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BRICK LANE One module, seven compressors!

KPop Demon Hunters

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Expressive E Soliste

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Apogee Symphony Studio Affordable entry to immersive monitoring

SSL Revival 4000

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ON TEST: BUCHLA / STEINBERG / POLYEND / VALETON / NORDIC / NOISEWORKS / LYNX / NI / BRAINGASM / BITWIG / HH / HOTONE

TECHNIQUE: MID-SIDES RECORDING / PLOSIVE CONTROL / DAW WORKSHOPS

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LE ADER

IDEAL WORLD I

t’s easy to think of emulation as belonging to the digital world. The Emulator was one of the first successful samplers, and today, plug-ins emulate everything from microphones to mixing consoles. But there have been many products that used analogue technology to emulate other analogue systems. The Emulator itself was pre-dated by the Mellotron and the Chamberlin. Flangers replicated an effect first created using tape machines. Even guitar amp simulators were originally analogue devices, such as the Tom Scholz Rockman. In the last two decades, digital modelling has become so powerful that you might think there wasn’t much mileage left in the idea of analogue emulation. If you can accurately mimic some piece of vintage gear in a few lines of code, why bother developing and building complex analogue circuits to do the same thing? One reason is that analogue technology has also moved forward, with some of the biggest strides coming in the field of digital control over analogue circuits, allowing them to fit into a digital world where recallability and software control are expected. And while no piece of hardware will ever be as cheap to manufacture as a plug-in, analogue circuitry still has qualities that can’t easily be replicated in software. As an example, take McDSP’s Analog Processing Box, which uses digital switching and control to configure sophisticated analogue

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processing. Sonically, it achieves results that simply don’t seem to be possible using software alone. The same is true of this month’s cover product. Cranborne Audio’s Brick Lane 500 is a poster child for what can be done with cutting-edge digitally controlled analogue circuitry. Its audio signal path is fully analogue, and uses an innovative implementation of PWM compression to deliver astonishing technical specs — some of which are arguably impossible to match in software. (If you’re sceptical, consider that the Brick Lane’s attack time can go as low as six nanoseconds. That’s at least three orders of magnitude lower than the duration of a single sample, even in a plug-in that upsamples to 96kHz.) Thanks to its extensive digital control, the Brick Lane 500 can also emulate other styles of compression including opto, FET and even variable-mu valve circuits. Having spent a lot of time myself with the APB, I’m convinced that digitally controlled analogue is the best of both worlds. It can capture the essential sonic qualities of classic dynamics processors in a way that many feel conventional plug-ins don’t, but retains all the versatility and controllability of digital. And best of all, on a per-channel basis, devices like the Brick Lane 500 and APB are actually more affordable than most vintage-style compressors. The future is hybrid!

“Digitally controlled analogue is the best of both worlds.”

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

October 2025 / www.soundonsound.com

124 INSIDE TRACK

IN THIS ISSUE

www.soundonsound.com

October 2025 / issue 12 / volume 40

FEATURES 8 SOS For Artists This month, we set ourselves a real-world challenge: compose and release a new tune, using the tools provided in an SOS For Artists subscription!

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NEVE 88C WOR TH $1 195

91 Modular: SchneidersKeller SchneidersKeller’s Edd Butterworth and Jean-Marcel Fricke haven’t just opened a shop: they’re trying to start a community.

110 Ultimate Plosive Control Pesky pops proving problematic? Here’s how to pull them down...

112 Stereo Miking: Mid-Sides Our series on the fundamentals of stereo demystifies Mid-Sides and explains the practical advantages of an M-S mic array.

PAGE 64

120 Spotlight: Dante Interfaces Once the preserve of large studio facilities, Dante is now becoming common in smaller setups. We talk you through some of the options.

124 Inside Track: KPop Demon Hunters

Recording a band live in the room is always a challenge — especially when it’s led by a jazz legend who switches instruments without warning!

Five years in the making, Netflix’s smash hit animated movie tested its music production team to the limit.

148 Q&A

134 How I Got That Sound

Your studio and recording questions answered.

Producer, engineer and mixer Dave Sardy explains how he turned an acoustic guitar into a synth‑like drone on Oasis’ ‘(Get Off Your) High Horse Lady’.

136 Isabel Gracefield: Mulatu Astatke

October 2025 / www.soundonsound.com

154 Why I Love... Working In The Box Atheen Spencer doesn’t miss the bad old days of lugging heavy gear around the country.

74 LYNX MESA

ON TEST 36 Apogee Symphony Studio

98 IK Multimedia Prism Reverb Reverb Plug-in

99 LANG Electronics 312L 500-series Microphone Preamplifier

USB Audio Interface

86 Bitwig Connect 4/12

USB Audio Interface

Braingasm Ocula

66 Native Instruments Circular Software Sequencer & Sound Engine

C O V E R

Analogue Synthesizer

58 Cranborne Audio Brick Lane 500 Modal PWM Compressor

146 Native Instruments Scene:

Nightshade Sample Library

52 Expressive E Soliste Software Instrument

46 Noiseworks DynAssist Vocal Pre-production Software

32 Nordic Audio Labs NU-24K

Analogue Mixer

96 Hotone Verbera Convolution Reverb Pedal

144 Sonora Cinematic Softstruck Sample Library

20 SSL Revival 4000 Input Channel

24 Steinberg SpectraLayers Pro 12 Spectral Editing Software

98 Strymon Cloudburst Reverb Plug-in

90 TiNRS Bopp & Steve Eurorack Module

28 Valeton GP-5 Multi-effects Pedal

Sample Library

94 HH Q16FX

Polyend Press Stereo VCA Compressor Pedal

144 Nine Angles For Forte Spat Curve

146 Fracture Sounds

Orchestral Atmospheres

Majetone Dream Kit: Zenith Software Synthesizer

Diffuse Field Microphone

80 Buchla Music Easel

Lynx Mesa Thunderbolt Audio Interface

Capacitor Microphone

Omnitone Rhythmi & Beatsi Eurorack Module

WORKSHOPS 100 Studio One 104 Logic 106 Pro Tools 108 Cubase www.soundonsound.com / October 2025

TECHNIQUE

This month, we set ourselves a real-world challenge: compose and release a new tune, using the tools provided in an SOS For Artists subscription!

THE SOS TEAM

he last few SOS For Artists (SOSFA) columns have been orientation sessions really, to take you through some of the key features. This time, we decided we should get more hands-on, so

we set ourselves a challenge: use only the tools available as part of SOSFA to create a track from start to finish — right through to mixing, mastering, release and promotion. As we do that, we’ll dig deeper into the many tools provided in SOSFA. You can find an audio example detailing the progress

An overview of the Logic project for the first stages of our nu‑disco track, the composition of which is explained in the main text.

made this month on the SOS website at: https://sosm.ag/sosfa-1025. The brief we gave ourselves was to create and release a nu‑disco track, but obviously the principles apply more widely and we’ll also be working with other genres in future workshops. Since lots of us are familiar with Apple’s Logic, we’re using that as the host DAW for this project, but pretty much everything we do here can be done using the ‘lite’ versions of Cubase and Ableton available as part of the SOSFA subscription — or, for that matter, any other DAW.

Electric Piano

The AAS Session Bundle, included with SOS For Artists, had a lot to offer for this project!

October 2025 / www.soundonsound.com

Every good nu‑disco track needs a funky electric piano (EP), and that’s catered for by the excellent AAS Lounge Lizard Session 4 plug-in, which is part of the AAS Session Bundle in SOSFA. So the first task is to get that installed. Installation is quick and painless (it took us less than five minutes and just 66.6MB of drive space). It’s probably worth noting at this point that to download and authorise the plug-ins that feature in your subscription, you’ll need to create an account on the relevant manufacturer’s website if you don’t have one already. We found that it made the EP

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TECHNIQUE

C R E ATING & R E LE A SING A N U ‑ DIS CO T R ACK : PA RT 1

The Lounge Lizard settings used for the electric piano sound described in this article.

more fun to play in this genre if we dialled in a touch of the built-in room reverb, and added a little bite to the sound by increasing the Hammer and Fork settings.

Into The Groove With Lounge Lizard up and running, our next step was to build a groove and sketch out some chords. The quickest way? Grabbing a few beats from the Sample section of the SOSFA dashboard. At this stage, you probably want to get something together quickly so you have something solid to write to — so your choices don’t necessarily need to make the final edit. In terms of tempo, 124 bpm seems to be around the sweet spot for nu‑disco that will keep it club-friendly and not seem out of place alongside other tracks in the genre. There’s so much content in the SOSFA Samples section that it’s a good plan, in the Search tab, to select the Advanced Filters to narrow your search options down. For this particular search we filtered by drums only, dance genre and a tempo in the range of 118-128 bpm. To make the search quicker, we then homed in on Disco and House, which returned three pages of options — with more time you might want to rummage through a few more genres to get some fresh ideas, but you get the point. We set the search to ‘loop’, so as to exclude one-shots — again, the aim is to be quick and keep the creative juices flowing, and you can always come back and select some one-shots later for the final beat construction, or even use the sampler to chop up the chosen loops, as we did in last month’s workshop. Auditioning beats is quick and easy, and a couple stood out right away, so we dropped them into the Creator section at

October 2025 / www.soundonsound.com

the bottom of the SOSFA Samples page, just to hear how well they worked together. Once the first sample is added to that panel, you can switch the bpm to match your project. You can load up to eight samples at once in the Creator section, and while you’re audition them it’s a great time to mark good loops as favourites, and add them to specific collections for later use. If you get into this sort of habit, writing should become even quicker over time. We ended up with eight loops for this project, and although the kicks in a couple of the samples didn’t quite gel, on balance we had a great-feeling groove, so we decided to move on, and made a note to address any clashes in the DAW later on.

With your sample auditioning complete, you need to download and import your loops into your DAW, and you have a couple of options. You can just download the original stereo files, to get started quickly. Alternatively, you can download the processed tracks from the player, in which case they’ll already be at the correct tempo for your project. Since one of the best tools for warping tracks is Ableton Live, and we have a version of Live Lite right here in the SOSFA dashboard, we decided to grab the unprocessed samples by dropping them into the Collections folder in our account, and download them from there. Handily, the Creator is still showing up at the bottom of the screen when you go into your Collections tab, so it’s easy to cross‑reference and make sure that you’ve got everything. You can also clearly see what you’ve already favourited and downloaded as the icons become highlighted, so there’s no duplication of effort now or in the future.

Warped Thinking Although beat warping can be done in other DAWs, the simplicity and cleanliness of doing this in Ableton is hard to beat. With the beats stored in a folder, it was quick to drag them each onto their own track in Ableton. As luck would have it, the original tempo for all of them was 120 bpm, so they could be warped in one go. Selecting all tracks, warping in Beats mode and using the command Warp From Here (start at 124 bm)

Using the advanced filters in the SOSFA sample search narrowed down the results, making it easy to find suitable loops.

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TECHNIQUE

C R E ATING & R E LE A SING A N U ‑ DIS CO T R ACK : PA RT 1

had this lot sync’ed in seconds at the new tempo. Then, it was a matter of exporting the Warped beats into a new folder in the main project, and shifting the original beats to the main Samples folder for future use (keep everything tidy like this, you’ll thank yourself later!). These were then dropped onto their own tracks in Logic, and grouped into a Summing Stack (again, keeping everything tidy to make processing easier). A happy side‑effect of the warping process was that it seemed to tidy up those clashing kicks.

Chord Sequence & Electric Piano Groove This beat now gave us a groove to play with — something to noodle over for the next few hours. Of course, it would be possible to speed this process up by grabbing some EP loops from the Samples tab and develop them, but for this project we wanted to come up with an original groove and chord sequence; really, that’s the fun part, isn’t it? For some of you, it might help to drop a looped bass line in to groove over too. In the interests of getting this project done by the deadline for this article, we opted for a very simple four-bar pattern as the main groove, and later added a B and C section for variety. Heavily quantising keyboard parts can kill that live performance feel, so here we just used it to tweak the odd notes, to bring them into alignment where they part clashed with others. Another trick to help retain the all-important vibe is to shift whole chords at once, rather than individual notes, making sure that the prominent note of the chord hits close to the beat.

Put The Bass In The next part of creating our groove would normally be to structure the entire piece, write out the chord sequence and then bring in a real bassist... And we have tools coming soon that can help you with that. But in the interests of demonstrating what you can do with SOSFA’s tools already, we created a synth bass line using the GForce Axess plug-in. We kept to a fairly simple bass line, just to flesh the track out a little, and developed it more during a dedicated arrangement stage, once other parts had been added. To add thickness, we doubled this part using another sound from the AAS Ultra Analog plug-in. During the mixing stage, the Axess handled more of the sub frequencies, and the Ultra Analog some of the higher, more

October 2025 / www.soundonsound.com

tonally distinctive frequencies, to make sure that the bass line remains audible even on phone-speaker playback.

A String Arrangement No nu‑disco track would be complete without a string part, and we found just the tool for the job in the form of Sonuscore Elements. This runs in NI’s Kontakt sampler, and while it’s a cut-down version, it still offers us plenty of useful options. We ended up using three articulations — the Staccato, Legato and Tremolo options — which added some nice variety to the part. The legato strings are actually doubled with the staccato, the latter set at a lower velocity, just to give a faster start to the legato note. That’s not always necessary, but can help when working in an uptempo genre like this.

Structure & Arrangement Next, it was time to start developing the structure of the track, so it was a matter of copying and pasting the beat to last for around three and a half minutes, then dropping the various parts across it to create plenty of variety, and prevent the track from feeling repetitive. There’s still a lot of work to do on the arrangement, and things will change further still during the mixing stage, where we can start bringing in some fun effects. The beat was continued for four bars at the end of the track too, with DJ transitions in mind, although you can also provide the track stems during release for this purpose. One of the first things we wanted to do was to bring in a piano part to double up on the EP for thickness, and to add some nice, jazzy incidental parts. The Spitfire Originals piano is perfect for this. Some piano plug-ins can sit very low in the mix so, for now, we

just gave it a boost using the Dynamics control, duplicated the EP groove to the piano track, then added a further track for incidental parts. The main string riff is played three times in this track, and we wanted this section to build with the inclusion of new, different parts each time. Normally we’d suggest hiring a real guitarist to add a new part, but since the brief was to demonstrate what’s available in SOSFA, we turned to the AAS Strum Session plug-in, to see if it could introduce more rhythm in the second and third instances of the main riff. This great little plug-in has a bank each of electric and acoustic sounds, and a selection of strum patterns that you can add to each. We used three MIDI notes to trigger a chord sequence, a minor key (Cm in this instance) and the chosen loop pattern. Although there are full strumming patterns available, we opted for a fairly sparse one, partly because it suits the genre but also because it allowed us to blend the part in more easily, to disguise the fact that it’s not a real guitarist! The rhythm it provided really lifted the main section, and drove things along nicely. When we get around to adding some effects to this part, it should sound reasonably convincing to the untrained ear. Next up, to increase the energy and sense of build‑up even more, we decided to add a synth part towards the end of the track. Since we already had the AAS Ultra Analog plug-in loaded, we used this. There are some really useful presets in this plug-in, and we opted for a multitimbral one that worked well for both pad and lead parts — a great starting point, though some slight tweaking was required to get it to sit nicely alongside the other sounds.

The Creator tab lets you audition multiple samples from the SOSFA online library together. You can match their tempo, and figure out what works well and what doesn’t before downloading.

A fresh part was created to lift the original chord sequence and fill in any gaps in the arrangement. A short riff created for the B section was pasted into the Strum track to see how it would sound doubled and the result almost sounds like a vocoder in the mix — a happy accident that worked, so we decided to keep it! At this stage, just two more elements were missing: percussion and risers. For this, we quickly scanned through the samples section again, and downloaded a percussion loop. Although it was OK ‘as is’, we ended up starting it on beat 2 of the bar, just to get a little more swing, with a view to tweaking it more later as the arrangement develops. Likewise, we found a couple of risers, one quite a high, swooshing sound and another deeper sound. These were dropped in just before the final instance of the main groove and will be developed further using effects and cymbal crashes at the end of the sounds to mitigate the abrupt endings.

Next Time... And that brings us up to date. For now, the main structure of the track is built,

The version of Ableton included in SOSFA provides a quick and easy way to time-stretch your samples.

and we have a very rough mix balance — albeit one with parts that currently overlap too much in terms both of frequencies and stereo positioning. The next stages will be to further develop the beat, and then dive into the mixing and mastering using the plug-ins available in SOSFA.

The final steps on this project will be to go through the process of creating artwork, adding metadata, releasing the track and promoting it. That’s not an easy feat without the assistance of a major‑label budget, but there are a few tricks that we can use!

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www.soundonsound.com / October 2025

ON TE ST

Majetone Dream Kit: Zenith

Software Synthesizer

Built entirely from percussion sounds, but not for percussion, Dream Kit: Zenith is a synth like no other. MARK GORDON

uestion: when is a drum kit not a drum kit? Answer: when it’s a synth. Let me explain... The drum kit, or indeed synth, in question is Dream Kit: Zenith by Majetone. It’s a VST and AU instrument for Mac and Windows that reimagines the drum kit as a fully chromatic, synthesizer-style sound source, featuring over 21GB of multisampled waveforms that utilise not just drums themselves, but a wealth of component parts such as lugs, shells, hoops and even squeaky pedals. Despite its name, Dream Kit: Zenith is not your regular virtual drum plug-in, but neither is it your regular synth. While all the waveforms originate from drum and percussion sources, the primary goal of James Yates, the creative brain behind Dream Kit: Zenith, was to make some part of a drum kit stand in for each of the major food groups found in synths: Rhythmic, Harmonic, Melodic, Bass, and Sound FX. So has he achieved this?

Kit & Caboodle The plug-in occupies a single window, its simple layout and colour scheme giving it a ’70s retro look and feel. The display is split into three discrete areas — Waveform 1, Waveform 2 and a central Master section — working in combination to create the 1000+ presets. The two Waveform sections on the left and right are identical and enable you to load and manipulate the 120 supplied waveforms, which include multisampled variations with up to 38 velocity layers, multiple round robins and unique one-shots. Recorded using a range of high-end studio mics, each waveform has been tuned across five octaves and re-amped at Otterhead Studios, to offer close, overhead and room options that can be independently balanced. Looking at the list of waveforms, you begin to appreciate what sets this plug-in apart and gives it its unique sound. In addition to drum shells hit with mallets and thumbs drawn across various drum heads (which sound incredible), there are also recordings of a broken china

October 2025 / www.soundonsound.com

Dream Kit: Zenith occupies a single window but includes a wide array of editable parameters. However, the 1000+ presets are so impressive you may not feel the need to edit anything at all.

cymbal thrown across a room and floor tom legs played with a cello bow! The Waveform section of the plug-in offers a wide range of re‑shaping and manipulation tools, with pitch, pan and reverse parameters, coupled with the familiar ADSR, EQ (based on a one-off German console) and LFO. A less conventional component is a discrete arpeggiator featuring Step, Octave, Speed and Swing parameters, enabling you to create ‘Reich-like’ rhythms and melodies between the two waveforms, which can be incredibly effective with the more percussive examples. Effects are also well catered for, with the inclusion of a great-sounding distortion, delay, and a unique convolution reverb featuring springs and plates, alongside less conventional offerings such as stairways, bunkers, and a tunnel on the A34. The centrally located Master section is where presets are loaded and saved, and the overall output volume is controlled. This is also where the sub‑oscillator resides, allowing you to introduce sine, triangle, saw, square and noise waveforms, which can add weight and body and make Dream Kit: Zenith effectively a three-oscillator synth. The presets themselves are divided into seven categories — Kits, Majetone, Reverse, Snares, Sustain, Synth and Toms — and cover all the bases you can imagine... plus a great many you couldn’t! You’d be

forgiven for thinking Dream Kit: Zenith is a synth-tweaker’s dream, but if you only ever used the 1000+ presets that ship with the plug-in and never adjusted a parameter in anger, I think you’d be more than happy with your investment.

Conclusion Dream Kit: Zenith has its origins in the world of drums and percussion, and this is certainly apparent from many of the presets. However, the vast majority have been manipulated and morphed into something unique, inspiring, and worlds away from what you might expect. Ethereal pads, organic basses, retro electronica and eerie soundscapes sit alongside processed concert toms and snare drums, creating a unique and inspirational palette of sounds that is exciting, creative and (most importantly) very usable. They say there’s a fine line between madness and genius. Dream Kit: Zenith sits very comfortably on the genius side.

summary Dream Kit: Zenth reimagines the drum kit as fully chromatic synth, transforming over 21GB of of drum samples into an extensive collection of processed percussion, ethereal pads, organic basses, retro electronica and much, much more.

$ $199 W www.majetone.com

ON TE ST

Polyend Press Stereo VCA Compressor Pedal M AT T H O U G H TO N

here was a time when the best you could hope for from a compressor pedal was a few knobs and a half‑decent chance of increasing an instrument’s sustain. I never really got on with that sort of thing — it just felt clumsy and unsatisfying to me, and I much preferred studio-style compressors, whose settings I could dial in more precisely. Thankfully, recent years have seen quite a few more capable dynamics processors packed into pedal-sized boxes. As well as the various excellent analogue-modelling digital devices, we’ve seen a range of good analogue options, including FET compressors (for example from Origin Effects), optical compressors (such as the JoeMeek Floor-Q) and some excellent VCA compressors (from Becos FX, amongst others). And some of them do much more than control the attack, sustain and dynamic range of a signal, for example offering side-chain control, dual bands, EQ and filtering, or saturation/distortion.

