in my case but can range from mono all the way up to 9.1.6. The same variety of channel formats is available on each of the 12 outputs, but along with offering channel formats that simply reflect the inputs (which will mean, for example, that a stereo output from an Atmos input will comprise just the front left and right), outputs 5 to 12 can additionally comprise downmixes to various formats, including Apple Spatial binaural.

This is particularly significant in the world of Atmos music mixing because, until recently, the processing that an Atmos mix undergoes when submitted to Apple before it’s made available in binaural stereo on Apple Music was something of a mystery.

It would sound somewhat different from a binaural downmix created using the Dolby Atmos Renderer, and this resulted in Atmos mix and mastering engineers berating Apple for, effectively, messing with their work. Apple have now made the Apple Spatial binaural processing algorithms available to developers, so Sphere isn’t the only route by which Apple Spatial binaural can be auditioned while mixing or mastering, but it is a great facility to have, and along with the Apple Spatial binaural downmix, Sphere offers personal HRTF and head-tracking integration.

Somewhat unexpected to me was that Sphere is able to connect with multiple audio interfaces simultaneously — this feature appears to be aimed at users who perhaps want to get into Atmos mixing but don’t yet possess a single interface with an adequate output count. A further unexpected feature is the incorporation of Ginger Audio’s Authentic Audio network streaming technology. An output can be selected and bussed to Sphere’s streaming module, where it can be downmixed if required and made available as a stream (with or without video) that remote users can access in a web browser using a link generated by Sphere 2. This is incredibly simple to use and, again, a quick audio-only check confirmed that it works well. Network streaming is not normally a function expected of a monitor controller, but it could be hugely useful, because it can potentially turn any mix or mastering session into an ‘attended’ event. You simply send the client a Sphere 2 streaming link and they can listen, in real time, from pretty much anywhere that has a reasonable Internet connection. And if they want to hear how the work will sound in Apple Spatial format, that’s just a couple of mouse-clicks away too.

are 12 input slots arranged under two tabs.

In each slot are a text field for a custom input name, an input level meter, input gain adjustment, a lock function (which effectively toggles the input on/off), a plug-in slot, and finally a cue send with an on/off button and output gain knob. The plug-in slot enables just one AU plug-in to be inserted, and said plug-in will then be installed across all the input channels. If the plug-in and input channel format don’t match, either some channels will be muted, or Sphere 2 will flag with a red exclamation mark that plug-in channel mapping is an issue. The same flag will appear if the plug-in is otherwise inappropriate.

Also in each input slot is a ‘settings’ icon, that opens a channel mapping window.

Here, specific inputs can be assigned to numbered Sphere channels. Screen 2 shows assignments for my Audient ORIA interface, and I’ll explain them shortly.

But first, the reason for the SPHERE 16 name in the Channel column is that, when installed, Sphere 2 creates three virtual audio devices (SPHERE 16, SPHERE 32 and SPHERE 64) that can be specified as the output device for the audio source apps (in my case, Pro Tools or Apple Music), and one of those devices then becomes the input for Sphere. The further explanation required for channel assignments arises

Ginger Audio Sphere v2 $500 pros • Incredibly versatile signal routing and control.

• Comprehensive traditional monitor control functions.

• Apple Spatial binaural downmix.

• Network streaming.

• Volume analysis.

• Plug-in slots.

cons • None.

summary As monitor control solutions go, Sphere 2 may not be inexpensive — but it is so massively capable that in the right studio and workflow context it could well feel like a bargain.

because, while the ORIA offers 16 surround outputs for monitoring layouts up to 9.1.6, my layout is ‘only’ 7.1.4. The ‘missing’ monitors are Left Wide, Right Wide, Left Top Mid and Right Top Mid, which would nominally be numbered 9, 10, 13 and 14, so my channel assignments omit those numbers. Along with channel assignments, the Input Settings screen also enables channel gain trim, solo and mute functions.

To the right of the input slots you find the features that comprise the bulk of

Settings & Controls Returning to the core functions of Sphere, once installed and licensed (using a simple licence code) Sphere launches to display its main screen, as illustrated in Screen 1. Pretty much everything required for day-to-day operation is presented on the main screen, so once all is configured, there’s little need for hopping between windows. To the left of the main screen

Screen 2: Sphere 2’s input channel setup and assignments.

www.soundonsound.com / June 2025

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GINGER AUDIO SPHERE V2

Screen 3: Sphere 2’s output channel setup, assignments and bass management.

Sphere’s conventional monitor control functionality. The 3D map of monitors reflects the input channel configuration and, depending on the solo/mute switch setting, each monitor can be soloed or muted. Monitors can also be soloed or muted in Front, Wide, Surround, Rear, Top or Sub groups, or in custom groups (combined by Shift-clicking monitors). At the bottom left of the monitor display are options for left and right front channel mono, Mid and Sides functions, but these only affect the front left and right channels so seem somewhat academic in a multi-monitor context.

Below are Sphere’s loudness analysis functions.

Pressing Play starts loudness analysis, and pressing Reset returns it to zero.

A drop-down menu on the left provides a list of the streaming platforms whose loudness specifications Sphere 2’s analysis can be set to reference. The peak loudness meter to the right of the output metering will show the loudness peak level for the streaming platform selected.

To the right of Sphere 2’s output level metering is its master output level control.

Sphere 2 is EuCon, StreamDeck, MIDI and OSC compliant for control by external hardware, so if you miss the tactile feel of a big volume knob, that can be sorted.

Around the volume control are dim and

mute buttons along with three reference level selection buttons that can be set via a settings display, launched by clicking on the three dots above each one.

Again looking right, Sphere offers cue and talkback functions that can be routed to any output, and then sub-routed to specific channels offered by the output device. Both the cue and talkback

7.1.4 format is routed to output 4 but stereo is selected in the output format, all but the left and right channels will be muted.

However if the same 7.1.4 audio is routed to output 5 and stereo is selected in the output format, the 7.1.4 will be downmixed so that the 5.1.4 channels are folded into stereo in compliance with the Dolby Atmos algorithms.

In common with a Sphere input plug-in, an output plug-in operates across all channels, and the same constraints apply: strap a stereo plug-in across a 7.1.4 output and Sphere will give you the red exclamation mark and mute all but the left and right channels.

To my way of thinking, the nailed-on application for the Sphere output plug-in slot is Sonarworks SoundID Reference. I say this because, somewhat frustratingly, the multi-channel version of Sonarworks operates as a DAW plug-in only — there’s no ‘systemwide’ version that can address and optimise channel formats beyond stereo. Drop Sonarworks multi-channel into a Sphere output plug-in slot, and that frustration goes away.

The final image, Screen 3, shows the output side equivalent of the input channel mapping window of Screen 2. It’s accessed by clicking on the settings icon in an output slot and, as with input channel mapping, enables Sphere channels to address the

“You simply send the client a Sphere 2 streaming link and they can listen, in real time, from pretty much anywhere that has a reasonable Internet connection.”

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feature an AU plug-in slot (a smidgen of reverb for the singer?), and the cue function additionally offers Dolby or Apple downmix functionality.

Beneath all this, you’ll find the Sphere 2 Authentic Audio network streaming controls, but before we dive into those we’ll head right again to the output section.

As with the input section, Sphere’s outputs are found in 12 slots across two tabs. Each output can be configured with any channel format up to 9.1.6 and each one offers an AU plug-in slot. Output slots 5 to 12 are slightly different from slots 1 to 4 in that they offer downmixing rather than simple channel muting. For example, if audio in

appropriate hardware interface output.

But it also provides bass management configuration, for which the term ‘comprehensive’ feels inadequate!

Next, we come to Sphere’s Authentic Audio network streaming controls I mentioned earlier, and the functionality is so straightforward in use that there’s not a great deal to say. Streaming simply requires a Sphere output to be selected, its channel format to be specified and, if required, a session name and password to be defined. Once that’s done, pressing the Start Streaming button generates a web link to send to the collaborator; click on that link and the audio will play in a browser. Perhaps the only complication is that simple browser streaming playback is stereo only. If you’ve selected a multi-channel output, the Sphere 2 stream will play just the left and right channels, so better to select a stereo downmix output. If a full-fat multi-channel stream is required, a Ginger Audio Authentic Audio plug-in is necessary.

Finally, we come to the question of latency and before we talk figures,

ALTERNATIVES I’m not aware of any software that compares directly to Sphere 2, though ListenHub from Sonnox provides some related functionality (but is stereo only).

Of course some hardware can offer the same features via their control software.

note that I used Sphere while recording and didn’t notice any significant issues.

Still, there is inherently some latency, and you can adjust it in the settings menu. For recording, you’d set it to a low value (which asks more of your CPU) and at its lowest setting (32 samples) I measured 10ms latency.

If monitoring a complex Atmos mix you can set it higher to free up your CPU. For the highest setting (2048 samples) latency rose to about 136ms.

Going Global I came to this review not quite sure what to expect from Sphere 2, or even why I might find a multi-channel monitor control app useful. After all, I already

have a multi-output interface with its own control app that together effectively provide monitor control. I’m leaving it as a fully convinced enthusiast. These are complex times for audio production in that satisfying disparate demands of a multitude of consumer formats requires considerable versatility, understanding and agility from us audio engineers. In this context, Sphere 2 is potentially of significant value because the capabilities and qualities it offers so effectively tie all the many contemporary monitoring control functions together in one place.

It dots the ‘i’s and cross the ‘t’s of a session really effectively. It’s a great product, made perhaps more so because I genuinely didn’t realise I needed it. £ Sphere Immersive perpetual licence

$499.99, subscription $24.99/month (discounted to $17.49 when going to press). Sphere Surround $399.99 or $19.99/month (currently $13.99).

Sphere Stereo $249.99 or $14.99/ month ($10.49).

E info@gingeraudio.com W www.gingeraudio.com

www.soundonsound.com / June 2025

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Slate Digital VMS 2 NEIL ROGERS

I

first reviewed Slate Digital’s Virtual Microphone System (VMS) way back in 2016. Microphone modelling wasn’t an entirely new concept at the time, but Slate Digital’s approach was unique: while other systems were entirely software-based, the VMS relied on the use of a specific reference microphone that could provide a blank canvas on which software emulations of classic mics could be projected. Eight years later, in January 2025, I took a fresh look at the VMS, to see how the system had evolved and expanded since its original release.

When I was asked if I wanted to look at a brand-new version of the VMS, my first

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Microphone Modelling System

Version 2 of Slate Digital’s Virtual Microphone System offers new hardware, a comprehensive software refresh — and the fruits of one of the best-stocked mic lockers in the world. question was whether there were enough changes to talk about since my last review! The answer was compelling: the VMS 2 features a brand‑new modelling microphone, as well as a dedicated plug-in for handling the software emulations (previously this was done using Slate Digital’s general‑purpose Virtual Mix Rack plug-in). We’ll be taking a detailed look at both of these new components, as well as exploring a brand new pack of microphone models based around selected items from the

extremely well stocked mic locker at Radio France.

The New Mic Slate Digital have given the new ML-1A microphone a cosmetic refresh, and it certainly has a more modern, stylish look than the original ML-1. And, although they are giving little away in terms of the origins of the capsule or electronics, the specs have also been improved, with lower self-noise, greater sensitivity and a higher maximum SPL rating. This last

improvement seems the most significant on paper, with a maximum SPL of 145dB (when the -20dB pad is engaged) being very impressive and perhaps allowing for more flexibility when it comes to replicating mics that are typically used for close‑miking very loud sources.

The new mic ships with a shockmount (but is also compatible with standard swivel mounts) as well as a compact Peli‑style case. This is something I’ve always considered a much more sensible option than a large hard case, and much more practical for a mic that will often be moving about between bedrooms, studios and practice rooms.

I mentioned in my previous review that the original ML-1 mic was a good-sounding capacitor mic in its own right, and I can confirm that the new ML-1A also ticks that box: I ended up using the ‘raw’ mic as part of my drum setup before getting the new software installed, and the ML-1A sounded clear and punchy above the kit. I feel this is important to mention for a couple of reasons. First, it allows you to use the mic even if your setup doesn’t let you monitor through the software in real time, so you can still focus on your microphone technique. Second, in a busy session it’s unlikely to be practical to scroll through the many emulation options available (this is one of the aspects of using virtual mics that still makes me feel a little uneasy), but the fact you’re getting a good-sounding mic in its own right allows you flexibility in terms of how you use the whole package. Finally, a good-sounding mic will always be a useful and future-proof investment!

your mic is positioned from a source, and the Intensity control, which gives you the ability to exaggerate — or reduce — the inherent character of a particular model.

I’ll share a few thoughts about how I got on with these features later but generally, I found the new VMS plug-in to be easier to use than Virtual Mix Rack, and very well thought out.

Finally on the plug-in, if you’ve already invested in the previous ML-1 mic you’ll be pleased to hear that this new software is also intended to work with that mic. When opening the plug-in you are asked which Slate Digital microphone you are using; it will then automatically switch to the correct model every time you open the plug-in.

Vive La France!

In the UK, the BBC is part of the fabric of our lives, so it was interesting to learn a little about the French equivalent. Like the BBC, Radio France was a pioneer of early broadcasting technology, and over the years it has amassed a huge collection of microphones — over 5000 in total, covering 300 different models!

Slate Digital chose to model 10 of those

Slate Digital VMS 2 £479 pros • The new ML-1A microphone is a significant improvement on the original.

• Dedicated VMS plug-in makes the system easier to use and offers useful new controls.

• VMS software benefits owners of the original VMS too.

• Radio France Signature pack includes some unusual but very useful mic choices.

cons • None.

summary The Slate Digital Virtual Microphone System 2 features a brand‑new mic, dedicated software and a new mic pack from Radio France to spice up what was already an extensive collection of impressive‑sounding mic models.

for their Radio France Signature Mic Pack, and it’s an intriguing selection (see box for the full list). Sensibly, Slate Digital have focused on adding variation here, as the heavy hitters from the large‑diaphragm capacitor mic world are already well

The Software I was pleased when I heard that Slate Digital had decided to give the VMS its own dedicated software. While the original system worked fine, it did sometimes involve having to make more mouse clicks than was ideal when auditioning the different models. As well as being more efficient, the new dedicated VMS plug-in features some interesting new options to play with, such as a stereo widener and an ‘Interpolate’ feature that allows you to combine the qualities of two mic models of your choice. The new software also maintains two features I found useful in my previous review: the Proximity control, which allows you to mimic the tonal differences imparted by the distance

www.soundonsound.com / June 2025

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S L AT E DIGITA L V MS 2

represented in the existing packs. It’s great to see ribbon mics as well as some dynamic and small‑diaphragm capacitor options included.

However, one of the key aspects of this technology is that it offers us engineers access to the sound of mics we’re highly unlikely to ever own. Whilst some of the ‘rarities’ included in this pack certainly tick that box, many of the models are still very much available.

What I can say, though, is that even as a studio owner of nearly 20 years, I haven’t personally used the majority of the Radio France mics that Slate Digital have modelled, which perhaps validates the choices made by the VMS team. I was especially keen to hear the RF-42B, which is an emulation of the largest ribbon mic ever made, and the RF-21B, which is based on a quirky omnidirectional dynamic mic that is used by Radio France for interviews ‘in the field’. We also have the RF-X25 (based on a well‑known American model), as well as the RF-H64, which is based on a German pencil

Radio France Models The Radio France Signature Mic Pack includes 10 microphone models, all based on specific mics in the Radio France collection. Slate don’t publicise the exact models they used, but between the model numbers and the VMS plug‑in GUI, the inspiration is pretty clear: VMS Name RF-X25 RF-88 RF-BB6 RF-21B RF-149 RF-H64 RF-H64 Bonnette RF Amethyste RF-42B Voix RF-42B Musique

Microphone Type American large-diaphragm capacitor German moving‑coil dynamic Latvian large‑diaphragm valve capacitor French moving‑coil dynamic German large‑diaphragm valve capacitor German small‑diaphragm capacitor German small‑diaphragm capacitor with windshield Estonian large‑diaphragm capacitor French long ribbon in Voice mode French long ribbon in Music mode

microphone designed for handheld use with speech, or as a targeted spot mic in busy recording sessions.

In Use Vocals were the obvious place to start with the VMS 2, and it was revealing to audition the options from the Radio

France pack on an excellent male rock vocalist I was recording at my studio. The top end of his voice opened up in a lovely way using the RF-BB6 model, and I also liked how the RF-X25 brought some extra presence to his voice. The 149 emulation was really interesting, as it seemed to retain more of the sound of the ‘raw’

The microphone modelling is now handled by a dedicated VMS plug-in, rather than in the Virtual Mix Rack host plug-in as before.

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mic but with a little extra body and a touch more softness. The small‑diaphragm options also worked nicely on my singer, with the reduced low end not proving to be an issue in the context of a mix. The RF-H64 Bonnette (a version of the RF-H64 with the foam pop filter attached) also worked nicely for taming a little sibilance. Although they didn’t deliver the right tone for the genre of this recording session, it was interesting to hear how the large ribbon mic models sounded on my singer. Any sense of sibilance was quickly reduced, and his voice seemed darker and more chesty-sounding.

These 42B models would work superbly on certain female voices, I think, and they do present an authentic ribbon mic flavour in a mix.

I mentioned earlier that I was looking forward to trying out the reporter’s mic, as I love the idea of options that present something obviously different or unusual-sounding. But, while the RF-21B did sound more ‘lo-fi’ than the other models, in this session it wasn’t distinct enough, and left me a little underwhelmed. It did sound more dramatic on drums, however, which is where we’ll head next...

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Drums On the same session, I put up the ML-1A as an additional mono overhead mic for the drum recording. How a mic fares with loud, often harsh-sounding cymbals is a great test for a new mic and, as I mentioned earlier, the raw sound from the mic was clear and punchy, with the cymbals sounding respectable if a little sterile. Auditioning a couple of the large‑diaphragm mic models had the desired effect of ‘sweetening’ the top end, and I liked how the RF-X25 sounded in that regard. The RF-149 model produced quite a dramatic change to the midrange, which had the effect of bringing out the transients in the ride cymbal in particular. It was also nice to hear how the dynamic mic models blended with the rest of my kit mics. I’ve long been a fan of having a dynamic mic above a drum kit in conjunction with more standard overhead options; it sounds unimpressive on its own but can bring out a nice sense of 1-2 kHz detail in a guitar-heavy mix, and the RF-21B model worked well for this.

My favourite model in this drum setting, though, was the RF-42B in its ‘Musique’ setting: the high end of the cymbals was rolled off and it was great to hear that ‘thumpy’ low end you can get using large ribbon mics on drums. It was also interesting to play with the new Interpolate feature here: whilst the sound of large ribbon mics can be pleasing on drums, I often find myself heading for the EQ to add more presence or definition in the context of a mix. By blending in something brighter — like the small‑diagphragm FR-H64 — I was able to find that ‘mix ready’ sound without reaching for an EQ plug-in. I should add that Interpolate is not the same as a mix function. It doesn’t simply blend between the sound of one mic and another: rather, it continuously morphs between two models. (If you do want a straightforward blend

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“As well as being more efficient, the new dedicated VMS plug-in features some interesting new options to play with, such as a stereo widener and an ‘Interpolate’ feature...”

Radio France’s unusual reporter’s mic being modelled by the Slate Digital team.

you can of course duplicate the track and use a second instance of the plug‑in).

Mics For Days!

I have focused on the new Radio France pack here, but do check out my review from January this year where I covered the Blackbird, Strongroom and Classic 87 packs that now come included with the All Access Pass subscription: www.soundonsound.com/reviews/slatedigital-vms. At the time of writing, that bundle will give you access to no fewer than 36 mic models.

Whilst not containing many especially rare or famous mics, the Radio France pack adds a good amount of variety, and it seems a sensible call from the VMS team to not go down the path of modellng more high‑end studio mics from Neumann and the like, as these are already well covered in other packs. The inclusion of more dynamics and ribbons is very

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welcome, and I’d like to see even more obvious character and effect options in future packs.

Final Thoughts The new ML-1A seems a solid upgrade on the previous version, but perhaps not to the extent that people who recently bought the V1 should feel let down! The big plus for me with this new release is the dedicated VMS software, which will benefit users both new and old. When feeling my way around a new piece of software, it’s quite rare for every guess I make in terms of trying to access a feature to turn out to be right first time, but that was definitely the case here.

There are a lot of mics to choose from now too, and it’s important that the software allows you to quickly audition the options intuitively. The extra processing features are also worthwhile. I found myself tweaking the Proximity and

Intensity controls the most when mixing, and found both to be very handy indeed for fine-tuning mics in the context of a mix. I also found the Interpolate option useful on occasion; as described earlier, it proved quite effective at giving darkersounding mics a bit more edge, and vice versa. I’m not a fan of stereo widening myself, but I know many engineers who are, and they will enjoy having that option included. The Radio France pack is a nice extra feature for this new version 2 launch, and it offers engineers like me the chance to try out a few mics we haven’t used before as well as offering a different ‘feel’ from the other mic packs.

You can hear some audio examples of the Slate Digital VMS 2 in action at: https://sosm.ag/slate-digital-vms2 £ £478.80 including VAT. W www.slatedigital.com

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ON TE ST

Mono Valve Compressor

LANG Electronics P.LANE Type 436 Through the ’50s and ’60s, Altec’s 436 compressors were often modified and hugely influential. But this keenly priced homage does more than clone these vintage devices.

BOB THOMAS

T

he LANG Electronics name is now owned by Heritage Audio, and with the P.LANE Type 436 variable-mu valve compressor they’ve moved the brand’s focus from New York to a “London studio” where a “particularly talented band from Liverpool” recorded during the 1960s. It doesn’t take Sherlock Holmes to figure out the reference to EMI’s recording studios on Abbey Road, and omnipresent in those studios during the 1960s were a number of Altec Model 436A & 436B valve-based variable-mu compressors, imported from the USA in 1958 and modified in-house by the EMI engineers. Heritage Audio’s engineers have taken their inspiration from a slightly different variant, the Altec 436C, which was developed in the early 1960s by Altec themselves, in response to the widespread practice (on both sides of the Atlantic) of modifying their earlier units.

Vibe & Valves Aesthetically, the P.LANE 436 nails the 1950s vibe of the vintage units, with a 2U grey fascia, chicken-head knobs, circular analogue meter and red jewel power indicator lamp. To the far left of that meter, which displays up to 30dB of

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gain reduction, sits a detented, passive rotary input level attenuator, followed by a smaller vertical pairing of Attack and Recovery (release) selection switches.

Each gives access to six settings, but the Recovery switch has an additional five Hold positions too, of which more later.

To the right of the VU meter another vertical control column consists of a six-position DC Threshold control and two screwdriver-adjustable trim pots: one called Zero, to set the meter’s 0dB point, and the other Balance, to rebalance the triodes of the P.LANE 436’s variable-mu valve, to minimise any low-frequency artefacts that may arise as that valve ages. Sitting next to these you’ll find the detented, passive rotary Output Attenuation control, and this is followed by an on/off toggle switch and the jewel lamp. At the rear are the expected XLRs and the less expected parallel 6.35mm jack sockets carrying the balanced audio I/O, a 220/110 V mains voltage selector, and a fused IEC mains socket.

As with all of the Heritage Audio products that I’ve reviewed, the PCB layout, its components (all full-size, through-hole) and the overall construction are of exemplary quality. A compact toroidal mains transformer is bolted vertically to the right-hand end of the

chassis. The main PCB carries the analogue power supply, side-chain and audio circuitry, together with the P.LANE 436’s three valves. The Heritage Audio‑designed, LANG-branded input transformer features a high-nickel alloy core and a mu-metal shielding can, and sits on a small board at the left-hand end of the chassis. The output transformer, another in-house design this time equipped with an M6 silicon steel core, to help create colour and harmonic distortion at the output stage, is bolted down behind the main PCB next to a large heatsink.

The increasing lack of availability of quantities of reasonably priced, high-quality NOS (new old stock) vintage valves in both the short and long term is a growing challenge, for both equipment manufacturers and guitarists like myself.

Heritage Audio, though, seem adept at overcoming this challenge, as illustrated, for example, by the substitution of 16

LANG Electronics P.LANE Type 436 €999 pros • Superb overall performance.

• Can control dynamics without supressing punch and high frequencies.

• Recovery Hold function eliminates compression release artefacts.

• Excellent value for money.

cons • Assuming you enjoy the vintage vibe, none.

summary An updated, attractively priced take on a 1960s variable-mu compressor that brings a legendary heritage and an iconic character and performance into today’s home and project studios.

6BA6s for eight 6386s in their Herchild 670 compressor. Such availability issues required the substitution not only of the 6BC8 variable-mu valve from the 436C’s gain cell with a NOS Russian-made 6H5P (which Heritage Audio’s Peter Rodriguez describes as “performing like a premium selected 6BC8”), but also of the original’s output stage double-triode 6CG7 by a JJ-manufactured E88CC. Known as a 6922 in the US, the E88CC is an upgraded version of the standard ECC88/6DJ8 that was originally developed by Mullard in the UK, in part for use in computers. The only valve from the 436C that didn’t require substitution was the 6AL5 dual‑diode in the side-chain, which generates the compression control signal and remains readily available as NOS.

In My Ears Gain staging with the P.LANE 436 requires some consideration. The input and output level controls are passive attenuators and, counter-intuitive though it may seem, these equip the P.LANE

436 with a kind of automatic make-up gain. To obtain greater amounts of gain reduction, you have to reduce the input attenuation, thus allowing more signal level into the compressor. This increased input signal (assisted, no doubt, by the 20dB of gain through the P.LANE 436 when no compression is taking place) helps to compensate for level lost due to compression. Finally, the output attenuator allows you to set the level leaving the unit. Since both attenuators sit after their respective transformers, with maximum input and output levels of approximately +24dBu at less than 1% THD, you can drop the hammer on both input and output transformers and still be able to avoid overloading the next link in the signal chain.

The attack times are given as “15ms to much slower” and increase linearly over the six switch positions. When it comes to recovery times, the picture is slightly more complex. These are given as 300ms-6sec, increasing in a linear fashion from switch positions 2-6. Position 1 on the Recovery dial (designated as

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SF or SuperFast) is the outlier, as this sets a recovery time of 150ms, a speed unknown in the Altec 436C. When the Recovery switch is moved to a Hold (red dot) position, the current level of compression is frozen until either another recovery time is selected or the frozen level is increased by a higher-level transient. This function allows you to prevent either unnatural initial compression at the start of a song or the rapid increase in level when compressed audio drops suddenly below the compression threshold, either during or at the end of the track. In the case of the former, having selected the recovery time for the track, you’d play through the start of the track, switch to the nearest hold position to freeze the resulting compression level, play the track from the start, and unfreeze the compression once the track was under way. In the latter case, you’d simply switch to an adjacent hold position before the audio dropped in level, and either switch compression back in when the quiet passage finished, or leave it frozen during the fadeout.

Ag

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L A N G E L E C T R O N I C S P. L A N E T Y P E 4 3 6

The XLR audio I/O are duplicated on quarter-inch jacks to make connection convenient.

Rather than simply adjusting the threshold, moving through the P.LANE 436’s DC Threshold control’s six switch positions also sets the compression slope, moving progressively from pronounced to gentle compression.

Attack and recovery timings interact with the DC Threshold to create the compression curve, and it is that three-control interaction that enables you intuitively to sculpt the style, character and amount of compression that’s best suited to your source signal.

Pleasure To Know With few visual references — only the meter indicating gain reduction, and the positions (not actual values) of the controls — getting the best out of the P.LANE 436 requires experimentation, but I found that this led quite quickly to an intuitive understanding of the interaction and relationship between the controls. To my way of thinking, there are two fundamental relationships within the P.LANE 436: one between the input level and the DC Threshold; and the other between the Attack and Recovery times. The former I think of as defining the character and amount of compression, and the latter I regard as creating the style of compression. These two relationships are interdependent, so I tend to adjust their control pairings in turn and, having got the compression heading the way that I want it to, I’ll start adjusting individual controls in order to zero in on my goal.

Unlike, say, a VCA or FET compressor, with the P.LANE 436 there’s virtually no

ALTERNATIVES The Chandler RS-124, developed in conjunction with Abbey Road Studios, is the way to go if you crave an authentic reproduction. Retro Instruments’ Revolver and Highland Dynamics’ BG2 might offer less authenticity, but no less validity.

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such thing as too much compression.

I found that there’s either too little or just enough — even if that “just enough” is keeping the compression level at its 20dB maximum. For many sources, vocals and acoustic instruments for example, gain reduction in the 10-15 dB range on the meter seems to me to be the P.LANE 436’s sweetest spot. Having said that, it also sounds great shaving just a few dB off a bus to pull everything together.

The P.LANE 436 has a smooth sonic character that’s no doubt in part due to that steel-cored output transformer, but it also subtly enhances the signal passing through it. Add in a compression that

“With the P.LANE 436 there’s virtually no such thing as too much compression.” has the ability to exert control without destroying high frequencies, crushing the life out of a sound, or sacrificing its initial attack and punch, and the result is a compressor that, to my ears, can pull a mono bus or source into aural focus — and most of the time it does so without it being too obvious. Even the SuperFast recovery setting doesn’t break this paradigm, as its effect is to push the source somewhat more to the fore — particularly, though not exclusively, in combination with the 15ms attack time. This makes those combinations ideal partners for drums, percussion and rhythmically complex guitar strums. The one circumstance where the P.LANE 436 can sound more muffled than I’d like it to is when, irrespective of the attack time, it’s delivering 20dB of compression at the six-second recovery time, but I doubt that I’d ever go there ‘in anger’.

Lastly, a special mention must go to the Recovery time Hold function, which stops noise rising during quiet passages as a track or source is being dynamically controlled. This was my first encounter

with this feature and it worked brilliantly, though there are some practical steps that you have to take to get the most from it: when tracking, rehearsal and a cue list are a must and, when mixing, a stopwatch and an old-school time-referenced cue list definitely help.

Verdict I have a real soft spot for variable-mu compressors. I find that their slow attack and long release times lead to a comparatively unobtrusive and relaxed performance that, for me, works particularly well on vocals, acoustic guitar and bass, electric bass guitar, small drum kits and percussion. The P.LANE Type 436 may be inspired by the EMI-modified Altecs but, really, it is best thought of simply as being a well thought-out compressor that puts a legendary heritage and an iconic sonic character within the financial reach of today’s home and project studios.

If you can’t live without a hard bypass switch, side-chain filters, wet/ dry mixing and a stereo link, you’ll have to look elsewhere. But if you can leave those behind, you’ll be rewarded with a compression style that’s seductive and artistically inspiring. Although its slow attack and release may prevent it being your only compressor, the P.LANE 436’s characterful performance, sound quality, control flexibility and attractive price deliver a value-for-money proposition that you’ll find hard to equal. An original EMI-modified Altec 436B sold on eBay in 2012 for £22,322, and that, in my view, makes the Lang Electronics P.LANE Type 436 an outstanding bargain that really should be auditioned by anyone considering adding a variable‑mu compressor to their arsenal. £ €999 including VAT. T Heritage Audio +34 917 266 189 E support@heritageaudio.com W https://heritageaudio.com

ON TE ST

SAM INGLIS

W

hen a plug-in developer adopts the name iamReverb Audio, it suggests they are pretty serious about reverb. No surprise, then, that the Vienna-based company’s first product is an eponymous plug-in dedicated to providing, you guessed it, reverberation; and what’s more, they claim a significant technical breakthrough.

iamReverb Audio iamReverb Convolution Reverb Plug-in

Getting Convolved About two decades ago, convolution was heralded as the technology that would revolutionise mixing, if only we could find computers fast enough to do it. Today, it’s a process that laptops can do in their sleep, and impulse responses are everywhere from amp simulators to virtual tape machines. But in terms of our use of reverb, it’s not obvious that it has revolutionised mixing. The same increases in computer power that have made real-time convolution trivial to perform have also enabled algorithmic reverbs to overcome many of their previous limitations, whilst the limitations of basic convolution were soon exposed.

One obvious limitation of a basic convolution plug-in is that it’s hard to edit or alter the sound. A well‑recorded impulse response captured in a nice-sounding space allows the character of that space to be faithfully added to any source, but engineers who have grown up with algorithmic reverbs missed the ability to vary the duration, tone, width and other qualities of the reverberation. Another limitation is that simple convolution is a static process that cannot capture

iamReverb Audio iamReverb £243 pros • High-quality factory library in which all the sounds are very usable.

• Presets are endlessly malleable using just a few simple controls.

• Friendly and straightforward user interface.

• There’s a sense of presence and immediacy to the sound that is not found in many convolution reverbs.

cons • No plate, spring or digital reverbs.

summary It’s simple to use, surprisingly versatile and sounds great. What more could you want from a reverb plug-in?

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IamReverb say that convolution reverbs share an inherent flaw — which only their novel IR recording technique can overcome. time-variant qualities of the reverberation.

The modulation options provided in algorithmic reverbs might not replicate these perfectly, but they are usually good enough, and are easy to adjust.

I’ve reviewed several plug-ins that aim to overcome these limitations, including Liquidsonics’ excellent Reverberate, HOFA’s highly versatile IQ-Reverb, Acustica’s novel Silver and Inspired Acoustics’ epic Inspirata. All of them, in their own way, make convolution more flexible, better sounding and ultimately more useful. But, according to iamReverb, a core problem with existing convolution techniques has remained unsolved until now.

Early Doors In a convolution reverb, the incoming audio signal is ‘convolved’ with a short audio file known as an impulse response. As the name suggests, this impulse response is a recording of the way in which an acoustic space (or, indeed, any other system) reacts when an impulse is fed into it. But what is an impulse? In theory, it’s a burst

of sound with zero duration, or at least, with a duration less than a single sample.

However, no real-world sound behaves like this, so the would-be convolution reverb maker has a choice. He or she can use something that is a fairly good approximation of an impulse, such as a balloon bursting or a shot from a starter pistol; alternatively, a sine sweep can be played back over a PA system, and the impulse response derived mathematically from the resulting recording.

Both techniques suffer from the limitation that iamReverb have identified, but it’s perhaps more easily understood in relation to the first. Even though a gunshot is a very brief burst of sound, it’s not a perfect impulse, and thus does not die away instantly. Consequently, if the mics capturing the impulse response are positioned in such a way that the first reflection paths are only marginally longer than the direct path, it becomes impossible to fully disentangle those first reflections from the direct sound. In other words, there is an inherent limit to the accuracy with which conventional convolution techniques

Photo: Bernhard Raab

can reproduce early reflections. Yet, as the makers of algorithmic reverbs figured out decades ago, those first reflections provide vital aural cues about the space we’re in.

The key innovation that iamReverb claim, then, is a new technique for capturing impulse responses that avoids this problem and faithfully represents early reflections even when the mics are placed very close to the source.

As this is a proprietary technique, details are secret, but they have at least given it a snappy name: Enhanced Reverb Modelling, or ERM. The company say that ERM makes it possible to recreate the effect of hearing a nearby source in a reverberant space with a realism that wasn’t previously possible.

The iamReverb team have developed a new technique for capturing impulse responses which, they say, allows early reflections to be faithfully represented for the first time.

