Sound On Sound [USA] October 2025 edition

1985 — 2025

THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

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

CRANBORNE AUDIO

BRICK LANE One module, seven compressors!

KPop Demon Hunters

The epic production behind the hits

Expressive E Soliste

Musically modelled strings

Apogee Symphony Studio Affordable entry to immersive monitoring

SSL Revival 4000

E-series channel in a 1U rack

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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.

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October 2025 / www.soundonsound.com

3

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’.

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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

74

USB Audio Interface

70

Braingasm Ocula

14

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

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

16

Capacitor Microphone

91

Omnitone Rhythmi & Beatsi Eurorack Module

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

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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

T

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!

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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

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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

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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

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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

Q

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

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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.

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ON TE ST

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

T

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

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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

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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

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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

A

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

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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

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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

T

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

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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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Soundtrack module useful from day one.

It generates three layers: dialogue, music and effects. As with all the Unmix modules, results will vary depending upon the original source material but with the few video clips I experimented with, I was impressed. Yes, in a busy action movie soundtrack, you might find some material gets directed into the wrong layer and requires a little manual editing, but obvious use cases include restoration of old film/video soundtracks, or providing alternate audio mixes with louder dialogue.

The DePlosive module is a useful addition. It does exactly what you would expect and does it very effectively. However, even more impressive is the new Voice Enhance module. Steinberg suggest that this can correct for clipping, compression artefacts, noise and excessive room ambience. While removing the unwanted elements of the sound is perhaps nothing new (although it is done very effectively here), what’s more remarkable is how well the module also restores the overall frequency balance of a voice. This ‘reconstruction’ element will undoubtedly be useful in audio restoration but, equally, might be useful for content creators looking just to sweeten their dialogue recordings if they know their recording environment is sub-optimal.

Go With The Flow It’s not all AI-based news, though, as SLP12 also brings useful operational improvements for various editing tasks. An obvious highlight here is the slicker process for cutting material from one layer and pasting it into another layer of your choice.

It makes a non-destructive tidy up from a challenging unmixing process much faster.

This is enhanced further by various improvements to the selection processes.

For example, it is now easier to change, edit and refine a selection prior to executing any editing process. In addition, the assisted processes for making selections based upon frequency, harmonics or transients have all been improved. The latter is particularly useful when working with drum-based layers. For example, you might select the transients only of a snare or kick layer, move these to a separate layer, then adjust the relative

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levels of the transient and sustained portions of the sound to change the dynamics or punchiness of the drum.

Not so many years ago, these sorts of spectral editing tasks SLP12 also makes manual editing workflows easier through refinements to features used to be such as the frequency, harmonics and (as shown here) transient selection tools. incredibly challenging and, while Steinberg have made amazing repetitive tasks knowing that processed strides with all SLP’s AI-based processing output files will get suitably organised while tools since they acquired the software in you go off and do something else (like get 2019, it’s also true that they have made a good night’s sleep!). rapid improvements to the manual editing Conclusions experience. This is no longer an ‘experts only’ editing environment and it’s great I know, it seems like the proverbial to see these sorts of accessibility barriers five minutes since SOS last reviewed continue to diminish.

SpectraLayers Pro (actually, it was the October 2024 issue). I can therefore And There’s More understand if some occasional users are frustrated by the rapid paid upgrade There is a whole swathe of other cycle and choose to move up at improvements contained within SLP12. a slower own pace.

These include refinements within the ARA However, if SLP is part of your daily support, an improved History panel, native workflow in a professional audio editing support of Windows ARM64 hardware and environment, the improvements that the display of a time estimate for completion SPL12 brings will be very welcome, when you trigger a module (so you can plan particularly if your primary use for the a suitable hot beverage). software utilises its various AI-based There are a couple of obvious modules. In a commercial environment, highlights here also. First, a new if those improvements mean even one Phase Rotation module allows significant job gets through the quality bar you to easily correct asymmetrical when it might not have done with SLP11, waveforms which can, in turn, then the upgrade will probably have paid make it easier to handle their for itself. And, with workflow refinements compression or overall volume. elsewhere, things will also get done Second, the Batch Processing faster and more efficiently. options have been refined, Personally, I can’t wait to see just allowing you more control over how much further Steinberg can push all the folder organisation and these boundaries. Same time next year filenames of any audio outputs for SLP13? Quite possibly... watch this generated by your sequenced space. Until then, as a suite of tools for processing. For those requiring unmixing, audio restoration and audio to run processes on multiple files enhancement work, I think Steinberg (for example, in post-production have set the standard for others to match. when passing multiple dialogue SpectraLayers Pro 12 is both technically files through the same noise impressive and — for a spectral editing reduction or enhancement platform — remarkably straightforward to processes), this makes it much use. It’s a compelling combination. easier to ‘set and forget’ some Many of SLP12’s existing modules have been improved, and new ones have also been added.

$ $349.99. Upgrade pricing available.

W www.steinberg.net

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hile Valeton might not have quite the same profile as the likes of Line 6, Fractal, Neural DSP or Kemper, their GP series has been quietly flying slightly below the radar, and has now developed into a very impressive line of guitar amp modellers, profilers and multi-effects devices that punch well above their price. The latest in this line is the diminutive GP-5: while it comes in a tiny mini-pedal format, it has a frankly amazing feature list, and a very modest price tag too. That list includes amp and cab modelling, the ability to use amp-plus-cab captures, effects modelling, a customisable signal chain, a rather nice collection of genre-based drum loops, and a mobile app (for both iOS and Android) for easy editing and control over all these features. But is the idea of squeezing an entire guitar (or bass) rig into a pocket-sized pedal like this too good to be true?

Time to plug in and find out...

Guitar Rig TARDIS The GP-5 offers a standard quarter-inch jack instrument input and a multi-function mono/stereo output, again on quarter-inch jack This can be used to feed other pedals or an amp, or to provide a stereo output for a mixer/PA or even for headphones.

A well‑written PDF manual does an excellent job of explaining the various configuration options, and these would cover almost any eventuality, from headphone practice through to various live-performance scenarios. The back panel has both a standard 9V power inlet and a USB-C port, and the latter facilitates firmware and file import options, as well as serving as a basic audio interface for recording.

On the top panel, you get a dedicated volume knob along with a multi-function push-turn Parameter encoder. This allows you to switch patches, edit some settings and turn the wireless Bluetooth connectivity on (more on this shortly). The tiny LCD screen shows patch names and numbers, indicates the Bluetooth status, and, if you long-press the single footswitch, provides access to a simple but effective tuner. The footswitch itself can be used in three different modes: to step forwards through the 100 patch slots; turn effects on/off; or as a global bypass. The provision of only a single footswitch here is understandable given the compact format, but if I had one gripe it’s that there’s no mode that simply lets you toggle back and forth between two adjacent patches. This would enable easy switching between, say, a rhythm and lead tone for a specific song, for example. Something for a future firmware update? Having said that, there is a Song List mode, so if your gig set list is pre-organised you could sequence patches within the GP-5 and simply step through that list.

An App For That For patch editing, the dedicated mobile app (which is free to download) is essential. It connects easily to the GP-5 over Bluetooth and, as shown in the

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The Valeton GP-5: an entire guitar rig that fits in your pocket.

Valeton GP-5 Multi-effects Pedal

Can this pocket-sized, pocket-money guitar rig really deliver the goods? You might be surprised... screenshot, its very intuitive UI allows you to select patches and edit all the elements in your signal chain. This includes the various effects slots: NR (noise reduction); Pre (preamp/boost style pedals); Dst (overdrive and distortion); EQ; Mod (various chorus, phaser and flanger options); Dly and Rvb. Within any patch, in each slot, you can pick one of the available pedal options; again, these are properly documented in the PDF manual. You can also reorder the signal chain by simply dragging and dropping within the GUI.

The Amp and Cab modules provide modelling-based options and cover lots of classic amp/cab targets, including clean to high-gain choices as well as a number of options for bass. As with the effects, each amp model offers a small number of editable parameters such as volume, gain and EQ. The two modules can be switched on/off independently so, for example, if the GP-5 signal were to be sent through an actual guitar cab, you could disable the Cab module. The Cab module also supports IR loading, by the way, with 20 slots available for user files.

The N-S module provides amp/ cab profile capabilities using Valeton’s SnapTone technology that lets you import captures from the open-source Neural Amp Modeler

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software. The GP-5 ships with a decent selection of 50 factory presets already, but with 80 slots available in total, you can load more, for example from the Tone3000 website, and when connected over USB to a computer you can use Valeton’s Mac/Windows software to manage this process. Note that if you enable the N-S module, the Amp and Cab modules are automatically bypassed. I’m not sure if this is a processing power limitation but, even if not, it’s a common-sense feature to avoid the potential sonic chaos of ‘amp/cab into amp/cab’.

Just for good measure, the app also offers a pretty impressive selection of tempo-adjustable genre-based drum loops, and if you link your GP-5 to your phone as a Bluetooth audio device, you can hear the drums over the GP-5’s output and blend them with your guitar signal.

Used with headphones, it makes for an excellent ‘silent’ practice platform.

Small Size, Big Sound The GP-5’s extensive feature list and compact cuteness are impressive but, wonderfully, they’re matched by some truly excellent guitar tones.

As we now expect from almost any modeller/ capture device, you have plenty of choices, spanning a wide range of guitar tones and musical styles.

There are amps inspired by classic clean platforms by Fender, crunchy options for a selection of Vox or classic Marshall-style tones, and a good selection of amp styles aimed at higher-gain rock and metal (think modern Marshall, Mesa, Peavey and others). In short, there’s something for everyone.

And if you want to know which real amp, cab or stompbox any of the GP-5 options are nodding at, it’s all covered in the manual.

You don’t get quite as many parameters to tweak as you might on larger and more expensive units, but there’s still plenty of scope to fine-tune the tone to meet your personal needs and preferences. The cleans

sound great to me, the high-gain sounds are suitably aggressive, and the in-between, edge-of-breakup/ gnarly crunch sounds very convincing. The GP-5 models also responded well to playing dynamics or volume roll-off. The bottom line is that the GP-5 sounds way better than its size or price tag suggest it should have any right to.

All of which means that the tones available from the GP-5 go well above the bar required for personal practice. It would make a great recording tool for those on a budget or who need an ultra‑portable solution. It would also make a great option for compact band rehearsals or as a backup rig for gigging musicians just in case that vintage valve amp goes ‘ppfff’ 10 minutes before showtime. No, it might not quite have the flexibility of a full amp, cab and pedalboard rig, but it would certainly let you access some very usable sounds that get the job done.

If you pay attention to the world of amp modelling and capture technology, then you’ll be familiar with IKM’s Tonex One pedal, which Paul White reviewed back in September 2024 (https:// sosm.ag/ik-tonex-one). While that focuses on capture-based technology, to me it’s the obvious ‘name’ competition for the GP-5, and during the review process I had access to a Tonex One. Sonically, I have to say that both really are very impressive, and while there’s no clear winner or loser in terms of sound, there are some underlying feature differences (as well as the price) to consider that could sway a purchase decision. But I reckon both are excellent for the asking price.

Rig In A Box Coincidently, Line 6 happened to announce their new flagship Helix Stadium amp modeller/capture multi-effects device at about the same time as the GP-5 became available. Undoubtedly, we’ll review the Helix Stadium when it’s available and I’ve little doubt that it will be absolutely fabulous (I’m a 10-year user of the original Helix Floor; it’s the best bit of guitar tech I’ve ever purchased). But while this tiny sub-$80 pedal can’t compete directly with that sort of multi-thousand-dollar flagship unit, amazingly, it does manage to perform a similar role, and with considerable style. Even as an IR loader alone, its price would be keen, but it does a heck of a lot more than that — if you need a compact guitar rig for personal practice, band rehearsals and the occasional gig, then the Valeton GP-5 is an absolute steal. I’m sure it will sell by the bucket load, and it deserves to.

summary Valeton have absolutely knocked it out the park with the GP-5. It’s a great‑sounding practice tool or a full live performance rig, all in a pocket-sized pedal for a ridiculously affordable price.

The free-to-download mobile app is an essential part of the overall GP-5 package.

It includes a good selection of drum loops for personal practice, and these can be streamed to the GP-5 via Bluetooth.

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$ $79.99 W www.valeton.net

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In-depth pro comments

ON TE ST

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

F

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

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ON TE ST

NOR DI C A U DIO L A B S N U - 24 K

that has neither a valve nor an output transformer, but uses a proprietary solid-state circuit called FloFET that is claimed to sound “smooth and flattering”.

Hey, Good Lookin’ The NU-24K ships in a small Peli-type case, with a Rycote InVision shockmount.

To say that this is an attractive mic to look at is a serious understatement. The proportions are spot-on, build quality is absolutely impeccable, the understated black finish with gold legending is classy without being tacky, and the open headbasket with its super-fine gauze is a work of art. The aforementioned D100K capsule is clearly visible through this gauze, but does not give up its mysteries easily. In some large-diaphragm capsules, the membrane is transparent enough that you can see the pattern of holes in the backplate, but not so here.

As I’ve already mentioned, the NU-24K is a transformerless, solid-state mic, and as such requires standard 48V phantom power, from which it draws a modest 3.5mA. The specifications make for interesting reading, if you’re so inclined. The polar pattern is described as “cardioid to supercardioid (frequency dependent)” — which, to be fair, is true of most large-diaphragm capacitor mics — and output impedance is an unusually low 22Ω. Self-noise is quoted at 17.6dBA “with 6dB/oct roll-off towards 2kHz”, which suggests that the FloFET circuit somehow shapes an otherwise fairly high noise level to be less obnoxious in practice. In my experience, noise is most troublesome when it’s higher up the frequency spectrum, so this isn’t a bad idea, and subjectively, noise is not an issue with the NU-24K. Sensitivity is a fairly standard 18mV/Pa. The NU-24K is marketed primarily as a vocal mic. This sometimes indicates that the manufacturer has deliberately attenuated the low end, but not here: the -3dB points are at 17Hz and 27kHz, and the frequency response chart shows quite a large bump centred around 35Hz, which is deeper than even the most Russian of basses can produce. From 100Hz or so upwards, the response is broadly flat to about

3kHz, whereupon a presence lift rises steadily, plateauing between 10 and 20 kHz and extending right up to that 27kHz mark. On paper, this make the NU-24K look like a very bright mic — or, given its strong bass, like a mic with a ‘scooped’ frequency response.

Off The Charts Any engineer who’s been around the block will tell you that microphone frequency response charts, at best, tell a small part of the story. A mic that employs a K67-style capsule without de-emphasis might measure somewhat like the NU-24K yet sound totally different, with an edgy character and an

component in any mic, and I’m convinced that Martin Kantola has come up with something truly special in the D100K. Nearly all capacitor mic capsules fall into one of very few design families, and the vast majority are based on the Neumann K47 or K67.

The D100K sounds nothing like any Neumann-derived capsule I’ve ever heard. By contrast, it does have a lot in common sonically with the fabled AKG ‘brass ring’ CK12, and of all the mics I put it up against, the NU-24K was most similar to my old C414 EB. In fact, it has all the strengths of that classic, but arguably sounds better. The 414 sometimes seems to add a touch of fur to the source, perhaps because of its relatively small output transformer; switch to the NU-24K and it’s like someone has taken a veil off the same mic. Turn again to a U87 and you’re left wondering why the Neumann sounds so nasal and honky. There were some sources, such as resonator guitar, where that quality was desirable, but in the majority of cases I preferred the NU-24K.

“Female vocals can be a challenge to record well, and whilst no mic is universally good for every singer, I’d go as far as to say that I don’t recall ever hearing a mic that works better on this source.”

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uncomfortable emphasis on sibilants.

That’s absolutely not the case here. That treble lift, or mid scoop if you prefer, manifests itself here as clarity rather than sharpness. And although we typically associate the quality of smoothness with a subdued high-frequency response, I think Nordic Audio Labs are absolutely correct to claim it for the NU-24K. I had the opportunity to try it on several different vocalists during the review period, and also tested it on instruments such as acoustic guitar.

Female vocals can be a challenge to record well, and whilst no mic is universally good for every singer, I’d go as far as to say that I don’t recall ever hearing a mic that works better on this source. It’s no slouch on male vocals either, and what particularly struck me in both cases was how natural its capture of the human voice is. It may well be emphasising the high end, or subtly attenuating the midrange, but what’s more important is what it’s not doing. This is a mic that doesn’t smear transients, or emphasise nasal resonances, or sound hard when the singer belts it out. Other large-diaphragm mics, even very good ones, often sound slightly artificial or woolly in comparison.

The unusually open headbasket probably plays a role here, but the capsule is the most important

A Bit Of All Right I’ve had the good fortune to review lots of new microphones for this magazine, including some very expensive boutique models. Many have impressed me, but few have impresed me as much as the NU-24K. This is a pretty special microphone, to the point of making me wonder if it’s time to retire some of my beloved vintage models. It’s clear, natural and smooth, but not at all sterile or boring.

I loved it on the human voice, liked it very much on most acoustic instruments, and several times wished for a pair so that I could try them out as drum overheads. The NU-24K has all the makings of a future classic, and belongs in the top drawer of any mic locker.

summary The NU-24K is beautifully made and looks stunning, but it’s the sound that will sell it.

This is just a superb vocal and instrument mic.

$ $1650 W www.onefsound.com W www.nordicaudiolabs.com

ON TE ST

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

PHIL WARD

A

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.

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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

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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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ON TE ST

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,

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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.

W www.apogeedigital.com

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

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We are proud to unveil the next milestone title in our collection of SDX expansions for Superior Drummer 3, the Real to Reel SDX by multi-awardwinning producer/mixer/engineer Kevin Killen (U2, Peter Gabriel, Elvis Costello).

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Noiseworks DynAssist

Vocal Pre-production Software With ARA2 compatibility, this clever software slots right into your DAW workflow and promises to take much of the pain and time out of getting your vocals mix ready.

M AT T H O U G H TO N

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bviously, a great vocal recording starts with a decent mic in front of a great performance. But once that performance is captured, there can be a lot of effort involved in improving the recording before it’s time to actually start mixing. Such prep work was traditionally done by an assistant but that’s only now the case for a select few in‑demand mix engineers — it allows them to mix quickly, and put their full focus on the more creative and musical side of mixing.

The rest of us must be our own assistants, but the tasks are the same: detailed clip-gain work to make sure the vocal part has a broadly consistent level and every word and syllable is audible, as well as cleaning up undesirable sounds that could interact unhelpfully with your processing chain — things like bumps of the mic stand, plosive pops, loud breaths, sibilance, any hum on amps and pedals,

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and tape hiss. Get all your ducks lined up at this stage, and you’ll find that it’s so much easier to bring out the magic in your recordings using your bread-and-butter mixing tools, such as compressors, EQs, saturation, reverbs and delays, along with the usual fader and pan moves.

It’s a similar (albeit slightly different) story when working with dialogue, where you’ll typically want to aim for a nice, consistent level, with every word and syllable clearly heard, whether it’s a whisper or an excited exclamation, and with the breathing, popping and essing and so on firmly and transparently under control — sometimes eliminated, sometimes just lowered, for a more natural feel. A key difference with dialogue is that recordings are often much longer, so a lot of these issues are necessarily addressed on the fly, along with your edits and fades.

There have always been tools that can take some of the sweat and tears out of this prep stage, but some can cause you

The default view, after the initial analysis has been performed, with the Ride tab open.

almost as much work as they save. Take breath control, for example: until recently, every real-time or offline breath control plug-in I’d tried was a case of ‘close but no cigar’, and while I might be able to correct any artefacts manually, doing that takes a lot of attention and time. That’s why

Noiseworks DynAssist $149 pros • Simple idea, brilliantly executed.

• A genuine labour-saver.

• AI used only for detection, not for the actual processing.

• User has complete control over the end result.

• One of the best breath controllers out there.

cons • Can’t protect manual edits if you need to adjust other settings.

summary A superb labour-saving plug-in, DynAssist can help you prep your vocals and dialogue for mixing in next to no time, and includes one of the best breath control facilities out there.

A company worth keeping an eye on.

I’ve tended to rely on a manual approach to breath control; it’s more effective, it’s quick to do for a 3-4 minute song, and with experience it can be done reasonably swiftly as you audition and edit dialogue.

DynAssist All of which preamble brings me to my first encounter with software developers Noiseworks, whose DynAssist software aims to perform much of this prep work for you, swiftly and, potentially, entirely automatically. In 2025, it almost goes without saying that DynAssist makes use of machine learning, but I feel it important to highlight that it only employs this ‘AI’ to detect the signal levels and identify different content in the signal; the actual processing it applies is conventional level automation, and the user has full DAW-automation-style control over the result.

DynAssist is a VST3/AU/AAX plug-in for 64-bit macOS and Windows systems, and for most DAWs the best approach is to use it as an ARA plug-in. In case you’re not yet familiar with ARA, this is an offline plug-in format that most DAWs, including Studio One, Reaper, Cubase and Pro Tools, now support, and it enables third-party software to function in much the same way as your DAW’s own offline tools such as pitch and time correction. There are other direct transfer systems for DAWs that don’t support ARA, though, and these are detailed in the PDF manual.

Different DAWs have slightly different ways of opening the ARA plug-in but generally it’s a case of right-clicking on a track or event and selecting the plug-in, or selecting a track/event and navigating your DAW’s menu to locate and load it.

Your audio will then open within DynAssist, typically at the bottom of the DAW’s GUI. Personally, I prefer to pop it out into its own floating window, as I can then resize to taste with a drag of the corner, and show/hide it as desired.

Within the DynAssist GUI are four main tabbed windows: Ride, Vox Gate, Breath and Sibilance, each providing control over a different function. At the bottom, whichever view you’ve selected, you’ll see a waveform with a node-based automation line overlaid, and switching tabs updates this view to highlight the relevant data.