Press Pack The latest company to tread this path are Polyend, who I’d hitherto thought of mostly as a company whose focus was firmly on

October 2025 / www.soundonsound.com

It may come in the form of a stompbox, but there’s studio-quality processing inside — and more than enough control for serious mixing. the synth market. But while their new Press VCA compressor pedal caters for that crowd, of course, it also boasts a number of interesting features that should mean it has just as much appeal for guitarists and bassists, and, in fact, pretty much anyone who’s looking for a pedal or tabletop compressor to use on stage or in the studio. With a footprint of about 105mm (a shade over four inches) by 120mm (4.75 inches), it takes up a chunk of pedalboard real estate if that’s where you want to put it, but it’s not unreasonably large. It’s powered by a 9V, centre-negative PSU of the sort usually used for guitar effects and pedalboards. There isn’t one included and there’s no internal battery option; you’ll need a PSU capable of supplying 350mA. The power inlet is on the back panel, of course, and there you’ll also find the main I/O on two quarter-inch TRS jacks. The input/output jacks cater for mono instrument or unbalanced stereo line-level I/O, selected using a slide switch. You could, for example feed the left and right TS jack outs from a synth or drum machine

into it, using a Y-lead, or connect a guitar directly to it with the usual TS jack cable. There are also two further connectors on the back: TRS mini‑jacks that provide a thru output, for passing the incoming signal on to other devices, and an external side-chain input, for triggering compression from sources other than the one being processed. So, for instance, you could send a kick signal to duck a bass, or to get a whole mix pumping in time to the beat. Everything else of interest is on the top. Here, there are seven bright-yellow, cylindrical knobs, accompanied by two smaller black knobs and two 10mm footswitches. You’ll also find a 10-LED gain reduction meter, and status LEDS for the switches. It looks great, but I did find that some of the labels were often unhelpfully obscured from view by the tall knobs, and personally I’d have preferred all of the labels to be as bright as those on the left — some are darker and harder to read against the black, and the numbers beneath the gain-reduction meter are so muted as to be almost unreadable. Still, you soon

MODULAR INSTRUMENT SYNTHESIZER WITH ANALOG AND DIGITAL SYNTH HERITAGE, ADVANCED PROCESSING AND HIGH PERFORMANCE SIGNAL PATH

ON TE ST

P O LY E N D P R E S S

get a feel for what everything does, so none of this really impacts on operation in the longer term. The footswitch on the right is for on/ bypass, while that on the left brings the external side-chain into play. The main compressor controls are on the yellow knobs, all of which turn continuous analogue pots. Again, dark labels beneath serve as an aide memoire of the values, so let’s put them in clearer black and white! The threshold can be set anywhere from -30 to +10 dB. The ratio ranges from a bus-compressor-friendly 2:1 right up to infinity:1 (effectively limiting). The separate attack and release knobs are both marked from 1 to 10, and there’s 0-30 dB of make-up gain available. The other two yellow knobs are Mix, a wet/ dry blend control that gives you instant access to parallel compression, and Tilt, which applies a tilt EQ with a fixed centre frequency. Turn this anticlockwise for more bass and less treble, and clockwise for the opposite. It acts on the wet signal only, but if you set the threshold control for no gain reduction and the mix control to 100 percent wet, you can use this on the input signal, which can be incredibly useful.

Filtration Station So there’s plenty of control to start with, but the final two knobs (the smaller black ones on the left) make this compressor particularly versatile: one is a high-pass filter (20Hz to 1kHz) and the other a low-pass filter (100Hz to 20kHz), and they act on the side-chain signal (whether internal or external, this is the control signal used to trigger the compression; shape this and you change how the compressor reacts). Most of us are accustomed to using high-pass filters in the side-chain to reduce a compressor’s sensitivity to low frequencies, but adding a low-pass filter to this lets you do so much more. For instance, if feeding a drum loop into the external side-chain, you could roll the high-pass filter down so it has no effect, and bring the low-pass filter down until compression is triggered only by the low‑frequency energy of the kick. Alternatively, you could use the two in tandem as a ‘bracket filter’, and really target a narrow band, for example as a de-esser. You get the idea: extra control and versatility.

In Use I tested the Press using a range of different material, including vocals, acoustic and

October 2025 / www.soundonsound.com

As well as the main mono/stereo I/O, there are also mini-jacks for an external side-chain input, and to pass the input/side-chain signal on to another device.

electronic drums, acoustic and electric guitar, bass, and some synths and keys. The first thing that struck me was just how clean and transparent the unit was when not applying any gain reduction. It may have unbalanced I/O but noise wasn’t ever a problem, and when sending line-level signals out from my audio interface there seemed to be plenty of headroom. The sound is everything you’d expect of a VCA compressor: it can act very fast if you want that, or you can set it up for a more relaxed feel. It might not deliver quite the sonic vibe of a FET or opto compressor, but dialling in the desired result is quick and easy, despite those ‘camouflaged’ labels. The yellow knobs have a black line to indicate the position, while the black ones have a white line, and these high-contrast markings make it easy to see what’s what. You soon find that a quick glance is enough to take in the whole unit’s settings, or to set things up from your notes or a recall sheet. There are some presets laid out like that in the manual, and if you’re less used to studio compressors, these can be great for initial orientation. The footswitches, given plenty of space, work well too: easy to hit with a foot, while pressing with fingers doesn’t require undue pressure. Polyend put great emphasis on the importance of the high- and low‑pass side-chain filters, and that’s understandable: they really help when you want to focus the action only on sounds

you wish to squish. But for me, Tilt is the star of the show. It’s a simple idea really, but I’ve only ever encountered a tilt EQ on one other company’s pedals (Becos FX’s Stella range), and they’re great for making broad darker/brighter tonal shifts that don’t otherwise mess with the sound. And that can be really handy for correcting any tonal changes resulting from more aggressive compression.

Hot Off The Press What this all boils down to is a compressor that is stylish, neatly assembled, versatile and easy to use. I love that it can be deployed on mono or stereo sources; it’s just as at home smoothing my (reasonable) guitar playing or pinning my (somewhat ham-fisted!) bass playing in place, as it is parallel-processing a drum bus, just tickling the mix bus, or injecting a side-chain-driven bounce into an EDM track. Given the quality, its stereo capability, and the bells and whistles, it’s decently priced too. So if you’re looking for a portable analogue compressor, this one’s well worth checking out.

“The sound is everything you’d expect of a VCA compressor... Dialling in the desired result is quick and easy.”

summary A high-quality analogue compressor in a format that you can take anywhere, and deploy on pretty much anything!

$ $399. W https://polyend.com

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S S L R E V I VA L 4 0 0 0

SSL Revival 4000 Input Channel

SAM INGLIS

n illustrious heritage can be a mixed blessing. On the one hand, you have brand recognition that newer companies would kill for. On the other, people can be reluctant to accept that a manufacturer with a heritage might want to move on from it. Shure, for example, have produced some outstanding mics in recent years, but the SM57 remains the best-seller. On this side of the Atlantic, Solid State Logic have continually pushed the technological envelope, first with super-sophisticated analogue consoles like

October 2025 / www.soundonsound.com

Solid State Logic dust off the blueprints of their E-series console to create a classy channel strip.

the J- and K-series, then with hybrid designs such as the AWS and Duality. Today, those who associate the SSL name primarily with innovation can point to the digital T-series mixer and the fully recallable yet analogue Oracle as beacons of this philosophy. But there will always be people who hanker after the ‘SSL sound’ of consoles from the 1980s, and the company have been pretty good at catering to this market too. As long ago as 2006, SSL’s XLogic range included a faithful 1U recreation of the G-series bus compressor, as well as a product called the E-Signature Channel, which offered pretty much the

full feature set of a channel strip from the revered E-series console, with some additional bells and whistles. Today’s range includes 500-series recreations of various elements from their mixers, as well as plug-in emulations of most of the sought-after SSL designs. And, as of now, it also features a 1U rackmounting hardware processor that offers pretty much the full feature set of a channel strip from the revered E-series console...

The Revival The new Revival 4000 is, however, far from being a reissue of the

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S S L R E V I VA L 4 0 0 0

XLogic E-Signature Channel. To the core feature set, that device added a switchable alternative mic preamp circuit employing SSL’s transformerless VHD technology, and a second dynamics stage that recreated the notorious Listen Mic Compressor. It also featured a slot for an optional A-D converter card. All three are absent from the Revival 4000, but this time around, SSL have added a de-esser and a balanced insert point. There are also numerous points of difference concerning how this is all switched and laid out, and the cosmetics of the two units are quite unlike, but perhaps the biggest difference concerns the price. The E-Signature Channel retailed at around $5000, placing it out of reach of most project and home-studio owners. Since then, SSL have built up offshore manufacturing facilities that have allowed them to drastically reduce prices, and the Revival 4000 costs just $1999. Offshoring once carried connotations of shonky build quality, but every SSL hardware product I’ve reviewed has been rock solid, and the Revival 4000 is no exception. With its thick metal fascia, smoothly and evenly weighted controls with detents at the zero positions, and clear, precise legending, it feels every bit a professional product. Nor have SSL cheaped out in any way on the design; the Revival 4000 has an integrated PSU, and practically every switched function has an LED to let you know whether it’s active or not. Internally, the mic preamp uses the same Jensen JT-115K-E transformer found in the original E-series channel strip. All the audio I/O is on the rear panel, starting with an XLR mic input and a combi XLR/jack line input. (The latter is not designed to accommodate instruments that need to see a high input impedance.) Insert send and return, on balanced jacks, are followed by a compressor side-chain input on another combi socket, and a separate stereo link jack. Finally, the output signal emerges on a male XLR.

True Grit The SSL mic preamp might not have the cachet of some other vintage designs, but it’s certainly versatile. The variable gain control runs from +20 to +70 dB, and there’s also an input pad and a trim control with a ±20dB range that applies to both mic and line inputs. Consequently, if you want to drive it harder for a bit of warmth, that’s

October 2025 / www.soundonsound.com

very much possible, while a huge amount of gain is available through the system as a whole, as there’s a further ±20dB trim on the output stage. The de-esser is a simple one-knob affair with a three-LED gain-reduction meter, but the main dynamics section presents the ‘full fat’ SSL channel compressor and expander/gate. In fact, it’s more comprehensive than the implementation in their recent 4K G plug-in, thanks to its switchable hard-knee and linear-release options. Release times are fully variable, as are the compressor ratio and expander/ gate range, but in both cases there’s the traditional choice of two fixed attack times. A single button switches between gate (1:infinity) and expander (1:2) mode, and the compressor and expander/gate have separate five-segment LED meters. A quirk of SSL’s design is that the gainreduction meters on linked units remain independent, even though they are both compressing by the same amount.

“There’s something that just feels indefinably ‘right’ about the SSL EQ.” For further control over the dynamics, it’s possible to switch either or both of the main filter and EQ sections into the compressor side-chain. However, the provision of a dedicated de-esser eliminates one of the main reasons to do this, allowing you to use the EQ for tone-shaping instead. And both the filters and the EQ are extremely powerful tools for this purpose. The filters have a huge range, extending up to 500Hz for the high-pass and down to 3kHz for the low-pass, while the main EQ section is highly versatile. The low-mid and high-mid bands are fully parametric, and their ranges overlap significantly, so you can for instance notch out a resonance whilst applying a broadband boost in the same region. The low and high bands can be switched between shelving and bell modes, and a BLK button switches the EQ response to ‘black knob’ mode, as found in most SSL consoles after 1983. In both modes, there’s something that just feels indefinably ‘right’ about the SSL EQ. With ±15dB range on every band, it’s capable of radically altering the timbre of any signal, yet it rarely sounds strained or harsh. Conversely, even small

adjustments have an authoritative effect, so you’re never left wondering whether your settings are actually doing anything.

All Singing I think most of us accept nowadays that if we want ultra-precise control, super-fine detail and comprehensive recall, software is the way to go. Our reasons for choosing hardware are more to do with the broader sonic qualities that it can impart, whether that be coloration from valves and transformers or the unique dynamic characteristics of opto cells and VCAs. In doing so, we are often sacrificing features in return for that character. What’s striking about the SSL Revival 4000 is that although it’s consciously billed as a vintage-inspired product, it never feels limited in terms of functionality. If anything, in fact, you could argue that it has more features than most of us are likely to need, and that tasks such as de-essing and gating are best left to the mix and handled by plug-ins. That’s no reflection on SSL’s circuits, which work perfectly well, but it’s not as though a de-esser really has analogue mojo. Personally, therefore, I’d perhaps rather SSL had retained the Listen Mic Compressor from the E-Signature Channel in place of the de-esser, though the expander/gate has always been a core feature of the original SSL console channel strip. Some vintage-inspired processors are one-trick ponies, imparting their sonic qualities pretty much regardless of how they’re set up. By contrast, the Revival 4000 is endlessly versatile. Yes, you can coax some grit and colour from it, but you have to make a conscious decision to do so. Used normally, it’s just a very good front end for any source in any style of music — and, thanks to its separate line input, an excellent line-level processor at mixdown. In fact, it says everything you need to know about the SSL console’s influence on music that this actually feels like a perfectly contemporary product. It may be billed as a revival, but it never really went away!

summary The Revival 4000 demonstrates exactly how much SSL got right when they designed the original E-series console!

$ $1999 W www.solidstatelogic.com

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Steinberg SpectraLayers Pro 12

Spectral Editing Software

It’s only 12 months since the last version, but SpectraLayers has made some huge advances. JOHN WALDEN

he world of spectral editing is on something of a roll, much of it driven by the integration of AI-based algorithms into many of the typical processing and restoration tasks for which it is most often used. It’s perhaps not surprising, therefore, that this product category is seeing a pretty rapid upgrade cycle as manufactures try to keep up with — and ahead of — their competition. All of which perhaps explains why Steinberg have adopted an annual upgrade cycle for SpectraLayers Pro, and we now have version 12. So, just what’s new in the world of audio editing magic within SLP12?

How To Unbake A Cake There is ‘new and improved’ scattered throughout the v12 update but, for many users, the highlights will be the further developments in SpectraLayer’s ability to do the audio equivalent of unbaking a cake. Four of the Unmix modules — Song, Drums, Noisy Speech and Transcription — plus the Voice DeNoise module get improvements in v12. For example, the Unmix Noisy Speech module, which separates an audio recording into ‘speech’ and ‘noise’ layers, generates noticeably cleaner results, making it easier to improve the intelligibility of a spoken voice in a post-production setting, even when working with what might seem like potentially challenging source material. From a music production perspective, it’s the Unmix Song and Unmix Drums modules that will attract the most attention. In testing both with a range of materials, the step up from v11 was very obvious. With Unmix Song, for example, the extracted vocal layer, which contains all the lead/ backing/harmony vocals within the original mix, is obviously cleaner than with v11. Indeed, the quality of all the resulting layers is getting remarkably close to having a set

October 2025 / www.soundonsound.com

SpectraLayers Pro 12 makes sophisticated spectral-based complex audio editing tasks incredibly accessible to any level of user.

of mix busses — drums, vocals, guitars, bass, piano, sax/brass and other — to work with, albeit with any effects associated with each instrument group already baked in. In use, the thing I found most remarkable was that, in the main, the majority of any remaining artefacts are focused on the effects elements of the original sounds (for example, sounds heavily processed by modulation effects); the fundamental sounds were generally dealt with very cleanly. This includes the drum and bass layers. Are they perfect? Well, no, but Steinberg have made significant steps forward here, particularly in terms of the cleanness of the drum transients (I’ll come back to transients later), while the bass appears much more solid. To my ears at least, SLP12 has now set a quality bar in terms of full song unmixing for others to match. The Unmix Drums module brings both improved results and new layer options as you now get six layers in total: kick, snare, hi-hat, toms, ride and crash. The kick/snare separation is generally very clean (and the snare layer does a respectable job with ghost notes) but perhaps the most notable improvement I found was with the hi-hat, which, in most cases, seemed crisper than before. Depending upon your use case, and the nature of the drum submix you are dealing with (straight from the mixing desk or an unmixed drum layer), you may still find yourself with some manual editing to do. However, if all you need to do is make minor adjustments to the drum balance or use the

resulting drum layers as triggers to replace or enforce the sounds with samples, it can facilitate either of these tasks.

New Modules SLP12 adds four new modules: Unmix Instrument, Unmix Soundtrack, Voice Enhance and DePlosive. Unmix Instrument is an interesting new option in that it allows you to identify any single sound as an unmixing target. The only catch is that you have to ‘register’ the required sound first and this really requires a few seconds (up to 10) where the sound can be heard in isolation. In practice, this may not always be possible but the concept is interesting so I hope it is something that is developed over time. In contrast, I think post-production audio engineers will find the new Unmix

Steinberg SpectraLayers Pro 12 $349 pros • Very noticeable improvements within the AI-based processing modules.

cons • Rapid upgrade cycle may frustrate more occasional users.

summary Steinberg continue to make rapid progress with SpectraLayers Pro, with this release bringing very obvious technical and operational improvements.

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Nordic Audio Labs NU-24K Capacitor Microphone

This highly impressive new large-diaphragm mic brings a touch of Nordic noir to the studio. SAM INGLIS

innish engineer Martin Kantola has been building microphones for more than 30 years. Like many ‘boutique’ makers, he was initially inspired by the classic Neumann valve mics, but in 2019, he started building a new large-diaphragm capsule to his own design. The D100K is a 30mm, edge-terminated capsule for which Nordic Audio Labs claim “excellent off-axis phase response, super realistic transient response and [a] very smooth top end”. So far, this capsule has been offered in the multi-pattern NU-100K and the NU-880K, which uses two D100K capsules mounted at a 90-degree mutual angle to offer stereo and horizontal B-format output. Nordic Audio Labs also manufacture the unique NU-314K, a large-diaphragm pressure mic with a pure omni pattern, and the NU-47V, which aims to match the sound of the immortal Neumann U47 but without using valves. Apart from the NU-47V, all Nordic mics share a similar look and feel, and they’re very much premium products, typically costing several thousand pounds. For those who want to explore the advantages of the D100K capsule in a somewhat more affordable package, Nordic Audio Labs have now launched the NU-24K: a no-frills, fixed-cardioid mic

October 2025 / www.soundonsound.com

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Apogee Symphony Studio Designed with immersive audio in mind, the Symphony Studio is also a top‑class all-round audio interface. PHIL WARD

pogee’s new Symphony Studio range of USB interfaces is aimed particularly towards immersive audio production duties. In that context, and like the Audient ORIA interface I wrote about in SOS July 2024, the headline specification that substantially defines each of the three Symphony Studio models is not so much their input count but their output count. To put some numbers on that, the three models in the range offer two inputs and 12 outputs (the 2x12), eight ins/eight outs (the 8x8), and eight ins/16 outs (the 8x16).

Apogee Symphony Studio From $2199 pros • Hugely versatile immersive output configuration. • Really great‑sounding inputs. • Comprehensive control from the Apogee Control 2 app.

cons • No I/O expansion through ADAT or audio over IP. • No integration with room/monitor optimisation.

summary Apogee’s Symphony Studio is a jack of all trades, but it has them all pretty much mastered. Highly recommended.

October 2025 / www.soundonsound.com

Now, you might wonder about how suitable the 8x8 model is for immersive duties, because with ‘only’ eight outputs the largest immersive monitor array the 8x8 can address would be, for example, 5.1.2 (left, centre, right, left surround, right surround, LFE, left height, right height). While perhaps being a viable initial route into the world of immersive monitoring, that falls short of the generally accepted Dolby Atmos bed minimum of 7.1.2. So the 8x8 Symphony Studio model feels like a kind of hybrid between an interface aimed squarely at the contemporary immersive world and a traditional one intended for more conventional recording and mix duties. In terms of retail price, the 2x12 is the entrylevel Symphony Studio model and its output count does place it firmly in the realm of immersive mix duties. The 12 outputs enable it to address what I’d consider the sweet spot of entry-level immersive monitoring: 7.1.4. But with just two inputs, you’re only going to be tracking the odd overdub. At the other end of the Symphony Studio range, the 8x16 is not only able to address a full-fat Dolby Atmos 9.1.6 monitoring system, but its eight inputs can potentially track a modest band. That 8x16 capability comes at a cost though, because it commands a significantly higher price than the 2x12 and 8x8. Before I leave the I/O counts behind, something to consider, especially in regard to the 2x12 and 8x8 models, is that their I/O counts are absolute. Symphony Studio

USB Audio Interface models offer no input or output expansion via, for example, ADAT or Dante. So if you were to settle on the 2x12 but then found you needed more inputs, or decided on the 8x8 and subsequently wanted to expand your immersive monitoring, you’d be out of luck.

Three Of A Kind Beyond their I/O counts the three Symphony Studio models are functionally the same. I was loaned an 8x16 model for this review, but everything I’m about to write

Round the back things are surprisingly simple, with just eight XLR inputs, twin D-Sub ports for the 16 outputs, and an IEC mains socket.

is equally appropriate to the 2x12 and 8x16. Symphony Studio models are all housed in the same 1U rackmount enclosure. The subtly profiled and classily finished aluminium front panel combines Apogee’s corporate purple with a gloss‑black centre section that houses an OLED display. Along with the display, the front panel is populated by eight buttons in two groups of four, and a rotary encoder/push button. Both sets of buttons switch the contents of the display and the action of the rotary encoder to offer the appropriate configuration functions. The left‑hand group of buttons provides input selection for level adjustment and configuration via the rotary encoder, enables input‑specific phantom power, and toggles the Apogee ‘Soft Limit’ function on and off. The right‑hand group of buttons provides access to basic monitor control functions (volume, mute, dim), headphone output selection and control, and a Home function that returns the menu to the top level. Finally on the right of the front panel are two headphone sockets (one 6.3mm and one 3.5mm) and a power switch.