Three Steps To Heaven The iamReverb plug-in is available for macOS and Windows in all major native formats, and is authorised using the iLok system. The 300MB-ish library of impulse responses seems to be installed separately for each format, and is invisible to the user; you can’t import or record your own IRs. The plug-in interface is cheerful and refreshingly bright. To my eyes, it draws on the same design language as the Teenage Engineering OP-1 and some Baby Audio plug-ins, which is no bad thing. And, like those Baby Audio plug-ins, it presents a compact and very manageable set of parameters. There are just two editing pages, one for actual reverb parameters and one for setting up a simple but effective four-band EQ. What’s perhaps most striking about the main editing page is that most of the controls on offer are parameters I associate with algorithmic rather than convolution reverb. For example, you can freely vary both the decay time and the size of the space. There’s also variable pre-delay, whilst the central part of the window contains three dials labelled Close, Mid and Far. On a Lexicon-style reverb, you’d expect these to let you balance the level of early reflections against that of the tail.

They do something similar here, setting the relative levels of three different, complementary IR captures made at different distances from the source.

A preset within iamReverb is thus much more than a static representation of a position within a space, which is what you usually get with an impulse response. Instead, it’s a starting point that you can fine-tune in much the same way as is possible in a typical algorithmic

“The key innovation that iamReverb claim is a new technique for capturing impulse responses that faithfully represents early reflections even when the mics are placed very close to the source.” plug-in. The factory library includes over 100 presets, of which perhaps half are named after a type of space (Small Church, Long Chamber, and so on), and the other half after their suggested application (Acoustic Guitar Hall, Drum Chamber, and the like). All are derived from real acoustic spaces, and in fact the entire library is based on different captures of around 30 rooms. However, there are no plates,

springs or vintage digital units, which means you can’t get the full gamut of useful reverb treatments from iamReverb.

It’s also strictly a stereo-only plug-in, with no surround options.

Impulse Purchase?

From a technical point of view, iamReverb’s criticism of existing convolution reverbs makes sense. However, I can’t say that I’ve

www.soundonsound.com / June 2025

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IAMREVERB AUDIO IAMREVERB

Photo: Bernhard Raab

The impulse recording process used a Dutch & Dutch 8C loudspeaker, captured using a variety of microphones and recorded through a Forssell SMP-2 preamp.

previously noticed anything audibly amiss with the early reflections in other plug-ins, so I was interested to discover whether this one would provide some sort of epiphany.

Since no two convolution plug-ins sample the same spaces from the same positions with the same equipment, there’s no useful sense in which you can do an A/B comparison between them.

Consequently, I can’t be certain whether iamReverb’s good qualities are down to ERM, or whether they’re simply the result of skilful IR capture and plug-in design in other respects. But what I can tell you is that this is a very fine reverb plug-in indeed. Having now used iamReverb pretty much exclusively on a number of mixes, two things stand out above all.

The first is a liveliness to the reverberation that I’ve not encountered in other plug-ins. It’s not that the sound is brighter, or that it’s compressed; more that it simply feels more present and immediate, even for large spaces. I don’t know whether this is a consequence of having early reflections more accurately represented, and it’s a subtle rather than a night-and-day difference, but it’s something that I consistently noticed across the entire library. There are no controls for adding or adjusting modulation to the sound, but I never felt they were necessary.

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The second is that this is a reverb plug-in made by people who actually use reverb plug-ins, and use them for mixing music. The philosophy here is demanding in terms of quality, but not precious in terms of allowing the sound to be manipulated. For example, during the review period, iamReverb issued an update that added a switch to reverse the panning of the Far component. This is hardly ‘realistic’ in the purist sense, but it’s useful and effective in situations where you don’t want the reverb tail to be lost behind the dry sound in the stereo field.

The plug-in’s second page is a simple four-band EQ.

As I’ve already mentioned, that library contains around 100 presets, which might not sound all that much. But there’s an unusually high hit rate here, and none of them fall into the “interesting, but”, or the “useful for post-production” categories.

There are no parking garages, shipping containers or kitchen cupboards.

Everything that’s included here has an obvious and valuable real-world application, especially if the music you’re mixing is based around real instruments rather than synthesized sounds.

Better still, every preset has a wide and useful range of variation. The Decay and Size parameters permit quite broad changes without making things sound artificial or unnatural, and the concept of balancing Close, Mid and Far mic positions is simple and highly effective. I was initially wondering why there’s no preset category for ambiences, given the emphasis on early reflections capture, but in fact there is no real need for one. You can create excellent ambiences from a number of the existing presets, simply by reducing the decay time or by emphasising the Close position. If it wasn’t for the realism of the sound and the fact that turning the Decay and Size controls briefly mutes the output, you’d think you were using an algorithmic reverb.

In short, iamReverb is a most impressive debut product. It sounds great, the user interface is faultless, and above all, it’s extremely usable. I just hope iamReverb Audio haven’t boxed themselves into a corner with their choice of company name, because I’m keen to hear what they can do in other fields! £ £243 including VAT. W www.iamreverb.com

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

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

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FE ATURE

Oberheim Two Voice ALEX BALL

T

he Oberheim Two Voice (or TVS-1) is one of Tom Oberheim’s favourite creations from his impressive back catalogue. Born out of a humble satellite unit designed by an all-star cast of engineers, the Two Voice was the beginning of a line of synthesizers that have made an indelible mark in contemporary music.

An Engineer Finds Music A Kansas native with an interest in electronics, Oberheim moved to LA with friends in the 1950s. As well as experiencing printed circuit boards for the first time whilst working for a camera shop, he also took a role as a draftsman for the electronics division of National Cash Register,

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who were one of the earliest companies to build computers.

Realising he needed to study further to get where he wanted to be in an ever changing technological world, Oberheim returned to education, studying Physics at UCLA in his 20s. This, fatefully, included a side course in art and music. Developing an appreciation for music through his studies, Oberheim started singing in choirs, including the UCLA choir. This moved him into circles of musicians and one day, one such musician asked Oberheim if he could use his skills to build him a ring modulator. Not really knowing what a ring modulator even was, Oberheim found an article written by Harald Bode, famously a collaborator of Robert Moog, and successfully managed to build a working device in his apartment in Santa Monica.

If any synthesizer can be considered to be greater than the sum of its parts, it’s Tom Oberheim’s enduring classic, the Two Voice.

Word of this must have gotten around as Oberheim then received a phone call from composer Leonard Rosenman, asking if he could bring a ring modulator down for use on a film he was scoring, which turned out to be Beneath The Planet Of The Apes. Whilst the ring modulator was only used on some piano parts, musicians in the orchestra were intrigued by the device and asked if he could sell them one so, still based in his apartment, he went into business.

Oberheim then got a call from Chicago Music Instruments (CMI), who’d also heard of his ring modulator and wanted to add it to their Maestro line of effects. And so it became the Maestro RM-1, which is why both Oberheim and Maestro badged versions of the same effect can be found.

Spurred on, Oberheim then decided to try to build a pedal that emulated tape

The Two Voice recipe is simple: take two Synthesizer Expander Modules...

flange, but came across an example circuit for phase‑shifting using op-amps and so built that instead. Again, Maestro added it to their line and it became the Maestro PS-1A, which sold a staggering 70,000 units in the 1970s, putting Oberheim well and truly on the map.

Pulling on his earlier experience working with computers, Oberheim then designed one of the earliest digital synthesizer sequencers on the market in 1973. The DS-2 stored control voltages in logic gates and could then play them back to reproduce the programmed sequence of notes. Unlike the first microprocessor‑controlled sequencers, such as the Roland MC-8, there was no ability to copy and paste or sequence multiple tracks, but the possibility of storing 144 notes in a world of 16‑step sequencers was a big deal for the time.

For all its exciting potential, however, the DS-2 created a problem: if a synthesist hooked up their synthesizer to it to play back a sequence, then what would the synthesist play? This realisation lead Oberheim to begin work on a satellite synthesizer unit that could be used with the DS-2 whilst the synthesist played along on the keyboard of their main synth. For this idea to sell, the satellite unit needed to be as affordable as possible, but also have the minimum required to make it usable.

The resulting design was dubbed the Synthesizer Expander Module, or SEM

for short. It contained two oscillators, a state-variable filter, an amplifier, two envelopes and an LFO. Deliberately designed without a keyboard, the rear of the unit housed the jack points to connect it up to the DS-2.

Collaboration & Experimentation In search of quality VCOs for the SEM concept, Oberheim approached E-mu Systems, who were already deeply involved with innovations in both analogue and digital synthesizer design. Dave Rossum of E-mu had previously cooked up some discrete VCOs for E-mu’s own modular synthesizer and so, in turn, he designed discrete VCOs for the SEM.

E-mu’s involvement didn’t stop there, however. Understanding the potential of digital technology, Rossum had been working on polyphonic keyboard designs, culminating in a microprocessor‑controlled scanning keyboard, possibly the first of its kind.

This would prove fateful for Oberheim and the SEM, as we’ll discuss later.

Next on the shopping list was a filter.

Oberheim already had connections with ARP Instruments as he’d approached co-founder David Friend at a trade show and offered to be their West Coast dealer, which they’d take him up on as he had a good list of movie composers in his contact list. One of ARP’s (then Tonus’) first engineers was Dennis P Colin.

A talented designer, Colin had a particular

A Life Of Its Own Out in the world, SEM synths have appeared in many famous recordings and live performances from Stevie Wonder, Weather Report, Chick Corea, Herbie Hancock, John Carpenter, George Duke, Rush, Vangelis, Vince Clarke, the Prodigy, 808 State and Edgar Froese, amongst others, but the TVS-1 has some specific appearances that are worth mentioning. Firstly, Mark Mothersbaugh used his unit for the sprinkling of electronic sounds and even entire electronic cues for the score to The Life Aquatic With Steve Zissou in 2004.

Meanwhile, on the other side of the Atlantic, Geoff Barrow and Ben Salisbury were originally hired to compose the music for the 2012 film Dredd and took the approach of a raw, minimal, dystopian synth score. They were subsequently replaced on the job by Paul Leonard-Morgan, but were thankfully able to release their work as an album titled Drokk: Music Inspired by Mega-City One. The unused score features heavy use of three original TVS-1 units owned by the duo.

One other use worth mentioning is that of Michael Stein and Kyle Dixon. After much speculation as to which synth provided the distinctive arpeggio in their theme for Stranger Things, the band revealed that it was their Oberheim TVS-1 and, furthermore, that the arpeggio was played by hand, rather than with a sequencer or arpeggiator.

www.soundonsound.com / June 2025

53

FE ATURE

OBERHEIM TWO VOICE REVISITED

...add an eight‑step mini-sequencer...

skill with filters and had created the 1047 Multimode Filter/Resonator for the ARP 2500. He’d written a paper on the science behind his design and Oberheim cut a deal and licensed use of the filter from him, which again proved fateful as it helped define the distinctive and desirable Oberheim sound.

Back at base, Oberheim had brought in Jim Cooper as a staff engineer, and he worked with him until the early ’80s when he left to form his own company and also become head of the MIDI Manufacturers Association. Cooper provided additional circuit designs for the SEM, including the triangle wave LFO. With his main circuits in place, Oberheim set to work tweaking the design, changing component values and experimenting with how the electronics affected the sound. After some time he found the point where the SEM sounded as good as he could make it and from then on, SEMs were always made to that

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spec. Time has shown that his instincts were spot on, as the SEM is amongst the most emulated and copied analogue designs of all time.

Wait A Second...

The SEM went out into the world to pair with the DS-2 as intended, but Oberheim had already put two and two together and realised that E-mu’s digital keyboard could open the door for something much more ambitious. Since a polyphonic keyboard can send out discrete control voltage and gate signals, these could be connected to multiple SEM modules to create a polyphonic synthesizer! Oberheim had already had experience of a crude version of this, as he had modded ARP 2600s so that they could be played duophonically, albeit with the oscillators sharing a signal path. This was the opportunity to take that to a whole other level.

I had the great pleasure of speaking to Dave Rossum at Machina Bristronica

in 2024 and in a conversation about this time he told me that Oberheim actually paid to get the patent sorted for the keyboard to speed up the production, and then licensed it back to E-mu.

Armed with all the components, the end results were the Four Voice (FVS-1) and the almighty Eight Voice (EVS-1), the former being the first commercially available polyphonic synthesizer. But before either of those came the humble Two Voice (TVS-1) in 1975.

Housed in a case containing two SEMs and the polyphonic keyboard, the Two Voice allowed the player to play both voices in unison or play two notes on the keyboard, and to choose the order of SEM that they were directed to with a ‘left voice first’ or ‘right voice first’ toggle switch. As a bonus, a two-channel, eight-step mini-sequencer was housed in the left of the instrument, with the dual‑concentric knobs providing parallel control voltages. Much like the DS-2/SEM

Don Lewis & LEO In the early ’70s, American musician Don Lewis (who’d already caught the attention of Roland’s future founder, Ikutaro Kakehashi, by modifying his Ace Tone Rhythm Aces) hatched an ambitious plan to make a synthesizer ensemble that could be controlled by a single person. The result was called the Live Electronic Orchestra, or LEO for short. Whilst Lewis and the LEO would prove inspirational for the later development of MIDI, there’s an interesting Oberheim connection entwined within. At around the same time that Tom Oberheim was working with E-mu Systems to

concept, one SEM could be sequenced while the synthesist played the other from the keyboard, or as a bonus, sequences could be transposed from the keyboard, and/or the second sequencer channel could modulate the sequencer clock if the clock knob was pulled up.

One of the defining characteristics of the SEM is the 2-pole state‑variable filter. As well as offering phasey notch filtering, vocal‑esque band-pass filtering and fizzing high-pass filtering, along with classic low-pass, the SEM filter sounds better and better the more resonance is added. This is counter to numerous analogue filter designs that lose bass and become whistley as the resonance is cranked. Maxed out, the SEM filter still sounds superb and this was the basis of one of the most famous synth sounds of all time, the filter sweep on Rush’s ‘Tom Sawyer’. Whilst this was performed on the later Oberheim OB-X, the filter was still the same SEM design and the sound is possible on the Two, Four and Eight Voice instruments with differing levels of unison stack.

pair up his SEMs with their polyphonic keyboard, Lewis had worked with engineer Richard Bates, but also Armand Pascetta, who had also created a digital, polyphonic keyboard. One of the parts of LEO was a bank of four Oberheim SEMs that were controlled by Pascetta’s keyboard, meaning that the Oberheim Four Voice was, in essence, independently developed twice at the same moment in history! Lewis’ homebrew Four Voice can be heard on Sergio Mendes’ Homecooking and on the Brothers Johnson’s Look Out For #1, both recorded in 1976.

a limited proto-memory found in the Four and Eight Voice models that could store offset CV values for some parameters.

Understandably, sales of Oberheim’s synths went immediately south following the announcement of the P5 and to keep up, SEM synths were abandoned and replaced with a new line of polyphonic

demand for discontinued Oberheims started to rise and with that went the prices. With relatively few units made, the appetite for the ’70s Oberheim sound has lead to all manner of clones, ‘inspired bys’, sample packs and software emulations.

Having spent time away from the industry, Tom Oberheim began remaking SEMs in the 21st Century. This lead on to two new versions of the Two Voice, produced on a limited basis: firstly the Two Voice Pro in 2015, and secondly the Special Edition in 2021. Both units kept the main features of the original, but included updates and improvements for the modern world. The latter was also the first synth to reuse the Oberheim brand after it had been returned to Tom Oberheim.

“Time has shown that his instincts were spot on, as the SEM is amongst the most emulated and copied analogue designs of all time.” synthesizers, the OB-X, OB-Xa and OB-8, which are a story unto themselves.

The SEM and its polyphonic pals were overnight dinosaurs and could have been lost to the 1970s as a relic of how things were achieved at that point in history.

But instead, as the decades drifted on,

Audio examples of the Oberheim Two Voice can be found at https://sosm.ag/OberheimTwoVoice

Short But Sweet In 1978, Dave Smith of Sequential Circuits famously flipped the synthesizer world on its head when he unveiled the Prophet‑5. Again, involving much technology from E-mu Systems, the Prophet had a single set of front‑panel controls changing the parameters of all of the voices simultaneously, plus there was the massive breakthrough of proper patch storage. This was the opposite of Oberheim’s SEM-based synthesizers, which had a full set of controls that needed to be manually set up for every voice, which equated to 160 knobs on the Eight Voice, just to change from one sound to another! Also, the only thing that resembled patch memory was

...and an E-mu keyboard and you’re done!

www.soundonsound.com / June 2025

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ON TE ST

Harrison Mixbus 11

Harrison’s distinctive console-inspired DAW is now more usable — and more affordable DAW Software For macOS, Windows & Linux — than ever.

Harrison Mixbus 11 £40/£120 pros • User interface overhaul provides some welcome workflow refinements.

• Revised pricing structure will make Mixbus very tempting to new users.

cons • As a Mac user, I’m now envious of the Windows/Linux touchscreen support!

summary Mixbus 11 brings some welcome user interface improvements for existing users, and dramatically reduces the cost of entry for potential new users.

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JOHN WALDEN

H

arrison’s Mixbus DAW celebrated its 15th anniversary last year with the milestone introduction of v10, as reviewed in the October 2024 issue of SOS. Yet such is the pace of development that, barely six months later, Mixbus 11 is already upon us.

This release sees a simplification down to two product tiers — Mixbus and Mixbus Pro — and some very tempting price reductions, which are explained in the box. Of course, both feature all the core functionality that made Mixbus Pro

10 a powerful platform for recording and editing audio and MIDI, mixing with a very console-style workflow, composing to picture and triggering loops and samples, whilst Mixbus Pro includes Dolby Atmos support. Previous SOS reviews will have you covered on that front, so let’s focus on the refinements in this new release that might tempt existing users to upgrade or encourage new users into the Mixbus fold.

U&I Mixbus 11 brings numerous UI refinements intended to make navigating core functions easier and more efficient. For example, on

Mixbus is designed to replicate the experience of using a large-format console, and its fully expanded Mixer view can be impressive!

the right side of the toolbar, the buttons used to toggle between Mixbus’ four main working environments — Cue, Record, Edit and Mix — are now larger and much more obvious. To the left of these, you now also get dedicated buttons for easy access to the Audio Setup and Preferences dialogues, and to the right, you’ll find new ‘Edges’ options, allowing you to toggle open various properties panels within each of the four main screens. Keyboard shortcuts are also available.

Depending on which of the main screens you’re in, these properties panes can open at the left, right or bottom of the window, with their own sets of options. On the Cue, Edit and Mixer screens, perhaps the most interesting of these is the ability to display the Channel Strip contents for the currently selected channel. The fully expanded Mixbus Mixer, with its core Gate, EQ, Comp and Send panels, makes the program’s console-based design ethos very apparent, but occupies a large amount of screen real estate. The new lower panel gives you a clear view of all the key Channel Strip controls, albeit for one channel at a time, even on a modest laptop-sized display.

These are laid out horizontally, with the signal chain running left to right. On the Mix page, this lower edge display also includes generic control panels for any plug-ins you have inserted on the selected channel, allowing you to tweak some key controls without always needing to open separate plug-in windows.

The transport toolbar has also been refined. The key controls are now larger,

Value Engineering With Mixbus 11, the range has been simplified to leave only two tiers: Mixbus and Mixbus Pro. In effect, the ‘Plus’ label has been retired, so what was previously Mixbus Plus is now simply called Mixbus. And, yes, that means all the features that were in the Plus version are now found in the introductory Mixbus tier, including the Gate module within the Channel Strip and the capable collection of 19 XT plug-ins. Mixbus 11 also sees a change in approach to the activation system, with a new machine-based ‘token’ system supporting two installs for Mixbus and three for Mixbus Pro. Perhaps even more interesting are the pricing changes that accompany the line-up adjustment. Previously, Mixbus Plus and Pro cost £274.99 and £399.99 respectively, although I would imagine most buyers took advantage of Harrison’s regular promotional offers to pay significantly less. With v11, the full prices have been slashed so that Mixbus 11 is now just £39.99 and Mixbus Pro 11 is down to £119.99. Existing v10 customers will also receive offers with discounted upgrade pricing options. Whichever tier you are now considering, the bang-for-buck ratio has become considerably more appealing. This change makes Mixbus much more accessible, and much more attractive, to potential users.

and some lesser-used options are hidden by default; the Preferences panel lets you customise what’s shown. If you open multiple main page displays as detached windows, each of these now gets a full transport toolbar. You can, of course, also open a dedicated (and resizable) floating Transport Controls panel containing key controls, should you wish.

As well as MIDI clips for editing, the new lower edge panel also provides a compact view of the Channel Strip and plug-in controls for the currently selected channel.

www.soundonsound.com / June 2025

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ON TE ST

H A R RIS ON MI X BU S 11

Other UI tweaks include an improved Locator Bar, new buttons within the Master Bus for easily managing visibility of busses, VCAs and the Monitor Section, navigation improvements within the right-hand sidebars, easier options to enable/disable the EQ, Comp and Gate sections within the Mixer, and a new Comments section within the Rec/Input panel for note-taking.

MIDI Made Easier Mixbus 11 also brings some very welcome additional MIDI editing options. For example, in addition to inline editing within the Edit page, if you have the lower-edge properties panel open, the contents of any selected MIDI clip are automatically displayed there, where they can be edited. Even better, if you double-click on a MIDI clip with the Grab tool selected, you can open a MIDI region in its own window (this action is also possible from the right-edge panel’s Regions list). As this is then fully resizable, including sub-panels for editing velocity, pitch-bend and other parameters, it provides a more adaptable environment for complex note or parameter editing.

It’s also worth noting that selecting (or right-clicking) on an audio region in the same fashion will display that region’s properties within the lower-edge panel, or in a detached window. This includes control

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panels for any of the region-specific Region Effects introduced in v10.2, for speedy access and editing.

Touchy Feely As I did my testing under macOS, perhaps the most notable of the other refinements is something that I can speak of only from a point of envy: Windows and Linux users get improved support for touchscreens, including multi‑touch.

Now, your mileage may vary, and may well depend upon how well you gel with any of the more upmarket DAW controllers that are currently available, but, having used Mixbus 11 on a large 4K display, the fully expanded Mix page, with all its core sections open, is a rather magnificent sight. A large-format 4K monitor with touch capability will cost you upwards of £1500, but I would love to experience the hands-on control such a screen would offer! Obviously, it would not be the same as laying your hands on an actual Harrison hardware console, but the idea of having 30-plus channels’ worth of faders, knobs and buttons laid out in front of you and all responding to as many fingers as you can muster undoubtedly has its appeal.

Progress Report Whereas Mixbus 10 introduced huge features such as Dolby Atmos

MIDI clips can now be easily popped open in a dedicated window for detailed editing work.

support and the SSL 9000J EQ module emulation, I think it is fair to say that the updates to Mixbus 11 are more evolutionary than revolutionary.

That’s not to say that they are not welcome, though. There is great value in improving the user experience and providing new ways to access and employ core tools. Mixbus 11 provides significant steps forward on that front that many existing users will appreciate.

For potential new users, the change to the pricing structure will undoubtedly be appealing and, presumably, the change reflects a desire to broaden Mixbus’ user base.

Mixbus is a DAW with a very particular character, which deliberately recalls the experience of working with a high-end, large-format mixing console. That might not suit every type of music producer but, if it’s a concept that you do find enticing, at under £40 Mixbus 11 — with all the features previously found in the now-retired Plus edition — is a bit of a steal. £ Mixbus 11 Pro £119.99,

Mixbus 11 (previously Mixbus Plus) £39.99.

W www.harrisonaudio.com

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ON TE ST

Toontrack Fusion

Software Collection

If you’re an EZ user with a hankering for fusion, Toontrack have got you covered.

JOHN WALDEN

T

oontrack have a thing for doing genre-themed releases. The most notable of these is their ‘Metal Month’, which is a regular feature in their release calendar when November rolls around. However, the most recent month with a musical theme was focused upon fusion. The two highlight releases were the Fusion EZX (for EZdrummer, but also usable within Superior Drummer) and the Funk Fusion EBX (for EZbass). Alongside these new sound expansions, there were also a couple of MIDI packs — the drum groove Foundational Fusion MIDI and the Jazz Fusion MIDI for EZkeys — and the Session Amps: Funk Fusion EZmix 3 presets pack. Ready for some

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Fusion-based inspiration? Armed with these new releases, let’s see what the EZ trio can do...

Drum Roll Please...

The samples for the Fusion EZX were recorded at School Farm Studios, Essex, England, with Pete Riley behind the various kits and Stefano Civetta behind the desk.

The two have impressive credit lists with artists such as Nile Rodgers, Ed Sheeran, Bruno Mars, Keith Emerson, Guthrie Goven and Ritchie Kotzen.

Pete Riley was also responsible for selecting the four primary kits used (by Sakae, Tama and two different Yamaha kits) and, as his playing resumé includes lots of music that blends jazz, rock and funk, he is clearly well placed to know what’s

The Fusion EZX features some classy acoustic drum sounds with added extras such as the octobans and gong drum.

required to represent the breadth of drum sounds that span the fusion style. The kits themselves were obviously well populated with items to hit (for example, two snares, seven toms and 10 cymbal options) and, as well as additional snare and kick choices, you also get various percussion instruments, the very distinct sound of octobans, and a gong drum.

Within the EZX, the various drum kit presets are divided into groups based upon these four source kits and, in each case, you get multiple preset options, each featuring different styles of mix processing.

These include different ambience options (from dry to roomy) and options with some

Toontrack Fusion pros • Suitable fusion fun across the whole EZ trio.

cons • You still need to be able to play the guitar.

summary Toontrack fill a whole month with fusion-themed sound and MIDI expansions. The drum and bass options make a particularly great combination.

rather nice tape-style saturation applied.

However, what’s most striking (d’oh!) is the undeniably natural sound and the limited use of compression. This is obviously a very deliberate choice and, for those presets that are not intentionally set up as ‘dry’ mixes, you get all the character of the sounds ringing out and the overtones that come with that. Obviously, that might not be appropriate in some genres of music or mix contexts, but in a jazz/funk/fusion setting, the natural vibe is absolutely right.

However, one of the Yamaha kits was also sampled in a very different configuration — less resonant and with damping — for a tighter sound when needed.

As demonstrated by the MIDI groove collection supplied with the EZX, Pete is obviously a brilliant drummer. These provide a selection of mid-tempo and half-time of grooves in straight 4/4 and

swung 7/8. There are some wonderful syncopated grooves that are very cool to play against and some great tom-based grooves. If you want to expand your drum groove options further, the additional Foundational Fusion MIDI groove pack — also played by Pete — is well worth a punt. This spans a wider range of time signatures and both straight and swung playing. Paired with these kits, these are some inspiring grooves to explore, with all the variations and fills you need to deliver a truly human-sounding drum performance. While Toontrack’s sound expansions often get the headlines, unless you are a top-notch drummer (or bass player or pianist), the value of the MIDI groove/pattern expansions should not be underestimated, especially when they feature musical styles that are technically more adventurous.

And On The Bass...

The Funk Fusion EBX for EZbass was sampled from a Franz Bassguitars five-string Sirius model. These are high-end instruments (around the £3500 mark) and this particular example featured both dual‑coil humbuckers and an active/passive dual-band preamp, making it incredibly versatile when dialling in tone.

That versatility comes across in the various presets within the EBX, which, as well as offering EZbass’ easy-to-use compact control sets for customising the sound of the bass, feature tones that add

some useful amp, saturation, compression and modulation options. There are also presets with octave options, auto-wah, a filter with a resonance control, and a delay for slapback effects. Every preset also includes a one-knob Synth option that you can blend into the bass sound if you need a little extra colour. Finally, the included DI preset is excellent if you want to apply your own choice of amp modelling and/or effects. In short, this is a great‑sounding EBX that can span warm vintage to high-definition modern tones.

The supplied MIDI pattern content is focused on slow and mid-tempo options in both straight and swung 4/4 time. There is some very cool playing amongst this collection, whether you are looking for something that lets the bass sit in a song while some other instrument shines or you want a little more flamboyance when the bass needs to feature. The latter is particularly so with the various slap bass grooves, which make full use of the range of performance articulations EZbass offers.

Slap bass is not to everyone’s taste but, in this case, it’s done with both style and restraint... and sounds awesome. Fingers crossed Toontrack have a further EZbass MIDI expansion pack in the works to provide additional options on this front.

Cue The Keys For the piano, the fusion theme was supported with the Jazz Fusion MIDI pattern expansion. This features the playing

The Funk Fusion EBX delivers some fabulous bass tones alongside some top-notch playing within its supplied MIDI.

www.soundonsound.com / June 2025

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TOONTRACK FUSION

Providing you have the guitar skills, the Session Amps Funk Fusion EZmix 3 preset expansion can provide plenty of guitar tones for your next fusion project.

of Manuel Valera, a Grammy-nominated virtuoso pianist originally from Cuba.

Whether you pair these with a standard acoustic piano or a Rhodes, there is some pretty sophisticated playing on display here. In typical Toontrack style, the patterns themselves are organised into ‘song’ folders based upon tempo (spanning options from 70 to 200 bpm) and time signature with options played in 4/4, 5/4, 7/4 and 9/8.

There is a strong jazz influence in the playing and, while that makes for some truly exquisite chord choices and voicings, it also means you might have to work a bit harder to get your other instruments to follow the piano’s lead. If these heady harmonic heights are your thing, Manuel will have you covered. However, those needing a somewhat gentler introduction to fusion piano will find the Fusion EZkeys MIDI pattern pack provides a suitable alternative.

I’ll Get My Guitar...

With your EZ trio of drums, bass and piano fully fusion‑fuelled, until Toontrack do eventually release EZguitar (yes, we all want it and, no, I’ve no idea if Toontrack will ever do it; I suspect it would present some considerable technical challenges to do it in a way that they felt matched the high standards set by the other EZ instruments), you will actually have to get your guitar out if you want to add suitable guitar lines to your fusion projects.

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However, at least the Session Amps Funk Fusion EZmix 3 presets pack can get you started with some suitable tones.

In terms of cleans, that includes very mellow jazz-friendly lead or rhythm tones and some suitably bright, compressed options that are idea for funk-influenced chords. At the other end of the gain spectrum, there are also some great saturated tones suitable for lead styles, where the distortion is smooth and never gets out of control. And, between these two extremes, there is a broad selection of clean, mildly driven and overdriven tones with and without some effects options such as delay, modulation, auto-wah and reverb.

With most recording guitar players having a hard drive cluttered with multiple amp and effects modelling software, for some, EZmix might not be the first option that comes to mind for a virtual guitar rig, but this preset pack is actually pretty impressive. As ever with EZmix, the streamlined workflow gets you a great sound with a minimum of fuss.

Fusing Fusion As standalone products, each of these releases is very good in its own right.

However, it’s also interesting to see how well they might work together: can you fuse together elements from these fusion-themed expansions to inspire a new idea? Well, obviously, I’ve tried,

and you can audition a few brief examples of my efforts on the SOS website (check out the link below).

As mentioned above, the stronger jazz influence within the EZkeys MIDI pack perhaps makes that fusion‑fusing process a little more challenging but it’s a process that works very well with both the sounds and the supplied MIDI patterns for the EZX and EBX. Obviously, some musical judgement (which I might well lack!) is required to turn performances that were not originally played with each other in mind into a coherent musical idea.

That said, the excellent drum and bass sounds are a great starting point; the rest is a matter of some suitable MIDI editing.

And the EZmix presets let you add some genre-appropriate guitar sounds with ease.

If fusion is your thing — or you want a great starting point to make it your thing — you’re in luck; Toontrack’s fusion-themed expansions provide a great resource, both in terms of sounds and performances.

Audio examples of these fusion packs in action can be found at: https://sosm.ag/ToontrackFusion £ Fusion EZX €89, Funk Fusion EBX €89,

Foundational Fusion Drum MIDI €29, Jazz Fusion EZkeys MIDI €29, Session Amps Funk Fusion EZmix pack €29. Prices include VAT. W www.toontrack.com

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TECHNIQUE

The finished installation: you can see the GK-5 pickup near the bridge, and the TRS data socket fitted to the side of the guitar. The electronics and woodwork are hidden behind the scratchplate.

Fitting A Boss GK-5 Kit

O

There can be advantages in permanently installing the pickup for your Roland/Boss guitar synth or modeller.

Usually, it’s a job for a luthier, but with some basic woodworking skills you can go the DIY route...

The most recent incarnation, the GK-5, is available for guitars with six or seven strings and basses with up to six strings.

As regular readers will know from my Boss VG-800 review in SOS March 25 (https://sosm.ag/boss-vg800), the standard version can be fitted to most guitars without modification. This pickup differs from the earlier 13-pin GK systems, in that it now uses a quarter-inch TRS jack data connection rather than a 13-pin DIN connector, and the signal is now digital whereas before it was analogue. This new Serial GK digital interface is compatible

with the latest Boss guitar synth (GM-800) and modeller (VG-800) but, courtesy of an adaptor box (the GKC-DA GK Converter) it can also control older 13-pin devices.

Another box, the Boss GKC-AD GK Converter, allows earlier analogue divided pickups like the GK-3, GK-2A and GK-2 to be used with the most recent synths and modellers.

The small connector box, which can be fitted to the guitar body using adhesive pads, or using a clamp arrangement that sits around the strap button, has no physical controls. The earlier GK control boxes have a volume control, a GK/Guitar/ Both switch, and buttons for moving up or down through the presets. In practical terms, the lack of such facilities on the new version means using an expression pedal

PAUL WHITE

ver the past few decades, Roland (more recently under the Boss name) have released a number of GK-series ‘divided pickups’ for driving their guitar synths and guitar modellers, with the GK pickups delivering the separate signal from each string that these devices need in order to function.

GK-5 Essentials

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to control the volume of the connected device. The GK-2, GK-2A and GK-3 also had an input for a standard guitar signal, so that it could be sent to the receiving device without needing a second cable, which you do with the latest version.

While the standard GK-5 provides a practical way to equip a guitar for driving compatible Boss devices, not everyone likes the exposed connector box and the loop of cable linking it to the pickup. It is admittedly tidier than the older GK-2, GK-2A, and GK-3 systems, but at the expense of losing physical controls and the ability to feed the regular guitar sound down the same cable. However, there is another option, in the form of the GK-5 KIT, a version that’s intended for more permanent installations

and which brings back the missing control and connectivity options.

The full kit. As the wires are terminated with connectors, there’s actually very little soldering required.

KIT & Caboodle Although the original guitar’s circuitry remains intact, meaning its usual output is not affected, installing the GK-5 KIT invariably requires modification of the guitar, because all the new electronics are hidden away inside the guitar body.

The extent of the modifications depends on the guitar’s existing body construction and control layout, and fitting a GK-5 KIT would normally be undertaken by a luthier.

But as I’m no stranger to DIY, I thought I’d see just how easy it would be to fit one myself. I chose to add it to one of my DIY S-type guitars, and it’s worth noting that I was happy to experiment on this one — if you want to use a GK pickup on a valuable guitar, I’d strongly suggest that you stick with the standard GK pickup and use the adhesive mounting pads for the connector box and pickup, since permanent mods of the sort described below will affect the resale value!

