SOUND ON SOUND [USA] April 2025

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LE ADER

OUTBOARD BOUND I

n the early years of this century, it seemed as though a different major studio was closing down every week. It was also a lean time for the manufacturers of high-end mixers and rackmounting gear. The ‘project studios’ that were pushing aside traditional recording spaces didn’t need or want large-format consoles, or racks of hardware effects.

The expectation in many quarters was that this trend would continue more or less indefinitely.

Big studios would go the way of the dinosaur, and software would eventually replace outboard completely. Yet, strangely, neither of those things has happened. Having reached rock bottom around 2010 or so, the market for commercial studios has bounced back, partly thanks to new business models that are turning the recording process into an Instagrammable experience. And although software is now at the heart of almost every studio, it’s no longer displacing hardware.

What’s changed, though, is that today’s hardware purchases are driven by choice rather than necessity. That means the heart plays more of a role in guiding them. Where we might once have had racks full of noise gates and workmanlike compressors, we now have a small collection of items that we’ve assembled because we love them. And we use them not instead of software, but alongside software.

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The stumbling block in setting up this sort of ‘hybrid’ arrangement has always been the interface between the two worlds. The convenience of instant recall and automation that DAWs offer tends to fall to bits as soon as you bring any hardware into the picture, and typical USB audio interfaces are not always designed for integrating outboard. So it’s pleasing to see that, so far, the gear trend of 2025 is for manufacturers to tackle this problem head-on.

More and more audio interfaces are sprouting insert points, including the Black Lion Revolution 14x16 reviewed last month. The Audient iD48 reviewed in this issue takes this concept and runs with it, allowing you to switch its insert points into the post-D-A signal path for use as hardware inserts during mixing. SSL’s forthcoming Alpha 8 looks to be a uniquely flexible take on the line-level ADAT expander for connecting external hardware.

And perhaps the most innovative solution yet is the Freqport FO1, also reviewed in this issue, which lets the DAW relate to outboard as if it were simply another plug-in.

Bringing hardware into a software world used to involve patchbays, soldering irons, recall sheets and lots of headaches. One by one, those obstacles are being overcome. If you’re one of those people who’s become trapped in the box by the sheer awkwardness of leaving it, now might just be a good time to reconsider your options.

“Bringing hardware into a software world used to involve patchbays, soldering irons, recall sheets and lots of headaches.”

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FEATURES 88 Modular Synthesist, producer and YouTuber Julien Guillot, better known as Stazma, talks modules, mastering and teaching.

100 The SOS Guide To Gigging

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118 Spotlight: Battery-powered

PAGE 36

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From busking to barbecues — gigging with no mains power.

124 Inside Track: Koreless FKA Twigs has worked with many big-name producers, but it was experimental electronic artist Koreless who guided the making of her new hit album Eusexua.

130 Talkback

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Artist and producer Ela Minus explains how synthesizers changed her life.

132 Restoring Led Zeppelin A hit movie about the world’s greatest rock band turns conventional ideas of audio restoration on their head.

138 How I Got That Sound Engineer Stefan Boman recalls the role of room miking on the Hellacopters’ ‘No Song Unheard’.

140 Panda Bear The new album by Noah Lennox, aka Panda Bear, features familiar collaborators pulling in a very different musical direction.

148 Q&A Your studio and recording questions answered.

154 Why I Love... Problem Solving Mark Gordon revels in the satisfaction of finding and fixing faults.

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Cytomic The Scream Overdrive Modelling Plug-in

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Boss VE-22 Vocal Performer Cycling ’74 Max 9

76 Safari Pedals Zebra Clipper

68 Music Thing Modular Modular Synthesizer

Qu-Bit Stardust Eurorack Module

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ON TE ST

If you’re a solo performer who needs to control vocal harmonies and effects on stage, could this little box be all you need?

AT H E E N S P E N C E R

W

hen you’re a solo artist performing live, you already have multiple things to worry about if you’re to keep the momentum of your performance going and the audience engaged, especially if you’re singing and playing an instrument too. So it’s always handy to have something that you can just stomp on for accessing and controlling your vocal effects. I always found Boss’s VE-20 pedal great for this but in the updated version, the VE-22, you get added functionality and a nice big (320 x 240) LCD screen to make your life even easier — even with my ever-increasing shortsightedness, I could see the selected patch easily, and the colour‑coding helps even more in a dark environment!

Orientation & Control The VE-22 is sturdily built and logically presented, with three button-style footswitches taking the place of the two larger rectangular pedals of its predecessor.

On the top panel, as well as the screen and footswitches, you’ll find a volume knob, and three press-turn encoders for accessing various functions and setting up your effects.

On the back, there’s a USB-C socket for recording, as well as a quarter-inch jack input for adding a footswitch (the Roland FS-6 and FS-7, for example, but with a splitter cable you could use two single footswitches such as the FS-5U), or an expression pedal. TRS mini‑jack sockets provide a stereo aux input and a line/ headphones output. Two XLR outputs can be configured to deliver a stereo or dual‑mono signal, or one wet and one dry mono out, and there’s an XLR input for the mic. This mic input has a phantom power option, as well as a small knob for adjusting the input sensitivity.

The VE-22 is supplied with four AA batteries, so it’s possible to get up and running right away, but when phantom power is enabled the current draw increases from 180 to 260 mA. If you’re gigging regularly, then, you’ll almost certainly want the external power supply

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Boss VE-22 Vocal Performer the signal-to-noise ratio, and Vocal Effects Pedal tothemaximise D-A converters are 32-bit, with 32-bit

floating‑point processing being used inside. and probably also the CB-VE22 carry bag, both of which are optional extras (though some retailers do seem to offer the VE-22 and a suitable PSU as a package). The bag is fairly rigid, with a large outer pocket, and offers space for a mic, cable, PSU and other accessories.

This pedal really is intuitive, and you can get up and running with the 50 factory presets quickly. The Menu button obviously takes you into a menu area, and when you’re there the Harmony, Effect and Echo knobs become controllers for whatever parameter you’ve selected using the Memory knob. You can press these knobs once to switch them on/off, which is great because these are easier to use than the main switches during setup, while a double‑press gives you quick access to the corresponding menu page. You can write to one of the 99 available user patches by pressing the Exit and Menu buttons together.

Although the VE-22 is designed primarily as a performance unit, the availability of phantom power and USB-C audio interfacing means it could also work well for anyone wanting to use the unit for recording.

The A-D conversion is 24-bit/48kHz, using Boss’ now-familiar Adaptive Focus method

Processing & Effects The incoming mic signal can be cleaned up a little using the low‑cut filter, which can be turned on/off; there’s no information given on its frequency or slope but it works well in cutting out rumble, and there’s a separate EQ that I found really useful in eliminating the boominess that I was hearing from a headset mic. You can also select Low, Mid or High parameters, to adjust the frequencies at which harmony and pitch-correction effects are added, and this can be handy for preventing unwanted artefacts or feedback. A dynamics section provides access to simple but effective compression and de-essing.

When it comes to the effects, there are three banks to play with and you can use them in any combination, including using all three banks at once. The first caters for the harmonies, while the second is for general effects, including delay, chorus, flanger, phaser, vibrato and distortion. The third focuses on delays and reverbs, with four main reverb types, a delay and then four additional options with the reverb and delay combined. You also have access to some combination effects, plus a few more esoteric options including ring modulation,

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ON TE ST

BOSS VE-22 VOC AL PERFORMER

Though designed primarily with performance use in mind, the VE-22 doubles up as a USB audio interface.

a slicer, and both radio and lo-fi effects if you like to get a little more creative, and you can edit these to make them as subtle or intense as you like. The effects are of the quality we’ve come to expect from Boss, which is to say generally pretty decent, and you have a reasonable amount of control over them too, being able to edit such parameters as level, low cut, high cut, rate and depth.

In the Harmonies bank you’ll find a few different intervals to choose from, and doubling and tripling the vocal really can add some great punch to a solo singer’s performance. The harmonies themselves, though, I sometimes found a little trickier to work with. For example, while you can select the key in which you sing, there’s no guitar input or MIDI control to change the chords, so you’re stuck with those fixed intervals and if your track modulates then you’ll need to get a little creative. There are two ways to get around this: you can bypass the harmonies in certain sections of a track; or you can set up patches for different song sections, stomping up and down between them as required. The change between patches is pretty much instant, which makes the latter approach possible, but you do have to be careful about cutting off the tail end of your delay and reverb-based effects.

For me, the best auto-harmony results came from dropping the harmony volume levels and panning them slightly left and right, to get them to sit behind the main vocal — this tended to sound more natural than placing the harmonies up front and centre.

Of course, Auto-Tune‑style pitch correction is a popular real-time process for gigs too, whether you’re going for smooth and natural‑sounding correction or extreme T-Pain effects, so it’s good that you not only

have access to such a facility here, but that you’re given a useful degree of control.

This effect is found in the central bank, and it was something I felt was a particularly useful feature when mixing in the harmonies — it can really tighten up the sound.

Looping & Other Features While the VE-22 isn’t a dedicated looper pedal, there is a simple‑to‑use looping facility, and it’s a nice bonus: you can either build up a track with overdubs, or simply use it for some fun little interludes during a performance. The learning curve

Feed your backing tracks in and put your headphones on, and you can fine-tune with ease — without wearing out your voice — before heading to the rehearsal room.

Verdict It seems to me that the VE-22 has been designed primarily for vocalists who need an easy-to-use and lightweight effects unit for either solo or band gigs, but Boss have also paid attention to the needs of creatives who might be looking to incorporate unique effects into their performances.

Of course, Boss have a range of devices based on similar technology, and singing musicians may well prefer the VE-500 model. While that device does lack the appealing screen of the VE-22, it includes some more advanced features, such as the sort of instrument control of your harmonies that I discussed above.

Boss aren’t without competition in this area, from the likes of TC‑Helicon’s VoiceLive range and Zoom’s V-6. But overall, this is an impressive package for the price. It’s easy to use, it boasts a fantastic amount of functionality, and it delivers some high-quality effects — and the external pedal support, looper and USB interfacing are pleasant bonus features. Well worth checking out.

“I could see the selected patch easily, and the colour coding helps even more in a dark environment.”

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isn’t daunting even if you’re not used to looping, it’s a lot of fun to play with, and it demonstrates just how clean and quiet the inputs are: I got up to eight individual parts going using the overdub feature without any noticeable degradation of the signal. The recorded phrase can’t be saved internally and will be wiped once you leave that mode, although you can record the output to another device using the USB connection.

Another nifty feature is the ability to sync effects to a tempo (in bpm) and, importantly, you can tap that tempo in if you assign this feature to a pedal or button. That’s ideal if your performance is totally live, as you can keep these things in sync even if the tempo changes.

Finally, the ability to process not just mic signals but also line-level sources is also a handy feature, not least when it comes to getting your patches set up.

summary

The VE-22 packs a great collection of effects in a compact pedal, and lets you quickly craft your patches, take elements in and out, and switch between patches during a performance. A very appealing tool for vocalists.

$ $349.99.

W www.boss.info/us

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ON TE ST

The Tube Screamer is one of the most modded pedals there is, and this plug-in lets you to mod the virtual circuit to your heart’s content!

PAUL WHITE

W

ith so many effects‑modelling plug-ins available today, you might wonder why Cytomic would focus their talents on an emulation of the classic Ibanez TS808 Tube Screamer. Indeed, when I first saw their announcement, I wondered that myself — but boy am I glad that I decided to see what all the fuss was about! I started by comparing the results of placing The Scream before a Blackstar St James amp model set to the clean channel of its EL34 mode. In comparison with other drive plug-ins, the Scream sounded more refined (to be fair, Overloud TH-U’s Tube 9 came close), reminding me of a real Tube Screamer running into a clean amp, with the classic smooth highs combining with some mid lift and a little taming of the low end. Tweak the gain and you can go from an almost clean, warm boost to a hint of bluesy drive and then on into classic rock territory, resplendent with smooth saturation.

Tweakers’ Delight?

But Cytomic have taken things much further. First, there’s a global control panel that allows adjustment of the pre and post levels, plus the ability to mix the wet and dry sounds, which can help when a little more transient clarity is required.

The ‘pedal’ itself sports the usual trio of Drive, Level and Tone knobs, but you also get a two-position Mod switch plus trimmer pots for adjusting the working voltage and the circuitry bias. The Mod switch’s two positions access differently modified versions of the pedal, and circuit mods can be undertaken by the user without the need for a soldering iron!