A final fifth tab puts all the parameters in one place; obviously this looks busier, but it’s sometimes convenient to have direct access to everything from one screen.

At the top left is a toggle that lets you select between vocal and instrument modes. Essentially, instrument mode omits the breath control and sibilance tabs, and offers a different approach to gating, of which more below. On the other side, another toggle determines whether DynAssist operates in Track mode (whereby it analyses all clips on the DAW track on which you’ve inserted it) or Clip mode (it analyses only the currently selected clip). The latter can be really

helpful in the initial tweaking stages, since you can leave DynAssist’s automatic re-analysis feature switched on so you can audition the results quickly, without unhelpful pauses. There are also the usual global facilities, including bypass, undo/ redo, a preset management section, a link to open the manual, and a basic settings pop-up, where you can enable tooltips, change the default Track/Clip mode behaviour, and adjust the GUI scaling.

At the bottom of the GUI are a number of useful features, not least buttons to engage auto-analysis and for loop selection. Not only does the latter sync with you DAW to facilitate loop selection for playback, but again, if auto-analyse is on, it will only analyse that section, making it much quicker to audition different settings.

It’s a good idea, and especially so for long dialogue recordings, as you can refine settings on a shorter selection before applying them to the full part, which will likely be very similar. The auto-analysis can also be switched off, by the way — again, this allows you to make changes more quickly, since DynAssist won’t have to recalculate its settings after every tweak, though obviously you won’t hear the result until the analysis has been performed.

A pen button lets you manually adjust the automation points (that are created automatically according to the settings in the main tabs) with the mouse cursor.

As well as directly editing individual points in this way, you can make marquee

The Vox Gate uses machine learning to identify non-vocal/dialogue areas in a file, and lets you decide how much they’re attenuated.

www.soundonsound.com / October 2025

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selections and pull all the contained points up/down simultaneously, which is useful.

You can navigate around the waveform using the mouse wheel to zoom in/out, with Cmd (Mac) or the Windows key acting as a modifier to scroll horizontally. It was a breeze with my MacBook Pro’s trackpad too. All in all, it’s intuitive and works well — very much like editing in your DAW’s own automation lanes.

Ticket To Ride Starting in Ride view, the aim here is to level the dynamics of your vocal part.

Conceptually, it’s not dissimilar to using two-stage compression, with a parallel compressor bringing up lower level details, and then a regular one pulling down the louder parts. A big difference, of course, is that it’s an offline process, not a real-time one, and that has a couple of potential advantages.

For one thing, DynAssist’s analysis means it ‘already knows’ where the signal level rises and falls, so it can be more accurate and transparent, and less ‘reactive’, side-stepping common issues such as pumping or overreactions to brief peaks.

Another key difference is that its output is editable level automation, so it’s super‑easy to control any issues thrown up by that ‘compression’. And since DynAssist’s level changes are all applied at the start of the signal chain, before your processors and effects, it’s arguably more akin to an automatic clip‑gain envelope process than either compression or fader riding. You don’t really need to think about ‘make‑up gain’ in the way you would with a compressor either, since DynAssist lets you specify a target LUFS-I loudness by dragging a tab on the left to the desired

value, and to my mind, there are a couple of benefits to this approach. For example, you can assess any tweaks you make safe in the knowledge that loudness isn’t skewing your perception. Also, you can make the signal levels predictable for any further processing. For instance, you might have a go-to chain for processing vocals, and within it various threshold-dependent tools such as compressors. You may even have this feeding into a similarly structured bus‑processing chain, with thresholds calibrated to specific levels. If you already know how loud your source must be to hit those chains in their sweet spot, you can make it so — increasing your chances of a decent ‘faders up’ mix, and thus making mixing quicker and easier.

It’s also worth noting that although it must take the time to perform an analysis when you update settings, DynAssist introduces no latency on playback in the way an insert plug-in might. So if you’re one of those types who tends to add more virtual instrument parts at all stages of your production, DynAssist won’t get in the way.

There are just a handful of user controls, and I don’t think you really need more.

A Speed slider determines how detailed the level correction will be, with fewer automation points being created if you drag the slider over to the right, and very detailed automation when it’s fully left.

Beneath, a Smooth slider effectively scales the range of gain/cut being applied, so you can decide just how assertive your vocal riding will be, while Max and Min parameters set the threshold boundaries within which the vocal riding will be performed; these can be linked so that both are adjusted at once, or unlinked, as you prefer. Finally, an Avoid Clipping

Sibilance is indicated in light grey on the waveform. In some places, other processes have applied gain such that the overall automation curve shows a boost (green) even where one of two esses (light grey)is being attenuated.

toggle prevents the level of the part being boosted above 0dB true peak when engaged.

I should note that performing the analysis function — whether automatic or not — will override any manual automation edits that you’ve made. And because whenever you make a change to any parameter, in any tab, DynAssist needs to re‑analyse the audio, you must for now either leave your manual edits until last, or render them before any re‑analysis.

Helpfully, you do get a pop up warning before losing any changes. Better still, when I suggested to Noiseworks that it might be good to allow the user to ‘protect’ areas from analysis, in much the same way I can in Synchro Arts’ Revoice Pro (another ARA plug-in), I was told they’d had the same thought, and that this is already very much in their development plans.

Gate Expectations

This might just be one of the most effective automatic breath-control tools out there!

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The next tab lets you engage or bypass the Vocal Gate and, courtesy of ‘AI’, this is a content-aware process: it knows which parts of the recording are vocals and which aren’t, and you don’t need to set a threshold. As I mentioned already in passing, Instrument mode works slightly differently: this does give you a conventional threshold control, the idea being that you can manually set up the gating to suit any other instrument/sound source you might want to process.

Essentially, this gate automatically detects any ‘silence’ between vocal phrases and attenuates it, to prevent

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NOISEWORKS DYNASSIST

As you become accustomed to using DynAssist, there’s a lot to be said for working in the All tab, where you have access to every parameter at once, but without the graphical control functions found in the other tabs.

unwanted noise coming through that might otherwise be brought up by any compression and limiting downstream in your signal chain. On the left, there’s a graphical control resembling automation points, and this lets you set the gate close and open speeds, and the amount of attenuation (ie. the ‘floor’). At minus infinity, it gives you silence when the gate is closed, while other values attenuate the noise without fully muting it; useful for continuity where a noise (like tape hiss) is audible during the vocal and not just in between phrases.

To the right are a couple of directly editable parameters. Min Noise Length prevents the gate kicking in overzealously during short breaks between words. This control is located away from the graphical one, but the latter is updated to indicate your changes. Loudness cutoff lets you prevent quieter sounds such as spill from making it through the gate, and this seems to work pretty well.

Take My Breath Away The next two tabs are almost identical to each other, but one targets breaths and the other sibilance. Again, DynAssist’s analysis identifies these sounds for you and changes the level according the parameters you specify in the relevant tab.

And I have to say it largely does a grand job of the detection, without you having to do anything...

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A Relative Threshold control determines how loud a breath/ess must be to be affected by your settings. At minus infinity, all instances will be targeted, and typically this is how I wanted things set up for breaths. That said, I can conceive of some situations where you might want to leave quieter breaths alone, and target only louder ones. You can also specify by how much the breaths/esses should be attenuated, again from nothing to minus infinity. And, lastly, there’s a Transition Speed parameter, broadly equivalent to a combined attack/release control.

As I mentioned, the final tab called All shows all of the other tabs’ settings as numeric/text parameters that you can click and drag to increase/decrease. While I did find the sliders and graphics of the other tabs useful for learning how to use this tool, as I grew more familiar with it, I spent more time here.

User Experience In general, DynAssist works superbly.

I tested it as an ARA plug-in in Reaper on an M1 MacBook Pro running macOS 12.4. The tests involved several male and female vocal tracks and a couple of long dialogue parts for a podcast I was producing at the time. The vocals included some recordings of my own voice where I deliberately sang more dynamically (erratically?) than I normally would, just to see how DynAssist could cope.

On the sung parts, the analysis was pretty speedy, but naturally it took a good while longer to perform the initial analysis for the 50-minute-long dialogue parts. It’s worth the short wait, though, as generally you’re going to need to do less by way of tweaking with simple dialogue than with singing. However, after the initial analysis, you may want to tun off auto-analyse while making adjustments to the settings, or to limit the analysis to a short loop so that you can hear the results without delay. In all cases, bar none, the initial levelling was truly impressive, and I found it super-easy to hone the results by juggling the Speed and Smooth parameters to taste, and only occasionally feeling a need to set minimum/maximum gain limits.

Much to my pleasant surprise the breath control was, for the most part, absolutely spot on, with breath onsets and endings detected beautifully, and none of the unacceptable artefacts I’ve endured with every other tool that’s got my hopes up with promises of perfect de-breathing!

There were just a few instances in a full track where the very start of the breath wasn’t identified, but these were few and far between, and as each breath is obviously highlighted you can quickly drag a single automation point left to solve that. I loved that I was able to turn all the breaths up/down to taste — with a vocal processing chain inserted after DynAssist, where a singer was using breaths

rhythmically, I could even judge how they interacted with the overall groove.

Though it was effective, I found the visual feedback of the de-essing tab a little more confusing. While the detected esses are highlighted, esses are briefer than breaths, and depending on your zoom settings they can be harder to see. Also, it’s not always immediately obvious what the de-ess contribution is to the overall automation — it’s possible to have the audio being turned up for other reasons, so that even when de-essing the net effect is indicated in green (for gain) on the automation curve. I’d quite like the option of dedicated lines to indicate the de-essing and de-breathing gain reduction — and again, when I put that to Noiseworks, they said I was not the first to suggest that, and it’s in their plans. The audible results are good, by the way.

On The Level When you consider that, for all the clever machine-learning-driven content recognition, all DynAssist is actually doing to your audio signal is turning it up and

down, it really is an incredibly impressive achievement. But it’s a very practical one too, which could well save you a lot of effort and frustration — and time. It can be your clip gain, your compressor, a de-esser and, I have to say, perhaps the most effective de-breather I’ve yet encountered, all in one. I will be using it for my dialogue and music work going forward, and reckon it will save me lots of time without obvious compromise — I love that, unlike many AI tools, it does what it does in a way that leaves me free to make manual adjustments, giving me complete control over the end result.

Of course, DynAssist doesn’t do everything that I might need to do to prep a vocal or dialogue part: there’s no de-noising, and you can’t apply temporary filters to counter plosive pops, for example. Also, like all ARA plug-in, it must be placed at the start of the chain, and you can’t instantiate other ARA plug-ins simultaneously on the same track. So if you wish to use other ARA plug-ins such as Melodyne for pitch-correction or iZotope RX to address some of the jobs

listed at the top of this paragraph, then you’re going to have to spend a little time planning the most efficient workflow. For example, if I were to want to use all three of those apps, I might look to use RX first for spectral de-noising, perhaps plosive control and a spot of mouth de-clicking, before rendering that result and loading DynAssist, and I might do pitch correction after that.

But in the grand scheme these are tiny niggles. And again, without giving away too many details, when I discussed this side of things with Noiseworks, I was told that they’re very conscious of these limitations of ARA, and that they have ambitious plans to address them in future versions. Already, though, I’ve found DynAssist most impressive. I’d heartily recommend that anyone working regularly with dialogue or vocals at least tries the demo. For me, it’s probably worth the price of entry for the breath control alone — and that’s not even the main selling point! $ $149.

W https://noiseworksaudio.com

www.soundonsound.com / October 2025

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EXPRESSIVE E SOLISTE

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

ROBIN BIGWOOD

W

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.

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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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EXPRESSIVE E SOLISTE

The busy bowing parameters page, a click away from the main interface. There is a lot of flexibility here, to change both the fixed behaviour of the physical model and the way it responds to control inputs from your controller.

bends also sound complex and natural.

Strike faster, or play with a detached touch, and attacks become crisper. The musical effect often tends towards sweet and impassioned, and it’s easy to create lines that surge and recede in much the same way that a real violinist achieves. Happily, not a great deal is given away when using a basic non-MPE MIDI controller. With those, by default, key-presses are responsible for a velocity-controlled attack phase and the mod-wheel dials in the intensity of the sustained part, and it can sound really good.

Back & Forth I expect the vast majority of users will end up using (whether they know it or not) Soliste’s Sustain bowing mode, most of the time. That simply means that while you hold a note, sound continues indefinitely, subject to real-time control of other parameters, as if your virtual string player was keeping the bow in constant motion. But there are other modes including a potentially very convincing held tremolo, and two ‘back and forth’ options which, on an Osmose (for example), can give rise to virtual bowstrokes on both the press and release of a key, with optional sustain on the downstroke phase. With no sustain an Osmose key effectively becomes a bow about 2cm long, which isn’t of much use other than for very fast passages or rapid repeated notes. However, movement of your modulation wheel can be brought into play to simulate longer bow movement. And it should be here that Expressive E’s Touché controller really comes into its own. I wasn’t able to test this combo, but everything suggests that stroking Touché’s larger surface will simulate yet longer individual bow strokes.

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One thing Soliste doesn’t support is double stopping: playing two notes at once.

Attempt it and Soliste just plumps for one note or another, monosynth style. I can see there could be challenges in implementing it while also supporting overlapping legato note inputs, and it’s probably impossible alongside the Osmose pressure glide feature. But to have no way of achieving it at all is an unfortunate omission.

So, an impressive range of control is available by varying keyboard touch, but additional playing styles — articulations, in normal sample library parlance — are very much available, accessed via a Presets sidebar. Spiccato, martelé, tremolo, pizzicato and others can be selected with the mouse, and many have a secondary Character option. They include general adjectival descriptions —

light, intense, and so on — or something more technical, such as a ‘harmonics’ option for the already glassy-sounding sul ponticello, or a gloriously ghostly ‘flautando’ for sul tasto. These variations all derive from settings in the bow page, including a normally hidden set of five advanced parameters that go deep into the underlying physical model, affecting bow position (relative to the bridge), pressure and noise. Explore extremes in those and Soliste can produce some quite striking, novel sounds, which can be saved as user presets. A preset, by the way, recalls just the entirety of these bow page parameters, not any other settings.

Further sound-shaping flexibility can come from real-time MIDI control: nearly all parameters (including the effects) can be mapped to up to six incoming MIDI CCs, pitch-bend or aftertouch, in the MIDI sidebar. A nifty combo of MIDI Learn for the message type, and click-to-assign for the destination, makes this a breeze to set up.

Performance Practice I’ve mentioned some aspects of the playing experience already. But what of the sound and character of Soliste? Well, because there’s no functional demo version of the plug-ins available, I would encourage prospective buyers to first listen carefully to Expressive E’s YouTube showcases. They’ll let you form an opinion much more rapidly than me waxing lyrical for paragraphs on end. However, I will offer a few insights.

I personally don’t think many listeners would mistake Soliste, in an exposed solo say, for a recording of a real instrument.

Actually I ought to nuance that: you might, for individual notes or some simple accompaniment figures, probably more

A Soliste preset is comparable to a sample library’s ‘articulation’, but currently there’s no way to select them on the fly with a key-press or other MIDI message.

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so for the bass and cello than the upper strings, and certainly some of the specialist ‘extended’ techniques are surprisingly convincing. But to my ear straightforward expressive melodic playing exhibits a certain quite strong synthetic quality: not unpleasant at all, but not naturalistic either. Attempt the main theme from Mendelssohn’s violin concerto or perhaps John Williams’ Schindler’s List on the Soliste violin and quite honestly you’re not fooling anyone. The reasons are hard to pinpoint, but it’s partly tone quality (which to my ear becomes actually quite flute- or tube-like in some tessitura) and also the quality of legato pitch transitions (which can have a little whiff of synth-y portamento even when the main portamento feature is turned off).

But here’s the question: is Soliste supposed to be a true-to-life replacement for real string instruments? Expressive E don’t claim so, and in the plug-ins themselves the funky instrument depictions also suggest not. If that’s the case, we can judge these instruments on different terms. You might say they form a class of instruments all their own: not synths, and not sample libraries either. In that I was

Effects Lurking in another side panel in the interface is a trio of effects processors.

The three-band parametric EQ offers boosts and cuts up to 10dB. It works just fine, but the labelling isn’t quite finished: the frequency range of all bands show as 40-4000 Hz. A Timbre knob adds a sort of edgy brightness, defaulting to differing low values across the different instruments. And there’s a very good, simple but smooth-sounding convolution reverb with impulse responses named seemingly after various UK venues, in York, Elveden and St Albans, amongst a few others.

continually reminded of the Osmose’s onboard synth, which has its own sophisticated physical modelling abilities.

It too is capable of a (different) range of quasi-acoustic sounds with remarkable harmonic complexity and expressive nuance, but which almost always remain outside the realm of naturalistic reality: to great effect. I was also minded of Yamaha’s 1990s VL-series synths, which offered their own brand of fascinating natural/synthetic hybrids, with expressive control inputs.

For me the proof of the pudding was that the Soliste sounds opened up new possibilities all their own. I absolutely loved playing them, especially with an Osmose. The experience is absorbing and addictive, the sound beguiling and often ravishingly beautiful, and lovely things keep happening that open up creative pathways.

These instruments feel alive, and I started dreaming up ways of incorporating them into song arrangements, even alongside traditional sampled strings.

Comparing Soliste to the best solo strings sample libraries is instructive. Yes, with painstaking work those can generate something that actually might fool most of the people most of the time: they’re using recordings of real instruments after all. But none I’m aware of are remotely as musically involving under the fingers, and they require much more learning and forward planning to achieve good results.

This broadly corresponds to the way I’ve known other samples versus modelling contests play out: sample libraries often triumph in sheer naturalistic accuracy, but acoustic modellers offer a more immersive musical experience. Samplers are work, modellers are play, maybe?

Conclusion

A flexible MIDI control scheme lets you extend the scope of Soliste’s response beyond what’s possible to generate from your keys alone.

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Looking at the broader implementation of the Soliste plug-ins there are few things to be aware of. The elephant in the room, for me, is the lack of sample library-like keyswitchable articulations. Soliste presets offer a range of articulations, but you can only select them with the mouse. So if you

want to perform a string part that has a sustained phrase then a spiccato section, and a couple of pizzicato notes to finish, your best option would probably be to create three instances of the plug-in set up accordingly, and do it in bits, or get painstakingly creative with your DAW’s parameter automation. MIDI CC control can take charge of bowing mode, and that counts for a lot, but it’s far from a full solution. I find this omission a great shame, and even if it doesn’t pop the expressive balloon it definitely deflates it a bit. There would be room for a bank of keyswitches too, as the lower pitch limit for each instrument corresponds exactly to its real-life counterpart: the violin, for example, bottoms out at G2.

Other things I was less keen on include the strange partial nature of the preset system, only recalling a subset of bowing-related parameter values. I’d have much preferred presets to include vibrato, portamento, effect and other settings. Oh, and at the time of writing there is still no manual of any kind: not a PDF, nor even a web page. Some website walkthrough videos are useful, covering some features in depth, but specific information relating to setup in different DAWs, non-MPE use, or incorporating a Touché, is currently almost non-existent.

Still, there’s a heck of a lot to like in Soliste, and way beyond a formal review phase I find myself continually drawn back to it. The asking price — $299 for the whole suite, or $99 for each instrument — is not inconsiderable, but it’s possible to make striking solo performances and arrangements with it that have a sound quality and expressivity like little else. $ $299 W www.expressivee.com

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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

I

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

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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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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!

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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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Control & Operation The Brick Lane’s control panel is styled like other Cranborne 500-series modules, with a quintet of large, gently detented knobs arrayed vertically to the left of the panel, and various displays, switches and a smaller knob to the right.

The input level is set with the top rotary control over a ±20dB range, while the threshold knob underneath is scaled simply from Min to Max. A large dual bar-graph meter to the right complements these primary level controls by showing input level (post input trim) on the left meter and gain reduction (GR) on the right. The input meter spans -24 to +24 dBu, with green LEDs up to +15dBu, then yellows at +18 and +21 dBu, and red at

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CRANBORNE AUDIO BRICK L ANE 500

+24dBu. The GR meter descends from the top, covering the range from 0.5 to 15 dB, with the finest resolution in the upper half — the middle of the scale indicates 5dB of gain reduction.

Below the threshold control are attack and release knobs, both scaled from slow to fast — the right approach to the labelling, given that their precise values are highly dependent on the chosen operating mode. The last large knob, at the bottom of the panel, adjusts the output level, again spanning a ±20dB range.

The Brick Lane’s compression is activated using a toggle in the bottom-right corner, and an adjacent LED lights blue when it’s active or white if being controlled by OptoSync (enabled by another toggle just above).

A third, centre-sprung momentary toggle switch above that one controls the Brick Lane’s clever features, responding differently to long and short presses left or right. For example, a short press left cycles through four side-chain high-pass filter options (off, 60, 100 or 200 Hz, with LED indicators just below) to reduce the impact of strong bass on the compression process.

A long press to the left accesses seven different stereo-linking modes, indicated by the number of illuminated GR LEDs. The side-chains can operate

ALTERNATIVES The closest Brick Lane alternative I’m aware of is the Great River PWM-01 500-series compressor module, but it’s both more expensive and less versatile than the Brick Lane. Rackmounting PWM compressors are available from Crane Song (the STC-8 and Trakker) and ART (Dual-Limiter). George Massenburg Labs made the excellent Model 8900 PWM compressor, but I believe that’s currently out of production. There are also a few niche and DIY homages to the original PYE 4060 design.

independently (unlinked), the loudest channel setting the GR (with two low-pass filter options), or with the two channels summed to determine the GR (again, with two low-pass filter options). The filter options are 5.4 or 2.7 kHz and help to smooth gain-reduction changes in the presence of transients. Another long press left exits this configuration mode.