The GK-5 KIT comprises the GK-5 pickup itself, some circuitry inside a low-profile metal case, all the necessary cables, a volume pot, a power LED, a couple of switches, and the output jack assembly. By way of instruction, there’s a very clear diagram that shows

where things plug in and how to wire the switches. Helpfully, all the included cables are colour-coded and terminated in mini connectors that plug straight into the circuit box, which means that the only soldering required is for the switches, the volume pot and the cable that links to the

regular guitar output. However, where to site the circuitry within your guitar is up to you. A common approach is to create a cavity in the back of the guitar, with the pot and switches fitted to the pick guard (as with a Strat) or fitted within the control cavity of an instrument that accesses

The GK-5 KIT comes with a clear diagram to show you how to hook everything up.

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TECHNIQUE

FIT TING A BOSS GK-5 KIT

Marking out the outline of the pieces to be installed, to establish the size and location of the cavity and holes that will be required.

the controls from the back (as with most Gibson models). The switches are small and the volume pot could replace one of your existing tone pots if there’s no space to add an extra control.

After some deliberation, I decided to see if I could fit everything to the back of my guitar’s pick guard, and it turned out that there was just enough room to fix the circuit box, allowing a couple of millimetres clearance at the neck end of the pick guard. To make this fit, though, I’d have to create a cavity around 18mm deep under the pick guard, and this would have to come to within a millimetre or two of the end of the pick guard, so precision would be required. So that I knew where it would be OK to remove wood without it being visible, I started by removing the pick guard, putting masking tape on the guitar body, and then replacing the pick guard so I could draw round it.

I wouldn’t recommend using a hammer and chisel to remove the wood!

The ideal tool is a hand router, fitted with a template-following cutter — that’s a cutter with a bearing at the router end of the shaft that follows a template to allow precise cutting. These needn’t be expensive, but a guitar is perhaps not the best first project, and if you aren’t handy with a router, I’d suggest you make contact with one of the local woodworking clubs and see if you can talk somebody

into doing it for you in exchange for a small donation.

Template Trick To create the template I simply used three straight strips of MDF, but anything solid and around 6mm thick will do the job. The trick is to fix this firmly to the body in such a way that it can still be removed when you’re done cutting, and a common way of doing this is to put masking tape on the guitar body and then another layer on the

The MDF template used for routing the straight edges. Note the drilled hole at the top left, as a starting point for the router.

so I used two cutters of different lengths.

If you can attach a vacuum clear to the dust port on the router, then cutting will be easier and there will be less in the way of wood dust floating around. I know this sounds scary but as long as you take your time it is actually very straightforward, and if you decide that a rear cavity is the way to go, you can use the same tape and MDF strip style arrangement to create your cavity template, after which you will need to drill one or more holes linking it to the usual control cavity to accommodate the wires for the pot and switches. A plastic cover made from pick guard material can be used to cover the cavity.

Having got that part of the job out of the way without incident, I now had to make a hole in the side of the guitar body to accommodate the output jack, which comes with a small circuit board attached (carrying a small connector for the included cable). The kit also includes a plastic jack plate, but to allow the socket to fit along with its little circuit board, the hole you drill needs to almost as wide as the plastic plate. To err on the safe side, I used a 22mm spade bit in my pillar drill and then used a wood rasp to shape the

“If you want to use a GK pickup on a valuable guitar, I’d strongly suggest that you stick with the standard GK pickup and use the adhesive mounting pads...”

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back of the template strips. You then stick the masking tape back-to-back using a few drops of superglue. Once the superglue has solidified, the bond is very strong but you can still peel all the tape away from the body when you are finished.

As router cutters are, by nature, circular (I used a 10mm cutter), it can help to first drill a hole in the corners of your marked aperture, so that you can use a chisel or sharp knife after routing to make the corners more square. As long as you make sure the bearing is running along the sides of your template strips the cut should be perfectly clean. It isn’t wise to try to remove too much wood in one go

The newly routed cavity and hole for the jack socket.

hole so that the socket assembly would fit. It was then easy enough to drill a hole linking it to the main control cavity.

That leaves just one more bit of woodwork to do, and that’s to remove a small amount of wood beneath the location of the divided pickup, to allow its cable a path into the control cavity. The pick guard also needs to have a slot cut in it to allow the cable from the pickup to pass through. The pickup mounts to the rear edge of the pick guard, close to the bridge, and height adjustment can be done either by packing thin strips of neoprene foam under it or by using height

adjustment springs. As the amount of adjustment needed on an S-type guitar is usually only a couple of millimetres, there’s no space for the springs between the pickup and pick guard but it is possible to use them by drilling a hole (just a little larger in diameter than the spring) through the pick guard and into the wood, around 10mm or so deep.

The spring will then sit in the hole giving it room to compress fully as the pickup mounting screws are tightened.

Rather than try to find space for yet another control pot, I rewired the (middle) tone pot to act as a global tone control, and then replaced the other tone control with the volume knob for the GK-5. There was plenty of space for the switches and LED, so I just placed these where they wouldn’t get in the way when I used the guitar’s volume control.

Drilling pick guard plastic is very easy after all that woodwork!

Finishing Off

All the components wired in place on the pick guard, ready to fit to the guitar. The finished result is in the main picture at the top of this article.

For my installation, I decided to fix the circuitry box to the back of my pick guard using double‑sided adhesive tape. One benefit of this approach is that the majority of the wiring is then confined to the pick guard, and once all the connections have been plugged in, it’s a simple matter to tidy up the wiring using thin cable ties.

The only cable I found a little tricky was the one that links the circuitry box to the output jack because it isn’t very long, it is screened (making it stiffer), and it has a ferrite filter in the middle. But, if you plug this into the connector box end just before fitting the pick guard to the guitar, there’s enough slack to do the job (providing you

haven’t placed the jack assembly too far from the circuitry box).

Once I’d wrestled the pick guard assembly back into place, I screwed down the pickup and made the necessary action and pickup height adjustments before testing. If you have a guitar that has insufficient space beneath the strings for the pickup, this can be fixed by putting a thin wood or plastic shim (a strip of old credit card or even thinner should be enough) at the back of the neck pocket, just to lift the end of the neck. This will allow you to raise the bridge saddles so that the GK pickup will fit comfortably beneath the strings.

I plugged my guitar into the VG-800, the blue LED came on and everything worked! For the pot and switches to work, it’s necessary to go to the GK setup page in the connected device and select the pickup type as GK-5 KIT. The centre‑sprung toggle then moves through patches and the three-way toggle selects GK/Guitar/Both.

Admittedly, this project is not for the faint-hearted, but a lot of guitar players do their own modifications, and may well have the skills to do this kind of work — at the end of the day, it boils down to some simple soldering and the ability to use a router safely. Soldering’s easy enough to learn, and if you’re unsure about the routing, then just a little assistance from someone used to woodworking tools should be enough to get the job finished.

www.soundonsound.com / June 2025

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ON TE ST

Arturia Pigments 6

Software Synthesizer

Pigments 6 is bigger, better and free to existing users!

ROBIN VINCENT

T

he latest version of Arturia’s flagship software synthesizer has a smattering of new features and functions that build on what is already a rather delicious platform. But does it serve up anything innovative to this smorgasbord of sound-generating engines, or is it just another layer of cream on top of an impressive and colourful pie? Let’s dig in.

The first thing you notice is that there’s very little to notice. Arturia mention something about a refreshed layout, but with version 5 sitting right next to it, I really

Arturia Pigments 6 €199 pros • Fantastic Modal Engine.

• Granular enhancements that sound great.

• Worthwhile filter improvements.

• Modulating modulators.

• A vocoder that’s more than disco.

cons • Engines need preset management.

summary Pigments remains comprehensive, approachable and thoroughly engaging, giving you even more reasons to jump on board.

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can’t see any difference bar a few swapped knobs. Arturia did darken the knobs in Light Mode for a bit more contrast, which is welcome, but on the whole it’s unchanged.

A more notable change in the UI is the way it handles modulation. The system is already very slick; you can drag any modulator from the central section onto any modulatable parameter, and it becomes assigned. When you drop it, a little display comes up with a coloured knob that you can use to change the amount of modulation being applied, and you can add as many modulators as you want. However, now you can drop another modulator onto the little modulator amount knob in order to modulate the modulator amount. A smaller knob appears underneath where you can set the amount of modulator modulation.

This creates a whole other level of nuance that deepens the sound design experience, although I imagine most users wouldn’t necessarily dig that far down. Sadly, it won’t currently let us modulate the amount that the modulator is modulating the modulator, but we shall see what version 7 brings.

Modal Engine The biggest new feature of version 6 is found in the strings and beams of the Modal Engine. It’s a physical modelling synthesizer that could easily be an

instrument in its own right. You have a choice of two resonators in the rasping of strings or the striking of beams. On either side, you have two exciters, one setting the stage through the initial striking of the resonator and the other providing the bowing through the application of friction.

You can experiment with different variations of excitement, like enforcing a collision or applying transient sample shaping all the way through to extracting granular occurrences, sniffing around noise maps and messing with friction density. At both ends you can use an audio input to provide the energy for the beam or string model. You have control over the number of partials being used, and you can warp in more harmonics, shape the timbre, push the stereo field, and fiddle with the decay.

The harmonic warping gets really interesting once you disengage the pitch quantiser. This brings in all sorts of percussive and atonal possibilities as the harmonics throw punches at each other.

Playing between the friction exciters can be a lot of fun as you collapse the Noise that’s dragging soundscapes out of beams for the twinkling windchimes and warbles of Granular.

The results are quite beautiful, fascinatingly organic, and have buckets of weird, ethereal and experimental vibes.

I particularly enjoy the way everything is visualised; the partials, harmonics and the impact of the exciters are all helpfully animated so that you can find the right focal point for your attention.

However, the biggest drag is not in the friction section; it’s the lack of any preset facility. For a synth engine this diverse and interesting there really should be a way you can save and recall your settings for this engine alone. It feels like an oversight that’s more noticeable with every update.

There’s so much you can do with each of the sound engines that having the ability to save engine configurations seems like a no-brainer. As Pigments gets bigger and more complex, the need for this sort of modularity becomes, I feel, more and more apparent.

A case in point would be the Sample Engine. You have the ability to edit and tune samples, filter, bit-crush, modulate and granulate, but you can’t save that as a thing. The Sample’s Granular processor has gained a Scan function which means that you can set the grains scattering back and forth along the sample. It’s a bit like sprinkling the sample with time dust. Again, this could be an instrument in its own right,

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ON TE ST

ARTURIA PIGMENTS 6

The new Modal Engine brings the sound of hitting things to Pigments.

and we would benefit greatly from being able to save our work at the engine level.

Filters Arturia have put significant work into improving the basic multimode filter section. Part of that is the psychological impact of renaming it the ‘Classic’ filter, which automatically makes it feel warmer, but the more important aspect is the appearance of an analogue overdrive to spice up the input and an analogue mode that compensates for the level drop at high resonances. The original digital mode feels quite weedy in comparison. You do have to shave a few dB off the output to keep things under control, but it definitely brings in some more character.

There are two new filter models, the first of which is called the Cluster filter. This pulls in a bunch of peaks to form what looks like a set of sharp teeth. You can select from one to five peaks, use ‘Spread’ to push them apart, and there are numerous modes that place it in the context of different filter modes. It has a somewhat throaty tone and offers a really interesting and dramatic spin on pads and sustained

sounds. It would be nice to be able to apply modulation to the number of peaks but that’s currently not a thing.

The second is the Lo-Fi filter which isn’t as much filtering as destructively downsampling. The cutoff controls the sample rate but is also tied to an optional pre- and post-filter. These two low-pass filters can offset themselves from the main cutoff to provide some really interesting shaping. It has a very different feel to the bit‑crusher found in the effects section and can add an unusual digital grunginess.

Extras Talking of effects, a new vocoder rears its band-pass filters in the effects section.

will have a different cutoff value. It brings in a lovely amount of variation.

Also within the Random section is a Random type called Random, which is easy to miss because you could have sworn it was there before. Previously, we had Sample & Hold, Turing and Binary, which are all a little bit wordy and overly complicated when really all you want is a good old straightforward bit of randomness. Random is exactly that. You can set the distance between the highest and lowest values, the speed of generation, and polarity, and then have it stepped or smooth, calm or jittery.

One more new feature is the Envelope Follower, which is hidden away in the Combinate section. You can use an audio input or the output of any other section within Pigments and grab the shape of the audio to use as a modulator.

It’s a great way to pull rhythms in from other places or to add some personal expression over the shaping of sounds using another instrument.

Conclusion When taken together, all of these additions, tweaks, and behind-the-scenes improvements make for a decent full‑increment update. That, and its being free to all existing users, isn’t going to disappoint anyone.

I am very taken with the Modal Engine. It’s capable of producing elegant sounds and yet somehow manages to circumvent the more complex aspects of acoustic modelling to be a really compelling and discoverable synthesizer. The granular aspects that I often find to be a bit bewildering seem to have found their footing in both the Modal Engine and the Sample Scanning.

The improvements to the Classic filter are very welcome to an analogue fan such as me, although I also have plenty of time for a good bit of sample rate deterioration.

The modulation of modulators will make many people smile, and I might have even enjoyed using a vocoder.

There was nothing wrong with version 5 that you could poke a stick at, but Arturia have managed to bring more to the table to make it even better. All it needs now is a preset library for synth engines, and I’d be a very happy synth noodler.

“There was nothing wrong with version 5 that you could poke a stick at, but Arturia have managed to bring more to the table to make it even better.”

The new Vocoder. Not just for robot voices...

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You can have between four and 40 bands, control the bandwidth, formant shift and sibilance, and swap between modern, vintage and dirty modes. A Gate control saves us from the rumble of unintentionally open filter bands.

I’m not a fan of vocoders generally, but what I like about this one is how it doesn’t immediately direct you towards vocal clichés. Sure, you can plug in a microphone and use it as the modulator, but you can also route in any of the engines, utility sources or filter outputs and that inspires many more creative uses.

Also new is a modulator called the Voice Modulator that’s popped up in the Random section. It’s a sort of polyphonic step sequencer that sends out a different value for up to eight voices. So, if you tie it to a filter and press a chord, each note

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EastWest get gritty with their latest Opus-fronted virtual instrument collection.

JOHN WALDEN

I

get that some folks are not fans of subscription-based access to their virtual instruments but, for working producers or media composers, it’s difficult not to be impressed by the bang-for-buck offered by EastWest’s ComposerCloud+. The latest addition to the line-up is LO-FI. Yes, you can buy a perpetual licence for the title, but for existing ComposerCloud+ subscribers, it is available at no extra cost, just adding further value. EastWest describe LO-FI as a sonic toolkit for hip‑hop and EDM but with a darker edge. So, if your projects need a little sonic imperfection to make them... well, perfect, is LO-FI what you need to add the necessary grit?

LO-FI Low-down LO-FI provides a virtual instrument library fronted by the latest iteration of EastWest’s Opus. This runs as either as a standalone application or in all the DAW-friendly plug-in formats you might need. With an installed size of just over 22GB, LO-FI offers a collection of nearly 350 individual presets organised into nine categories; Bass, Keys & Synths, Leads, Pads, Guitars, Drums & Percussion, Orchestral, Textures and Vocals.

While these are all conventional enough instrument groupings, as we will see in a moment, the underlying design strategy for LO-FI’s sound designers seems to have been to give all these instruments a suitable sonic twist by any means necessary.

The Sound Of LO-FI My initial exploration of those sounds took way longer than it should have. That’s generally a good sign as it is usually because the sounds are inspiring to play, but it did leave me with two initial thoughts.

First, the sounds are full of character.

I suspect the sound design team had a lot of fun as pitch‑shifting, distortion, modulation and other sonic degradation abound.

However, this is all done with some taste.

The underlying sounds are not trashed within an inch of their life; just given enough colour to impart some sonic personality.

Second, that character is — in general at least — as advertised, which is to say the sounds have a dark, gritty, flavour. When it comes to your sonic expectations, therefore, this is ‘lo-fi’ in an edgy hip‑hop way rather

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EastWest LO-FI Sample Library

than a lazy, lounge-tinted, study music way.

You could undoubtedly drop some of these sounds into that more chilled context, but I’d guess that perhaps wasn’t the primary target EastWest had in mind.

There are some inspiring sounds within every instrument category. However, from a production perspective, it’s the Bass, Keys & Synths, Leads, Pads and Guitars that provide the core of the library’s material.

There is good stuff wherever you look here. That includes plenty of excellent bass options and, whether you just need something super-solid to support the low end (Big Boy Bass, for example), something more EDM friendly (such as Deep Ghost), something more attention-grabbing to provide a hook-like element (Bass Stab), a sub-shaker (plenty of options here), or something a little more obviously aggressive (Overdrive Crank ticks that box), LO-FI has plenty to offer.

The ‘dark’ is particularly obvious within the Keys & Synths category, which is also LO-FI’s largest instrument group.

Yes, there are some dusty pianos that might do the more chilled lo-fi style (such as Dark Piano), but these sounds undoubtedly encourage you down a somewhat more sinister musical route (for example, Danger Pluck, Cold Pressed Chords, Organ Of The Night, and many others). That darker mood is also present

within many of the Leads presets (such as Formantico, Hijacked or Holloween Lead).

The same is true of the Guitar category.

There are some excellent options here for adding melodic elements and a number of presets where the overdriven or distorted sound design is taken a little further. There are cleaner options if you need them (the Big Ben preset sounds great) but also plenty of choices that are a little more obviously processed and/or meaner sounding (such as Broken Wing Guitar or Axes Edge).

While the above might represent the most abundantly stocked categories, there is plenty to enjoy elsewhere. For example, if you want some ear-candy to add the occasional textural element, then, yes, the Textures category provides some rather cool options. Equally, alongside some very playable sounds (such as the Vox Triple

EastWest LO-FI $299 pros • A great collection of sounds for darker hip‑hop and electronic music production.

cons • Perhaps not an obvious first choice for more chilled jazz/lounge flavours of lo-fi.

summary EastWest’s LO-FI delivers sonic imperfection of the highest quality. This is a great palette of sounds for producers or composers who need a darker edge to their hip‑hop or electronic music.

In essence, LO-FI can be thought of as a hybrid instrument.

It combines an underlying (and extensive) sample-based sound set but then lets you modify those sounds with a number of more synth-like options built into the UI. So, for example, at the top left are options to engage portamento, legato and polyphonic options, to LO-FI’s preset collection is well stocked with sounds to get inspired by. transpose pitch, and to adjust the MIDI velocity response.

Stack or Distro Vox Oh presets), there are Top right you can configure full amp and some quirky instruments within the Vocals mod ADSR envelopes. Bottom centre of the category (including the rather wonderful UI is home to independent low-pass and Chopsy Vox preset). high-pass resonant filters. There are also The Drums & Percussion category LFO and step-based modulation options. provides a selection of both kits (a kit The UI also provides a range of effects of drum sounds mapped across the and, in the centre, an X/Y pad with the keyboard) and individual drum elements choice between four more creative sound (a single pitch-mapped sound). The modulation options. The Automation page hip‑hop vibe comes across clearly here, provides further possibilities for modulation, whether it’s chromatic 808s, crunchy and MIDI Learn lets you link LO-FI’s controls acoustic kits, collections of heavily filtered to a suitable MIDI hardware controller. percussion sounds, or single sounds that And, of course, you can easily stack sound have been heavily processed. I’m sure you layers within Opus. could find more comprehensive ‘hip‑hop The bottom line here is that you get meets electronic’ drum and percussion a bit of the best of both worlds; instant collections in a dedicated library, but the gratification from the excellent collection of sounds here cover plenty of ground and presets and a very accessible set of tools for are full of character. manipulating those presets further via some Within the Orchestral category, presets DIY sound design. Hardcore synth sound such as Celli Stac, Violin Stac and Sad designers might still find themselves with an Pizzicato provide options for some classic itch to scratch but everyone else ought to pizzicato-esque hip‑hop countermelodies, be more than happy. all served with a touch (or a lot) of saturated warmth. Equally, there are some It’s LO-FI, Jim... instruments — featuring brass, woodwind and strings — with pitch modulation if At the time of writing, LO-FI was still you want to get a little wonky with your available at a discounted price for the tuning. Whatever you pick, the sounds perpetual licence ($199), and there were are suitably different from your typical deals on the ComposerCloud+ subscription orchestral palette. (a 10‑year anniversary offer of $9.99/ month). I don’t know if those prices will still DIY Design hold by the time you read this but, if you are willing to embrace the subscription Colour scheme aside, LO-FI’s Opus model, ComposerCloud+ — now including front end is the same as EastWest’s Iconic LO-FI — has to be one of the best deals (which I reviewed in the January 2025 issue currently available for working producers of SOS). I won’t cover all the same ground or composers. here, but the key elements are worth I’ll finish with a reminder of the point repeating to give some sense of the DIY I made earlier though. As EastWest make sound design potential if you want to go very clear within all LO-FI’s product details, beyond the supplied presets.

they have aimed for a darker mood with this library. The sounds have undergone a range of processing options to give them a lived-in, gritty, organic sense of sonic imperfection, but that’s imparted a somewhat sinister, and sometimes aggressive, quality to the sounds.

So, yes, it’s lo-fi, but not as everyone might know it. If you wanted a palette of sounds for a more chilled, jazz-meets-lounge form of lo-fi, then perhaps this isn’t the most obvious first choice. However, if you are a music producer or media composer who creates harder-edged hip‑hop or EDM songs or TV/film scores, LO-FI is right on the money and will offer you plenty of sounds to get creative with. There are other virtual instruments that can do ‘dark and sinister’ very well but, as a single, unified collection of sounds, LO-FI is an impressive palette from which to work. Is it possible to do high-quality sonic imperfection? Obviously, it is... and with LO-FI, EastWest have done just that.

As well as all the options provided by the front end, Opus makes it easy to stack presets for further DIY sound design.

£ $299 or via ComposerCloud+

subscriptions ($19.99/month or $149/year).

Prices include VAT. W www.soundsonline.com

www.soundonsound.com / June 2025

73

FE ATURE

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

track/5f0W1kEx5bgjMm77kPAEUz

W www.youtube.com/

watch?v=8INQz5_3N80

Hunter Lea • ‘Real Shoring’ J O E M AT E R A

G

rammy-nominated producer and musician Hunter Lea is also a curator and co-producer of Light In The Attic Records’ Nancy Sinatra & Lee Hazelwood Archival Series. He currently produces and engineers out of Uptone Recorders in Tacoma, WA, a fully functional vintage recording studio based around equipment from the golden age of recording technology. Some of this was used to produce his nominated favourite sound: the swelling oceanic tones on the 2024 track ‘Real Shoring’ he made with Mike Dumovich.

“I first ran a Vox Continental organ right into my favourite amp, a ’62 Fender Vibrolux, a brown Fender amp that has a single 12-inch speaker. It sounds amazing for guitar, but for keyboards, a lot of stuff sounds really good on it too. I miked it using an AKG D19C microphone, but upon listening to it,

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it sounded too dry. So, I added spring reverb and liked that, but then I added a Maestro Echoplex EP-3 tape echo, and that was sounding cool. The final missing ingredient was a volume pedal, but I put that at the very start, so I could swell the sound. I would hit a note and swell it up with my foot and it would go through the echo and then the reverb and it created this washy oceanic sound.

“I did a random take just improvising to hear what it sounded like with this new sound I’d found. And that was the perfect feeling. So, I just improvised the entire take. I didn’t write anything.

I just did a pass where I was sort of swelling the notes in between to not cover up the beautiful melody or any of the piano or anything.”

Seascape “In the end, it was more like a sound, an ambient sound that complemented the piano, this washy oceanic feeling.

I didn’t want it to be too literal with the song, but I liked the idea of creating the sound of an ocean wave with a Vox organ because it’s not typically how these organs are used. And it gets a really unique sound that kind of sounds like a synthesizer, and kind of sounds like an organ, but is definitely unique.

“In addition to that, the Vox has four drawbars, where the tones go from

basically the brightest to the darkest.

So, I scooped all the highs, so that the right three drawbars were all the way down, and had no sound coming from them. And because it’s just the lowest drawbar, it almost sounds like a sine wave. A really gentle, nice tone. Voxes are usually known for their cutting, harsh tones, and while I love those tones too, this sound is really subdued and mellow and unique. It was perfect for the track because it was kind of haunting and ghosty, yet didn’t take up any of the frequency of the piano and vocal, because I didn’t want to obscure that in any way.

“It all was put through a sidecar that was made up of channels from a late ’60s Quad Eight console, which was the console Neil Young recorded the Harvest record on. So, the same mixing board that recorded ‘Heart Of Gold’ also recorded my organ track.”

PODCASTS PEOPLE &

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Creating A Filter Pan Effect

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

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

Behind The Brand

Practical Tips For DAW Users

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Martin Kohlstedt

Steve Dub - Producer And Mixer

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

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

Piano & Electronica

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

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www.soundonsound.com/podcasts

INSIDE TRACK

Having proved his worth on Tate McRae’s second album, producer and songwriter Grant Boutin played a pivotal role in the third.

PAUL TINGEN

R

eleased in September 2023, Tate McRae’s hit ‘Greedy’ helped steer her second album Think Later to platinum in the US and her native Canada. This year, McRae’s third album, So Close To What, went to number one in at least dozen countries, including the US. McRae’s first album, I Used To Think I Could Fly (2022), was made with famous writers and producers including Blake Slatkin, Charlie Handsome, Charlie Puth, Greg Kurstin, Finneas and Louis Bell, all of whom have appeared in SOS. Kurstin also featured on Think Later in addition to Ryan Tedder, lead vocalist of OneRepublic and one of the most successful writers and producers in the industry. Tedder has many writing and production credits on So Close To What, and other big names like Rob Bisel, Emile Haynie, Lostboy, Ilya and Blake Slatkin also contributed, along with the relatively unknown Grant Boutin. Boutin co-produced five tracks on Think Later, including ‘Greedy’, and on So Close To What, he has writing and production credits on eight songs, including four songs on which he is sole producer.

Moreover, Boutin was again involved with some of the big singles, including ‘Revolving Door’ and ‘Sports Car’.

Planting Seeds

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Growing up in San Diego, Grant was inspired to start making music by a very young Tom Norris, now one of the world’s top producers and mixers. (Norris, who was featured in SOS August 2020, this year collected two Grammy awards for his work on Charli XCX’s Brat.) “When I was a kid, Tom was my next-door neighbour.

I would often go over to Tom’s house, and he was an early teenager willing to hang out with an eight-year‑old. He’d be making house music in Fruity Loops, and I thought it was the coolest thing I had ever seen, even as I was too young to understand what he was doing. When I was 11, I downloaded Fruity Loops, and started messing around with it.

“Tom was not only my introduction to Fruity Loops, but also to electronic music. He made me a CD with music

by Pendulum and Junkie XL, and so on, that I listened to non-stop during family vacations. My parents also required me to take piano lessons. I hated it when growing up, but by age 15-16 I enjoyed being expressive at the piano. It made me a better musician than I would have been otherwise.”

The young Grant Boutin grew up fascinated by EDM and future bass in particular. “I listened a lot to Timbaland, and when I was making dance music, I never gravitated towards four-to-the-floor stuff. I always wanted interesting syncopated drum rhythms. Whenever I tried a house track, I never felt inspired or that I had an edge in originality in that lane. So all I did was EDM. Whenever I created a cool groove and added some melodies on top, it felt fresh.

“As a teenager I developed the idea of combining chords and melodies with hip-hop beats and electric music, but didn’t know how to achieve it. But when I heard Wave Racer in 2014, I thought, ‘Yeah, this is exactly what this combination should be, but I’m going to do it my way.’

Wave Racer was doing future bass, and I was into that genre really early. I watched it become big, and started my electronic project Grant Bowtie when I was in middle school. I signed with Monstercat, and released my music via them.”

So Close To What has been Tate McRae’s most successful album yet.

to making a new sick drop, I wasn’t sure what to do that hadn’t been done 100 times before. The favourite parts of my tracks became the vocal

That song, a ballad, has yet to be released, but Grant and Rubin contributed to another OneRepublic song, ‘Mirage’. Tedder’s company Runner Music partnered with Avex to publish Grant, and his career was beginning to take off.

“Ryan started bringing me in for sessions, and found out that I could do other things too, like K-pop, and that I could turn things around really quickly.

One could say I was in a producer boot camp with him for the first year of working with him. He’d send me demos and things that he had started and asked me to flesh out the production more or just to try to make it sound more finished. He’s doing so many songs every day, he doesn’t necessarily have the time to tweak everything.”

One of the things that Tedder sent to Grant during that time was an early version of ‘Greedy’. “He sent it to see what I could come up with. I don’t think there were any instructions. It already had quite a bit of what’s in the final version, including the drums and the vocals, so I started filling up the ambience, swapped out some of the drums, and added more bits and pieces. My tendency is to overproduce and throw too many little sound bits into every crevice of the

Grant Boutin: “When I did dance music, many of my drop lead synths included me humming a melody, and then formanting it a certain way, and layering it with a synth.

Having a vocal element in a song offers nice ear candy, and it’s super fun to do.”

Networking The pandemic then steered Grant’s career in another direction. “I went to the USC Thornton School of Music where I did their Music Production Major. Straight after I graduated in 2019, Covid hit. I was living in downtown LA with a friend, stuck in a dark room with a window looking out at a wall. Until then I had only been doing my artist project, and dance music was dead at that point.

I had been DJ’ing since the end of college, and really loved it. But no more live shows were happening.

“At the same time, I had been getting a bit tired of dance music. When it came

sections that lead up to the drop. My old manager had also pushed me into exploring songwriting and production for others, so I started doing more collaborative sessions, and writing songs all by myself. I wrote tons of really cliché songs!”

Grant eventually came up with what he calls “some cooler stuff”, and managed to sign a publishing deal. In 2023, Lou Al-Chamaa joined Avex Publishing as Senior VP and Head of A&R Publishing.

He liked Grant’s stuff and put him in a room with fellow developing writer Coleton Rubin. The two wrote “a really cool song”, which Al-Chamaa emailed to Ryan Tedder, who, says Grant, “became obsessed with the song and really wanted it for OneRepublic”.

www.soundonsound.com / June 2025

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INSIDE TRACK

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song. They ended up loving what I did for the chorus, because they wanted that really full, and they deleted everything I had added to the verses and pre, so there was a big contrast between the two sections.

“I thought it was a cool song, and I really got into a flow state when working on it, but I had no idea that it would become a single, let alone a global smash hit. Because it did, I was asked to come in to help finish several of the songs on Think Later.

I showed up for the sessions at Ryan’s studio, and hung out with him, Amy [Allen], Jasper [Harris] and Tate. They’d all be leaning over my shoulder while I was at the computer, asking me things like, ‘Yo, can you make this synth sound really edgy, dark and gritty?’ and throwing out song references, and it all needed to happen in an hour. Honestly, it was really challenging. For a month I went to Ryan’s house every day, and did what I could to make these songs sound as good as possible and help finish them.”

Close Up Grant was not part of the very first steps of creating So Close To What, but soon became deeply involved. “I think the songs ‘It’s OK I’m OK’ and ‘2 Hands’ might already have been in existence, and perhaps some others. Some songwriting had already been done. Then I did two sessions with Tate and Amy Allen.

Tate had a very specific sound design request, and I guess I did a decent job of executing it. We wrote a really cool song, and that snowballed into me doing a lot of writing and production on the album.

“With the first album, I was mainly a finisher. I added a lot of cool creative elements to several songs, to service ideas that already existed. On the new album I got to start songs from scratch.

I did as much preparation as I could for these sessions. My brain just shuts down if it’s like, ‘OK, time to make a hit,’ when there’s nothing to start with. So I connected with Ryan and Tate before sessions to talk about stylistic inspiration, and Tate would have playlists with the moods she was looking for. So at home I was making song starts and sounds and drum tracks for inspiration. Others came in with their own ideas as well.

Ryan is really good at being aware of the big picture, while Tate knows exactly what she wants. She has so much inspiration all the time.

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Working with high-profile producers and songwriters helped develop Grant Boutin’s existing ability to get sounds fast.

“‘Sports Car’ is one of the songs for which I came in with the beat, which was pretty much some form of the currently existing song. I was really leaning into an early 2000s lane, using the Phrygian dominant scale, which I have always liked, though Tate has shied away from that specific sound in the past.

In trying to emulate early 2000s sonics I used many detuned saw-wave hits, which I created using Xfer Serum. That lead thing in the intro of ‘Sports Car’ is a detuned single saw wave, going through reverb and then some plug-in with tape wow after the reverb, to kind of glue and detune the whole thing.

“A lot of the bass stuff is Serum, layered with 808s, and me humming underneath. When I did dance music, many of my drop lead synths included me humming a melody, and then formanting it a certain way, and layering it with a synth. Having a vocal element in a song offers nice ear candy, and it’s super fun to do. It’s a creative thing of getting on the microphone, messing around, and seeing what you can come up with.

“I like to layer my vocals quite a bit, and I’ll usually have Melodyne on, so it’s in time and in tune with the other instances of the recordings. Sometimes I’ll formant it down or speed it up or pitch it, depending on what I’m going for. I’m also a big fan of the Output Thermal Interactive Distortion plug-in. I run a lot of things through that, and just go through the presets, or tweak them a little bit. There’s really no exact method. Essentially I have no idea what I’m doing. I just start recording myself and throw weird effects on!

“For synth sounds I use a lot of [U-he] Zebra, and I’m also a Native Instruments Kontakt library hoarder. I really like the Slate + Ash and Teletone Audio Kontakt libraries. But a lot of my secret sauce is not necessarily the synths or samples I’m using, but the way I treat them. I come from dance music, so I side-chain all kick drums to some extent, using Cableguys ShaperBox. It’s part of my template. In a dance song, that side-chain will dip the entire frequency spectrum, but if I’m mixing a song that needs to have a more transparent or gluey-type punch, I’ll just dip the low end.

“Because of my dance music background, I want the drums to hit as hard as humanly possible. A friend taught me how to get a harder punch by simply drawing automation curves for every single kick. So every kick has volume automation going to zero, to maximise the transient.

I do a lot of side-chaining with Oeksound Soothe as well, mostly for vocal effects.

I’m really detailed about delay throws and reverbs, also putting on choruses and tape effects and so on, though in pop music you don’t have the room to go really crazy in the same way with sound design and other elements because you don’t want to crowd out the vocal.”

Sporting Chances Grant applied many of these working methods to ‘Sports Car’, the biggest hit from So Close To What. “I brought my beat for ‘Sports Car’ in after we had already written a full song that day. They heard that glitchy intro in the beginning, and were like, ‘Oh, that’s sick.’ Usually when

“I’m out there with my Windows PC with Fruity Loops, my Focusrite 2i2 interface, and my clunky wireless keyboard that looks like it’s from the ’90s...”

I’m in the room with Tate, Ryan and Julia [Michaels], I tend to be at the computer, but when it’s just Tate and me, I’m also contributing melodic ideas. I’m not very helpful with lyrics, but particularly when Tate and I were at Jungle City Studios in New York later on in the project, where we wrote and recorded ‘Bloodonmyhands’ and ‘Like I Do’, I had a Shure SM7B next to me, and we’d both be ad-libbing with Auto-Tune and full vocal chains.