Upping the Voltage appears to give a smoother sustain while Bias, which essentially offsets the centre rail voltage, produces authentically spitty and spluttery results if turned anticlockwise. For most users that will be enough, but the Mod button gets you into a world of diode swapping that can be applied separately

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Cytomic The Scream Overdrive Modelling Plug-in to the Mod switch’s two positions. The available diode types are Custom (the default), Silicon, Germanium, LED, Schottky and Zener, with a number of choices for each. The Silicon category offers 13 diode choices, and there are five Germanium types. Each of the circuit’s two diodes can be changed independently, and there’s the option to place multiple diodes (of the same type) in series; a common trick is to double up just one diode, to add some asymmetry to the clipping. A further button randomises component values.

This tweakers’ rabbit-hole keeps on getting deeper! Click on the Diode symbol (top right), and a Tube Screamer circuit schematic opens. Here, you can change the resistor or capacitor values — it’s a great way of designing and testing your own modded Tube Screamer before you go out and buy the bits to build one. The factory presets already include several well-known variations on the circuit, including various TS9 models, a Keely‑modified TS808, a Boss SD-1, an EQD Plumes and a Mesa Grid Slammer. As you select these, the circuit diagram updates accordingly.

Various mods can be saved as presets to load into your next virtual design session.

Switching to HD mode, which asks more of your CPU, models the virtual circuitry as one complex large-equation

model, and includes more precise modelling of the op‑amps, including slew rate, diode capacitance and power supply sag. A bar of dots (the Explorer Bar) stores each change, allowing you to go back to earlier versions very quickly, and further diode characteristics can be imported in the Spice model format for those who wish to get super-geeky.

In stereo mode, different component variations can be added to each channel.

A Real Scream!

If you just want a great-sounding TS808 with its familiar three knobs, that’s all you need to use. But this is so much more than a detailed plug-in version of the beloved Tube Screamer. Not only has it been modelled in fastidious detail, but you can tweak to your heart’s content and really dig into the fine detail of circuitry design — with no risk of burning out any chips!

And all this for only $29? It’s a steal!

summary So much more than a Tube Screamer emulation, The Scream could be as valuable to pedal builders as a design tool as it is as an effect for those wanting to shape their own guitar sounds.

$ $29.

W https://cytomic.com

ON TE ST

Z&H Designs 1178 Stereo Peak Limiter

Dual-channel Dynamics Processor

This UREI-inspired compressor includes some useful features for modern hybrid studios.

NEIL ROGERS

Z

&H Designs started life in 2019 when Jaky Zheng, an audio engineering instructor and studio owner, gathered together a team of designers, engineers and co-creators from around the globe to flesh out and develop equipment designs and prototypes to work in real-world studio sessions.

The final designs that are now coming to market are all hand-built in Shanghai by an experienced team who, they say, are now able to manufacture not just the devices themselves, but also many of the components used in them. Some of those, including op-amps and transformers, are available through their website. The idea is that Z&H can have full control of the quality of their designs, as well as the ability to build custom units to order.

Two 1176s?

We were invited to test one of Z&H’s takes on a classic studio processor: their 1178 Stereo Peak Limiter is clearly inspired by one of the less-well-known 1176-style FET compressors, the UREI 1178. Intriguingly, for this particular model, Z&H claim to have exclusive access to the remaining stocks of some discontinued components used in the original UREI 1178, including the NPD5566 FET transistor, which they say makes it possible to build a faithful recreation of the original signal path.

Released in 1979 (around the time of the Revision H 1176), the 1178 was UREI’s solution for engineers who wanted to use two 1176s as a stereo pair. Previously, getting two 1176s to behave as a stereo

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device wasn’t straightforward, and involved a battery-powered adaptor being connected to each device. The 1178 was conceived from the outset as a true stereo device, so the two channels shared a set of attack, release and ratio controls (making it less versatile when operating on two different mono signals than an 1176). Long since discontinued, Universal Audio briefly returned to the idea of a stereo 1176 in the early ’00s, with the release of the 2-1176.

Unlike the 1178, this model features fully independent controls for each compressor.

I’ve had a 2-1176 at my studio for well over a decade, and it gets heavy use as a dual-mono device with the bonus of stereo operation if needed, though stereo operation requires more manual control matching than on an 1178.

“One feature that might help Z&H’s 1178 stand out in this crowded marketplace is the option to operate as a Mid-Sides compressor.”

When it first arrived at my studio, I was impressed with the aesthetic qualities of the Z&H 1178. With a solid build quality and a nice finish, it gives the impression of being a ‘full-fat’ professional audio product. I especially liked the two large, slightly dimmed ’70s-style VU meters, and the various knobs and push-switches felt authentic too. But while it might appear very familiar, it’s worth looking more closely at the controls, because Z&H have sensibly

decided not to remain slavishly faithful to the original 1178 design. For instance, the user can configure the device for stereo or dual-mono operation, and there’s even a Mid-Sides mode. Z&H have also included other features that have become popular in modern studios: there’s a side-chain filter, to reduce the compressor’s sensitivity to lower frequencies, and a wet/dry blend control that allows instant parallel compression without using a mixer.

In At The Deep End! My studio is a busy place, with frequent recording sessions going on, and I like to slip review pieces into use on these sessions where I feel it’s appropriate, particularly with gear based on classic designs like this where the controls seem fairly straightforward. It means I’m able to judge how intuitive a device is to use, and how it compares with the tools I use on a day-to-day basis. As I often use one channel of my 2-1176 when tracking vocals, I thought this might be a good place to start — but it was a matter of seconds before I realised that this Z&H device was a rather different beast! Relative to my 2-1176, the input controls seemed very sensitive, and I found it tricky to set the threshold where I wanted it. I could also hear my vocalist sounding ‘compressed’ even when there was no activity showing on the gain reduction meter. It didn’t sound bad, but it certainly had a different sonic character, being somewhat darker and generally more colourful than my 2-1176.

So I lost my nerve on that session, and decided this was a piece that I should spend more time getting to know in the safer

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ON TE ST

Z & H D E S I G N S 1 17 8 S T E R E O P E A K L I M I T E R

The IEC mains inlet feeds an internal toroidal transformer and linear power supply, and can be switched for use with different AC mains voltages.

confines of an unattended mixing session!

This proved to be the right decision: like a lot of good outboard equipment, the Z&H 1178 is a device that rewards time spent learning and experimenting with how it behaves on a range of sources.

Before delving into stereo sources, I spent time using just a single channel on some of the typical sources on which an 1176 tends to excel. Trying out the Z&H 1178 on both male and female vocals, I began to get a better feel for how this unit behaved.

I didn’t have an original 1178 to compare it against, and was a little thrown by just how sensitive the controls seemed to be (which explained my experiences when tracking!).

With the gain reduction was engaged, and balancing the input and output levels in the usual way, it felt like small adjustments made quite dramatic changes to how much compression was being applied — I did grow used to it, but despite there being a substantial 41 steps on the input and output control knobs, I often felt that I’d dialled in either too little gain reduction or too much!

What this behaviour did encourage, though, was use of the side-chain high-pass filter and the blend control, and these certainly helped me to dial in the sounds I wanted. In fact, this workflow let me to some great results on a DI’ed bass guitar part, bringing stability and a sense of depth to the sound. It sounded very familiar on vocals too, with perhaps just a little more colour added than my own 2-1176 — in a good way!

Adventures In Stereo Most users will no doubt be drawn to the Z&H 1178 for its ability to control and shape stereo sources, such as when used on the drum bus or even a full mix. There’s some nuanced debate around the best way of using compressors on stereo sources, especially with a potentially fast and aggressive‑sounding compressor like an 1176. It’s good, then, to have options,

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which this device does. A toggle switch decides whether it operates in Stereo or A/B mode. In the former, the channels’ gain-reduction elements react to a common signal: the gain reduction is determined by the loudest signal on whichever channel at any given time. In A/B mode, it functions as a dual-mono compressor, with each channel reacting to its own control signal and, other than for the ratio, giving the user independent control of each channel.

Playing with these stereo options got me right into the heart of what this product has to offer. I found that A/B mode often worked well, with the caveat that I often had very carefully to nudge the settings to get the levels feeling nicely balanced across the two channels. 1176s are much-loved on drums, of course, so it’s perhaps not surprising that my favourite use for this 1178 was on a drum bus. With a medium attack and fast release, I found it really satisfying to dial in a healthy dollop of gain reduction and then use the blend control to fine-tune just how the drums sat in the mix. Punch, depth and lots of the good stuff could be heard.

I also really enjoyed the extra ‘zing’ and control I achieved on a pair of hard-panned acoustic guitars, with the side-chain filter contributing a sense of fine control. On that point, I often found myself engaging the side-chain filter at 60Hz rather than turn it off, as it just seemed to make the unit more predictable and less prone to big jumps in gain reduction. The side-chain filter’s other options of 120 and 180 Hz brought welcome flexibility too, of course.

Generally, I found myself favouring the Stereo setting for stereo sources, probably because the sensitive input controls meant I was constantly nudging the dials to get an even-sounding balance between the two sides. But while I’ve seen those two options on other UREI-inspired designs, one feature that might help Z&H’s 1178 stand out in this crowded marketplace is the option to operate as a Mid-Sides compressor.

In M-S mode, the left set of channel controls operate on the Mid component of a stereo signal and the right set on the Sides, and I quite enjoyed using it on a stereo mix bus. Because of the relatively aggressive nature of this style of compression it can leave a big impression on your audio, but sometimes that’s what’s needed! For example, I could achieve firm control over the vocals, kick and snare whilst allowing sounds in the Sides a little more dynamic freedom. I’m not sure I could imagine a mastering engineer using this type of compression on a day-to-day basis, but again the parallel mix control can help you tone down the results a bit — I can certainly see potential here for mix bus processing and stem mixing applications.

Summing Up Despite our slightly rocky start, I ended up enjoying my time with the Z&H 1178.

I haven’t used a vintage 1178 so I can’t comment on how closely to one of those this device sounds and performs, but to my ears it has a more aggressive and colourful tone than other 1176-style compressors I’ve used, and certainly that’s the case when comparing it with the UA 2-1176 in my studio. I mentioned some unpredictability to the input controls, but despite that — or perhaps because of it — this device often took me down some unexpected paths, and experimentation was generally rewarded with satisfying results. This company already have some other intriguing options in their range, and I’ll be very interested to see how they develop their portfolio in the years to come.

summary The Z&H Designs 1178 is a high-quality interpretation of a vintage stereo FET compressor, offering a colourful character and some useful functionality.

$ $2,199 plus shipping.

W https://zhdesigns.audio

Motorized MIDI controller for software and hardware

Rethink software and hardware interactions via fully assignable motorized touch-sensitive knobs, user definable haptics for detailed control, high-res screens providing easy-to-read labels for every control, and internal memory with storage for thousands of possible assignments.

Powerful, compact and portable, Roto-Control features three modes of operation: MIX, PLUGIN and MIDI mode with Motion Recorder. Get ready to control.

melbourneinstruments.com/rotocontrol | @melbourneinstruments

ON TE ST

Native Instruments Session Guitarist Electric Storm Deluxe Virtual Instrument

NI put the pedal to the very modern metal with their latest Session Guitarist title.

JOHN WALDEN

N

ative Instruments now have a pretty diverse selection of styles within their Session Guitarist series and the latest title — Electric Storm Deluxe — expands that further. If you follow electric guitar trends, then you won’t have failed to notice the rise of modern metal into (almost!) the mainstream: for example, Karnivool, Periphery, Animals As Leaders and, if you watched the Paris Olympics opening

18

ceremony, Gojira. Often combining the chug of djent, highly technical riffs, and ambient chord voicings that might make even a jazz player squeak, it’s a style that has also seen 7- and 8-string guitars come to the fore. It has already embedded itself firmly into video and film soundtracks.

It’s perhaps not surprising, therefore, that developers are paying attention and providing software instruments that deliver this sound and style.

The Oncoming Storm As with other more recent Session Guitarist titles, Electric Storm comes in two Kontakt versions. One provides the usual pattern-based Session Guitarist engine, while the second provides a combination of patterns plus a fully playable ‘melody’ virtual instrument with multiple articulation options. Under the hood, the sample base has been created using a high-end Framus 8-string electric guitar. The engine includes a comprehensive suite of amp, cab and effects modelling, and provides plenty of creative sound‑shaping possibilities. The tones are pretty epic but you can also

April 2025 / www.soundonsound.com

opt to use just the DI signal and add your own choice of third-party guitar rig modelling if you prefer.

Electric Storm provides a collection of 270 riff, chord and arpeggio patterns, organised into 63 ‘song’ based groups. Like other titles in the Session Guitarist series, the clever part — and the bit that allows you to generate instant results — is the engine’s ability to adapt these performance patterns to your own choice of MIDI‑triggered chord changes. By keyswitching between patterns, playing your own chord triggers, and adjusting the performance dynamics (via the Impact control), you can coax plenty of performance variety out of the base pattern set.