Flicking the toggle to the right cycles through seven different compressor modes, with LED indications above the switch. These modes are labelled Velvet, Float, Smash, Tame, Glue and Polish — this last with white and blue options (see box). Selecting a mode recalls the appropriate parameters for things like feedback/feed-forward topology, RMS or peak-sensing

side-chain, initial compression ratio, knee shape, side-chain emphasis, and saturation character (amongst other things), as well as optimised ranges for the attack and release controls.

The initial ratio value increases automatically in most modes as the threshold is lowered but it can also be adjusted manually, if desired, via a long toggle press to the right. When this function is accessed the GR meter turns blue, and further short flicks to the right alter the ratio value as indicated on different GR LEDs. Another long press to the right exits this function.

The small rotary control labelled Stress, just below the bar-graph meter, adjusts the amount of multiband saturation. Fully anticlockwise turns the function off, and rotating clockwise increases the saturation effect for the selected mode. For example, in the Velvet mode the Stress feature introduces a creamy valve-style saturation with some low-end thickness, whereas in Float mode Stress smooths out the high end and softens transients. Adjusting Stress in Smash mode adds a bright, aggressive and gritty character, while in Glue mode Stress increases the overall mix thickness. Tame mode is pure ultra-clean PWM compression and Stress is quite subtle and restrained. In the two Polish modes (these are look-ahead

Construction & Technical Performance The Brick Lane 500 is constructed to a very high standard, of course, with a main circuit board mounted against a metal side panel and a smaller daughterboard sitting on top. Both boards are covered densely with surface‑mount components including an STM32 ARM Cortex microcontroller, which I gather is the heart of the PWM system. Given the very high-frequency (250kHz) PWM switching I was surprised that the whole unit wasn’t enclosed in a metal case, but I could detect no interference into adjacent modules in the rack, which is reassuring.

The published technical specifications are exemplary, of course, with maximum I/O levels of +26.5dBu (the input meter turns red at +24dBu), a frequency response within ±0.5dB from 2Hz up to 80kHz, and a noise floor below -100dBu (unweighted). This is all outstanding in a 500-series module.

Other specifications are equally astounding. For example, the fastest attack time (with the look-ahead mode enabled) is just six nanoseconds, and THD is amazingly low at around 0.00025% at nominal operating levels (with zero gain reduction). In most compressors, distortion rises with increasing gain reduction, but even at 20dB gain reduction the Brick Lane’s THD figure rises only to 0.00085% — such is the benefit of PWM! It seems rather contrary, then, that the Stress control deliberately introduces up to 20% distortion at extreme settings... but it’s nice distortion, so that’s all right!

One spec that raised an eyebrow is the current consumption. This is quoted as 140mA per rail (with a peak requirement of 160mA). It’s by no means the highest we’ve seen, but it does exceed the original API (and current VPR Alliance) specification for 500-series racks of 130mA per rail. That said, I tested a pair of Brick Lane modules in an original API Lunchbox frame with two preamp modules also installed and detected no problems at all. No doubt Cranborne

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The Brick Lane’s signal path is entirely analogue, but an STM32 ARM Cortex microcontroller lies at the heart of the PWM control circuitry that makes this compressor so incredibly versatile. designed the Brick Lane module for use in their own 500R8 and 500ADAT racks, both of which provide up to 250mA per rail, and many other 500-series rack manufacturers now use equally generous power supplies, including API’s own ‘high-current’ racks. So, I don’t believe the current consumption will be an issue in practice, but it’s worth being aware of if using an elderly API 500-series rack.

PWM limiters), Stress changes the character of peak clipping.

Additional configuration options can be accessed using a six-way DIP switch on the upper circuit board. The first switch selects the 3.5mm socket for sync’ing modules instead of the OptoSync facility, and the next two switches determine whether the level trim and bypass switch settings are mirrored on OptoSync’ed units. Switches 4 and 5 decide whether the input and output level controls remain active when the unit is bypassed, while switch 6 accesses the mysterious Enigma mode — for more on that see the separate box.

In Use Operating the Brick Lane compressor is perfectly straightforward, and the seven compression modes offer more than enough variety and controllability for most applications and requirements. I found it pleasingly easy to extract whatever flavour of compression I wanted for a given instrument, vocal or mix, fine-tuning settings as desired. It’s just a case of selecting an appropriate emulation mode, setting the threshold for the desired amount of squash, the attack and release controls to suit the material’s rhythm and dynamics, and the Stress control for character. Occasionally I manually changed the ratio value, but I rarely found that I wanted to delve into the nitty-gritty Enigma parameters — purely because Cranborne have optimised the parameter sets for each mode so incredibly well.

In my view, it’s so creative to choose a suitable-sounding mode and then adjust the few front-panel controls while listening to how the music is affected, and far more productive than focusing on too many control parameters and scale numbers, which only distract from the real job in hand.

In terms of sound quality, I can’t fault the Brick Lane at all. Like the Camden preamp, it covers every inch of ground from sublimely clean and transparent to in-your-face pumpy aggression, with everything you could ever want in between! If you want to shape drums or rein in a slap bass, the Smash mode will do it well. If you’re tracking vocals, try the Velvet or Float modes. Glue is ideal when Mixing, and the Tame and Polish modes are wonderfully transparent for mastering.

Enigma Mode The Brick Lane’s Enigma functions are enabled using DIP switch 6. Long presses left or right on the mode toggle can then access hidden parameter options to tailor the unit’s sound character to suit specific requirements and preferences. I doubt most users will ever want to meddle, but advanced users — and particularly mastering engineers — may find the adaptability and customisation extremely useful. Of course, a factory reset is available, should this inner tweaking go horribly wrong!

The Enigma functions are quite complex.

It’s all explained well in the user-manual but, as an overview, there are options for adjusting

The core strength of PWM compressors is how hard the compression can be driven without audible artefacts, and in that respect I’m hugely impressed with the Brick Lane 500 — particularly in the Tame compression and (blue) Polish limiting modes (which I note both use the triple side‑chain sensing configuration of peak, RMS and long RMS). The Velvet, Float, Smash and Glue modes all represent

the peak-clipping character, side-chain high-frequency emphasis settings, and side-chain detector modes (RMS, peak, and the weighting between the peak and RMS values for the dual modes), compression knee characteristics, progressive weightings for the attack and release times, and multiband crossover and polarity options for the Stress function.

There’s an immense capability here for fine-tuning the sound character and behaviour of different compressor modes, going far beyond most users’ requirements, but it’s fascinating to see the detailed modelling involved — and Cranborne’s fantastic attention to detail.

the key characteristics associated with analogue compressors so effectively that the Brick Lane genuinely can be used for any source, in any mixing or mastering application, and it’s this versatility that makes it a true classic in the making. Highly recommended! $ $599 T Cranborne Audio +44 (0)1707 656500 E marketing@cranborne‑audio.com W www.cranborne-audio.com

The Past Reinvented. The Future Redefined. www.soundonsound.com / October 2025

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To enter, please visit: https://sosm.ag/neve-comp-1025 desktop‑friendly format. The 88C offers all the controls you’d expect, including fully variable threshold, ratio and release time, plus provision for external side-chaining and stereo linking. The compressor’s attack employs Neve’s proprietary Adaptive

Attack Technology, which automatically adapts to the incoming signal, making the 88C ideal for both individual tracks and bus compression.

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“An all‑analogue dual‑channel VCA compressor, the 88C uses the exact same compressor circuit found in Neve’s flagship 88RS console...”

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October 2025 / www.soundonsound.com

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

JOHN WALDEN

D

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.

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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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N ATI V E INS T RU M E N T S CIR CU L A R

The Sequencer page is where Circular gets deep.

and then drag and drop the full Circular performance to (up to) four tracks within your DAW should you wish.

Not So Simple Sequencer However, it’s the Sequencer page where you begin so see Circular’s true depth.

Again, selecting one of the four slots brings its sequencer settings into view. There is obviously a lot going on here but a key feature to note is that, like Patterning, the sequences for the four layers are completely independent of each other.

Should you wish, you can therefore specify different settings for elements such as step counts (up to 64 steps), step playback rates, playback directions and step lengths in each layer. If polyrhythmic patterns appeal, Circular has you covered in a rather impressive fashion.

The pattern lanes are arranged in three vertical sections with Step Duration, the Parameter Lane (actually five lanes for Velocity, Probability, Note (pitch), Length and Step FX) and Sound Lane. Amongst a whole host of options here are the ability to specify per-step values for Step Duration, Step FX and the sound used. Yes, you can load a different sound (from the presets or a user sample) on a per-step basis. If you want, for example, a single lane to provide a compact set of drum or percussion samples, then that’s totally possible.

For each of the four layers, you can choose from a number of different sequencing modes. This includes a No Sequencing option so the sound within a layer can just be freely played via the MIDI keyboard. The PDF manual does a good job of explaining the purpose of these various modes but, whether you want sustained polyphonic playback for chords, or arpeggiation for more melodic playback, it’s all possible. Interestingly, the step counts, loop range, and loop starting point can be

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With comprehensive effects, modulation and mixing features, you are not short of sound‑design options.

configured differently within each of the three lane zones of the UI (and differently between each of the four layers). And, as well as lots of randomisation options within the sequencing engine, this includes the option to randomise the starting step.

At the base of the Sequencer page is a further panel with two tabs. The Sound tab gives you a set of synth-engine-like controls, including tuning, filter, amp and envelope options that can be configured differently for each sound within the Sound Lane as opposed to just the overall A, B, C or D layer. The Modulate panel lets you configure the two LFO and Shaper modulators.

Other modulation sources include velocity, aftertouch, the macro knobs, key tracking, MPE slide, and a few others. Clicking on one of Circular’s parameters in the Sequencer, FX or Mixer pages brings that parameter’s modulation setting into this panel, where you can link it to any two modulation sources and edit how the modulation is configured. Again, it’s powerful stuff.

Full Rack Circular doesn’t skimp on the effects options either. As well as insert (four per layer), global (again, four slots) and two send effects, you also get three per-step effects.

The layer effects are fairly conventional and mainly draw from Kontakt’s built-in repertoire of effects (reverb, delay, compression, EQ, modulation), but the per-step effects — which operate on the sequencer notes — have all sorts of creative options amongst them including bending, ornamentation, gater, tape, arpeggiation, rachet and grain.

The latter are very cool and operate via the Grain FX bus. Again, this offers four slots.

There are some truly weird and wonderful options to be found here.

However, this is topped off by the fact that you can use modulation to control in

real time which of the three per-step effects is active. The possibilities are mind-boggling.

The Argument For Circular?

The included Snapshot presets provide an excellent demonstration of how good Circular can sound but, even here, some time is needed to fully grasp what can be done.

The various presets exploit the engine’s capabilities in different ways and you do have to put some effort into working out just how to exploit each one in a musical context, particularly in terms of the modulation and multi-layered sequence triggering.

However, did I mention that Circular is deep? I did... and it is. It’s also not likely to be in the casual purchase price range for most potential users so, before you take the plunge, do be aware that I think it requires a genuine investment of time to realise the very considerable creative possibilities it has to offer. That said, the rewards are worth it; Circular sounds fabulous in a quirky, hypnotic kind of a way and, while it might not be for everyone, working film, TV and other media composers looking for a modern hybrid sound palette that combines organic and contemporary elements will undoubtedly appreciate the scoring potential. This is the kind of tool where you could easily create a complete cue from a single Snapshot once you have mastered the various modulation possibilities. It’s also an instrument that electronic music producers could put to very good use. OK, so this is perhaps not an instrument for those that prefer to paddle in the shallower end of the virtual instrument world, but if you are prepared to dive deep, then hold your breath; Circular has plenty of wonders to explore. $ $199 W www.native-instruments.com

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

HUGH ROBJOHNS

B

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

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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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and easily, and then locked securely in place, and since the Artcise mounts weigh only 52g each, the total weight of a stereo Ocula array is still only around 215g, and a Decca Tree array about 325 grams. Compared to most Decca Tree mic solutions that’s amazingly lightweight, and would therefore allow a much more slender (and thus less visible) support structure than usual, which is a very appealing feature.

In Use The electret capsule has evolved to the point of truly professional quality over recent decades, with many manufacturers embracing their benefits, including DPA, Earthworks, AKG, Line Audio and others. Indeed, the Line Audio OM1 omnidirectional microphone has become very popular in the semi-professional classical recording genre in recent years. I’ve used them myself on occasion with good results, so they seem a valid comparator for the Braingasm Ocula, albeit considerably less expensive.

I found the Ocula mics to be very easy to rig and configure on their versatile Artcise mounts, and their sound quality is impressive — crisp, clear and detailed, with a wide bandwidth extending well to the lowest frequencies.

The ‘enhanced’ directionality courtesy of the spherical body is subtle — akin to a hypo-cardioid rather than a full cardioid — but it definitely makes a worthwhile difference to the stereo imaging when the mics are angled outwards in a spaced array configuration.

Inside the XLR barrel is a circuit board, which converts 48V phantom power into the 3V required to bias the electret capsule, and also balances the output signal.

In a direct comparison with the OM1, the Braingasm Ocula definitely has a lower noise floor, and the increased HF sensitivity, airy presence, and subtle directionality make them far more versatile in distant‑miking applications, in my view, while remaining very discreet with minimal impact on sightlines.

Although clearly not intended for use in close-miking applications, the Ocula can be employed in those roles too, of course, bearing in mind that the fixed HF lift can be a little intrusive at times, depending on the source and exact positioning. That said, the HF boost is very smooth and natural, and it responds nicely to taming EQ, if necessary. In many cases, though, I found that its brightness added welcome detail and

Going APE The Neumann M50 became an extremely popular mainstay of classical recording from 1951, particularly when employed in the Decca Tree array. Neumann ceased production of the M50 in 1971, partly because of the complexity of its manufacture and the difficulty in maintaining consistent results.

However, they later revived the same ideas using newer technology, firstly in the TLM50 (released in 1990) and currently in the M150 Tube microphone. Meanwhile, a number of other manufacturers such as Flea and Wunder have been building homage microphones based around the classic M50.

An alternative approach employed by several manufacturers is to offer dedicated APE accessories that can be fitted temporarily over their own small-diaphragm omnidirectional microphones. DPA and Microtech Gefell have just such accessories, for example, and in most cases these are simple plastic spheres that slip snuggly over the microphone such that the capsule surface lies flush with the sphere.

Often different sizes of APE spheres are available, too, typically spanning 30-50 mm — the diameter determining the frequency at which the HF boost and narrowed directionality starts to take effect (the larger the sphere, the lower the starting frequency).

The advent of commercial 3D printing has also made third-party APE designs readily available, both for specific microphone models and as generic designs to fit specified mic diameters. And for DIY enthusiasts, home-grown solutions can be also constructed. A good few years ago, I made a set of APEs to fit my own Sennheiser MKH20 microphones from beech wood spheres bought online. I drilled the spheres with a 25mm Forstner bit, sprayed them black, and fitted grub screws to secure them to the mics. They have proven to be highly effective in many orchestral and choir recordings since.

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attack in a natural manner, especially to percussive instruments, helping them to really cut through well in a mix. The slight HF directivity usefully allows the mic’s focus to be directed to specific areas of an instrument in a beneficial way, too — not as obviously as aiming a cardioid mic, but without the proximity effect and phasiness associated with most pressuregradient designs.

The recommended price of the Ocula feels a little on the high side for me, at over twice the cost of the OM1, making it a big stretch for semi-pros.

That said, the Braingasm Ocula is a well-designed and well-engineered product filling a particular niche in the microphone market very nicely indeed, and that alone guarantees its popularity. Its technical performance is also very good, outperforming the Line Audio OM1 in my view and competing remarkably well with established small-diaphragm capacitor mics.

For both of those reasons I’m sure the Braingasm Ocula will become a popular choice for anyone seeking omnidirectional microphones optimised for use in the diffuse field, with minimal size and weight. And for anyone contemplating building a Decca Tree array, the Ocula is a total no-brainer as it is perfectly optimised for that role and is capable of very good quality results. $ Stereo pair €700 (approx. $812),

triple‑matched set €1000 ($1160).

Prices exclude tax & shipping.

W www.braingasmlab.com

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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

I

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.

To find a dealer visit www.musicmarketing.ca

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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

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ON TE ST

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

T

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

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Buchla’s standard vactrol is not RoHS compliant because it contains a very tiny amount of cadmium. And while it’s perfectly legal to buy an Easel direct from the USA, I suspect most of us would rather avoid the import process. Anticipating this, Buchla have decided to offer the ‘proper’ LPG free of charge, dispatched directly to the user.

I duly requested one and, after a wait of just a few days, it arrived and I popped it in.

To my ears, the replacement LPG is an instant, audible improvement, one that new Easel owners should definitely make the effort to obtain. If you were ever curious about the famed ‘Buchla Bongo’ sound — the result of a very short envelope pinging a Lo Pass Gate — then you absolutely need the real thing. Substitutes do not cut it.

Via switches, you select the mode for each LPG channel, choosing either a (non-resonant) 12dB/octave low-pass filter, a VCA or a combination of both. It’s this Combination Mode — filter plus VCA — that produces the pleasing effect we’ve come to associate with vactrols. The resistance of these opto‑electric circuits depends on the measurement of light from an LED — a response sometimes described as ‘sluggish’. More importantly in this context,

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The Music Easel Modern comes in a squarer box and adds the EMBIO (Electronic Music Box I/O) module.

fast hits in succession translate to an accumulated brightness. The results feel somehow alive, or maybe suggestive of an acoustic instrument.

The second gate may be fed by the Modulation Oscillator, the aux in or the first LPG output. If no connection is made at the aux input, a white noise source is helpfully normalled in. Finally, while the Easel doesn’t have a high-pass filter, routing the first LPG into the second produces some rather nasal, phase-cancellation type effects.

Further Into The 208c The Complex Oscillator and its wavefolding are key elements in the Easel’s sonic footprint. The process of flipping parts of simple waveforms to create new shapes has become better known in recent years, thanks to various Eurorack modules and standalone synths from Make Noise, Arturia and even Moog. Raising the timbre slider gradually warps the oscillator’s plain sine wave, turning it increasingly raspy and gnarly. If you sweep the fold amount with an envelope, it releases a rich seam of hard, biting tones. For extra depth, the sine can be mixed with a second waveform — either

a thin pulse, a square or a triangle — and the mix modulated in a subtle, almost PWM way.

Moving on, the Modulation Oscillator might initially seem like the poor relation.

After all, it lacks wavefolding and offers just a simple selection of sawtooth, square or triangle waves. Fortunately, these new Easels have a little extra something — as revealed in the Quick Start patch. There, the TiniJax cable is employed to patch the Complex Oscillator’s (audio) output to the Mod Oscillator’s FM input. And while it might seem contrary to the Buchla ethos, this bonus flavour of FM is well worth having.

As its name suggests, the Modulation Oscillator also provides a source of audio-rate modulation. Working together, the oscillators conspire to produce analogue FM and ring‑modulation that wouldn’t sound out of place on classic Radiophonic Workshop soundtracks. You can apply AM to external signals too, an option I recommend even if you only have loose mini-jack leads to hand. My first choice was to draft in a drum machine, because percussion’s clear transients are ideal for testing envelope followers. Then,

by applying low levels of AM, a humble Korg Volca was transformed — Cluster-style — by tremolo, while high AM levels yielded crunchy ring‑mod grooves. If at this point you were to patch the envelope follower to an LPG or oscillator, you might not emerge from your lair in time for tea!

In its Lo mode, the Modulation Oscillator behaves like a regular LFO, especially if you disengage the keyboard tracking. In the Modern Easel, a second LFO is available via the EMBIO, and it’s one characterised by its slowness (around 30 seconds at the lowest rate). So, while it’s never quite fast enough for vibrato, not even the wavery Boards Of Canada variety, this LFO can supply welcome life and movement to just about any parameter.

Positioned to the left of the oscillators, the envelope suits a right-hander who’s forever adjusting note shape while playing (eg. me). The envelope is comprised of three stages — attack, sustain and decay — with switches to determine the mode and source. Unusually, it needn’t be triggered exclusively by the keyboard — the Pulser or the sequencer can serve instead. Its versatility is further extended by voltage control of each stage, and by the choice of sustained, transient or looping operation. When looping, the decay value sets the speed of repeats.

The adjacent Pulser is a decaying sawtooth that can function either as a trigger source or a basic envelope. It, too, offers transient or sustained modes, so it can remain on during a keypress. A small button will manually start it and, in self-triggering mode, the Period slider sets the loop speed.

This, naturally, is voltage-controllable.

If we rope in the EMBIO’s Slew processor, the Pulser becomes a quite different animal. By slewing the start and end phases of a control signal separately, you can soften the Pulser’s otherwise instant attack and/or blur its release. And thanks to the stacking power of banana cables, you never lose access to the raw Pulser output, so, at times, it seems like you’ve got three envelopes.

Most of the coloured CV outputs are identical copies of each other, distributed around the panel for convenience.

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MIDI For a monophonic instrument, the Easel must surely hold the record for the greatest number of MIDI channels hogged. Courtesy of the 208c MIDI Implementation document (and some trial and error) I concluded that only channels 11-16 are truly safe from the Easel’s clutches. The oscillators respond to notes, portamento amount and pitch-bend on two opposing sets of MIDI channels and there are other channels allocated to the sequencer, the envelope and the Pulser. It’s comprehensive, but somewhat mind-boggling and I felt it was probably best suited to a computer and USB connectivity. Several MIDI CCs are assigned to the 218e keyboard, enabling you to capture the motions of the knobs and Strip.