“At some point when working on ‘Sports Car’, Julia Michaels had the idea of doing a whisper chorus, like the Ying Yang Twins’ song ‘Wait (The Whisper Song)’ [2005]. We were all into that idea, but for a while Tate wasn’t sure. In the vinyl version of the song there is no true whisper-only chorus, there’s a whisper with talking underneath it. That was how the song existed for a very long time.

But Julia was really passionate about the whisper idea, saying it’s what makes the song. Eventually Tate hit me up one day and said, ‘Hey, let’s try it with the whisper

again.’ After a few more versions and turning up bass sounds and incorporating some other requests, she was into the whisper as well.”

Grant also contributed a small but distinctive element to the intro. “One of my favourite things is the glitchy bit right before the song starts. It’s a bunch of small sound bites, like radio sounds and guys’ voices and all this stuff panning around. It’s microscopic and happens very quickly. Sometimes a song needs to have a really attention-grabbing intro. I had no idea what to do for this one, so I simply threw some things together. For a while others didn’t want it in the song, and then everyone decided to keep it. I also added a guitar in the second chorus.”

The most essential element of the song, the vocals, were cut by Grant and McRae at Tedder’s studio. “Only towards the end, when Tate wanted to recut some lines, did we record at Westlake. Pretty much all Tate’s vocals that I recorded were cut with a Neumann U67 or U87,

Against The Grain Whereas the pro audio world is predominantly Mac-based, Grant Boutin is a diehard Windows user, and works exclusively in FL Studio.

“I’ve been using it for 15 years and I know all the hotkeys. It’s just what I’m lightning fast in. And deep down, there is this fun feeling of carrying the torch for Fruity Loops in the pop space. Every session I go to, my co-writers say, ‘Whoa, I’ve never even seen someone use this program’ or ‘I didn’t even know you could record vocals in Fruity Loops.’

Sometimes producers will say, ‘Don’t worry, bro. I’ll cut the vocals,’ and I’m like, ‘Hang on, I can do it too!’

“I’m out there with my Windows PC with Fruity Loops, my Focusrite 2i2 interface, and my clunky wireless keyboard that looks like it’s from the ’90s. But I’ve always wanted a wireless mechanical keyboard, and I love it. It means I never have to readjust my brain to the layout of the laptop keyboard. When I go out to sessions it can be embarrassing when people want to AirDrop me stuff and I have to say, ‘Sorry, can you email me or send it via a file sharing website?’ There are some websites that let you do AirDrop between Windows and Mac, but they’re finicky and sometimes don’t work. So I’ll also bring a flash drive because it’s just too much of a pain

to do everything to email. Sometimes I think I should get a Mac, but don’t know how to use one. I’ve been on a PC my entire life.

“Having said all that, there are some things that I wish were different about Fruity Loops. It’s like an endless blank canvas, where you can drag anything anywhere.

None of the tracks conform to effects. You have to manually click each audio clip and assign it to a mixer track. It doesn’t have the same playlists and comping options as Pro Tools or Ableton, and because it doesn’t really matter where anything is in the arrangement window, it can get super disorganised very quickly.

Recording audio in Fruity Loops can be a pain, but the fact that I don’t have to interrupt my workflow to switch DAWs makes it worth it.

“Despite the drawbacks I just described, Fruity Loops is totally capable of doing vocal production.

I recorded almost all of Tate’s vocals on the songs I co-produced on the new album in it. On the song ‘Sports Car’ I maxed out my Fruity Loops project. It was literally up to track 500! And probably 80 percent of that was muted vocals.

I didn’t want to touch a single thing because I was so scared that I would mess something up.”

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INSIDE TRACK

GR A N T BOU TIN • TAT E MC R A E

going through a Neve 1073 and then the Tube-Tech CL-1B. Pretty standard. My vocal chain in Fruity Loops also is very simple. I’ll do a high-pass, and then have an Xfer OTT compressor, Waves RVox and CLA Vocals, a de-esser, an exciter, and if there’s a resonance that is hurting my ears, I’ll add Soothe. One thing I do that people think is dumb is have Auto-Tune baked into the track. But I don’t have heavy settings, and it saves a lot of time, especially with the Fruity Loops workflow.

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“I’ve worked with artists that love to do 20 to 40 takes of each line, which to me feels a lot. With Tate, we go line by line.

Sometimes she kills it on the first take.

Sometimes we’ll do five, sometimes we’ll do 10, but it rarely is more. I do a lot of vocal doubling, grabbing takes that were good but not as good as the lead, and I’ll turn those into doubles. Sometimes I’ll pitch a vocal an octave down, and mix it in so you can barely hear it, but it’s reinforcing things. Another thing that

I started doing is either manually turn all breaths down to super quiet before all the compression hits it, or I’ll separate them all out onto their own track and turn them down, just because when you’re compressing the heck out of a vocal, the breaths get super loud.”

Arriving At Perfection Grant says he did “probably 50 versions” of ‘Sports Car’ as he and the others tried to arrive at the perfect result. “It’s

Unusually, Grant Boutin still works entirely in FL Studio. This screenshot shows the Tate McRae hit ‘Revolving Door’.

probably the only time in my life I’ve put that much work in a song, and loved the song more at the end than when it started. Normally when I go through that many versions of a song, I’ll be sick of it, but ‘Sports Car’ was getting cooler all the time. Tate really drove that, and I really respect her taste and vision!”

Throughout his work on the songs he was involved in, and for each of the 50 versions of ‘Sports Car’, Grant was creating rough mixes. He often did this

back at his own studio, where he also has “Yamaha HS8 monitors, a Manley microphone, a Neve 1073 mic pre, a MIDI keyboard, my guitar and my bass, and Audio‑Technica ATH-M50X and Audeze LCD-X headphones. I like mixing on headphones. The Audeze LCD-X phones are open-backed and really good for mixing, but I’ve kind of stopped using them as much recently because I’m not the most skilled mixer as far the correct EQ curve for the general song is concerned, and

because the Audezes are so forgiving on the ears and not super bright, I tend to mix overly bright on them. When I then put some normal consumer headphones on I’ll be like ‘Ah, this is hurting my ears.’ So what I do now is mix on my Audio‑Technica M50Xs, because they are a lot harsher. It stops me from mixing too bright.

“My go‑to plug-ins during mixing are the Soundtoys EchoBoy and Valhalla delays, and the Valhalla Vintage Verb.

I’ve recently also been getting into the Liquidsonics Seventh Heaven reverb. For more creative effects I love the Valhalla Super Massive, and the Shimmer. The latter has such a specific sound that’s great for Tate McRae’s stuff. She loves that reverb.

She loves everything to sound airy and atmospheric, and the Shimmer can fill out space. It’s also really fun to suddenly go from a section of a song that is completely dry to sections with huge reverb tails. That is kind of an EDM-ism as well. I got used to a sound where every breakdown before the drop had some eternal reverb sound on the vocal, and then suddenly that disappears.

“I’m always side-chaining vocals to their reverbs because it can otherwise get really muddy, using Soothe for the higher‑frequency stuff. It kind of does what the Wavesfactory Trackspacer does.

It notches the actual resonances, and is not dipping the whole thing. I also use a lot of chorusing and tape effects on everything, which I think are nice additions to electronic sounds to make them sound more warm and less purely digital. I love the Baby Audio tape emulation plug‑in, TAIP, which is supposed to be AI, and the Aberrant Audio DSP Sketch Cassette II. I’ll also use the Wavesfactory Spectre saturation EQ plug-in a lot. I love clipping, and tend to clip all my drums, using the Kazrog KClip Zero plug-in, and Decap’s Knock plug-in. A lot of the drums in ‘Sports Car’ are pretty clipped.

“I have to make these rough mixes sound pretty finished and powerful, to get everyone impressed by them. I try to make everything pretty maxed out on the loudness scale and make sure everything is high-passed when it needs to be. Plus everything’s side-chaining and punchy and the vocals have to be sitting right. After that I count on a mixer to help me take it that next five to 10 percent, and of course, on Tate’s album we had great mixers like Serban Ghenea, Manny Marroquin — and Tom Norris!”

www.soundonsound.com / June 2025

81

COMPE TITION

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date. You can use these emulations either directly in your DAW, or via UA’s latency‑free Console front end, which in addition to providing a flexible and professional digital mixing platform with world-class emulations, also integrates Sonarworks’ industry-leading room and headphone correction system.

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www.soundonsound.com / June 2025

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ON TE ST

Active Monitors

IK Multimedia iLoud Micro Monitor Pro IK’s newest iLoud speakers pack serious monitoring into an ultra-portable format.

PHIL WARD

P

aul White reviewed the original version of IK Multimedia’s iLoud Micro desktop active monitor very favourably back in 2016. The intervening decade has seen the monitor establish itself as one of the benchmarks when it comes to ultra-compact portable monitoring, but its time is now over, because its replacement has arrived: the iLoud Micro Monitor Pro. The iLoud Micro Pro and its predecessor share an almost identical aesthetic and at first I imagined that their injectionmoulded thermoplastic enclosures would be of the same dimensions, but I was wrong; the Micro Pro has grown a little, with its cabinet enclosing around three litres internal volume compared to the previous model’s two. This internal volume increase is significant in that it will help square the circle that comprises one of the fundamental constraints of all loudspeaker design: the trade-off between low‑frequency bandwidth and maximum volume level.

Along with being dimensionally inflated, the iLoud Micro Pro differs from its predecessor in boasting increased amplifier power, by incorporating IK Multimedia’s ARC room correction system and X-Monitor control app, and through the inclusion of an ARC measurement microphone. The increase in amplifier

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power apparently arises because, rather than working in a primary and secondary arrangement, where one monitor houses the amplification for both, the iLoud Micro Pro works as a conventional active system with amplification present in each monitor.

The addition of X-Monitor integration and ARC brings the iLoud Micro Pro up to the same feature level as IK Multimedia’s existing iLoud MTM and iLoud Precision monitors. ARC, and room correction generally, is potentially of significant value in a monitor such as the iLoud Micro Pro, which is likely to find itself pressed into service in all sorts of non-optimal spaces and installations.

Other minor changes and improvements to the iLoud Micro Pro are the deletion of the previous 3.5mm TRS jack input and its replacement by a balanced XLR input, the inclusion of a slightly larger‑diameter tweeter, and, in acoustic specification terms, a marginally lower low‑frequency cutoff (50Hz against 55Hz) and a slightly higher maximum continuous sound pressure level (94dB against 92dB). The iLoud Micro Pro also incorporates an

internal power supply rather than working from an external mains power adaptor.

Monitor Emulation In addition to managing the ARC room‑correction process and general housekeeping functions (firmware updates, delays and standby setting), X-Monitor offers speaker emulation that aims to impart the characteristics of a range of alternative monitors onto the iLoud Micro Pro. The monitors emulated aren’t fully identified in X-Monitor, but the names drop some broad hints: I’d hazard a guess that White ’80s is the Yamaha NS10, and going by the image in the X-Monitor promotional material, High-End 3-Way is an ATC SCM25A. X-Monitor also provides emulations of generic hi-fi speakers, TVs and mobile phones.

Of course X-Monitor/ARC is not the only room-correction product that provides such functionality, and I’ve expressed my doubts before about its genuine value. To my way of thinking, although it is fun and even perhaps useful at a very rudimentary level, speaker emulation has about as

much chance of making a pair of iLoud Micro Pros sound truly like, say, an ATC SCM25A, as me donning a blonde wig will make my bass playing sound like Tina Weymouth. I say this because the subjective sound of any speaker is defined by so much more than simple frequency response. Speakers display a unique set of characteristics in terms of distortion profile, compression behaviour, time‑domain performance, delayed resonance signature and dispersion pattern that will define their sound as much as, if not more than, their axial frequency response. So it’s simply not possible to make one speaker genuinely sound like another through simple EQ. Having said all that, the X-Monitor/ ARC speaker emulation is technically accomplished as far as it goes and I’m sure is intended to be useful rather than just a gimmick, even if it (and systems like it) can provide just a vague hint of the sound of an alternative monitor. Much better, though, to learn and understand through practice and experience how your monitoring translates to alternative playback systems and contexts.

Driving Force The front panel of the iLoud Micro Pro carries a nominally three‑inch (75mm) bass/mid driver, a one‑inch (25mm) fabric-dome tweeter and a generously flared reflex port exit. Said port curves upwards within the cabinet to a flared entrance high up behind the back of the tweeter. The generous length of the port is necessary to achieve the desired low tuning frequency; however, long ports often come with an undesirable side-dish of organ-pipe resonance, so it will be interesting to see if the iLoud Micro Pro designers have swerved that issue.

The diaphragm of the bass/mid driver is described as a “composite cellulose fibre” item, which I think translates into non-marketing speak as “paper”. There is nothing remotely wrong with paper as a driver diaphragm material. The fact that it is still used so commonly is witness to the fact that it is generally so well suited to the role. Both iLoud Micro Pro drivers are protected by sturdy perforated

metal grilles, which seems very sensible considering the highly portable nature of such compact monitoring, and the tweeter is mounted at the apex of a subtle elliptical waveguide moulded into the front panel. Speaking of moulding, one great advantage of constructing speaker cabinets from plastic is that edges can be generously softened, and that’s the case with the iLoud Micro Pro. Softened edges will help significantly to suppress the diffraction effects that would otherwise cause various frequency response discontinuities.

Under Pressure The fundamental physics of loudspeakers defines a fixed relationship between the volume of air that must be moved by a diaphragm and the sound pressure level (SPL) at a certain frequency: as frequency falls and SPL rises, the volume of air required to move grows exponentially.

The diaphragm moves air by pumping backwards and forwards, with the resulting SPL defined by the amplitude of the movement.

In a reflex‑loaded speaker the port helps through the mass of air it contains, pumping similarly backwards and forwards — in phase with the driver at the port resonance frequency.

Now, air volume is a multiple of diaphragm/port area and linear distance, so a diaphragm and port that are not much blessed with area, like those of the the iLoud Micro Pro, need to move a long, long way to generate reasonable SPL, and that’s potentially a problem, because both are limited by mechanical constraints (for example, the diaphragm can only move as far as its voice coil and suspension allow).

To illustrate all this, I built a simulation in some speaker modelling software of a reasonably high‑performance three‑inch OEM bass/mid driver in a three‑litre cabinet with

IK Multimedia iLoud Micro Monitor Pro £599 pros • Tiny dimensions.

• Well designed and engineered.

• Highly capable monitoring performance.

• ARC and X-Monitor integration is effective and works well.

cons • Struggles with low bass and high volumes.

summary The iLoud Micro Pro continues the theme of its predecessor but brings both significant performance improvements and very effectively integrates ARC and X-Monitor.

It absolutely hits the target for ultra-compact and portable monitoring.

Connections On the rear panel we find a gain control, balanced XLR and unbalanced RCA phono input sockets, a mains input socket and power switch, a USB-C socket for connection of the supplied measurement microphone, and a matrix of push buttons that offer some broad-brush tonal adjustment, and which can also used to put the iLoud Micro Pro into ARC equalised mode.

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a port tuned to 55Hz — a system that’s approximately representative of the iLoud Micro Pro. In my simulation, the diaphragm excursion (the distance the diaphragm has to move) reaches its effective mechanical linearity limit of ±5mm when generating around 93.5dB SPL (at 1m) at 70Hz. I wrote its “effective” limit because actually, diaphragm excursion rockets further upwards at frequencies below the port resonance; however, that can be fixed in the simulation by adding a high-pass filter at around 35Hz, and it’s likely that the iLoud Micro Pro incorporates something similar. The filter protects the bass driver from over‑excursion and the amplifier from wasting precious power ineffectually pumping the diaphragm backwards and forwards (“ineffectually” because, in common with all reflex‑loaded speakers, the iLoud Micro Pro is able to generate only minimal useful output below the port resonance frequency).

But it’s not just the driver that has a linearity limit. The reflex port has one also, and my simulation shows that the airflow velocity through the port when the system is generating 93.5dB is well into the non-linear, turbulent‑flow region and likely to be noisy. Now, obviously, without knowing all the electro-acoustic details of the iLoud Micro Pro, I can’t be certain that my simulation is perfectly representative, but I’d bet the house on it being not far off.

I say that because my simulated maximum sound pressure level of 93.5dB, combined with a low‑frequency cutoff of 55Hz (-3dB), is within a gnat’s whisker of the iLoud Micro Pro published specs. Its “continuous max SPL” specification is 94dB and its low‑frequency cutoff is 50Hz. However, along with a continuous SPL specification, IK Multimedia publish a “peak SPL” spec for the iLoud Micro Pro of an eyebrow‑raising 109dB. I’m somewhat at a loss to understand how such a level is realistically possible from the iLoud Micro Pro, and I think 109dB needs to be accompanied by a large pinch of salt.

There’s an added complication too, because the real‑world maximum SPL of a speaker that is likely to be massaged by significant room compensation EQ, such as the iLoud Micro Pro, is almost impossible to predict. For example, in my installation, the ARC room compensation introduced a significant EQ dip between 70Hz and 90Hz, which will benefit potential maximum SPL. But ARC also introduced around 3dB of gain at around 200Hz, and that will not only work the bass/mid driver harder, it will

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The iLoud Micro Pro ships with an ARC measurement microphone, for room correction using the X-Monitor app. X-Monitor also lets you emulate a selection of studio and domestic playback systems.

also ask questions of the iLoud Micro Pro’s amplification headroom, because 3dB of gain requires double the amplifier power.

All this analysis and explanation is to show just how critical these electro-acoustic calculations become when monitors are miniaturised. The compromise reached by the engineers behind the iLoud Micro Pro is, I think, pretty well judged in its ability to offer worthwhile low‑frequency bandwidth combined with more than usable maximum SPL. It’s not, as described in the iLoud Micro Pro literature, “physics-defying”, but it’s quite an achievement, especially at the price.

Setting Up Instead of installing the iLoud Micro Pros at the acoustically treated ‘business end’ of the studio, I decided to set them up at a secondary workstation at the other end

of the room. The intention here was to use the monitors in the kind of a less‑than‑ideal environment that they might well find themselves in the real world, while also giving the ARC system a thorough workout.

The ARC process, managed through the X-Monitor app, is pretty quick and straightforward. With the monitors connected to the computer via USB, X-Monitor automatically identifies that the monitors are present, and when you’ve confirmed that the correct monitor is in the appropriate location, hitting the X-Monitor Calibration button begins the room compensation process. X-Monitor presents a series of screens which guide you through the process of connecting the supplied ARC measuring microphone and taking the measurements. ARC requires sine‑sweep measurements taken from four locations around the listening position

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for each monitor, and when those are complete it calculates the EQ required to flatten the frequency response of the monitor at the listening position.

ARC optimises each monitor individually, and this has both up sides and potential down sides. The main up side is that it’s easy to scale up a monitoring system to include multiple monitors — in fact iLoud Micro Pro packages up to 7.x.4 are available. I’ve substituted an “x” for the usual .1 LFE because, at present, there’s no compatible IK Multimedia subwoofer, and X-Monitor offers no bass management facilities. A possible down side of ARC treating monitors individually is that it might apply radically different EQ to each monitor of a stereo pair, and that will have the potential to degrade stereo imaging.

Neumann’s MA 1 room‑compensation system, for example, treats stereo monitors as composite systems for this very reason.

ARC is easy to use and although relatively limited in its implementation with the iLoud Micro Pro, it works well. It is clearly less technically sophisticated and capable than, for example, the Trinnov system, but then it is also very significantly less expensive.

Measurements I took one iLoud Micro Pro to my usual empty church space and made a few measurements. Diagram 1 shows the iLoud Micro Pro’s axial frequency response, and it demonstrates a reasonably flat characteristic with no obviously issues, although the tweeter level looks a little exaggerated. However, there are two curves overlaid on Diagram 1, one taken at 90dB SPL and one taken at 80dB SPL, and that’s because I quite quickly discovered when taking data that the iLoud Micro Pro incorporates a limiter function intended to protect its small bass/mid driver from overload. This is really no great surprise (numerous much larger active monitors incorporate limiters) because the laws of physics are inalienable, and despite the aspirations of marketing departments, ultra-compact, inexpensive monitors have their limits. So, the red curves of Diagram 1 shows that the iLoud Micro Pro begins to turn its bass/mid driver down once the SPL gets to about 90dB (it will let louder short‑term peaks pass unmolested). The remainder of my iLoud Micro Pro data was captured at 80dB. Lastly on Diagram 1, phase response data is inherent to the capture of a frequency response and

Diagram 1: The on-axis frequency response of the iLoud Micro Pro, measured at 80 and 90 dB SPL at 1m (green and red curves, respectively). The measurements have been normalised to allow for direct comparison.

Diagram 2: The on-axis response, and distortion products at 80dB SPL (at 1m). Second and third harmonics are shown in green and blue, respectively.

although I’ve not complicated Diagram 1 by including it, I can confirm that, as claimed in its promotional material, the iLoud Micro Pro has a generally linear phase characteristic from around 300Hz upwards.

Diagram 2 again shows the axial frequency response (at 80dB), but this time with the second- and third‑harmonic distortion displayed. In the context of its very small dimensions and low price, the

option is also overlaid. The rear‑panel EQ is perhaps somewhat academic thanks to the availability of the X-Monitor app and ARC, but it’s useful to see what it does. It works pretty much as expected although the LF and HF EQs operate over relatively restricted bandwidths.

They are quite subtle.

I wrote earlier about the experience of using ARC to compensate for my studio acoustics, and as the ARC parameters remain with the monitor rather being stored within the X-Monitor app, it is possible to engage and disengage them from the iLoud Micro Pro’s rear‑panel Cal button. So, Diagram 4 shows the default axial response overlaid with the result of my ARC room compensation. The dip aimed at suppressing the primary low‑frequency mode in the room is the most significant feature.

Finally, I mentioned the risk of organ-pipe resonance in long reflex ports, and Diagram 5 shows the result of placing the measurement mic right in the port mouth. The desired port resonance at between 50Hz and 60Hz is obvious, and its decay thereafter is very clean. The feature around 1kHz probably is a port organ-pipe effect, but because the port

“ARC mode did bring some significant benefits in terms of even tighter imaging and a better sense of tonal accuracy.”

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distortion performance of the iLoud Micro Pro is perfectly reasonable. Distortion, especially with bass/mid drivers, is very much a function of diaphragm excursion, and with so little diaphragm area available, more excursion is required. Through the critical mid band, distortion runs at around -40dB to -50dB, or between 1% and 0.3%. This is pretty typical of inexpensive compact monitors.

In Diagram 3 I checked the function and effect of the rear‑panel EQ switches, The diagram shows the axial response overlaid with high‑frequency and low‑frequency attenuation and boost. The desk EQ

mouth is so close to the bass/mid driver on the iLoud Micro Pro it’s actually quite difficult to disentangle their outputs.

A bit of further geeky data processing I carried out suggests quite strongly that the 1kHz feature comes from the port, but it is somewhat academic because its level is significantly below that of the bass/mid driver. I’d give the port a clean bill of health for this reason. It would be interesting to know how the iLoud Micro Pro designers have suppressed the organ-pipe resonance, but disassembling review samples to look inside is generally frowned upon.

Diagram 3: Comparing the speaker’s response in its default flat mode (red trace), with +2dB on the HF and LF bands (blue), with LF and HF set to -2dB (orange), and with the desk EQ option engaged (green).

Listening In Measurements reveal that the iLoud Micro Pro is a very competently engineered monitor. It has a generally flat frequency response, linear phase response, acceptable distortion and no sign of any unexpected and significant flaws. It is a very small monitor, though, so volume levels that stray into loud territory are challenging. But what does it sound like?

I spent some time with the iLoud Micro Pro in both its default mode and then ARC optimised for my studio installation. In default mode I found the overall balance of the iLoud Micro Pro slightly bright, so very quickly went for the tweeter cut option.

That helped, but I then felt that the iLoud Micro Pro lost a little of its clarity. Bass is surprisingly extended for such a compact monitor, but port noise could become audible with bass‑heavy material at volume levels not far beyond gentle. Between bass and high frequencies, however, lives the all important midrange, and the iLoud Micro Pro does a pretty good job of reproducing it. The iLoud Micro Pro’s midrange character doesn’t have the showstopping, see-through clarity and detail of the best nearfield monitors, but it’s revealing enough for mix work and is tonally neutral.

Stereo imaging is strong, with good focus and satisfying depth. Reverb tails are nicely detailed and firmly located in space.

In ARC optimised mode I’d hoped that the low‑frequency attenuation applied to suppress the room mode would also

ALTERNATIVES Unusual as it is, the iLoud Micro Pro doesn’t have the field entirely to itself.

If you feel the urge to build your own monitors, there’s the AudioSmile LBM, but if not, the AIAIAI Unit 4 and ADAM DV3 are worth investigating, as is the more conventional Genelec 8010A.

Diagram 4: Before and after ARC processing (red and green traces, respectively).

Diagram 5: A close-mic measurement of the port output.

benefit the low‑frequency power handling and allow the iLoud Micro Pro to play louder without port noise. That wasn’t initially the case, because the default ARC equalisation, as well as attenuating between 70Hz and 90Hz, also added some gain at lower frequencies. However, using the X-Monitor EQ facilities to add a little low‑frequency attenuation worked wonders to restrict the iLoud Micro Pro’s over-enthusiastic ambitions to perform like a somewhat bigger monitor, and allowed it to play a little louder. ARC optimisation did bring some significant benefits in terms of even tighter imaging and a better sense of tonal accuracy. It also brought a little more detail to the midrange and simultaneously fixed my slight concerns around the inherent high‑frequency balance.

The iLoud Micro Pro is a very small monitor, and despite the claims of 109dB, its genuinely high level of performance is only really available at modest volume level — the kind of levels suited to very nearfield desktop use. Push the iLoud Micro Pro too hard and its performance quite quickly becomes a little loose and slightly messy. But in the end, the combination of ARC optimisation with a little more targeted X-Monitor EQ tweaking resulted in the iLoud Micro Pro working very well in its intended role and sounding really very good indeed.

So good I stopped listening analytically and started just enjoying the music. £ £599.99 per pair. Individual speakers

£299.99 each; 11-speaker immersive bundle £2999.99. Prices include VAT. W www.ikmultimedia.com

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Acustica THING

Software Synthesizer

Acustica’s soft synth debut is a recreation of everyone’s favourite ’80s polysynth.

ROBIN BIGWOOD

T

HING is the first VST/AU/AAX virtual instrument from Italian software developers Acustica, joining their vast line-up of audio processing plug-ins in a new ‘Marea’ (the Italian word for ‘waves’) range that promises more synth emulations down the line.

It’s an odd name for a synth, THING, and especially so for one which (as many aficionados will already have spotted) is so clearly referencing Roland’s Jupiter‑8, the high-end analogue synth produced from 1981-1985. But why not? It’s certainly distinctive, and makes a change from yet another arbitrary code-like name or metaphorical whimsy.

Actually THING is two synths, THING8 (TH8) and THING5 (TH5), which are both part of a standard installation. TH8 is the more sophisticated, with a user interface

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that’s very close to the original Jupiter‑8 top panel, while TH5 has a friendlier, simplified parameter set and no arpeggiator.

It Don’t Mean A Thing Focusing on the full-fat THING8, to describe its basic topology is very much to describe the Jupiter‑8. There are two VCOs with switchable waveforms (sine, triangle, square/PWM, with a noise option in VCO2) feeding a filter section with a simple static high-pass stage and a much more fully equipped 12/24 dB/octave low-pass. There’s a single multi-waveform LFO (though VCO 2 can also be switched to a low‑frequency mode), and two envelope generators.

Notable synthesis niceties include sync of VCO 2 by VCO 1, and variable cross modulation (bidirectional linear FM) between the oscillators, for clangourous noises. There’s a fixed key follow option for envelope times (getting shorter as you

play higher), and plenty of ways to deploy modulation concurrently but independently.

Jupiter‑8s have an interesting section to the left of the keyboard, where the pitch bender and a modulation button (instead of a modulation wheel) could be set up to control the pitch of the VCOs (separately or together), the filter frequency, and modulation amounts going to oscillators and filter. That’s intact in TH8, albeit with controls set above the keyboard instead, and with the modulation button replaced with a more useful and standard mod wheel (which explains the related loss of the LFO Mod Rise Time fader). A welcome trade-off of authenticity for versatility.

It’s a legendary synth of course (and one for which, as a kid growing up in the early ’80s, I’d have happily traded a body part or two) but this topology is not in fact especially complex, and Acustica have largely held firm to that original conception.

Acustica THING €229 pros • A lively, elastic, analogue-like quality.

• Switchable filter characteristics add huge range.

• Useful onboard effects.

cons • Some aspects of installation are needlessly invasive.

• The feature-stripped TH5 could feel redundant for many users.

• Documentation lags behind current feature set.

summary A superb-sounding Jupiter-8 emulation, with abilities that extend to radically different, US-flavoured synth characters as well. Not cheap, but it’s quite special.

The control layout is very similar, give or take a bit of horizontal condensation.

Terminology is the same throughout too, and the eight‑note polyphony limit is retained. In fact the only significant changes, apart from the switchable filter emulations and effects we’ll come to in a minute, include jettisoning the original’s dual-layer architecture (and a repurposing of the original’s Lower/Upper Balance knob as a pan control), and an understandable axeing of preset and memory switches.

There are a few novel enhancements.

An additional, second RESON[ance] knob can be found in the filter section: Acustica tell me they added it to compensate for the inability of the hardware Jupiter‑8’s filter to go into self-oscillation. So it’s a variable boost, and for the filter types that could already self-oscillate it can make for some glorious harmonic chaos. It’s a really welcome feature. TH8 also gets velocity sensitivity knobs for the VCF and VCA sections, but there seems to be no facility for the synth to respond to aftertouch.

Going further, TH8 is equipped with an effects section that many early 80s Jupiter owners, having already shelled out the equivalent of about £15,000 in today’s money, could only dream of. (A Jupiter‑8 has no effects at all: not even the chorus that would become an iconic feature in the Juno series.)

A Saturator module offers 15 different hardware amp and pedal emulations.

Chorus then has obvious similarities with the Roland Juno chorus, though here increased in scope with a harmonic saturation bias control, and variable noise.

The delay section offers emulations of 20

historic analogue, digital is really useful. And finally and tape-based delays, there’s an Age parameter: as with a stereo ping-pong you turn it up, the front panel option and MIDI Clock sync graphics get progressively divisions. Finally a reverb more knackered, and instead offers 10 different algorithms of voices just going wonkily (and CPU-saving equivalents out of tune against each alongside) with pre‑delay other the most obvious and size parameters, and feature becomes an ongoing a ‘dark’ option. pitch wobble. It’s really nice, Many other sound and loosening things up, but playing possibilities come surprisingly subtle: even from combinations of the maxed out things are only controls at top left of the just unpleasant, and I couldn’t plug-in window and the discern any decorrelation green voice assignment of envelope times or filter buttons. There are more response (as with the Vintage interactions and subtlety parameter of a hardware than I can describe here, Prophet‑5 Rev4, say). but basically a Superstereo Going back to those effect duplicates sounding enticingly colourful buttons, Many of the effects are taken from Acustica’s extensive range of voices, and to varying the blue Hold latches any processor and effects plug-ins, and played notes to turn them degrees pans the resulting include some rare and unusual voice pairs and de-correlates into never-ending drones. hardware models, like the delays them in phase. They can be Unfortunately though it detuned too. This works both for polyphonic continues to behave in a strictly cumulative playing and in the monophonic Solo way when using the arpeggiator. Whatever modes, making for some beautiful effects notes you hold and release to begin with varying from subtle thickening to luxurious, are just fine, but pressing others thereafter loose chorusing. Unison mode plays all just adds to whatever is already latched. oscillators at once, but counterintuitively That’s not the way most arpeggiator does not make TH8 monophonic: holding latches work, nor how we’d want them to: down extra notes gives you a chord, with you just end up with messy note clusters. a redistribution of sounding voices, the Currently MIDI CC64 messages activate maximum number of which you’ve chosen Hold, so a sustain pedal doesn’t do from the Voices sub‑menu. An interesting anything different. The arpeggiator itself is if not especially useful effect, and with fine, and sticks to the original design, with a significant weakness in that it provides a 1-4 octave range, and four familiar basic no way to play the ensuing oscillator stacks patterns including random. It can be sync’ed monophonically, for domineering bass or to your DAW or an internal clock. lead duties with portamento.

Sound & Substance In any mode, a sub‑oscillator is available for extra welly, and you can choose I won’t beat around the bush: THING is between sine and square waveforms, which one of the best-sounding plug-in synths,

Housekeeping As is common nowadays, Acustica use a separate application called Aquarius to handle plug-in installation and updates. It’s clear and functional, and includes an uninstall option.

A THING installation is more invasive than most other software instruments though, at least in macOS, which I used during testing. Where almost all plug-ins are typically represented by a single file in the OS’s VST and AU folders, THING writes nine files and a sub‑folder containing dozens more in each. Worse, an Acustica folder appears in the top level of the macOS user Home folder, and despite containing nothing but temporary and other system-like files it had grown to an inexplicable and unwelcome 1.2GB in size after several weeks of testing. Talk about making yourself at home.

Even in use, you sense some underlying mechanisms might be quite complex. THING’s graphic interfaces took about three seconds to render when first opened on my M1 Max Mac, and about 0.5s thereafter, when every other plug-in on my system opens instantly. In use THING is not particularly demanding of CPU, but the Superstereo effect (which doubles voice count) and the optional Mega quality setting in the effects section increases the load. The CPU-saving ECO quality mode, incidentally, sounds really lacklustre with some kinds of presets.

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ACUSTICA THING

or analogue emulations of any kind, I’ve ever heard. It’s characterised by real presence, elasticity, and a sparkling, open, three-dimensional quality. If you associate virtual analogue with sterility or that sort of soapy lack of focus and clout (things start getting quite metaphorical when discussing these things) THING might make you think again.

On both TH8 and TH5 it’s the default VCF-8 filter mode that for me is the jewel in the crown. Even when the filter is already fully open, VCF-8 has noticeably more high‑frequency content (and can be louder) than all other filter types.

So if your notion of the Jupiter-8 sound is something fundamentally bright, sparkling and liquid, this will not disappoint. Having said that, the other filters are good too, and help the synths to span a huge timbral range. The Moog-inspired LAD is much darker and heavier, while the OMX is fizzy and light, and there’s massively different resonance behaviour and quality to boot. Does THING become a Memory Moog when you engage LAD? Not exactly — synths are always sums of all their parts — but having these options available hugely increases scope for sound design.