Beyond the preset patterns, there is a modest learning curve to master all the performance options.

However, if you dig in, the ability to drag-and-drop the MIDI pattern triggering notes to your DAW for editing, and options to switch riff playback to ‘single note’ mode (the pattern retains the rhythmic structure, but the riff sticks to a single pitch; you just play different trigger notes to revoice the melodic element of the riff) make a big difference in terms of crafting your own performance. Of course, the playable ‘melody’ instrument lets you add your own touches to the patterns and, with suitable use of the various articulations, lets you create very convincing solos.

In use, Electric Storm Deluxe undoubtedly hits its style and sonic targets. The pattern options provide instant gratification but the engine offers a surprising amount of flexibility. Pair it with suitable drum and bass performances (real or virtual) and you can get that super-heavy, down-tuned, modern metal vibe without having to master an actual 8-string guitar (my brain and fingers can’t cope!), while the more textural or ambient sounds are just as convincing.

Conclusion Of course, Electric Storm is not your only option.

For example, check out Impact Soundworks’ various Shreddage products or KeepForest’s Appex Modern Trailer Guitar — different price points, different feature sets, but the same general sonic target.

However, NI have captured the sound and spirt of the style really well in Electric Storm, and the Session Guitarist engine has something for everyone in terms of the performance features. Low, loud and heavy, or delicate, melodic and ambient; take your pick. If you want to bring the sound of modern metal guitar playing into your own projects, NI have made it easy to do; Electric Storm brings modern metal to the music production mainstream.

summary NI’s Electric Storm Deluxe lets you add the sound and style of modern metal guitar playing to your own projects.

The Session Guitarist engine, while easy for new users, also manages to offer plenty of performance flexibility for those prepared to dig in.

$ $149 W www.native-instruments.com

ON TE ST

Musik Hack Fuel Dynamics & Enhancement Processor Plug-in

Can just a small handful of controls give you the power to shape a range of sounds?

M AT T H O U G H TO N

I

was hugely impressed by Musik Hack’s Master Plan (https:// sosm.ag/musik-hack-masterplan). Essentially a DIY mastering tool, its stand-out feature is the Loud knob, a clever turn-for-more ‘maximiser’ that delivers proudly loud yet subjectively clear and pleasing results, all made easy thanks to a loudness-matching Unity button. I still use Master Plan, but Musik Hack have now released another plug-in called Fuel, featuring the same Loud knob and, again, supporting AU, AAX and VST3 formats and both macOS and Windows.

Fuel Injection This time, the Loud knob is flanked by controls designed to enrich and enliven individual sources, such as vocals, bass, guitar, synths and drums, and the control set is again deliberately simple, the idea being that you turn just a few knobs and decide if you want more or less of each process. The tweakers out there may not approve, but Musik Hack have done a decent job of judging the balance here. My only criticism is that the wet/dry Mix knob applies globally; having that option for each section could facilitate speedier results.

In and Out level knobs (each ±12dB) have a short-term LUFS readout beneath. Bypass and Unity buttons allow instant comparison with the unprocessed signal (Unity doesn’t affect the output LUFS figure), and each processing stage can be bypassed individually by clicking on its text label. A low-latency Live setting reduces the quality of some processing a tad while opening up the option of using Fuel to process a performer’s instrument while they play, and an Oversampling button delivers better quality at the expense of more CPU cycles.

To the left of the Loud knob is a compressor, with a knob (ranging from 0-100) and a Comp/OTT toggle switch. In Comp mode, it’s a standard single-band downward compressor, while OTT is a multi-band type. The compressor’s forgiving nature suggests there could be more going on than ‘turn for more gain reduction’ but from a user point of view you do indeed just turn and listen. Beneath, a horizontal fader labelled Body controls a separate upward compressor. Body’s an apt name given the effect on various sources, but I also found it handy for improving lyrical intelligibility by lifting up lower‑level vocal details; not a replacement for fader riding, but certainly a help.

To the right, a Soft Clip knob (again, 0-100 and ‘clockwise for more’) is joined by a Crunch distortion slider that ushers in a bright, edgy fuzz that can work wonders in getting a sound to ‘cut through’ in a mix. It’s pretty full on, though, so you won’t want to use it on everything (if everything ‘cuts’ then nothing does!) and

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when processing ‘natural’ sources you’ll rarely want to set it above 20-25, except when using the Mix knob for parallel processing.

The two remaining knobs are for the saturation processors, Bass and Thick. Bass seems to be a combination of EQ and LF saturation, while Thick fills out the midrange. Further features include a range of useful presets and the ability not only to scale the GUI but also to change its background colour per instance — a handy navigation aid if using instances on different sources.

Fuel’s Gold With Fuel, Musik Hack continue to impress. It never overwhelms you with parameters: you really can just turn knobs and listen. At the same time, though, there’s plenty of scope for enriching and filling out various sounds, from vocals to drums, and it’s really hard to make things sound bad unintentionally as long as you keep Unity engaged. Other than sometimes forgetting that certain stages were bypassed, it really couldn’t have been more intuitive.

If you have a limp sound that seems to be running on empty, then, you could try adding Fuel to thicken, enrich and sweeten it into usability. You might tame a rock vocal’s peaks while making its details heard, lending it a sense of power, and making it cut through a busy chorus. Or look to it to thrash the living daylights out of your drums, or give a lead guitar or synth an edge worthy of a katana...

But I reckon Fuel’s greatest potential in real-world mixes probably lies in exploring the more subtle treatments it can deliver, which can make pretty much any instrument sound just that bit larger than life — processing just a few select sources in that way could well work wonders in the context of a mix. Well worth a demo!

summary A strong second offering from Musik Hack, Fuel is an easy‑to‑use plug-in that combines upward and downward compression, saturation, clipping and distortion, for results ranging from subtle to extreme.

$ $79 (discounted to $59 when going to press).

Rent to own option also available.

W www.musikhack.com

To find a dealer visit www.musicmarketing.ca

ON TE ST

This modern take on a vintage germanium preamp offers bucketloads of character.

From top: SoundGas 636P2, Chandler Germanium Preamp and vintage Pye mixer.

SAM INGLIS

S

oundGas will be well known to British readers as one of the country’s leading specialists in vintage gear, especially old tape and disc echoes. Their USP is to offer comprehensively refurbished equipment, backed by a warranty and service options. They’re rarely the cheapest option, but when you’re spending thousands, this sort of peace of mind is invaluable.

Alongside the Space Echoes and Binsons, one of the mainstays of SoundGas’s vintage trading has been the Grampian 636 spring reverb.

This owes its cult status partly to its use by dub mixers such as Lee ‘Scratch’ Perry, and partly to its unique saturation characteristics when overdriven — the 636 was reportedly a mainstay of Pete Townshend’s guitar sound. A design from the early days of solid-state electronics, it uses germanium transistors to amplify the input signal and drive the reverb tank.

Germanium circuits frequently offered significantly worse noise and distortion performance than the valve designs they replaced, and were made obsolete in the ’60s when silicon transistors became widely available.

Germanium semiconductors are also notoriously inconsistent, which is a serious problem when trying to match the sound across multiple channels or units. But they have character in spades, and hence there’s a thriving cottage industry today in audio devices that use germanium circuits. In the studio world, Chandler have an entire range of germanium-based studio processors, and there are also plenty of guitar fuzzboxes and preamps based around germanium circuits.

Cooking On Gas Over in Derbyshire, meanwhile, SoundGas struggled to find enough Grampian 636 units to meet demand, and many of those they did find required complete rebuilds. Hence the decision was eventually taken

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SoundGas 636P2 Stereo Preamp

to develop a modern recreation.

This has been produced in very limited numbers, largely dictated by the challenge of finding the correct reverb tanks, not to mention germanium transistors that were anywhere near in tolerance. The SoundGas Type 636 is also labour-intensive to make, and consequently retails for almost £6000 on the rare occasions when

units are available. As was already mentioned, however, the cult of the 636 revolves as much around the preamp circuit as its ability to add reverb to things, and so SoundGas have developed several further 636 products that omit the reverb tank.

These include the pedal-format 636P, the 500-series 636-500, and the product under review here.

The SoundGas 636P2 is a 1U rackmounting two-channel preamp designed for use with microphones or instruments. Each channel has a phantom power switch and selectable input impedance, and SoundGas have also added Carnhill input and output transformers to balance the rear-panel input and output XLRs. Front-panel unbalanced jack sockets labelled ‘break in’ and ‘break out’ allow you to tap into the circuit after the input transformer and/or before the output transformer respectively.

The 636P2 is very nicely made, with a sturdy metal chassis and faceplate.

Apart from the aforementioned XLR and jack I/O, the only extra socket is an IEC connector that accepts a standard mains cable. You will need to specify your local mains voltage when ordering, as it does not use a universal PSU. Gain is controlled using vintage-style chickenhead knobs, and to the right of these you’ll find the 636P2’s only concession to onboard metering, in the shape of amber-coloured lamps labelled Overdrive. These don’t really provide useful information about signal levels, but they are an important part of the 636 circuit, the bulb serving as a crude limiter.

White City television studios, where it was employed as a ‘floor mixer’ to submix signals on their way to the main console.

However, a direct comparison between these three units is hampered by their wildly different operating levels. No specs are provided for the SoundGas preamp, but even with the gain cranked, the hottest output you can get from it seems to be around +12dBu. By contrast, the Chandler preamp will happily kick out +34dBu — way more than any of my A-D converters can handle — and the Pye mixer is specified for “at least 90dB” end-to-end gain, which translates in practice as “hard to use unless you turn the master output way down”. The Chandler also has a unique approach to gain setting, wherein its Gain and Feedback controls interact in unpredictable fashion.

Germanium Whip When tested with microphones, the 636P2 and the Pye mixer sounded pretty similar. In both cases, it’s possible

timbres from the same patch on my Korg MS20 simply by choosing different I/O paths. Going straight into and out of the XLRs delivers a warm, soft tone with a hint of growl when pushed, while the break-in jacks produce a more aggressive yet still sweet-sounding result. However, with unbalanced cables, a significant level of hum was audible unless I patched a DI box between synth and preamp.

What was most surprising about recording synths through the 636P2 was the effect it had on modulation.

The same degree of LFO pulsing, in the same patch, sounded much more obvious on the SoundGas preamp than on any of the others. I can only assume that this too is somehow due to the asymmetric clipping, which results in the positive-going waveform being trimmed while the negative-going side still has the headroom to modulate.

Finally, the break-in jack can of course be used to plug in a guitar or bass, whereupon the 636P2 becomes the polar opposite of a clean DI. With humbuckers in particular, in fact, it can be plain filthy, with a deliciously complex, chewy distortion that’s perfect for psychedelic or garage rock tones, and which can easily be further processed using a cabinet simulator if you want to bring it under control.

The SoundGas 636P2 won’t be your only preamp, but what it does is pretty much unique. Whereas some preamps aspire to offer ‘straight wire with gain’, this is more like a rusty coathanger. It’s neither clean nor quiet, but it adds instant character to anything you run through it. And although it can get pretty wild, there’s nothing accidental about the way it sounds — the two channels on the review unit were perfectly matched both in terms of level and tone. If you’re afflicted by sterile synths, dull drum machines or soulless samplers, the 636P2 will sort you out a treat.

“Whereas some preamps aspire to offer ‘straight wire with gain’, this is more like a rusty coathanger.”

Unclean Machine After this preamble, it shouldn’t come as a surprise to learn that the SoundGas 636 is not intended as a transparent, low-noise mic preamp for classical recording. I was particularly interested to try it because, as you can see from the photo, I have a couple of other germanium devices in my studio to compare it with. Chandler’s Germanium Preamp was reviewed by Hugh Robjohns back in November 2006 (www.soundonsound.com/ reviews/chandler-germanium), and remains a current product. It is an original design, not based directly on any vintage unit, and is transformer balanced on both input and output.

The four-channel mixer below it, meanwhile, belongs to the very earliest generation of solid-state audio equipment, from around 1959-60, and features huge Gardner mic transformers on the inputs. It was made in the UK by Pye Electronics and saw duty at the BBC’s

to get a relatively uncoloured sound by turning down the input gain, though the 636P2’s relatively low overall gain means you may then need to run its output into another preamp to achieve a satisfying line level. Turn up the gain, and both develop the same sort of crunchy saturation on peaks. The character is nearly identical to the ear, with the Pye perhaps sounding a touch ‘heavier’, but it’s clear from inspecting the waveforms that the clipping in the 636P2 is asymmetrical whereas that of the Pye is more or less symmetrical.