On the rear panel is a line of XLR input sockets that reflect the particular model’s input count. The input sockets are conventional XLRs, without combi instrument jack sockets, so if you want to connect a guitar, for example, some species of unbalanced jack to XLR conversion will be required — although at the kind of professional level a Symphony Studio is likely to find itself, I’d imagine a guitar preamp or at least a DI box will be available. Adjacent to the input XLRs are, depending on the particular Symphony Studio model, one or two DB25 D-Sub output sockets for monitor connection. Each DB25 provides eight balanced outputs. D-Sub sockets are reasonably common on pro hardware that’s intended for permanent rack installation, but they are less often seen on products, such as the Symphony Studio, that might find themselves in less permanent home or project studios. In a rack‑installed scenario, a D-Sub socket will likely be connected to a patchbay of some description that offers signal access via jack or XLR, but in studios a rung or two down the

Screen 1: The Apogee Control 2 home display shows inputs on the left and outputs on the right.

professional hierarchy, connecting monitor cables terminated in XLR or TRS jack plugs demands some cumbersome D-Sub to XLR or jack break-out cables. An option of course would be to have some custom fulllength D-Sub‑to‑monitor cables made, but that won’t be inexpensive if the monitoring system comprises, say, 12 speakers. I can see why Apogee would go the D-Sub route for Symphony Studio outputs (cost saving and rear‑panel real-estate) but for some users I think D-Sub might fall into the curveball category. Apart from the mains power input, the last rear‑panel feature to mention is the obligatory USB-C socket for DAW host connection. A cool touch is that the particular USB socket fitted, and the USB-C cable supplied, incorporate a retaining thumbscrew feature that ensures there’s no chance of the plug being inadvertently pulled. I liked that — it makes USB feel a bit more ‘pro’.

Hardware & Software While the core functionality of the Symphony Studio can be configured from its front‑panel buttons and rotary encoder (or

www.soundonsound.com / October 2025

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APOGEE SYMPHONY STUDIO

to gently limit signals that come within 2dB of A‑D clipping. The Soft Limiting occurs before the input signal reaches the DSP‑based ECS channel strip, so if it were to be engaged after the ECS compressor/ limiter parameters have been configured, those parameters might require revisiting. Apogee claim that the ECS results in no extra latency and a quick check I made confirmed that to be the case (although ECS did appear to add about 1dB to wide-band level even when set to 0dB gain). While I’m on the subject, I measured basic in/out latency of the Studio Symphony at under 3ms (48kHz sampling rate).

Mostly Software

Screen 2: The Apogee ECS channel strip plug-in can be inserted on each Symphony Studio input.

a USB‑connected Apogee Control hardware desktop controller), its partnering DAW host application, called Apogee Control 2, both duplicates the front panel functions and offers considerably more — and that’s an understatement. For example, the first Apogee Control 2 screenshot illustrates its home display with general settings on the left and the eight input channels adjacent to the right. The input channel button functionality is all pretty much self-evident, but hidden behind the FX button is a feature that sets the Symphony Studio apart from lesser interfaces: a fully featured, DSP‑powered Apogee ECS channel strip plug-in. Pressing the FX button opens the plug-in, illustrated in Screen 2. The ECS plug-in, rather than consuming DAW host processing power, resides within the Symphony Studio hardware and there’s effectively no limit on the number of instances that can be used. If you want the ECS channel strip active on all inputs that’s no problem. The mic preamps offer up to 75dB of gain combined with an EIN (equivalent input noise) spec of -129dB, which puts them towards the top end of mic preamp spectrum. Downstream of the inputs, the Symphony Studio, say Apogee, “harnesses the same cutting-edge A‑D/D‑A converters

October 2025 / www.soundonsound.com

found in the renowned Symphony MkII, delivering unparalleled sonic accuracy and transparency”. I’d not indulge in such hyperbole in my analysis, but Apogee have long been renowned for very high‑performance converter technology, so I’d fully expect very high quality. Also available on Symphony Studio inputs is Apogee’s Soft Limiting technology. Apogee Soft Limiting operates in the analogue domain (rather than within the DSP engine)

Returning to the Apogee Control 2 app, to the right of the input section on the home display screenshot are output channel strips that offer metering, gain adjustment, mute and solo functions. At the bottom of each channel are Speaker Set buttons, and these enable outputs potentially to be assigned to alternative monitors. Speaker Sets, and what Apogee have christened Monitor Workflows, are fundamental elements of the Studio Symphony operational infrastructure and to a large extent a consequence of its immersive audio role. A Monitor Workflow is a stored Symphony Studio preset that covers channel routing, room correction (output/room EQ), speaker delay and bass management — the last three of which I’ll cover a few paragraphs hence. The Symphony Studio 8x16 ships with four Monitor Workflow presets pre-configured in terms of channel routing for stereo, 5.1, 7.1.4 and 9.1.6 monitoring. Four further preset slots are available. The pre-configured Monitor Workflow presets have no room

Screen 3: Selecting Monitor Workflows in Apogee Control 2 opens the monitor configuration window.

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APOGEE SYMPHONY STUDIO

Screen 4: The Room Correction window provides 16 bands of EQ, and adjustable delay on each monitor output channel.

correction or delays applied and have bass management disabled. One important point to make concerning Monitor Workflows is that selecting a Workflow comprising fewer channels than the output material (say you select a 2.0 workflow when outputting 7.1.4 material) doesn’t fold the multi-channel stream down to stereo (in Atmos terms, it’s not a re-render). It will simply mute all but the left and right outputs. This is by no means a criticism of the Symphony Studio or Apogee Control 2, because re-renders are not its job (the Atmos renderer is responsible for creating re-renders), but worth bearing in mind. The second element of monitor management is Speaker Sets. Apogee Control 2 provides switchable Speaker Sets that enable Monitor Workflow presets to be assigned to different sets of speakers. This enables, for example, alternative monitors to be used for 2.0 and Atmos playback. My studio is a perfect example of this, where 7.1.4 Atmos monitoring is done over a system comprising 11 Dynaudio BM5 MkIIIs and one Dynaudio Sub 18, whereas stereo mostly happens on a pair of Neumann KH150s. Up to three different Speaker Sets are available and each can have different Monitor Workflows assigned; however,

October 2025 / www.soundonsound.com

alternative Speaker Set options are only available if the Symphony Studio has free outputs beyond those consumed by Atmos monitoring. Again, my studio works well in this respect because the 12 outputs of 7.1.4 monitoring leave four free for alternative Speaker Sets. Returning to Screen 1 for a moment, it reflects a custom Monitor Workflow I created (called 7.1.4B) where Symphony Studio outputs 1‑12 are assigned to Speaker Set 1 (the Dynaudio 7.1.4 system), and outputs 13+14 are assigned to Speaker Set 2 (the stereo Neumann KH150s). For completeness I also assigned channels 15+16 to Speaker Set 3, although they were left unused. Speaker Sets and Monitor Workflows can be selected on the fly from buttons in the panel on the bottom right of the main display. Configuring Monitor Workflows is done in the Apogee Control 2 Monitor Workflow home display, illustrated in Screen 3. My 7.1.4B Monitor Workflow is selected in the screenshot but the assignment of channels 13‑16 to Speaker Sets 2 and 3 can also be seen. The monitor icons in the display can be clicked either to solo or mute monitors depending on the setting at the top

right of the display. Along the top of the Monitor Workflow main display are buttons that open the In/Out routing, Bass Management setup and Room Correction windows. The In/Out routing and Bass Management windows and options are generally self-explanatory and intuitive, but Room Correction deserves some explanation so I’ve illustrated it in Screen 4.

Read The Room The room correction EQ and delays, powered by the Symphony Studio’s internal DSP, offer 16 bands of EQ for each output. Each band can be set to peak, low-pass, high-pass, low-shelf or high-shelf functions. Delays can also be applied to each output channel. You’ll see from the screenshot that some EQ has been applied and this was generated by an Apogee Control 2 room acoustics correction routine, accessed by clicking on the cube line drawing at the top right of the Room Correction display. Screen 5 illustrates said routine. The dimensions and room surface materials I’ve input in the screenshot reflect my studio, although the reverberation time (2.83ms) that the routine predicts is nowhere near the measured value of between 0.3ms and

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ON TE ST

APOGEE SYMPHONY STUDIO

0.6ms. The proposed EQ curves it applies looks somewhat generic too. Furthermore the correction routine also applies the same EQ to all the outputs but the LFE, and that’s not really adequate for serious room correction, to my way of thinking. This is all of course because the routine takes no account of actual monitor locations in the room, their inherent acoustic signature or any absorption or diffusion. In Apogee’s defence, they do state in the Symphony Studio user manual that the room correction routine is only a starting point, and they suggest the use of third‑party applications that can comprehensively analyse room/ monitor acoustics and export EQ and delay coefficients (Sonarworks SoundID or Room EQ Wizard, for example). I’d endorse that wholeheartedly, although I’m not sure I’d want to be responsible for manually inputting frequency, level and Q values for 16 EQ bands on up to 16 output channels. What’s missing here of course is full integration with a room analysis and correction system such as the Audient ORIA or Universal Audio Apollo, for example, have with Sonarworks. Finally, in Apogee Control 2, there’s the headphone configuration and control section on the right‑hand side of the display.

ALTERNATIVES USB interfaces with enough outputs for Atmos are still relatively thin on the ground. I’ve already mentioned the Audient ORIA, and if two inputs are enough, it competes quite closely with the Symphony Studio 2x12. With its more restricted output count the Symphony Studio 8x8 has numerous competitors from brands such as Focusrite, SSL and PreSonus, but the Symphony Studio 8x16 is pretty much out on its own in the USB interface world.

recordings of a variety of basses and acoustic guitars and came away from those experiments deeply impressed by the effectively silent and wonderfully transparent Symphony Studio mic preamps, and by the sound‑sculpting opportunities of the ECS channel strip plug-in (I found its preset patches are notably worthwhile). There’s something genuinely engaging and creatively inspiring about tracking through the Symphony Studio and ECS.

Conclusion I’ve been describing the Symphony Studio 8x16 as a 16‑output device but, of course, two stereo headphone sockets effectively add another four. Each headphone output can be independently configured within Apogee Control 2 to be fed from any two DAW playback channels, from the Apogee Control 2 internal mixer or directly from a Symphony Studio input. You couldn’t really wish for any more headphone configuration options.

One Last Thing Before I wrap up, in all this talk of outputs and monitoring functionality I’ve perhaps lost sight of the input and tracking side of things. So I made a few simple test

The Symphony Studio is deeply impressive. It’s a hugely capable and versatile interface that does pretty much everything required in a contemporary immersive studio environment, and more. On the tracking side, the very high‑performance mic inputs in combination with the ECS channel strip plug-in offer genuinely professional studio quality, while on the multi-channel output side, the degree of versatility and configurability and control is exactly the kind of thing an immersive mix facility needs. $ Symphony Studio 2x12 $2199, 8x8 $2999, 8x16 $3999.

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Screen 5: Apogee Control 2 incorporates a rudimentary room modelling function that generates EQ curves based on room size and predicted acoustics.

October 2025 / www.soundonsound.com

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ON TE ST

EXPRESSIVE E SOLISTE

Expressive E Soliste Expressive E’s trademark physical modelling paired with MPE control delivers an extremely playable instrument.

ROBIN BIGWOOD

ith the French company Expressive E, the clue is most definitely in the name. These days they may be best known for the Osmose, their groundbreaking MPE (MIDI Polyphonic Expression) synth and controller keyboard, but they develop virtual instruments too and Soliste is the latest. It’s a suite of modelled solo string instruments — violin, viola, cello and double bass — that run in recent versions of macOS or Windows as VST2, VST3 or Audio Unit plug-ins, with three activations possible via a virtual or hardware iLok. And it’s definitely built with real-time expressivity in mind. Acoustically modelled instruments typically promise greater versatility and a more involved playing experience than their sample-based counterparts, and are often magnitudes smaller in installation size. It’s an especially attractive concept for a virtual bowed string instrument; in a solo guise they’ve always been notoriously difficult to deliver in sampled form. On top of that Soliste has enhanced compatibility with specialised MIDI controllers: the Osmose first and foremost, and also Expressive E’s standalone gesture controller, Touché. But it will work with other MPE controllers and even quite basic non-MPE keyboards too.

October 2025 / www.soundonsound.com

Fired up in your computer, all the separate Soliste instruments present with identical controls in clean, modern-looking, three-way resizeable windows. They’re designed to give musical results straight off the bat, but there’s some considerable sound‑design depth available should you want to explore it.

Open Strings A Soliste plug-in window presents three main control areas. In the middle, below the big graphic depiction of the instrument, you get to choose a timbre characteristic based on a notional ‘resonant body’ material. VLN.356, Soliste’s violin, offers a choice of six: Spark, Hollow, Sleek, Antique, Lyric and Mellow. As you step through them the large instrument graphic changes colour and texture, so if you fancy trying a Mellow yellow and white violin, or a Hollow one made of cork, it’s there. The other instruments offer either four or five materials, incidentally. To the left are a few controls relating to pitch, and particularly vibrato. Assisted vibrato will do a pitch wobble for you, with variable amplitude, frequency and fade-in time. If you use your pitch-bend wheel, or a key-generated bend from an MPE controller, that’s taken into account too, adding to the effect. You can also turn the whole thing off, for completely manual control and sleek non-vibrato tones.

Software Instrument

Then, to the right, appropriately, are controls relating to the bow, and thus the actuation and excitement of the virtual strings. To dig in with this aspect of the plug-in you’ll want to click the ‘more’ button, which opens up a page-full of parameters. Pitch also gets a ‘more’ page, offering deeper, more finessed control of the auto-vibrato (especially in its sensitivity to MPE pressure, aftertouch, and configurable MIDI CC messages). There’s also a portamento section there, that’s compatible with the Osmose’s see-saw Pressure Glide feature, or can introduce inter-note swoops based on key velocity thresholds. Jumping straight in, using an Osmose as the MIDI controller, and using the default Classic Mode preset, a freshly instantiated Soliste instrument delivers impressive expressivity with essentially zero effort. Typical MPE behaviour is to the fore: sounds can creep in, triggered by slow and shallow key-touches, but then swell and intensify radically as you press deep into the aftertouch. Play legato and the melodic line is sewn together with brief but complex transitions; in particular, wide jumps down to a lower note sometimes take a moment to settle, and for the fundamental harmonic to fully emerge. Key-actuated vibrato and

Expressive E Soliste $299 pros • Engaging, complex sound quality, broad timbral range and first-class playability. • Bespoke handling of MPE MIDI controllers like the Osmose, but usable with almost all. • Comparatively small installation size of about 400MB per instrument.

cons • Not notably naturalistic-sounding... but that’s arguably not what Soliste is ultimately about. • Lack of keyswitched articulations hampers real-world use, for some jobs. • Double (let alone triple) stops are not supported.

summary A physically modelled quartet of orchestral string instruments, with great expressive potential for both MPE and conventional MIDI controllers. The sound strongly channels acoustic behaviour but offers a specific, attractive quality all its own.

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ON TE ST

Cranborne Audio Brick Lane 500 Modal PWM Compressor

Cranborne say their Brick Lane is the fastest compressor in the West — but it’s also capable of modelling the characteristics of classic opto, FET and valve units, and much more. HUGH ROBJOHNS

n just seven years, Cranborne Audio have built a very fine reputation for their steadily expanding range of innovative, high-performance products. Cranborne’s ethos is to push the boundaries of what’s possible with analogue audio electronics, both technologically and in terms of what the user can achieve with their products. Their approach has resulted in remarkably transparent products with exemplary specifications, combined with the ability to introduce delicious musical character at the twist of a knob or flick of a switch. For example, the company’s debut product, the Camden mic preamp (reviewed in SOS October 2018: https://sosm.ag/cranborne-camden500), remains unquestionably amongst the finest mic preamps available at any price, yet it also offers a glorious range of vintage tonalities if required. Their Carnaby EQs (the latest being the Carnaby HE2, reviewed in SOS October 2024: https://sosm.ag/ cranborne-carnaby-he2) offer a new approach to EQ and tone-shaping. And talking of exemplary specs, their 500-series racks incorporate exceptional digital conversion that comfortably outperforms many high-end mastering converters.

Overview The latest addition to Cranborne’s product portfolio is their first compressor — a single-slot, 500-series mono module that continues the company’s predilection for London-inspired names, called the Brick Lane. Its full title, though, the Brick Lane 500 Modal PWM Compressor, hints that this is no run-of-the-mill compressor... Whereas traditional analogue compressors might employ an electro-optical cell, vari-mu valve, FET or VCA as a gain-controlling element, the

October 2025 / www.soundonsound.com

Cranborne Audio Brick Lane 500 $599 pros • Astounding technical performance and highly skilful implementation. • Incredibly transparent PWM compression with beautifully optimised parameters. • Seven highly effective emulations of traditional compressor topologies, with variable saturation effects. • Simple control set allowing fast, intuitive operation. • Comprehensive deep-dive parameters for detailed customisation. • OptoSync module control linking. • Astounding value for money.

cons • None.

summary Another extraordinarily versatile and great-sounding product from Cranborne! In the Brick Lane they’ve extracted phenomenal performance and variety from a skilful implementation of sophisticated PWM compressor technology.

A Brick Lane installed in an API 500-6b Lunchbox rack. Note the Stress control beneath the meters, and the seven operating modes below it — for more about those, see the ‘Compression Modes & Emulations’ box.

Brick Lane 500 uses a sophisticated pulse width modulation process instead (see box). This technology has been employed in dynamic range compressors before, but remains relatively unusual; it offers

significant technical benefits over more traditional solutions, but it’s also very complicated and, in the past, expensive to implement. Its implementation has therefore been restricted mostly to

high-end manufacturers like Dave Hill/ Crane Song, George Massenburg Labs and Great River. Nevertheless, PWM technology theoretically affords the very best technical performance that an analogue compressor can deliver, particularly in terms of speed, ultra-low distortion (especially at high gain reductions), control linearity and consistency. And, since digital side-chain control suits the technology very well, this approach can also allow unrivalled operational versatility.

PWM Compression: A Primer While Cranborne’s implementation is novel, pulse width modulation compression itself is not a new technology. It evolved in the late 1960s, when fast transistors became available at low cost. A few compressors employed the PWM technique in the ’60s and ’70s, including the EMT 156 and PYE 4060 — models that now command huge second-hand prices, and have a reputation for sounding smooth and transparent, with significantly lower distortion at high gain reductions compared with contemporary designs that used other topologies. Conventional compressors employ a gain-control element in the signal path to reduce the amplitude of the output signal, as determined by ‘side-chain’ circuitry, which analyses the signal (either from the input or output) to produce a control voltage that varies the gain reduction. That gain-control element is usually configured as a continuously adjustable voltage divider, essentially acting as a remote-controlled volume knob. In contrast, the PWM concept is a radically different approach. It’s equivalent to an output switch being flicked on and off at an incredibly high speed (typically around 250kHz): when the switch is open, the audio signal’s

energy is allowed through to the output, and when it’s closed, it isn’t. If the ratio of open and closed durations is varied for each switching period, a precise amount of audio energy (and hence signal volume) is allowed through to the output. For example, if the pulse width is 100 percent, then all of the source energy is passed to the output and there is no attenuation, whereas if the pulse width is reduced to 50 percent of the switching period, only half the energy is passed, giving a 6dB reduction in level. This is clearly a form of audio sampling, yet the pulse-width modulated samples are never quantised and so never become ‘digital’ — it remains a fully analogue signal path. However, both anti-alias and reconstruction filtering are still required to adhere to the Nyquist theorem. Those early PYE and EMT compressors (and some more modern PWM compressors too) rely on very complex analogue circuitry to control the PWM process. By contrast, the Brick Lane uses a digital microcontroller that affords incredibly accurate and consistent control, but also allows compression parameters to be modelled in myriad ways to perfect absolutely precise and transparent dynamic control — or to emulate the subtle musical imperfections of vintage compressors.

www.soundonsound.com / October 2025

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CRANBORNE AUDIO BRICK L ANE 500

So, right off the bat, the PWM technology (carefully optimised by Cranborne’s team, which is no small challenge, actually!) makes the Brick Lane an inherently ultra-clean, transparent, and stunningly fast compressor. And that’s great, if that’s what you need — but we all know that compression is often much more about musicality and character than it is about speed and precision, so Cranborne have conceived the Brick Lane to be configurable to accurately emulate a variety of conventional compressor technologies, too.

idea to control fast transients without distortion. Remember, this is an analogue processor, so the sort of look‑ahead we might be used to in plug-ins isn’t available. And rather than pass the signal through a delay line to allow the

enough for very effective transient control with a near-zero attack time, allowing true brickwall limiting when required. Like other Cranborne products, the Brick Lane compressor also incorporates its own form of saturation processing. The Camden preamp has its Mojo control, for example, while the Carnaby EQ has its Harmonic EQ, and in a similar vein, the Brick Lane offers the user a Stress control. This feature involves a multiband saturation circuit that, combined with other compressor parameter variations, replicates the musical characteristics associated with different analogue compressor technologies. Six distinct flavours are available, with the Stress knob adjusting the strength of the carefully tailored saturation effects.

“The Brick Lane 500 includes a novel variation on the ‘look-ahead’ idea to control fast transients without distortion.”

Looking Forward There’s some genuine innovation here: the Brick Lane 500 includes a novel variation on the ‘look-ahead’

side-chain a sneak preview (which is how the HUM Audio LAAL’s analogue look‑ahead function works, but introduces latency), the Brick Lane uses a new digital technology called ‘negative group delay’. This fascinating science essentially preempts transient peaks by a quarter-wavelength (at any frequency). That might not sound much but it’s

It serves no purpose and might end up hidden in the rack... but you have to admire the graffiti-adorned brickwork aesthetic!