Optionally, the keyboard can function as a polyphonic controller.

In general, five-pin MIDI users will face the most limitations (for example, the arpeggiator can sync to MIDI Clock but only if it’s received at the keyboard’s USB-C port). Still, at least the MIDI and the CV/gate inputs coexist without conflict, or at least I had no issues when sending notes, gates and MIDI CCs together. Generally, if what you mostly need is no-nonsense playback, the CV and gate inputs perform admirably, especially as the input is calibrated for 1V/ oct — unlike the 1.2V/oct standard employed by other Buchla gear.

www.soundonsound.com / October 2025

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It’s OK to want a Music Easel just for the cool Buchla branded case.

source, which generates a unique random value at each of its four outputs.

You’re therefore free to exploit the many possibilities this presents, with new values created for every trigger of the keyboard, Pulser or sequencer.

It’s easy to be dismissive of a mere five‑step analogue sequencer, given the ubiquity of eight or 16 steps in classic sequences and bass lines. But this is a timely reminder that sequencers are also useful voltage sources with a variety of uses. For example, when driven by the keyboard, this one can supply accents, stepped timbral changes, variations in envelope length and more. And, given that panel space is finite, Buchla deserve kudos for choosing five rather than four stages.

Switches above each slider play their part when the sequencer is used as a trigger source. It offers the possibility of triggers that occur only on every other note, or in other patterns within the five‑step limit. Sequence length may be set by switch or CV. Finally, a single knob is provided for the spring reverb — a splendid‑sounding example of the species with a range stretching from completely dry to a dense, reverb-only mulch.

A somewhat avant‑garde musician himself, Don Buchla was a keen advocate for innovative controllers, preferring them to conventional musical keyboards.

Nevertheless, the ‘almost normal’ 218e is a pleasant alternative to regular keys and, at just under 2.5 octaves, is fast and fun to play. If necessary, it can serve as a MIDI controller and is even fitted with pressure and velocity outputs.

In case you were wondering, keyboard pressure is generated by measuring the surface area of finger in contact, whereas velocity is a product of finger speed.

Admittedly, there’s not a massive range of either available but, via calibration

routines detailed in the user guide, pressure sensitivity can be tailored to distinguish between light finger-tip contact and a more full-on press. Similar routines fine-tune the velocity response, add slew to the pitch Strip and generally make the Easel your own.

The four Preset Voltage Sources are invaluable for numerous roles throughout the Easel and even within the 218e itself.

They can supply favourite portamento amounts, quirky pitch transpositions or specific arpeggiator speeds. And in another user customisation, the fourth knob can be assigned to octave transposition, and over a wider range than is otherwise possible.

New to this keyboard, the Strip is a convenient source of pitch-bend and finger-wobbly vibrato. For each contact, a pulse output is generated, which is useful for actions such as triggering the envelope or stepping the arpeggiator.

Dedicated controls supported by a few hidden powers bestow a wider appeal on the arpeggiator than its simple ‘up’ and ‘random’ directions promise. It can be latched with a sustain pedal or a long press of two buttons and earns bonus points by including random velocity variations. When cranked to maximum speed, the arpeggiator can go very fast indeed, turning the Easel’s output into a shimmering wall of sound.

For added complexity, it can be paired with the step sequencer or Pulser, and external synchronisation is available from the pulse input or MIDI Clock. Finally, another user configuration option, ‘directed random’,

The 218e Once upon a time, I tried and failed to get the older keyboard model to work reliably, so I’m very relieved to find the 218e a massive improvement in responsiveness. Indeed, I have yet to resort to the supplied earth cable, worry about what type of slippers I’m wearing or blame my weed-infused carpet for the static it generates.

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The Retro version of the Music Easel keeps things aesthetically closer to the original.

uses the Preset Voltage Sources to fluidly switch between up and down or descending random motions.

Retro & Modern The larger of the two models, the Modern Easel (58cm x 39cm x 16cm) is housed in a hard black plastic suitcase with a blanking panel indicating its capacity to house an extra module. The Retro has an even more solid aluminium case (55 x 42 x 15cm) as well as old-style knobs, but it sacrifices the EMBIO, additional module space and the CV/ gate inputs. Otherwise it has all the enhancements of the modern 208c and 218e.

Although I’ve mentioned the EMBIO throughout the review, there are still a few points about it worth making.

The first is that the slew knobs were rather wobbly, evoking memories of my earlier, less satisfying Easel encounter. The second is that its CV mixer is wonderful. The primary CV input can be attenuated and/or inverted and the crossfader (complete with CV input) facilitates patching marvels such as velocity-driven envelopes, the morphing of CV sources and much more. Lastly, I should briefly draw your attention to the CV to pulse converter. From triangle or sawtooth input, it generates a square/pulse wave suitable for clocking and triggering operations.

Conclusion It’s been a real pleasure to once again spend time with a Buchla Easel and ponder its place in the grand scheme of things. My overwhelming impression is of an instrument that inspires you to think and work differently — and maybe even to revise your approach to analogue synthesis.

It’s always a challenge, updating classic gear without breaking what made it classic in the first place, but in this respect Buchla have succeeded admirably, refining features and boosting flexibility without compromising the essential Easel nature. It also looks strikingly attractive and is one of those instruments that just sounds great, even in relatively simple patches.

In a recent book about the Minimoog, many users agreed it was ‘good for bass’, but I doubt there would be such a broad consensus for the Easel, which is more open-ended and therefore harder to pin down. Yes, it happily delivers bells, basses and buttery solos, but there’s no denying its gravitational pull towards the experimental and weird. If you’re curious about just how ‘out there’ it can get, it’s well worth trawling YouTube for self-playing instances of the ‘Krell patch’.

Compared to the Easel I bought some years ago, these updated models have been a delight and I’m really going to miss them. Finished to a much higher standard and with a keyboard that actually works, so many of the issues I experienced have finally been laid to rest. And while this is obviously not a budget instrument, neither were the reissues of the Minimoog or ARP 2600. Ultimately, both of the new Easels are a fitting tribute to Don Buchla.

Old designs they might be but they still hint at fresh and untapped possibilities, which isn’t bad for an electronic music box from the 1970s!

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$ $4999 W www.buchla.com www.soundonsound.com / October 2025

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USB Audio Interface

Bitwig Connect 4/12 NICK ROTHWELL

I

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

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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

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controller, except that rather than 16 sliders or eight knobs it offers its single dial. Glancing around my studio at my collection of multi-fader control devices, I was initially sceptical of the Connect: how can a single dial be any kind of control surface?

The answer is that it can only control one parameter at a time — no surprise there — but the parameter is chosen by mousing over the controls in Bitwig’s visual interface on screen. Click the rightmost Bitwig button to put the Connect into controller mode, and any moused-over control (fader, knob) gains a small white indicator triangle. The Connect’s dial jumps between controls as you move the mouse pointer, showing the current value on its LED ring, and even lighting the LEDs in the appropriate colour for devices with differently coloured control sections.

At first it feels a rather odd, two-handed workflow, and moving the mouse pointer doesn’t offer any physical feedback, but the system works surprisingly well. It doesn’t matter which track or device is currently selected in the session.

Click the Bitwig button again, and the dial locks to that parameter regardless of mouse pointer position, with a subdued white indication of this mode on the LEDs — useful for automation recording where you don’t want to accidentally knock the mouse pointer off target mid-record.

Virtual faders and knobs are afforded very fine control — the Connect dial’s LEDs fade in or out individually to indicate the slightest move. Some discrete parameters can be controlled, others not: in the Polymer synth, I can spin the dial to select my sub‑oscillator waveform, but not the main oscillator type (perhaps understandably, as this requires a swap of the internal modular synth modules). Sample scrubbing is rather fun: put a Sampler instance into ‘freeze playhead’ mode and you can sweep the sample, positioning with 0.1 percent accuracy, a good deal better than with a conventional MIDI controller. Support for third-party plug-ins is partial: you have to click on a control in a VST window, at which point the parameter becomes live in Bitwig’s generic device view and you can control it with the dial: not ideal, but certainly usable.

momentarily. The extent of the dial’s lit LEDs gives an indication of the zoom level. Scrolling is not compatible with playhead following: if you scroll the start marker, following is turned off.

(It feels to me that there should also be some button combo to re-enable following.)

Modular Interfacing To put the Connect 4/12 through its paces as an interface for controlling modular synthesizers from Bitwig Studio, I reached out to my colleague Tom Armitage at University Of The Arts London who duly turned up with a Music Thing Modular Workshop System (reviewed in Sound On An LFO-modulated control voltage comes out of Bitwig Studio, Sound April 2025), a self-contained through a patch cable and back into an oscilloscope. modular synth that’s not much bigger Automation recording is supported than the Connect itself. only in the linear Arranger view for now, The most direct way to hook up and integrates with the Connect’s transport a modular is by using the HW CV controls. The Record button enables MIDI Instrument device, a Bitwig instrument or audio recording on armed Arranger which takes in MIDI, converts it to CV/ tracks, as you would expect. Hold the gate, sends that to the hardware and Bitwig button and press Record to arm takes in the resulting audio. You select the two interface ports for CV and gate output, and the input port (or port pair for stereo) for audio coming back. All of Bitwig’s CV I/O endpoints detect when a Connect interface is attached, and rely on its internal 1V/ automation recording only. The actual octave calibration. The HW CV device also automation mode is a Bitwig preference: has an embedded device chain purely for latch, touch or write. For latch or touch, hosting additional CV outputs: these are the Connect’s dial touch-sensitivity comes fixed, although you can alter them with into play: latch starts recording when the modulators, automate them and so on. dial is touched, while touch punches in In the screenshot, I’ve demonstrated by only whilst it is being touched. The dial’s modulating a CV out with a Bitwig LFO, sensitivity is a slight liability here: I noticed physically patching it up as a loopback on occasional automation points being the Connect itself, and have an oscilloscope registered even when I was sure the dial display of the result coming in. wasn’t being moved, but these can easily At a more granular level, CV input is be removed afterwards. available as a modulator for devices in The Scroll button enables scrolling of Bitwig Studio. We spent a happy hour or the Arranger view. Scrolling moves the so routing a meandering LFO from the playback start marker without affecting Workshop System into a sampler, using playback immediately; to jump to the the signal to perform a granular sweep scrolled position while playing, tap the over a drum loop.

Play button. Tapping the Scroll button To really explore the control potential again toggles between scrolling and of analogue sources, you need to dig into zooming, with LED colour indication to Bitwig’s Grid modular environment. Here, match — or you can double‑tap and hold CV and gate inputs and outputs can be the dial itself to switch to zoom mode wired up more flexibly, and you can send

“It is a fascinating process to be patching two modular systems at the same time — one physical, one virtual — and the Connect allows them to be linked together with minimal fuss.”

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control and audio back and forth between the real and virtual modular worlds in all sorts of combinations. It is a fascinating process to be patching two modular systems at the same time — one physical, one virtual — and the Connect allows them to be linked together with minimal fuss.

Add non-modular software devices to the picture and the potential is enormous.

As with my loopback experiment, the Grid’s oscilloscope comes in useful here to see what’s going on. Each of the 3.5mm jacks has its own multi-colour LED, which tracks the colour of the Grid module’s input or output on screen (though the colour indication was not totally reliable for me). Multiple signals to the same output are summed.

At this stage I did start to question the choice of six main audio outputs and only two inputs (not counting headphones or the 3.5mm jacks). I would have thought it more likely that users would want to bring more audio channels into Bitwig Studio than drive lots of outputs: for most gigs, stereo + headphones is plenty, and if you want to do serious surround then you’ll

Two oscilloscopes in Bitwig Studio’s Grid device: control voltage in (white) and corresponding filter-controlled audio from the Workshop System (red).

want more than six outputs, maybe even an ADAT connection. A Connect 6/10 (or even an 8/8!) would probably suit my needs a little better.

Conclusion I was initially sceptical of a control surface with only one dial, but the workflow of the Connect 4/12 has been really well thought out, the routing and control functions

are easy to reach, and the mouse-over navigation works much better than I expected. Its interfacing with modular gear is easy to use and just works, which fits in perfectly with Bitwig Studio’s existing modular environment. I just wish it had a few more audio inputs! $ $599 W www.bitwig.com

www.soundonsound.com / October 2025

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TiNRS Bopp & Steve EURORACK MODULE

T

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

W

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.”

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Omnitone Rhythmi & Beatsi EURORACK MODULE

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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

E

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.

W

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

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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

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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

W

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!

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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

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

PAUL WHITE

Y

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.

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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

97

ON TE ST

IK Multimedia Prism Reverb Reverb Plug-in Whether added to your T-Racks 6 Max ecosystem (in which case it’s free) or purchased separately, IK Multimedia’s Prism reverb is a creative reverb that IK claim goes “beyond shimmer” to deliver “a new dimension of sound”. A bold claim, so let’s see how it stacks up...

Prism comprises four clearly defined, colour-coded processing modules: Reverb, Mod, Pitch and Feedback, each of which has a bypass switch. Reverb takes care of the initial reverb generation, creating the essential spatial characteristics of the effect, and the reverb size can be anything from short and tight to a very long wash. There are controls here for size, diffusion, low-pass filter and high-pass filter, and all values are shown in small print next to the control names. A display above the controls shows the reverb development as a cloud of pixels.

At low diffusion settings individual repeats are clearly audible. The decay tail doesn’t have a very natural shape — but as you’ll see below, that changes when we come to the Feedback section.

In the Mod section are two identical sets of controls for the left and right channels, and these can be linked by clicking a chain icon. Here you find pitch modulation with controls for rate and depth, as well as random and tempo-sync options. A display shows the modulation for each channel in red and purple. Pitch is where you go to create shimmery reverbs or grumbly depths,

as there are two individually pannable pitch‑shifters (up to ±24 semitones). A fine tune button allows for more subtle detuning, and a display shows how far the pitches diverge.

Lastly comes the Feedback section I referred to above, with a control for level (amount) and two filter controls: Range and Centre (frequency) that change the shape of a feedback filter, which is displayed above the controls.

Additionally there are more conventional high- and low-cut filter controls, and Flip and FB Pitch buttons — FB Pitch takes the pitch‑shifting in or out of the feedback loop, while Flip swaps the stereo channels on each feedback pass, to add spatial depth and complexity.

Sitting above these four modules is a Master section, providing global wet/dry mix, input and output gain with metering, a stereo width control and a bypass/power button. A Reset button sets all parameters to their default positions.

While each section has a fairly limited number of controls, the range of effects on offer is impressive — and the shimmer reverbs are amongst the best I’ve heard.

Without adding in some Feedback, the reverb decay does sound somewhat unnatural, but once you bring that into play, smoothly decaying, dense reverb tails are in plentiful supply. And if you max out Feedback, the whole system just keeps on recirculating sound indefinitely.

Strymon Cloudburst Reverb Plug-in Strymon’s Cloudburst pedal (reviewed in SOS April 23: https://sosm. ag/0423-strymon-cloudburst) won a lot of friends amongst those into ambient, shoegaze and cinematic-style composing.

Now, it’s available as a plug-in, which should make life a lot easier for many of them! It’s authorised via iLok, and it supports VST3, AU, and AAX hosts on Mac and Windows platforms. It has a low CPU usage and very low latency, making it practical for live as well as studio use.

Both the Cloudburst pedal and its new plug-in sibling are based on an algorithmic reverb effect that doesn’t set out to emulate a real space, but rather to create harmonically rich, bloomy reverb textures.

As with most algorithmic reverbs, the core algorithm utilises a series of interconnected

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delay lines with complex feedback paths, filtering and some time-based modulation to avoid the metallic ringing artefacts that beset some simpler digital reverbs. The signal path handles any permutation of stereo or mono in and out, with the left and right signals mixed to mono before hitting the reverb engine. The reverb algorithm then creates the wide, spacious effect users of the pedal will find familiar.

While that approach to reverb generation is not unique, the Ensemble mode makes Cloudburst very different from the

The pitch‑shifter allow for the creation of wonderfully lush shimmery effects that sound really smooth — added to a piano it sounds like a string section in the background. The effect on clean guitar is equally magical. IK say that Prism is more than a shimmer reverb, and that is certainly true, but for me its shimmer-tinted reverbs are what it does best.

I haven’t heard a more mellifluous shimmer since the Strymon Cloudburst, the plug-in version of which is also reviewed in this issue. Although Cloudburst works in a very different way, Prism manages to deliver that very same sense of smoothness. As with most long, shimmery reverbs, ambient composers, bands who stare at their footwear and those working on sound for picture will probably be the core users. But anyone who’s into the more experimental side of pop could get a lot of out of Prism too. Paul White $ $99.99

(Free for owners of T-Racks 6 Max.)

W www.ikmultimedia.com

norm. Strymon explain that it works by splitting the incoming audio into multiple frequency bands, and then resynthesizing a new harmonically rich signal — which has an almost orchestral texture — from the divided input. It’s this signal that’s fed into the reverb engine.

Strymon are keen to point out that it’s much more than the usual ‘octave up’ shimmer effect, and say it’s more akin to polyphonic additive synthesis, which tracks the dry input signal’s pitch and reacts dynamically to the input. The outcome is that the harmonically related layers evolve with the input audio in a very natural way.

An adjustable pre-delay feeds into the reverb engine, which on the pedal version offers three Ensemble settings: off, and two preset depths. On the plug-in, the Ensemble effect is fully adjustable courtesy of the large Amount knob, placed between the two harmonic displays, and the level

of reverb is set using the Mix control.

The Decay time control can go from short reverbs to huge sustaining clouds of washy loveliness, and there’s a Tone control that affects only the reverb sound. Modulation is a key element of the Cloudburst’s sound too: the modulations within the reverb algorithm are controlled by low-frequency oscillators that are subtly randomised in phase and rate, independently for each channel. The Mod knob allows users to control this modulation, which becomes clearly audible as an effect at higher settings, adding both pitch drift and, when used in stereo, spatial movement within the reverb tail. The Mod control functions as a modulation depth control up to the mid point, then from there onwards it increases the modulation speed. Small input and output gain controls, with metering, are located at the top left of the GUI. There’s even a Hold button with two modes, for infinite decay time or freeze. Infinite allows the reverbs to sustain indefinitely while each new

note played adds to the reverb signal.

Freeze provides infinite sustain, but allows new (unprocessed) notes to be played on top of the sustained signal without adding to it. While the plug-in retains the straightforward operational paradigm of the pedal, it also benefits from useful enhancements, including MIDI parameter control, DAW tempo sync for the pre-delay, host-based preset management and host parameter automation. There’s also a Dry Kill button to make it easy to set up as an aux send effect.

Dialling down the adjustable Ensemble effect allows the plug-in to be used as a straightforward reverb when necessary, but its real strength is in creating harmonic washes of sound.

As with the pedal, the wonderfully lush harmonic reverb tail is unique amongst reverb effects and it feels very liberating to be able to use it within the plug-in environment. I’ve spent many a happy hour trying to recreate the sound using combinations of other

plug-ins but nothing really comes close in creating that ‘Do I hear a string synth in the background?’ reverb character.

Cloudburst also has that uncanny ability to really lay on large amounts of reverb without obscuring the original sound.

I already have a Cloudburst pedal for live use, but I write a lot of ambient music these days, and I can see the plug-in getting a lot of use in my forthcoming ‘music to soak lentils by’ projects. The plug-in costs just over a third of the pedal’s asking price, but it sports some genuine enhancements, such as fully variable Ensemble and the two Hold modes, and in the studio it’s often just more convenient to use the plug-in.

So it is definitely going into my favourites folder. This won’t appeal only to ambient composers, though — Cloudburst delivers a genuinely unique alternative to shimmer reverb that exudes class and could fit into any genre where a bit of textural magic is needed. Paul White

LANG Electronics 312L

in a rackmounted 500-series chassis), or instrument‑level signals on the jack, the input impedance being 2MΩ. And while I’m on the subject of impedances, at 75Ω the output impedance is low enough to drive the 600Ω loads presented by some older processors.

The circuitry is a combination of surface-mount and through-hole devices.

The input transformer is custom one designed by Heritage and is shielded by a mu-metal can, while the chunkier-looking output transformer is again a custom one, but made by Cinemag. Heritage also designed their own op‑amp, the HA-880, and this is removable — you could substitute it with others, including those used in API’s consoles if you wanted.

I tested this preamp on a number of mic sources, including with some passive ribbons and, since I don’t have an SM7 here, an SM57 and an older Shure Unidyne mic. With all those mics, while I didn’t find that I needed anything like the full 90dB of gain, engaging the booster made the main gain control easier to judge, and resulted in a change (for the better) in tonality, not just in

gain. The subjective qualities of a mic preamp are always relatively subtle in the grand scheme of a signal chain, and there’s no attenuation control after the output transformer so if you want to really ‘drive’ this one for deliberate saturation, you’re going to need something after it to take care of that. That said, there’s definitely an API-ish vibe to the sound — a punch, presence or life if you like, that can be delightful on electric guitar for rock in particular. But the noise is decently low, as befits a preamp with so much gain available, and as with most console-style preamps, you can attain decent results on pretty much anything. I also tried running some mono stems and mixes through it (I had only one module available), and it did add a certain richness that could be appealing on the right material, which suggests that a pair of these might make a pleasing partner for a passive summing box.

All in all, then, this is an appealing mic preamp, in terms of its technical performance, its sound and its feature set. One that should appeal to users of low-sensitivity passive ribbons and moving-coil mics in particular, but perfectly capable of amplifying capacitor mics too.