The Chandler preamp, by contrast, is impossible to drive into obvious saturation without massively overloading the A-D converter that follows it.

This sort of saturation on microphones has its place, but I personally wouldn’t require it often enough to justify the 636P2’s asking price, and it lacks enough clean gain to be useful as a general-purpose mic preamp. Where I think the unit comes into its own, though, is as a (nearly) line-level processor for synths and other electronic instruments. I was able to extract a surprisingly wide range of

summary Behind the 636P2’s innocent exterior lurks a seriously spicy preamp that excels for recording electronic instruments.

$ £2450 (approx $3043).

W www.soundgas.com www.soundonsound.com / April 2025

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ON TE ST

APL Virtuoso v2 From $149 pros • Fabulously effective virtual monitoring.

• Multi-monitor immersive mixing on headphones.

• Wide variety of modelled channel formats.

• Easy to use.

cons • Adjusted room settings can’t be saved.

summary Virtuoso v2 leverages cutting-edge psychoacoustic research to create a genuinely effective and valuable product, and I can see myself using Virtuoso v2 very regularly.

over headphones a representation of the sound and spatial quality of monitors in a room. However, rather than modelling only stereo monitoring, Virtuoso is able to model a comprehensive variety of multi-channel immersive formats.

Everything from, say, 5.1.2 Atmos to Auro 3D 13.1, and numerous varieties in between. At the time of writing, I believe only Acustica Audio Sienna, among the similar products I mentioned above, can work beyond stereo.

Overview Screen 1: Virtuoso’s default View page.

APL Virtuoso v2 Binaural Rendering Software

Based on cutting-edge psychoacoustic research, Virtuoso v2 aims to deliver for headphone users the most convincing emulation of ‘speakers in a room’ to date.

PHIL WARD

A

PL Virtuoso is a plug-in and standalone binaural renderer built on technology developed by Dr Hyunkook Lee, Professor of Audio and Psychoacoustic Engineering and Director of Applied Psychoacoustics Lab (APL) and the Centre for Audio and Psychoacoustic Engineering at the University of Huddersfield. The core psychoacoustic

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3D modelling technology of Virtuoso is known as ASPEN (APL Spatial Audio Engine) and, along with its use in Virtuoso v2, APL say that it will soon be available for licensing.

Virtuoso lives in the same kind of territory as Stephen Slate Audio VSX, Sonarworks SoundID Reference Virtual Monitoring, Acustica Audio Sienna, Dsoniq Realphones, and Waves Abbey Road Studio 3, in that it aims to deliver

The standalone version of Virtuoso is available for both Windows and macOS, and its plug-in version covers AU, AAX and VST3 formats. The specifics of the integration of the plug-in within a DAW session will depend very much on the channel width of the session and personal routing and workflow preferences, so beyond simply installing the plug-in and checking its functionality in Pro Tools, I concentrated for this review on using the standalone version.

Once installed and running (which requires iLok authorisation), Virtuoso can model 53 different monitor configurations, from mono to 26 channels, including immersive audio formats such as Dolby Atmos, ITU-R, Auro-3D, MPEG-H and Ambisonics arrays. Any of those monitor configurations can be heard in seven different virtual rooms with customisable acoustic parameters such as ambience, reverb time and listening distance. Virtuoso also offers headphone head tracking support, headphone response compensation with 100 stored headphone models, monitor frequency

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ON TE ST

APL VIRTUOSO V2

response modelling with 11 stored monitor models, routing and downmix options, and 10-band parametric headphone EQ. Finally, the standalone version of Virtuoso offers WAV and ADM (Dolby Atmos delivery format) file playback.

The View Page Screen 1 illustrates the Virtuoso View page; effectively its home page. The main graphic displays a representation of the currently selected monitor format (Atmos 7.1.4 in this case), which it (somewhat confusingly, I found) refers to as the Input Config. Each monitor in the display can be clicked to solo and Option-clicked to mute. Before I move on to describing the functions located below the main graphic, there’s actually a really important configuration parameter hidden behind the Options menu at the top right of the page — one that took me a while to find (before I’d read the manual of course).

The Options menu includes an Audio Settings sub‑menu where Virtuoso inputs and outputs can be specified. You won’t get very far with Virtuoso ’til at least the output setting has been attended to. And if you’re wondering what the input is for, it provides an alternative routing option for playback. For example, I was able to route Apple Spatial (Atmos) material from the Apple Music desktop app into Virtuoso.

Moving down the View page, Room presets are selected from a drop‑down menu and include a variety of studio control room models along with a dubbing stage, a domestic living room and a domestic home cinema room. The room models, say APL, were “scientifically designed with ASPEN’s 3D geometric acoustic simulation and carefully tuned by Tonmeister ears. The APL Listening Room preset models the actual APL critical listening room that complies with the ITU-R BS.1116-3 recommendations. All of the preset rooms have different room dimensions and acoustic characteristics.”

Alongside the Input Config and Room preset menus are Ambience, RT60 (reverb time), Listening Distance and Gain adjustment knobs. The first three enable some modification of the Room preset and the last one is effectively the output gain control. Option-clicking the four controls returns them to their default values but they don’t retain any adjusted values when a different room is selected.

I found this slightly inconvenient because it means if I made adjustments and

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Screen 2: The Settings page allows you to tweak parameters relating to the HRTF and room modelling.

then switched to a different room, the adjustments were gone when I returned to the first room. I would have liked Virtuoso to have a facility to store user room presets (although of course that can effectively be done in the plug-in version by saving a global plug-in preset).

Below the room adjustment knobs are drop‑down menus to select headphone (HPC) and monitor (SPK) EQ presets.

The headphone menu lists 100 popular studio headphones, including, as is apparent from the screenshot, one of my old favourites, the Sennheiser HD650. The range of monitor options is much less comprehensive at just 11 models, and in the drop‑down menu they are only identified by letters, so you have to read the manual to find out what they are. Worry not however, ’cause I’ve done that for you: Monitor A: ATC SCM11; Monitor B: ATC SCM7;

volume level that would otherwise change when room parameters vary.

On the left of the Settings page, starting at the top, are facilities to select between five generic HRTF presets or load a personal SOFA format HRTF. A channel mapping option comes next.

“Soloing speakers from the Virtuoso View page was shockingly effective in demonstrating its power to localise sounds around, behind and above.”

Monitor C: Barefoot MM45; Monitor D: B&W 705 S2; Monitor E: Dutch & Dutch 8C; Monitor F: Genelec 8331A; Monitor G: Harbeth M30.2 XD; Monitor H: Neumann KH120A; Monitor I: PMC6; Monitor J: Q Acoustics 3030i; Monitor K: Yamaha HS5.

Finally on the View screen is a file player that enables the selection of an ADM or WAV format immersive file package for playback. Play, stop and scrub functions provide playback control.

The playback function is not implemented in the plug-in version of Virtuoso .

The Setting Page Screen 2 illustrates the Virtuoso Settings page. Controls on the right of the Settings page enable further modification of the room models with room dimensions, damping (absorption), and low‑ and high‑frequency reverb decay. A Room Loudness Normalisation option is also provided that equalises the apparent

Channel mapping provides a solution to the issue of different DAWs and immersive workflows potentially assigning different channel orders to immersive channel widths. Channel mapping offers 1:1 and Auto options.

In 1:1 mode, channels get mapped to each Virtuoso input sequentially. This, say APL, is recommended for those working with Ambisonic material, or who find that their particular DAW operates in this way. In Auto mode, the channel names are used to map each channel to a corresponding virtual loudspeaker. This is the recommended mode for spatial audio formats such as Dolby Atmos.

Next comes the Downmix menu.

Downmix enables immersive multi-channel material to be heard downmixed as if the room were home to a monitoring or playback system comprising fewer speakers. Below the Downmix menu, a low‑pass filter can

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ON TE ST

APL VIRTUOSO V2

Screen 3: The Matrix page.

ALTERNATIVES Finally, the HPC EQ page, shown in Screen 4, presents a 10‑band parametric equaliser that, if a headphone EQ profile is assigned, shows the profile curve and enables its adjustment. If no headphone profile is assigned, the EQ will be flat and adjustable to taste.

Virtuoso’s Virtues

be engaged to enable manual routing to a Virtuoso virtual subwoofer. In the majority of cases I think such LFE routing or bass management is likely to be handled within the DAW session or embedded in the ADM playback, so the filter would remain off.

The final feature of this page is management of head tracking. My first exposure to head tracking was with the Audeze Maxwell I reviewed in the January 2025 issue, so I’d refer you to that for my thoughts on head tracking generally but, coincidentally, one of the headphone models with which Virtuoso head tracking is compatible is the Maxwell. I still had the Maxwell review sample to hand and firing them up with Virtuoso head tracking engaged worked extremely well.

Matrix & More The Matrix page, shown in Screen 3, shows the current channel assignments and provides Mute and Solo options for each monitor.

These are the same Mute and Solo functions available by clicking on a speaker in the View page. The Matrix page also enables trim/gain to be applied to each channel. The assignments and channel gains shown on the Matrix page will vary depending on the downmix settings selected in the Settings page.

Screen 4: The HPC EQ page, which allows the headphone correction to be modified.

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I’ve been pretty comprehensively through the functions and aims of Virtuoso but does it work?

Yes it does — and then some!

I was genuinely blown away

by Virtuoso’s ability to render over headphones a convincing multi-monitor system and room acoustic. I began by listening to a particularly familiar ADM and found myself instantly at home with the sound. In the Virtuoso Settings page, I ‘built’ a virtual room, modelled on my own in terms of dimensions and reverb time, and picked not dissimilar virtual monitors (Neumann KH120) to play the role of my Dynaudio BM5 Mk3 Atmos system.

Then I routed the Virtuoso output to one of the headphone sockets on my Audient ORIA interface so that I could quickly compare my real-world Atmos monitoring with the Virtuoso virtual version over headphones. It was a genuinely uncanny experience; all the panned Atmos elements ‘appeared’ in the appropriate places with a satisfying

While there are plenty of systems doing some of what Virtuoso v2 can do, those that are able to model multi-channel monitoring are thin on the ground. To my knowledge, only Acustica Audio’s Sienna currently offers comparable functionality.

sense of precision and focus, and the room acoustic sounded not just real, but familiar. Soloing speakers from the Virtuoso View page was shockingly effective in demonstrating its power to localise sounds around, behind and above. More than once I removed my headphones to check that the actual monitors were muted.

Virtuoso’s virtual room and monitoring isn’t perfect, though, with the most noticeable divergence from reality for me (apart from the physical experience of wearing headphones) being level differences in terms of the intensity of some rear‑panned mix elements. However, tweaks to the channel gain settings in the Virtuoso Matrix page were pretty effective in bringing the presentation in the virtual room closer to the real room, and I’m sure more refinement in that respect would only improve things. Picking the most effective HRTF is also a significant factor in this respect.

Once I’d started getting such great results from Virtuoso and familiar ADMs, I couldn’t help but start trawling around Apple Music for favourite Apple Spatial tracks to see how Virtuoso would handle them. At that point however, any thoughts of formally reviewing Virtuoso pretty much came to an end — I moved on simply to enjoying what it can do. And what it can do is deeply impressive.

But does it signal the end of conventional monitoring? I don’t think so, because there’s still nothing quite like the sense of air moving in a room.

But if you want to mix in Atmos (or any other of a multitude of immersive formats) and you can’t make space for, or afford, a multi-channel monitoring system, Virtuoso is a hugely effective and capable alternative. $ Standalone version $149 (discounted to

$59 when going to press). Plug-in version $199 (discounted to $79). Virtuoso Bundle (both versions) $299 (discounted to $99).

E support@apl-hud.com W https://apl-hud.com

ON TE ST

Freqport FreqInOut FO1

Outboard DAW Integration System

A clever new USB-powered unit aims to make it easier to access hardware in the box.

SAM INGLIS

I

n the early days of DAWs, software alone couldn’t provide the necessary processing power to handle a busy mix.

Even if your computer could take care of tracking and editing, a hardware console and a few racks full of outboard were required to distil the results into something suitable for mass consumption.

Today, few of us could say that we really need hardware outboard equipment.

Computers are now so powerful, and plug-ins so sophisticated, that there are no real limitations on what we can do with them. Yet hybrid setups combining software and hardware are growing in popularity. In particular, we’ve witnessed the rise of what might be called the hybrid desktop studio. Regardless of need, small studio owners are tempted by the idea of having a few choice items that they can integrate into an otherwise software-based studio environment. And this is a trend that extends beyond recording and mixing.