October 2025 / www.soundonsound.com

Compression Modes & Emulations The Brick Lane features seven different operating modes, which have been designed to emulate more traditional compressor topologies: • Velvet represents the characteristics of a vari-mu valve compressor in a feedback topology, with RMS sensing, an initial 2:1 ratio, and programme-dependent release, giving a smooth, warm charm. • Float emulates a typical gentle 3:1 opto-compressor, again with RMS sensing, fast attack but with a 60ms lag, two-stage programme-dependent release, and side-chain pre-emphasis creating a brighter character than the Velvet mode. • Smash replicates a stiff, feed-forward, 8:1 FET compressor with fast peak detection, giving a punchy, dynamic sound character. • Tame mode is another feed-forward configuration, with a 4:1 ratio and a wonderfully smooth triple side-chain sensing format combining peak, RMS and long-RMS sensing. This is effectively a pure PWM compressor with a stunningly clean, transparent character, but retaining tight control of transients. • Glue is a typical bus-style feed-forward compressor with a 2:1 ratio and RMS sensing intended for mixing and mastering, creating a very cohesive sound and a smooth “transformery warmth”. • The two Polish modes are intended for peak limiting in mastering, using a feed-forward topology with infinity:1 ratio. The white version provides very fast and hard peak limiting with a look-ahead function for ultimate control of transients, while the blue version has a soft-knee character and gentler clipping, which sounds more like analogue tape limiting.

Also, like the Carnaby Harmonic EQ, two or more Brick Lane compressors can be linked for stereo or multi-channel operation in two ways. A 3.5mm TRS socket near the back of the module caters for wired connections between modules, but most users will employ the company’s ingenious OptoSync feature where an LED transmitter on the right of each module near the front panel beams across to a corresponding opto-sensor on the left of adjacent modules, sharing control information. The leftmost module in the OptoSync group determines control settings for all linked modules. The main benefit of the wired alternative is that it allows the two modules to be placed anywhere in the host 500-series rack — it might be more convenient, for example, if you were to assemble two identical channel strips, each comprising preamp, EQ and compressor.

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ON TE ST

Native Instruments Circular Software Sequencer & Sound Engine NI’s Circular offers a dizzying range of creative possibilities. JOHN WALDEN

esigned by Frank Elting (the mind behind three of NI’s popular granular Kontakt instruments, Straylight, Pharlight and Ashlight), and with sound‑design input from the Most Human Colors, Samuel Estes and the Solos, NI’s Circular comes from a stellar team. It also boasts an intriguing concept, UI and feature set. So, is Circular going to help you keep your musical world going round? Time to explore...

Circular Tour Let’s start with a brief summary of Circular’s main features. In essence, Circular provides a four-layer, sample-based sound engine for use within Kontakt (full version or Kontakt Player). It ships with some 9GB of sample data which suggests the 168 included sounds are deeply sampled. The sounds themselves are derived from a wide range of sources including guitars, music boxes, prepared pianos, tuned metalwork, various brass instruments, vocals and a range of synths, some familiar and some processed to give a more unusual sonic flavour. The individual sounds are very classy and easily good enough to make the sonic cut in even the most demanding of musical contexts. From these source sounds, the design team have built around 250 presets (Kontakt Snapshots) divided into Atmospheres and

Native Instruments Circular $199 pros • Fantastic sounds for modern scoring applications and electronic music styles. • Very deep feature set.

cons • For some, the very deep feature set!

summary With a high-quality sound set, and a powerful four-layer sequencing feature, Circular has plenty to offer media and electronic music composers provided they are prepared to dig deep.

October 2025 / www.soundonsound.com

The central display in Circular’s Play page provides easy access to some of the sequencing options while also being hypnotic to watch.

Sequences. You can, of course, build your own Snapshots based upon the included sounds and, rather wonderfully, you can also load your own (single) samples into any of Circular’s four sound layers and then make use of the full sequencing, effects and modulation options the engine provides. A four-layer sound engine is a familiar virtual instrument format but where Circular gets interesting is in the features it combines to trigger, sequence, modulate and process those sounds. This is also where things get pretty deep and, while there is a lot of instant gratification to be had from the supplied Snapshots, I think it’s fair to say that Circular does require you to dip more than a toe in the shallow end if you want to fully appreciate what it is capable of. Space precludes a comprehensive description here, but I’ll try to summarise the main elements before exploring one or two of the more interesting elements in more detail to give a sense of what’s possible. Hold on tight...

Circular Simplicity? Circular’s controls are laid out across four main pages — Play, Sequencer, FX and Mixer — accessed via the tab headers contained within the ever-present toolbar at the top of the UI. The toolbar also

provides access to three macro-style knobs (part of the modulation system; by default, the Variation knob is linked to your mod wheel for hands-on operation), the main Snapshot preset browser, key/ scale settings (no duff notes allowed from Circular’s powerful sequencing/arpeggiation features) and global Settings (Circular can support polyphonic aftertouch and MPE; the response can be adjusted here). The Play page provides easy access to the four sound layers. You can pick the individual sounds, adjust their relative levels, and decide which incoming MIDI notes (highest, lowest, all) each layer will be triggered by. However, via the very attractive central circular graphic, you can also begin to create and edit sequences for each of the four layers. This graphical interface is obviously where Circular derives its name from. It reminds me of Patterning, one of my favourite iOS music apps (see the October 2018 review) and, as we will see in a minute, it shares one key feature with that app. Selecting one of the four layers — A, B, C or D — brings that to the foreground for manual editing within the circular display and also pops open a panel that provides options for generating either random or Euclidean patterns or enabling a MIDI record function. The latter lets you play Circular in real time

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ON TE ST

Braingasm Ocula Diffuse Field Microphone This unusual-looking mic is designed specifically for Decca Tree arrays.

HUGH ROBJOHNS

raingasm are a specialist microphone repairer based in Rome, Italy. In addition to servicing, restoration and modification of vintage capacitor microphones (including re-skinning capsules), the company’s lab also undertakes repairs of a wide range of outboard studio

October 2025 / www.soundonsound.com

equipment, and even offers an audio electronics design service. These days it seems quite a lot of microphone repair companies branch out into manufacturing their own microphone designs, and Braingasm are no exception, offering a small but very interesting range of studio capacitor microphones. The current portfolio includes the MP1 small-diaphragm cardioid mic

(sold in stereo pairs), along with three large‑diaphragm models: the Ferro (fixed cardioid) and Multideluxe (multi‑pattern) mics both have FET-based impedance converters, while the Magnitude is a multi-pattern valve microphone. These microphones employ a range of different technologies including transformer or transformerless outputs, valve or FET impedance converters, fixed or variable polar patterns, large or small diaphragms — and so offer not only different features but also different tonalities, making each one interesting and characterful in its own unique way. Yet they are all comfortingly familiar to anyone conversant with the typical fare of standard studio microphones. However, the subject of this review has a far less familiar format — indeed, it’s unique as far as I know! The Ocula is described as a Diffuse Field Recording System, but if that means nothing, perhaps if I say it’s available in both dual‑ and triple‑matched sets, the latter might give a better clue as to its design. For these are compact, omnidirectional electret microphones with the capsule housed within a custom-made Acoustic Pressure Equalising sphere (see the ‘APE Concept’ box). This configuration was first introduced in the revered Neumann M50 microphone, which has become virtually synonymous with the Decca Tree stereo technique employed widely in classical orchestral and film soundtrack recordings. Although Neumann ceased production of the M50 in 1971, it was replaced by the TLM50 in 1990 (now discontinued) and then the M150 Tube model from 2001. The APE concept is not unique to Neumann, though, and several other manufacturers offer APE spheres as accessories for their own small-diaphragm omnidirectional microphones. Nevertheless, the Braingasm Ocula is unique, as far as I know, in being a compact omni microphone with a permanently attached APE.

Ball Games Pub quiz afficionados will no doubt be aware that an American pool ball is 57mm in diameter, while a British snooker ball is slightly smaller at 52.5mm. I mention this only to give a familiar reference: the shiny black Ocula microphone is a little smaller than both, having a diameter of 40mm — the same diameter as the Neumann M50 design.

Each Ocula mic weighs around 56g and has an integral 3/8-inch threaded insert at the bottom for attaching to a mic stand. The model name and serial number are engraved onto the top of the sphere, while the audio output is carried on a fixed, 5m long (3.4mm diameter) Sommer cable exiting from the rear of the sphere. I found this cable prone to tangling, but a Velcro tie is attached to help maintain the coils in some semblance of order when stored. A genuine Neutrik XLR plug is fitted to the cable, but inside is a tiny circuit board carrying four SMD transistors and a handful of resistors and capacitors to convert between the unbalanced capsule signal and the electronically balanced XLR output. Standard 48V phantom power is required, and this is converted by the circuit board to the 3V bias required by the capsule. The capsule is mounted in a rubber grommet, with the grille slightly proud of the ball surface (presumably because the diaphragm is recessed within the electret’s metal housing). Braingasm have used a high‑quality 10mm omnidirectional electret capsule manufactured in Japan by Primo, the EM-1272-Z1. Specs-wise, Braingasm’s published specs align exactly with Primo’s, stating self-noise to be a very respectable 14dBA, with a high sensitivity of 39mV/Pa, and a maximum SPL of around 119dB. That self-noise figure is comparable to many

small-diaphragm DC-biased capacitor microphones, and is truly excellent for a 10mm electret. Rather than state the complete microphone’s frequency response or provide a response chart, the specs only give the ‘amplifier response’ (10Hz to 35kHz), which isn’t very helpful... but a datasheet on the Primo website shows the full frequency response chart of the raw capsule, which is essentially flat from 60Hz up to 2kHz, above which the response rises gently, reaching +4dB at 10kHz before tailing off to -6dB at 20kHz. I suspect the absence of data below 60Hz relates only to the size of the company’s measuring facility — the Ocula delivered a fine and well-extended LF response during my evaluations. However, that raw capsule response will be substantially modified by the presence of the APE sphere, resulting in a boosted high-end response due to the physics of acoustic pressure

Braingasm Ocula From $812 pros • Intriguing entry into true Decca Tree mic techniques. • Elegant engineering. • A very compact microphone housing with low visual impact. • Supplied with a versatile ball-mount for easy positioning. • Low-noise, high-output capsule.

cons • Not quite as affordable as one might hope.

summary A unique and very practical self-contained diffuse-field omni mic with integral APE sphere, as popularised in the Decca Tree mic technique.

As I mentioned earlier, the Ocula microphone is sold in stereo‑ or triple‑matched pairs, with the matching said to be within ±0.5dB across the whole usable bandwidth. I was supplied with a stereo matched pair for this review, which shipped in a cardboardbox,although each microphone is protected in its own velvet bag, with the cable coil secured by a Velcro strip. Also included with the set are a pair of high-quality and remarkably compact orbital (ball) mounts, made by a company called Artcise. When attached to these ball mounts each Ocula mic can be angled exactly as desired very quickly

“The Braingasm Ocula is a well-designed and well-engineered product filling a particular niche in the microphone market very nicely indeed...” build-up. So, I estimate the Ocula’s actual frequency response rises to a peak of around +9dB at 10kHz, before falling back to about +3dB at 20kHz. Hopefully, Braingasm will publish a genuine measured frequency response chart of the complete Ocula mic before too long.

The APE Concept The Acoustic Pressure Equaliser (APE) is a concept developed in the very late 1940s by the NWDR (Nordwestdeutscher Rundfunk) laboratory — the R&D department of the Northwest German broadcast organisation. The NWDR team was led by Herbert Großkopf, carrying out research into how differently shaped microphone capsule housings affected the overall frequency response and directivity. Großkopf, by the way, is probably better known for patenting the continuously variable polar-pattern control, the rights to which Neumann acquired and launched on the studio world in 1949 as the M49 multi‑pattern studio microphone. The APE concept was first introduced in a product just two years later, in 1951, when Neumann brought out the M50 omnidirectional microphone. The M50 actually looks very similar to the M49, as it shares the same outer casing and has similar electronics. Inside, though, the capsule diaphragm is just 12mm in diameter and is embedded flush at the surface of a 40mm Perspex ball. In terms of the sphere’s acoustic physics, as on-axis soundwaves reach the diaphragm, the presence of the sphere’s surface leads to an increase in air pressure for wavelengths comparable to, or smaller than, the diameter of the sphere. In the case of a 40mm sphere, that translates into a graceful HF boost above 2.5kHz, rising to about +5dB, and this is very handy as it almost

perfectly counteracts the natural loss of HF energy as sound travels through the air into the diffuse field. And that was the aim, of course — to provide a flat overall response for a microphone placed in the diffuse sound field, as it would be for orchestral recording, for example. Additionally, though, the sphere also causes a high-frequency shadowing effect for sounds arriving off axis, and this correspondingly reduces the microphone’s sensitivity to off-axis high‑frequency signals. In other words, it makes the omnidirectional microphone behave more like a directional microphone at high frequencies — not quite as much as a true cardioid, but more than a hypocardioid. So, for orchestral recording, where the microphone will be working at a distance in the diffuse field, an omnidirectional capsule embedded within a sphere combines the low-frequency extension of a standard omni with a natural high-frequency boost compensating for inherent air losses to maintain detail and brilliance. And, when making stereo recordings, the HF directivity allows the microphones to be pointed outwards to increase the inter-channel level differences between channels, enhancing the stereo imaging in a similar way to traditional pressure-gradient mics — and all without sacrificing the low-end performance of pressure-operated mics. Hurrah! Jägermeisters all round!

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Lynx Mesa Thunderbolt Audio Interface Lynx have drawn on their existing Hilo2 and Aurora(n) designs to create a powerful desktop audio interface. SAM INGLIS

t’s one of the more niche league tables in the world, but Hugh Robjohns’ chart of measured dynamic range performance on the SOS Forum tells you a lot about the current state of audio interface design. In general, newer interfaces score more highly than older ones, though a few classics such as the UA 2192 and Lavry AD11 still compete. And, as you’d expect, the upper reaches are dominated by relatively costly products. The Lynx Hilo2, released late last year, is the only current interface that features in the top five for both A-D and D-A measurements. Such performance naturally comes at a price, and depending on which variant you opt for, you can expect to pay upwards of £4k for a two-input device that operates only at line level. Mastering engineers and serious classical music purists will love it, but many small studio owners will find it unaffordable and its I/O too limited.

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The idea that Lynx could address both concerns in a single product sounds counterintuitive. Surely there would be no way to simultaneously increase the Hilo2’s input count, add mic preamps and monitor control, and reduce cost, without resorting to drastic measures like offshoring their manufacturing, or drastically reducing sound quality? Well, apparently there was a way — and the result is the Lynx Mesa.

Going Hi How have they achieved this? For one thing, whereas the Hilo2 can be configured as a Thunderbolt, USB or Dante device using optional cards, those options have been closed off on the Mesa, which is available only in a fixed Thunderbolt variant. Lynx have also saved a few pennies by not including any way to actually connect it to your computer, so you’ll need to add a certified Thunderbolt 3, 4 or 5 cable with Type‑C connectors to your shopping basket. Additionally, the Mesa lacks the Hilo2’s internal PSU and battery

power option, drawing its current instead from an external ‘line‑lump’ PSU. On the analogue side, the Hilo2 is a two-in, four-out device, whereas the Mesa is the other way around, losing the XLR line outputs but gaining a second pair of inputs. What’s more, all four of its analogue inputs are on combi XLR/jack sockets and feature the same comprehensive digitally controlled mic preamps that are available for the same company’s Aurora(n). The Mesa has two built-in headphone amps to the Hilo2’s one; these have independent volume controls but always present the same signal. Another Mesa feature derived from the Aurora(n) and not found on the Hilo2 is built-in multitrack recording to an SD card. In terms of digital I/O, the Mesa retains the Hilo2’s single optical input and output, which can operate either as ADAT Lightpipe or stereo S/PDIF, as well as the word‑clock I/O on BNC connectors and the dedicated coaxial S/PDIF I/O. However, it lacks the Hilo2’s dedicated AES3 in and out on XLRs. A single Type‑C socket takes care of connection to the host computer.

Touch Typing Although there’s a lot in common under the hood with other Lynx products,

Lynx Mesa $2499 pros • Slick design with superb touchscreen implementation. • Good audio specs and subjective sound quality. • Easy to set up and use. • Built-in SD card recording. • Highly flexible internal routing and mixing.

cons • Limited monitor control. • Internal mixer has no EQ, dynamics or solo/PFL bus. • Both headphone sockets carry the same signal. • No Thunderbolt cable included.

summary The Mesa is a stylish, powerful and good-sounding desktop Thunderbolt interface, with an excellent touchscreen implementation and the useful ability to record directly to SD cards.

the Mesa represents a clean break in terms of industrial design. Its stylish, asymmetrical case is made from chunky metal, and seems designed to act as a heat sink; at any rate, it gets pretty warm in use. The gently sloping top panel makes the ideal home for the full colour touchscreen, which is impressively clear from all viewing angles, bright enough to use in direct sunlight, and precise and responsive to touch. It’s joined there by four buttons, a large rotary encoder and potentiometer volume controls for the two headphone sockets. These are found to the right of the curved front panel, with a slot for an SD card to the left. All other I/O is tucked away on the rear panel.

The obvious comparisons are with the Neumann MT48 and its close sibling, the Merging Anubis. The Mesa does indeed bear some similarities to these devices, both in terms of physical design and functionality, and perhaps the most striking of them relates to the use of the touchscreen. I’ve reviewed other touchscreen-equipped interfaces where the screen felt at best a luxury and at worst a form of window dressing. That’s not the case with either the Mesa or the MT48/ Anubis, which exploit their screens to the fullest. In both cases, in fact, the screen offers pretty much all the functionality that’s available from the control panel application running on the attached computer. So, not only can you configure the Mesa and perform basic housekeeping, but you also get full control over its internal mixer. There’s even a built-in Help system, which is a nice touch. There are also some important contrasts between the Mesa and the Neumann/ Merging products. I’ll get to the functional differences later, but from a control point of view, the main one is that the MT48 and Anubis serve the control panel and mixer as web pages. This means they can be accessed from any device that can run a browser and connect to the same Wi-Fi network as the host computer, so you can for example make adjustments from the live room, or allow musicians to set up their own cue mixes. By contrast, the Mesa currently offers no remote control options, but on the plus side, its graphical interface is rather more attractive to look at than those of the MT48 and Anubis.

Megamix At base sample rates, the Lynx Mesa presents a total of 16 inputs and outputs to your DAW. On the input side, these comprise four mic/line/instrument inputs, coaxial S/PDIF, a stereo loopback input, and up to eight optical digital inputs. On the output side, your DAW simply sees

eight pairs of outputs labelled Play 1-2, Play 3-4 and so on. Each physical output pair in effect has its own mixer, with its own independent balance drawn from both the inputs and the DAW playback channels. Playback from any of the 16 SD card channels can also be freely mixed in on a per-output basis. It’s an extremely flexible mixing and routing system, with virtually no limitations as to what can be sent where. Even the outputs have their own dedicated faders. Having said that, there are no signal-conditioning facilities at all apart from polarity switches and high-pass filters on the four analogue inputs. If you like your low-latency cue mixes to feature EQ and dynamics, or effects such as reverb and delay, that’s not possible here. Curiously, there’s also no solo or PFL bus, which seems a strange omission in an otherwise comprehensive mixing arrangement. On a Mac or PC, control over the mixer is handled using a utility called Mesa-Remote. As previously mentioned, this pretty much duplicates the functionality that’s available from the touchscreen. It also uses similar graphics and colour schemes, though Lynx have chosen to lay things out differently to take advantage of the larger display area available on a typical computer. One of the few functions exclusive to Mesa-Remote is the ability to save and load mixer Scenes. These can be assigned to six on-screen buttons, for instant recall. Inputs, playback channels, SD playback channels and outputs each have their own bank of faders, which appear in four rows. The output faders are always visible, and indeed it’s necessary to select an output pair in order to see what mix is being adjusted. The other three rows can be collapsed, but I do feel the Mesa-Remote user interface could use a little refinement. With more than two fader banks expanded, the entire interface won’t fit on a laptop screen, and if the output channels are out of view, there’s no colour coding or

The Mesa features four analogue inputs, all with preamps and all capable of accepting DI guitar signals.

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Mesa-Remote runs on your Mac or PC as an alternative to the touchscreen. In this screenshot, the fader banks for the DAW and SD card playback channels are collapsed.

other information to tell you which one is currently selected. And if you want to scroll with a trackpad or wheel, you’ll need to be careful where the mouse pointer is positioned; if you leave it over a fader or pan control, your gesture will move the fader rather than scrolling the window.

Talk & Listen Although there’s no ‘master channel’ as such in the Mesa’s mixer, it does have some monitor control and master section features. The main rotary encoder can govern the output fader level either for the main monitors or the headphones, and it has a secondary press action that can be dim or mute. There’s also a dedicated mute button, which can be assigned either to Current Output or to any combination of physical outputs. And although there’s no built-in talkback mic, it’s possible to repurpose one of the Mesa’s analogue inputs for this function. Speaker switching is obviously not an option, since there is only one pair of line-level outputs. Sadly, there’s no way to generate a mono fold-down of the monitor signal within the Mesa mixer. The two top-panel buttons to the left of the group of four are assignable. A nice touch is that they can have different functions depending on whether an SD card is inserted or not. The physical buttons display the universal circular record and

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triangular play icons, reflecting their default functions when a card is inserted, but they can also be used as shortcuts to reach different mixer pages, or as either latching or momentary talkback buttons. The third button toggles between displaying the Mesa’s metering view, which can represent signal levels at all of its inputs and outputs, and the main menu.