The instrument input is a plus that makes a great electric guitar or bass DI. At this price, it’s very tempting. Matt Houghton

500-series Microphone Preamp The LANG 312L, made by Heritage Audio, is a transfomer-balanced mic preamp whose design is broadly rooted in old API one, but which, like all the best ‘homages’, brings something extra to the party. The main gain control at the top is a pot, and beneath it there’s button to engage a Cloudlifter-style booster preamp that’s included within the module. Marked Ribbon & SM7, this not only adds a fixed 25dB of gain (making the gain range a generous 15-90 dB) but it also changes the preamp’s input impedance from 1.2kΩ to 10kΩ, which means vintage ribbons and moving-coil dynamics will see a suitable load. The bottom of the four buttons engages a 20dB pad, which makes it possible to accommodate line-level signals.

The other buttons switch on 48V phantom power and invert the signal polarity, as you might expect. I was a little surprised to find no high-pass filter option included, to remove rumble or counter the proximity effect, but there is a handy front-panel Neutrik Combo XLR/jack input. This can accept mic signals on the XLR (potentially a handy shortcut when the preamp’s

$ $79 W www.strymon.net

$ $349.

W https://heritageaudio.com www.soundonsound.com / October 2025

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Studio One

TECHNIQUE

The Chord Selector Wheel.

The Chord track continues to mature, and now lets you use Nashville Numbering.

ROBIN VINCENT

W

e’ve followed the development of the Chord track over the years, and have got stuck into it a few times in these workshops. We marvelled at its introduction in version 4, revelled in its appearance in the Show Page in version 5, and admired the quality of the printout on lead sheets and guitar tablature in version 6. In version 7.22, the Chord system gets an additional approach, and I thought that would be a good opportunity to check in with how chords flow through Studio One. This new facet is called Nashville Numbering.

There’s nothing new about Nashville Numbering; it’s a tried and tested way of representing chords that’s terribly useful for session musicians, cover bands and worship environments where keys can find themselves changed without a moment’s notice. It’s the idea that instead of naming chords in a song by a set key (using C, F and G and so on), you would use a numbering system where the key root major chord would be 1 and the rest would be numbered relative to that. So, in our C, F and G example, they would be 1, 4 and 5. If you’ve had any musical training, those numbers would already be familiar, and you might have even come across them as roman numerals such as I, IV and V. The fabulous thing about this system is that if someone decides to change the key, you don’t have to print the whole sheet out again, or scribble new letters over your music. All that’s required is

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that you know a bit about how chords in different keys relate to each other. So if you’re playing in the key of C, then the number 5 chord could be G, but if you’re playing in E, then the 5 chord would be B. In G it would be D, and so on. So, regardless of whatever key you wish to play in, the Nashville Numbered chord sheet will remain the same.

When using major chords, you can simply use the numbers 1 to 7 to represent all the notes in a major scale. Once your chords become more exotic, it can get a little bit more complicated. You can use sharp or flat symbols, which is easy enough, but getting into chord types you have a ‘-’ for minor, a little ⁷ for dominant 7th, a triangle for major 7th and a circle for diminished. Many of these are outside the realms of a standard alphanumeric keyboard, but what’s important is how Studio One presents and interprets the information.

Striking A Chord Let’s recap how and where the chord system gets its fingers into Studio One. At the top of the arrange page, along with things like the Marker, Video, Lyric and Arranger tracks, you will find the Chord track. Double-click (or Ctrl/Cmd‑click) in space and it will create a bar-sized, resizable box with the chord C written in it. Double-click the box, and the Chord Selector Wheel will pop up, giving you access to every chord you ever thought of. You can select chords by name, type or interval, or you can click in the keyboard to find the name of any combination of

notes you wish to represent. It’s a handy and educational tool.

However, all this adding chords by hand seems a little long-winded considering that the necessary musical information is probably already in the tracks of your project. Well, Studio One has considered this and gives you the convenient opportunity to extract chords from either MIDI or audio tracks to populate the Chord track automatically.

For a MIDI instrument track, you’ll find the Extract to Chord Track option under the right-click Instrument Parts menu.

For an audio track, you’ll find it under the right-click Audio menu. You can also pull a MIDI file into the Chord track directly from the browser, and Studio One will lay out all the chords for you without your having to create a single track. In all instances, the chord detection is pretty good. You might need to do a bit more editing and adjustment when extracting from an audio track, but that does depend on the source material.

Once your Chord track is fully populated, it has a few tricks up its sleeve.

The first one will bring up a large display of the current and next chord. So, if you were recording a new guitar track, you could easily follow the chords of the song on screen. To pull up the Chord Display, click on the little note symbol on the Chord track header. There are three views you can look at. The Chord Track view gives you the current chord and next chord so you can play along with your song. The Input Chord view will show you the name of whatever chord you are

playing on a connected MIDI keyboard.

The Editor view will show you chords that are currently selected in the note editor.

The next trick is that you can pull the chords from the Chord track into an instrument track, and it will create an event containing the MIDI notes of each chord. It’s just simple triads that last for the length of the chord box, but it could be a great way to turn a guitar track into a different instrument. You are of course free to edit the chord notes, but this will not be reflected in the Chord track unless you pull the edited event back over the original chord box. This interplay gives you a lot of versatility as you shift chords from Chord track to Instrument track and back again.

all your instrument and audio parts. The Chord track becomes a bit of a harmonic playground. To make this happen, you need to enable the Follow Chords function for both the Chord track (press the big Follow button) and the tracks that are going to be following it. You’ll find this as an option in the Inspector for the track. Halfway down there’s a drop-down

Remarkably, Studio One can also do this for audio tracks. Now, polyphonic pitch‑shifting is not always fabulous and can produce a lot of artefacts. In the manual, PreSonus suggest that this is very useful for experimentation and trying out ideas, but you will likely have to re‑record the audio track once you’ve decided on the changes. To help with the accuracy, there are some Tune Modes on audio tracks that optimise the algorithm to the source material.

Editing the chords in the Chord track can be done in a couple of ways.

Double-clicking the chord will bring up the Chord Selector Wheel that we saw at the beginning of the workshop, and you can use that to change every aspect of the chord. But there’s a simpler option, which is to right-click the chord.

With this menu, you get quick access to the root, type, bass and intervals used in the chord. You can also transpose or select all the chords of that type to change them together.

The last area of play is in the Chord Track Inspector. The top section of the

“With Nashville Numbering, the key becomes very important, as that denotes the chords that the numbers refer to.”

menu for Follow Chords. Initially this is set to Off, but there are three other options.

In Parallel mode, the notes align with the Following Suit root note of the target chord and existing note relationships are maintained. In You can take this a step further by Narrow mode, all the notes shift to the imposing the Chord track onto tracks that nearest note in the target chord. And Bass already exist. The brilliant thing about this mode is the same as Parallel, but where is that you can experiment with different the bass or root note shifts to the nearest chord progressions and harmonic note of the structures without having to re‑record Sound On Sound - PSI Audio AVAA C214 Ad – July 2024_PRINT.pdf 1 target 7/1/24chord.

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www.soundonsound.com / October 2025

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101

TECHNIQUE

THE CHORD TRACK

Studio One

The Chord track can now show your chords in the Nashville Numbering style in the Chord Display.

inspector is all about the auditioning of chords within the Chord track. In a way, it turns the Chord track into a musical track of its own. With Audition Chords enabled, the chords will sound whenever you click on them. But sound with what?

Well, you can select any loaded virtual instrument or MIDI output from the Instrument drop-down menu. If you have nothing else to hand, you can choose Chord Preview and get treated to some

The Extract to Chord Track feature will populate the Chord track with all the chords in your selection, be it audio or MIDI.

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terrible piano sound Studio One has hidden away within itself. Enabling Play Track will allow all the chords to sound in order as the song plays back.

You may have noticed the list of chords in the inspector that look suspiciously like the entries in the Arranger track for different song sections. These are a list of all the chords in your song as they happen, and they act in exactly the same way as the Arranger sections. They let

you dash about your song by clicking alongside the chord you want to get to.

Nashville Numbering Finally, after journeying through the chord system, we find ourselves ready to check out the Nashville Numbering options. You’ll find them in the Chord track inspector under Chord Display. You have a couple of options here, beginning with Regular, which shows the chords

using standard definitions. Alternatively, you can display the chords as numbers like they do in Nashville, or you can opt for Roman numerals if you were perhaps more classically trained. We then have some additional options in terms of style.

Small Caps uses two‑ and three-letter abbreviations, so ‘mi’ for minor and ‘aug’ and ‘dim’ for augmented and diminished, respectively. Minimal reduces the letter count a little bit with ‘m’ for minor and ‘M7’ for major 7. And then we have Symbol, which introduces the aforementioned Nashville Numbering.

With Nashville Numbering, the key becomes very important, as that denotes the chords that the numbers refer to.

If you want the Chord track to play back in the correct key, you’ll need to define it. This can be done in a couple of places. You’ll find an entry for Key on the transport bar, and you’ll also find the option in the Chord Track Inspector.

However, there’s a little bit of a disconnect between the key and chords. If you change the key, nothing actually happens to the chords; they play

back the same, and in Regular chord mode, they remain named the same.

So, a C chord is still a C chord whether you are in the key of E flat or G minor.

However, as the Nashville Number system defines chords in relation to the key, you’ll see the numbers change as you change the key, in order that the played chord remains the same. This sort of defeats the point of the Nashville Numbering system, but there’s a way to make this right through the use of Transpose. To make it work, start by defining the original key and set out the numbered chords. Then, if you wish to change the key being played, use the Transpose option next to the Key on the transport bar. This changes the key and pulls the Chord track along with you.

There’s one other box you have to tick and that’s in the inspector of each of the rest of the tracks. It’s called Follow Global Transpose — make sure it’s ticked.

The chord system in Studio One is super-powerful and can be a terrific aid to songwriting and playing with harmonic content. Nashville Numbering is also a useful and very welcome option.

www.soundonsound.com / October 2025

103

Logic Logic

TECHNIQUE

Stem Splitter is an amazing feat of technology, but is it actually useful? Yes!

Using Stem Splitter couldn’t be easier: just import an audio file into Logic and select Stem Splitter from the Functions menu.

DAV I D R I CA R D

W

hen the Stem Splitter function appeared in Logic Pro 11, I brushed it off as a somewhat alluring gimmick. The ability to separate individual elements from a mixed track had already become mainstream in standalone applications, so the novelty had come and gone for me.

Logic Pro 11.2 and Logic Pro for iPad 2.2 introduced further features to the function, and it was then that I had to admit that perhaps I had been overlooking the power of the Stem Splitter.

The operation of Stem Splitter couldn’t be more straightforward: drag or import an audio file into the Main window, and, with the region selected, choose Stem Splitter from the local Functions menu. The menu defaults to having all the elements (Drums, Bass, Vocals, Other) ticked and the Separate All Stems option selected.

Of course, if you don’t need each stem extracted, go ahead and tick what you want, or use the preset menu to find a combination that suits your needs. Logic Pro does the busywork of creating the stems, placing them in a summing folder and muting out the original.

If you unmute the original region and mute the track, you can easily toggle the track’s solo button to switch between the original track and the sum of its stems. And while you definitely hear a slight difference, the two are essentially the same.

OK, cool trick. We’re all impressed.

But now what?

Remixing For many musicians and producers, remixing existing commercial releases would be the obvious use for Stem Splitter.

After following the aforementioned procedure for extracting stems, I usually put the BPM Counter plug in (found in the

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Metering folder) on the drum track. Play back a few bars and this indispensable tool displays the song’s tempo. After typing the bpm value into the Tempo window, I’m almost ready to get started.

It’s always a good idea to select all of the regions and move them forward or backward to line the beats up to the grid.

Try to get the first discernable downbeat to hit at the start of a measure. Now when you loop a section with the Cycle range, it should play back as if it were written in the Logic Pro session all along.

If it’s not lining up or staying in the grid, it might be because it was recorded without a click (or didn’t strictly adhere to one), or there are tempo changes throughout the song. But for our purposes, let’s assume the tempo is consistent and the tracks are perfectly locked to our grid.

At this point, you can let your creativity guide you. Add instruments, mute instruments, add different effects to anything and everything. You can mute the bass, guitars and piano and totally reharmonise the song. You can even grab the drum track and create a groove template from it and apply it to your next masterpiece! The possibilities are endless.

The ability to remix existing music without having to obtain the original multitracks is a gamechanger. This use of Stem Splitter alone is very powerful, but let’s keep searching for other ways to exploit this tool.

Raiding The Archive Commercial music is the go-to candidate for the Stem Splitter, but what about our own music? Every month I get a request from a music house to pull tracks (and have stems ready) from my library in a different style for a creative pitch they are working on. Some of the tracks I submit are from so long ago that, even if I could find the original files, opening

them and having them play back in Logic would be challenging.

Stem Splitter means that — as long as I have the stereo mix — I can extract enough material for any tweaks that may be needed.

This could be invaluable for post-production professionals who feel limited by only having a stereo mix of a track that they’ve licensed. Rather than ducking the volume of the track, they could pull out different elements to make room for dialogue and effects.

In a similar vein, I have a box of cassettes from the ’80s and ’90s that could benefit from the Stem Splitter — high school and college band rehearsals, demos, and live shows, as well as mixes from my four-track days. Sure, the cringe factor alone has prevented me from revisiting these chestnuts, but maybe, just maybe, with this new functionality, I could finally face the music and… Who knows? Maybe there’s a song in there that just needs a little additional production to make it sound professional-ish.

Crate Digging This leads me to another potential goldmine. Maybe in the bridge of our worst song from the good old days, the drummer played the most inspiring and hypnotic groove. How great is it that we can extract it and use all of our modern tools to create a retro drum loop with a flavour that would be difficult or impossible to convincingly create from scratch? The same could be said for a quirky guitar riff or a lead vocal.

You’ve essentially found yourself in possession of your own royalty-free retro sound library.

Mix & Match When the concept of stem separation first came to my attention, my first thought

wasn’t about remixing or scouring my archives for drum loops. No, my immediate impulse was to recognise the potential this could have on my own mixes.

I’m a huge fan of mix referencing.

But Stem Splitter allows us to go a bit more granular. Instead of A/B’ing your whole mix against a reference track, why not compare your drum stem to that of the reference?

When I listen to a reference mix, it can be deceiving how much low end is actually present in the bass. But now I can solo the bass stem in my reference track, open up a Channel EQ and hear (and see in the EQ’s analyser) exactly what’s going on in the low end. I could even use the Match EQ plug‑in to match my bass’s frequency content to that of the reference track. Or I can compare my vocals with commercial recordings to see how much air there truly is in the reference track’s vocal.

bass stem and play along with the band like karaoke! When I recently had to learn a set for an ’80s tribute band I was subbing for, I created the whole set list in one Logic session.

After learning the bass lines, I would mute the bass stem, press Play, and run through the entire set.

It’s head-scratchingly amazing that Logic can pick out the reverb and delays for each instrument group and keep them contained in their instrument’s stem.

Restorative Properties Earlier this year, a friend called me in a panic. “You’re the audio expert, I need your help!” This could be the day that my immodesty and reality collide, I thought. Turns out she had made a man-on-the-street type video, and the dialogue and background noise was all over the place. I assured her I could solve the issue — while I was simultaneously downloading trials for iZotope’s RX. But before installing anything, I imported the audio from the movie into Logic and checked to see if EQ would solve the issue. No dice. Then I took a chance and ran it through Stem Splitter. It interpreted the dialogue as Vocals and the noise as Other.

The results were impressive. After removing most of the background noise, I was able to salvage the dialogue with my usual EQ and dynamics processing tools.

This is just the beginning. I’m confident that in future updates, Stem Splitter will become more powerful still, and the tool’s possibilities will only be limited by our imaginations!

“I’m a huge fan of mix referencing.

But Stem Splitter allows us to go a bit more granular. Instead of A/B’ing your whole mix against a reference track, why not compare your drum stem to that of the reference?”

Learn & Play We described using Logic Pro as a tool for transcription last month, but I’ll add my own experience of using Stem Splitter in this role here. As a bassist, I’ve used this function many times to isolate a track’s bass line so I can not only hear it better, but I can actually learn what’s being played more easily. Alternatively, you can mute the

Stem Splitter also allows us to get a peak behind the curtain to understand what’s happening on our favourite records. For example, I like to zoom in on the bass and drum stems and try to see whether they are playing together, or if the bass is just a little ahead of or behind the beat.

I was also surprised to hear how little reverb was used on the horn section from one of my favourite big band albums.

The hugeness in the mix was not coming from reverb in this case, but there was no way to know that without Stem Splitter.

Not just for music: Stem Splitter did a surprisingly good job of removing background noise and hiss from a badly recorded audio interview. It interpreted the dialogue as Vocals and the noise as Other.

www.soundonsound.com / October 2025

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TECHNIQUE

Pro Tools offers a number of ways to implement side‑chaining in your mixes.

JULIAN RODGERS

I

n audio production circles, ‘side‑chaining’ is one of those terms which is often used in a rather narrow and not entirely accurate way. You’ve probably encountered the term as shorthand for one particular use: compressing a source using an external signal as a trigger, to create a ‘ducking’ or ‘pumping’ effect.

While valid and very effective, that technique represents only one of the ways side‑chains are used. For simplicity I’ll refer to compression, but these principles apply to any dynamic process: compression, gating, expansion or limiting.

In Pro Tools, as in other DAWs, dynamics processors rely on side‑chains. Whether you’re using Smack or Pro Compressor, you’re working with a detector circuit that operates with reference to a side‑chain.

The better you understand that, the more control you’ll have.

What Is A Side‑chain?

All conventional dynamics plug-ins control the level of audio passing through them by monitoring the level of a control signal and comparing it to a reference level: the level at which an action should begin. This reference level is called the threshold, and what happens in response to the control signal (the side‑chain signal) crossing the reference level (the threshold) depends on the type of process (compression, expansion and so on), and the settings of the other controls (principally ratio, attack and release times).

The side‑chain is the path carrying the control signal that feeds the detector in a dynamics processor. It’s what the plug‑in is “listening to” in order to decide when to apply gain reduction, and how much.

By default, the side-chain signal is derived from the input signal to which compression is being applied. This is known as an internal side‑chain.

All compressors, even simple ones like Dyn3 Compressor/Limiter, have a side‑chain path. What varies is how accessible and adjustable that side‑chain path is. Some plug‑ins, like the 1176‑style BF76, have unconventional control sets

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Screen 1: A common production technique is to high‑pass filter a compressor’s internal side-chain, so that kicks and basses don’t trigger excessive compression. Some compressor plug‑ins have these filters built in, but if not, you can set up your own filtering in the Pro Tools mixer.

with no threshold control or access to the side‑chain path for filtering, but the side‑chain path is still there. More advanced Pro Tools plug‑ins like Channel Strip and Pro Compressor expose additional controls for shaping the side‑chain signal, allowing you to fine-tune how the detector responds.

Internal Vs External Some compressor plug‑ins allow you to override the default internal side-chain routing by offering an external ‘key’ input. This allows you to employ a completely different audio signal as the side‑chain source. One application for this is to set up a compressor to ‘duck’ audio out of the way of another signal. For example, if you use the bass drum signal as the side-chain source for compressor on a pad sound, you can get the pad to ‘pump’ in time with the beat.

To use an external side‑chain signal in Pro Tools, you assign a bus to the Key Input selector in the plug‑in’s header section. You can feed that bus using a send, or you can hold Control (Mac) or Start (Windows) while clicking an output assignment to route a track’s output to more than one place. Pro Tools indicates this multiple routing with a ‘+’ next to the output. This is neat, but gives less control as the fader controls both paths, so the

same level is going to the mixer and also to the compressor’s side-chain input.

While useful, external side‑chaining is the exception rather than the rule.

Most compression tasks rely on internal side‑chains. The real power often lies not in choosing a different signal entirely, but in shaping the internal side-chain in various ways.

Filtering Much more common than using an external side-chain is applying a filter to the internal side‑chain. High-pass filters are most commonly used, and for good reason: low-frequency elements such as kick drums and basses often dominate the action of compressors, triggering excessive and distracting gain reduction.

For example, if you’re compressing the mix bus and every kick drum causes an audible dip in volume, a high-pass filter on the side‑chain can moderate that trigger.

This lets you focus the compression to react more to the snare or midrange energy, giving a more musical rhythmic response as the movement created by the compressor is focused away from the bass energy.

In Channel Strip, Pro Tools users have access to various filter types including high-pass, low-pass, band-pass and notch.

These can be used to isolate or suppress

specific frequency bands in the detector path. This is how some de-essers work, by making the side‑chain sensitive to energy around 6kHz so that the compressor cuts only when there is loud sibilance. Pro Compressor takes this further, with more precise control of the filters. Both offer a key listen button, indicated by a small speaker icon in the side‑chain section, allowing you to audition the side‑chain signal directly — hugely helpful for tuning by ear.

Even the older Dyn3 plug‑in includes basic high- and low-pass filters, making it more capable than it might first appear. On the other hand, plug‑ins like Smack, which emulates vintage hardware with simpler controls, offers three preset side‑chain filtering options. It’s better suited to coloration than surgical level control. Try using Smack more as a saturation plug‑in than for level control. It’s very good at that.

If you are using a compressor without filters in the side‑chain, you can get the same result with a bit of manual routing.

Staying with Pro Tools’ stock plug‑ins, Impact is a very respectable SSL‑style compressor and doesn’t have a filter in the side-chain (the original hardware SSL one didn’t either). But all you need to do to achieve a filtered side‑chain is set up a signal path that gives you somewhere to insert an EQ. In Screen 1, I’m routing my mix through an aux input (Mix) and then on to a master fader. I’m then sending the Mix to another aux (Impact SC) where I have a high‑pass filter inserted, and have routed that aux on to the Ext Key input in Impact, thereby feeding a filtered version of the

mix to the detector to exclude the bass energy from the compressor action.