THING sounds very good just in its raw state, without the Superstereo feature, or the onboard effects, but it can do even more with them. Superstereo is valuable because it’s so controllable: a sniff of it lends some size and dimension without taking over the stereo image. It also maintains positive stereo phase correlation until quite extreme values. The Saturator stage does everything from subtle harmonic enhancement to fuzzy

destruction, and Tone’s apparently simple three-band EQ can actually have its band frequency centres adjusted, graphically, in the otherwise blank panel at top right. The chorus mimics a Juno chorus in that there are three modes, but whilst the third (‘I+II’) correctly drops the pseudo-stereo phase opposition, its very fast choppy warble is an acquired taste. You’d need a forensic approach to separate the many modes of the delay, at least for simple treatments, but suffice to say there are digital as well as analogue/BBD style sounds, and

and when comparing those TH8 still has considerably more presence and sparkle, with a character that is not matched by opening the Jup-8 filter (or switching to 12dB/oct mode, for 24dB/oct sounds). This is not to say that Jup-8 V4 is bad. Far from it: it’s also a great-sounding virtual synth that offers some more possibilities around unison mode than TH8, and has valuable additional modulation sources and effects processors. But TH8 more often made me think ‘Oh I see!’ in a way that was normally reserved for the increasingly rare occasions I’ve met a hardware Jupiter‑8 (or Prophet‑5, or Minimoog...) in a studio. In that respect it deserves some serious kudos as an emulation: it’s probably the finest virtual Jupiter-8 out there, currently.

One thing I haven’t tackled yet is the question of value. It’s pertinent because the THING ‘suite’ is costly as plug-ins go: €229. Yes, Arturia’s Jup-8V is only a little cheaper, at €149 (though it’s much better value as part of the bigger V-Collection bundle), but another well-regarded Jupiter-8 emulation, TAL’s J-8, is well under £100. Roland has its own ‘official’ Jupiter-8 emulation too, that can be had through a subscription model or bought outright at around that €/£150 price point again.

I think it’s likely, after comparing all these options, that many users would decide that THING has qualities that make it worth the premium. Currently though, I feel the total package, from installation to ongoing use, has too much unevenness to earn a simple unqualified recommendation.

I’d have expected to see full, updated

“I won’t beat around the bush: THING is one of the best-sounding plug-in synths, or analogue emulations of any kind, I’ve ever heard.” a ping-pong option. Amongst the reverbs there are some quirky and characterful things, though sadly nothing really luxurious and modulated, and definitely no shimmers.

The maximum decay time is no longer than about eight seconds, which is a bit lacklustre for some jobs, but hey, we’ve all got other reverbs in our DAWs.

Conclusion To give some additional context, during testing I made some extensive side‑by‑side comparisons with Arturia’s Jup-8 V4. The two soft synths occupy the same sonic territory, as you’d hope, but for similar settings the Acustica synth always produced more high‑frequency content. Both emulations include original Jupiter-8 presets like ‘Negative Pluck’,

Learning & Sharing THING has a funky, retro-style PDF manual, which is detailed in some places and infuriatingly incomplete in others. For example, the most recent version I had makes no mention at all of the extra resonance control, and one of the filter types, despite THING having been out for a while.

A particularly poorly documented feature is THING’s facility for hardware MIDI control via incoming MIDI CC messages. Through trial and error I found it can be made to work, for individual or all instances of the plug-in, but the implementation isn’t much better than the documentation. You can never actually see your control assignments once made, you can’t remove them individually, and the only way to delete global assignments is to fish around deep in your computer OS and manually throw away an entire configuration file. Babies and bathwater stuff, which feels extremely clunky compared to some of the competition.

The preset system is better: hundreds of factory presets are provided, and you can recall them through a pop-up menu or via a preset browser that includes descriptions, author info, and a possibility to ‘favourite’ or rate individual presets. Presets don’t exist as separate files on your computer, but it is possible to export them for sharing.

Meanwhile, THING uses a Waves-like system whereby two current states of the plug-in, A and B, sort of coexist, allowing you make straightforward comparisons, easy variations of patches and so on.

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TH5 If TH8 is a Jupiter‑8, you might at first wonder if TH5 is a Prophet‑5... ‘No’ is the answer, although this smaller sibling does have the same ‘VCF-PRO’ filter option. What you get instead is many of the core TH8 synthesis features intact, with the exception of there now being only one envelope generator.

Virtually all performance-oriented modulation options are missing: it doesn’t have a virtual mod wheel even, no sustain pedal response either, and pitch-bend is fixed at ±12 semitones. The GUI is simpler and smaller.

TH5 can sound good, but I can’t help wondering who amongst THING’s potential buyers would ever bother firing up this simplified synth when TH8 is available. Acustica’s claim that TH5 is a CPU-saving option didn’t play out in reality for me, especially as it’s still capable of eight‑note polyphony (not just the five mentioned in the manual). At the same time, it’s odd that a genuinely unique and quite esoteric feature, alternative envelope modes, is found only here. The four different modes (of which three are documented) seem to adjust filter response and interaction just as much as timing curves. Curious.

documentation and much nicer installation etiquette. On the synth itself a more useful Hold implementation is needed, along with better MIDI Learn facilities, and the strange mono Unison limitation should be sorted out. I wonder if we couldn’t have an option for greater polyphony too, even just in the Poly modes — really long sustaining sounds would benefit from it hugely.

However, ultimately it is sound quality, much more so than mere features, that makes any synth truly valuable, and THING gets that right in a way that’s still uncommon in the virtual world. It can sound absolutely glorious, alive and responsive.

I can imagine it becoming many users’ ‘go-to’ synth, with the multiple filter models covering hugely more ground than

a Jupiter‑8 could by itself. To the extent that it could even be a single synth that could fulfil all roles. THING is quirky, yes, but it’s also already very impressive. A free demo will let you see if it can do its thing (sorry...) in your studio. £ €229 including VAT. W www.acustica-audio.com

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And that pretty much describes Tascam’s new FR-AV2, a miraculously compact professional stereo audio recorder with a robust fibreglass casing. It measures just 99 x 80 x 40mm and when loaded with a trio of AA batteries it weighs only 267g.

If the numbers are hard to appreciate, they make the FR-AV2 fractionally larger and heavier than the Zoom F3, which is its closest equivalent, but the differences aren’t significant in any practical sense.

Features

Tascam FR-AV2 Portable Audio Recorder

With superb mic preamps, 32-bit recording and comprehensive timecode facilities, the Tascam FR‑AV2 is a miniature marvel.

Tascam FR-AV2 £370 pros • Class-leading, clean and quiet mic preamps.

• 32-bit floating‑point format negates worrying about recording levels.

• Comprehensive timecode facilities.

• Good battery life and powering options.

• Bluetooth remote control and remote monitoring features.

• Nice clear screen and simple menus.

• Extremely compact.

cons • Single microSD memory card.

• Bluetooth adaptor a cost option.

summary A superbly engineered stereo audio recorder with superlative audio quality, the freedom of 32-bit floating‑point files, sophisticated timecode implementation, easy operation, good battery life, and great Bluetooth remote control features.

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HUGH ROBJOHNS

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hen I started working in pro audio, a professional two-channel audio recorder was the size and weight of a concrete block. It consumed 12 D-cells each working day, could only record 25 minutes of audio (at 7.5ips) on a five-inch reel of long-play tape, and required a skilled operator to maximise the audio quality because the usable dynamic range was barely 50dB! By the mid-1990s, of course, digital audio changed all that, first with the 16-bit R-DAT format, then with 24-bit solid-state recorders and, more recently, with 32-bit floating-point recording. Each technological evolution reduced the recorder’s size and weight, and today’s stereo recorders are barely large enough to accommodate the batteries and XLR connectors. Meanwhile, solid-state media and lithium batteries allow continuous all-day recording with ease...

Tascam’s FR-AV2 delivers professional-quality, two-channel audio recording, with sample rates of 48, 96, or 192 kHz in either 24-bit linear or 32-bit floating-point files, using the broadcast WAV (BWF) format. Recordings are made onto a single microSD, microSDHC or microSDXC card, and data capacities of up to 512GB are supported. The memory card plugs into a slot behind a rubber seal on the right-hand side of the recorder, alongside a USB-C socket and on/off slide-switch. Usefully, the FR-AV2 features an automatic file closing system to prevent loss of already recorded material in the event of power failure.

The miniature memory card format is not a very practical one for physical exchanges with a card reader to transfer data to a computer. Instead, a more practical solution is linking the recorder directly to the computer over a USB 2.0 connection, and accessing the data as a remote storage device — it’s compatible with Windows 10/11 and macOS 13/14/15. A secondary application for the USB socket allows the FR-AV2 to be configured as a 2x2 USB audio interface (supporting 48 and 96 kHz samples rates only), again with 24-bit fixed or 32-bit float quantisation.

If the quarter-inch threaded insert on the base of the recorder and 48kHz sample rate option didn’t give it away already, the FR-AV2 is heavily optimised for use with video cameras. It boasts rock-steady timecode synchronisation features with 3.5mm mini-jack input and output sockets on the lower side panel for physical connections to any third-party timecode sync’ing equipment. This is a really useful facility, with support for all the standard video frame rates. Moreover, a socket on the top panel of the machine accepts Tascam’s optional AK-BT2 Bluetooth adaptor. This not only provides wireless timecode transmission/reception, but also remote stereo audio monitoring, BEXT and iXML metadata entry, and full remote control via Tascam’s Recorder Connect app.

Both iOS (16/17/18) and Android (12/13/14) operating systems are supported for the Bluetooth app, and it’s also compatible with Atomos timecode devices.

Even more usefully, when not required for timecode I/O, those same mini-jack connections double up as unbalanced stereo line inputs and outputs, which is handy for sending or receiving audio to/from a video camera. The nominal working level is -19dBV (maximum +1dBV), while the nominal unbalanced output is -10dBV (maximum +6dBV), making it well suited for consumer equipment.

The input can also provide plug-in power (2.5 or 5 V) for electret mics.

A third 3.5mm mini-jack socket on the same panel provides a dedicated stereo headphone output, with adjacent rocker volume control. This delivers up to 50mW per channel, optimised for low-impedance headphones (32Ω seem to work best), and while a little more

headphone power would be useful, that would harm battery life.

Two professional electronically-balanced analogue audio inputs are presented on combi XLR sockets at the left-hand side of the device. The XLR mic inputs accept signals up to +4dBu, and the centre TRS sockets take line levels up to +24dBu. A menu page allows the recording source to be selected from the XLR (mic), TRS (line), mini-jack (Ext), or USB audio. Phantom power can be switched on/off and set to 24V or 48V using the menus.

Tascam say the FR-AV2 mic preamps are “ultra‑low-noise HDDA” with an EIN figure of better than -127dBu. That’s on a par with the best portable recorders on the market, as well as many high-end studio mic preamps.

The analogue front end feeds dual high/low gain converter stages, which are combined to capture a claimed 132dB dynamic range for the 32-bit floating‑point mode. Although it’s possible to overload the mic preamps it’s

unlikely in practice, and the dual converters ensure an incredibly low noise floor. To my ears, the FR-AV2 sounds very quiet and clean indeed — quieter than the Zoom F3 and my own F8n Pro, actually — and I found there’s still more than enough (digital) gain available to capture low-output mics like the Shure SM7B without noticeable noise.

Power Play While miniaturisation can be a blessing in many applications, a small battery compartment inherently restricts the recording time, which is always a concern in portable applications. In the worst-case scenario for the FR-AV2, with 48V phantom powering of two mics and 32-bit floating‑point recording, you have about 4.5 hours recording time with alkaline batteries, 5.5 hours with NiMH batteries, or 10 hours with lithium cells. Switch to dynamic mics or line sources and 24-bit recording, and those times extend to around 9, 8.5 and 15 hours, respectively. Considerably longer recording durations, along with power source redundancy, can be achieved by using an external USB power bank connected via the USB-C socket. For non-portable applications, an optional mains power adaptor is also available.

On The Menu

While it works perfectly well as a high-quality audio recorder in its own right, the FR-AV2 includes a number of features aimed at audio for video work.

The FR-AV2’s operating status is shown on a 50mm (two-inch) colour LCD screen, with four soft keys along the bottom edge to navigate a simple menu structure. Below these is a quintet of standard transport buttons, and menus enable various options for initiating and stopping recordings. The three LEDs on the panel’s chamfered edge below the transport keys illuminate when the unit is recording, and if either input channel exceeds -2dBFS. A five-second pre-record buffer is enabled by default, meaning each recording contains five seconds of audio from before the record button was pressed, to ensure that nothing is missed if recording in a hurry.

Returning to the display, it is crisp, bright and informative without being cluttered. Battery status and Bluetooth activity is shown in a blue bar across the top, and below that, on a grey background, is the elapsed record-time display, file name, recording format, sample rate, and remaining SD card duration. By default, files are date-stamped and numbered sequentially, but the date section can be replaced by user-entered text (6-9 characters), if required, to indicate a specific project or scene.

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TA S C A M F R-AV 2

Moving on down, the selected input sources are indicated to the left of a horizontal stereo input meter, scaled from -48 to +6 dBFS. The meter is green up to -12 (the mid-point), yellow to -6, and orange above that — an arrangement that I found worked pretty well — and if the A-D converter is overloaded, the entire meter turns red. A second stereo bar-graph meter underneath shows the output level and, at the bottom of the screen, four labels indicate the soft-key functions, defaulting to Menu, Browse (file finder), Input and Output.

Pressing F3 (Input) accesses the input signal conditioning facilities, starting with two digital channel gain controls (0-60 dB) that can be ganged or operated independently. By default, digital gain is set to 18dB, presumably on the assumption that this is low enough to avoid overloads in most situations, and the 32‑bit format allows low‑level sounds to be retrieved without significant noise.

A stereo-link option applies most signal conditioning options to both channels together instead of independently, as well as encoding the recorded audio as a single stereo file, rather than two mono files. Processing options include polarity inversion, low-cut filter (off, 40, 80 or 120 Hz), input delay (up to 300ms), a limiter/compressor (with preset parameters but pleasingly effective),

a noise gate (with three threshold presets), an equaliser (with presets for ‘interviews’ and ‘music’, as well as a manual setup — the latter accessing a four-band EQ with high and low shelves and two parametric mid sections).

There’s also a Mid-Sides decoder which can be configured either to decode Mid-Sides inputs and record the left-right stereo equivalents, or to record the raw Mid-Sides signals directly and place the decoder only in the monitoring signal path.

In both cases, an on-screen slider adjusts the stereo width and, if recording raw Mid-Sides signals, the decoder can also be enabled on playback. However, there is a hidden trap: although these M-S decoder functions appear in the Input and Output

ALTERNATIVES While the market is awash with stereo recorders of varying qualities, the closest equivalent to the FR-AV2 is undoubtedly the Zoom F3, which is slightly less expensive. However, the Tascam’s better timecode options, clearer screen and quieter preamps will justify the extra cost for many. The Tascam Bluetooth adaptor also costs a little more than Zoom’s equivalent but, again, it allows far greater functionality, which will help to justify the expense.

Marks are added to recorded files automatically, indicating specific periods of time (5, 10, 15, 30 and 60 minutes), or where the input signal exceeds the maximum A-D level, or where there are ‘buffer overflows’ in the SD card. Marks can also be added manually using the F1 soft-key and, on playback, marked positions (all, or a selected type) can be accessed using the F2/ F3 buttons to fast wind in either direction.

A menu screen also lists all recorded marks with their type and time location, if needed.

“There’s more than enough gain available to capture low-output mics like the Shure SM7B without noticeable noise.” menus, they are disabled by default and can’t be enabled from within these menus at all. Instead, it’s necessary to navigate to Menu / I/O Settings / MS Decode, and then select either Rec (to convert on the way in) or Monitor (to convert on the way out). This seems unnecessarily cumbersome to me: surely these functions could be activated directly from the Input and Output menus, something that I hope can be addressed in a future firmware update.

The outputs have similar signal processing facilities, including independent or ganged level attenuation, a peak limiter, an adjustable output delay (up to 300ms) to compensate for latency between vision and sound in the camera, and Mid-Sides decoding.

The FR2-AV is almost as small as it could be while still housing batteries, XLR connectors and the microSD card it records to.

Best In Class?

Tascam’s FR-AV2 is a seriously impressive and capable device. It is very well engineered, with blissfully clean and quiet mic preamps, excellent 32-bit float implementation, rock-steady timecode, and remote Bluetooth monitoring and control options. Also, despite the fibreglass casing, I had no problems at all with RF interference, even when deliberately abusing the recorder using a mobile phone or placing it next to radio mic equipment. That’s not something I can say of all the portable recorders that I’ve reviewed!

I found the screen larger and generally easier to read than that on the Zoom F3, although possibly not in very strong sunlight, and the Tascam’s menu structure is, for the most part, very simple and logical. The battery life is quite reasonable, and most importantly, I can’t fault the sound quality at all — it’s comfortably amongst the best I’ve heard from any portable recorder at any price. And at the asking price, that makes the FR-AV2 really very special. £ FR-AV2 £369.99, AK-BT2 Bluetooth

adaptor £54.99. Prices include VAT.

T TEAC UK Ltd +44 (0)1923 797205 W www.tascam.eu

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ON TE ST

Møffenzeef Mødular DitDit Eurorack Module

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nless you’ve been living under a rock over the past few years — or, less judgementally, if you’re new to modular synthesis — you’ll have noticed a tangible increase in developers looking for new and novel stimuli to put to musical use. ADDAC’s wind controller, Schlappi Engineering’s binary counting or the Clatters Machines Garden Listener, for example.

But if, like me, you’re prone to immediately wonder how many such designs are in fact more interesting in theory than they are in practice, you will be heartened to hear the DitDit from Møffenzeef Mødular is as musically useful as it is conceptually rich. The DitDit takes text as its stimulus and spits out Morse code, which makes for ideal translation into a language of triggers and gates, creating intriguing polyrhythms and non-linear combinations.

The DitDit has a USB input on the front panel, which accepts a PS/2 computer keyboard. This presented the only real headache. I got one keyboard, which didn’t work. I got another. That didn’t work either. Eventually I did what is recommended in the manual, and bought an old-school PS/2 keyboard with a six-pin connector and a USB adaptor. Success! So if you have a local computer shop (you know the type: a grimy haunted-office-garage filled with dusty early-2000s tech), head on down and watch them raise their eyebrows when you ask for a PS/2 keyboard for, er, your synthesizer. While you might sniff at the idea of buying an extra component for your module, it’ll set you back around £10 at most and is totally worth it (once you get the right one). It makes inputting text quick and spontaneous — far more so than an encoder, say.

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As for the panel of the DitDit itself, it features an OLED screen, a Start/Stop switch, three inputs, three outputs and the USB input.

The screen itself is detailed enough to handle some lovely retro-style lettering and rudimentary menus (more on that shortly). The three inputs consist of a clock input — here labelled Bang! — as well as Reset and Start/Stop gate inputs. The outputs are where things get more interesting, not to mention elegantly designed: there’s an overall output for both the ‘dit’ and ‘dah’ content of the signal, but ‘dit’s and ‘dah’s each get their own outputs as well.

This was excellent: I could send the main output to a sequencer, for example, and then allocate the ‘dit’s to one timbral destination being sequenced and the ‘dah’s to another. However you route it, the result is more often than not quick and instinctive, but also complex and — importantly — not random. I posit that balance is not an easy thing to strike for a Eurorack developer.

Using a keyboard, of course, has benefits beyond typing. The DitDit allows for words or phrases to be saved to one of eight slots for quick recall. How to save? Ctrl+S of course! And it can only be Ctrl+O to open a saved phrase. Need to exit a menu or clear the screen? Hit Esc. The usefulness extends beyond

nostalgia: the F-keys are used to configure several useful things, for instance the weighting between ‘dit’s and ‘dah’s (‘dah’s can be worth the length of two or the conventional three ‘dit’s) and spacing. Spaces between characters or words can be enabled, for more traditional phrase-based flurries; or they can be disabled, for gapless successions of gates and triggers. Both are musically applicable in their own way.

I found that gapless made for better drum sequencing, while spaces worked very nicely for modulation or sending via a slew limiter.

If you’re a conceptual thinker, the DitDit is a theme park. Perhaps it’s the fact that I’m currently writing a PhD, so forgive me, but the prospect of sending Morse code of the lyrics of a song to create rhythm for that very song is rich indeed.

You could encode secret messages into your music, or toy with the timbral monotony of traditional Morse code. Even just on the surface, it draws a lovely parallel between the musicality of speech and that of coded language. Once again I return to one of my favourite things about modular, which is the artistry and poetry of the whole thing, once you really get going. Simple, elegant and a lot of fun, no ... --- ... needed.

William Stokes £

$250

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www.moffenzeefmodular.com

Erica Synths Perkons Voice Eurorack Module

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rica’s Perkons HD-01 is one of the most inventive drum machines of recent years. Its four voices combine digital synthesis and samples with analogue filters and overdrive, all controlled by a WYSIWYG X0X‑style sequencer that mixes traditional TR-type programming with Elektron-style p-locks, chance, probability and ratchets. The result is something that’s truly original, but it’s not without its frustrations. For example, there’s a 16-step pattern limit when there are 64 step keys crying out to be used on a per-track basis, and while you can modulate all the voice controls, the modulation itself is global.

And wouldn’t it be nice if you could use multiple instances of the same voice? It’s also far from cheap, so all those sounds and idiosyncrasies

have so far been out of reach to most.

The Perkons Voice is, as its name suggests, a single Perkons’ synth voice in Eurorack form.

Unsurprisingly, it’s considerably cheaper than the HD-01, but it may also answer a lot of the other drawbacks with the original. For example, it has modulation inputs aplenty, it can switch between algorithms instantly, and it’s Eurorack, so you can sequence it with whatever you like. OK, so it doesn’t come in a sturdy metal box in a divisive blue colour, but again, it’s Eurorack, so you can put it in whatever colour box you like. As you get to know the Perkons Voice you might even start to wonder if, far from being a cut-down version of the original, in many ways it might be better...

At its heart are the same 13 synth algorithms as the HD-01, a mix of analogue and digital styles — the latter including wavetable and Karplus Strong — and a sprinkling of user-replaceable samples (256kB is a sprinkling, right?). You can scroll through these algorithms using the data wheel, edit your selection with the front‑panel controls and save it as a patch.

The Voice controls are also the same as the HD-01: Tune, Decay, Drive, Parameter 1, Parameter 2 and Cutoff, and there are CV inputs for each of these alongside inputs for Trig, Accent and Patch CV. The innocuous sounding Patch CV could be the Perkons Voice’s killer feature. All it does is change the Patch when you feed it CV, but this changes it from being a drum voice to drum voices, and while linear drumming is obviously compulsory, the four-note polyphony eliminates voice stealing on all but the most sustained sounds. So, when paired with any CV and gate sequencer you can carefully program multi-part drums with, say, kick, snare, tom and hi-hat sounds. Or, and it’s a big ‘or’, you can point something like a Turing Machine at it, turn the tempo up and sit back to enjoy the absolute chaos that ensues.

A more measured approach is obviously possible, but even turning the knobs on a CV sequencer is a hugely creative way to program percussion, immediately taking you away from the considered ‘select an instrument, pick a step’ method of TR-style programming into something much more fluent,

Modular Profile: Ross Fish

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efreshingly shrewd and not a little iconoclastic, Ross Fish founded Møffenzeef Mødular with all the creativity and risk you might expect, given the modules in the Møffenzeef range.

On his entry into modular My foray into Eurorack started in 2012, at college.

I studied Sound Design for Film at Savannah College of Art and Design. Very quickly into the start of my degree, I realised that I didn’t like working on films very much. The school had a Doepfer A-100 system that

I learned from in a modular synthesis class taught by Professor Matt Akers, who was very influential on my studies and interests. The two of us started performing together as Moffenzeef, playing live improvisational experimental sound stuff entirely on Eurorack systems that we assembled from bits and pieces of the school’s system and his own personal system. Most of the ‘music’ we made sounded pretty terrible, but we had a lot of fun performing around town. At the time that I was graduating, the job market for audio engineers was pretty

where you’re feeling your way, making adjustments until it sounds right.

The Perkons Voice, like the HD-01, has its own sound. It’s possible to coax TR-type sounds from it, but it sometimes feels as though you’re swimming against the tide. For the most part the sounds here are weird. They can be brash, aggressive, unruly, abrasive and all sorts of other adjectives you wouldn’t want to see on your child’s school report.

That said, time spent getting to know the Perkons is rewarded with a huge variety of sounds, and the extra modulation on offer here adds to that considerably. Having just two parameter controls per algorithm can feel frustrating on the HD-01 (as is trying to remember what they control with each different algorithm), but it’s less of a problem here in Eurorack world where you can use additional envelopes and modulate to your heart’s content. It’s not Plaits, but there’s a huge range on offer here beyond the obvious percussion duties. Erica’s trademark sound is often in evidence, but this is much more than a drum voice for making very hard techno. In fact it’s much more than just a drum voice.

The Perkons Voice obviously isn’t an instrument in the same way the HD-01 is, but it delivers the same sounds at a fraction of the cost — and with far greater flexibility than the original.

David Glasper £

£285

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www.ericasynths.lv

grim and I didn’t really know what to pursue after school.

Professor Akers suggested that I learn how to design and build Eurorack modules, so I took his advice. I had no electronics experience at all and just dove straight into it, learning as I went along.

On the DitDit DitDit is truly the most experimental thing I’ve ever designed and was definitely an intellectual exercise.

I wound up creating a pretty unique sequencer that makes patterns that you probably wouldn’t come up with on your own. It started from a conversation I had last year with my older pal Marty

Morgenbesser. Marty was telling me about his ham radio obsession and how when he was younger he had to pass a Morse code efficiency exam to get his license. He told me “musicians always seemed to get the hang of it because it has a sing-song‑like rhythm.”

I started thinking and realised that the rules of Morse code aligned with modular clocking concepts; gates with varying pulse widths happening at divided clock intervals.

After looking up the rules of Morse code, I then realised that there was an underlying polyrhythmic structure imposed on the sequences.

I thought about the musical implications for these rules

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and thought about ways the user could alter them while using the device to get different results. I designed what I believed to be the most flexible real-time Morse code sequencer I could think of, and as far as I know, I’m the only person to make something like this. I’m really proud of it and think it’s one of the cooler things I’ve designed. I couldn’t have done any of it without the programming expertise of my good friend Ben Davis, who wrote the firmware.

On the story of Møffenzeef Mødular Møffenzeef started with blind ignorance, abundant arrogance, and a lot of youthful naïveté. Towards the end of college, I came up with this noise box thing called the GMO. I decided that I wanted to turn it into a Eurorack drum module and try to start a company like Akers suggested.

I was Facebook friends with Lars Larsen of LZX Industries and we were in the same synth groups. I saw that he was moving to Portland, Oregon to work for Darkplace Manufacturing, which he described to me as “a record label for synth companies” where they would fund, manufacture and distribute modules. I told him that I was a Eurorack designer — I wasn’t — and that I had a working module prototype — I didn’t. He invited me to Portland to discuss future prospects. I wound up flying out there with a half working circuit encased in a cigar box. I presented it to Lars and Josh Holley, the Darkplace owner, and did a sales pitch that should have been featured in a Twilight Zone episode. The entire angle of the pitch was “none of these modules are supposed to sound like they actually work, they’re all supposed to sound broken and fucked up” (this statement later became a core ethos of all future designs I worked on).

I still to this day don’t understand how or why they said yes to me, but they did.

They wound up giving me a contract to fund and produce my first Eurorack module, the GMO — a ROMpler module that was designed to sound like a broken drum machine. My girlfriend Courtney (now my fiancée) and I sold all of our stuff and moved to Portland to start the company out of Darkplace. After a few years, I moved production in house and started doing stuff on my own.

On the culture of modular Lately I feel like modular synthesis culture is stale and uninspiring. There is a hyper emphasis on consumerism and a fervent

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desire from customers for new things constantly. I’ve come to terms with the fact that I’m making art therapy tools for wealthy people who want to have a hobby. I’m not knocking that — believe me, I’m thrilled to take your money — I’m just acknowledging that very few of my customers are musicians or are even interested in making music to begin with. If you ask the average person at Superbooth why they are interested in modular synthesizers they will give you a similar answer. Most people who buy this stuff use it as a decompression tool at the end of their work day with no real objective or creative task in mind.

I approach designing all of these tools with the goal of music-making in mind.

Most of my energy goes into thinking about the physical interface and how

a musician would pragmatically use them. A lot of the designers in Eurorack are software developers or engineers and not musicians. I’ve always been a musician first. I feel like a lot of people in the community don’t understand my motivations or why I would make such strange things to begin with. As odd as this sounds, coming from someone who makes dial tone drums and Morse code sequencers, everything was designed with the goal of making music. Running this company has always been odd and frustrating for me. I feel conflicted and misunderstood! I’ve distanced myself from electronic music outside of work.

These days I spend most of my time playing Klezmer music with old people.

My bass has become my security blanket.

William Stokes

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“I admire Sound On Sound as the survivor amongst the professional media. I knew the people that founded it, I read technical articles and have even lifted some of the articles and applied them to a few of my own designs!

Long may you last!”

The Late, Great Rupert Neve Electrical engineer, founder of Rupert Neve Designs.

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INTER VIE W

Jason & Sarah Dasent: Making Music Tech Accessible A successful musician and producer who happens to be blind, Jason Dasent is helping manufacturers to implement text‑to‑speech and other vital accessibility features.

SAM INGLIS

“I

Jason Dasent and his wife Sarah have become the industry’s leading consultants on accessibility.

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make light of being blind, because it could be a heavy topic,” explains Jason Dasent.

“People tend to handle it in different ways. People tend to be embarrassed.

British people have a thing where they say, ‘I don’t mean to be rude...’ But the nicest people, and the people I have remained lifelong friends with, are the rude people. Ask whatever you want to ask. Nothing is off limits, you know. That’s how we’re gonna break down the barriers to accessibility. I think I’m offence-proof!

I had to make myself that way, because I need to make people comfortable.”

As well as developing a career as a producer, songwriter and keyboard player, Jason has also found himself becoming the industry’s leading adviser on accessibility, working with almost all the major hardware and software manufacturers to help them make their products usable by vision-impaired people. It is, as he acknowledges, quite a responsibility, especially as there often aren’t existing standards or conventions to guide the implementation of accessibility features.

“I’m inventing them!” he admits, “and it’s a really scary thing. Sometimes I’ll stop and think and I’ll just freeze and say, ‘I hope I’m giving people good advice.

I hope I’m not leading a whole industry down the road of nonsense, my nonsense.’

But then you see a product become accessible and say, ‘OK, maybe it’s a good thing.’ But it’s really exciting, like when you have an idea and you discuss it with a software engineer and they come back to you next week: Jay, check this out.

Jason’s studio setup is largely keyboard-based, and includes many of the products that he has helped to make accessible.

“I’m sure there’s lots of things that people just don’t think about. If it doesn’t affect you directly, then you wouldn’t know. You have a good Samaritan situation where, for example, you may have a professor out of college and a blind student comes in. The professor has never heard anything about accessible music tech. So in the goodness of his heart, he would stay back in the afternoon after school is done with the blind student and he would say, let me put a piece of tape on the number five button, right? Which is all well and good, but that’s no longer necessary. Just because I’m blind, I don’t want my equipment to be big and clunky.

Just make the technology available. Give me a sexy touchscreen that talks!

“I always say: make it possible, don’t make it easy. I want the same experience as a sighted person. I want the same learning curve.”

Memory Man Jason’s own learning curve began early.

“I was in primary school, and another student had a little keyboard. I picked it up, played a few notes and it actually sounded like music. It was close to Christmas, so guess what I got for Christmas? A little Casio keyboard. I was about nine or 10.

My family kind of realised that I was a bit serious about it, you know, maybe there was something there. So my brothers and sisters and all my working siblings and everybody bought me my first synth, a Casio CZ-101. And my brother Dane was not musical, but he was technical. He started teaching me how to program and do sound design and make little patches and stuff. One thing led to the next, and I started becoming interested in drum machines and early sequencers, the Roland MC‑50 and stuff like that.

“And there was no accessibility in those days. There was no speech. So if, for example, I wanted to bring up preset number five, I might have to do something like turn a dial 15 times. You just had to remember where things were, and I had to build a photographic memory of the layout. And then you would turn the dial 13 times and the phone would ring, or somebody would call you. I could give you lots of embarrassing moments, because I started playing with bands pretty early, from like 15 years old, and you’re doing a big gig and the crowd is hyped and ready to go, and then you hit start and nothing happens, because obviously I couldn’t see the screen, right? But anyway, I guess it built character.

“So at some point along the line, computers started to talk. At least, it became affordable to get a talking computer, and things became a little bit easier, one step at a time. I started using Cakewalk and Sonar back in the day. And then Pro Tools became accessible. So one thing led to another, and I was able to build a pretty fulfilling career, doing both studio and live and even film and post and stuff.”

Jason decided that the next step in developing his musical career should be to study at college in England. He and his wife Sarah duly crossed the Atlantic, only to find that the higher education sector was really not set up for his needs. “A lot of times, people need help in terms of getting the message that accessibility in teaching and in schools and in universities and stuff like that is really important,” says Sarah. “Dealing with a visually impaired person is different from someone having a wheelchair, because at least they can see. When somebody can’t see, it’s a different thing, especially for us sighted people. We just don’t think about it, you know, and it’s something that you need to know. For example, one of the most basic things to do when a blind person comes into your programme is to just get them a layout of everything. And sometimes you

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just don’t know where to start, but that’s like the starting point: a layout of the space and then a layout of each unit.”

Maschine Learning At this time, buying a new item of music technology was often an expensive gamble for Jason, because there was often no way of knowing whether he would be able to use it. “Sarah and I purchased a Native Instruments Maschine. We were in Trinidad, thousands of miles away. We had everything shipped out to us, plugged it into the computer, installed the software, opened it up, only to find out we might as well have bought a doorstop, because it was totally inaccessible. It was like a blank screen to me, because it didn’t support text to speech. So it was either sink or swim. We could have sold it and admitted defeat, but I have a little bit of programming knowledge, using a program called Keyboard Maestro. I thought, ‘OK, maybe we can build some macros. Sarah, can you help me build some macros?’ That would do something like: you execute the keyboard shortcut, the mouse moves to the Play button and clicks it.

“So we started working. Sarah mapped the Play button. It worked. So we then did Stop, it worked. How about we get adventurous: let’s map Record! And slowly but surely we chipped away, along with a colleague called John de la Bastide, and in about a year, we had like 600 macros done to basically control the entire program. Native Instruments got wind of it and invited us to their office in Hackney, and we went and we showed them what we did. Long story short, they were impressed and they hired us to work with them on making it accessible from the back end.

Cool. So it’s now fully accessible.

“While we were on that trip, Focusrite heard about us and they called us and said, ‘Look, we’re doing the Vocaster and the new Scarlett’s coming.

We want you to work with us on making the Focusrite stuff accessible.’ Then it started to spread. Ableton called us and we worked for three years with them to make Ableton Live fully accessible. Sarah was working with me to start with, and then I was able to take it on my own. So we were able to break down that barrier

Since demonstrating that the Maschine (centre, right) could be made accessible, Jason Dasent has worked with NI to achieve the same goal with numerous other products. He has likewise worked with Ableton and Focusrite on products like Live/Push and the Novation Circuit (centre).

because a lot of people had reservations about hiring a blind person. Then Audio Modeling got in touch, then Arturia and SSL, and most recently Rhodes, and we worked with Softube to make the Console 1 and Console 1 Fader the first 100 percent accessible control surfaces.

Slowly but surely, I think a barrier is being broken down.”

document. And then, because he has such an amazing memory, after a time, he remembers the layout and it makes it really easy for him to do his testing.”