Hot Mesa As I mentioned at the start of this article, the Lynx Hilo2 is one of the best all-round

performers on the market in terms of A-D and D-A quality. Some of the Hilo2’s more esoteric or audiophile options are omitted on the Mesa, notably the choice of converter filtering modes, and it doesn’t quite achieve the same numbers, but it’s very impressive nonetheless. Output dynamic range is 120dBA both on the monitor outputs and the headphone outputs. The latter are punchier than those found on most audio interfaces, delivering a maximum output level of +18.4dBu and up to 383mW into a 60Ω load. In line input mode, its analogue ins achieve an A-weighted dynamic range figure of 119dB, and are flat to within ±0.01dB from 20Hz to 20kHz. Switch the inputs to mic mode, and the dynamic range falls a little, but only to 116dB. More importantly, Equivalent Input Noise is an impressively low -129dBu (A-weighted). Gain is variable in 1dB steps over a 63dB range, with a switchable pad extending this at the lower end by 13.6dB. The hottest level that can be accommodated is a beefy +28.7dBu, whereas in line mode, the Mesa is aligned such that 0dBFS corresponds to +20dBu. The jack part of the combi input is used only for high-impedance sources, and presents a 1MΩ impedance when used unbalanced or 2MΩ balanced. In terms of its technical performance, in fact, the Mesa mirrors Lynx’s Aurora(n) rather than the Hilo2. You could think of it as a product that combines that interface’s audio circuitry with the Hilo2’s touchscreen-based control paradigm. And subjectively, it sounds very good indeed. I would have no qualms whatsoever about using it on any recording, and I’d

SD Card Recording Like Lynx’s modular Aurora(n) system, the Mesa has the ability to use a Micro SD card for 16-track recording and playback. This process is not covered in depth in the PDF manual, but it’s clear that a lot of thought has gone into making it as intuitive as possible. Insert a card into the front-panel slot, and (unless you’ve modified the settings) the F1 and F2 buttons light up red and green respectively, indicating that they have now become Record and Play controls. A fuller selection of transport controls appears on the touchscreen and in the Mesa-Remote software. What’s more, the Mesa automatically senses which inputs have signal coming in, and record-arms only those tracks — though, again, this can be overridden if you need to. Recordings are organised into Sessions, each of which can comprise multiple Takes, which are recorded as single poly-Wave files. There is no way to drop into or overdub onto an existing take, or reorder the channels, so a guitar that was recorded through input 4 will always be played back on SD Play channel 4, regardless of whether tracks 1-3 are used. As far as I can tell, there’s also no way to manually locate the playback head or cursor within a take, so playback always starts from the beginning. Takes can also be organised into Playlists in the standard XSPF format. Takes within a Playlist are automatically played back one after the other. Oddly, takes and sessions can be managed, selected and named from the touchscreen, but not from Mesa-Remote. Nor is there a straightforward way to transfer audio files from the SD card to the computer; you’ll need to remove the card and insert it into your Mac or PC’s card reader.

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be perfectly happy to employ its built-in preamps on absolutely anything. What’s more, although it’s arguably better featured than the Hilo2, the Mesa is little more than half the price of that flagship converter. However, half of the price of a Hilo2 is still a lot of money. For the price of the Mesa, you could have your pick of 1U rackmounting interfaces from the likes of RME, Universal Audio, MOTU or Metric Halo. Most of these would offer broadly comparable tech specs and more comprehensive I/O, albeit without the Mesa’s rather lovely touchscreen. In the desktop arena, the Mesa faces more affordable competition from the Antelope Audio Zen Tour Synergy Core, which has extra I/O and features an onboard plug-in-style effects platform; it even has a touchscreen, though its potential is not fully exploited. But, as I’ve already mentioned, the most obvious rivals for the Mesa’s crown are the Merging Anubis and the closely related Neumann MT48.

No Wrong Answers You’d have to be an exceptionally picky engineer with golden ears to base your choice on perceived sound quality. The Mesa is faultless in this regard, but the Anubis platform actually betters it on paper, with exceptional specs and subjective performance. The touchscreen implementations are about equally excellent, and leave products such as the aforementioned Zen Tour and the Apogee Symphony well behind. In terms of built-in local I/O, the Anubis and MT48 are perhaps more like the Hilo2 than the Mesa, with multiple outputs but only a single pair of inputs, though these do have preamps. They also have two fully independent headphone outs, but the Anubis lacks the optical I/O found on the MT48 and Mesa. In short, it’s a tough choice, and to my mind it comes down to three factors: expandability, ease of use, and connection protocol. To take the last of these first, the Anubis connects to a computer over Ethernet, originally using the RAVENNA protocol but now with the option of Dante as an alternative, whilst the MT48 offers the choice of Ethernet or USB.

The Mesa’s mic preamps are the same design used in the LM-PRE4 module for Lynx’s Aurora(n).

On my Mac, Ethernet operation with the RAVENNA firmware delivered a round-trip latency of around 3.6ms at 44.1kHz with a 16-sample buffer size. With the MT48 on USB, the smallest buffer size on offer is 32 samples, which results in around 5ms total latency. Being a Thunderbolt interface, I expected the Mesa to beat this, but in fact it’s marginally less good, with the smallest 32-sample buffer size producing a round-trip latency of 6ms at 44.1kHz. All of these are very respectable figures, but they don’t give the Mesa a point of difference.

a RAVENNA network isn’t too bad as IT admin goes, and Merging’s UNITE setup utility takes quite a lot of the pain away, but I don’t think the Anubis will ever be ‘plug and play’ in the way that the Lynx unit is. And although Merging’s touchscreen implementation is equally powerful and permits remote control, Lynx’s is more polished and graphically sophisticated. The Mesa also abounds with thoughtful design touches, such as the ability of the inputs to auto-sense what sort of source is connected and adopt the correct mode without user intervention. Finally, the value of the Mesa’s built-in audio recording shouldn’t be underestimated. I can only think of a handful of other interfaces that support direct recording either to SD cards or USB devices. You wouldn’t choose to leave the laptop at home and track a crucial session this way, but the benefits of having a (somewhat) redundant recording platform churning away in the background are obvious. Whether it’s acting as a backup in case of computer failure, or simply documenting the session so that you can recall ideas and jams that didn’t get tracked to the primary system, it offers peace of mind that isn’t easily obtained in other ways.

“The Mesa abounds with thoughtful design touches, such as the ability of the inputs to auto-sense what sort of source is connected and adopt the correct mode without user intervention.”

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Meanwhile, the Anubis and MT48 score heavily on expandability. Yes, they lack the Mesa’s dedicated S/PDIF I/O and its word‑clock sockets, but both can operate as part of an audio-over-IP network, opening up more or less endless possibilities. Better still, Merging’s ‘peering’ technology allows the Anubis and MT48 to control other Merging interfaces in a seamless fashion, whereby the additional I/O is simply incorporated into the Anubis mixer. Said mixer is also even more powerful than that of the Mesa, with built-in EQ, dynamics and reverb available on top of an equally flexible routing and mixing architecture. In the case of the Anubis, it’s even possible to carry out monitor calibration internally. The flip side of this is that ease of use certainly favours the Mesa. Configuring

$ $2499 T Lynx Studio +1 714 545 4700 E sales@lynxstudio.com W www.lynxstudio.com

Radial Engineering creates reliable, high-fidelity audio gear to ensure you achieve the best sound and create music and mixes that are truly inspiring. Our goal is to help you live up to the highest professional standards, improve workflow, and capture the subtleties of every take whenever and wherever you hit record. Our mission is to provide world-class audio equipment to support your success as recording technology continues to evolve while listening to our customers as closely as we listen to the music.

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Buchla Music Easel Analogue Synthesizer

Fifty years after Don Buchla first released his portable electronic music studio, is the world finally ready for the Music Easel?

PAUL NAGLE

he development of synthesizers into the myriad forms we know today can be traced back to the 1960s and two American designers, one whose name became synonymous with ’synthesizer’, the other who thought the word was too limiting for his instruments. The two men are, of course, Bob Moog and Don Buchla and both leave legacies far greater than implied by terms such as East Coast or West Coast Synthesis. Although it’s true they were geographically separated, Moog and Buchla invite as many comparisons as contrasts, but there’s no denying that Moog’s impact has been the most keenly felt across the industry. Even to this day, Don Buchla remains a much more elusive character, his contributions less widely understood and appreciated.

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While I can’t hope to redress the balance in just a few thousand words here, I would like the opportunity to dally in the Buchla universe and check out a recently reissued classic, the Buchla Music Easel. Born out of the artistic melting pot that was the San Fransisco Tape Music Center, this was an electronic instrument designed for portability, patchability and live performance. It seems criminal that it’s not as well known and loved as the Minimoog, ARP 2600 or Synthi AKS. Around eight years ago, having long been curious about Buchla and his instruments, I treated myself to an Easel — as a kind of retirement present. For reasons I’ll come to later, it was eventually returned, but I’ve always regarded it as the missing link in my personal quest to learn about every important synth. Without giving away too many spoilers, much has

changed since then, not least that the Easel is now available in two updated versions: Retro and Modern.

First Steps Drawing from modules Buchla had already designed, the Music Easel first appeared in 1972. While the bulk of this review will concentrate on the more feature-rich Modern version, both it and the Retro have the same long pedigree — and have been given enhancements to extend their life far into the future. Both models are supplied in robust cases, ideal for the performing musician. Upon connecting the external laptop-style power supply, you’ll probably need to reach for the Quick Start patch sheet (unless you’re already up to speed on the Buchla way). Disappointingly, a full printed manual is not supplied. Indeed, you

have to assemble its components yourself from resources on the Buchla website. Accordingly, to supplement the original 1974 ‘Operating Directive’, I added the 208c addendum, the 218e V3 user guide and the EMBIO manual. Incidentally, all the components of the Modern Easel (which I’ll refer to as simply Easel from now on) are available individually. The 208c or Easel Command is the main instrument, the 218e the updated keyboard and the EMBIO is the ‘Electric Music Box I/O’. Ten banana cables are supplied, along with four shorting bars, a MIDI adaptor (Type‑A) and a TiniJax lead. This lead closely resembles a 3.5mm mini-jack lead, but it’s not; it’s a fraction wider at 3.58mm. Before making any assumptions, it’s therefore worth taking stock of the sockets strewn about the panel, ready to baffle then hopefully delight. Starting with the EMBIO, its CV and gate connections are 3.5mm, ready to talk to Eurorack. MIDI is present in TRS mini-jack form and there’s another mini-jack located on the 208c, the headphone output. Every other small jack, including the mixer on the EMBIO and the audio outputs on the 208c, are in the archaic TiniJax format — because that’s how Buchla have always done it. It would have been nice if they were more obviously different because, as you can imagine, it makes things slightly awkward when interfacing with non-Buchla gear or when using the external audio input. I’m told

TipTop’s stackable Eurorack cables work reliably in both types but haven’t tried them personally. There are a couple of USB-C connections (also for MIDI), and the audio outputs and sustain input are reassuringly normal quarter-inch. Everything else is, well... bananas. The Easel is mostly modular in concept; there are some hard-wired connections and a multitude of switches, but you won’t get far without patching. In a Buchla system, this means banana cables, which have one advantage over Eurorack (and 5U format) in

and a spring reverb. Additionally, there’s a voltage inverter and an audio input (with envelope follower), plus a Program Interface slot. Volume controls for the two channels and a headphone output round things off. Courtesy of the 218e keyboard, the Easel gains portamento, an arpeggiator, four assignable voltage sources and several control outputs. While both Retro and Modern Easels include MIDI, only the Modern’s EMBIO provides those handy CV and gate inputs, along with a set of extra utilities. The Easel is one of those rare vintage synths capable of patch storage and recall. Thanks to voltage control and the Program Interface slot, prepared cards could be inserted and even hot-swapped during performance, invoking radical shifts at the flick of a switch! That option still remains, but if you purchase the Program Manager, it will provide a solder-free alternative, one which is supported by an app for Mac and Windows.

“Buchla have succeeded admirably, refining features and boosting flexibility without compromising the essential Easel nature.”

Photo: Tim Rafferty

Buchla Music Easel $4999 pros • A self-contained performance synth with a long pedigree. • Distinctive wavefolding, audio-rate modulation and Lo Pass Gates offer worthy alternatives to more familiar subtractive synthesis. • It’s an instrument to fire the imagination. • Features MIDI and, in the case of the Modern version, CV/gate inputs.

cons • Not cheap — but what classic is? • The mix of 3.5mm and 3.58mm jacks is initially hard to fathom (and identify). • Its MIDI is not for the faint-hearted.

summary A beautiful-sounding electronic music box that deserves to be better known. The updates in these models are significant, particularly if you ever struggled to make the earlier keyboard work consistently. Created for the electronic composers of the ’70s, the Easel is still a unique voice today.

being naturally stackable. Multiples are not required. The cables are colour-coded for length and the sockets coloured by function, with black denoting a CV input and various hues representing the available modulation sources. Furthermore, pulse outputs, which are used for timing information, envelope triggering and so on are red, making it easy to see your options at a glance. Unlike Moog, Buchla preferred to keep audio and voltage connections separate, considering the differences as ‘sound versus structure’. Thus, bananas carry control signals (0-10 V) with shielded cables reserved for audio. One small exception is found in the Quick Start patch but we’ll leave that as a treat for later. I should mention that the banana sockets are all pretty tight at first and tough to disconnect one-handed, at least for this puny, partially retired knob‑twiddler. Fortunately, they do loosen up after about a week of regular use and, long-term, I suspect they’ll prove more trustworthy than, say, Eurorack connections.

The Eclectic Electric Music Box The Easel is a collection of tools assembled to serve electronic musicians from the tape music era — and beyond. Central to operations are two oscillators, the Complex Oscillator and Modulation Oscillator, the former notable for its wavefolding. The latter is a modulator in either low or high frequency ranges as well as being an audio source in its own right. In supporting roles are a voltage-controlled envelope, a ‘Pulser’, a five-step sequencer, a source of random voltages, two low-pass gates, white noise

What, No Cutoff Knob? The central playground of the 208c is a feast of helpfully coloured and responsive sliders. Admittedly, an over-stacked banana can sometimes get in the way, but that’s where those shorting bars come in handy — for connecting common source and destination pairs without impeding access. Typically, the sliders are divided into two types, one providing direct access, the other for voltage control via an adjacent CV input. In the case of the Dual Lo Pass gate (LPG), raising the Level 1 or 2 sliders will open the gate and produce a drone. If you’ve played an ARP Odyssey or other synth with a VCA gain control, this will be familiar. Before plunging in, it’s worth adjusting to the clean, raw tonality on offer. For those of us who grew up with synths from Roland, Korg and Moog, the Easel hints at an alternate reality, one in which filter cutoff and resonance don’t take centre stage. Actually, the Easel does have filters — two of them; they’re components of the Dual Lo Pass Gate — but they behave rather differently to the low-pass filters we’re used to. And, actually, the Lo Pass Gate module included isn’t even the one Buchla want us to have... The tale goes like this: Easels sold today via the UK distributors are shipped with a Lo Pass Gate module that is fully RoHS compliant, meaning all its components can be legally tossed into landfill(!). However, www.soundonsound.com / October 2025

ON TE ST

USB Audio Interface

Bitwig Connect 4/12 NICK ROTHWELL

t seems to be something of a rule that every music software company must, at some stage, get into the hardware business. Ableton played the control surface game with the Push and Push 2, and added audio with the Push 3. Now Bitwig have entered the ring with a desktop audio interface, although its controller features are also deceptively powerful, and it has connectivity clearly aimed at the modular synth market.

First Impressions The Connect 4/12 is visually striking. It sports a colour scheme incorporating Bitwig’s signature bright orange, and prominently features a large dial on its top surface, much in the style of Universal Audio’s Apollo series or the Mackie Big

October 2025 / www.soundonsound.com

Bitwig’s first hardware product takes a characteristically individual approach to audio interfacing. Knob. The Connect’s case is all metal, and the unit is reassuringly heavy. The top surface presents the interface’s most unusual feature: a row of 3.5mm sockets, intended primarily for modular synth control but usable for (unbalanced) audio at a pinch. To the left of the big dial are stereo meters calibrated in 4dB increments from -40 to 0 dBFS, and status LEDs for audio input type, phantom power and MIDI I/O. There are transport controls bottom left, and mode switches below and to the right of the dial, which itself is set in a 360-degree ring of LEDs (32 of them). Mini-jacks aside, all other connections are on the back panel. Sadly, there’s no silkscreened guide on the top, so you do have to peer over the back and squint

a little to see the labels. There are two balanced quarter-inch audio inputs (one a combi jack to support a microphone), six balanced line outputs, headphone out (annoyingly this isn’t on the front of the device), and two mini-jacks for MIDI in and out via the bundled adaptors. Power and connectivity (USB 2) are via a USB-C socket, and there’s no facility to use an external power supply (though at no point in my testing did I feel like I needed one). There’s not even a power switch. Buttons on the back select the input type for inputs 1 and 2 (instrument/line). If there’s an XLR plugged into input 1 then the device assumes a microphone is attached and goes into microphone mode, otherwise not; there’s no manual microphone selection. A dedicated

Bitwig Connect 4/12 $599 pros • Well designed ‘hands on’ functions on the front panel. • High-resolution dial works well for editing, scrolling and scrubbing. • Neatly integrated into Bitwig Studio as control surface and modular interface. • Good build quality.

cons • Limited number of audio inputs. • Some functions disabled at high sample rates. • No scroll or zoom functionality for Launcher clips.

summary Bitwig have scored a triple whammy with the Connect 4/12: it’s a solid hands-on audio interface, a wellintegrated control surface and a powerful link between the Bitwig DAW and a modular synth environment. It pushes all the right buttons.

button toggles phantom power. Inputs 1 and 2 can be ganged into a stereo pair, in which case they both operate at line level. So far so good — and the Connect is class-compliant so you can just plug it in and you’re away. (Some brief testing confirmed that it also works on a USB-C‑enabled iPad.) The dial, which is aluminium and feels smooth and solid, lets you set input and output levels: click the relevant button marked In 1, In 2, Main or Phones, then turn the dial, and the ring of LEDs unfurls in orange for visual feedback. The maximum gain for the main two inputs is 30dB for instrument and line input and 62dB for microphone. The minimal setting (no LEDs lit) is unity gain — there’s no way to attenuate. Output is attenuated only, there’s no signal boost. The Connect claims to be compatible with the Mackie Control protocol. Sure

The Connect’s control panel application.

enough, when I fired up Ableton Live and configured the Connect as a Mackie control surface, the transport buttons worked, as did the dial for scrolling. The integration is, shall we say, minimal; the behaviour with Bitwig Studio is much more immersive, as we’ll see.

Audio This is probably about as far as you’ll get without reading a manual, and since my unit arrived without one, I had to do a bit of hunting around on the Bitwig site to find one, at which point I discovered that there’s also a dedicated control panel application for setup and mixing on-screen. If you’ve already gone ahead and connected the device to a computer and fired up your DAW of choice, you’ll notice that it actually claims to have six inputs, not four. Inputs 5 and 6 are a dedicated loopback connection from the (pre-fader) main outputs, useful for capturing live mixes or non-DAW computer audio. It’s a credit to the device that pretty much all of the functions of the control panel app

can be accessed pretty immediately from the front panel. There’s a hardware monitor from input 1 and/or 2 direct to main outs or headphones — the dial’s LEDs turn red to indicate monitor level. The audio features are rounded out by three dedicated buttons to the right of the dial. Mono switches the output to mono, useful for checking a mix. Alt toggles the main mix to an alternative pair of outputs, such as a second pair of speakers. Dim is a toggle which attenuates the main mix by a configurable amount. I recently played a couple of small live gigs with the Connect, and Dim is exactly what you need to talk to your live engineer during a soundcheck. The audio interface will run at high sample rates (176.4 or 192 kHz) but doing so disables inputs 3 and 4, outputs 9 to 12, and direct monitoring.