Channel Linking Most Pro Tools users work in stereo, and most stock Avid compressors treat stereo signals in a particular way: the External Key input is mono, while the internal side‑chain signal is stereo. But the detector is linked between channels, and responds to the loudest signal present in either the left or right channel. This preserves the stereo image by ensuring both channels are compressed equally. Otherwise, the stereo image would shift when more compression is applied on one side than the other.

This approach mirrors how VCA meters behave in Pro Tools: unlike aux or master fader meters that show the summed level of a group, VCA meters show the peak level of the loudest contributing track. The detectors in Dyn3, Channel Strip and Pro Compressor all work the same way; they react to the highest input level across the stereo field.

Whether of not you want this linked behaviour is a creative choice, and many third-party plug-ins such as FabFilter’s Pro-C 2 offer variable channel linking, allowing users to partially or fully unlink stereo channels.

This isn’t currently available in Avid’s own plug‑ins, but there is a workaround.

If you want to try unlinked compression, instantiate a multi-mono plug‑in instead of a multi‑channel version. If you use the Unlink button in the plug‑in header, the two channels will operate independently.

Screen 2: Most of Pro Tools’ bundled compressors will only work in linked mode on stereo tracks.

If you want to unlink the left and right channel side‑chains, you need to use a multi‑mono instance.

If you maintain identical settings on both channels, you’ll essentially have unlinked stereo (dual mono) compression. To see the different gain reduction amounts on each channel, Option/Alt‑click on the channel selector in the plug‑in header to view both channels in separate windows.

ARA Compression?

Traditional compression using AAX plug‑ins like Dyn3, Channel Strip and Pro Compressor happens in real time, following the principles established in analogue hardware. But Pro Tools has recently opened up new possibilities through ARA integration.

Plug‑ins like NoiseWorks DynAssist or Wavelab Go offer analysis-based level control. Rather than reacting to peaks as they happen, these tools can access the entire audio clip and apply processing accordingly. This allows for highly transparent loudness normalisation or intelligent dynamics shaping. Ideal for mastering or loudness compliance.

With these tools taking care of level management, compressors are freed up for what they do best: adding energy, movement, and tone.

In Pro Tools, plug‑ins like Pro Compressor and Channel Strip give users the power to shape the internal side‑chain. Whether you’re aiming for mix bus glue, vocal consistency, or drum punch, understanding the role of the detector is key.

Compression is ultimately about timing, emphasis and context. And all of that begins with what the compressor hears, which is the side‑chain.

www.soundonsound.com / October 2025

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TECHNIQUE

Ducking your reverbs and delays can bring greater clarity to your vocals.

JOHN WALDEN

A

mbience effects, such as reverb and delay, are an essential part of vocal production in many genres. But slapping on these effects indiscriminately also risks compromising vocal clarity, and even the whole mix: long effect tails can mask or otherwise clash with different sounds, while if you keep the tails too short, then they might not create the sense of space and dimension you’re looking for. In this month’s workshop I’ll explore some ways to unpick this conundrum.

Options In a DAW like Cubase, one way to tackle the issue is to set your effects up on FX Tracks rather than as inserts, and create an automation envelope for the send levels from the vocal track. Use a lower send level in the busier sections of a vocal to keep things sounding cleaner, but allowing more of the effect to be heard between well-spaced words or phrases.

Alternatively, you can use automation of the level of the FX Track — similar, but this time, we’re controlling the ‘return’ from the reverb or delay, not just what’s sent into it. Either tactic can work well and they both give you very precise control, but it can also be a fiddly and incredibly time-consuming approach.

The classic alternative is to use ‘ducking’, whereby a gate or compressor reacts to the vocal to pull the effect’s output level down automatically. Some newer reverb and delay plug-ins — including the reverb and delay modules in Cubase Pro/Artist’s VocalChain plug-in — have this option built in. Another, perhaps less obvious alternative for Cubase Pro users is to use the new Envelope Follower Modulator: you can set this up to control the send level to the FX Track. If you want to try this, be sure to dial in a negative Modulation Depth, so that the send level is lowered when the vocal signal increases!

I want to be inclusive here, though, so I’m going to take you through an approach that any user can employ, from Cubase Elements upwards, and with any reverb or delay plug-in. We’ll start by

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Automated effects ducking can give your vocals a little extra clarity within the mix.

setting up a conventional ducked effect, before refining it to give us finer control over the result.

Line Up Your Ducks The main screenshot shows the key elements of of our basic ducking configuration. You can hear this in action in the audio examples on the SOS website at: https://sosm.ag/cubase-1025. My first step was to create conventional sends from the vocal track to two FX Tracks.

Respectively, these have instances of RoomWorks (for reverb) and MonoDelay (for delay) inserted. You can set the usual send level controls on the vocal track to configure an initial static wet/dry balance.

Next, insert a compressor after RoomWorks, and another after MonoDelay. I’ve used instances of Cubase’s vanilla Compressor plug-in here but any compressor with a side-chain input option will do the trick; we’re not really looking for ‘flavour’ here, but simply

level control. Note that both Compressor instances have their side-chain input activated, and that the vocal track has been selected as the side-chain source. This means that although the compressors are acting on the effects, their gain reduction will only be triggered by the vocal level; in essence, the louder the vocal gets above the compressor threshold, the more the effect will be pulled down.

The final stage involves adjusting the various compressor settings (threshold, ratio, attack and release), the side-chain send levels and the FX Tracks’ own faders. It’s pretty easy to influence when ducking will occur (threshold), the amount of ducking (ratio) and, broadly, the speed with which any ducking is triggered and then recovers (attack and release). A fast attack will pull the reverb out of the way quickly so that your vocal words can be heard clearly. You have more room for creativity on the release.

Pro users can experiment with some frequency-based ducking using Frequency 2.

But in both cases, you don’t want the transition to sound too obvious, so some experimentation may be required to find the best combination for the project in question. The side-chain send level and the FX Track faders both influence the overall wet/dry balance, and so let you finesse how the effects are tucked out of the way of the vocal as it varies in level throughout the performance to (hopefully!) achieve a suitable end result.

Freq Out The classic approach described above might well give you all the control you need. But it might not!

If you pay attention to third-party reverb and delay plug-ins you’ll notice that some newer designs, such as iZotope’s Aurora reverb, do more than simply drop the reverb level — they adopt the kind of spectral balancing algorithms you find in plug-ins like Soundtheory’s Gullfoss, Oeksound’s Soothe, Baby Audio’s Smooth Operator and iZotope’s Clarity. Aurora, for instance, ducks the reverb return more at those frequencies where the vocal is loudest, and adjusts this response in real-time as the vocal’s frequency balance changes throughout the performance. The idea is that the reverb gets out of the way only where it needs to, but the overall sense of the effect can still be heard most of the time.

It’s a neat trick!

While the algorithms behind those plug-ins vary, the concept is rather like using many bands of dynamic EQ — and those bands in turn are similar in concept

For Artist and Elements users, TDR NOVA is a very credible free alternative to Frequency 2.

to frequency-specific compressors, each focusing its efforts on a very narrow band. All of which raises a question: can we achieve something broadly similar if we use a dynamic EQ plug-in instead of a compressor to duck the effect? Yes, we can! Well, sort of... The second screenshot shows an instance of Cubase Pro’s Frequency 2 plug-in configured for such a role, and it could be substituted for the instance of Compressor in either/both of the FX Tracks in our earlier approach.

I’ve set bands 1 and 8 to basic low-cut and high-cut filters, respectively, and these are used to simply shave off

to try and lets you focus your automated reverb (or delay) ducking (or ‘unmasking’ if you prefer that terminology) on the frequency range within which your vocal is sitting at any given instant. Of course, Frequency 2 is only available to Cubase Pro users, but you can use the same approach with any dynamic EQ plug-in that supports external side-chaining. If you don’t currently own such a plug-in, you could always download a suitable free option, and top of my personal list would be Tokyo Dawn Labs’ NOVA. This free download supports all the major plug-in formats and provides a four-band dynamic EQ with side-chain support and additional low‑/high‑pass filters.

The final screenshot shows a configuration that might be used to match what I set up for Frequency 2, and it responds equally well to a side-chain input in Cubase Elements, Artist and Pro.

“Can we use a dynamic EQ plug-in instead of a compressor to duck the effect? Yes, we can!” the low and high ends of the reverb return. I configured the other six bands to operate in their dynamic mode, and centred each on a different frequency.

To keep things simple to follow here, I used identical settings for each band’s Q, gain, ratio, attack and release (shown for band 4 in the screenshot).

I then adjusted the threshold of each band to control the amount of ‘ducking’ generated when the vocal contained energy within that frequency range. At any instant the different bands will apply different amounts of gain reduction — in the screenshot, band 4 (at about 1kHz) is providing a 10dB cut, while band 5 (around 2kHz) is only pulling the sound down by about 2dB. It’s not quite as sophisticated a solution as that found in Aurora, but it’s interesting

Duck Duck Go! You won’t always need to duck your effects tails, and on some more spacious arrangements it might even be counter-productive. But if you’re looking to coax maximum clarity out of a busier mix, ducking can be really beneficial. It’s also possible that you’ll occasionally want so much control that it’s worth making the effort to automate the send and return levels very precisely. But the sort of frequency-specific automated ducking I’ve just described might well do the trick very nicely. Whichever route you choose, Cubase Elements, Artist and Pro have multiple options. www.soundonsound.com / October 2025

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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

N

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.”

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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

L

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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Diagram 4: Adjusting the relative sensitivity of the Sides microphone moves the points where the Mid and Sides polar patterns are equal in amplitude, which defines the edges of the SRA and thus the perceived stereo width of sources placed in front of the M-S array.

stereo-width controls do in mixers).

The first matrix outputs an M-S signal that we route into another matrix that outputs L-R. The stereo channels L and R that we start with become M (=L+R) and S (=L-R). Those M-S signals go into a second matrix to derive L and R again (L=M+S and R= M-S). But look what happens if we substitute those M and S terms in the second matrix for the Left and Right terms that we started out with:

matter, as it’s not difficult to adjust the signal level manually, but usually we need to preserve unity gain and headroom throughout a signal chain. A simple solution is widely (but not universally) adopted: each matrix is designed with 3dB attenuation at both its outputs. That way, when two matrix processes are combined, the overall signal maintains unity gain. If you notice a slight (3dB) loss in level when passing a signal through a single matrix, this will be why!

• Left output = M+S = (L+R)+(L-R) = 2L • Right output = M-S = (L+R)-(L-R) = 2R

Practical Benefits Of Mid-Sides

As you can see in the rightmost part of those two equations, we now have twice as much output signal level as we started with. Put another way, the output of the second matrix is 6dB louder than the original signal. Sometimes that won’t

OK, enough of all the maths! What is M-S actually useful for in practice? The main advantage of the M-S format is that varying the ratio of the Mid and Sides components adjusts the width of the stereo sound stage. With no Sides signal you’re left with pure mono. And as

Diagram 5a: The equivalent L-R pickup pattern for an M-S setup with Mid mics of different polar patterns.

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the Sides level is increased, the stereo width increases with it. Full (100 percent) width is when the Mid and Sides signals are exactly equal in level, and pushing the Sides level higher still creates an ‘out-of-phase’ effect, whereby sounds appear to come from ‘outside’ the speakers, but do so in a very unstable (and, often, arguably unpleasant) way.

The Mid and Sides signals can also be processed independently too, and a very powerful technique is to boost and/or cut the level of the Sides signal using an equaliser. This allows the stereo width to be adjusted in a frequency-selective way. For example, rolling off the low end in the Sides leaves all the bass in the Mid channel only, so it’s all in the centre of the stereo image — this is a standard process for cutting vinyl records. Alternatively, using a high-shelf cut in the Sides channel narrows the stereo image slightly at high frequencies, and that’s exactly how EMI’s Stereosonic Shuffler works (Diagram 2), as mentioned last month in Part 3.

Mid-Sides Stereo Array So far, I’ve discussed creating Mid-Sides signals from left-right stereo using a matrix process. But we can also capture M-S stereo signals directly at source, using a pair of microphones — an idea that Alan Blumlein came up with almost a century ago. Like conventional L-R stereo, two microphones are required,

Diagram 5b: The M-S arrays on the top row give the same SRA (indicated in grey) as the X-Y arrays shown on the bottom row.

TECHNIQUE

MID‑SIDES MIKING

and they must be mounted coincidently, such that they capture only amplitude differences between the two channels (ie. with no time-of-arrival differences).

The Mid signal can be captured using any desired polar pattern that ‘hears’ all the sound sources consistently, and provides a suitable perspective (in terms of the direct sound:reverberation ratio).

Blumlein originally used a figure-8 ribbon mic for this technique, but hypercardioid, cardioid and even omnidirectional patterns are now common options, too.

The Sides mic must reject sound sources at the middle of the stereo sound stage, so must have a polar pattern with a null facing forwards, but good pickup to either side. The perfect candidate for that role is obviously the figure-8 pattern, and the fact that its front and rear lobes inherently have opposite polarities is essential in making the maths work when decoding the M-S signal to L-R — those opposite polarities uniquely identify which sound sources are on the left or right.

Blumlein’s original M-S array comprised a forward-facing figure-8 mic for the Mid channel, and a sideways‑facing figure-8 mic for the Sides channel. Importantly, the face of the Sides mic facing left must have the same polarity as the front of the Mid microphone — in other words, the front of the mic should face left and its rear should face right. If it doesn’t, then the decoded stereo image will be leftright reversed.

Diagram 3 illustrates how the different polarity lobes of the two figure-8 mics add and subtract through the decoder to create a virtual L-R stereo mic array (which, in this case, happens to be a conventional Blumlein array).

A central sound source will only be ‘heard’ by the Mid mic because it’s precisely in the null of the Sides mic.

However, as that sound source moves off to the left, the Sides mic starts to pick it up and the Sides output level rises. At the same time, the Mid mic’s output level falls,

as the source moves progressively off axis — there are pure amplitude differences between the Mid and Sides channels.

Use the M-S equations I set out earlier, and you’ll find that when the sound exists only in one channel (eg. left only), it’s true that Mid=L and Sides=L. In other words, when a sound source is at the extreme edge of the array’s SRA — as far left or right as it can be — the Mid and Sides channels will have equal amplitudes. Put simply: M=S. It follows that if the level of the signal captured by the Sides mic is then increased at or after the preamp stage, then a central sound source won’t need to move as far left or right to generate the same output level in the Sides channel as it does in the Mid mic. In other words, the SRA is reduced! Conversely, if the Sides mic level is reduced, a central sound source needs to move much further out from the centre to generate the same signal level captured on the Mid mic, so the SRA increases.

M-S Matrix Technologies

Diagram 8a (left): A traditional M-S matrix, using a transformer for each channel with a single primary coil and two secondaries. Diagram 8b (above): The electronic equivalent.

Converting from L-R to M-S (or vice versa) can be achieved in a variety of ways, but originally it was performed passively, using transformers. Each channel used a transformer with a single input (primary) winding and two secondary output windings, typically with a ratio of 1.4:1:1. Each of the matrix outputs are derived by wiring a secondary in each transformer in series, either in the same polarity (for the sum output), or in opposite polarities (for the difference output), as shown in Diagram 8. However, this is a bulky and expensive solution, and analogue matrix systems today typically use op-amps to add and

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subtract the input signals (Diagram 8b), usually achieving better frequency and phase responses, too.

The same conversion processes can also be achieved by routing signals around an analogue console using channel pan pots and polarity inversions to add and subtract the Mid and Sides signals en route to the left-right stereo bus outputs (remembering that mixing two signals together when one is polarity-inverted is mathematically the same as subtracting one signal from another). There’s a more detailed explanation of these arrangements in the ‘Decoding M-S Mic Arrays In A Mixing Console’

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box, and I’ve included a couple of examples on the SOS website.

The same console signal routing concepts could be employed within the virtual mixer of a DAW, of course, but why would you when it’s so

much easier to use an M‑S decoder plug-in. Most DAWs include this functionality but, if not, Voxengo’s MSED is free, very easy to use, includes useful stereo metering, and is highly recommended.

Diagram 9: Voxengo’s enduringly useful MSED plug-in.

Diagram 6: Neumann’s Recording Tools app.

Diagram 4 illustrates this using a cardioid Mid microphone with lines drawn from the coincident centre of both mics out to where the Sides and Mid patterns meet. These lines define the edges of the SRA and, as you can see, their position is dependent on the relative sensitivity of the Sides mic.

While it is certainly possible to decode the outputs from an M‑S array to record a normal left-right stereo signal, the real benefit of the M-S format is obtained by recording the M-S signals directly, and decoding them to left-right in post-production, during the mix, when the appropriate stereo image width can be determined, or even adjusted dynamically.

Example 7 in the associated choir audio files was captured using an M-S array and is recorded in the M-S format. The output from a forward-facing cardioid mic (a Sennheiser MKH 40) is on channel 1, and a sideways figure-8 mic (MKH 30) on channel 2. When these tracks are loaded into a stereo track in a DAW, the resulting raw material will sound noticeably left-dominant, because the Mid mic in the left channel is facing the choir whereas the Sides mic in the right channel isn’t. Inserting an M-S decoder plug-in into the stereo track will decode the signal to normal left-right stereo, and produce a stereo image similar to that from the coincident hypercardioids used in Example 5. If the level of the Sides channel is adjusted (whether using the decoder plug-in’s controls, or the right channel’s input gain), the stereo

image width can be varied all the way from mono (phantom centre) up to ‘outside the speakers’.

Equivalences When a figure-8 mic is employed for the Mid channel in an M-S array, the equivalent left and right ‘virtual’ mics are a classic Blumlein array, as already illustrated. But if an omnidirectional mic is used instead, the decoded virtual mics turn out as back-to-back cardioids, with a mutual angle of 180 degrees — a handy format for capturing ambient atmospheres.

More commonly, though, cardioid or hypercardioid patterns are employed for the Mid mic, creating virtual left-right mics which are a little more directional than the real Mid mic’s pattern, and with a mutual angle determined by the ratio of Mid and Sides levels through the decoder (see Diagrams 5a and 5b).

In addition to being able to adjust the SRA during or after the recording, another important advantage of the M-S array is that central sound sources are always directly on-axis to the Mid microphone. The relevance of this is that, as already mentioned, most directional microphones get more directional with increasing frequency.

With conventional left-right coincident arrays, a central source is always inherently off-axis to both mics, and therefore potentially coloured due to the imperfect polar pattern frequency response. This is especially true where very wide mutual angles or highly directional polar patterns are employed. Using the M-S format guarantees that a central source www.soundonsound.com / October 2025

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— which is usually the most important element — is captured with the mic’s ideal on-axis frequency response.

SRA Visualisation Tools When rigging any stereo microphone array, it should be clear by now that two key parameters are paramount: the sound-stage perspective (the direct/reverberation balance), and the width of the sound sources across the stereo image. The first is dependent on the room’s acoustics and the mic array needs to be moved closer or further away from the sound sources to achieve the preferred perspective.

Once that position is known, the stereo recording angle that’s needed in order to encompass all the sound sources can be assessed. But how, then, can a specific mic array setup be found to match that SRA requirement?

When it comes to working out how to achieve a specific SRA, or trying to visualise how changing polar patterns and mutual angles interact, there are a couple of interactive tools that I find particularly helpful (though note that neither have options for M-S arrays).

When working on location, I mostly use the Neumann Recording Tools app (Diagram 6), which I reviewed in SOS August 2018: www.soundonsound.com/ reviews/neumann-recording-tools. It’s free and available for both Android and iOS. This is actually two apps in one, with one section offering a comprehensive gain and signal level calculator for the entire recording chain. The second section is

the part that’s relevant here — it’s an SRA calculator, allowing the user to select and adjust polar patterns, mutual angle, and capsule spacing, with the resulting SRA illustrated graphically, along with indications of the relative proportions of inter-channel level and timing differences.

This app is both accurate and easy to use and, by holding the phone horizontally you can squint across the top of the screen to see if the SRA is adequate for the sound sources in front of the mics.

The second option I’d recommend is a web page on the very comprehensive Seng Piel audio tools website (https:// sengpielaudio.com/HejiaE.htm). Called ‘Visualisation of all Stereo Microphone Systems with Two Microphones’, it is similar to the Neumann app, but has some useful additional features (see Diagram 7). For a start, it has presets for a number of well-established stereo mic arrays, which are selectable from a drop-down box on the left-hand side.

Alternatively, polar pattern, mutual angle and capsule spacing options can be entered manually.

The graphic display shows the selected mic polar patterns in red, and their mutual angle can be adjusted by dragging one of the circles at the top edges of the individual patterns (or by entering a value in the corresponding data box to the right). Similarly, the capsule spacing can be adjusted by dragging the circle in the centre of the mic array (or by entering a value in the data box). The grey outline surrounding the red mics is the equivalent polar

Diagram 7: The Seng Piel audio tools website: a wonderful resource for visualising two-mic stereo arrays.

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pattern if the two channels are summed to mono (so equivalent to the Mid mic), while the blue-grey wedge indicates the SRA for the combination of selected microphone array parameters.

Five coloured bars arced across the SRA region represent notional (orchestral) sound sources in front of the mics, and their spread angle can be adjusted either by dragging the outer bars or entering a value in the Orchestra Angle data box. These virtual sources are mapped between the loudspeakers above, indicating any spatial distortion generated by the particular mic array parameters. The SRA for the displayed mic array is given in another data box on the right, along with the maximum inter-channel amplitude and timing differences, and their contributions to the overall imaging positions.

I find it highly instructive to experiment with both of these applications to help develop a mental picture of the way adjustments to the physical parameters of a stereo mic array impact the stereo recording angle. With practice, adjusting mic array parameters will become instinctive if you need to increase or decrease the mutual angle to compensate for moving the mic closer or further away, for example.