“The concepts are the same, but it’s sometimes different, depending on what framework they use,” adds Jason. “Usually I don’t need to worry about that too much because that’s the engineering team’s job. A lot of stuff is built on JUCE, and that’s pretty standard, but Softube for example have their own protocol. So it would be a little bit different, but the goal is the same: what the software does, what the hardware does, what you need it to do, what you need to see. But I need to say, let’s make this talk. When you hit Record, say ‘record on’, things like that. Sometimes people attempt to do it on their own, without the experience of somebody vision impaired. They may be well-intentioned, but if you click a Record button, they might have it say something

Jason Dasent: “Everyone who’s blind, mummy, daddy, auntie, uncle, says they need to do music.

Why do they need to do music? Because Stevie Wonder is blind and Stevie Wonder did music.”

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“When we get a piece of gear from some company that wants to become accessible, I will sit with Jason and label everything for him in a Word document,” explains Sarah. “We just go from row to row, going down. You know, first button is the scroll bars or whatever, and then we would just go right across the keyboard or drum machine, whatever it is, row by row, and just put it into a Word

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INTER VIE W

MAKING MUSIC TECH ACCESSIBLE

Taking electronic music gear on stage has always been a particular challenge for blind musicians like Jason, because of the added potential for things to go wrong.

like, ‘This is a Record button. If you press it the record function comes up.’ But all I want to hear is ‘record on’.”

Not Special A point that Jason repeatedly emphasises is that accessibility isn’t about making it artificially easy for visually impaired people to make music. Rather, it’s about removing additional barriers, so the blind experience the same challenges as everyone else, and enjoy the same opportunities.

“You come across a lot of insecurity among visually impaired people. They would do something at home, but they wouldn’t go apply for a job. And where I grew up, in Trinidad, there was no infrastructure for visually impaired people already. It worked out for me because my mum said: ‘You know, no matter what, you have to grow up and live as a normal person. There’s nothing special about you because you’re blind.’ And that’s the best gift I’ve ever been given. ’Cause I had to go out there. I had to go and sit in front of 15 execs at an advertising agency and say, ‘I’m blind but I’m gonna score your next commercial, right?’ And it made me strong.

It made me able to become who I am.

“A lot of people are buying the equipment just to use it in their bedroom, but I’m applying for jobs. People get a lot of grants and stuff like that so they can live comfortably without working, you know, but that’s not the situation for me. You work to make a living, right? Not because you’re blind or disabled. That’s that’s what life is about. That’s what makes life worth living.

And I’m trying to kind of instil that culture

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without changing people’s beliefs. I’ve worked for everything that I have, and I’ll have it no other way. The technology is there, it’s just getting it out there to the people.”

Meanwhile, for everyone whose expectations of blind people are too low, there are others who make the opposite assumption. “That leads me to the next thing, which I call Stevie Wonder Syndrome. Everyone who’s blind, mummy, daddy, auntie, uncle, says they need to do music. Why do they need to do music? Because Stevie Wonder is blind and Stevie Wonder did music. So I have situations where there’s somebody who I teach, who is extremely intelligent, but he’s not cut out for music. But his circle, his parents, and whoever is around him, insists he must be into music because he’s blind and Stevie Wonder is blind.

And that’s not an isolated case. I can’t teach him music if music is not for him.”

Hearts & Minds Making music tech equipment truly accessible can be a big job, which ties up precious development resources. At the same time, it’s rarely glamorous, and doesn’t always offer obvious commercial benefits. Nevertheless, Jason Dasent has found that manufacturers are often very willing to invest in accessibility once he has demonstrated the possibilities of the technology. “Sarah and I always have a thing we say: inspiration, not obligation.

Nobody’s obligated to do anything for you, but they can be inspired. If it’s an obligation, they’ll do the minimum just to get over

the line so that they’re not doing anything illegal. But if they do it by inspiration, it’s going to be done out of love and passion, you know? I don’t know if that’s the only way, but that’s the way that I work.

“I’m often sent petitions by forums and stuff. Would you please sign this petition to say that the government needs to make it mandatory? Things need to be accessible by law, right? I’m not going to sign that petition, because if you do that, you turn people off. If somebody tells me I have to do something, I’m probably not going to do it. What I would do is go to events and talk to manufacturers. And my thing is, ‘How about if we work together to make it accessible? Let’s do it together. Let’s have some fun!’ And that approach works 99 percent of the time.

“I’m learning these things, and working with people and inspiring people.

Some people are more inspired than others. It’s about choosing who you want to work with, what kind of work you want to do with them, how you bring people and manufacturers together. But my thing is I want a situation where visually impaired people are employable easily.

I wanna make sure that that technology is there so it can do things that can reduce the rate of unemployment. And so now we have the fully accessible studio. I can do everything from recording to mixing and mastering. The thing is just getting more brands and different companies on board so that there’s choice. So now there are a few things that are accessible in all the different categories. But what we aim for is everything to be accessible in every category.”

Watching Jason Dasent at work in his fully accessible studio makes clear how text-to-speech has revolutionised music technology. Both Pro Tools and the other software tools he works with are adapted to talk back, reporting the function of whatever control is currently in focus.

Working from a carefully constructed and consistent template, it quickly becomes obvious that the only thing slowing him down is having to explain to me what’s going on. It’s also instantly apparent that Jason’s abilities as composer, keyboard player and arranger are top-notch. In the space of half an hour or so, he has put together a more or less complete soul track, working both with software instruments and with hybrid devices like the NI Maschine. Accessibility is simply the key that opens the door to allow talent and musical ability to flourish.

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TALKBACK

Aron Kobayashi Ritch WILLIAM STOKES

L

os Angeles-based Aron Kobayashi Ritch first fell in love with production in a friend’s father’s garage studio, recording local bands while skipping school to assist at the legendary Sound City Studios with engineer David Boucher.

It was in this period that he met Momma, the rock band he would come to join and whose records he would come to produce, not least latest LP Welcome To My Blue Sky. At the moment I can’t stop listening to I’ve been listening a lot to the first two Cranberries records. That was a band where I always knew the hits, but I never fully understood why they were important. They’re just so fucking good! The stuff that Steven Street was doing on them is pretty mind-blowing.

I’ve always been a fan of his, but yeah, those records are amazing. I also have a constant love affair with the band Frou Frou. They only have one album [Details], that’s it! But I think that is such a landmark for me, in terms of sound and songwriting and production. What Imogen Heap and Guy Sigsworth were doing on that record is so insane. I’ve always loved that record. And lastly, I’ve revisited the first Third Eye Blind record. Growing up in that era, the late ’90s and early 2000s, a lot of the music I make sort of dips into that. I like records that had radio pop appeal, but like, were still ‘indie records’, or at least had the stylings of alternative music. It’s such a feat to combine the two and not make it incredibly cringey. That’s a fine line to toe, and it’s something we think a lot about in my band, too.

The artist I’d most like to record It’s hard for me to take artists who I love, whose records I think are amazing, and to think about how I would somehow

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alter their career. Like, yeah, I’d love to work with Radiohead, but would I really do a better job than Nigel Godrich?

Probably not. But the reality is I enjoy the process of finding an artist’s sound, so the prospect of working with someone that I’ve looked up to is sort of like, “Well, what would I do with this?” I do think it would be really cool to work with Weezer. We toured with them a long time ago, and they’re one of my favourite bands, but mostly it’s because I have a specific idea of what they should sound like. They had an aesthetic in the ’90s that I think they kind of stumbled upon, which was this very dry, honest representation of their music that I think gets lost when you get popular. This is a debate that’s raged on for decades with Weezer fans.

The first thing I look for in a studio I think a really sizeable live room is pretty important to me. I make a lot of records with bands, and with bands I prefer to start from a place of like, “How would you guys play this all together?” So, having a room where they can all be together and look at each other, with enough places for isolation, and to get a drum sound that has a lot of natural ambience, all those things are pretty important to me.

And then: Neve! I’m pretty Neve-focused.

A Neve desk, Neve pres: those are pretty important to me too, to how I think about starting a record. I don’t do much to tape anymore, and Neve tends to get me closer there just because the transient response is a little softer than something like an API — which I do also love.

The person I would consider my mentor Probably David Boucher. While I can’t say I learned everything I know during the years I worked with him, I was really young and it was just about experiencing what an actual workflow is and what a professional scenario is, and also what the lifestyle is like.

Like, “OK, this is how you make a living doing this.” That was pretty important for me coming up, and I got to do a lot of really cool projects, for instance the Andrew Bird record that I sat in on was just really amazing to watch. He does a lot of orchestral stuff for Disney, also.

So, in my senior year of high school, I would skip school to go and assist and take notes and put up mics for, like, the Moana soundtrack! It was very strange to be there, and very cool of him to bring me along. That’s him putting his neck out in this very professional scenario. “Yeah, I’m going to bring along a kid to watch, and he’s going to skip school to do it! Is that cool?” And everyone’s like, “Yeah, sure!” My dad always said, “Don’t let school get in the way of your education,” so I think he really liked it when I told him I had done that. It was cool to start my career from more an engineering perspective. It’s not really what I get hired to do now, but having that basis was really helpful to me thinking about how I want a record to sound.

My go-to reference track or album There’s sort of two. One that I used for a long time was a song from that Andrew Bird record! Tchad Blake mixed it and I love his mixes. There’s a song called ‘Capsized’, and I would listen to that a lot as a low‑end reference for things, because it has a really nice presence to it. It wasn’t incredibly subby, but he incorporated a lot of low end into rock records, like he did with Arctic Monkeys, or the Black Keys. There’s like an extra octave that people weren’t even touching before. So that’s one. And then, honestly, that Third Eye Blind record! Eric Valentine is an incredible mixer. I think he is really good at compartmentalising things. That’s something I look for a lot.

He’s really good at things being… not harsh. So, that one for the clarity of it, and then the Andrew Bird for the low end.

The producer I’d most like to work with Probably Guy Sigsworth. I just think he’s a genius — and is completely different to me. I don’t think he approaches records from a band perspective like I do. He seems very digital-focused, like, the way he constructs his productions. He did a Madonna song that I’m obsessed with, too, called ‘What It Feels Like For A Girl’. He did it right before Frou Frou, I think. But it sounds like he just made a Frou Frou song and put Madonna on it — like, you can’t go wrong! He just does stuff where I’m like, “What are you doing?” These insane sounds and the way he transitions between sections...

He’s a very interesting person to me, so it would be awesome just to meet him one day. I would love him to do a remix of a song for my band. I think that would be really cool.

My secret weapon in the studio is I use the [Soundtoys] Tremolator plug-in a lot. I’ve never been a synth person; I don’t have anything against them, it’s just not the language I speak. But I do a lot of stuff where it’s about getting ambience out of other instruments, like guitars and stuff like that, and having a way to control them. Tremolator allows me to get a really cool combination: you get to have this unruly, atmospheric thing, and then to control it through the beat of the song, almost like side-chaining it.

Cymbals are also really important to me. I’m not a drummer. No one should put me behind a kit. But I started collecting cymbals that I think sound good and that I know are good, and I bring them to almost every session that I do, no matter what. Nothing’s worse than bad cymbals. I have a I have a 22-inch [Zildjian K Constantinople] that I bought a few years ago, and that’s been on almost every record I’ve made for the last three years, because that’s what I hear from a ride cymbal.

The studio session I wish I’d witnessed I think, probably Radiohead making Kid A. That record confounds me. I listen to it and I have no clue how they got to here, because it’s just so unworldly. And to go from OK Computer to Kid A is such an insane transition. What I imagine happened is a lot of deconstruction, like, recording the song and then taking bits away, redoing bits. You sort of start in one place, which feels like a band, and then it sort of just morphs into something else because you find sounds along the way that become the crux of it. That’s how my brain thinks about it. But to be there and witness how they did that, I think that would be really amazing. There’s no way they got to this without just being like, “Let’s fuck around for like, a week, and then we’ll cut away the fat.” It’s the brilliance of knowing what to cut away. I think that’s the hard part, which they have down. There’s not that much going on, but also there is. They have chosen the sounds so specifically, there might only be three things happening at once, but it’s somehow so dense.

The studio experience that taught me the most I think, producing the Momma record Household Name. Making a record where there’s a label budget to be considered, and because there’s a lot riding on it, you have to plan how you are going to manage everyone’s time and get the most out of the money and the people. That taught me a lot, and a lot of that carried over into records I’ve made after. We made it in a studio I work out of a lot in Brooklyn called Studio G. It’s a great place. It’s just a huge live room, an amazing Neve console in the B room, and then the A room has an SSL. A lot of studios in New York have basically sold off all their outboard gear.

But at Studio G, you can show up with nothing and just make a record.

The advice I’d give myself of 10 years ago Enjoy the ride! There are a lot of stresses involved in this industry, particularly when you’re also in a band. I think that has its own difficulties; you’re basically managing two very different hectic schedules.

It’s just very complicated. But I think that if you just learn to enjoy the moments that are fun, that’s really important. You have to find the joy, and you have to remember why you’re doing it, because that’s the joyful part.

If you’re doing it for a living — and working more hours than people should — then you have to enjoy it.

www.soundonsound.com / June 2025

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ON TE ST

Harrison D510r

500-series Chassis

With up to 10 module slots on board, and both Dante converter and mixer expansion cards available, could this rack widen the appeal of the 500-series format?

M AT T H O U G H TO N

T

he D510r is a 500-series chassis with a difference. It can, if you wish, be used pretty much like any other 10-slot chassis: you can pass analogue signals into and out of the modules hosted within by using the rear-panel XLRs. But this rack also has expansion options.

Currently, there’s a Dante interface card and a mixer/headphone amp, but also firmly in Harrison’s plans (the launch date is yet to be announced) is a USB and ADAT converter card. In combination, I reckon these expansions mean this rack has more potential applications than almost any other, including for large-scale use of networked 500-series modules in a live sound context, or in analogue-digital hybrid studios as an expandable computer audio interface system.

Overview Sent for review was a D510r fitted with the optional D510mx Mixer Interface and D510Dante card. As you can see from the photos, it also came loaded with Harrison’s own 500-series preamp, EQ and compressor modules, but while I was reminded that there’s plenty to commend those, we reviewed them in SOS October 2024 (https://sosm.ag/harrison-500-series) so I won’t revisit them here.

The whole ensemble is housed in a 3U, 19-inch rackmount steel box that extends

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85mm (7.25 inches) behind the rack ears (these form part of the enclosure so can’t be removed). While there are plenty of ventilation holes on the top, bottom, sides and rear of that box, these don’t compromise the impressive robustness of this chassis. Top marks for that. Physically fitting and removing modules was just as I’d expected, given my experience of many other 500 series chassis, which is to say that it’s usually a breeze but seating a module in that final slot is always a fiddle! There aren’t many controls on the standard rack, but there are plenty on the Mixer Interface card, of which more below. The need to pack so many knobs in such a small place has left that feeling a tad cramped — not inoperable by any means but, for example, when turning a headphone level knob my thumb always brushed against the adjacent controls.

Viewed from the front, module slots 1-10 are laid out from left to right, and to their right is a faceplate, the standard version of which carries the brand name and model number, and has four buttons with some indicator LEDs along the bottom. A red LED indicates power on/off, and the other buttons and LEDs are currently “reserved for future functions”. These appear to include the ability to select the sample rate and clock settings manually for an installed converter card, and to indicate the status of USB, ADAT and clock connections (during the review period, I could only set

the sample rate remotely, using the Dante Controller app).

On the rear, the 10 pairs of XLRs carry analogue audio into and out of the rack (balancing is performed by the 500-series modules themselves). To the left, an IEC inlet with on/off switch feeds mains power (100-240 V, 50/60 Hz) to a built-in switch‑mode power supply.

The 10 500-series slots share up to 1700mA of current, the average of which is comfortably (40mA) above the VPR Alliance spec. The Dante card and planned USB/ADAT card fit into a slot in the lower portion of the rear panel, and on the Dante

Harrison D510r £699 pros • Robust construction.

• Mixer option, with dual headphone amps and cue-mix facilities.

• Optional bidirectional Dante converter card.

• USB and ADAT connectivity in development.

cons • Only internal means of selecting the module source.

summary The D510r is an impressive 500-series rack, with equally impressive expansion options — not least the multi-channel Dante A-D/D-A conversion that really sets it apart from the crowd.

card’s own panel are two RJ45 connectors, for Primary and Secondary Ethernet connections, as well as a BNC word clock connector and a recessed Reset control (that I never had cause to use).

The Mixer Interface card occupies channels 9 and 10, and extends to replace the standard faceplate. The XLR analogue inputs 9 and 10 feed this 10:2 summing mixer directly, and it takes the other eight inputs from the first eight module slots. The stereo mix is routed in parallel to the main mix out (channels 9 and 10), whose large Main level knob has unity marked around the one or two o’clock position, plus two front-panel quarter-inch TRS headphone jacks, each with a smaller level pot. The headphone outs also have a Blend knob, to set the balance between the stereo mix and a separate stereo feed for each headphone amp that can be sent from a computer or other Dante device when a converter card is fitted.

Mixer channels 1-10 each have a level and pan pot, and a red button with red indicator LED to route that channel to the mix bus (or not, as you prefer). These controls are arranged in two columns, with odd-numbered channels on the left and evens on the right, such that each row controls the five obvious stereo pairs (1+2, 3+4 etc.). Beneath this are the same buttons/LEDs as on the standard card, the only difference being that the Mute button lights at power up to indicate a lack of connection of the Dante interface to a network; at the time of writing this feature was still in development.

As with most 500 racks, a large, main PCB is mounted inside the back panel, and its most obvious features are the 10 card-edge connectors. Adjacent to each of these, though, are two switches...

The first switch determines whether that slot receives its analogue input from the rear-panel XLR or from the D-A converter. This gives you the welcome flexibility to, for example, route from a DAW to a multi‑module channel strip and back, using only a single Dante input and output, avoiding the need for several stages of conversion and the cumulative latency that would entail. Less welcome is that these physical switches on the inside are the only way to change this setting — if you

The other ‘switch’ is a Stereo Link jumper, which uses (presumably) pin 6 of the 500-series connector to sum the control signal of a dynamics module installed in an odd-numbered channel with that of one in the even-numbered channel to its right. Again, a welcome feature, but one that’s inconveniently placed for those wanting to change this setting after the initial setup, and there’s no external visual feedback on the current status.

Dante When fitted to the D510r, the D510Dante interface can carry up to 14 channels to the rack and 10 channels from it, at your choice of 44.1, 48, 88.2 or 96 KHz sample rates, over a Dante network. The ‘additional’ four sends to the unit are the PC 1+2 and PC 3+4 signals that I alluded to above and which, with the Mixer Interface fitted, you can use to route DAW mixes or backing tracks/cue mixes to the headphone amps. All the inputs and outputs are combined by an Ethernet switch fitted inside the D510r so that it appears as a single device on the Dante network, with a primary/secondary redundant configuration.

There are other 500-series racks out there with A-D and D-A conversion built in, Cranborne Audio’s 500R8 interface and 500ADAT expander being the most prominent examples. But I can’t think of one that also offers Dante connectivity, and the potential scale of setup into which you can integrate one or more D510rs without requiring separate converters or lots of analogue cabling is significant.

For instance, it means that you could create a network of several of these racks in a studio setting, with each and

“The headphone outs have a Blend knob, to set the balance between the stereo mix and a separate stereo feed for each headphone amp.”

Internal Options With no 500-series modules fitted, much of the internals are open to view, and some of what can be seen deserves our attention.

wish to change it after the initial setup, then you must power off the rack, remove a module and use a ‘long pokey thing’ (in my case, a wooden chopstick!) to push up a latching button. A task best done while shining a torch inside, since there’s no external indication of the status. In most traditional Dante network setups this is unlikely to be problematic, as you’ll likely configure the device once and leave it that way. But those, like me, working in a music production environment would be justified in wishing for more flexibility. I’d have preferred to see external switches (there’s space on the standard faceplate) or maybe remote-switching options, such as using MIDI over MADI.

The Dante interface occupies the lower half of the review unit’s rear panel. An alternative USB/ADAT interface is also planned.

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The optional Mixer Interface card, with individual level and pan controls for each channel, sends a stereo mix to XLR outputs 9 and 10, as well as to two headphone amps.

every channel directly accessible from your main DAW machine. Even though there’s a USB interface card on the horizon, a Dante setup could have some advantages in that context too, whether simply the reduction in cabling required around the back of your rack, or the ability to place racks of headphone amps, cue-mix facilities and 500-series preamps, processors and utility modules further away from a computer than USB typically allows.

I could imagine that appealing to the sort of shared-live-room, multi-control‑room setups that you find in higher education facilities. Alternatively, a single D510r rack with the Dante Virtual Soundcard software gives you an eight‑ or 10-slot 500-series audio interface that’s expandable and so, to some extent at least, future-proof.

But while we’re increasingly seeing Dante and other networked audio in music studios, it’s already widely used in live sound, and there are definitely applications for bands and engineers wanting to take high-quality analogue outboard out on tour.

A rack like this, that’s relatively compact compared with 19-inch processors and

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effects and can integrate with existing Dante setups without external boxes, could make a lot of sense.

I’ve used Dante devices previously, but this was my first time setting up a ‘network’ from scratch, and connecting the unit directly to my MacBook Pro (rather than a router) running Dante Virtual Soundcard initially caused some head scratching.

My computer has access to multiple networks, and it took a while to figure out the settings needed to get the computer and the rack talking to each other (I didn’t find Audinate’s Dante troubleshooting documentation the most accessible!).

But having cleared that hurdle and, in Dante Controller, set the D510r to the same sample rate as my DAW project, I could easily route audio from the DAW to the rack and back with impressively low latency. Setting up headphone cue mixes and balancing them with an analogue foldback was also a breeze.

Sound & Specs Subjectively, the D510r sounds good, and I’d characterise it as ‘clean’. I wouldn’t

expect the module slots themselves to add anything untoward, as there’s very little going on there until you plug a module in, but I’d include the mixer and headphone amps in that description too. I noticed no problems in terms of undue noise or distortion, and I found that my headphones of various impedances could be driven louder than I needed. It’s perhaps worth noting that in the ‘off’ position the headphone amps still pass a faint signal, but it’s not loud enough to leak onto a mic or be a bother.

The published specs perhaps don’t put the D510Dante converter card up there in ‘best-in-class’ territory, but it’s my sincere belief that people massively overthink converter specs today — for the vast majority of applications, converter quality just isn’t the issue it once was, and it’s unlikely ever to be the weakest link in your chain. In this case, the converters are perfectly capable of professional-quality results, offering plenty of dynamic range for the sort of signals that will be run through this rack and, importantly, I couldn’t hear any problems. In short, I’d have no qualms about using this box with the Dante card for recording, mixing or live sound.

Neither did I detect any of the problems I’ve observed in some host racks with the built-in power supply compromising the performance of nearby preamp modules.

Verdict So, Harrison’s D510r is a welcome addition to the 500 series. It brings convenient Dante conversion to the format for the first time. As well as being a 10-slot rack, this device has the potential, with the right expansion cards, to be a summing mixer, an audio interface, to deliver two headphone mixes... the list goes on, and when the USB and ADAT card becomes available it will lengthen further. Sure, there are some things that I’d like to see done differently, not least those internal-access switches. But there’s scope for future development here, and the convenience of having the converters built into your 500 rack rather than relying on standalone units should not be understated. Definitely one to put on the audition list.

£ D510r 500-series chassis £699, D510mx

Mixer Interface card £549, D510Dante converter card £749, D510usb USB/ADAT card price TBA. Prices include VAT. T Harrison Audio +1 (615) 768 5501 E info@harrisonaudio.com W harrisonaudio.com

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ON TE ST

Boss Katana:Go Modelling Headphone Amp For Guitar & Bass Boss’ Katana:Go has been overhauled, and is now more compact, sleek and portable than the original. It plugs into your guitar or bass jack output, and can not only deliver a virtual amped guitar/bass sound to your headphones, but also has a USB port that allows it to act as a computer audio interface. There’s Bluetooth connectivity too, to allow playback of backing tracks and communication with the Boss Tone Studio app for iOS and Android.

The unit has an internal rechargeable battery but can be powered directly over USB. The provided USB-C to USB-A cable is long enough that you can sit by a computer and record but, as with most USB cables, not long enough that you have the freedom to move around with your instrument.

Normally, then, you’ll want to make sure the Katana:Go is charged, and an LED that indicates various power-related things helpfully alerts you when the remaining battery life is 30 minutes or less. Another reason to run off the internal battery is to avoid noise, which proved a problem when connecting over USB to my MacBook Pro when the computer’s mains adaptor was also connected. Disconnecting either the USB cable (which of course prevents recording) or the computer’s power supply side-stepped that issue.

The headphone output is a TRS stereo mini‑jack and I had plenty of level through my various headphones. At the other end, a quarter-inch jack plug fits directly into your guitar’s jack socket. The jack’s hinged joint makes it easy to connect the Katana:Go to pretty much any guitar and allows the device to swivel with the headphone cable when you move, preventing undue strain.

It also makes it easy to move the device into the optimum position for operating the controls or viewing the screen. Speaking of which, the screen is bright and easily viewed, and while the handful of user controls is ‘hidden’ on the side panels, you soon develop an instinct for where they are.

On one side, there’s a volume thumb-wheel, an on/off slide switch, a Bluetooth button and a Stage Feel button. The purpose of the others should

be obvious, but we’ll come back to that last control later. On the opposite panel a momentary lever switch scrolls in both directions through the preset numbers (1-10), and doubles as a button to engage the monophonic but auto-ranging tuner. Three adjacent buttons select the preset ‘channel’ (Boss’ term for a bank) A, B or C, so you have access to 30 onboard presets in total.

What presets are available depends on the mode you’re in. The default is Guitar mode but powering up while holding down the Bluetooth button lets you change to Bass mode for the next power-up. The default preset in Guitar mode, 1A Brown Lead, delivers a compressed blues-rock crunch with some reverb and delay; a decent enough sound but as with many of the factory default ones, there’s a touch of ‘preset zero’ about it. The others have vaguely descriptive names, and feature effects like chorus, reverb, delay, harmonies, ring‑modulation and phasing. You get virtually no control on the hardware over these presets on the unit — you can’t even turn off any of the effects, never mind tweak the parameters — and I’d have liked more by way of cleaner tonal options for guitar by default. But if you like what the presets do, then the overall quality of the sounds is generally good, and I didn’t perceive any latency over headphones, making the playing feel decent when I had the guitar’s volume control in the preset’s sweet spot.

I was more impressed by the factory presets for bass, as I felt there were more instantly recordable sounds. These include

“The jack’s hinged joint... makes it easy to move the device into the optimum position for operating the controls or viewing the screen.”

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clean patches, compressed pop sounds, crunch, fuzz, chorus, phaser, pitch-dropped dub effects, presets for slap, fretless and acoustic, and even organ bass and bass synths. The final preset (10C) is a quirky Slow effect that fades the level up for each note played and chops the signal rhythmically as it does so.

It’s worth noting that one of the effects is a threshold-based noise gate. This can be useful but, without onboard control over that threshold, I found some default presets truncated quieter note tails.

I also found it frustrating that there’s no hardware control over the tempo/rate for effects such as tremolo and delay; I’d have loved to find tap tempo somewhere as a secondary button function.

But we don’t need to dwell on all that negativity, because you have plenty of control when connected to the Boss Tone Studio app over Bluetooth. Here, you can change the amp model as well as the drive and volume settings (so you can adjust the breakup point to suit your instrument). You can switch effects on/off, access their key parameters and position them in the signal chain. Changes are almost instant and can be saved over any of the 30 memory slots. You can also connect to Boss’ free Tone Exchange, to upload your preset collections or download those made by others. So there’s plenty of scope to save patches that suit your instruments and playing style, and access these using the hardware’s preset selector.

Bluetooth can also be used to for MIDI control, and Boss themselves offer a wireless switch and pedal. Frustratingly, though, I couldn’t find the MIDI parameter documentation needed to discover the full possibilities with other devices. For practice, the Bluetooth audio support also allows you

Though they’re not easily visible when plugged into a guitar, you soon develop a feel for which controls are where.

to stream a backing track to the Katana:Go, and in this scenario you can try the Stage Feel function. This uses binaural technology to place the backing track and your instrument in different positions, with you in the centre. For example, it can attempt to put your instrument in front of you, as if monitoring through a stage wedge, while shifting the backing track behind you and to the right, as if you were standing front-left of the rest of the band. It’s clever, and some might find it useful to simulate that live-on-stage situation, or simply for better separation between their instrument and the backing track. But honestly? I almost always preferred to switch it off, since the instrument always seemed to sound better to me that way — and if you start out thinking the presets all sound a bit ‘mushy’, then I’d suggest turning this off and giving them another listen.

I tested the Katana:Go’s audio interface functions with my Reaper DAW software, and both recording and playback worked well. Note, though, that it always captures the processed sound. I couldn’t find an option to capture a clean ‘DI feed’ either, but I’m probably more fussy about my guitar tones than the bass ones, and certainly I’d be happy using the bass sounds I captured in this way in a mix.

In summary, this tiny package packs a lot of punch. The factory presets perhaps didn’t rock my world, and I might have liked more hardware controls, but the sound quality is good and when you start to explore the control app you’ll find the Katana:Go capable of a range of good sounds. The price is right too! Matt Houghton £ £109.99 including VAT. W www.boss.info/uk

Heritage Audio Synth Buddy Passive Source/Destination Selector The Heritage Audio Synth Buddy uses only passive circuitry, and that means that it can be used with all manner of unbalanced audio signals, including synths and guitars.

It also means that it can operate in either direction — so you can select which of up to 10 mono or stereo sources are routed to a single destination, or route one instrument to your choice of up to 10 destinations. For example, I was able to sit it on my guitar pedalboard and use it to route the main output signal to the various different amps and amp emulation hardware that’s set up in my studio. I was also able to use it to feed different sources to my looper pedal.

The I/O are all on the back panel, laid out as vertically stacked pairs of TS sockets. The upper row is labelled as the left channel and the lower as right, though there’s nothing to stop you reversing that. The ‘output’, which could equally well be used as the single input, is in the middle, with five pairs for ‘the other end’ either side. On the top, you turn a large chicken‑head knob, which operates a switch

that changes the routing. Everything’s clearly labelled, so you’re never in any doubt about which numbered source is selected, or which physical jack leads for that matter. (That said, one change I’d like to see here is a ‘scribble strip’ or space at the top to accommodate user labelling.)

The switching itself makes a quiet acoustic click, which gives reassuring confirmation that you’ve changed position.

In terms of the impact of switching on the audio signal, it is almost silent — it never caused the sort of loud nasty pops that could make your heart skip a beat or damage a speaker, but there’s always a quiet ‘click’ when you change the signal path. As it is audible, it’s worth knowing about and, for example if planning to switch sources/destinations during rather than in between performances you’ll want to weigh up whether this is for you. Personally, I reckon it’s quiet enough that if I made the change on a beat few people listening would notice, but our personal mileages may vary on that point.

There are some other practical considerations. As with any passive system, long cables running to/from it could have an impact on tone, for

The I/O are on unbalanced TS jacks on the rear, with two rows to accommodate stereo signals (you can, of course, use just a single row for mono source-to-destination switching.

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example. And while the unit does look very neat on its own, remember that you can have as many as 22 cables trailing from the back. There are rubber feet that help to keep it in place on a desktop, but you’re going to want to consider cable dressing if you like to keep things tidy.

Personally, I didn’t find any of that a problem. In fact, the biggest challenge for me was removing the 16 screws and the main knob and the nut that holds the pot in place so I could see what was going on inside, which no user should need to do! For its main declared purpose — patching synth sources to an audio interface or other destination — the Synth Buddy works really well. For anyone with a smaller (eg. two-input) audio interface and a collection of guitars, pedals, synths and drum machines, then, the appeal should be obvious: you can leave everything hooked up to the Synth Buddy

Korneff Audio Shure Level-Loc Compressor Plug-in Released in the 1960s, Shure’s M62V Level-Loc Audio Level Controller was a mic-level device intended for use with public address systems. It had few controls and delivered no-nonsense, high-ratio compression for microphone users who, from an engineering perspective, weren’t expected to know what they were doing!

But in the decades that followed, engineers began to notice that this humble-looking box had a unique sound that could be used creatively in the studio. One famous proponent is Tchad Blake, who discovered the Level-Loc when working on the Tom Waits album Bone Machine and described it as a “life-changing” discovery. In the decades since, Blake has found the Level-Loc a willing accomplice for his distinct mixing style.

This new plug-in is the result of a collaboration between Blake and Korneff Audio, and is officially endorsed by Shure too. You can find out more about Tchad Blake’s use of the Level-Loc and Korneff’s approach to this plug-in in Sam Inglis’ interview-based article that appeared when the plug was announced at NAMM this year: https://sosm.ag/RecreatingLevelLoc. But here, I’ve put it to the test to see if it delivers on its promise of being an authentic

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A peep inside reveals neat but simple construction, with the large rotary switch attached to a single circuit board, to which the jack sockets are also soldered.

and simply turn the big chicken‑head knob to choose which source is routed to your interface, making it quick, easy and convenient to overdub different parts. No patching or mixer routing required, and you don’t need a bigger, more expensive interface with lots of I/O. Some might consider the headline price on the chunky side for what, at the end of the day, is a simple switch in a box: it retails for about the same price as some budget eight-channel

emulation with useful digital trimmings.

The original’s user controls are as simple as they come: an input control is paired with a ‘distance’ selector, which acts as a crude threshold. Other than a Level‑Loc/bypass switch, that’s it. The plug-in replicates these functionally and visually, but also lets you switch between three different modelled units — these devices have often been repaired and modded over the years, so have slightly different sonic characteristics. Models TB1 and TB2 are from Tchad Blake’s studio, while DK is modelled on plug-in designer Dan Korneff’s own. In use, switching between the units revealed very subtle tonal differences that were more audible when applying heavy compression. It certainly made me appreciate just how thorough Korneff Audio have been in remaining faithful to the spirit of these often quirky vintage devices.

In addition to the skeuomorphic GUI, a selection of controls lets you blend the sonic character of this traditionally heavy-handed tool into your mixes, without you having to perform additional routing

interfaces, for example, or double the price of some of the more affordable jack patchbays out there. But it allows you to handle more physical inputs than such an interface, and it’s more convenient than plugging and unplugging patch cords. The 22 jacks, good‑quality switch and robust steel enclosure account for a good chunk of the asking price, and, frankly, if I were to price up the time it would take me to make something like this I’d quickly conclude that it really wouldn’t be worth the bother. All in all then, I reckon this is a great idea, and it’s been executed in a way that could meet a lot of people’s needs.

Matt Houghton £ £185 including VAT. W https://heritageaudio.com

in your DAW or resort to third‑party routing plug-ins (as I’ve often had to do when using other Level‑Loc‑inspired plug‑ins). As well as the standard input and output level controls you can see in the main image, there are Pre and Post filters that, it’s important to note, filter the audio itself as opposed to the side-chain control signal. As you’ll see/hear if you explore the excellent presets provided

by Tchad Blake, these filters can be used in combination with the overall parallel wet/ dry control to access a surprisingly diverse amount tone-shaping options.