Bitwig Control The Connect 4/12’s control functions fall into two groups: parameter control and transport. As a parameter controller, the device operates rather like any other MIDI

Round the back we find a USB‑C port, 3.5mm MIDI I/O sockets, and the analogue audio I/O.

www.soundonsound.com / October 2025

ON TE ST

TiNRS Bopp & Steve EURORACK MODULE

his is Not Rocket Science (TiNRS) thought it might be nice to create a space, within space, where cosmic entities exist as the elements of a deep galactic reverb. The reverb is based upon a conversation between entity Bopp and entity Steve. Bopp is the originator of sound, whereas Steve is the receiver. Filling the void are elements such as Delay, Tail, Diffuse, Pitch and Envelope; all parts of a reverb that we usually experience together, but here, Bopp and Steve let us approach each one separately as the reverb is deconstructed before us in a signal flow tempered by individual wet/dry mix knobs. The results are epic, exciting and more than a little peculiar. So, to get a handle on it in simpler reverb terms, we have a Delay section which includes pre‑delay and early reflections spilling into diffusion and defining the flavour of space; and we have the Tail section where echoes and reflections are devolved into a backwash of noise. You can

then define how much of all this your input signal encounters along its journey through the space. Along the way, you can push the pitch of various elements around and pull an envelope off your signal to control other things. It can function as a captivating and sometimes badly behaved delay effect with a choice of Ping‑pong, Waveguide (your sound trapped in an angry pipe) or Multi‑tap modes. It has CV controllable Length (delay time) and Feedback, and a quirky Pattern control that shapes the character of each mode. At the Tail end, you can explore three modes of space reverb, from the ridiculously intense Dense mode, the pared-back Float mode with dampening or shimmer, and the granular Sparse mode that is like throwing together slices of audio in any order or direction the mode desires. A Temperature knob can act as a tilt EQ, throwing your reflections quite dramatically into high or low registers, or, in Float mode, as a shimmer generator. Most knobs seem to do more than one thing depending on the mode. There are other elements too, like the Diffuser, a flavour-enhanced all-pass filter that can place your sound in space before it enters the rollercoaster ride of Delay and Tail. The Chorality knob more often than not introduces multiple voices in a similar way to chorus, except when it’s guiding the probability of the playback of granular slices. With every element alive and working on your audio you can find yourself falling into a blissful state as you are smeared across the universe. But it’s in the pulling apart of the reverb that we find unexpected sound‑design entities that can take us in completely different directions. On its own,

the Diffuse introduces strange widenings, the Delay scatters and manipulates, and the Tail has an unusual foggy focus, much of which you’d miss in a regular reverb. It’s one of those rare cases where the sum of the parts is perhaps less interesting than the parts themselves. And then there’s the Pitch section. Bopp & Steve has three mind-bending pitch‑shifters that interrupt the deconstructed reverb conversation and impose wild shifts that may or may not be related to what you’re feeding it. There’s a Regular mode where all three shifters play loosely around the same note, a Chord mode where they all shift to a different note, and then there’s Free mode, which is unquantised and follows the Transpose and Control knobs. You can use those same knobs to mess with the pitch changes in either mode but you can also plug in a MIDI keyboard and turn your reverberations into an otherworldy playable instrument. Overall, in pulling apart the often taken-for-granted reverb, TiNRS have unearthed some versatile sound‑design elements that can be completely at odds with the usual generation of huge spaces. While the epic sound of cosmic overload that pours out when everything is pushed to the max is completely awe-inspiring, I’m finding the individual elements more fascinating because there’s so much to discover here. If TiNRS could grant one wish it would be some kind of preset system so that I could revisit some of the destinations I find on my travels. But then Bopp & Steve is not a set-and-forget reverb, it’s a playground of spatial effects that will leave you spinning, excited and at times happily baffled. Robin Vincent $

$550

www.thisisnotrocketscience.nl

“Bopp & Steve is not a set-and-forget reverb, it’s a playground of spatial effects that will leave you spinning, excited and at times happily baffled.” 90

October 2025 / www.soundonsound.com

Omnitone Rhythmi & Beatsi EURORACK MODULE

rom Omnitone of Canada, we have a pair of colourful rhythm‑makers that can operate separately, or as a percussive pair. Rhythmi is a pattern and trigger generator with an interesting angle on groove generation. Beatsi is a digital drum synthesizer rolled into a compact grid of easy access sound-shaping. Both employ a slightly cryptic use of colour to show activity and value, giving the modules a unique style and workflow. Rhythmi is not a sequencer as such; rather, it is an explorer of patterns that’s reminiscent of Mutable Instruments’ Grids. It has five channels: kick, snare, hi-hat, tom and crash, and it comes up with an open hi-hat by lengthening the hi-hat gate output. A pattern can go from two to 32 steps and from stripped-back laziness to full-on mania. You have a choice of five unique base patterns. These are influenced in the quite crowded Customise section by the Density knobs, which increase the number of hits, and Synco knobs, which define how often they are on or off the beat. The crash is different; it just triggers when the pattern energy is high enough. Once set you can then turn the Energy knob to make everything faster and harder, and spin the Evolve encoder to branch off into variations. It’s busy, exciting and full of surprises, which is not always what you want, but once you’ve accepted that you are not entirely in control, it’s a fun ride. Meanwhile, Beatsi is dutifully pumping out a corresponding collection of sounds. It has

three kits: orange is acoustic, blue is lo-fi, and green is an alien kit, although those labels shouldn’t be taken too seriously. There are individual inputs for each of the five channels and a pitch input for the toms. The toms are a bit of a special feature in that Rhythmi has a Tom 1V/oct CV output that can play a tune on the tom sound in the Beatsi, which is very cool. Otherwise, there’s a single mixed output and everything else is done in the grid. You use the Parameter encoder to zip through the 5x4 grid one node at a time. The colour band at the bottom gives you a rough idea of the value and changes colour to show which kit you’re in. The grid gives you control over pitch, level and decay, and you can also assign parameters to the two CV inputs. Once you’ve figured it out, it’s really quite powerful, if a bit laborious, but so much more preferable to fiddling around with a menu and values on a little display. You don’t have to use these two modules together, although they do make a great combo. Beatsi would make for an interesting and compact drum module for anyone into synthetic percussion. Rhythmi offers programming-free drum patterns with lots of variation for small adjustments and is so

Modular Profile: SchneidersKeller

dd Butterworth and Jean-Marcel Fricke are the joint helmsmen of the recently opened SchneidersKeller, an offshoot of Berlin’s venerable SchneidersLaden situated on Denmark Street in the heart of central London. Butterworth has long been a pillar of the London scene — not least as part of the much-missed boutique London Modular. Fricke, meanwhile, cut his teeth working with Andreas

Schneider at SchneidersLaden; though he also has control voltage in his blood, being the son of the late, legendary drum machine and synth designer Manfred Fricke. On their entry into modular: Edd Butterworth: I was working in a studio that had a bunch of classic analogue synths and outboard, mostly, Moogs, Korgs, Rolands... Everything was patched into the patchbays,

engrossing that you can forget to do anything else with your patch. The coloured interface is delightful but can be a bit mysterious, while Rhythmi in particular feels busy in form and function. However, there’s a little universe of rhythms and weird sounds in here that offer lots of fun, unexpected avenues and an alternative to programming and probability. Robin Vincent

including external inputs into the filters and CV/gate, et cetera. I was fascinated with routing things in and out of the synths and outboard. I got really into setting up these complicated routing scenarios. So when I wanted to buy some of my own gear I wanted that flexibility and decided on Eurorack as the best solution. I spent all the money I had on a Tiptop Happy Ending kit, the Expert Sleepers ES-5, Intellijel Dixie and Quadra, Doepfer VCA/Mixer and standard VCO and a Malekko Wiard Borg 2 filter. I still have

Rhythmi $299, Beatsi $280.

www.omnitone.ca

the Borg, the Quadra and the Doepfer VCAs! Jean-Marcel Fricke: I was familiar with modular synthesizers through my dad’s work, but I never took a serious interest in them. To me, they mostly sounded like quirky bleeps and bloops without much musical context. That changed when I did an internship at SchneidersLaden in Berlin when I was 18. During that time, I met Mark Verbos of Verbos Electronics. He took me to a gig and played a fully improvised live set using his system and a TR-909.

www.soundonsound.com / October 2025

ON TE ST MODULAR

I thought, ‘I could do that too!’. I convinced my dad to give me a Tanzbär [drum machine], along with an MFB OSC-03, MFB Dual LFO, MFB VCF, and ADSR — basically everything I needed to build a monophonic synth voice. When he saw how much I enjoyed it, he let me start doing demos at conventions like Superbooth, and I began performing live gigs. After my father passed away, I became a salesman at SchneidersLaden. On their go-to modules: EB: Doepfer VCAs: cheap and useful. Also, the Doepfer A-199 spring reverb. It’s cheap and nasty, a little bit noisy, but I just love it, feedback patching with it is wicked too. I also blew mine up by reversing the power and they fixed it for free. Then, Pamela’s Pro Workout. Can’t leave home without it. It constantly amazes me with the things it can do and how programmable it is. J-M F: The MFB Dual LFO: two LFOs in just 8HP with switchable waveforms, cross-modulation, and attenuated CV inputs — a solid modulation source. Also, the MFB Digital Delay, which brings vintage delay vibes in just 8HP and is, in my opinion, pretty underrated. I’ll use the Bastl Instruments Pizza for precise and bitey sounds, and the Verbos Electronics Complex Oscillator for warmer, organic leads. For sequencing, the Verbos Electronics Multi-Stage. As you may realise, I prefer everything hands-on without too much menu diving! The only exception is the ALM MFX. I just love experimenting with it. On creating SchneidersKeller: EB: It was important for us to have lots of places to sit in the space so that people could relax and also not be too

intimidated by massive racks of modular. Yes, a massive system is wicked, but we really want to present the modular as distinct instruments that can interact with one another. I want people to feel comfortable coming to the space, take their time learning or checking out a module or instrument and leave with the correct tool. A lot of the time when buying stuff online you might think that something is exactly what you want, then once you get it in your hands or start using it, it just doesn’t work for you. J-M F: For us, having a shop also means building a community. Engaging with other artists and synth enthusiasts is always fun and inspiring — not just on a technical or musical level, but also through the simple joy of connecting with people who share the same passion. On what’s next: EB: More live shows... We recently had our first live event with Yann Tiersen. It was really exciting to have a live show in the basement. Yann is a master, total synth nerd and just such a nice guy! We have a great‑sounding Neumann system, the space is super flexible so we can move

October 2025 / www.soundonsound.com

everything around and make the space suit the event. I’m not sure we’ll get away with a full‑on rave, but an intimate space where people can get up close and personal with the artists and their gear is what we’re aiming for. J-M F: And workshops! Sometimes people wander down the stairs [from Rough Trade] looking for more records, only to discover our little synth dungeon. Often, they’re curious. Once we encourage them to tweak a knob or push a button, they usually get excited — so why not show them the basics in an intimate setting? We also want to inspire those who are already a bit more advanced. Our goal is to keep the momentum going and help them get even more out of their systems. We also plan to have workshops which are hosted by the manufacturers themselves, who dive deep into their own creations, explain what drives them, and share their unique design philosophies. On the culture of modular: EB: I think there can be a culture of ignoring the basics... Doepfer absolutely has you covered here. People want the sexy things. But the

basics are really where the fun starts. Understanding the basics means that the sexy things are incredibly more useful. I am constantly amazed by what I can get out of the Doepfer Basic System 3 we have in the shop. Once you start thinking about truly modular systems you can build all sorts of unique sequencers, or weird modulations with switches, clock dividers and function generators. You don’t need that new, exciting thing because you can create a totally different thing each time you patch it. J-M F: In my opinion, modular synthesis is rooted in experimentation, individuality and hands-on sound creation. It encourages building a personalised system and engaging deeply with the creative process — while also requiring discipline to explore the full sonic potential of your setup. However, I feel that parts of the scene have shifted: modular has increasingly become a collector’s game. Limited releases, aesthetic-driven purchases and rising prices can sometimes overshadow its original purpose, which is sonic exploration. William Stokes

Mesa. The only desktop interface with legendary Lynx quality.

• Four low-noise microphone/line/Hi-Z switchable inputs • 14 total inputs and outputs with 2 loopback channels • Dual headphone outputs with dedicated D/A and independent level controls • Compact and portable for studio and live recording • Aurora(n) audio quality and transparency • Thunderbolt 3 interface for connection to Mac® and Windows® computers • High-contrast capacitive touchscreen interface • Flexible routing and monitor control • Optical I/O for S/PDIF or ADAT; Coax S/PDIF I/O; Word Clock in and out • SynchroLock 2™ low-jitter sample clock • MicroSD card slot for recording and playback

Get complete specs & info at lynxstudio.com

Lynx Studio Technology • Costa Mesa, CA • 714-545-4700 • lynxstudio.com

ON TE ST

Analogue Mixer

HH Q16FX Digital mixing might be all the rage, but sometimes a simple analogue desk is exactly what you need. PAUL WHITE

hile we now have a variety of digital mixers from which to choose, many with quite different modes of operation, there’s something reassuring about a conventional, knob-laden analogue desk that you can pick up and use straight away with barely a glance at the operation manual. The HH Q16FX is exactly such a product, offering all the essentials for hands-on mixing. It is solidly built in a metal case with a lifting handle inset into the left‑hand side, and has direct mains power (100-240 Volts) rather than an annoying external PSU. Practical colour coding is used for the control knob caps to aid navigation.

HH Q16FX $499 pros • Rugged construction. • Intuitive layout. • Good audio quality with sensible effects.

cons • No obvious cons, as long as you don’t need more than one foldback monitor mix.

summary A straightforward and easy-to-use mixer that will serve well on a range of gigs. The USB interfacing and built-in feedback detection are handy extras to have!

October 2025 / www.soundonsound.com

If you’re not familiar with the HH name, the original company launched some of the first solid-state guitar amps and PA systems, their most distinctive feature being their electroluminescent front panels. Today the name belongs to UK brand Headstock. Designed in the UK and then built ‘off shore’, the HH Q16FX is one of a range of Q-series analogue mixers offering from six to 16 channels. These seem to be designed primarily for live-sound applications, though they also function as 2‑in/2‑out 24-bit audio interfaces running at 48kHz. In the Q16FX, the 16 channels are deployed as eight mono mic/line channels, two stereo line-only channels and two stereo/mono mic/line channels. Channels 1‑8 include a single-knob compressor, and every channel has a three-band EQ operating at 12kHz, 2.5kHz and 80Hz. The mono channels have a separate 75Hz low‑cut filter too. DSP effects are included, with a choice of 16 effect types based around the mainstays of reverb, delay and echo. A nine-band graphic EQ is available to process the main or monitor mix, and this includes a feedback detection feature so you know which band is causing the problem. Channels can be routed to the man output and/or to a pair of subgroups — useful if you want to control the whole drum mix using a single fader. There’s a single pre-fader aux send per channel for setting up a monitor mix, and there’s also

a post-fader send for the internal effects, but with the option to also send that signal to an external effects unit. The mono mic inputs and main mix outputs are on balanced XLRs, with all the other connections being quarter‑inch jacks, other than the inputs for the USB/phono channel (which are on RCA phono sockets and USB‑B).

Channel Features The eight mono channels have identical features, starting with separate inputs for mic (XLR) and line (jack) sources. A further TRS jack functions as an insert point. Phantom power is toggled globally from a rear‑panel switch. The input gain control and low-cut button are at the top of the channel strip, followed by the one-knob compressor and the three-band EQ, which has a ±15dB range. The aux and effect sends come next, followed by the pan knob. The single pre-fader send means you can only set up one dedicated monitor mix, though for smaller gigs I have to confess that I usually take the lazy approach and just feed a copy of the main mix taken from the monitor outs to the foldback speakers. Stereo/mono channels 9+10 and 11+12 lose the insert point and compressor but otherwise have similar appointments to the first eight channels. Stereo operation is for the line inputs only — if used as a mic channel, then the channel is mono. That leaves channels 13+14 and 15+16, which are line-only stereo channels with no input gain control and no low-cut filter. All the controls from EQ downwards are the same as for the other channels.

The channel faders run from -∞ to +10 dB, and a peak LED warns of overloading. There are the usual mute and solo buttons, as well as routing buttons for main and submix, with the pan control steering between sub 1 and sub 2. The solo mode can be switched between solo-in-place and PFL modes (the latter routes soloed channels to the monitor and headphone outs) using a button next to the main output meters. A channel’s signal can be sent to the subgroups and/or main bus (or neither), and the subgroups can themselves be sent to either the left or right channel of the main stereo bus, or both (the former scenario allowing for a single stereo bus; the latter for two mono busses feeding the main mix).

Master Section The master section of the mixer is dominated by a nine-band graphic equaliser, which may be inserted into the main output or the aux output. There are rotary controls for the effects send and aux return, and a volume control for the phono/USB inputs, with a button to route these to the main mix bus. The headphone output can carry the main output, the phono/USB input or the subgroups (and also the solo signal when solo/PFL is in use). The graphic EQ panel houses buttons for Feedback Detection, EQ bypass and main/aux routing for the graphic

EQ. Note that feedback detection is not the same thing as automatic feedback suppression: red LEDs within the sliders illuminate if the energy in that band seems excessive, indicating that feedback may be occurring, leaving it up to the engineer to pull that band down. If the feedback detection is turned off, all the fader LEDs stay illuminated. Faders are used for the aux and effects sends, the two subgroups and the main left/right outputs, with a pair of 12-section LED meters showing the output, PFL or solo levels. The effects section is tucked away in the top right‑hand corner of the panel with a rotary switch selecting which of the 16 effects is in use. These are all variations on delay and reverb, which is by far the most sensible choice for a live desk. A button activates the effects and there’s a knob that adjusts one key parameter for each effect type — typically delay repeats or reverb decay time. I would have liked a tap tempo button for the delays as that’s a feature I use a lot when mixing live shows, but here you have to use the parameter knob to set the delay time. A footswitch jack allows for remote effects bypass. All the connections for the master section run along the top edge of the panel, and these comprise stereo returns for the aux and effects (if using an external processor), the mono aux and effects send outputs, footswitch and headphone jacks,

All the controls and connections are on the top panel, with the exception of the power and phantom power switches, the IEC mains inlet, and the USB port, which are on the rear.

two subgroup outs and stereo monitor outs. The two main outs are XLRs and there are also stereo RCA phono ins and outs. The USB‑B socket for interfacing is on the rear panel, along with an IEC mains inlet, power switch and the 48V phantom power switch.

Impressions This mixer is delightfully predictable and intuitive with status LEDs where needed, which is just what you need when mixing live sound. The bright red LEDs can be a bit dazzling when working in a dark corner, but then that’s true of most similar mixers too. As long as you pay heed to gain structure when setting the input and fader levels, the mixer operates with minimal background noise, and the one-knob compressors work really well, even on electric guitar. The effects section offers useful delay and reverb treatments, but note that when you change presets it can take a couple of seconds for the new parameters to settle down. Having a footswitch option to bypass the effects is very welcome. The built-in USB audio interface means that you can record a stereo mix, either at gigs or as part of a recording project at home. The interface is class compliant so should work without the need for additional drivers, and the USB return could come in handy for backing tracks or intermission music. It worked right away on my Mac, identifying itself as an HH Q Series Mixer in System Settings. For most applications the fixed three-band EQ works fine, and unless you are a seasoned sound engineer, dealing with a pair of fully parametric mid bands can sometimes be more trouble than it is worth anyway. The EQ frequencies are well-chosen and there’s plenty of range to get the job done. The graphic EQ can help smooth out room issues, and the feedback detection could be useful if you’re not certain which frequency band is howling away. Taming feedback with a graphic EQ isn’t as benign as an automatic feedback suppressor that deploys very narrow notches, but on a mixer of this type it is a welcome and practical addition. To summarise then, there’s nothing revolutionary about this mixer but it offers all the essentials, plus sensible effects and feedback detection, in a very practical and affordable package. It is solidly built, well thought-out and should meet a range of live sound and basic recording needs. $ $499 W www.hhaudio.com www.soundonsound.com / October 2025

ON TE ST

Not only does this compact pedal offer convolution reverb — it can also capture impulses from your hardware verbs or room mics. PAUL WHITE

es, I know... the world already has plenty of reverb pedals! But Hotone’s Verbera takes a somewhat different approach from most, as it’s based on a convolution reverb engine. Guitarists may be familiar with convolution processing, which uses impulse responses (IRs), for speaker simulation, but convolution reverbs require longer IRs, and while they’re now common in the DAW plug-in world, they’re pretty rare when it comes to physical stompboxes.

Overview The Verbera is a fairly compact pedal, featuring six knobs, two buttons and two footswitches. Two of the knobs (Preset and IR) are push/turn encoders, and one button doubles as Menu and Alt, allowing the knobs to take on alternative functions. The knobs are labelled Decay, Atk (attack)/Pre-delay, Tone/Mod, Preset, Mix, and IR. The second button is intriguingly labelled Clone — I’ll come back to that. Coloured LED rings around each knob light up orange, but when Alt is engaged the relevant knobs display turquoise rings. A small colour display shows the preset name by default, but this flips to showing function and value when a control is adjusted. The dual footswitches handle bypass (which can be set to buffered or true bypass) or preset

Hotone Verbera

Hotone Verbera Convolution Reverb Pedal up/down in one mode, or to bypass plus a Freeze effect when in Control mode. Presets can be selected directly by using the Preset encoder, with the display showing the current preset name as well as those immediately before and after it in the preset list. Up to 200 presets (100 factory and 100 user) can be stored, and a free desktop app called Neon Collector handles preset and IR management, including the import of third-party IRs. The Verbera supports third-party IRs at 24- or 32-bit wordlengths, and 44.1, 48, 96 or 192 kHz sample rates. The input is on a quarter-inch TRS jack that accepts mono or stereo signals. The stereo output is presented on

two quarter-inch jacks or, alternatively, a single TRS output jack can carry the stereo signal. There’s a USB port for connecting the Neon Collector app, which also manages firmware updates, and there’s a pair of 3.5mm TRS jacks for MIDI in and thru, with the MIDI input jack also able to accept an external expression pedal. Conversion is 32-bit at 192kHz. Note that although the PSU requirement is the standard 9V centre-negative, the specified current draw is a chunky 1A — though I’m informed that the pedal will in fact run fine from a 500mA supply. It can also be powered over USB, which requires 5V 2A, and a suitable PSU is included.

$300 pros • Capable of excellent real and imaginary reverb types. • Can load third‑party IRs. • Can capture room mics or hardware as IRs. • Very easy to use.

cons • The app feels a little basic.

summary The Hotone Verbera offers something genuinely fresh in the reverb pedal market — and at a very sensible price.

October 2025 / www.soundonsound.com

The Verbera has pretty versatile I/O for a compact pedal. For example, the MIDI input jack can alternatively accept an expression pedal, while a stereo output signal can be delivered on dual jacks or a single TRS jack on the rear.

The convolution engine provides an impressive range of reverb types — up to 10s stereo or 20s for mono — courtesy of 120 onboard IRs, though with the ability to import your own there’s plenty of scope to create different sounds. Importantly, there’s also a second reverb engine onboard, which is a fairly basic but nonetheless good-sounding algorithmic reverb labelled the XR engine. This has controls for mix, decay, pre-delay and modulation, and it can be used alongside the IR engine; it’s a great way to extend reverb decay times. The presets showcase a wide range of real spaces, from rooms to caves and cathedrals, but also include some quirky IRs for that ‘reverb but not as we know it’ vibe. The quality is excellent, and I was genuinely surprised at how easy this pedal is to use. The addition of the algorithmic engine, with its longer decay potential, means that combining both engines can yield patches with very long decay tails. While I would personally have liked to see a shimmer option in the algorithmic reverb, generally it performs very well.

Cloning While unusual in the pedal world, that’s all pretty familiar to anyone who has used convolution reverb plug-ins. But what I didn’t expect in a pedal of this size is the ability to create new impulse responses from existing hardware or real spaces! Hotone refer to this process as Cloning, rather than the more familiar term Capturing normally used for amp/cab setups, and the pedal supports up to 1024 IR slots. The sine-sweep test signal and the processing that’s required to convert it into an IR are all built in. To clone an existing piece of hardware, you simply press and hold the Clone button to enter Cloning mode. Then, connect the device you wish to clone (in mono or stereo) to Verbera’s inputs and outputs. Pressing the Clone button again initiates the sweep, after which the IR is created. The pedal even generates a name for the new IR, but you can of course rename it. You’re then prompted to save or discard the IR. As cloning processes go, this really couldn’t be simpler: just two button presses from start to finish. I had no issues using this feature, and you can load your newly created IR into any existing preset or create a new one from scratch.