That concludes this month’s instalment. Next time, I’ll explore what happens where the microphones in a stereo mic array aren’t coincident, and some of the popular arrays that deliberately exploit this technique.

Decoding M-S Mic Arrays In A Mixing Console The key to decoding Mid-Sides signals through an analogue mixing console is to keep in mind the core equations: Left = M+S and Right = M-S. The act of mixing signals together is the equivalent of adding them, and while subtraction isn’t a standard feature of a mixer, a little algebraic rearrangement gives us: Right = M+(-S). In practical terms, that minus sign simply indicates a polarity‑inverted signal — in other words, we simply need to mix a polarity-inverted Sides signal with the Mid. With all this in mind, the simplest console setup to decode an M-S mic array to the L-R stereo bus output is to start by setting the Mid channel’s pan to the centre, thus ensuring that it contributes equally to both the left and right outputs. The Sides signal feeds the left output, where it is summed with the Mid — so the Sides channel must be panned fully left. We also need the polarity-inverted Sides signal summed to the Mid at the right output. The classic way to achieve this is to duplicate the Sides channel, put this duplicate on a third channel whose polarity-inversion (erroneously called ‘phase’ on many mixers) button is engaged, and pan it hard-right. The faders of both the original and duplicated Sides channel need to be linked so that both are moved together when adjusting the stereo width.

This arrangement (Diagram 10) works perfectly well, especially with line-level sources. But if decoding an M-S mic array, splitting the Sides mic across two channels isn’t the best idea because it halves the input impedance seen by the mic. The arrangement also occupies three adjacent mixer channels and relies on perfect tracking of the two Sides faders. A better-engineered solution is to connect the Mid and Sides mic outputs to a pair of mono channels in the normal way, with the Mid channel panned to the centre, and the Sides channel panned hard-left, as before. To generate the inverted Sides signal,

use a post-fader aux send and connect the corresponding aux master output to a spare channel somewhere out of the way on the desk. That channel should have its polarity‑inversion button engaged and be hard-panned to the right, as before, but its fader should be set to unity and taped down so it can’t move!

The next stage is simply to match the aux send signal with the Sides channel’s level so that the +S and -S contributions to the left and right outputs are perfectly balanced. To achieve that, set the Sides channel’s aux send and the corresponding aux master to their unity reference levels. Open the Sides channel fader to 0dB, and switch the console monitoring to mono. Now adjust the aux return (-S) channel’s gain for the deepest possible null, indicating a perfect balance of the +S and -S contributions to the stereo L-R mix. Once set, you will not need to touch the aux controls of the -S channel gains again — but don’t forget to deselect the mono monitoring!

This alternative arrangement (Diagram 11) makes it easier to adjust the Sides level without having to worry about ganged faders, and it also presents both mics in the array with standard mic input impedances. In situations where you are using more than one M‑S array, this arrangement also saves channels as each new array just needs two (one for Mid and another for Sides). The aux return channel already set up will also generate the -S contributions from the second array! Just fine-tune the post-fader aux send level from the second Sides input while listening for a mono null.

Diagram 11: A better M-S approach on a console is to connect the Sides mic’s preamp to one console channel, and use a unity-gain post-fader aux send to mult the signal to a unity-gain channel elsewhere on the console.

Diagram 10: The classic way of accommodating a Mid-Sides array on a mixer, with a splitter allowing the Sides mic to feed two channels. It works, but is not ideal!

P8

Eight-channel microphone preamplifer with discrete technology The discrete preamp design is one of the main reasons for the outstanding sound.

With the optional eight-channel ADAT converter, the purely analog P8 can be digitally connected to audio interfaces.

P8 paves the way for high-quality multi-channel recording sessions.

www.spl.audio

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

S

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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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 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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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

R

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

Music mixer Curtis Douglas works out of an impossibly minimalist studio in Phoenix, Arizona.

the movie. You can have maybe one of those songs, because there are only 80 to 90 precious minutes to tell a story.

“My role was working closely with the directors, producer and songwriters to help craft the brief for each song, what they would do, what we needed from them dramaturgically. Spring and I worked closely together to decide what writers should be launched on each song. We ended up with five or six different songs for each moment, and every song had multiple versions, often dozens. It certainly wasn’t clear from day one what team would end up being the writers for the song for particular moments. It took a bit of time to find out which writers work best together, but it was a gift to realise for that EJAE and Mark are this incredible super team that when they joined forces with The Black Label created something that I don’t think has ever been heard before in film and K-pop.”

Story Time The Black Label was founded in 2015 by YG producers Teddy Parks and Kush, in association with YG Entertainment, one of the ‘big four’ companies in K-pop. “The directors, producer and I would meet

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remotely with The Black Label at least once every couple of weeks,” remembers Eisendrath. “We spent a lot of time working through the vibe and feel of the music.

These tracks were sent to EJAE and Mark, and they would add their toplines and

a bunch of melodies and some rough lyrics and concepts. Mark and I were given guidelines of what the song needs to say in the scene, and what the characters were trying to emote. Me being a K-pop person and him being into theatre was a very interesting mix. Our aim was to make K-pop bangers, but lyrically we kept coming back to the storyline. We wanted it to be authentic K-pop, so bringing Korean lyrics into it also was really important to the film‑makers and I. “I write in my small bedroom studio, where I have Ableton, a Universal Audio Apollo Twin, and I love my Neumann TLM 102. I’m also very proud of my vocal chain, which has a gate, a de-esser, EQ, Waves CLA-2A, Valhalla reverb and Waves H-Delay. I know how to vocal produce and all that stuff. So I imported the track, and put down some ideas for the lead melody, and if I have ideas for the background harmonies already, which I usually do, I recorded them as well. I record every idea I have. Once I had a bunch of ideas, Mark and I would meet on Zoom and we dissected each melody, edited it, and he also would have his input. That was our writing process. Once we Frankensteined

Curtis Douglas: “Of course, I’m using my ears all the time, but I look at a spectrum analyser a lot.” lyrics. Then we’d send them back to The Black Label, and discuss what adjustments they needed to make. Since we had so many writers in so many time zones, my role was to be the hub of the songwriting process, to make sure everyone’s connected, and to keep the songwriters up to speed with all of the story and picture development happening at Sony Pictures Animation in LA.” “Ian worked with The Black Label and all these amazing producers to create beautiful tracks,” confirms EJAE. “Their awesome fire beats were sent to Mark and I, and to start with, I would lay down

I will never forget when we first heard the instrumental track. It was the first time the directors, producer and myself were like, ‘That’s it. This is thrilling.’ It was like we’d gotten to a half point of a marathon, and the end was in sight. After that, EJAE and Mark got going on it. When we heard the result, I’ll never forget the joy we felt, having gone through so many versions, ‘This is perfect!’ Plus it sounded like it could be a number one. Of course, it’s incredibly ironic that it actually went to number one!”

the melody to what we thought was great, we worked on the lyrics.”

Black Gold

Rumi At The Top By contrast, EJAE recalls ‘Golden’ as one of the easier songs to write. “I’m not kidding, Photo: John Ward

‘Golden’, the biggest hit from the soundtrack and movie, also plays a central role in the film’s narrative. Eisendrath explains, “Before the song that you heard was written, there were five songs also using the word ‘gold’ or ‘golden’ in the hooks, written by three different teams. None of them quite worked for the film. The directors and myself spent a long time talking about this. This song was particularly hard. It is what’s called an ‘I want song’ in film and theatre. It articulates what the characters are going to run after for the whole film, what they want. And the audience has to get invested in that. In this song, it’s Huntr/x’s desire to dominate the world. It’s a first-half-of-life goal, a very immature, binary, simplistic thing: ‘Kill the demons, become superstars.

We’re good. That’s what we want.’

“At the same time, the audience doesn’t know the characters that well yet, so the song also has to provide some exposition.

We had a breakthrough when we realised that this song is also our opportunity to show the audience a little bit about each of the girls, with them singing about their pasts, and feeling victorious about what’s going to happen. Of course, something is going to knock them over, so that what they want is thrown into disarray. Then it becomes one of those stories where, through facing trials and tribulations, they change. And the thing they want changes.

It’s not what they thought they wanted.

That’s how we get to the finale.

“We asked The Black Label to work on the music and then we pulled in Mark and EJAE to work on the melody and lyric.

that melody came out really naturally, and fast. I wrote it in the car on the way to the dentist! I have a voice memo of that. By contrast, the song ‘How It’s Done’ had 30 demos or so! When I spoke with Mark on Zoom, I said, ‘I have this melody, and I think it’s kind of cool.’ He replied, ‘Oh, this is great,’ and then the lyrics came quite naturally.”

EJAE had, unwittingly, laid a trap for herself in the melody: “I sang the high note an octave lower when creating the melody, but the filmmakers were like, ‘Hey, Rumi needs to belt, and she has to show her range and her vocal abilities.’ I could have gone somewhere else, melodically,

Co-writer Mark Sonnenblick helped shape song lyrics to move the film’s narrative forward.

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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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INSIDE TRACK

KPOP DEMON HUNTERS

own vocal arrangements or a collaboration, like with EJAE, I’ve scored exactly what everyone’s going to sing. Many backing vocals had 30 to 40 different parts, that would be recorded at least tripled, if not more. In the songs that had bigger vocals, we’d bring in an ensemble of 16 singers.

For most of the recordings the directors and producer were beside me giving me feedback, offering insight and clarity about what they needed from each moment of every song.

“Once we’ve captured the audio, Derik and I would comp the vocals. This would be word for word, syllable by syllable.

Some people want straight full takes, but I feel that with this kind of music, it is never going to be the way. We had recorded three to four measures, one phrase at one time, many, many times, and then comped together the best bits. ‘I love that syllable.’

‘I love that word.’ ‘I love that breath.’ We settled on really detailed comps, resulting in hundreds of vocal tracks for each song, which we sent to Curtis, who worked his brilliant magic, editing the comps and all the songs, then mixing the songs in a way that defied our imagination.

The whole thing was like a song production factory that never stopped working until we had to put our tools down. We could have kept going forever.”

Eisendrath’s “song production factory” also hired the legendary Rob Mathes, whose string arrangements were played by a 26-piece string section at AIR Lyndhurst Hall in London, with the ceiling baffles lowered as much as possible, to get “the driest, most immediate sound with more of a pop edge”. Eisendrath conducted and Andy Dudman engineered. All these recordings added up to many hundreds of tracks for each session, which were sent to Curtis Douglas, who worked on them for close to two years, off and on, at his studio in Phoenix, Arizona.

Douglas’ studio is extremely minimalist, because “I want nothing in it that can distract me. I’ve gone to extreme lengths to get the room as symmetrical as possible. I even brought in lasers to perfect the azimuth and elevation of my monitors. I did most of the acoustic design myself, with help from a company called Music City Acoustics. I added a Trinnov Nova room correction unit as the icing on the cake. The monitors I use are Amphion Two15s, which I love and couldn’t live without, and my interface is the RME Fireface UCX II with the ARC controller. It’s the best interface, in the sense that I never think about it. I never touch it. It just works.”

The same attention to detail also informs his work, particularly on the gargantuan KPop Demon Hunters project.

“When I first got involved, two years ago, Ian asked me what mic, pres, and compressor I thought would be best for this project. I suggested the Telefunken

“The vocal editing alone on ‘Your Idol’ and ‘Soda Pop’ took me a week per song, before I even touched the mix. On ‘Your Idol’, there are 16-piece choirs with each singer tracked separately, so I had the control to edit every singer for each take. I would nudge them to be really tight timing-wise, then pitch-correct each vocalist separately. With that many backing vocals, you have esses and tees all over the place, competing with each other and with the lead vocal, so I would dramatically reduce or cut out those consonant sounds in the backing vocals. After weeks of questioning my sanity, hearing the vocals edited in such detail made the time and effort spent well worth it! The vocal editing honestly took just as long as the mixes.”

Tune, Tune, Tune Again “In terms of tuning, on a lead vocal, I’m very particular. Often I’ll tune each part three times using Melodyne, Auto-Tune and Waves Tune, and I then make the final comp from those.

They each work and sound slightly different.

I also manually tuned and time-corrected all the backing vocals using either Auto-Tune Graphic Mode or Melodyne. Luckily, the approach to tuning for this project matches my personal taste, in the sense that I like vocals to be in tune, but I don’t want to hear any obvious tuning or have retune speeds too high. My goal is to make tuning as invisible as possible, but also make sure that things are very, very in tune.

“After finishing the vocal editing, I’d do a first pass of the entire mix and send that to Ian and the team for feedback. Ian and I would then get into the details of the vocal mix, fine-tuning automation on harmonies, vocal arrangement dynamics, trying to get one syllable of the backing vocals to pop, and so on. When dealing with multiple vocalists in each song, which is typical in K-pop, the best approach for me was to not think of them as separate vocalists, but as one singer with different personalities. This mindset helps me craft a cohesive lead vocal throughout the song. If one singer was much brighter or darker than another, I’d have to tame down the bright singer, or brighten up the darker singer more than I normally would to make them feel right with the others.

“In addition to the hundreds of vocal tracks, when it came to mixing, I’d have

EJAE: “Whether it’s chord or melodic changes throughout the songs, or tons of stacked vocals, K-pop is very maximalist.”

Perfect Symmetry A Phoenix native, Curtis Douglas spent 15 years recording and producing rock and indie bands. Five years ago, he made the switch to mixing pop music, with remarkable success. ‘Golden’ is his sixth Billboard number one. Douglas has worked with the likes of John Legend, Steve Aoki and Snoop Dogg, and enjoyed number one hits with K-pop acts Twice and Stray Kids.

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ELA M 251, going into a Neve 1073, into an 1176-type compressor. Derik ended up using a Serpent Splice 500 compressor for most of the vocal tracking, which is one of my favourites. The 251 is the Holy Grail for pop music, if you can find a nice one. We also decided to use a Neumann U87 as a failsafe. The 87 generally went into an API or a cleaner mic pre, without compression.

Eighty percent of the vocals were cut on those two mics. Multiple times, the U87 saved me, although it was laborious to have two mics for hundreds of vocal tracks on certain songs!

“I would often receive a lead vocal — and this is not an exaggeration — with 100 takes or more for each part.

On a song like ‘Golden’, there were only 30 vocal tracks, but ‘Soda Pop’ and ‘Your Idol’ each had over 300. In general, it was my job to clean up the comps, look at alternative takes where needed, pitch‑correct and time‑correct, or ‘pocket’ all the vocals — really whatever it took to get them feeling and sounding good.

Ian was involved in every step of the process, via Zoom and Audiomovers.

He’s an exceptional talent.

300-plus tracks of drums, bass, synths and other instruments. The mix sessions for ‘Your Idol’ and ‘Soda Pop’ were over 500 tracks in total! ‘Golden’ and ‘Free’ were the two most conventional mixes that came across my desk. With ‘Golden’, being a straightforward pop song, the mix really came together in just a few hours.

“My mix template also made consistency from song to song much easier. I’ve really refined it over the years.

The very first thing I do is make sure that, with everything off the master bus, the entire mix is hitting at around -10dBFS. When I then instantiate my master bus, it’s already hitting each plug-in, particularly any compression/limiting, in the sweet spot.

From there, it’s almost immediately obvious whether there’s enough or not enough low end, and whether vocals are too loud or quiet. I have a system of checks and balances to ensure consistency from mix to mix. I use the same master bus chain on every song, and being so comfortable with the chain makes it easy to tweak each plug-in for the desired result. That said, I will go through upwards of eight different

limiters on every single song, just to find the one that feels right to me.”

Compare & Contrast “Generally, when I start a mix, my main focus is on kick, snare, bass and lead vocal.

If I can get those feeling good, I’m headed in the right direction. Then I’ll spend the rest of the day on all the other stuff. But really, once you have the kick, snare, bass and the lead vocal right, a mix feels right.

Everything else is a cherry on top. I don’t think of mixing K-pop any differently than mixing American pop. I think of the songs as what they are — big pop tunes. My main goal is always ‘How can I make this sound really big, futuristic, larger than life? How can I make everything extra, everything a little brighter, a little heavier, with a little bit more low end?’

“To achieve that, referencing is an important part of my workflow. Referencing is a really underappreciated aspect of mixing. I use the ADPTR Audio Metric AB plug-in and also look at a spectrum analyser in real time to see how my mixes compare to others and find ways to push

mine beyond them. How close am I to my brightest reference, or my darkest reference? For example, a song like ‘Soda Pop’ needed to be really bright, so I compared it heavily to the brighter mixes in my reference library.

“Of course, I’m using my ears all the time, but I look at a spectrum analyser a lot.

Where is my sub relative to other songs?

I pay close attention to headroom. If I’m not able to get a song as loud as I want it, where am I losing headroom? Often, there’s too much unnecessary low end somewhere, so I EQ low end out of every track that doesn’t need it. Most of all, there’s tons of side-chaining. The amount of sidechaining that went on in these KPDH mixes was enormous. Every time the kick hits, the 808, the bass and the low end of my entire instrument bus are ducked. Every time the lead vocal hits, I duck a little bit of the mid/ centre of my instrument and bass busses.

I’m using the Wavesfactory Track Spacer for that. It creates micro‑movements in the mix that keep it exciting while making room for everything. There’s never a time something isn’t being ducked to some degree.

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Curtis Douglas: Plug-in Choices Instrument bus “I used the Waves Abbey Road TG Mastering plug-in in this case only for the Spreader. I wanted to get all the instruments out of the way of the lead vocals. I find this widener to be exceptional.

I have 10 or more stereo wideners, but this one seems to do what I want it to without any negative effects. I often get pretty tweaky with the FabFilter Pro-Q 3. It feels like home to me, and it’s easy to make moves in Mid-Sides, which I do a lot.

I wanted a little bit more midrange in the sides, so I added a little bump at 1.2kHz, and took out a lot of mud in the middle around 200Hz to make room for the bass.”

Kick drum “I love the [Wavesfactory] Spectre plug-in. I dialled in some punchy upper mids, so the kick could be heard on systems that don’t have tons of low-end bandwidth. The Waves CLA-76 is on my Curtis Kick Tighten preset, to tighten up the low end of the kick. I increased the mix until the kick felt as tight as I wanted it to. There’s also some Pro-Q 3 going on, not too much, just getting rid of some of those honkier upper mid frequencies. Last, I wanted a little bit more sub at 45Hz just to make the kick feel more weighty.”

Master bus “The mix knob on the Plugin Alliance Brainworx Black Box is pretty low. I use this plug-in to increase perceived loudness without lowering

Instrument bus.

Master bus. headroom. It’s on nearly every mix I do, and I mix into it early on. The Submission Audio Flatline 2 is a hard clipper. I had it right before my final limiter to catch any stray peaks. Sometimes it’s doing nothing, and other times it’s knocking off

three or four dB. I didn’t want random bursts of energy to hit the limiter, because they can cause unpredictable moments of distortion and make the limiter momentarily work too hard. The clipper gives the limiter a more consistent level. It also has

Curtis Douglas’ snare processing chain.

“Ian would regularly check in with the directors and come back with their combined feedback. Like in ‘Your Idol’, they wanted the vocals to sound more demonic, so I experimented with five or six different ways to formant‑shift the lead vocals and the choir. Do you shift every single person individually, or do you shift the entire choir?

What sounds better? When you’re dealing with 200 tracks, sometimes just rendering them to listen back can take quite some time. I tried Melodyne, Auto-Tune, zplane Elastique Pitch, Soundtoys Little AlterBoy, just to see what sounded coolest. I also

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mixed in distortion and a gothic-sounding reverb with the formant‑shifting.”

Hitting Harder Right at the end of the music-making process, when re-recording mixer Michael Babcock blended all the audio for the movie, the team took another unusual step to maximise impact. Ian Eisendrath explains: “Normally, you first mix the film versions, with all their quirks and visual references, and then you go back and mix the soundtrack versions.

For this, because the directive was to

have really hard-hitting K-pop tracks, we did the soundtrack mixes first and then pulled them into the film and made the necessary adjustments.

“When everything went to the re-recording mixer, who was also working in Atmos, there was a lot of concern from all of us about not losing how hard these songs are hitting once they were brought into a theatrical space. Ultimately, a crazy thing ended up happening. We realised that there’s no way to deliver 20 to 50 mix stems, as is normal, because of how Curtis had mixed and mastered.

a handy delta button that allows you to hear the distortion that it’s adding in solo.

“The iZotope Ozone Maximizer is set to IRC IV Modern. Oddly enough, this is the only song on the album that ended up with the Modern algorithm. The UAD SSL G Bus Compressor has an attack of 0.1ms. If I were to put that up to say, 30ms, the low end would get much looser, as more lows would be sneaking through to hit the limiter. I took quite a lot of time choosing the attack time of the SSL, and almost always ended up in one of the fastest three attack times. The attack time, in combination with the amount of side-chain filtering, helped shape the low end.

The higher the side-chain filter, the more low end is getting through to the limiter, which ends up making it feel looser.”

Vocal reverb “This one was interesting. I had something very specific in my head and experimented until I got there.

First, my usual de-esser, set fairly aggressively, into the Lexicon 480L, with a mix of 25 percent into the main vocal verb: the LiquidSonics Verb Suite Classics, fully wet. It’s the same verb I used on the delay. Then a fairly intense EQ, because I really didn’t want low end in the reverb, or any of the mids clouding up the fundamental vocal frequencies.

I wanted the verb to be heard, but not get in the way.”

The vocal reverb chain features two reverb plug-ins in series.

Snare drum “The Baby Audio I Heart NY is a parallel compression plug-in. Spank is how much it’s compressing, and the Parallel Signal slider is the blend between the original and the Spanked signal. I didn’t blend in too much, but it made the snare feel as if it was being hit a bit harder.