Korneff Audio are known for providing lots of ‘under the hood’ controls to fine-tune and customise what’s possible with a device, and in this case there are a number of tweakable controls, including

input and output impedance settings, as well as a few component-swap options like resistor and capacitor values, which can affect the release time of the compression circuit. If that weren’t enough, we can also mimic powering the unit using a battery that’s running low, something which apparently can affect the saturation and ‘bloom’ of the compression. I must admit, I tend not to get overly ‘tweaky’ with plug-ins and I found many of these additional settings to be subtle on the ear.

But I could hear small changes in tone and level, and if you like to experiment and set your gear ‘just so’, you could have a lot of fun playing with this section!

Many people will think of the Level-Loc as an aggressive ‘character compressor’ to be deployed when things need shaking up. This plug-in delivers that, and especially when used on drums you can quickly achieve that signature, pumping ‘smacky’ sound that can either be used as a blunt tool or carefully blended in to taste using the wet/dry control. What struck me about this plug-in, though, thanks in part to the excellent presets, is how it can be used as a much more subtle tone-shaping device used to add presence and shape to guitars, vocals, piano and... well, pretty much anything! Some of Tchad Blake’s presets pair aggressive compression with assertive use of the high‑ and low-pass filters to focus in the processing on elements in the midrange. Often the wet/ dry blend will only be set to 10-20 percent, but it still has the effect of making elements cut through the mix or just have a little more attitude. I used this technique many times over a couple of mixes I was working on during the review period, and it was instrumental in getting a couple of stubborn tracks to feel more exciting.

We have so much choice these days, and I’m sure we can all relate to the experience of thinking a new plug-in is the ‘bees knees’ only to then realise you’ve not used it for six months! I try to take that into account when reviewing, and I really don’t think this plug-in will end up being like that.

It offers me something different to the many, many other plug-ins I have to play with. You can throw it on a track or a bus and quickly make a call on whether it adds something helpful. If it’s working, you then have a load of options for fine-tuning and tweaking. In short, I’m impressed, and at the very least you should have a little fun trying out the free trial. Neil Rogers £ $149.99 W https://korneffaudio.com

TC Electronic Ditto 2 Looper Pedal

additions. As before, the instrument signal passes through at unity gain and the high input impedance provides a good match for electric guitars. The rotary control adjusts the playback volume of the loop without affecting the dry sound.

The user profile settings offer two useful alternative modes of operation, each with or without ‘LoopSnap’: conventional Ditto looping or SingleTap looping. LoopSnap helps avoid a problem that everyone who’s used a looper before will have experienced — slightly misjudging the loop end point.

LoopSnap uses intelligent beat tracking to correct the loop start and end points, so as long as the input isn’t too rhythmically complex, it could get you out of trouble.

The LoopSnap algorithm can move the start and/or stop point up to 150 milliseconds in either direction, based on timing information detected in the recorded loop.

SingleTap mode uses the footswitch to cycle round record, play and stop/delete, making for very simple operation if you don’t need to overdub loops. By default the LoopSnap function is turned on along with SingleTap looping in the user switch position, though this can be changed via the app. Though the mini manual doesn’t seem to mention it, the last loop is stored at power down in either looping mode.

In summary then, the Ditto 2 is a delightfully simple and affordable looper, designed to meet the needs of those who don’t need a Ditto Plus or something more esoteric. It’s compact and easy to use, and the sound quality is excellent.

Having the status LED change colour to show the current operating status is also very practical. The LoopSnap function works well for most conventional music as long as your timing is somewhere close, and the app is very easy to use if you need to switch the user mode from the default. In all, it’s a worthy update to TC’s popular looper.

Paul White

TC’s diminutive one-knob, one-switch Ditto looper has proven very popular, but TC still wanted to improve it — and indeed they have. The Ditto 2 has the same form factor and control layout, but the pedal now has a larger, redesigned footswitch for more positive control, as well as a new toggle switch that changes the operational mode from that in the original Ditto to a Ditto 2 user profile, as set up using a free smartphone/tablet app. A sprung position on the switch allows for Bluetooth pairing (hold up the switch until the status LED pulses blue and your pedal should link to the app), but USB-C can also be used for app connection and performing firmware updates. Pairing has to be initiated from within the app, which had me fooled at first — simply holding the pairing button and looking in your tablet/phone devices menu doesn’t show anything until the app is running and a TC device selected.

The output, which is mono, can be set to buffered or true bypass and there’s now 10 minutes of loop storage at 32-bit/44.1kHz resolution. A standard 9V pedal PSU capable of applying 100mA is required for operation, but not supplied.

The normal Ditto pedal mode goes Record > Play > Overdub > Play > Overdub, with a double‑tap to stop and a maximum loop length of five minutes. The LED changes colour to indicate the current status: white for clear, red for recording, green for playback and purple for overdub. In Stop mode the LED shows yellow and holding down the footswitch clears the memory, albeit playing a second or so of the stored loop in the process. To restore the last loop during playback, the footswitch can be held down. To clear a loop while it is playing involves double-tapping the footswitch, holding down the second A smartphone app allows tap — both very practical you to change the looping mode.

£ £89.99 including VAT. W www.tcelectronic.com

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Mackie ProFX10 Go

Mixer & Audio Interface

With its compact footprint and the option to run from a built-in battery, the ProFX10 Go is Mackie’s most portable mixer ever.

PAUL WHITE

T

he Mackie ProFX10 Go is a compact mixer that can also double as a 24-bit/192kHz USB recording interface, and which also incorporates bidirectional Bluetooth to allow audio streaming to and from a nearby mobile device. However, mobility is the USP here, with the ProFX10 Go having the unusual ability to run on battery power. A 12V power adaptor is also included, and the replaceable lithium battery is automatically recharged when this is connected. Overall weight is 2.7kg with the battery fitted, and the size is a rucksack-friendly 70 x 270 x 292mm.

The GB-10 rechargeable battery ships with the mixer as standard, and Mackie say that it provides for up to eight hours of use between charges. As well as recharging when the mixer is plugged into mains power, the battery can also be charged outside the mixer.

With the ProFX10 Go, Mackie have made it possible for phone calls to be relayed via Bluetooth with a Mix Minus feature so that the caller doesn’t hear

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themselves repeated, making the mixer suitable for live streaming and podcasting phone-ins. There are also digital effects, employing what Mackie term their GigFX+ processor, which provides reverbs, delays, chorus and so on. Effects presets can be saved and recalled for live performance.

There are three different USB recording modes. Standard mode records the full mix with effects, Loopback mode brings in computer audio along with the mix for streaming, and basic interface mode uses channels 1+2 as the inputs without recording the effects. A separate USB 3+4 return can be brought into channels 7+8 — useful for incorporating software instruments or other DAW audio. Used with my Mac, the mixer shows up as a class‑compliant 2-in/4-out interface.

A licence for Tracktion’s Waveform OEM recording software is included.

Channel Hopping As with most small mixers, there are rather fewer mic preamps than input channels — in this case one each for the first four mixer channels, which also benefit from TRS insert points and a ‘Level Set’ LED for

basic metering. Channels 1 and 2 feature one-knob compressors and use combi XLR/jack connectors, with Hi-Z switches allowing for instrument use with the jack inputs. Channels 3 and 4 have separate XLR and line input jacks. The mic preamps use Mackie’s proven Onyx design and offer up to 60dB of gain, with switchable

Mackie ProFX10 Go £465 pros • Low noise and distortion.

• Versatile USB interface features.

• Bluetooth for phone interviews and music streaming.

• Excellent effects.

cons • Battery powering comes at a cost.

• Knobs rather than faders for channel and master level control.

• No monitor send or PFL/solo controls.

summary With Bluetooth connectivity, USB interfacing, a powerful effects section and the option of running off-grid, the ProFX10 Go offers a huge amount of flexibility for content creators, DJs, podcasters and more.

100Hz low-cut filters and global 48V phantom power switching.

Channels 5+6 and 7+8 are stereo channels, but can operate in mono using the odd‑numbered inputs. As mentioned, channels 7+8 can also accommodate USB outputs 3+4, with selection made using a switch. Channels 9+10 handle the stereo Bluetooth input (this section includes a pairing button for connecting devices), though it can also take a stereo analogue feed from a mini-jack. All channels have a fixed‑frequency three-band EQ, a pan control and a level pot. Channels other than 9+10 include an effects send, but there are no monitor sends and no solo or PFL provisions. There are large mute buttons on all the input channels, the effects return and the main mix. A headphone output provides personal monitoring.

The master section is clearly set out, with separate strips for the main mix output and the effects section. Rotary level controls again take the place of faders, with further level controls for the control room and headphone outputs. A Blend control mixes the Control Room output with the USB 1+2 return, which can also be sent to the headphone/control room outputs if desired. LED metering is provided on

the main output. The main outputs are presented on both XLRs and quarter‑inch TRS jacks, with a further pair of jacks accommodating the control room out, one more for the headphones, and a final jack for connecting a footswitch for bypassing the effects. The effects send mix has its own output, too, so you can send to an external processor if required. However, there is no dedicated aux return, so you would need to redeploy one of the other inputs.

Effects The GigFX+ panel is where you control the effects, and I’ve saved this bit ’til last as it is quite sophisticated for a mixer of this size!

saved for instant recall, and the treatments on offer largely comprise useful delays, echoes and reverb-type effects (including a combined delay and reverb option), along with a few modulation effects. All have simple controls similar to what you might expect from a guitar pedal, and the quality of the effects is excellent; the reverbs are plenty good enough for studio use.

Finally, the GigFX+ panel is also where you’ll find a Rec button for stepping through the different recording modes.

In Use In my tests the mixer worked perfectly, both running from its PSU and from battery. The preamps are clean-sounding, which is what you’d expect from Mackie’s Onyx circuit, and the EQ is workmanlike and effective — with the mid frequency set at 2.5kHz and shelving filters at 80Hz and 12kHz, there’s plenty of scope to address most common tonal issues. The USB interfacing came to life as soon as I selected the interface in my Mac’s Audio Settings, and recording worked without a hitch.

There are many less costly small mixers available if you don’t need the battery power, and if your main application is live music mixing, you may want more mixer channels than the ProFX10 Go offers. But what you get here are additional features that make the ProFX10 Go effective for general-purpose mixing and also as an audio interface in situations where there is no mains power. It also holds appeal for desktop use as an audio interface, where its built-in effects and extra USB return could come in very useful. The Blend control and phone-friendly Bluetooth connectivity obviously lean towards streaming and podcasting, too. So, while the lack of PFL/solo buttons and monitor send controls, as well the use of knobs rather than faders, could be offputting for those wanting a traditional ‘front of house’ desk, the ProFX10 Go is actually an extremely versatile little mixer that should hold a special appeal to those doing live streaming and podcasts. It’s rugged, easy to use and delivers the high standard of sound quality that we’ve come to expect from Mackie.

“The ProFX10 Go is actually an extremely versatile little mixer that should hold a special appeal to those doing live streaming and podcasts.”

It benefits from a small but clear colour LCD which aids navigation and depicts the editable parameters (as virtual knobs) for the currently selected effect. A push/ turn encoder plus a ‘back’ button handle effects selection and parameter selection/ adjustment. Up so six effect presets can be

The ProFX10 Go can accommodate up to four microphones, with the first two channels able to accept instrument signals on combi jacks. There are two stereo USB returns, a high-quality digital effects section, a Bluetooth input and separate outputs for the main mix and control room/headphone out.

£ £465 including VAT. T Source Distribution +44 (0)20 8962 5080 E sales@sourcedistribution.co.uk W www.sourcedistribution.co.uk W www.mackie.com

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SPOTLIGHT

StudioDesk Dominator

Studio Workstation Desks LUKE WOOD

W

hen you’re looking at a new purchase for the studio, it’s easy to be tempted by things that are fun rather than practical — a new mic, synth or outboard processor has a more immediate appeal than the piece of kit that holds or houses it! There’s a lot to be said for having a well-organised, practical workspace, though, and getting all of your gear set up using ‘normal’ furniture usually involves some significant compromises: making sure your monitors are in the best place can be a pain, controller keyboards can end up occupying too much desk space, and rack gear seems to have a habit of ending up just out of arm’s reach. So, this month, we’re focusing the SOS Spotlight on a range of dedicated studio workstation desks designed to neatly accommodate all of your most-used kit.

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The right studio furniture can transform your workflow.

We round up some of the best current options.

AKA Design

As well as offering a bespoke design and build service, AKA Design manufacture a comprehensive range of off-the-shelf workstation desks aimed at music and post-production studios. There’s the ProMedia, a compact desk that can be specified with a pair of top-mounted 3U racks that double up as a monitor/ display shelf, while the ProEdit is a slightly larger offering that can be fitted with three 4U or 6U top racks. Both can be expanded with AKA’s 12U rack, which boasts a modular design that allows it to be mounted underneath the desk or fitted as a worktop extension. The ProLite, meanwhile, offers a similar design to the ProEdit but at a lower cost, and although it’s not compatible with the modular

racks and worktops, it can be expanded with its own dedicated 12U floor rack.

If you’re after a sit-stand desk then it’s worth taking a look at the ProView and ProCreate ranges, and if they don’t quite tick the boxes, then both the ProEdit and ProMedia can be customised with electronic rise-and-fall frames. Finally, the ProWave makes it possible to combine the company’s racks and worktops with a range of centre sections designed to accommodate popular consoles and control surfaces from the likes of Avid, SSL and more. If you’re still stuck for ideas then the company offer a free design service, and will happily create a bespoke setup based on your studio’s floor plan. £ From £2010 including VAT. W www.akadesign.com

AKA Design ProEdit Argosy Dual 15

Argosy Argosy’s extensive product catalogue includes eight distinct model ranges, each of which are available in an array of different sizes and include options for integrating outboard gear, consoles, control surfaces and controller keyboards.

First up is the Halo series, which boasts a simple, stripped-back design and can be specified with angled top racks and a lowered desktop section designed to accommodate a controller keyboard. The Eclipse range then provides a modular solution that can house a console or control surface — specific models are available for Avid’s S1, S4 and S6, SSL’s Origin and AWS Delta, and Audient’s ASP4816, and there’s a universal version, too. There’s also the Eclipse Edit Desk, a smaller workspace that can be specified with or without a lower rack attached. For those who like to keep all their favourite gear close to hand, the AURA and AIRE ranges offer a wraparound design with rack spaces both above and below the worktop; the former sports Argosy’s signature chunky design, while the latter takes a more stripped-back, minimalist approach. Next up is the modular V Series, which comes in three frame sizes and allows you to ‘mix and match’ its angled rack and worktop components to create everything from a compact, mastering-style console loaded with rack space to a large workstation with plenty of space for keyboards, mixers, desktop processors and more. The Dual 15 then places a pair of angled racks either size of a large desktop surface that can be ordered with or without a large controller keyboard cutout. Finally, the G Series and 50, 60 and 70 Series make it possible to integrate a range of popular control surfaces with rack and desktop gear. The

former is available as a universal design, or a purpose-built variant designed to accommodate an Avid S4 or a PreSonus StudioLive II; the latter offers three frame sizes designed to fit around Avid’s C24. £ From £2645 including VAT. W www.argosyconsole.com

Buso Audio

Buso Audio’s simplest offering comes in the form of the Solo series, a collection of four desks fitted with a sliding tray that will hold either a 61- or 88-note controller keyboard. The Composer series then expands on the design of the Solo with a rear raised shelf, while the Producer series takes things a step further, comprising nine models that incorporate angled racks into the desktop — there are models with racks spanning the entire width, as well as versions with a central desktop. The Artist series, meanwhile, includes two compact models that can house either a 61- or 88-key controller keyboard, and feature a rear shelf that sits above either two or three 3U racks.

A selection of both the Producer and Artist models are also available fitted with a sit-stand frame. The company’s flagship Studio series is designed for users who require more rack space, and includes three models that pair two angled racks with a central worktop, as well as the Studio Classic, which features a trio of 13U rack bays. The Master series includes models equipped with one, two or three

10U racks, each with a full-width sliding shelf below the desk, along with an XL variant that pairs two 10U racks with a large central worktop. Next up is the Control series, which includes two models designed to house control surfaces such as Avid’s Artist Series and SSL’s UF-8 and UC-1, and one that houses Slate’s Raven MTi MAX. The company also offer a comprehensive range of workstations designed to fit around popular mid-size to large-format consoles and control surfaces. Finally, there’s the Modular range, which includes two sizes of desk that can be fitted with standard rack strips, or configured to house a Eurorack modular synth system. £ From £1353.60 including VAT. W www.busoaudio.com

De-Fi

De-Fi’s line-up is simpler than most, with just a single model on offer! Their Platform Studio Desk includes a pair of 3U racks that double up as raised rear shelves, and with the latest iteration of the design, the company have

Buso Audio Master 3

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S T U DIO WOR K S TATION DE S K S

De-Fi Platform

Glorious Sound Desk Pro

redesigned the centre section and added an adjustable central shelf that can either sit flush with the top of the outer racks or be secured at a lower position. The desk comes supplied with three pairs of rack ears, allowing the centre section to be used as a rack and increasing the desk’s total space to 9U. De-Fi have also made some changes to the design of the Platform Studio Desk’s sliding keyboard tray, which is now large enough to accommodate just about any large controller keyboard. £ From £855 including VAT. W www.de-fi.com

Glorious

Along with their collection of hi-fi and DJ-focused furniture, Glorious manufacture three different workstation desks for music studios. The most affordable model in the range is the Workbench, which features a pair of vertical 4U racks below a full-length shelf that can be used to mount monitors and displays, while a pull-out drawer provides a home for a controller keyboard. For those who are tight on space there’s the Sound Desk Compact, a narrow desk equipped with two vertical 3U racks, a rear shelf and a sliding

SessionDesk Gustav 2.0

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keyboard tray that’ll accommodate a 61-key controller. The Sound Desk Pro then expands the design, adding some additional storage space between the two racks and extending the keyboard tray to leave room for an 88-key controller. £ From €349.99 including VAT. W www.glorious.de

Sessiondesk

Sessiondesk’s entry-level Home Series desk can be ordered in two sizes, both of which can be fitted with an optional pair of top-mounted 3U racks and a sliding keyboard drawer. The Tav Series then includes three models, each equipped with two 9U angled racks and a rear shelf: the Gustav, Oktav and Quintav. All three are height-adjustable, and both the Gustav and Oktav feature an independent centre section that can be set at three different heights; the Quintav is a narrower design that places both racks together with no central workspace. The Modular System, as its name suggests, makes it possible to create an array of different configurations using four key components: the Solo, Duo, Trio and Trapez.

The Solo and Duo offer single and dual angled racks — both are available in 8U and 12U variants — while the Trio combines two angled 8U top racks with a lower 12U rack and a central work surface. The Trapez 60, 80 and 100 then offer three sizes of workstation that feature two 12U angled

top racks and a pair of 12U lower racks. The Modular System offerings all boast slotted rear panels that have been designed to help with ventilation and minimise reflections. Finally, there’s the Air Desk, an innovative design that has been created with composers in mind.

Available in two sizes, the model features a fixed keyboard shelf placed below a sliding desktop section, offering greater stability than many rival designs while still saving on space and placing computer peripherals in the optimal location. £ From £699 including VAT. W www.sessiondesk.com

StudioDesk

Another company that have just about every base covered are StudioDesk, whose product range includes no fewer than 12 core models! The entry-level Harmony features 8U of rack space underneath a monitor/display shelf and an adjustable keyboard tray large enough for most 88-key controllers. The Spirit and Fantom then offer a pair of compact workstations fitted with keyboard trays; the former is equipped with built-in swivel speaker stands and a central 3U rack, and the latter places 6U of rack space underneath a full-length shelf.

The company’s most popular offering is the Beat series, a versatile workstation with a rear shelf above either 6U or 9U of rack space and a sliding keyboard tray. Opting for the Dominator gets you a similar design but with the vertical racks supplemented by a central, angled 8U section, and the option of adding a large pull-out keyboard stand capable of supporting heavy controllers. Following a number of user requests, StudioDesk developed their first sit-stand desk in the

Zaor

Wavebone Space Station

form of the Xtreme, a simple design that pairs a large worktop with a rear shelf and boasts an electric lift mechanism.

The Orbit series then makes it possible to configure your own workspace by combining an electric sit-stand platform with 2U rack, keyboard tray and speaker shelf add-ons. One of the company’s popular early designs lives on thanks to the Commander V2, a sleek workstation fitted with 8U of angled rack space and enough room underneath for a pull-out keyboard stand or extra rack cabinet.

The Enterprise has been designed as the perfect centrepiece for a modern DAW setup, with a lowered display shelf and central angled rack joined by swivel speaker stands and support for either a sliding keyboard tray or pull-out shelf — the Enterprise is available in Classic or Electric variants, the latter of which offers height adjustment. Lastly, the Pro Line series comprises three core models aimed at large studios that require plenty of rack and shelf space. The largest version, the Pro Line XXL, is nearly two metres wide and has 20U of rack space as standard, with the option of adding an extra 32U underneath. All of StudioDesk’s offerings come fitted with built-in cable management, and many are also kitted out with ambient RGB lighting! Helpfully, the company’s website features a Quick Comparison sheet that places all of the key information about each of their designs into a table so you can find out which best suits your needs. £ From £848 including VAT. W www.studiodesk.net

while a sliding height-adjustable keyboard tray leaves room underneath for a 61-key controller. The Star Ship then offers the company’s take on a sit-stand design, with a single-tier surface and sliding keyboard tray mounted on a dual-motor electric lift platform — helpfully, it’s possible to store and recall up to four preset heights.

The Headquarter then offers a larger workstation equipped with 9U of angled rack space, and comes with a separate rolling keyboard stand that allows an 88-note controller to be housed neatly underneath the desk. For those in need of more desktop real estate, there’s the Space Station, a modular workstation that comprises four components: the Side Desk, a short desk and frame section designed to form either end of the Space Station; the Rack Desk, a desktop section fitted with an angled 6U rack; and a pair of Main Desks that offer 600mm and 800mm-wide desktop sections. Its modular design makes possible a range of different configurations, and there’s a perforated back panel designed to help keep cabling neat and tidy. £ From £349.99 including VAT. W www.wavebone.com

Zaor’s range of studio desks is divided into four families. The Vision Series includes desktop-only (Vision O and OS) and rack-equipped (Vision W and WS) models, both of which are available in two different lengths — the W and WS variants feature angled top racks with 6U and 9U of space respectively. The Vision K and KS then offer standalone keyboard stands that are designed to tuck neatly under the Vision-series desks. The Miza Line then includes a selection of models that vary in size from compact units designed to fit a 49-key controller, to large workstations that will accommodate 88-note piano keyboards. All feature pull-out keyboard shelves, and the flagship Miza X2 Flex includes 12U of angled rack space below a large shelf equipped with IsoAcoustics speaker isolators. Next up is the Classic Series, which includes the Maestro 12+6, 24 and 36, a trio of mastering-style desks that put an abundance of rack space within easy reach, along with the Onda Angled and Onda Angled MkII, which both combine a pair of ‘sidecar’ racks with a central worktop. The Classic Line also includes a selection of models designed to accommodate consoles such as the SSL Matrix and Behringer X32, as well as the Marea O, a universal workstation that houses a mid-sized console beneath 6U of rack space. Finally, the iDesign Series takes a minimalist approach, and includes models with or without built-in rack space, as well as the Vela Klavi, a built-to-order model that features a built-in 88-note controller. £ From €395 including VAT. W www.zaorstudiofurniture.com

Wavebone

There are four models to choose from in the Wavebone range. The first is the Star Rover, a compact desk designed for studios where space is at a premium. Its two-tier design places a pair of angled 3U racks below a rear display/monitor shelf,

Zaor Miza X2

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ON TE ST

Emergence Audio Time Crystals

Kontakt Instrument

HHHH

I’ve experienced the calming sounds of a singing bowl in person exactly once in my life, but I couldn’t tell you much about the experience because not long after my friend at university began playing the bowl, I fell asleep. I have, however, managed to stay awake and alert while exploring Time Crystals, the new library from Emergence Audio, which pairs their very modern and now well-known Non-Static Sampling and Infinite Motion Engine 2.0 technologies with the resonance of quartz singing bowls that have calmed humans for centuries.

It’s a simple concept: textures in the form of drones, pads, pulses and strikes originating from the sounds of singing bowls, equipped with detailed MIDI CC control to further shape the tones.

Creatively, it’s a double‑edged sword because while there are no static loops in this Kontakt library, things are so fluid that if you aren’t recording your experiments, you may stumble upon a wonderful sound but never find it again as you go through long evolving performances with subtle variations.

As important as the singing bowl is the tool you play it with, and Time Crystals offers three options: mallet, suede and wooden striker. The differences in tone are most apparent up to C4, after which they all start to generate bell-like tones with more muted or sharper attacks, depending on the selection of tools. The basic patch for each striker works as a building block to create your own textural pieces. For layered sounds, you have more than 160 presets to work with, including single patches or more elaborate multis.

While the library boasts an eight-octave range, there is definitely a sweet spot for each patch in terms of where it unfolds best. Each time I loaded a patch, the first thing I found myself doing was auditioning octaves until the sound seemed to come alive. Across the library you have everything from ominous low drones to crystal clear high notes that ring out delicately.

Modulation and movement are central to the library, and riding the mod wheel brings interesting results at the default level, even before you assign specific parameters

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to MIDI CC according to your preference.

The velocity sensitivity slider (found on the GUI’s Main page as ‘Vel’) is particularly useful for this library. Since a lot of the patches are played with long-held notes to let the sounds evolve, having control over velocity means that you can press down on the keys as hard as you like, without suddenly generating a sound that is too loud in a uniformly quiet passage.

The use cases for this library are gentle, ambient cues, textural underscore, and straight up meditative music, but there’s also some fun to be had with more gritty, percussive sounds, like in the Termination Pulse patch, for example.

With their previous collections, Emergence Audio have already established that they offer solid source material with precise control, and with Time Crystals they continue to deliver on that promise. After weeks of playing with it, I had one minor quibble. They say Time Crystals “invites you to shape time”, but I found it most suitable for pieces where you want to lose all sense of time — not slow it down, or speed it up... just forget that it exists. Sonal D’Silva $149 www.emergenceaudio.com

Klevgrand Elpiano

Plug-in Instrument

HHHHH Elpiano is an electric piano plug-in that offers 10 different piano models. All the familiar plug-in platforms are supported, including AAX, and the coding includes optimisation for Apple Silicon.

Elpiano covers a range of mechanical and electronic piano models that may then be processed via a choice of five different speaker cabinets. There’s also a generous effects section populated with reverb, Room, Delay, Chorus, Phaser, Tremolo, Vibrato and EQ, with adjustable Drive to add a bit of dirt. The modulation effects

are located at the bottom of the window and have typical stompbox‑type control configurations with just one or two knobs each, while Reverb, Room and Delay have horizontal amount sliders and a further drop-down panel for the Reverb and Delay that offer additional parameters, including tempo sync options for the delay. Room offers a choice of six room types from Large Room to Booth. The main rotary controls adjust Bass, Treble, Drive and Output level.

The sounds themselves are very responsive and range from late-night lounge smoothness to FM funk and bell-like tones.

All five cabinet models, which I suspect employ impulse responses, can be used in both mono and stereo configurations with a choice of Small, Medium, Large, Hi-fi or Toy, the latter delivering a very small speaker sound. While 10 pianos might not seem like a lot, the many permutations of piano, room and speaker cabinet provide a large range of usable options that can be further treated via the effects section.

If no cabinet model is needed that can be switched off, and when a cabinet is active, further buttons select mono or stereo miking arrangements. A master settings panel can be opened to adjust the GUI size, global tuning, velocity curve and keycap weight, the latter operating as a tilt control adjusting the sensitivity between low and high notes. A small piano keyboard graphic shows the notes that are being played.

As to the actual range of sounds, they cover pretty much anything you could want from an electric piano, hinting at Rhodes, Wurli, Hohner and FM voices with an authentic bell-like attack. The timbre changes smoothly with playing velocity, implying the sounds are modelled rather than sampled. The styling of the controls changes depending on the piano model, possibly giving more of a hint as to what type of piano is being modelled, and the rate and depth for the modulation effects are handled by the X/Y pad visible behind

the effect waveform. The Tremolo waveform can be changed using the Shape knob.

On the surface Elpiano might seem like a very simple instrument but Klevgrand have brought these electric piano sounds to life in a such an organic and responsive way that just makes you want to keep on using it. Paul White $149 www.klevgrand.com

Zero-G BioRhythms

Kontakt Instrument

HHHH Zero-G’s BioRhythms provides a loop-based drum and rhythm sample library for the full version of Kontakt (v7.8.1 or later). The library contains approximately 2300 loops coming in at just over 3GB of 48kHz/16-bit samples, and the Kontakt front-end presents these loops within a 16-channel playback engine. By default, the loops are organised into 13 Kontakt Snapshot presets but each of these contains 12 banks and 12 variations, each of which can be selected and/or triggered via the two sets of keyswitches. The library is NKS‑friendly so can be used with Komplete Kontrol or Maschine 2. Obviously, the loop-based format will not appeal to those who like to roll their own grooves but, if you need some instant rhythmic inspiration, there is plenty here to explore. Stylistically, the loops themselves could be used in a range of musical contexts from pop, electronic styles or in music-to-picture work.

The UI is perhaps not the most beautiful Kontakt instrument you have ever seen but it does pack in quite a lot of functionality once you have found your way around. It’s dominated by the 16-channel mixer panel and, by default, the loops are laid out in a sensibly consistent fashion with, for example, kick loops on channels 1 and 2, snare on channel 3, and so on. The other channels include hi-hats, claps, bongos, congas, shaker, tambourine, other percussion and then a selection of effects and riser sounds.

As well as the usual fader, mute, solo, pan and send (to the reverb and delay) controls, you also get the Var, Tune and Speed controls. The latter two do what you might expect, while the Var control lets you switch the loop variation for a specific channel, allowing you to easily

mix and match within the loop sets contained within a specific Snapshot. You can also trust to luck and randomise the Var setting for each channel.

At the top of the mixer you can select a specific channel, and its waveform is then displayed in the top-most panel.

All the various effects tabs can then be popped open and you can tweak the various effects — EQ/ Filter, Compression, etc — on a per-channel basis, or ‘link’ the settings to make changes across all channels. Loop playback can be switched between slice and time-stretch modes (the latter is a little more CPU intensive).

You can, of course, also save your own Snapshots if you create a set of loop variations and mixer settings that you might wish to reuse.

OK, so BioRhythms is essentially a loop playback engine. It doesn’t break any particular new ground in terms of functionality or musical content. However, the loops themselves are very usable, and you can get a lot of mileage out of the supplied content once you have found your way around the toolset. It’s also sensibly priced. For those on a budget looking for some instant drum and percussion inspiration, the demos on the Zero-G website would be well worth auditioning.

John Walden £62 www.zero-g.co.uk

Sonora Cinematic Emma Legato Kontakt Instrument

HHHH

Featuring Emma May Price — the same soprano heard in the Sonora Cinematic Aria libraries — Emma Legato is a very simple and inexpensive vocal instrument that runs on the free version of Kontakt (version 8 or higher). The instrument has three main sound voice options: Humming, Poco Vib and Aah. The Aah voice has a very expressive vibrato that creeps in as the note evolves. The true legato part is important as the transitions between notes are also sampled to produce a very natural-sounding

portamento. Keyswitching is available in the lower register of the keyboard to switch between the three voicings, and also to turn Auto mode on or off. Auto is a clever feature that automatically adjusts the legato transition time according to the incoming note placement, though it can be turned off to allow the user to set a fixed speed. Clicking the settings cog button opens up adjustable fade-in, fade-out and release times for the legato as well as attack and release times.

To the right of the three voice select buttons are fuzzy blobs for Reverb and Atmo. As you hold/slide up and down over these images the blob grows larger and brighter and the amount of effect is adjusted accordingly. Reverb does as you’d expect while Atmo appears to be a macro control that brings in multiple effects at the same time. Clicking on the dot to the right of the word Atmo changes its colour from blue to pink.

In blue mode the Atmo control brings in an endearingly shimmery, diffuse type of ambience with a long decay tail, which gives the voice an ethereal quality. In pink mode the effect combines modulated reverb with some subtle granular processing that creates a very spacious, textural effect that also seems to push the sound forward in the mix as you increase the amount of processing.

There are a few presets to get you started but with so few controls, they are barely necessary. Admittedly there are dozens of samples of Mmms and Ahhs on the market already, but the combination of a talented performer and true legato note transitions gives Emma Legato a hauntingly endearing character that will find many uses.

That really is it by way of controls so the learning curve is almost non-existent.

Despite its apparent simplicity, Emma Legato produces really great-sounding results so if you still need to be convinced, go to the Sonora Cinematic website and check out the video demo. Paul White £29 www.sonoracinematic.com Audio examples of this month’s libraries are available at www.soundonsound.com.

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TECHNIQUE

Studio One

VCA faders offer a powerful way to control your levels.

ROBIN VINCENT

W

hen you have large projects containing many tracks, there comes a point where mixing starts to feel a bit unwieldy. Studio One has a few tools that can help you handle that, such as routing busses and groups.

However, there’s another way to take the leg work out of mixing, and that’s through the use of VCA channels. We’re going to look at how the VCA channel could be the secret mixing weapon you’ve never got around to trying.

The easiest way to control multiple sources at once is by having a bunch of channels selected. You can do this by Ctrl-clicking (Command-click on Mac) individual channels or Shift-clicking across a range of channels. With several channels selected, anything you do to one will also be done to the others. The faders and pan controls will move together, the mute and solo buttons all operate at the same time, and routing and effects all get changed together as if they are one channel. All parameters maintain their relationship to each other and are scaled as they move.

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You can turn this selection of tracks into a group from the right-click menu, which means that when you click away from the channels to do something else, the connection between those selected channels is retained. Otherwise, you’d have to keep re-selecting them.

Grouping channels is certainly useful, but requires a bit of management to be versatile. Maybe you don’t want the panning linked, you might want different effects on different channels, or you just want to group the volume control and nothing else. If you bring up the Groups column to the left of the mixer and right-click the group in question, you can add or remove various mixer attributes from the group. Another useful tip is that you can hold the Alt (Option) key and move any parameter on a channel within the group without affecting any of the others.

So, the flexibility is there, but sometimes it takes a lot of clicks to get it to be what you need it to be.

Another option is to create a bus and then route your channels to it. This will give you control over the overall volume of those channels. You also have mute/ solo control and stereo balance. But this

control is applied to the mixed sound of those channels, rather than acting on the channels individually. While that is absolutely useful, it works very differently. You can’t ‘exempt’ channels in the group from pan changes, for example, and if you’ve got any post-fader effects on the channels being routed to the bus, you’ll still hear them even with the bus fader pulled down.

Both groups and busses have their uses, but throwing in the odd VCA channel can take on some of the heavy lifting and smarten up your mixing in a way that retains the intention of your mix.