Electronic musicians want hardware synths and samplers to complement their virtual instruments, and guitarists devote scary

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amounts of time and money to curating the perfect pedalboard.

The pain point in these scenarios concerns exactly how this hardware can be integrated into a software environment, and that’s what Freqport are targeting with their second product. To the outside world, the FreqInOut FO1 is a line-level USB audio interface; but to your computer, it’s a VST, AU or AAX plug-in. The goal is to make adding outboard into a DAW session as easy as loading up a software effect.

Hub Tropicana On the digital side, the key to making this work is the Analog Hub software developed for Freqport’s first product, the Freqtube FT1. Reviewed in SOS February 2023, the FT1 is a desktop processor that likewise connects over USB and presents a plug-in interface, and which offers four mono or two stereo channels of authentically analogue valve warmth to be applied to the DAW signals of your choice. Successfully implementing this involved overcoming one of the main limitations associated with USB audio, especially on Windows, where the ASIO protocol allows only a single device to be

addressed. Freqport succeeded, and the Analog Hub software allows the FT1 to run simultaneously with, and independently of, your main audio interface.

Freqport FreqInOut FO1 $599 pros • Greatly simplifies the process of integrating outboard equipment into a DAW mix.

• Plug-in can store photos of your hardware settings for recall.

• Analog Matrix allows multiple hardware processors to be chained without multiple conversion stages.

• Built-in wet/dry balancing, Mid‑Sides matrixing and wide-ranging level trim.

• Up to two FO1 and two FT1 units can be used together, and USB hubs are supported.

cons • Plug-in interface could be streamlined.

summary An innovative, easy‑to‑use and highly effective one-stop shop for integrating hardware into a small software-based mixing or production setup.

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FREQPORT FREQINOUT FO1

Analog Hub runs in the background on your computer, and can integrate multiple FO1 and FT1 units.

Analog Hub is also the tie that binds the new FO1 into your computer system, and it’s not limited to a single device. Two FT1 and two FO1 units can all be connected simultaneously to a single system, with no need for additional clocking connections.

This would give you eight mono or four stereo channels of line-level I/O, and the same number of channels of valve processing: not quite the Power Station in its heyday, but enough to make a big difference.

As a fully self-contained unit, the Freqtube FT1 supports preset storage and recall, and parameter automation, just like any other plug-in. Latency is also compensated for in the background, so the only difference compared with digital plug-ins is that you can’t perform offline bounces. Moreover, the FT1 is also a hardware controller, with eight rotary encoders that can be assigned to parameters within the Freqtube plug-in. By contrast, the FreqInOut FO1 is inherently open-ended. It’s designed to work with whatever hardware you care to connect; as long as something can accept and return a line-level signal, it’ll work with the FO1. The flip side of this is that the FO1 can’t control anything. It’s up to the user to configure and recall settings on connected hardware, although, as we’ll see, the associated plug-in lends a helping hand where it can.

a great deal more than is needed to get the best from most outboard.

As each has only a single USB port, multiple FO1s cannot be daisy-chained.

However, unlike many conventional USB interfaces, Freqport units are happy to connect to a powered USB or Thunderbolt hub. I had no problem simultaneously running the FO1 from a built-in USB port on my Mac, and the FT1 from an attached hub. There is a list of recommended hubs on the Freqport website, and they plan to make their own available soon. (Using a hub would also have the advantage of allowing you to position the FO1 further away from your computer.)

A single installer installs both the Analog Hub background utility, and the FreqInOut plug-in. Firmware updates are handled within Analog Hub, and it does have a basic user interface, but under normal circumstances you can pretty much forget about it. The Freqport devices automatically adapt to the sample rate of your chosen DAW project, and Analog Hub runs at its own buffer size, which is set independently of your principal audio interface.

Where The Magic Happens The first time you open the FreqInOut plug-in, you’ll be greeted with a Setup Wizard page that introduces you to the basics. This uses minuscule white text

Joined Up Thinking The FO1 is a smart but low-key half-rack desktop unit (an optional rackmounting kit is available separately). The front panel is largely empty apart from an RGB indicator.

Turn it round and you’ll find a power switch, a button labelled ID and a small white LED display, which is used only for identification and diagnostic purposes, plus two banks of four balanced jack sockets, a USB Type‑C port and a socket for an optional DC power supply. The PSU is not included, as the FO1 would normally be bus powered. The FO1 uses 32-bit ESS converters, and a dynamic range of 120dB (D-A) and 116dB (A-D) is claimed: a few dB short of what the very best audio interfaces deliver, but probably

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The FreqInOut plug-in can store and display a photo of the piece of equipment inserted in each slot, making it easy to recall settings after the fact.

which I found hard to read, but nothing it describes falls outside the realms of common sense. The Wizard can be recalled any time you need a refresher course, but before you can use the plug-in, you’ll need to visit the Global Hardware Configuration page.

Reflecting the FO1’s complement of physical I/O, the FreqInOut plug-in has four ‘slots’, each corresponding to an output/input pair. In the most basic use case, each of these four slots would be accessed through a different instance of the plug-in, in effect delivering four mono hardware inserts that can be used on different DAW channels. It’s also possible to link slots 1+2 or 3+4 to accommodate stereo processors, and to add Mid‑Sides matrixing to a linked pair of slots (which gets matrixed back to conventional stereo again on the way in). Slots that are in use by an existing instance of the plug-in are greyed out when you add another instance.

More sophisticated setups are also possible, courtesy of something Freqport call the Analog Matrix. This permits multiple slots to be chained within the analogue domain, allowing the output of one hardware device to be routed directly into the input of the next without any additional stages of A-D or D-A conversion.

The FO1 also has adjustable analogue trim on both inputs and outputs, and the main purpose of the Global Hardware Configuration page is to set this up for each slot. The amount of trim available has increased with firmware updates, and at the time of writing is ±12dB. The maximum signal level that can be delivered at the outputs and accommodated at the inputs is +18dBu, so in effect, the default zero position represents an operating alignment of 0dBFS = +6dBu. The +18dBu ceiling might be a limitation with some rackmounting compressors that expect hotter signals, but the extensive trim options make it easy to accommodate things such as guitar pedals that don’t expect a full line level. You can also add custom names for each slot within Global Hardware Configuration, and write the entire configuration of the FO1 to the unit itself. This isn’t necessary in normal use, as everything is recalled when a preset or DAW project is reloaded, but might be valuable if for example you want to use it with a different computer.

In both Global Hardware Configuration view and the normal view, you’ll see input and output meters at the top and bottom of the screen respectively. These relate to

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FREQPORT FREQINOUT FO1

the signal emerging from and returning to the instance of the plug-in you’re viewing, and both have gain sliders superimposed on them that allow the signal level to be adjusted in the digital domain, over and above any analogue trim settings made in the Configuration page. Unlike the trim controls, these send and return level sliders can be automated.

Picture Palace The idea is that you’ll only visit the Global Hardware Configuration page when your hardware setup changes. In normal use, you’ll work in the default view, where settings related to the particular instance and slot(s) in use appear. Initially, you’ll see the central area taken up with a photo of the rear of the FO1. Click on the camera icon above this, press the dustbin icon to ditch this default photo, and a QR code is displayed. By scanning this with your phone, or clicking the adjacent upload icon, you can then replace the default image with a photo of whatever piece of kit is plugged into the selected FO1 slot. Basic cropping and image rotation are available, and there’s also a magnifying glass that allows you to inspect bits of your photo in more detail. This will be particularly useful with complex rackmount gear, as the square‑ish shape of the photo area lends itself better to pedals and desktop units. The idea, of course, is that although the FO1 can’t control or recall settings on connected devices, it can at least store photos of how they are set, allowing

The Global Hardware Configuration pane is where trim values are set and slots are named.

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the user to quickly and easily return to whatever preset or setting was used on a given project.

Last but not least, there’s a wet/dry mix control for each slot. Like the Analog Matrix routing, wet/dry balancing takes place in hardware, ensuring that there shouldn’t be any phase issues when the wet and dry paths are combined, at least as long as the gear you’re connecting is analogue and doesn’t delay the wet signal. A firmware update (which may be available by the time you read this) will add a configurable delay to the dry path so that digital devices can also be accommodated, plus switchable polarity reversal. Finally, the plug-in also has an internal bypass buttton, which is important, because the usual bypass mechanism within the DAW mixer and plug-in header is not supported.

The Ins & Outs In practice, all of this takes longer to describe than it does to learn. Once you have the I/O configuration and gain set up appropriately for your outboard, there’s really very little that needs to happen in software beyond opening up the plug-in, choosing which slot or slots you want to use, and uploading photos. After a while, in fact, I started to wonder whether there was any need for the Global Hardware Configuration settings to be on a separate page. I don’t really see why its functionality couldn’t be integrated with the Analog Matrix in the lower half of the default view, which would simplify operation even further. At the very least, it would be good if there was an easier way to access it than by clicking on a tiny cogwheel icon and choosing Global Hardware Configuration from a drop-down menu. (While I’m at it, the dustbin icon in the preset view also presents an unnecessary step, as the only reason to click on the camera icon is when you want to want to replace the image.)

Beyond that, there’s little to say about the FO1 other than that it just works. I didn’t experience any audio glitches or refusal to cooperate with my main audio interface, the FO1 and FT1 coexisted seamlessly, and before long I was loading and saving DAW projects without even thinking about whether they employed external outboard.

Despite having hundreds of plug-ins installed, I’m as vulnerable as the next SOS reader to a tasty piece of analogue outboard gear. And, like many of us, I’ve found that mucking about with hardware inserts in DAWs quickly puts me off. The outboard that I do use nearly always gets

printed, just because using it any other way is a pain. If I had racks full of the stuff and relied on it for everything, I’m sure I’d learn to manage the inconvenience, but because I mainly mix in the box, the benefit of including just a couple of hardware processors or effects units never quite seems to outweigh the awkwardness.

And when it does, I live in the constant fear that I’ll return to a mix two years hence to be greeted by a session that contains no record of what hardware device was used in a given insert, or what settings were employed.

The other great benefit of the FO1 is that it doesn’t take up any of your main audio interface’s I/O. Most project and home studios nowadays are based around USB interfaces, and there’s almost always a hard limit as to how far these can be expanded. If your ADAT sockets are needed for drum mics or synths, you’d need to engage in a complicated dance involving patchbay wiring to use the same I/O to integrate hardware at mixdown.

And even if you have a spare optical port handy, the FO1 is a neater solution than adding a line-level ADAT expander, and likely no more costly. It also retains its full I/O count at all sample rates. Moreover, FO1 features such as the Analog Matrix, wet/dry mixing and Mid‑Sides matrixing would be difficult or impossible to reproduce using conventional hardware inserts.

Studios that operate large-format consoles with racks and racks full of outboard have always had interfacing options available to them, albeit at a price.

The FO1 is the first attempt I know of to create something that specifically targets a very different user: the engineer, producer or musician who works mainly in the box, but wants a seamless way to integrate a few pieces of outboard, or get more from their guitar pedals. Indeed, thanks to its form factor and relatively conservative operating levels, it’s arguably even better suited to working with desktop gear and pedals than traditional studio outboard, though it does the latter perfectly well too. The FreqInOut FO1 does exactly what it sets out to do, rebalancing the gain-versus-pain equation and turning what was, for many of us, a theoretical possibility into a practical reality. $ $599 T RAD Distribution +1 845 378 1189 E sales@raddist.com W www.raddist.com W www.freqport.com

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COMPE TITION

Win! Kali Audio & LEWITT Studio Bundle F

or this month exclusive SOS competition, we’ve teamed up with not just one but two renowned pro audio brands: US monitor gurus Kali Audio and Austrian microphone specialists LEWITT. The prize will be nothing less than a complete recording bundle, including a 2.1 monitor system, a premium audio interface and two top‑quality studio microphones — everything you need to turn your computer or mobile device into a fully functional music, podcasting and content‑creation studio!

Monitoring comes courtesy of Kali Audio, who’ll be providing the winner with a pair of LP-6 active monitors and a WS-6.2 subwoofer.

Part of Kali’s critically acclaimed Lone Pine series, the LP-6 boasts 80W of amplification, split between its 1-inch tweeter and 6‑inch woofer, and a frequency response that extends from 25kHz all the way down to 39Hz (-10dB).