“What I didn’t expect in a pedal of this size is the ability to create new impulse responses from existing hardware or real spaces!” There are relatively few parameters to manage, so a software editor isn’t necessary, and the display provides helpful prompts where needed. When creating clone IRs, it’s important to note that non-linear elements, such as distortion or fuzz, are not captured correctly, and the same applies to effects involving time, like amplitude or pitch modulation. If cloning a delay or reverb, it’s best to disable any modulation and set the mix control to fully wet. Used this way, you’ll get a good capture of the original character. Note that the IR is always 20 seconds long for mono or 10 seconds for stereo, regardless of the actual decay time of the captured source. To hear the full tail, set the Decay knob to maximum. If you need an even longer decay then, as I discussed above, you can use the algorithmic reverb to extend it. While you can’t accurately capture devices that use modulation, sometimes breaking the rules can still yield interesting results, and I’d say there’s scope for experimentation. For example, I attempted to clone a patch from my Particles II granular delay pedal, which adds random octave shifts to the

delay segments — the result was a reverb tail with strange but surprisingly charming filter-like effects.

Summary There’s a lot to like about the Hotone Verbera. It combines studio-quality convolution reverbs with a basic but great-sounding algorithmic engine, and it’s very easy to use. The IR decay time can be shortened, the attack profile adjusted, and tone and modulation applied. There’s MIDI control for parameters and presets, and creating your own clone IRs requires only two button presses and re-plugging a couple of jack cables. The build quality is reassuringly solid, the screen and LED rings are thoughtfully designed for intuitive operation, and there’s ample preset storage. And the factory presets are also easy to tweak. It’s great to see something a bit different from the crowd coming to market at such a sensible price — well worth checking out. $ $299.99 W www.halleonard.com W www.hotone.com

The Neon Collector app, which handles preset and IR management, the import of third-party IRs, and firmware updates.

www.soundonsound.com / October 2025

TECHNIQUE

Ultimate Plosive Control Initial track (top) and isolated pops copied to second track (bottom).

Pesky pops proving problematic? Here’s how to pull them down... CRAIG ANDERTON

o matter how carefully you set up a mic’s position or pop filter, when vocalists sing close to the mic some pops can still make it through, so you have to deal with them after the fact. Traditionally, most pop removal relies on filtering extremely low frequencies. But reducing these low frequencies often also affects the warmth and depth of the vocal. This may or may not matter, but when it does then filtering out pops with EQ becomes problematic. And while there are clever AI‑based tools out there now, in my experience they don’t always quite hit the mark. Thankfully, attending to pops manually is really easy...

Cancel Culture My preferred method uses good old-fashioned editing and polarity inversion: first I copy only the pops to a second track, then I flip that track’s polarity so that, with the levels at unity, the copied pops cancel those on the original track. You can then adjust the second track’s level and use processing to ‘uncancel’ some of the pop sound and shape its timbre to taste. Not only does this technique not affect the vocal itself, but also the pops themselves don’t sound ‘thin’, and changing the reduced pop’s timbre doesn’t affect the timbre of the vocal itself. Even a little high-pass filtering

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at a very low frequency will usually do copy — perfect alignment on the timeline what’s needed. is crucial for proper cancellation. In the It really is that simple, and you can MixConsole’s Pre section, simply invert do this in any DAW, but below I’ll take the second track’s polarity. Then play you through it step by step in Cubase. the two tracks together: the pops will First, identify the pops at the beginnings be gone! of words, and slice at the beginning Tonal Tweaks and end of the pop. Remember to zoom in close enough to make sure that the Then, simply adjust the second track’s slice is before any attack-related part level to set the desired amount of pop of the pop, and that the cancellation across the slice after the pop is on whole track. If necessary, a zero crossing (there’s add a touch of high-pass an option in Cubase and filtering to alter the tone of most other DAWs to do this the pops. Of course you’ll automatically when you probably want any more make a cut). Then, create general vocal processing a second audio track. to apply to the whole vocal Now, select all the pops including your treated on the first track, and click plosives, and you can do on a selected pop and that if you submix the two hold the mouse button tracks to a group track down. While continuing (use that track for your to hold the mouse button processing and effects down, press and hold the sends), or by rendering Command/Ctrl and Alt/Opt a sum of the two tracks. keys. While still holding If you’d like to hear these modifier keys, drag this technique in action, the pops down into the there’s an audio file on the second track and release SOS website at http://sosm. the mouse. The order of ag/ultimate-plosive-control clicking, holding down the in which you can hear mouse button and using the raw track and the these modifier keys will Cubase’s MixConsole showing pops fully cancelled, prevent the pops from and then the pops the setup for this simple moving left or right as you pop-cancellation technique. blended to taste.

“Simply adjust the second track’s level to set the desired amount of pop cancellation.”

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

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

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

Universal Audio Est. 1958

TECHNIQUE

Part 4: Mid-Sides & Visualisation Tools We conclude our primer on coincident arrays by exploring the theory and practice of Mid-Sides miking.

HUGH ROBJOHNS

ast month we continued our exploration of coincident (X-Y) stereo microphone arrays, and looked at how the Stereo Recording Angle (SRA) can be altered, either by changing the mutual angle between the two mics, or by changing the mic polar patterns. In this instalment, I’ll delve into another form of coincident mic array — the mysterious Mid-Sides format — and discuss some tools that can help visualise and calculate the SRAs of different stereo arrays. As always, you can find the audio examples that accompany this series at https://sosm.ag/this-is-stereo-media.

in the left (red) channel. But as soon as it starts to move towards the right, some left-channel information is also carried partly in the right (green) channel too. The further the source moves right, the stronger it gets in the right channel.

A central sound source is carried equally in the left and right channels, of course. For the Mid-Sides format, the Mid channel carries everything, no matter where it is positioned in the stereo image. But a central sound is 6dB louder than if

Mid-Sides The term Mid-Sides seems to induce fear or confusion in many and, because the explanations often use equations, some seem to think it will be hard work! But Mid-Sides is really nothing more than the conventional left-right (L-R) stereo format viewed through a different lens, and in practice it’s very easy. L-R and M-S both carry exactly the same information, and are completely interchangeable through a simple ‘matrix’ process. Diagram 1 illustrates how audio information is portrayed in each channel in both formats. In the L-R format (top), a fully-left sound source is carried only

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Diagram 1: M-S and L-R are both stereo formats, and carry exactly the same stereo information, but different parts of the signal are conveyed on each of the two channels.

Diagram 2: EMI’s Stereosonic Shuffler employs a high-shelf cut in the Sides channel to narrow the stereo image slightly at high frequencies.

that sound was fully left or right, because the same signal exists in both channels (if you add two identical signals the result is 6dB louder). The Sides channel, as the name implies, is strongest for signals that exist only in one channel (whether left or right) at either side of the stereo image. The signal strength falls as the source moves towards the centre, and a central sound source is not carried in the Sides channel at all. And all of this behaviour is defined by two simple equations: • M id = Left + Right (the sum of both channels). • Sides = Left - Right (the difference between channels). Hopefully you can see that, since you’re adding the left and right, the Mid channel is exactly the same as a conventional mono sum. The Sides channel, on the

other hand, essentially carries all the stereo-width ‘goodness’ that you lose when summing to mono — the wider a sound source in the stereo image, the stronger it will be in the Sides channel. Importantly, exactly the same sum and difference matrix processing converts the other way, too: • Left = Mid + Sides (sum) • Right = Mid - Sides (difference) Consequently, whatever format you feed into an M-S matrix, the opposite format will appear at its output. L-R becomes M-S, and M-S becomes L-R; it’s the same process for both conversions. There’s a small but important caveat, though! Imagine you want to convert L-R to M-S to adjust the stereo width, and then convert the narrowed/widened signal back to L-R (this is precisely what most

Diagram 3: A Blumlein Mid‑Sides array, matrixed to create a ‘virtual’ L‑R stereo array.

www.soundonsound.com / October 2025

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SPOTLIGHT

Universal Audio Apollo x16D

Dante Interfaces

Once the preserve of large studio facilities, Dante is now becoming common in smaller setups. We talk you through some of the options...

LUKE WOOD

ince its introduction in 2006, Audinate’s Dante AoIP (Audio over IP) protocol has become hugely prevalent in the worlds of high-end live sound and broadcast. It’s easy to see the benefits for such applications: the ability to run over standard Ethernet networks, support for high channel counts and sample rates, being able to route input

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sources to multiple destinations, redundant backups that’ll help to avoid show-stopping failures, and so on. Those same benefits make Dante a compelling choice for large, multi-room commercial recording and post-production facilities, too, but there are now a growing number of products aimed at smaller studios. Generally speaking, Dante interfaces fall into two categories: simple devices that offer a selection of Ethernet ports that can be used to

connect your I/O devices of choice, and more traditional audio interfaces that feature their own selection of local I/O and use Dante connectivity as a means of expanding their channel count and integrating with other networked devices and systems. In this month’s Spotlight, we’ll be taking a look at a mixture of the two, but specifically those that are capable of passing audio between a DAW and Dante network via USB or Thunderbolt.

Avid MTRX Studio

Avid MTRX Studio / MTRX II Designed to bring the power of Avid’s large modular I/O systems to smaller spaces and budgets, the MTRX Studio is a 1U unit that includes 64 channels of Dante connectivity alongside 18 analogue mic and line-level connections, 64 DigiLink channels and 16 ADAT channels. Connectivity with Pro Tools HDX systems is complemented by an optional Thunderbolt 3 card that allows for a native connection to Macs and PCs, and comprehensive routing functionality via DAD’s DADman software application makes it possible to pass audio from any input to any output, allowing users to share connectivity across multiple devices and different DAWs simultaneously. There’s built-in support for monitoring Dolby Atmos in layouts up to 9.1.6, and EuCon integration means that the interface’s monitoring functions can be controlled directly from Avid control surfaces. Its larger sibling, the MTRX II, then offers a modular platform that supports 256 Dante channels in its standard form (along with 64 DigiLink and MADI channels), and can be expanded to accommodate a massive 1280 channels that can be distributed to multiple audio networks — even if they’re running different sample rates and sync’ing to different clocks. $ TRX Studio (Thunderbolt) $5798, MTRX II from $7798. W www.avid.com

DAD AX Center / AX64 / Core 256 The newest addition to Digital Audio Denmark’s Thunder Core family packs a huge amount of connectivity into a compact 1U unit — fully expanded, the AX Center boasts a whopping 984 inputs and outputs! In its standard form, it supports 256 channels of Dante alongside 64 channels of MADI, 16 channels of ADAT, as well as featuring a pair of mic/instrument inputs and eight analogue outputs (two stereo monitor pairs and two stereo headphone outputs); the unit’s Thunderbolt 3 connection is capable of carrying 256 inputs and outputs to/from a host computer. A pair of expansion slots make it possible to add yet more I/O, with options including analogue mic pres and line-level I/O, AES, MADI and yet more Dante — with the appropriate cards installed, the total Dante I/O count tops out at 512. If that’s not enough, then its larger sibling, the AX64, includes eight expansion slots and supports a staggering 1872 input and output channels! Finally, there’s the Core 256, a digital-only interface that features 256 channels of Dante and 64 channels of MADI, and has been designed with live sound and post-production duties in mind. Routing and monitoring control for all three interfaces is taken care of by the company’s DADman application. $ AX Center $4099, AX 64 $5099, Core 256 $3099.

Dante Virtual Soundcard An alternative to a hardware interface comes in the form of Audinate’s Dante Virtual Soundcard application. As its name suggests, the software allows an Ethernet-equipped Mac or PC to appear as an audio interface on a Dante network, making it possible to route audio between computer applications and any connected hardware devices. Two variants are available: Standard and Pro, both with a selection of licensing options — Standard can be rented for 30 days or bought as a perpetual licence, Pro is available either as an annual subscription or on a fixed-term licence. The key distinction between the two is channel count: Standard will accommodate up to 64 inputs and outputs at 44.1/48 kHz (falling to 32x32 at 88.2/96 kHz and 8x8 at 176.4/192 kHz), while Pro offers 128x128 operation at sample rates up to 96kHz, and 16x16 at 176.4/192 kHz. Because the necessary processing takes place on the host computer, DVS does incur higher latency than opting for a hardware interface, although for situations that don’t require real-time monitoring — capturing a live performance from or offering playback for a virtual soundcheck on a Dante-equipped console, for example — it can offer a neat, cost-effective solution. • • • • •

Standard 30-Day: $19.99. Standard Single Machine: $49.99. Standard Transferable: $99.99. Pro Annual: $124.99 per year. Pro Term: 1-Year $124.99, 3-Year $299.99.

W www.getdante.com

W www.digitalaudio.dk DAD AX Center

DAD AX64

www.soundonsound.com / October 2025

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SPOTLIGHT

DANTE INTERFACES

Focusrite Red 8Line

Focusrite Red 8Line/16Line

Focusrite’s RedNet range includes a versatile selection of interfaces, converters and monitoring controllers that all feature Dante connectivity, catering to everything from compact studio and live recording rigs to modular, scalable systems for large, multi-room facilities. The Red 8Line is a 1U unit that, as its name suggests, features eight analogue line-level inputs and outputs, along with 16 channels of ADAT I/O, stereo S/PDIF connectivity, a pair of the company’s Red Evolution mic preamps, two high-impedance instrument inputs and 32 channels of Dante. The Red 16Line then increases the local analogue line-level I/O count from eight to 16. Both units can be hooked up to Pro Tools HDX systems thanks to DigiLink ports, while native Mac and PC connectivity is provided over Thunderbolt 3. $ Red 8Line $3699.99,

Red 16Line $4499.99.

W www.focusrite.com/red Focusrite RedNet TNX

Focusrite RedNet TNX

The latest addition to the RedNet line-up offers a Thunderbolt-equipped alternative to their PCIeNX card, bringing high-channel-count Dante connectivity to laptops and desktop users with no spare PCIe slots. Out of the box, it’ll support 128 inputs and 128 outputs to/from a Dante network at 192kHz, and that count can be doubled to a whopping 256x256 channels by purchasing an optional software upgrade from Audinate’s Dante Activator software — when running in 256x256 mode, the maximum sample rate is 96kHz. Connection to a host computer is handled via Thunderbolt 3, and a pair of rear-panel ports that each provide 15W of power make it possible to run the RedNet TNX alongside other Thunderbolt devices. Dante connectivity is taken care of by two Ethernet ports that can be configured to operate in switched or redundant modes; with the former, the TNX will act as a network switch, allowing it to be daisy-chained

with other Dante devices, while the latter can be used to create a redundant backup system. Drivers are available for both Mac and Windows systems, with all of the device’s routing and monitoring functionality handled by Focusrite’s RedNet Control software. $ $2199.99. W www.focusrite.com

Neumann MT 48

While the MT 48 was originally designed to support RAVENNA and AES67 networked audio, Neumann say that their users requested a version that would be compatible with Dante. From the end of 2024 onwards, the MT 48 has been shipped with a Dante Ready sticker on its packaging. By default, it still comes loaded with the original firmware, but those who wish to add Dante connectivity can do so by installing a free firmware update from Neumann, and purchasing the appropriate licence from Audinate — there are 8x8- and 64x64-channel options available. With the appropriate firmware and licence installed, the MT 48’s Ethernet port can be used to connect the interface to a Dante network, expanding its built-in I/O with up to 64 channels of Dante inputs and outputs. Local connections include a pair of built-in mic/line inputs and four line-level outputs, along with ADAT I/O and a pair of quarter-inch TRS sockets that can function either as MIDI or GPIO ports. Connectivity to a host computer is provided by a USB-C socket. $ $1850. W www.neumann.com

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RME Digiface Dante

RME’s Digiface range includes five simple interfaces that cater to the most commonly used digital and AoIP formats. The Digiface Dante packs in 64 channels of Dante and 64 channels of MADI I/O, offering a massive 256 channels of audio over USB 3.0. Dante connectivity is provided by four gigabit Ethernet sockets (two Primary and two Secondary), while a pair of BNC connectors can be configured either as word‑clock or MADI I/O. As well as serving as inputs and outputs, the dual Primary and Secondary Dante ports allow the interface to function as a network switch, and also make it possible to set up a redundant secondary network for some peace of mind in mission-critical applications. Despite the focus on digital formats, the Digiface Dante is equipped with a quarter-inch TRS socket that can function either as a headphone output or a stereo line-level analogue output. Signal routing and mixing is carried out by RME’s renowned

TotalMix software, although the lack of DSP in the Digiface line-up means that the range of built-in effects and processors included in TotalMix FX are not available. In addition to its primary function as a USB audio interface, the Digiface Dante is capable of acting as a standalone 64-channel Dante/ MADI converter. $ $1749. W www.rme-audio.de

Universal Audio Apollo x16D Designed with both live sound and studio applications in mind, the Apollo x16D combines Universal Audio’s

renowned DSP-based UAD plug-ins with the flexible routing and redundancy options provided by AoIP networking. For live applications, the interface offers a convenient way to integrate UAD plug-ins into a live console running on a Dante network, while studio users benefit from the ability to easily set up a multi-room Apollo-based system over a standard network. The latest addition to the Apollo line-up complements the standard Apollo x16’s connectivity with 16 Dante inputs and outputs, and it’s possible to link up to four units for a maximum of 64 channels of Dante I/O at sample rates up to 96kHz. Monitoring options cover everything from stereo to a variety of immersive formats (Dolby Atmos, Auro-3D, Sony 360 Reality Audio) in layouts up to 9.1.6, with built-in room-correction facilities provided by Sonarworks integration. The Apollo x16D can be purchased in two versions: Essentials+ and Ultimate+, which come bundled with 33 and 124 UAD plug-ins/four LUNA Extensions respectively. $ Apollo x16D Essentials+

Neumann MT 48

$2999, Ultimate+ $3999.

W www.uaudio.com www.soundonsound.com / October 2025

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INSIDE TRACK

The Making Of KPop Demon Hunters Five years in the making, Netflix’s smash hit animated movie tested its music production team to the limit. PAUL TINGEN

eleased on 20th June this year, KPop Demon Hunters has already become the second most-watched movie on Netflix, with over 200 million views. Its soundtrack album has reached number one in more than a dozen countries, while the single ‘Golden’ has topped both the UK and US singles charts. At one point, seven of the soundtrack album’s songs could be found in the Billboard Top 100.

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Ian Eisendrath was producer on all seven tracks, in addition to producing all of the vocals, jointly vocal arranging alongside EJAE, conducting the orchestral sessions, and supervising the music editing and mixing processes, which were handled by music mixer Curtis Douglas. A graduate from the University of Michigan who has always been strongly interested in “music that drives story”, Eisendrath has worked as executive music producer and music director/supervisor on many Broadway productions, as well as several movies

and TV productions. In 2022, he was hired by Spring Aspers, President of Music at Sony Pictures, to be at the heart of the music-making for the KPop movie. “We had a really wonderful process where the directors, Maggie Kang and Chris Appelhans, Michelle Wong, and I, would spend many hours in meetings, delving into what each moment in the movie should sound like. What was the character arc? The tempo and tone of the music? What did the feeling need to be? How does the story progress from the first line of the lyric to the last? What are the references we want

Producer Ian Eisendrath coordinated the fantastically complex production process behind the music for KPop Demon Hunters.

to share with the songwriters? Sometimes these references were the demos that EJAE and [Costa Rican composer] Daniel Rojas had made. More often, we worked with various groups that had stabs at creating these musical moments.” Many different songs were written for each moment in the film, of which there often were dozens of different versions. Vocal sessions were extraordinary, with hundreds of backing vocals and over 1000 vocal takes for most songs. Eisendrath worked with recording engineer Derik Lee to carefully comp the vocals on each song, often “syllable by syllable”, and they handed up to 500 vocal tracks per song to mixer Curtis Douglas, who spent weeks working on them in similar levels of detail.

To The Max EJAE started work on the project at the end of 2020, before the movie had been given the green light. Born Kim Eun-jae in Seoul, EJAE is currently living in New York. A singer, songwriter and producer, she has written material for K-pop acts like Aespa, Twice and others. She also ended up being the singer for the main character in the movie, Rumi.

As well as co-writing many of the songs, EJAE is also the singing voice of lead character Rumi.

“I got onto the project via a friend, Daniel Rojas. In the beginning it was just the two of us creating demos for the directors to use in early screenings, there wasn’t a script yet. Our job was to try to figure out the sound of the girls in the movie, to develop a blueprint for the kind of K-pop songs in it, and for the traditional Korean aspects. None of our early demos ended up in the movie, apart from some melodies of ‘How It’s Done’. “With regards to K-pop, it differs from American pop in that there are many more things going on. Whether it’s chord or melodic changes throughout the songs, or tons of stacked vocals, K-pop is very maximalist. K-pop is usually girl groups with multiple members, and each member has to shine. There may be a rapping section, or a high note section, where a vocalist shows off her vocal skills. Because it’s a group thing, there are many background vocals. Harmonies are huge in K-pop.” “EJAE and Danny wrote many, many songs,” recalls Ian Eisendrath. “When I joined the project, these demos allowed me to understand what kind of songs the film-makers were thinking of in specific moments. Spring [Aspers] then handpicked

and hired an unbelievable stable of writers who were all experienced K-pop hit‑makers. They included The Black Label producers like 24, Ido, Teddy Parks and Kush, as well as Lindgren, Stephen Kirk and Jenna Andrews. “Early in that process, we realised that bringing in EJAE as a collaborator for The Black Label instrumentals was a real win, because they have their distinctive sound, and EJAE has hers, and combining their forces worked really well. After things started to get going, a good friend of mine, Mark Sonnenblick, who I met while working in theatre and on other film projects, and who is great at writing lyrics for narrative songs, came in to collaborate with EJAE, as well as with Jenna and Stephen. “The challenge for them was to write lyrics that work as a pop lyric, while also fitting the story and the character. These songs needed to have great hooks, and also allow the movie characters to reveal themselves and push the narrative forwards. You can have the most beautiful songs, but if nothing happens during them, except for someone sharing a feeling, the song will be cut down or cut entirely from www.soundonsound.com / October 2025

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KPOP DEMON HUNTERS

to make my life easier. But I had to hit that high note, or it would not be an earworm. Honestly, at that moment, I forgot that I was also the lead singer. It’s not the easiest to sing. I found my range writing these songs!” Mark Sonnenblick: “We knew it’s important to the story that there’s this crazy high note in the song, because she’s not going to be able to sing it by the end of when ‘Golden’ first appears in the movie. Her voice is failing her. So you’re figuring out what the character is feeling, and what information has to be in the song. And the high note first happens at the beginning of the song, so how do you build up to that and after that? In this situation, lyric writing is kind of problem solving. It was like, ‘There’s six things this song needs to do aside from being a great song.’ It’s part of what I love in lyric writing.” More twists ensued before ‘Golden’ was finished, as Ian Eisendrath recollects: “At that point we had a song that tells you what the girls tell the world about their stories. Then we realised that the film was missing a vulnerable moment from Rumi, our hero. So we decided we needed a bridge with Rumi singing her innermost thoughts, and there is this huge visual reveal. I’ll never forget the gasps around me at that moment the first time I watched the film with an audience. To have an intimate moment of close to 45 seconds in the middle of what we hope would be a number one pop song was a risky thing to do, it was wild! The moment made sense in the movie, but I worried that it might not work for the soundtrack album. I remember sending the entire song to Spring Aspers by the time we started to look at the soundtrack versions of the songs, and I asked, ‘Do we need to do something else here?’ And her reply was, ‘No. Leave it.’ Fantastic! That’s how the song was released.”