VCA Channels A VCA channel is an assignable fader that can take on the role of multiple volume controls and automation. It’s essentially a controller. What it does is send an offset value to the faders of whatever channels are assigned to it. All the faders assigned to a VCA channel will move together, but rather than all moving to the same value, they will maintain their relative relationships. It’s much easier to see than it is to explain, so let’s set up a VCA channel on a bunch of tracks and see it in action.

Find a bunch of tracks that you want to control the volume of. This could be

spanner icon on the left and make sure that ‘VCA connections’ is ticked under Channel Components.

For now, don’t put these channels into a group, or you might get confused about what’s controlling what. If you know what you’re doing, having a bunch of channels working as a group but with volume control deselected is probably a very clever way to take advantage of what a VCA channel offers without doing away with the advantages of a group.

However, for the purposes of focusing on VCA channels in this workshop, don’t have them in a group!

Special Relativity

Here, all the drum channels in a mix have been assigned to a VCA channel (the red fader on the right). Moving the VCA fader will change the levels of all the assigned channels, but unlike with a bus, no audio is routed through it — the VCA simply controls the level of channels assigned to it.

the individual channels of a drum kit, several vocal harmonies, a couple of guitar tracks: you get the idea. Select them using Ctrl-click (Command-click on Mac) or, if they are consecutive channels in the console, click on the first channel and then Shift-click on the last. Right-click on any of them and choose ‘Add VCA for selected channels’. The VCA channel will appear, noticeable with its red fader cap, all the way over to the right, next to the master output. The position might not be ideal, so if you click on the little spanner icon to the left of the mixer you can untick the box ‘Keep VCA channels to the right’ and then move it so it’s alongside your VCA group.

You don’t have to do it this way: you can add a VCA channel that’s not assigned to anything, just like you would add any other type of channel.

When you add a VCA channel, a little scribble strip will appear beneath the fader of every other channel. If you click on it, you can assign the fader to any available VCA channel. If your VCA boxes are missing, check the

Anyway, let’s not get distracted as we’re about to reveal the big point about the VCA channel. If you move the VCA channel fader, you’ll see all the assigned channels moving in concert with it. Now, here’s the coolest part. You can adjust the individual volume on all or any of the other channels without having to ungroup, deselect or hold a modifier key, and the VCA fader will still be in overall control. So, if you set the levels differently for each channel and then play with the VCA fader you can see how the channel faders maintain their relationship to each other. If one channel is 3dB louder than another, it will always be 3dB louder until you push it to the maximum at the top or to nothing at the bottom.

So, the effect of the VCA channel on volume is very much the same as it is with grouped channels, except that it leaves the individual channels free to make their own changes. It also stands out more in the mixer, like a bus does, and so it gives you a nice visual indication that it has control over a bunch of other things. Compared to using a bus, a VCA channel will pull the individual channel faders down, so you don’t get the issue with post-fader send effects still coming through when you turn the VCA down.

VCA channels are otherwise pretty basic, although you can rename them, recolour them and add an icon if you wish. A VCA channel is not an audio channel and nothing is being routed to it, so it does not need insert or send effects.

The mute and solo buttons are useful and act on all the assigned channels.

It appears to have a panning control, but this actually doesn’t do anything, which is a shame and feels like a missed opportunity or perhaps a feature yet to be implemented. VCA channels could be

useful in all sorts of places for control or modulation of parameters, but currently they are only for volume control. Besides, MIDI and macro controllers can probably take care of everything else.

VCA Channels & Automation VCA channels can be automated, which effectively means you can automate the mixing of a bunch of channels at once. But it’s also slightly cleverer than that. If an individual channel already has automation, the existing data can be offset or scaled by the VCA. If you look at the automation on a track, you can see the colourful curve of the movement you’ve recorded. In the background, there’s a grey version of those curves that represents the actual automation offset by the VCA. So rather than the existing automation fighting the VCA fader, it becomes scaled by it.

If you untick the ‘Keep VCA channels to the right’ box in the mixer preferences, you can move your VCAs to anywhere in the console. Making VCA connections visible (under Channel Components) will give you a drop-down menu on each channel for assigning it to a VCA.

www.soundonsound.com / June 2025

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Studio One

TECHNIQUE

VCA CHANNELS

The blue line on the ‘Bossa Loop’ track is that track’s original automation. The VCA channel below it has its own automation, shown in red, and the combination of the two automation lines is shown in grey on the audio track.

If you’ve ever tried adjusting levels after you’ve already implemented a bunch of automation then you’ll know it can take a lot of editing and using the Range tool to scale the automation to fit with the new situation. Slapping on a VCA channel can really help with that.

To automate the VCA fader, you choose a Write mode from the automation box and move the fader on playback, just like any other channel. However, you may have noticed that there’s no VCA ‘track’ in the arrangement window. Never fear; one will appear once you start writing automation or if you select ‘Edit automation’ from the right-click menu of the VCA fader.

There’s a neat feature where you can ‘print’ VCA channel automation directly onto the other tracks. This is the Merge VCA Automation option from the right-click menu. All the tracks then get their own editable volume automation, and the VCA fader returns to zero,

where it can get on with the business of scaling the assigned tracks.

What do you do if you’ve automated the VCA channel, then want to scale that automation back a bit but don’t wish to merge it into the tracks? Easy: just add another VCA channel to control the first VCA channel! If you think about it, if you have a bunch of VCA faders controlling

Studio One. But as it is, the VCA channel is a humble fader level control that’s a brilliant and versatile way of mixing multiple channels without losing the flexibility of adjusting and automating them individually too.

Working Together A last point to make is that VCA channels shouldn’t be seen as a replacement for groups or busses. You may still want to route your channels to a bus to use group processing or simply to organise your signal flow.

That’s fine; all the VCA channel is doing is taking over the level control so you can leave the bus level alone.

With groups, I mentioned earlier that you can still use all the other group facilities like panning and effects, but let a VCA sort out the level control. So the VCA channel augments and complements your other mixing tools rather than replacing them, and once you start using them, you’ll never go back.

“The VCA channel augments and complements your other mixing tools rather than replacing them...”

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a lot of different groups you could add another VCA channel to control all of those VCA channels controlling all those other VCA channels...

That can get interesting very quickly, and is precisely why I’d love to see PreSonus develop the VCA channel into a self-contained modulation device with LFOs and possibility engines that could be assigned to any parameter in

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Andrew Scheps Engineer, producer, mixer, Grammy Award winner (Lana Del Rey, Hozier, Jay-Z).

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Pro Tools Live

TECHNIQUE

We check out what’s new in Live 12.1.

OLI FREKE

F

ollowing the smorgasbord of new features in 2024’s Live 12 update, 12.1 brings even more devices and enhancements. Let’s take a run through what’s new and improved in the latest version...

Auto Shift First up, and possibly most significant, is the new Auto Shift device — Ableton’s answer to vocal pitch correction.

Something of a cross between the simple-but-effective Little AlterBoy

by Soundtoys and the high-end vocal processing of market leaders Antares Auto‑Tune and Celemony Melodyne, Auto Shift is an intuitive and ‘free’ alternative.

Whilst it doesn’t necessarily have the audio polish or graphical display of the latter two, it does have some impressive features they lack.

The first of these is the ability to use a separate MIDI clip as a side‑chain input. In the simplest use case, the external MIDI part might contain only the ‘correct’ version of the melody, allowing the vocal to track it for a fully corrected vocal line. But things get much more interesting if you use a clip with a different melody, or a counterpoint, or even a mad arpeggiation to create wild and almost impossible-to-achieve-otherwise pitch effects. And, of course, you can control any monophonic line, it doesn’t necessarily have to be a vocal. This alone will reward much creative experimentation.

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The Main Quantizer tab with Scale Aware and MIDI side‑chain engaged.

Not only that, but the incoming MIDI part doesn’t even have to be monophonic — you can program a chord sequence in ‘Poly’ mode, and Auto Shift will produce an instant backing harmony (up to four parts). Use the Wet/ Dry balance to make sure the lead is prominent, and job done. (Of course, for more control, you may wish to duplicate the lead part and create a harmony Group on a new channel, with each part having its own pan and volume settings).

Real-time pitch correction is another welcome feature. Whilst more widely supported in other auto‑tune plug-ins, this is a trick still not available in Melodyne.

All of this is controlled within the Quantizer tab of Auto Shift, with LFO being the secondary tab. The Quantizer tab also contains a horizontal pitch meter which shows how much the incoming audio is being corrected by, and a one‑octave keyboard highlights the note to which the audio is being quantised. If you click the Scale Aware button, the keyboard then highlights the notes of the current Global Scale and

quantises only to those. In a neat twist, you can choose additional accidentals to include, or indeed not use Scale Aware at all and leave it chromatic, or choose an entirely arbitrary selection of notes.

‘Strength’ adjusts how strictly Auto Shift moves the incoming note’s pitch to the nearest quantised note, and ‘Smooth’ controls the rate of transition between pitches. Set Strength to 100 percent and Smooth to zero percent for the full T-Pain; or something a bit less extreme for a more naturalistic outcome. Formant can also be set to track pitch (to help avoid munchkinisation), but of course, formant can also be used as an effect too — and as we have come to expect with Live, automation is easy and can unlock the doors to many exciting discoveries.

I was initially a bit confused as to why there were both ‘Shift’ and ‘Pitch’ controls in the Quantize tab. At first glance they both seem to do the same thing — shift the incoming audio by a chosen amount. But it would appear Shift is for a ‘permanent’ key change and therefore is only adjustable in semitones. Pitch,

The LFO panel allows for direct control over four parameters — Pitch, Formant, Volume and Pan — as well as reset on note onset, Delay, and Attack (time to peak value).

however, has both semitone and fine control, and is therefore perhaps more useful for automation and for use as an effect in its own right (chorusing, etc). And you don’t even have to use these pitch options for vibrato as there’s a dedicated vibrato panel with Amount, Rate, Fade In, and a ‘Natural’ toggle.

Having said all that, you don’t even have to quantise the pitch of your audio if you don’t want to. Just turn the whole Quantize tab off and use the LFO tab by itself for a different approach.

So, what can you do with it? Well, the LFO tab is basically an LFO tool that has been hardwired to Pitch, Formant, Volume and Pan. There are, of course, the expected LFO-specific parameters of Shape, Rate and Reset (LFO phase returns to zero with each new MIDI note).

This control of Auto Shift you could just do with a standalone LFO tool mapped to some of the Quantizer parameters, sure, but things like resetting the LFO on MIDI note-on would be much harder to achieve. And having these LFO pitch options ‘hardwired and ready to go’ in Auto Shift is undoubtedly handy.

Auto Shift may not quite match the sonic quality of Auto‑Tune or Melodyne, but it’s not far off, and to expect it to do so might be a bit unreasonable for a stock plug-in. Especially given that Antares and Celemony have been developing and refining their vocal pitch correction algorithms for many years now. And neither can Auto Shift pluck out and tune individual notes from within a chord like Melodyne. But then neither of those plug-ins currently give MIDI control over the vocal part, which is so easy to do in Auto Shift.

Given the plug-in’s ease of automation, ability to correct in real time and the novel MIDI input feature, it really feels like it’s encouraging you to experiment, which is an exciting contrast to the more professional ‘let’s fix these vocals’ attitude of the big boys. All of which is fine, it just depends on what you want to do. Either way, Auto Shift is a great new offering.

Note the X/Y pad with the two macros controlling Factor and Grain for the Stretch effect.

set of preset kits that really showcase the full range of sonic treatment possible.

Drum Sampler’s interface has four simple panels: the first shows the loaded waveform (you can drag and drop your own samples here) and provides a set of simplified envelope controls (Attack, Hold, Decay), Gate/Trigger mode and Tuning.

The second panel is where the fun starts — it’s dominated by a square X/Y control ‘pad’ where the effect’s parameter macros are adjusted by the X/Y values of its cursor. Like Meld, the two macros are appropriately pre‑assigned for the

distortion, updated metering, and new features include Mid‑Sides routing, True Peak, Soft Clip, and Maximize modes.

Saturator has been updated with a Bass Shaper curve designed for 808‑style bass and low-end parts.

MIDI editing improvements: The MIDI Editor now includes a toolbar that allows filtering and selecting notes based on attributes like time, pitch, velocity, chance, and duration. 12.1 includes the two new MPE-specific MIDI Transformations Glissando and LFO, which were profiled in the April 2025 edition of SOS, along with the other new and updated MIDI Transformations.

“Auto Shift may not quite match the sonic quality of Auto‑Tune or Melodyne, but it’s not far off...”

Drum Sampler Another nice addition to Live 12.1 is the new Drum Sampler device. Designed for one-shot percussive samples, Drum Sampler focuses on the processing that is often used on such sounds to give extra grit, dirt and punch. It comes with a large

effect selected. (Those effects being: Stretch, Loop, Pitch Envelope, Punch, 8 Bit, FM, Ring Modulation, Sub-Oscillator, and Noise). The third panel has a fully featured filter section, and the fourth has output controls like volume and pan.

This gives a huge range of options to adapt drum and percussion sounds to your taste and to a production, and the great thing is that the extremes of the macro controls are not necessarily very subtle either! Splendid for really pushing a sample into new sonic territory.

Work Smarter...

Alongside Auto Shift and Drum Sampler, Live 12.1 includes a set of very useful workflow improvements: Auto-Tagging for samples and plug-ins: Live 12.1 can now automatically assign tags to short‑duration audio samples (less than 60 seconds) and to VST3 plug-ins based on their metadata.

Enhanced Limiter and Saturator Devices: The Limiter has been redesigned with a smoother release curve to minimise

Updates for Push 2 and Push 3 controllers: Enhancements include the ability to design sounds with greater detail using Macro Variations, apply groove to all MIDI clips with a single encoder twist on Push 3, and improved browser filtering. Users can also save Default Sets and Template Sets directly on the hardware and organise projects more efficiently by rearranging tracks and scenes.

MIDI Tools and Sequencers Packs: Live 12.1 introduced two new Max for Live packs — MIDI Tools by Philip Meyer is a collection of MIDI generators and transformations, including devices like Phase Pattern, Polyrhythm, Stages, Retrigger, and Slice Shuffler, offering diverse rhythmic and melodic possibilities. Sequencers

by Ableton features creative step‑sequencing tools such as StepArp, SQ, and Rhythmic Steps, inspired by hardware systems and classic Live devices, to generate complex and evolving patterns.

www.soundonsound.com / June 2025

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Pro Tools Cubase

TECHNIQUE

Cubase’s new Modulator has many superpowers — it can even ride your vocal levels for you!

JOHN WALDEN

W

aves Audio’s Vocal Rider plug-in has proven enduringly popular (it was introduced some 15 years ago!) with good reason — it can automate, or semi-automate, what was once a time-consuming task: riding a channel fader to ensure the vocal isn’t ‘lost’ at different points in a busy and dynamic mix.

In this article, I’ll take you through how you can use Cubase 14’s new Modulators to achieve similar results to Vocal Rider.

Rider Essentials Under the hood, Vocal Rider does a number of things. First, like a combination of compression and expansion, it can be used to make the overall dynamic range of vocals, dialogue or solo instruments (despite the name, it works well on various sources) more consistent. Second, as it can accept a side-chain signal (such as a vocal-free backing track, or specific elements from the mix) and track that source’s dynamics, it can be used to adjust how the vocal level is balanced against that mix, so that the lyrics remain clearly audible even as the arrangement changes. Third, all of its automatic level adjustments can be written as automation data to your DAW. So you’re able to finesse the result with targeted manual edits — something that’s much harder to do with a compressor.

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In the first stage of our DIY vocal riding, the combination of the Envelope Follower Modulator and Volume plug-in lets us reduce the dynamic range of the original vocal.

It’s possible in most DAWs to achieve similar ends using the built in tools, but generally speaking it can be more fiddly to set up. For instance, Mike Senior outlined one such approach in a Cubase workshop back in SOS June 2010 (https://sosm.ag/ cubase-0610), shortly after Vocal Rider came out. With the introduction in Cubase 14 of the new Modulator facility, though, Cubase Pro users now have another option...

On The Level First, we’ll use Cubase’s Envelope Follower Modulator and the new Volume plug-in in combination to reign in the dynamic range of a spoken or sung vocal.

A suitable configuration is shown in the first screenshot. Next to my main lead vocal channel is a mono Group Track set to receive a pre-fader send from the main vocal. The processing (courtesy of the Envelope Shaper Modulator and an instance of the Volume plug-in) is done to this Group Track, before both tracks are sent to and blended on a further Group Track.

As the name suggests, the Envelope Follower creates a control signal based on the level (amplitude) of the incoming audio signal. By adjusting the Attack, Release, Threshold and Gain knobs, you can modulate the amplitude curve displayed in the Modulator’s main panel. This curve

For the second stage of the process, this instance of the Envelope Follower uses its side-chain input to let it track the volume of key elements in the backing track.

can then be used to drive changes in any of the Modulator’s target parameters. In this case, that’s Volume’s Gain parameter. Note that the Modulation Depth has been set to a negative value (-50%) — this way, as the incoming audio gets louder, the Modulator will turn down Volume’s Gain, thus reducing the signal level at the track’s output.

Clearly, with Attack, Release and Threshold all present, this is not a million miles away from being a regular compressor’s control set. Meanwhile, the Envelope Follower’s Gain control acts as an input level control, equivalent to pushing a compressor’s input harder. Adjusting the negative Modulation Depth then lets you control how much gain reduction is applied once the input signal passes the threshold.

You can also adjust the blend of these two vocal channels (one unprocessed and one processed), which makes it easy to dial back the overall impact of the gain reduction to taste. You might also have noticed that in this example, this blend is then sent to a further Group Track that I named ‘Vocal Ride Stage 1’ — I’ll come back to this in a minute!

Before I do though, it should also be noted that Vocal Rider, while letting you reduce volume peaks in a similar fashion, also raises the levels of quieter sections. I suspect you could configure an additional Modulator stage to simulate this, but I’ve kept things simple here and, in most cases, you can achieve a perfectly adequate reduction in overall dynamic range by attacking the problem from one direction (top down, so to speak) rather than two (top down and bottom up).

The Envelope Follower modulates the Volume control on the Vocal Ride Stage 2 track, increasing the track’s volume when the level of the backing track gets louder.

loudest parts of any mix, so best not to let them drive the automatic vocal levelling too much — otherwise we can easily end up with a vocal that pumps in time with the kick and/or snare!

Two further points need noting. First, the EQed Backing track’s main output is set to No Bus, so no matter how much signal is sent here it won’t be heard in the main stereo mix. Second, this channel has a steep high-pass filter applied, so that the kick/bass elements in the backing

You can set the Attack, Release (both set to act more slowly, for a smoother response), Threshold, Gain and Depth of the Envelope Follower to adjust the degree of volume control, so that the vocal signal on this track follows the dynamics of the backing track. The main vocal bus then sums this Vocal Ride Stage 2 signal with that of Vocal Ride Stage 1. Finally, you can just use these two tracks’ faders to set how the overall vocal levels ride the dynamic changes in the overall mix level and remain audible. Hey presto! Two stages of vocal riding that help to prevent your vocal getting lost.

“Drum elements can be amongst the loudest parts of any mix, so best Can’t Catch Me? not to let them drive the automatic I found this approach very effective on the whole, but there’s one caveat vocal levelling too much.”

A Bit On The Side The second stage of vocal riding involves getting the overall vocal level to ride along with changes in the dynamics of the instrumental backing in which it sits. As shown in the first screenshot, the approach I’ve employed (which leans on Mike’s article that I mentioned above) requires two further mono Group Tracks, that I’ve named ‘EQed Backing’ and ‘Vocal Ride Stage 2’).

The EQed Backing channel receives post-fader sends from any of the backing instrumentation tracks you want your vocal to ride its volume against. In this case, I’ve used two such feeds. One comes from an ‘all instruments’ track, and the other, with a much lower send level, from an ‘all drums’ track. Drum elements can be amongst the

track can be controlled. This will let the midrange frequencies in the backing track, which are the most likely to mask our vocal frequencies, dominate the second stage of vocal riding.

The Vocal Ride Stage 2 track is set to receive a send from Vocal Ride Stage 1, and is then routed to the main vocal bus.

It also has both an Envelope Follower Modulator and a Volume plug-in inserted on it. In this case, though, the Envelope Follower uses a side-chain input signal from the EQed Backing track, and changes in the level of this signal therefore dictate the envelope shape in the Modulator’s display. This envelope is then used as the modulation source to target the Gain control in the Volume plug-in. But in this case, the Modulation Depth is positive so, as the level of the side-chain signal from the EQed Backing signal increases, so too will the level of the Vocal Ride Stage 2 track.

that some might find significant.

As I said above, Waves’ Vocal Rider can record its volume automation data to your DAW, and that’s useful because, as with most automated processes, although it takes a lot of the hard work out of vocal level automation, there can still be sections you wish to adjust. Unless I’ve missed something (quite possible, so please do feel free to put me right if you know something I don’t!), then it doesn’t seem possible to ‘catch’ the Modulator‑driven changes to the Volume plug-in’s Gain control, and record them as standard automation data.

That’s a shame as the option for a final ‘tweak it manually’ stage would be great to have. Of course, this is the first iteration of the Modulator feature set and I’m sure Steinberg will be keen to expand upon what is a mighty impressive start. Let’s keep our fingers crossed that a means to record Modulator-driven automation data is on the development team’s ‘to do’ list.

www.soundonsound.com / June 2025

133

Pro Tools

TECHNIQUE

Find your way around large mixing sessions with these power-user tips.

JULIAN RODGERS

F

ilm mixes are known for their astronomical track counts, but even music sessions, once limited by tape formats or host computer constraints, now regularly run into triple figures. With modern productions becoming more complex and timelines tighter, the ability to get around a session quickly and intuitively isn’t just convenient, it’s essential. Here we’ll concentrate on ways to get the track or tracks that need your attention into view quickly and intuitively — from faster ways to scroll directly to the track you need to techniques to bring only the groups of tracks you want to see onto your screen or directly under your fingers.

Dizzy Heights A computer screen has fixed dimensions.

Once a session has more tracks than can fit vertically, you have two options: shrink the tracks, or start scrolling. If you shrink them too much they become unusable, so in practice most of us end up doing both and relying on a mix of navigation techniques we’ve picked up over time.

A common first step when space runs out is to reduce track height in the Edit window.

You can do this by dragging the lower edge of a track header, but Pro Tools offers faster methods. Use Control+Up/Down (Mac) or Start+Up/Down (Windows) to cycle through height presets on selected tracks. To resize all tracks simultaneously, hold Option (Mac) or Alt (Windows) while adjusting one track.

In the Mix window, toggling Narrow Mix View with Command+Option+M (Mac) or Ctrl+Alt+M (Windows) allows more tracks to fit horizontally — this is particularly useful when working without a control surface.

A must‑have keyboard shortcut is Command+Opt+Ctrl+down arrow (or up arrow, both work the same) on Mac, or Start+Alt+Ctrl+up/down on Windows.

This keystroke adjusts track height to fit

Scroll to Track lets you instantly jump to any track using an autocompleting text search box.

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The Narrow Mix View lets you shrink the width of your channels so you can see more on screen at a time.

all tracks into the Edit window, or down to a minimum track height if there are too many. Very useful.

Scrolling In The Deep The simplest way to navigate large sessions is often just scrolling. Use the scroll wheel to move vertically in the Edit window, and use Shift‑scroll to move horizontally in the Mix window. Dragging the scroll bars works, but it’s inefficient.

Two much faster techniques are using Scroll to Track and Control+Shift-clicking in the Tracks List. Scroll to Track, found under the Track menu, opens a searchable dialogue of all tracks. Type a few letters and the dialogue autocompletes; select the track, and Pro Tools brings it into view instantly. It’s even faster with the shortcut Cmd+Option+F (Mac) or Ctrl+Alt+F (Windows).

You can also Control+Shift+Click (Mac) or Start+Shift+Click (Windows) a track in the Tracks List to scroll that track into view in both the Edit and Mix windows.

These techniques are ideal for any session as they require no setup, just quick, keyboard-friendly access.

even if they don’t control any parameters, especially since, once a group is created, you can also assign a VCA. We’ll discuss that later. With up to 104 groups available, a ‘null group’ with no active attributes can serve as a recallable visibility shortcut.

Memory Locations Memory locations in Pro Tools can store more than markers — they also recall track visibility, zoom settings and selections. Setting up memory locations to show specific groups of tracks (for example drums, vocals, effects and so on) allows you to jump between them using the same keystrokes as timeline markers.

To use this feature, display the tracks you want visible, create a new memory location, set Time Properties to None, and enable Track Visibility.

The benefit is that it builds on a familiar workflow. Many users already navigate the timeline using markers, so this adds track visibility control without requiring new muscle memory. For consistency, use the same numbering scheme across all your sessions (10 for all tracks and 11

Group Therapy As well as being essential for editing and mixing, groups can also be a helpful way to navigate large sessions. In the Groups List, Control‑click (Mac) or Start‑click (Windows) on a group name in the sidebar to hide all tracks apart from the member tracks of the clicked group. Repeat the action with Shift held to add additional groups to the view.

Groups are easy to create: just select your tracks and hit Command+G (Mac) Control+G (Windows). It’s entirely reasonable to set them up purely for navigation,

Groups aren’t just for mixing and editing: you can also use them to switch between different track visibility schemes.

for drums, for example) to make keypad recall instinctive. Just remember to update your ‘all tracks’ memory location when the session changes, to avoid recalling hidden or inactive tracks unintentionally.

Universal Pictures

Into The Folder Folder tracks are one of the most effective ways to manage track visibility. By building a folder hierarchy in your sessions and closing the folders you’re not working on, you can drastically reduce the number of visible tracks in your session, meaning a previously sprawling session fits neatly onto your screen.

Basic Folders group tracks for organisation only, while Routing Folders also handle audio routing. Folders can be nested, so you can have separate drum, bass and guitar folders, and then put those into an overall ‘band’ folder, collapsing or expanding them as needed to control what’s shown on screen.

You can open or close folders by clicking the folder icon in the track header. Use Shift+F to toggle the state of selected folders. Option-click (Mac) or Alt-click (Windows) opens or closes all your folders.

The Tracks List reflects this structure using indentation, allowing you to visualise the folder hierarchy at a glance.

You can expand, collapse, show or hide tracks directly from the list, which is often more convenient.

VCA Spill If you’re using VCA Masters, you gain more than just the equivalent of a master fader for the group and more convenient soloing for the grouped tracks. VCA Spill is also a powerful navigation tool when used with a control surface. When you double-tap Select on the VCA Master using your control surface, the surface will ‘spill’ the tracks controlled by that specific VCA, reassigning each fader to a track within that

When screen space is tight, the Universe Window provides a miniature, top-down view of the entire Edit timeline, with the currently visible area shown as a highlighted rectangle. Clicking and dragging this rectangle scrolls the Edit window

VCA group. Double-tapping Select again toggles it back to the un-spilled state.

This makes it easy to collapse a group of tracks and expand it again only when needed. Folder Spill also offers similar functionality, but VCAs had it first and because, unlike folders, VCAs can contain any combination of tracks, they remain a valuable tool for non-hierarchical workflows.

Control Surfaces Control surfaces bring their own navigation challenges. While mouse users are limited by screen real estate, control surfaces are limited by fader count. That said, even small surfaces can be effective when used well.

EuCon-enabled surfaces like the Avid S1 or S6 integrate deeply with Pro Tools.

Selecting a track in the Edit or Mix windows automatically brings it to a fader (unless disabled in EuCon settings). Folder and VCA Spill are both supported, making complex sessions far more navigable.

If your surface doesn’t follow selection automatically, the same Control+Shift‑click Memory locations don’t just have to relate to the timeline: they can also store zoom settings, track visibility, track heights and more.

accordingly. It’s not especially fast in practice — mouse-driven navigation tends to slow down power users — but it’s a very accessible option for those new to managing large sessions.

Toggle it with Option+7 (Mac) or Alt+7 (Windows).

(Mac) / Start+Shift‑click (Windows) shortcut mentioned earlier brings a clicked track to fader 1. This makes banking fast, even on simpler hardware. Used well, Folder Spill, VCA Spill, and selection-based banking make even an eight-fader controller feel much more capable.

One of EuCon’s most powerful features is Layouts — custom arrangements of tracks that can be recalled instantly.

A Layout can contain any tracks, in any order, independent of how they appear in the Edit or Mix windows. For example, one Layout might show only vocals, another just drums or keys. Layouts are often underused, but they’re a powerful way to bring the tracks you want to you rather than you having to go to them.

This round-up of tips only scratches the surface, but navigating your tracks, not just your timeline, is a core skill. As sessions grow larger, fluency in these navigation techniques allows you to work faster, stay focused, and concentrate on what really matters: making creative decisions.

Dual Screen Tip A great tip for dual-screen setups: you can Control+Shift‑click (Mac) or Start+Shift‑click (Windows) a track in one window (Edit or Mix) to bring it into view in the other, either at the top of the Edit window or far left of the Mix window. This shortcut is especially helpful when you’re working with one window per screen and need to keep both views in sync.

www.soundonsound.com / June 2025

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Q

Is it better to record through the Unison plug-ins on my UA Apollo Interface?

I’m curious about using UAD plug-ins.

Would there be a noticeable difference in recording directly and committing (printing) to a Unison plug-in, versus recording the vocal dry and afterwards adding the preamp plug-in as an insert?

I saw a few comments mentioning impedance, but I don’t understand how that will dramatically affect the sonic quality/characteristics.

SOS Forum post Sam Inglis, Editor In Chief The answer is... it depends! The usual recommendation is that the input impedance of a preamp should be at least 10 times the output impedance of the mic. That way, the mic is not ‘loaded’ by the preamp and, in theory, you get the truest representation of the signal generated at the capsule. However, some older mics, such as Shure’s SM57, were designed to work into preamps with much lower input impedances, and they can sound different when they do — you could say that the designers ‘voiced’ them with that loading effect in mind.

Mics with electronically balanced output stages (in other words, most modern active mics) are pretty much immune from these loading effects, and will sound the same regardless of preamp impedance.

Passive and transformer-balanced mics, on the other hand, can sound different.

So, in theory, if you’re using a modern capacitor mic with an electronically balanced output stage, it should sound the same whether you record through the plug-in in Unison mode, or record through a clean preamp and apply the plug-in later. But if you’re using an older mic, there may be a difference.

UA’s Unison plug-ins don’t just model analogue colour — they can also change the impedance of the preamp itself, which in turn can alter the response of some microphones while recording.

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Q

What’s the best way to treat noise on my guitar recordings?

When I record guitar in my little ‘studio’ at home, it always sounds kinda noisy.

I have a little old Fender Champion 600 tube amp, and I put a Shure SM57 on it. But there’s a bit of buzz (hum?) and it sounds kinda hissy too. I tried using the gate in Cubase, and it sort of works, but it only really gets rid of the noise when the guitar isn’t playing, and sometimes it cuts notes short too. What’s the best way to tackle this?

Darren Davies via email Matt Houghton, Reviews Editor I had a Champion 600. While not exactly versatile, it’s a real valve amp and I thought it sounded very decent for the price. Still, I remember it not being the quietest and there were issues with the small speaker’s sound and the cabinet vibrating — I had mine modded, and fitted the amp in a bigger cabinet with a nicer, larger speaker. Still, assuming all is well with your guitar and pedals, you should be able to get a reasonably good sound out of this amp through an SM57.

It’s always best where possible to minimise noise at source, so first I’d examine your whole signal chain: your guitar/pedals could well be contributing to the noise. For more on that sort of thing, check out our SOS February 2025 feature: https:// sosm.ag/tackling-noisy-guitar-recordings.

To then tackle any unwanted noise you record, I don’t recommend gates, mostly for the reasons you state. But hiss, buzz and hum can usually be pulled down effectively using a ‘fingerprint’ style spectral noise remover.

Spectral noise removers such as the one in iZotope RX are a far better option than a noise gate for removing hiss and buzz on guitar recordings.

I tend to use iZotope RX, but if your budget’s tight there are lots of options. Essentially, you record some amp sound without any playing then feed that sound into the plug-in so that it can learn the ‘fingerprint’ of the noise you want it to remove. To make this easy, I always try to capture, say, 10-30 seconds of ‘silence’ before or after the guitar take I’m recording and use that. The software can then remove the noise either offline (as with RX) or in real time during playback (like ReaFIR, from the free ReaPlugs bundle). Either way, you can’t play through them (the real-time ones add latency that will affect your ability to play well and in time), but you can always monitor your playing using the amp itself. There can be undesirable side effects if you push the processing too far, but you should be able to make very worthwhile improvements to your recorded sound this way, such that later using compression or other tools that can bring up the noise floor won’t sound unbearable!

“You should be able to make very worthwhile improvements to your recorded sound.”

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Scoring Squid Game

With the new series of Netflix’s record-breaking drama about to land, we find out how Squid Game and Parasite composer Jung Jae-il creates his mesmerising scores.

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137

MUSIC TECHNOLOGY MARKETING CHRIS KORFF

I

f your inbox is anything like mine, hardly a day goes by in which you don’t receive a flurry of emails telling you about the latest innovations in music technology. I can live without the emojis in the email headers (yes, I’m sure my mixes would indeed be ‘fire’ if only I bought your software), but I’ll never tire of scanning through a press release to see if a new product could slot into my studio workflow.

Scanning music technology press releases is my job, so I’ve got quite good at it, but it can still take me a surprisingly long time to sift through the waffle before

I learn that a “Game-changing innovation leveraging the latest technology” is a USB interface with a DI input on the front. A picture, though, speaks a thousand marketing words, so it’s often quicker to just spin the mouse wheel down to the bottom of the email and click the ‘images’ link so you can see whatever it is you’re being marketed.

From a purely practical point of view, a nice hi-res photo of the back panel of a piece of hardware will tell you volumes about it in a matter of seconds.

Take a peak at the rear panel of any audio interface, for example.

Does it have eight XLRs? If so, I can record drums with it.

Does it have ADAT I/O? Then

it could probably replace my current interface. Does it have a Dante port? Then I can’t afford it. And so on. A front-panel photo will likewise be instructive (although, curiously, the blanker and plainer the front is, the more likely it’ll cost thousands, especially if it also has a Dante port round the back).

But the real fun is in the ‘in use’ or ‘lifestyle’ photos.

They serve a useful purpose in giving you a sense of scale, so you can see just how tiny the latest Teenage Engineering gizmo is, or whether a new groovebox could fit in that bit of space between your mouse and your MIDI keyboard. They are also, of course, intended to inspire you to buy the product

“I’ll never tire of scanning through a press release to see if a new product could slot into my studio workflow.”

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June 2025 / www.soundonsound.com

in question. But for me, those photos are a chance to enjoy a world in which home studios are spacious, welcoming rooms, tastefully decorated in the Scandi-chic style, bathed in soft natural light, where there are no unsightly tangles of cables, where desks are free of clutter, and gear is artfully placed so that everything is in reach, inviting you to play. I can (and do) spend hours looking at these photos, only to look down at my own desk and see coffee rings, crumbs, cables and chaos. And I wonder if I could ever live like the invisible producers in these minimalist marketing studios do.

So, in their own way, these marketing departments do inspire me — if only to have a little tidy-up every now and then. And perhaps to wonder if I should buy a pot plant...

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