And to further extend the low end, the WS-6.2 subwoofer uses two 6.5-inch bass

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April 2025 / www.soundonsound.com

Worth $1900 drivers in tandem, powered by 1kW of Class-D amplification, for a response that goes right down to a bone-shaking 27Hz. In terms of recording, LEWITT are offering their powerful desktop USB interface, the CONNECT 6. Compatible with both Apple and Windows computers,

To enter, please visit:

https://sosm.ag/kali-lewitt-0425 as well as iOS/iPadOS and Android mobile devices, the CONNECT 6 offers two channels of premium microphone preamplification, a pair of headphone outputs, stereo line outs for monitoring, an aux input, and the ability to connect to both a computer and a mobile device at the same time — ideal for streaming,

podcasting and more. The CONNECT 6 is loaded with DSP too, with compressors, EQs, expanders, maximisers and more all on hand to help you optimise your sound.

Finally, LEWITT will also be providing not one but two of their innovative RAY large-diaphragm capacitor microphones.

The first mic with ‘Autofocus for your voice’, the RAY pairs LEWITT’s proprietary capsule design with a unique proximity sensor, allowing the mic to tailor its response as needed, effectively eliminating the proximity effect. The RAY won this year’s prestigious SOS Award for best microphone, and will shine on any source, whether you’re recording vocals or instruments, streaming, or producing podcasts.

To be in with a chance of winning this fantastic prize bundle, simply follow the URL shown, and answer all the questions there, by Friday 2nd May 2025.

Good luck!

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W www.kaliaudio.com W www.lewitt-audio.com

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500-series Mixer

Looptrotter Modular Console We take a look at what has to be one of the most configurable analogue consoles out there.

M AT T H O U G H TO N

T

hat analogue consoles are still ‘a thing’ in 2025 doesn’t surprise me, because they make great recording front ends: not only can they provide lots of high-quality recording channels, but they also make it really quick and easy to manage multi-miked band recordings, since the engineer has one button/knob per function when it comes to things like soloing, muting, polarity inversion, and building truly zero-latency headphone mixes. But when it comes to mixing, our DAW software outguns them in so many ways, so many of us feel that it only really makes good sense to bring sounds ‘outside the box’ when we’re looking

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for the character that specific analogue hardware might contribute.

While there are exceptions, most professional analogue consoles released in recent years have fallen into two camps.

The first comprises high‑quality but efficient consoles with preamps, EQ and sometimes dynamics processing on board, and a specific workflow almost ‘baked in’ — the Audient ASP8024, Trident 78 series and Harrison 32Classic, for example. The second type has been a somehow ‘slimmed down’ offering that takes up less space than a traditional console and needs augmenting with other hardware: Neve’s 8424, for example, has no preamps of its own but provides remote control over the company’s separate rackmount preamps, as well as

providing a couple of 500-series module slots so you can tailor the bus processing.

The SSL XL‑Desk and API’s The Box are even smaller examples, but offer many more 500-series slots, allowing you to customise any channel.

Modular Thinking Looptrotter’s Modular Console is a bit different. It certainly falls into the latter category in that every channel is intended to be populated with 500-series modules.

But like the Audient and Trident, it comes in a range of sizes and offers more ‘big console’ features. It’s been available for a few years but as it isn’t the sort of device that’s easy to line up for review — it really needs populating with more modules than I think anyone at SOS owns — we’ve not had the opportunity. So when Looptrotter’s UK distributors KMR Audio, who also sell a huge range of 500-series modules, recently invited me to their showroom to put their demo rig through its paces, I jumped at the chance and caught a train to London.

Although it has lots of I/O and provides aux and summing busses, a master section and metering, the Modular Console lacks any preamps, EQ or dynamics processing.

Instead, each input channel is given a pair

of 500-series module slots, arranged one above the other, where you might usually expect to see the preamp and EQ sections of a console. The big idea, of course, is that you can configure it with whatever preamps, EQ or dynamics processors you want, and that your choice can be different on every channel.

A less obvious benefit is that, unlike with a conventional console and patchbay system, this arrangement allows you to put pretty much all the hardware controls you might need on a session within easy reach of the listening position.

The desk is made up of one or more Input Modules and a Master Module, each just wide enough to accommodate eight 500-series modules. This eight-channel ‘bucket’ approach influences the Modular Console’s off-the-peg sizes, of course. The current options have eight, 16, 24 or 32 input channels, and the larger versions can be arranged as you prefer, for example with all input channels on the left, or with some to the left of the Master Module and others to the right. If you wish, Looptrotter can also incorporate a desk surface for your screen and computer peripherals, and Looptrotter tell me they’d be happy to build larger custom consoles too, with additional channels added as eight-channel Input Modules.

The whole ensemble is encased in solid-but-stylish wooden furniture, which extends out the front to form an armrest.

There are a range of different wood veneers available from California Trading, a supplier in Looptrotter’s native Poland.

At the time of writing, these could only be viewed on a Polish-language website, but Google Translate makes it easy to see the options that are available, including European oak, American walnut, bubinga and rosewood. Power comes from a fanless, 3U hybrid power supply (Looptrotter say that “switching power supplies provide a lot of power, and behind them are linear regulators that thoroughly clean the power lines of all the garbage.” It remained whisper quiet for my several hours with the console.

to process channels 8+9 with a stereo compressor you could find a way.

The input channels each have four aux send pots, and each of those is switchable pre/post fader in the aux master section.

This, of course, means the pre/post status for each aux send applies to all channels simultaneously — you couldn’t set aux 1 as pre-fader on one channel and post-fader on another, but that’s probably a limitation more in theory than practice. There are mute and solo buttons, a pan pot and a long-throw ALPS fader, ranging from +10dB to full attenuation. After the fader and pan pot, four buttons route the post-fader signal to the four stereo subgroups and/or the L-R master bus, and each destination bus has a single 500-series module slot per channel.

There’s a final set of channel controls that I’ve not yet mentioned. A Tape pot sets the signal level that’s sent to your recording device, of course, but adjacent to this is a button that toggles the channel between to operating modes, called Record and Mix. In Record mode, the channel’s signal is received from a Mic/Line DB25 D-Sub connector on the rear. This flows through the channel’s 500-series slots and insert point. The latter has separate send and return connectors, again on rear-panel DB25s, and the send remains active even when the insert is bypassed. The signal is then routed out of the DB25 Tape Output and comes back through the DAW Input (on, yes, you guessed it, a DB25), before flowing through the rest of the channel. It’s

Looptrotter Modular Console $11,498 pros • Supremely versatile.

• High standard of construction.

• Good connectivity.

• Channel modes for recording and mixing.

• Custom finishes available.

cons • Manual rather light on details!

summary A highly configurable, high-quality analogue mixing desk with plenty of I/O and intuitive control.

this DAW Input ‘tape return’ signal whose level is displayed on the channel meter. In Mix mode, it’s the DAW Input connector that feeds the first 500‑series slot, and the meter displays the level after the second 500-series slot. It’s from this point once again that the Tape Output takes its signal.

This intuitive dual-mode system means you can capture a performance through your 500-series modules (and any inserted hardware) before it hits the fader, then flip any channel into Mix mode and use the same gear to build a mix from your DAW tracks. If you didn’t want to commit to the analogue mix, you could print each processed channel back to the DAW. But there’s another option: if you don’t need the inserts, you could use the always-on insert

Input Channels Each input channel features an insert point, and this and both 500-series slots are individually bypassable. It’s possible to use double‑width stereo 500-series modules in adjacent channels. As the inputs come in buckets of eight, you couldn’t physically place a stereo compressor in, say, channels 8+9, but the module slots all have I/O on the rear panel so if you really wanted

Each 500-series slot can be individually engaged/bypassed.

www.soundonsound.com / April 2025

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and you should plan accordingly if bringing everything out to a patchbay.

Master Module

Most of the plentiful I/O are on DB25 connectors, and those intended for connection to your converters are protected from phantom power.

send as a direct out after the first 500-series module: you could quickly EQ the signal in the monitor mix without printing the EQ, which you could then take the time to refine in Mix mode.

The DAW Inputs have blocking capacitors to protect your converters from any stray DC voltages. That’s important,

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since the first thing the signal hits is a 500-series slot, which could potentially be hosting a mic preamp with switchable 48V phantom power: connect your interface to the Tape Out and DAW Input, and you won’t fry it! There’s obviously no provision for blocking phantom power on the regular Mic/ Line input used in Record mode, though,

In the Master Module you’ll find a range of controls for all of the desk’s global facilities: the aux sends, the subgroups and master bus, monitor control and talkback, as well as four stereo input channels.

Top left are the master level, solo and pre/post controls for the four stereo aux sends, and beneath them are rotary level controls, mute and solo buttons, along with bus routing buttons for the four stereo inputs. (These can serve as aux returns if you wish, but could be used for anything.)

To the right of the aux controls is a 2x24 LED bar meter showing the solo level (set in the Monitor section, of which more shortly). Positioned between this and the L-R bus’ two 500 slots on the right is a pair of moving-coil VU meters for the master bus output level (post fader), and that signal’s sent both to the monitor section and to the main L-R output. The two final controls in this ‘upper section’ are level knobs for the dedicated External and DAW stereo

inputs, of which more when I discuss the Monitor section. Beneath all this are the eight 500-series slots for the four stereo input channels/subgroups and, as with the master bus slots, these are accompanied by bypass switches for each slot pair and for the stereo insert.

Moving down to the bottom section, you’ll see a talkback section on the left, the five stereo group and master faders on the right and, in between the two, a Monitor section that allows you to select different sources and to route them to different destinations. As well as the main mix, those sources can be the talkback mic, the solo bus, and/or the aforementioned External or DAW stereo inputs. As for destinations, there are three stereo monitor outputs (on XLRs), each with its own selector button and level trim pot. There’s also a level control for the single headphone output (you could of course also feed separate headphone amps from other outputs).

The External input is a line-level input accessible both on a rear‑panel DB25 and, for convenience, on a jack at the front, hidden underneath the armrest. These

Each input channel has four aux sends, each of which can be switched, globally, to send the pre‑ or post-fader signal.

www.soundonsound.com / April 2025

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inputs are summed, rather than the front one taking precedence when plugged in.

The stereo DAW input comes in on the pins of a DB25 connector that’s shared with the L-R bus’ insert sends and returns, and the main L-R output. An efficient though unusual arrangement, this connector adheres to the usual Tascam channel format, but it combines inputs and outputs so could mean using a custom cable or adaptors if interfacing with XLRs. On this subject, I don’t propose to list every input and output here: you should be able to see from the rear‑panel photo what signals are on what connectors, but it’s also very clearly set out in the PDF manual. Generally, signals that you might expect to connect directly to a patchbay or audio interface are on DB25s, while other connections are on XLRs or quarter-inch jacks.

The talkback system, which includes a preset limiter, is well specified. As well as an internal mic, there’s the option to connect an external one to an XLR on the rear. There’s not only a gain pot and phantom power switch here, but a momentary footswitch input too and even a USB MIDI input that allows you to control talkback using a signal from your DAW — a thoughtful touch. The manual talkback button in the Monitor section can be set to

ALTERNATIVES While there are very few other console designs that are quite so versatile, there are plenty that can host 500-series modules. API mixers have long been used this format, for example, and today API, SSL, Neve, Rodec, Tonelux, Wolff Audio, Tree Audio and Interphase Audio all have mixers that can host at least a few modules (and some can accommodate many). The closest point of comparison is probably API’s The Box 2, which similarly has two module slots per input channel, but it’s not scalable in the way the Modular Console is. For larger installations, Wolff Audio’s Console ST and Interphase Audio’s Ark are probably the most interesting modular alternatives.

momentary or latching behaviour, or auto (quick press for latch, hold for momentary) and a dedicated SLS output allows you to route the talkback mic elsewhere.

There’s also a listen mic input with a gain control on the rear, an integrated limiter, and a top-panel level control, to facilitate two-way communication with the live room.

As well as the speaker trims, headphone and solo level knobs and the larger main control room level control, there’s a dim facility comprising a button and level control, and there are mono and mute buttons

for the whole control room section (ie. independent of the main mix ones!).

Sum Thing Important The idea, of course, is that you can use 500-series modules to tailor this console to your own needs and/or tastes. But while the Modular Console doesn’t give you any mic preamps and EQs, it functions as a line-level mixer even before any 500-series modules have been installed. I could find no published technical specs and wasn’t in a position to measure things on my visit, but as I understand it, the channel signals are summed separately from the group signals on separate op‑amps, and the summed signals are then summed on the transformers, with a pure Class‑A transistor amplifier being responsible for the signal amplification. Subjectively it all sounded beautifully clean and pristine, with just a nice little touch of ‘life’ to it, which is as I’d want it; if you want more colour than that, you can use 500-series modules!

Configuration Deliberations So how might you configure a Modular Console? Well, given the range of 500-series modules now available, there are few limits on how you can configure this console other than your budget and your imagination. One option, for instance, would be to populate all the channels with Neve preamps and SSL EQs, and matching compressors on the busses. Or you might put different ‘flavours’ on channels dedicated to different instruments, or spend big on a select few channels, while putting more affordable options in channels you’ll use less frequently.