One’s A Crowd After EJAE and Sonnenblick had written the toplines, Eisendrath collaborated on the vocal arrangements with EJAE, who explains: “As a K-pop writer I have vocal arrangements in my demos. I’m recording tons of harmonies and vocal stacks. I also did the crowd noises in some songs. That was me in my room screaming. I felt so bad for my neighbours! And if there was a lyric change, I had to do it again. I’m talking stacks of 200 vocals that I had to re-record in some cases. “Many of the effects, the crowd noises, harmonies, doubles, ad libs and so on, I did at home, but I sang the leads and main

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Rob Mathes’ string arrangements were tracked at AIR Lyndhurst, with Ian Eisendrath conducting.

harmonies in recording studios, mostly the Power Station in New York. The chorus in the beginning of ‘Your Idol’ in the movie is my voice, that I transposed down four half steps, because they were supposed to be sung by a dude. There were 200 vocals that I had to transpose down, it took so long! Ian would then come in and transcribe my arrangements and make them way better. He’s a musical genius. He’d edit what I’d done and add new harmonies and parts. It was a great collaboration.” As the vocal arrangements were finalised, Eisendrath embarked on the task of recording and then comping the main vocals, with engineer Derik Lee. “There was a lot of recording all over the world over the course of a year and a half. I would do sessions in South Korea, LA, New York, Thailand, London, and if I could not be

present there were many remote sessions, using Zoom and Audiomovers. Modern technology being what it is, this was shockingly seamless. “I produced all the vocals during the sessions, one singer at a time. The first thing I did was teach the melodies and rhythms to the singers, with me at the piano. Derik and I would record four bars or so at a time. We kept extensive notes about every single take, because some songs had over 1000 takes, if not more. I spent time talking with the singers about pitch, tonality, sound, phrasing, timing and groove, vocal style, what the primary word to hit is in a phrase, when to breathe, and so on. And we’d focus on storytelling. What is the character trying to communicate, how do they feel? “Once we had the lead vocals, we’d record backing vocals. Whether they’re my

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FE ATURE

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Dave Sardy • Oasis ‘(Get Off Your) High Horse Lady’ J O E M AT E R A

merican-born producer, engineer and mixer Dave Sardy has worked with household names including Bush, Johnny Cash, Oasis and the Who. A favourite sound from this impressive discography is the guitar drone on ‘(Get Off Your) High Horse Lady’ from Oasis’ 2008 album Dig Out Your Soul. “I loved the song when Noel Gallagher first played it for me, he had a kind of a drone string vibe happening with the way he was playing it on the guitar. He writes a lot on acoustic guitar, so he’ll just sit and play acoustic guitar and sing the song a bunch of times to rehearse and warm up his voice. So, you get an idea of what the song is. With this song he was kind of droning on this one string

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throughout the song, which gave me the idea of using the technique of sympathetic vibrations — the harmonic phenomenon wherein a passive string or vibratory body responds to external vibrations to which it has a harmonic likeness. “Knowing that, I got his guitar, which was a Gibson acoustic, and tuned it all to a chord in the key of the song. I then took his guitar and put it through a Neve Melbourne sidecar console. Because we were at Abbey Road in Studio 2, and because we wanted that punchy Neve sound, I brought over my own Neve Melbourne — which is a 12-channel sidecar — from the States and set it up next to the EMI REDD console. The studio also had a Neve VR console, which we basically used as a table and monitor controller.”

Drone Warfare “To achieve this sound, I had the acoustic guitar set up on a stand on top of the console, and had the drum beat playing from the song onto the speaker, going directly onto the back of the body of his acoustic guitar, where I had a Neumann KM84 mic set up close to the guitar strings, using the monitor speakers to create the vibrations. The KM84 gave it a very focused sound. I then DI’ed

it out of the acoustic as well and sent that to a Vox Super Beatle amp in the live room. Because the KM84 is a really small capsule, it makes the perfect mic for this type of close-mic approach. You need the guitar vibrating and the speaker triggering it, but without getting too much of the speaker. So, you kind of need a really tight‑pattern microphone. “We did a bunch of different takes and kept re-recording it. After four or five iterations, it becomes this sound that has overtones of the room that you’re in, so everything turns into a vibration, giving you this sort of beautiful droning sound that is created out of the rhythm and the key of the song. The finished sound ends up sounding a little like a synth, a drone synth, which it’s not. It’s all created out of that guitar vibrating and the song being played into the guitar and then resonating and being recorded over and over again.”

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INTER VIE W Photo: Alexis Maryon

PLA Y ON

Isabel Gracefield: Recording Mulatu Astatke

Recording a band live in the room is always a challenge — especially when it’s led by a jazz legend who switches instruments without warning! SAM INGLIS

“I

Mulatu Astatke, now 81, is considered the father of Ethio-jazz.

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n engineering culture, there’s this idea that everything is always being recorded with the perfect microphone in the perfect position,” says Isabel Gracefield. “But my experience of really big, high-pressured sessions is that often it’s that first, instinctive decision. You check it’s OK and it’s working, and then you move on. “A lot of times you’re doing these quite prestigious projects and it’s just: that mic was spare, it was sat on top of the piano, so I grabbed it and put it on the stand. And I’ll often say to the assistants, ‘Can you just put it somewhere you think is sensible?’ and I’ll listen to it — and if it’s OK, great, because there’s eight other things that we’re dealing with. And so, it’s not always micro-finessed. It’s just if you don’t like it, it changes.” The sessions for Mulatu Plays Mulatu were at the top of the scale both for pressure and prestige. Over six days at London’s RAK Studios, with Isabel and her

The sessions for Mulatu Plays Mulatu took place at RAK Studios in London, under Head Engineer Isabel Gracefield.

Photo: Alexis Maryon

team engineering, Mulatu Astatke took his band through new interpretations of 11 compositions drawn from his own back catalogue. Producer Dexter Story then flew to Addis Ababa to track overdubs by musicians there, before Gracefield mixed what may prove the valedictory album by the 81-year-old father of Ethio-jazz.

Slow Cooking The complex live sessions tested the resources of both RAK and its Head Engineer Gracefield. Meticulous planning was vital, but so too were pragmatism and adaptability. Above all, it highlighted the importance of the technical team’s ability to serve the artist and the music. “We started talking with the label at least three months beforehand,” recalls Gracefield. “We had multiple Zooms with the label and the producer and James Arben, who plays sax on the record and is also the MD. They were

really clear that they needed to keep to the stage plot as close as possible in the studio, because when Mulatu turns to this side, he expects to see this person. Obviously their live plot is amazing for playing live, but instinctively I looked at it and just went ‘That’s not a recording plot!’ All the drums and percussion were on one side. “So there was a lot of negotiation about whether we could put the drums and the percussion on opposite sides, because I wanted to have a coherent stereo picture. All those choices had to go through the label, the producer and the MD, to make sure Mulatu was going to be comfortable when he walked into the room. It took a while to negotiate how everyone was going to be in the space.”

All Together Now The Studio 1 live room at RAK is a large space that can optionally be divided

into two, isolated areas. Gracefield’s original plan had been to set up the louder instruments such as drums and brass in one of these, with Ethiopian instruments and other delicate sources in the other. However, rehearsals for the sessions forced a major change of plan, when Mulatu Astatke decided that RAK sessions would not include the Ethiopian instruments after all. It also became clear that dividing the space wouldn’t work from the point of view of Astatke communicating with his band. This left Isabel and her team with some serious engineering challenges, because alongside drums, congas, trumpet and saxophone, the band also included much softer instruments such as cello, upright bass and Astatke’s vibraphone. With lines of sight being vital, hiding these away in booths was not an option, so microphone choices became as much about managing spill as they www.soundonsound.com / October 2025

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IS A BE L GR AC E F IE LD • M U L AT U A S TAT K E

The live room at RAK Studio 1 was configured to allow Mulatu Astatke to move freely between instruments.

were about getting the perfect sound for each individual source. Screens and gobos were used to reduce spill as much as possible, and Gracefield employed a lot of dynamic mics, often positioned very close to the instruments they were capturing. Happily, the results were often in line with what the musicians had hoped to hear. “We originally had a Neumann FET U47 on the cello, and we were just getting two percent cello and 90 percent percussion. I grabbed a Sennheiser MD441, and interestingly, the cellist was like, ‘That’s much closer to the

sound that I want to hear on myself anyway.’ It was much more pokey and nasal. There were a couple of tracks where there was no percussion, and

the only microphone you need.’ He put it right at the mouthpiece, and it sounds amazing. It is quite a classic sound, and I hadn’t come across that before. I’m always doing posh things with 87s and Schoeps, but I’ve used it quite a lot since. The trumpet was a Coles 4038, and the bell was almost touching the mic. “We did stick with Schoeps as the overheads on drums, on a stereo bar very close above the kit. They are just so soft in the top end, and I think especially in an ORTF setup, you get this very natural stereo picture with a ton of detail but you

Isabel Gracefield: “The studio environment is really different, and it requires everyone to recalibrate who they’re listening to and why.”

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I asked if he would like us to swap, and he was like ‘Nope, this is the sound.’ So that was a really interesting lesson for me, because I would never have considered a 441 on cello. “Then James came in and he just went, ‘It’s an SM58 on the flute. That’s

Photo: Alexis Maryon

don’t get harshness off the cymbals. The internal balance of the drummer’s got to be immaculate and it helps if they’re a quiet player, but I use them a lot. “The first couple of days that the band were in, they were still very much in live mode, and I had no separation on the drums. So the drums were just in everything, and we were getting this big boomy room sound. The studio environment is really different, and it requires everyone to recalibrate who they’re listening to and why. And so there was a real dialogue between Dexter and the players to try and get a softer sound.”

PZMs & Piano Recording grand piano in a band context can be challenging, because most techniques that work well in other scenarios capture too much spill from other sources, and rely on having the piano lid open. Isabel Gracefield has devised an ingenious trick using inexpensive Tandy / Radioshack PZMs, or pressure zone microphones. “I tape them to the inside of the piano, on the curve of the wood. Basically, one goes down at the bottom to catch more of the low end, and then the other one goes nearer the hammers, and that’s the brighter of the two mics. So that’s sort of roughly your left-right, and then that enables you to close the lid. They can be pretty great. And I prefer PZMs that still have the battery pack attached. When you buy them, off eBay or whatever, they’ll have an AA battery and then people mod them to make them phantom-powered. But if you use the battery, the amazing thing is you get this compression curve on them because the battery can’t draw enough power. So, you get a little, free, very natural compressor in your microphone! “The PZMs are good from a separation point of view, but they are really bright, and sometimes that brightness wasn’t sitting with everything else. So then we had to switch to the [Neumann] U87s, and I think I had some Coles up at one point as well.”

Good Vibes

and less conventionally with a pair of Tandy PZMs (see box).

Maxed Out Mulatu Plays Mulatu was recorded to Pro Tools through the vintage API console in RAK Studio 1. This is not an inline console, and is usually used in a split configuration,

with 24 of its channels handling mic inputs and the other 24 tape returns. However, this wasn’t an option for the Astatke sessions, which required up to 70 inputs to Pro Tools. “Normally, channels 25 to 48 are your mic pres and sends to tape, and then you do the returns on the left side,” explains Gracefield. “But in

Jon Scott’s drum kit was tracked using an ORTF pair of Schoeps pencil mics with MK4 cardioid capsules, plus close mics.

Photo: Alexis Maryon

As well as leading the band, Mulatu Astatke himself switches between a variety of instruments, often spontaneously, and the setup had to be flexible enough to accommodate this. “We had a set list, but we never knew exactly what instrument he might play. So we had a Wurlitzer piano, we had the timbales, or there’s quite a lot of places on the record where all he’s doing is he’s playing wood block and he’s setting the time, but he’s not really playing anything else. Or he might play vibes. Sometimes it would be piano. He would just get up and move, and we would just have to chase after him and make sure we were recording it!” Probably the most important, and the most challenging of Astatke’s instruments to mic was the vibraphone. “Vibes is an instrument where I don’t think there’s a definitive sound,” explains Gracefield. “People aren’t going to know if you’ve put something slightly weird on the vibes. So I think it’s an instrument where you can grab whatever’s handy, have a listen, and if it’s not good, change it. Where spill’s not been a concern, I’ve done C414s on vibes before and they’re lovely. Sony C48s are really lovely. “On this record, it was like, ‘I’ve got to be able to put them on a stereo bar. They’ve got to be directional enough to control spill, and they’ve got to have some definition.’ It could have been SM57s, but I think they’re a bit too aggressive. The 421s are a bit fatter and it could have been Schoeps, but they’re not directional enough. So we just narrowed into that.” Sennheiser MD421s were also used on congas and timbales, whilst the studio’s grand piano was miked both conventionally with Neumann U87s

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SAMPLE SHOP

variation to keep them interesting, and given the low cost of the instrument, it has to be seen as good value. Paul White $34 www.sonoracinematic.com

Fracture Sounds Orchestral Atmospheres

Kontakt Instrument

HHHHH Fracture Sounds’ libraries excel at two things: detailed orchestral sampling and atmospheric textures. Orchestral Atmospheres brings these strengths together in one elegant and extremely expressive Kontakt library. More than just an ambient/atmosphere tool, Orchestral Atmospheres offers a deeply inspiring sound palette for film composers, sound designers and even pop/hip-hop producers. The library, which runs in Kontakt Player 7.6 or later and requires about 2.25GB, is built on 45 textured layers — mainly strings, brass, and vocals — drawn from Fracture’s own orchestral recordings. These layers are warm and dynamic, working as well in isolation as they do layered together. The 20 snapshot presets offer a quick tour of the possibilities, from cinematic beds to shimmering blends that combine orchestral timbres with ambient design. At the core of the library is a three-layer engine. Users can stack or crossfade layers in real time, making each patch feel alive and constantly evolving. The modulation wheel controls these blends smoothly, adding movement and expression. A Random button instantly creates new combinations, encouraging experimentation and opening up endless possibilities. On top of the engine, the instrument’s controls are straightforward: octave‑shifting, soloing, attack/release shaping, filters and tempo-sync’ed LFOs. The Lo-Fi section adds tape saturation, noise and speaker emulation for extra colour and grit. The reverb section is especially effective, with five options — convolution, room, hall, shimmer and lush. Delay and filter controls further refine the texture, making it easy to sculpt each sound. The GUI is elegant and easy to navigate. It’s fast to load a patch, make a few adjustments, and arrive at something inspiring. Most factory presets lean towards slow-attack, reverb-heavy pads, but tweaking the envelopes quickly produces

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punchier, more traditional orchestral results. While this isn’t a library for sharply articulated lines or solo passages, it excels in creating movement and depth, and can be very effective for ensemble pads in orchestral mock-ups. Overall, the emotional impact is immediate. The textures have warmth, the blending is seamless, and the overall design feels thoughtful from start to finish. Benjamin Boukris $59 www.fracturesounds.com

Native Instruments Scene: Nightshade Kontakt Instrument

HHHHH When I first opened Nightshade, the third release in Native Instruments’ Scene series of cinematic sounds for media composers, the patch that loaded (seemingly randomly) was The Omen. A fitting start for an instrument filled with an abundance of ready-to-go creepy sounds at your fingertips. A veritable all-star ensemble of eerie sounds have gathered in this light-on-disk-space (728.5MB) Kontakt 8 instrument: low rumbles, subliminal subs, baleful winds, breathy whispers, detuned notes, sharp scrapes, sinister organs, crackling transmission, ominous drones, granular voices and more. The groundwork for ghoulish score has been laid perfectly, straight out of the box, before you even begin to customise the settings to your taste. The other impressive feature of the instrument is how easy it is to use while also allowing an enormous amount of flexibility. The core functionality is simple: each patch consists of two layered sounds, and adjustments are made via a large X/Y pad. Crossfade between the two sounds on the X axis; adjust modulation intensity on the Y axis. Both parameters are easily controlled through MIDI CC making it possible to modify sounds in real time, while playing. Further fine‑tuning is possible with macros (Attack, Release, Motion, Mask, Tape) and effects, which come with creative names like Crushed Sky, Introspection and Double Ring but are in effect a function of simple or complex reverb, delay, degradation and modulation. The drop‑down menu for each individual sound allows you to select a scale and mode of your choice for the composition.

Nightshade is polyphonic aftertouch-enabled, so those with compatible MIDI controllers are in for a treat. You can adjust volume, tuning and motion for individual notes as you play, adding even more expression to your performance. Even for basic controllers, the patches respond very well to velocity sensitivity, which is great, say, for choosing if you want the sounds to slowly fade in and creep up on you, or attack immediately. Playing with release time is particularly satisfying in this instrument: sudden stops and starts lend the music an unpredictable, abrupt feeling that adds to the disquiet. The sounds in Nightshade are spooky across the spectrum, and while billed as a horror instrument, I found it capable of quite nuanced dark compositions; there are varying levels of intensity, it’s not all jump scares. While dread, doom and otherworldly soundscapes are serviced well, there are some delightfully old-school patches that bring to mind more subtle movie moments of suspense and rising tension that were scored orchestrally in old-timey movies. The patches span several octaves of playability, adding more colours to your palette. As they rise in pitch, the sounds gain texture and sometimes even movement. You’re not limited to a sweet spot for each patch, instead you can use the different colours of the low to high registers to shape the sound. The instrument handles movement wonderfully with rhythmic pulses, jagged LFOs, and even what feels like triplets. It’s actually startling to realise that, in a horror context, a steady low tone has an air of despairing finality to it while a more kinetic, panned rumble promises escape for your protagonist. The darkness hasn’t taken root, you can still run! If you’re the type to yell at the screen, there’s a sound here that will inspire you to do that. At only $29, this library is an absolute steal. It has great sounds out of the box, and the flexibility to customise them extensively. I can’t think of what else you might need to start setting a properly sinister mood. Sonal D’Silva $29 www.native-instruments.com Audio examples of this month’s libraries are available at www.soundonsound.com.

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WORKING IN THE BOX AT H E E N S P E N C E R

here was a time when performing at a gig meant 10 trips to a Transit van, carrying kit that was primarily constructed from wood, metal and carpet. As for recording, that didn’t become mobile until the introduction of the Portastudio, where we suddenly started to get a taste of the locational freedom that was to come. However it wasn’t until an early encounter with an Atari Falcon that things really began to change for me, as the idea of being able to operate purely in the digital realm started to become a reality. By the early 2000s I had switched my MDF2, weighted controller, synth, double

X-stand, 10U wheeled rack unit, desk, eight PAR cans (with stands), Hysis tops and 18‑inch bass bins for a Toshiba laptop, a small keyboard and portable wheeled PA — not because it was necessarily better, but because this five‑foot‑two female was physically breaking from carrying too much kit and not having a roadie! I could also now own a nice, feminine car instead of a bright yellow ex-British Rail DAF van (although I do still miss blasting down the M1 in that van, affectionately known as Big Bird). Having always been a fan of minimalism in my home, and with the live rig shrinking, my studio started to go the same way. At first I adopted Cakewalk and could record

a small number of audio tracks into my computer. Later I switched to laptops and Logic. Fast forward to today and not only do I have total locational independence but I also have a Rhodes, Moog, Sequential, Oberheim and symphonic orchestra tucked into the laptop bag on my shoulder. In recent years, plug‑ins have become cone-rippingly good, and with a top pair of studio headphones and a small controller I can work from absolutely anywhere. If I need to upload the end result to share with session musicians or remixers then I just plug in my phone and use the hotspot setting, or find a coffee shop with speedy WiFi and a good selection of biscuits.

“Removing options definitely speeds up the workflow, and if you’re not feeling inspired on any particular day, you can easily take your studio to the beach.” 154

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When it comes to performing smaller local gigs, I can hop onto public transport with a five‑octave keyboard that weighs less than 6kg, a lightweight column keyboard stand and a Bose S1 Pro, with all of the sounds coming from either standalone versions of my favourite plug‑ins, or from presets set up in Ableton or MainStage. If needed I can also drop an interface into my bag (often the AG06) for submixing into a bigger PA system at the venue. Removing options definitely speeds up the workflow, and if you’re not feeling inspired on any particular day, you can easily take your studio to the beach. While it’s true that you can miss the tactile joy of real knobs and faders, I just love the flexibility that working in the box offers, and besides, it’s easy to take a portable studio into a professional studio, so I still get to play with plenty of fun toys.

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Sound On Sound [USA] October 2025 edition