Spectre added a bit of harmonic content to the upper mids, trying to add some weight.

I wanted more low end in the snare, so I added some with the Pro-Q 3. The 4kHz area was a little harsh, so I took some of that frequency out. Then I used the Kilohearts Transient Shaper to make the snare feel longer, and fairly significantly so.”

Vocal delay “I generally put a de-esser as the first insert on every delay or reverb because I don’t want consonant sounds flying all around the mix.

I want the verb and the delay on the meat of the vocal. I had one side of the H-Delay phase inverted, so it pushed the delays to the outside of the speakers. The EQ, post-delay, took out some midrange. One thing I did to help homogenise the delay with the main reverb in songs is put the main reverb on the delay as well, just with a fairly low mix, 12 percent in this case. The LiquidSonics is one of the verbs I used as the main vocal verb.”

There’s too much side-chaining and dynamic EQ and every track reacting to every other track, which is what gives the songs their glue and makes them sound like music today. So we ended up delivering just an instrumental stem, a bass stem, a drums stem, and a vocals stem. They were all pre-mastered, and Curtis’s mastering chain for each song was then run during the mix stage. I don’t know if that’s been done before, but I really think that helped the songs in the movie feel and sound as vibrant and hard-hitting as a record.”

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FE ATURE

Photo: Wyatt Sardy

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track/4gDGqWJ5fkAHRbCVpQzsWP

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

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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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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RAK’s vintage API console was the main source of mic preamps and EQ for tracking.

On Talk

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Photo: Matty Deveson

A universal challenge when running large sessions is for the engineer to notice and hear comments made by musicians between takes. It’s not hard to create a separate cue mix for listenback or ‘reverse talkback’, but routing that to the main studio monitors can cause problems, as Isabel Gracefield explains. “You often have that experience of turning it up to hear them talking, and then they’ll start playing, and suddenly you’ve got this really intense drum sound that isn’t the drum sound. And if you’ve got someone from the band in the control room, they’re like ‘Why does it sound like that?’ It can become quite a stressful thing to look after.”

Gracefield’s solution is simple and effective. “I set up one of those Fostex mains-powered speakers [6301NB] that has a jack input next to my rig, so I could route any signal from Pro Tools to that Fostex speaker. And the idea was that talkback did not come out of the PMCs in the control room. It came out of this speaker, and I could just switch it on and off with a manual switch when I wanted to hear people. I needed a way to be able to hear the band at any time, but in a distinctly different world than on the main speakers. This separate speaker just solves it, because everyone understands that’s a different signal path. You just set up a Pro Tools send that’s labelled talkback, and then at any moment I can use that person’s mic to hear them talking to me.”

this case I just reserved two channels for my output, and then 46 channels going to Pro Tools on the preamps. It’s a very simple desk. It’s just mic pres and EQ. Every channel’s got either a 560 or 550, which is incredible. And in general, I drive the preamps quite hot, because they just sound really good hot.”

Additional signals were tracked through standalone preamps, including the studio’s unique ‘traffic light’ units that were built by a former tech to match those of the API console. The complexity of the session necessitated running cue sends for the musicians from Pro Tools rather than the mixer.

“I think the biggest aspect of this session, in a lot of ways, was headphones,” says Gracefield. “We’ve got 16-channel Aviom [personal headphone mixers] and we were using all 16 channels. We had one stereo channel,

which was Mulatu on vibraphone, because he’d been really clear that he wanted that stereo picture, and then everyone else was in mono. While we were setting up, we would keep going out onto the floor and putting their headphones on and adjusting their mixes. I had an assistant, Sam, in the control room with me, and then we had another assistant, Adele, who had a set of headphones that were a copy of Mulatu’s mix, and she was in there 24/7 with him, constantly adjusting the mix. And one of the key things was that his stereo field matched what was in the room behind him, so he could always just turn his head and know what was going on. And that was a really tough job, because his level on his headphones was just the loudest thing you’ve ever heard. But it actually became this really beautiful relationship between them. He’d start playing one

playing the piano, and it was like, ‘Are we in record?’, and we just caught it. And as far as I understand, there’s no piano recording of him playing like that, and this Tizita is an incredibly important cultural piece that we just managed to catch. And then everyone’s sort of creeping in the background of the live room to do their bits with him, a little bit of saxophone and cello. I think he played for half an hour. Just absolute magic. It’s quite a big thing to get him comfortable enough to play in that way and feel relaxed enough to do it, and a lot of that was to do with Adele’s work on the headphones.”

In The Cans

Isabel Gracefield’s hand-drawn sketch showing the planned layout of RAK Studio 1 for the sessions.

instrument and then with no warning move to another instrument, and so she could just move a microphone in.

“There’s a piece we recorded called the Tizita. There was one lunchtime where he just went in and he started

“For this project, I mixed in the box,” says Isabel. “I did the initial month of mixing on my own, 80 percent on headphones.

I’ve got Genelecs at home so I was checking with those, and then we had three or four days attended in RAK Studio 4. Since lockdown, I’ve been mixing on headphones so much that as long as I know that I’m going to go into a space

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Photo: Alexis Maryon

Mulatu Plays Mulatu was produced

by Dexter Story, who also flew to Ethiopia to record overdubs by local musicians.

where I can hear everything on really good speakers, I’m happy to defer some decisions until that point. I also use a little Bose Mini SoundLink because if you’ve got too much low end or midrange, it really shuts down and sounds crappy. In headphones you can have way too much midrange detail, so I use the SoundLink to counteract the headphones and then just check it on my Genelecs to make sure nothing crazy is happening.”

The main decisions that were deferred until the attended sessions in RAK Studio 4 related to the low end.

“Studio 4 in RAK has Genelecs with two subs, so I really get to see what’s going on. A lot of the low end got carved away in that environment. On this project, the mixes are 90 percent EQs, very little compression. if it needs something, I’ll ride it manually. I wanted the record to feel soft, so there’s a lot of pulling out midrange resonances on things to just soften all the time, and I think my master bus was an SSL compressor just barely moving, and one limiter, and then again a bit more corrective EQ.” “I had to do a lot of really careful riding of the piano mics, because there were a lot of places where the piano is quiet.

Where there’d be a specific movement

Adventures In Addis “In rehearsal, we had an Ethiopian band and his European band, and the original intention was to record it all at once,” says producer Dexter Story. “Ethiopian musicians have a different relationship to rhythm and pulse and energy, and those two approaches confronted each other at that rehearsal.

We only had one day of rehearsal, and it was not going to happen in one day. Even Mulatu said ‘We need to use musicians from Ethiopia.’ So, we’re looking at each other like: somebody’s going to Ethiopia to record...”

A plan was quickly hatched for Story to fly to Addis Ababa after the RAK sessions and record local players at Mulatu Astatke’s own club. “Mulatu is very sensitive about who’s in the room when he records. He has a jazz club, and I knew I had to come with the recording gear, because he did not want to go to a studio.”

Things started badly when Dexter’s entire recording rig was confiscated by customs officials,

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who demanded $15,000 for its release, and only relented when Astatke himself intervened.

“I brought a small interface, the [Universal Audio] Apollo x4, which allowed me four ins. It has two headphone outs as well, and two quarter-inch jacks in the front if I need them, because a lot of those instruments they play from their pickups now. It allows me two monitor outs, so I was able to play through the house PA and also play through some small monitors. I brought a little two-channel Rupert Neve Designs 5211 mic preamp. I took my Neumann U87, my AudioTechnica 3035, and some small stereo Sennheisers to get a stereo impression on drums. I took my Sennheiser MD421, that worked really well, and then I took a couple of [Shure] SM57s, of course.”

The next challenge was to fit the recording sessions into the club’s busy schedule. “I took a fixed amount of cash with me, and I don’t have access to

October 2025 / www.soundonsound.com

keeping a musician all week long. I have one musician a day, one musician per afternoon or per morning to capture all that.

So, I had to grid out where these musicians are going to be on songs and make sure I can capture them for what Mulatu envisions them doing, and I had to be clear with him: ‘Hey, are these the only pieces you want him on?’”

Recording the instruments themselves also required some lateral thinking, especially as the jazz club was anything but a quiet and peaceful environment. “The instruments that we recorded are the traditional instruments of Ethiopia: the massenqo, which is the bowed lute; the krar, which is the five‑ or six-string lyre; and the washint. The washint is the end‑blown flute; they blow it with the side of their mouth. It’s one tonality, but pentatonic, so the musician has to bring 24 of them!

The begena is the 10-string upright plucked lyre. It’s only used in the church. Then there’s the two-sided

drum called the kebero, both the religious one, which is huge, as well as the kebero drum set, which is more of a one-sided set with about four drums, sometimes five, which are made with hardware that sits them above the ground about two feet, and they put mics right inside of the tub, right up against the head, to get the sound.

The larger one is used as more of a kick drum, but it’s played with the hand, and then the smaller ones are kind of used like toms and snares, like high-pitched darbuka kind of sounds, so they got a spectrum of tonalities and pitches.

That’s played like a drum set but with no kick. And we recorded traditional handclaps. Ethiopians have a particular, intentional way of clapping on their own music.

“The massenqo and the krar have now been electrified, thanks to someone that I study named Ateweberhan Segid, an Eritrean man, and Tewolde Redda.

These two geniuses electrified the krar, so now that’s the modus

or note I’d just let the piano come up, and then it would have to pull straight back down, because otherwise you get this big roomy drum sound and then you lose all the detail on the close mics.

Cello and percussion were also a tricky combination. In the mixes the cello mic is only open when it absolutely has to be, so that I think if you concentrate you can hear during cello solos that suddenly the percussion sounds roomier, but hopefully you’re so drawn to the cello that you’re not thinking about it!”

The other major challenge at the mix was integrating the overdubs recorded

by Dexter Story in Ethiopia. “There was an instrument called the begena, which is very buzzy and aggressive-sounding, but it’s got a huge cultural importance.

Mulatu really wanted it to run all the way through the track, but we were all finding ‘We’re not sure this is listenable!’

And that was a really complex thing to handle, because it’s so important to the label to honour his culture and his history, but also the audience that they’re selling to is primarily listening with Western ears and is maybe not capable of listening to eight minutes of begena 8dB louder than everything else.

And so we really had to think about how

operandi, as they say, and they like to go direct. I tried miking them as well, but what worked were actually the direct ins.

“I had to put four mics on that drum set.

I had two mics spatial on top so I would catch the little snare, high pitch, and then two up underneath it then catching the lower tom tones. And then for the big drum, I had to put it by her hand. I wanted to catch the slap of the hand, but then also caught the room a little bit. With the washint, I put stereo mics above him, but I caught something with the Neumann U87 catching him right below where I saw the wind coming out, so tried to catch the flute in that way.

“Fortunately I knew the session enough to know how Isabel records. I watched her and her people and then tried to do the same. I knew I had a big job to do, and once I knew I had takes, then I was able to kind of package it up. I took it back home. I did some of the edits and stuff while I was there. But for the most part I didn’t put many effects on the instruments; I just kind of placed them within a Pro Tools session and shipped them to her, ’cause I still wanted her to have control of mixing her record.”

to mix the cultural instruments for our ears while still keeping Mulatu’s vision intact. So that was a really complex, sensitive area of the record, and I think they’ve managed to find that line in the final masters where you do feel the cultural instruments, but you’re not overwhelmed by that aspect of it.”

Make It Personal Mulatu Plays Mulatu isn’t only a great-sounding album. It’s a project that illustrates how the role of the engineer extends beyond purely technical realms, into enabling the artist to deliver performances that are worth capturing in the first place. “This is a group of people who’ve been playing

together for a long time, so there’s a really deep interpersonal history,” reflects Isabel Gracefield. “You’ve got to get inside all of that and make sure that everyone feels really looked after and respected in that space, and that their wishes are being met. And I think in all my years of doing this, this was the session that was most about that interpersonal care: making sure we had extra assistants who were really carefully briefed and able to look after everyone, and sending people out on the floor to check on people, and managing talkback.

You can’t allow there to be one person in the band who doesn’t feel like their instrument has been given the time and respect that it needs.”

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Nine Angles For Forte Spat Curve Kontakt Instrument

HHHH Dolby Atmos has undoubtedly generated a new surge of interest in multi‑channel music creation, and that has been further encouraged by a number of the mainstream DAWs adding support for the format. However, while a number of specialised effects plug-ins (for example, LiquidSonics’ Cinematic Rooms Professional Atmos-ready reverb) have appeared, items for a dedicated Atmos/spatial audio recording/mixing toolkit are fairly thin on the ground. All of which makes Nine Angles For Forte’s Spat Curve — described as a 3D sample-based instrument collection — rather intriguing.

Spat Curve is designed for Kontakt (full or Player version) but was created from the ground up to support — and exploit — the spatial audio opportunities provided

by Atmos/Ambisonics in both a studio/ recording context and a live performance context. The library provides 16 different individual instruments (more on these in a minute) but each is constructed so that, when played, its performance dynamics include spatial components, with both the instruments and the effect elements of the sounds making full use of the spatial positioning that Atmos, Ambisonics and other multi-channel playback system provide; these are true ‘3D instruments’.

The source instruments are an eclectic selection. For example, they include ensembles for bassoon, French horn, trombone, marimba and guitar, a 3D acoustic drum kit (this is particularly impressive), a collection of four synth-based sounds (two basses and two leads), several different whistles and a stick-based percussion instrument. Each has been programmed with different spatial responses. For example, within the drum kit, two versions are provided for some kit elements. The first maintains a conventional front/left/right location but the second is spatially dynamic. For both snare and hi-hat, that involves changing their spatial position on each hit letting you create ‘spatial rhythms’. The stick-based percussion instrument is similar in this regard. All the instruments also feature spatially based effects but with full control over the wet/ dry balance for flexibility. Spatial dynamics aside, there is a creative edge to the sound design, and all the instruments

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are very playable and responsive.

Spat Curve is undoubtedly both unusual and novel.

Indeed, I think ‘experimental’ is also an appropriate description, and Mexico-based Nine Angles For Forte won financial support for the development process. Obviously this is a niche product and, while it is not cheap, I suspect it will be of most interest to media composers who have a suitable multi-speaker monitoring system. That said, it translates well enough to binaural (which is how I did my own testing). It is also worth noting that configuring both Kontakt and your DAW to fully exploit the multi-channel nature of the instruments is quite a technical process. The documentation and tutorial video are essential references on this front.

The website also has session templates for Pro Tools and Reaper, as well as output CFG files for Kontakt to get you started.

Hopefully, similar templates might be offered for other DAWs at some stage.

Is Spat Curve the first of a new frontier of virtual instruments designed specifically for composing in spatial audio? Maybe so.

Watch this space... John Walden $400 www.naff.mx www.native-instruments.com

Sonora Cinematic Softstruck Kontakt Instrument

HHHH Sonora Cinematic already offer a felt piano instrument and now they’ve applied the same approach to guitar. Felt piano works by placing felt between the hammer and strings to produce a soft, muted sound, and this library takes the guitar in the same direction using commercially available felt guitar picks normally used with nylon string instruments. This produces a soft tone that lends itself well to further processing.

Softstruck runs on both the free Player and the full version of NI’s Kontakt.

The recording sessions used three electric guitars: a Butterscotch Telecaster with an added middle pickup, a Les Paul with humbuckers, and a semi-acoustic Duisenberg with a P90‑style pickup, used

for the flatwound string samples. The user interface is intuitive with a simple layout and enough control to allow for sound‑shaping without getting bogged down with complexities. There’s blending between four sound sources and the controls can be automated for creating evolving textures.

On opening the library in Kontakt you’ll see three Softstruck instruments: Butterscotch, Flatwounds and Single Cut. From these are derived a number of presets created by blending four faders designated Amp 1, Amp 2 and two pedalboard chains, Board A and Board B. Board A seems dedicated to fluttery, ambient delays while Board B creates a reverse delay type of effect. Two further controls add Reverb and Atmo, the latter having two switchable modes that add either a fluttery delay-meets-reverb ambient effect or a lush granular type of reverb. There are also five different master bus effects: 1969, Ensemble, Gentle Move, Miroirs and Memoria. These add subtle but useful changes to the overall sound. Ensemble creates an almost honky-tonk piano sound while Miroirs adds a reversed delay and 1969 a delay that sounds like a tape emulation. Gentle Move adds a thickening effect and there’s also a dedicated double‑track button at the bottom right of the window. Memoria seems to add an interesting ambience to the sound.

As you might expect, the raw sounds are all clean plucks influenced by the use of a felt pick, giving them something of an electric piano, toy piano or music box character.

Once you start blending the sounds it turns out there’s quite a lot of scope for creating different pluck, pad and underscore sounds.

Board B combined with Atmo adds a lot to the overall texture, and keeping Board B set high and the other faders low creates useful long-decay pad sounds with a softer attack.

Conversely you can emphasise the attack by bringing in more of the amp sounds. The sounds also lend themselves well to further processing using your existing plug-ins, from rotary speakers to lo-fi wobble.

While simple in concept, the sounds produced by Softstruck are very usable across a range of genres, with enough

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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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Q

What batteries should I put in my CAD Equitek E-200 mic?

I have a CAD Equitek mic that I bought second hand a while back, and I love it. It’s the multi-pattern E-200 one, and I’ve been using it happily for several years. Recently it stopped working properly, becoming increasingly distorted shortly after powering on.

Since it uses phantom power, I hadn’t realised until I downloaded the manual that it also needs internal batteries to work properly! It’s an old mic, and the manual says I should put rechargeable NiCad batteries in there. But these were banned here in the UK several years ago [indeed, they’re now heavily restricted in many territories — Ed.], and I can’t find them anywhere! What should I put in the mic? Is it a write off, or is it safe to use a different type of battery?

James Thompson via email Matt Houghton, Reviews Editor The short answer is that you can safely put rechargeable NiMH batteries in your mic. It will work just fine. But there’s a slightly longer explanation that might give you reassurance and stop some head‑scratching by other E-200 owners!

Like you, I have an Equitek E-200 that I bought used.

I can definitely appreciate why this mic appeals to you, and certainly wouldn’t want to write it off! In the cardioid mode, as well as conveying the usual detail you get from a capacitor mic, I find that this one exhibits a really smooth and classy character that makes it great for my own vocals. It’s something that I’ve always put down to the fact that there’s not the significant top-end lift you find on some vocal mics, and a dip in the response, centred around 6-7 kHz, seems to tame harshness. That, in turn, tends to make dynamics processing more forgiving, and I find this mic can be really handy if I want to track vocals through a hardware compressor. It might not be the perfect match for every voice, but for some it sounds great.

Anyway, back to the question. Earlier this year, my mic also started to exhibit the sort of problems you describe, and

I tracked this down to failing batteries.

The date printed on those batteries indicated that they’d been in my mic since at least the early ’00s — over two decades! — and this makes me think that quite a few unsuspecting ‘second owners’ could be running into this issue around now, if they haven’t already.

As you’ve probably already gleaned, the mic employs a pair of rechargeable 9V batteries to supplement the phantom power supply, a sophisticated arrangement that makes more current available for its internal amplifiers. When not in use, the mic has internal circuitry, powered by phantom power, to trickle-charge those batteries. The manual specifies NiCad (nickel-cadmium) batteries and, like you, I discovered that these are simply not available to buy today, due to 2016 legislation restricting the use of cadmium in the EU and UK. A little digging revealed that the more modern NiMH, or nickel-metal hydride, batteries are available in the same physical form factor, and that these can often be simple drop-in replacements for NiCad ones. But I also learned that while you can keep trickle-charging NiCad batteries when they’re fully charged, that’s usually not the case for NiMH ones.

They are more ‘fussy’ about the charging circuitry and can, if overcharged, overheat.

Not knowing the details of the mic’s charging circuitry, and fearing the worst, I emailed CAD’s customer support for advice.

They confirmed (almost instantly — great support for a nearly 30-year-old product!) that the Equitek E-200’s trickle-charging circuitry is compatible with rechargeable NiMH batteries, and that these are, in fact, preferred to NiCad ones. So I bought and charged a couple, and popped them inside. And my mic has performed just as well as it always did ever since.

“I can definitely appreciate why this mic appeals to you, and certainly wouldn’t want to write it off!”

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The Equitek E-200 capacitor mic uses (and charges) two rechargeable 9V batteries as a power reservoir that makes more current available for its built-in amps than is available over standard 48V phantom power. It’s an old mic and though NiCad batteries were originally specified, NiMH types will work just fine.

AWARDS The Best New Products Of The Year, chosen by the readers of Sound On Sound Voting will soon commence for the 16th annual SOS Awards, and will continue until the end of November 2025 at www.sosawards.com. The results will then be compiled, ready for announcement during January 2026.

Each category consists of a shortlist of nominations, chosen by the SOS editorial team, and we’d like you to tell us what you think are the outstanding products in each of the groups. As always, you are not required to vote in every category — if you don’t have any strong opinions on some of the product groups, there’s no need to vote for anything in those categories.

The categories are: Audio Interface DAW Effects & Processing Hardware Guitar & Bass Technology Software Plug-in Music Software Performance Controller Keyboard & Synth Drum Machine, Sampler & Sequencer Microphone Mixer & Mixing Controller Monitor Hardware Recorder Mic Preamp Software Instrument Studio Headphones & IEMs Live Sound Product To be nominated for an SOS Award a product has to have been on sale, or tested and reviewed by us, in the 12 months prior to the voting period. This year’s nominations can be viewed at the URL below until the end of November 2025, and we very much look forward to seeing your choices for all the best new products of the last year in music technology and recording.

www.sos awar ds .c om (voting closes 30th November 2025)

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153

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

T

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.”

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October 2025 / www.soundonsound.com

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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