An option that appeals to me would be to use separate rackmount mic preamps, and pack the channels with my favourite EQs, compressors, saturators and so on, to put lots of control at my fingertips. I might even reserve a few channels for utility modules such as re‑amp interfaces. And remember, you still have the insert points should you want to add more processors (some might wish for a third row of 500-series slots, but that would risk making parts of the console hard to reach).

Value In Versatility?

The upper section of the Master Module hosts controls for the aux sends, the main stereo inputs, and a pair of stereo inputs for external sources and DAW playback.

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No high-quality console would be a modest investment today, and a fully loaded Looptrotter is no exception. But assessing the value for money it offers isn’t easy, as 500-series modules vary greatly in price; a preamp could cost a couple of hundred dollars or over a thousand! With that in

mind, I thought I’d run through a couple of examples that I think show that it’s a pretty attractive proposition to someone contemplating a new high-end mixer.

Our first point of reference is the Audient ASP4816-HE, a 16:8:2 mixer that can be yours for £17,500 including VAT here in the UK, and in terms of inputs, auxes and busses is broadly the same configuration as a 16-channel Looptrotter Modular Console.

The latter costs £15,500, and the £2k difference clearly won’t buy 16 channels of 500-series preamps and EQ. But the Looptrotter has the equivalent of over four eight-channel 500-series chassis and one 10 slot one, with a decent power supply.

That lot could easily cost £3k or more — and £5k is enough to buy you, say, 16 SSL SiX Channel modules, each with a preamp, a two-band bell/shelving EQ, a one-knob compressor and an instrument input, and leave you change for a bus compressor. Or you might pick up two or three eight-channel Audient mic preamps, and have plenty left over for a handful of choice 500-series modules to start your collection. Either setup would also leave you with plenty more slots should you want to add more over time.

At the other end of the food chain, you could blow a fortune packing out a Modular Console with ‘big name’ modules, but it could still have more of an edge over the competition in terms of price. Compare and contrast, for example, a Neve BCM10/2 Mk2 and a Looptrotter Modular Console with the same number of channels populated with Neve 1073LBs and

The Monitor section of the Master Module, with talkback and listen mic facilities, alongside source and monitor selection and level controls.

1073LBEQs, or a Harrison 32Classic with a 32-channel Modular packed with Harrison 500-series modules. Yes, there would be differences, but I’m sure you get the idea: while we’re not talking pocket change here, when judged against the competition the Looptrotter offers decent value for money.

It says a lot that my biggest criticism is of the owner’s manual: an 11-page PDF, this contains neither technical specs nor yet any description of the operation of the desk. What it does provide is helpful — it comprises a detailed set of signal flow and pin-out diagrams — and to be fair, Looptrotter agreed with my assessment and say that a much more comprehensive

manual is currently being written. That niggle aside, I’ve been very impressed by the Looptrotter Modular Console. It’s been designed with modern studios in mind, and that shows: it’s stylish, scalable, highly configurable, and incredibly easy to operate. I’m not in a position to replace my own mixer right now, but if I were in the market I’d be giving this one serious consideration. $ Eight channels $11,498. 16 channels

$16,497. 24 channels £21,496. 32 channels $26,495.

E contact@looptrotter.com W https://looptrotter.com

Patented. Legendary. Spectacular.

The patented Vitalizer circuitry makes the Vitalizer Mk3-T an unsurpassed sound designer.

Audio productions sound more alive, have more details, more transparency, a higher perceived loudness and an overwhelming bass fundament.

www.spl.audio

www.soundonsound.com / April 2025

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Portable PA System

Mackie ShowBox $799

Mackie’s new all-in-one battery-powered PA rig puts the controls right where you need them.

pros

PAUL WHITE

Mackie ShowBox • Mixer allows four mono sources and a stereo source to be mixed, with independent EQ for each input.

• Bluetooth streaming and aux mini‑jack input for connecting mobile devices.

• Removable mixer module makes live control easy.

• Wide range of effects.

• Long operating time when on battery power.

• Battery included in the price.

cons • EQ tweaks may be necessary to get the best vocal and guitar sounds.

• No phantom power available on the mic inputs.

• USB-C port doesn’t supply power.

summary A flexible and powerful system that would work for a wide range of gigs, whether or not you’re near a mains outlet!

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W

e’re now well used to seeing small PA speakers with built-in mixers, and the option of battery power is also becoming increasingly popular. Mackie’s new ShowBox ticks both of those boxes, while cramming in a few unexpected features. The most notable of these is the ‘breakaway’ Mix Control panel, which connects to the main unit via the included Cat5 cable, and can be used to provide hands-on control of the most important functions (levels , effects, tone control, snapshots, looper, tuner and so on), directly from your mic stand using the included standmount. Alternatively it can be left docked into its recess at the side of the cabinet. Unlike some of the

competition, the battery is included in the price of the ShowBox, and is accessed through the base of the cabinet.

Everything is housed in a polypropylene enclosure with rounded edges and a recessed handle, while the drivers are protected by a perforated metal grille. No cover is included, but an optional cover is available. Measuring 500 x 300 x 249mm and weighing just 9.5kg, the ShowBox is easily moved with just one hand. It can handle up to six analogue inputs: four mono (two mic inputs and two instrument inputs), and a stereo line-only pair, which is internally mixed to mono. Bluetooth streaming feeds into the stereo channel and, for the Ed Sheeran wannabes out there, a built-in two‑minute looper is available, as is a guitar tuner. While it works in the usual

way, the looper doesn’t have undo or redo capability, and loops are lost when the power is removed or the breakaway module disconnected.

USB-C interfacing is available for live streaming, playback and recording, and direct recording to micro‑SD card is also supported. The USB-C interface port doesn’t supply power, but a second USB port on the breakaway controller can supply power to your mobile devices (but does not carry data). A Class 10 or better SD card is needed for direct recording and it must be formatted before use as detailed in the manual.

An optional dual footswitch can be used to control the looper and to bypass the effects for live performance, and there’s also a headphone output on quarter‑inch TRS jack, as well as an XLR mix output for connection to a further powered speaker.

adjusts the sound according to location — for example, Outdoor mode boosts those frequencies that tend to get lost outside.

Most requirements are catered for here, including the option to use the speaker as a floor wedge at 45 degrees, or on a pole mount.

The Class‑D amplifiers and switch‑mode PSU ensure low weight and minimal heat dissipation while boasting

which helps focus the power where it is most needed. At nominal performance volumes, the ShowBox system can run for up to 12 hours on the included lithium‑ion battery pack. The battery charges in around three hours when mains power is connected, or around four hours if the amp is in use at the time.

The connectors are located on a recessed, metal rear panel, where you’ll find the IEC mains inlet and power switch, inputs for all the channels, send and return jacks, footswitch jacks, a headphone jack and the XLR mix out. At the top of the panel are the voicing selection buttons, USB‑C interface port, the Cat5 connector for the breakaway module and the micro‑SD card slot for recording. There are also buttons to engage the feedback eliminator and to switch the front-panel LED on or off.

A 3.5 mm aux input next to the Bluetooth pairing button is available for connecting phones, iPads and the like.

Ejected using a large ‘push to release’ button, the lightweight breakaway module

“That removable mixer panel is a great idea and makes self-mixing from the stage (or corner of the pub) much easier...”

The Tour Each of the four main channels is able to host its own effects, with a maximum of six effects deployed at the same time: one insert effect per channel and two further send effects, drawn from a selection of reverbs and delays. The insert effects include delays, reverbs, modulation and guitar distortion-type treatments. Three-band EQ (80Hz, 2.5kHz and 12kHz) and single-knob compression is available on each channel, including the stereo inputs 5+6. Should you want to add further effects there’s an external send/return loop, which you can send to from any of the mono inputs.

A feedback eliminator can activated by hitting a rear-panel button when ringing out the system, and works by deploying up to eight narrow notch filters.

My tests confirmed that it worked as intended, with no serious impact on the overall tonality. Ringing out does mean creating deliberate feedback, so that’s best done before your audience turn up, otherwise they might be inclined to leave again! This, of course, is true of any feedback-reduction system working on the same principle.

The onboard DSP offers a choice of switchable indoor or outdoor voicing as well as Amp or PA voicing. Amp voicing emphasises the mids slightly while PA voicing imposes a gentle smile curve, dipping the mids. Personally, I would have liked a flat option too! A separate button

an impressive peak power rating of 400 Watts, of which 350W goes to the eight-inch woofer and 50W to the one-inch titanium-diaphragm compression driver. The frequency response is quoted as 55Hz to 20kHz at the -3dB points, with a maximum SPL of 122dB at one metre.

The compression driver flare is designed to provide a 60‑degree vertical dispersion and a 90‑degree horizontal dispersion,

The breakaway control unit can be attached to a mic stand using the supplied mount.

www.soundonsound.com / April 2025

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MACKIE SHOWBOX

The first four (mono) channels comprise two mic and two instrument inputs, while channels 5+6 are for stereo line sources. There’s also an external effects send/return loop, a footswitch jack, a headphone out, a mix output, an aux mini-jack input, a USB-C port for computer interfacing, a micro SD card slot, and an Ethernet port for connecting the control unit remotely.

is made from tough plastic, and a mic stand adaptor is included so that it can be mounted securely part way down a mic stand. There’s a CAT5 connector on the rear edge along with the USB charging point.

SD card recording is activated using a large button on the breakaway module.

Take A Break The breakaway module has a friendly and intuitive layout, making good use of status LEDs for the buttons and around the relevant rotaries. The one stereo and four mono channels each have Select buttons to bring your chosen one under the control of the rotary/push controls, which offer adjustment of gain, compression and low, mid and high EQ, plus controls for FX 1 and FX 2. The integral push switches on these controls activate the tuner and snapshot functions respectively.

As a general rule, a quick double‑press bypasses or mutes the function as appropriate. LEDs around the knob show sharp or flat in tuner mode, while the mini display shows the note name.

Up to five snapshots can be saved and recalled. Other control presses allow the compressor threshold and so on to be adjusted. Two small amber displays show details of the effects or snapshots, and there’s also a level control for sending to the effects loop. In a separate section to the right are master controls for SD record, main volume with mute, headphone level, loop (return) level and the looper, which follows the usual routine of hold to clear, press to record, press to play, press to overdub and double-press to stop.

An automatic gain setting feature on the inputs reads the incoming signal level and then adjusts the gain to be as high as possible without the risk of clipping. This is activated by holding

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down the volume control; a brief press gets you back into manual gain mode, where a red peak LED shows if things are running too hot.

Impressions I tested the Showbox at a pub gig and, with some EQ tweaking, it gave us a well-balanced vocal sound and a solid acoustic guitar sound, with good resistance to feedback. When you dial in a dirty guitar effect such as Overdrive or Crunch, there’s no nasty fuzziness, so it seems that some form of speaker emulation must be built into those effects. However, I did find that to get the electric guitar to sound tonally balanced, I needed to add mid boost and a touch of treble while cutting back the bass. With the EQ set flat, I felt that the sound was somewhat bass-heavy and rather dull, though the room acoustics and the way the speaker is mounted will also affect the tonality. Likewise, although my usual vocal mic is well balanced, through the ShowBox, with the EQ set flat, the sound

seemed lacking in presence and again a little over-generous at the low end. A couple of EQ tweaks to bring up the mids and highs while pulling back the bass end gave me a nicely ‘present’ vocal sound, with plenty of EQ range left if needed.

It should be noted that there’s no phantom power available, and it can be tricky to operate the looper without the optional footswitch, because it is usually necessary to press the button at exactly the same time as you start playing. That being the case, I’d recommend adding a dual footswitch, as in a live situation it is also useful to be able to kill the effects when chatting between songs, and the second footswitch will let you do just that. Undo and redo would have been nice too, but then the looper is an extra so you can’t really expect all the functionality of a dedicated unit.

In all, then, the Mackie Showbox is a very flexible, self‑contained and eminently portable sound system for small pub/coffee shop gigs, garden parties and quality busking where mains power is not available. It accommodates acoustic guitars (with suitable pickup systems of course), electric guitars, keyboards and microphones. That removable mixer panel is a great idea and makes self-mixing from the stage (or corner of the pub) much easier without having to rely on phone or tablet apps. The ability to record to SD card and to hook the ShowBox up to a computer for live streaming is also welcome, as is the useful range of effects that can be independently deployed for each of the four main channels. I also found the Bluetooth playback to sound pretty good with the tone controls set close to flat, and I had no trouble getting the ShowBox paired with my Mac laptop.

The rechargeable battery is also included in the price, and it is easy to remove should it ever need replacing. All in all, an impressive package. $ $799 T Mackie +1 800 258 6883 E support@mackie.com W www.mackie.com

ON TE ST

Cycling ’74 Max 9 Programming Environment

Max 9 introduces Ableton devices and brings a host of welcome updates and improvements.

NICK ROTHWELL

I

t’s a half-decade since the last major release of Cycling ’74’s Max media programming environment, a period which saw a pandemic (giving many of us much more time in our home studios) and Apple’s shift to their own ARM-based silicon for the Macintosh. The previous release of Max (version 8) came shortly after Cycling ’74’s acquisition by Ableton, so we’ve also had five more years for that partnership to deepen, with ever more technological crossover between Max and Ableton Live.

So, how does the Max landscape look with the release of version 9?

Ableton Devices For a software platform aimed at building audio processing systems, one could argue that Max actually doesn’t make the process that easy, at least from the perspective of the jobbing electronic musician. Recording and manipulating audio buffers is relatively straightforward, but if you want to dive into

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the construction of non-trivial oscillator and filter systems or effects processing, perhaps adding control parameters and modulation on the way, things get complicated, and creating a good-sounding synthesizer voice or effects system is not a small amount of work.

Way back in the days of Max 7, Matthew Davidson developed a library called BEAP for teaching electroacoustic composition at Berklee, with graphical control panels allowing simple modular synthesizers to be constructed easily. In Max 9 we now have a more sophisticated library of native objects, under the generic name of ABL, which have been ported more or less directly from Ableton Live, including some freshly minted offerings from Live 12. Unlike BEAP, these objects don’t have graphical user interfaces — you’ll need to build those yourself (and there is some new UI machinery to help with that, as we’ll see).

What ABL offers instead is high-quality audio generation and processing at the level of fidelity and detail you’d find in any modern DAW. The ABL objects fall into two categories: a set of 55 (so far) low-level ‘DSP’ objects with names prefixed by ‘abl.dsp’, and 12 high‑level ‘device’ objects prefixed by ‘abl. device’. The selection of DSP devices includes traditional audio functions (oscillators, filters, phaser, flanger, ring modulator, some simple reverbs). Some of these operate in mono, some are stereo. All

are accompanied by comprehensive help patchers which explain their parameters and demonstrate their features.

The DSP objects are drawn from library components developed for devices in Ableton Live: for example, darkhall, quartz, shimmer, tides and prism are the five algorithms from Live’s new Hybrid Reverb device, whilst all two dozen or so oscillator types from Live’s Meld instrument are present as discrete objects. Meld in particular has been well received by Live users as an appealing and versatile sound source, so having access to its oscillators within Max really expands the kinds of synthesis that are possible without all the hard work.

Quite apart from the audio generating and processing objects, there are various LFOs, modulators, envelope followers and other components to allow you to build out sophisticated processing units, although something of the complexity of Live’s Meld or Roar would still be quite an undertaking. I was pleased to find that the DSP objects are ‘MC-aware’, which means that a multi‑channel audio feed can be split across an array of such objects running in parallel, with a spread of parameter settings.

So, if a granular oscillator like Swarm isn’t rich enough, mc.abl.swarm~ will deliver an entire swarm of swarms, while an MC instance of a simple FM oscillator offers the potential of some rich, multi-tonal additive and FM layering.

Cycling ’74 Max 9 $399 pros • Full integration of a comprehensive suite of Ableton Live DSP components and devices.

• A solid reboot of JavaScript support, with a modern V8 engine.

• Code boxes for JavaScript, dictionaries and more.

• A REPL feature for live coding and debugging scripts and code.

cons • Only one Ableton instrument (Drift) included in the ABL library.

• The REPL needs a bit of work to make it more user-friendly.

• No trackpad support for macOS.

summary Max 9 is a solid upgrade, with the JavaScript enhancements, REPL and code box features appealing to seasoned Max coders. The stand-out new feature is the integration of DSP objects and devices ported from Ableton Live, which brings high‑quality, easy-to-use DAW-style audio processing to Max for the first time.

The abl.dsp objects are powerful additions to Max’s audio processing arsenal, but with the possible exception of the obvious ‘effects’ objects they aren’t really full devices in the Ableton Live sense: apart from anything else, the oscillator objects are not polyphonic. However, ABL also brings us a handful of ‘full’ devices, including a complete version of Live 12’s Roar saturation/distortion effect. Instruments are, sadly, a little thin on the ground: we have abl.device.drift~, the analogue-style synth that arrived with Live 11. Drift in Live has a simple modulation system: three source/amount/destination slots rather than a complete matrix. This is reproduced in Max. As an instrument, Drift is fully polyphonic internally (though it will also operate monophonically or in unison mode), which means that it needs to be fed actual MIDI messages (a sequence of byte values) rather than frequency values like the ABL oscillators. In that respect, Drift behaves similarly to embedded VST instruments.

Sadly, we don’t get a complete port of Meld, perhaps thankfully as it is pretty complex, and the modulation matrix alone would be a challenge to manipulate in Max. I can imagine there’s also demand for Live’s venerable Operator FM synth, but this wasn’t built with Ableton’s latest libraries, making porting across to Max infeasible.

since the early 1990s when MIDI message processing was the only game in town. The arrival of the Jitter visual system, with its suite of complex 3D graphics and video processing objects, required Invisible linkage between control knobs and an ABL DSP object. the introduction of named object attributes, which in turn except that it looks and behaves like an led to dedicated attrui (attribute interface) actual control. Unlike attrui (but rather objects. The original preset object, used like the original preset object), there’s to store and retrieve object values, was no patch cord between control and ABL superseded by the pattr system. Complex object, making the connection process data values led to dictionary processing a little fiddly: you have to choose from and messages. And this is all before the a potentially enormous menu of target additional complexity caused by Max For devices and parameters. And the Live controls and automation. The arrival of connection is not very discoverable: you the ABL objects, with their large numbers have to Option-hover the mouse pointer of parameters and complete lack of visual over a control to see what it might be interface, has led to another kind of data mapped to. linkage: parameter connection. This new V8 & Code Boxes feature allows knob-based control panels to be built around ABL objects without the Max supports JavaScript coding in two traditional plumbing of patch cords, ‘set’ ways: the node.script object for hosting messages to avoid data loops, and so on. a Node.js server (for networking, database User interface objects (dials, sliders, access and more) as well as the js object toggle buttons) can be invisibly attached for scripting inside Max itself. While Node. to ABL device parameters, and once js uses a modern, sleek JavaScript variant, Patching attached will automatically track changes. js is tied to an old version of the language In this respect, the UI control behaves with a rather archaic syntax, unable to run The sophistication and capabilities of very similarly to an attached attrui object, modern code. Rather than an updated js, Max objects have increased immensely which might break legacy code, Max 9 provides a brand new object called V8, named for the V8 JavaScript engine it uses (the same as the Chrome browser and Node.js). V8 JavaScript is vastly more efficient than the legacy engine (a quoted speed‑up figure is a factor of 25 times).

The v8ui object supports graphics and interaction, the equivalent of the legacy jsui object. v8ui comes with support for PointerEvents, a standard API supporting digitiser tablets (including pressure and tilt) and multi-finger touch. For years, Max users have been making the most of experimental and unreliable objects for multi-touch, so proper support is welcome.

Unfortunately it’s still a work in progress, and there’s no multi-touch support yet for us MacBook users with our large, responsive trackpads. One point to note about V8: just like the legacy js object, it runs in Max’s user interface thread, not the high-priority scheduler thread, so it’s not ideal for anything requiring really accurate The Ableton Drift synthesizer driving a guitar pedal distortion effect. timing, and if embedded in a patcher www.soundonsound.com / April 2025

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C Y C L I N G ’ 74 M A X 9

This provides access to a REPL — a read-eval-print loop — which is a line-by-line interface to Max objects, patchers, help information and an additional dedicated JavaScript engine. This seems to be a nod to the Live Coding movement, where musicians improvise on stage by writing code live (and showing it to the Some sine and cosine list generation in a JavaScript code box. audience as they go). The Max REPL is that runs at high priority, there’s a risk of a bit of a polymath: it can send messages messages being transmitted in a different to Max objects just as if they came in via order than would happen in ‘pure’ Max message boxes, it can interrogate V8 code. This isn’t a showstopper, but requires JavaScript instances, and can evaluate care when patching. one-line JavaScript expressions. One of The arrival of a modern JavaScript its coolest tricks is to redirect the result of engine is accompanied by user interface one segment of code into another, so if enhancements to improve code editing you want to (say) calculate something in across the board. Code boxes are blocks JavaScript (a random number, say, or the of scripting text embedded directly inside number of milliseconds since the start of Max patchers, useful for one-liners or the year 1970) and send that to a named very short segments of code. Max 8 Max object in a patcher, it’s easy to do. featured code boxes in gen and RNBO, The state of the art in text-based two embedded toolchains for working coding environments has advanced with lower-level DSP code. In Max 9, code considerably in the last few years, with boxes have spread to Max more generally: editors like the popular Visual Studio they can contain JavaScript (both for V8 Code offering syntax highlighting and for Node.js), dictionaries, colls (indexed and ‘IntelliSense’, a posh name for tables of values) and OSC messages, as autocompletion of names, pop-up hints well as code related to graphics rendering. of function arguments, and so on. Judged The Node.js case is complicated slightly by against this standard, Max’s REPL feels its reliance on a separate Node process, spartan: without autocomplete you have to which still needs to be started (and type the full name of any object you want potentially stopped), and Node also reads to talk to (although there is also a pop-up its scripts from the file system, so there’s menu which lists them) and the name of a bit of machinery behind the scenes to any parameter you might want to alter. save the code box content somewhere Some text colouring would make it easier Node can find it. to read lines of code which mix JavaScript and Max elements. But if you’re writing REPL your own patchers and V8 objects, and Max has always been a graphical environment, which appeals to non-programmers who often find text-based systems too challenging or difficult to learn. The JavaScript objects allow Max coders to drop into a textual language for calculations which might be complicated or inconvenient to code up in a graphical system, but there’s now an additional text-based interface sitting snugly underneath the Max console.

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want a focal point to create messages and exercise or debug your code, REPL might well help. One day soon we might even see a brave live coder play a gig purely by typing lines into the REPL text area.

The Best Of The Rest I’ll round off this brief overview of Max 9 by mentioning some of the general interface improvements. There’s a ‘patcher list view’ pane, which lists all objects in a patcher, showing their arguments and scripting names in one place (useful as a reference when using the REPL). Text objects now have syntax colouring to distinguish object name, arguments and attributes (though integer and floating‑point arguments are the same colour, and it would be good to get an indication that an argument has been changed by an incoming message — these are the top two issues which confuse novice Max users).

There’s a debugging ‘illustration mode’ that animates a patcher’s message flow in cartoon-like slow motion. Graphical display objects like multisliders and function generators can now be shown with tasteful alpha-blended gradients.

Conclusion Having been through Max upgrades from version 2 all the way to the new Max 9, this new release feels like something of a solidification: Ableton’s DSP and device suite adds considerable heft to Max’s mainstream audio processing, while the code box objects blur the sometimes awkward division between patching and text-based coding, and add more visibility to storage objects like colls and dictionaries. The new REPL is a more esoteric feature, and still in need of some refinement, but looks as if it could steer the whole patch development process in new directions. Overall, as the cliché goes, a no-brainer of an upgrade. $ $399 or $120 per year/$12.99 per month.

Permanent upgrade licence £199.

W www.cycling74.com

A JavaScript one-liner to populate the first 20 cubic numbers.

ON TE ST

Synthesizer, Sampler & Sequencer

Teenage Engineering OP-XY The OP-XY is a portable sequencing powerhouse.

Teenage Engineering OP-XY $2299 pros • Fantastic sequencing with step components and performance effects.

• Unlimited step components per step.

• Rich range of sampling and synth sources.

• Per-scene engine assignments.

• Brain and Punch FX generate interesting tracks from simple starting points.

cons • No export or multi‑channel output.

• Brain doesn’t work with the Aux MIDI/ CV outputs.

summary A genuinely useful and accessible portable workstation, marrying the OP-Z’s sequencing with the OP-1’s beauty.

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SIMON SHERBBOURNE

I

doubt I’m the only one who followed an arc from mad desire for Teenage Engineering’s OP-1, to getting one, to discovering that what I really needed wasn’t a portable synth with a four-track audio recorder but a portable synth with a multitrack sequencer. We got this with bells on with the OP-Z, but I wished it had the same form factor as the OP-1 with its own screen. The OP-XY delivers something close to this, evolving OP-Z DNA into a workstation-level groovebox/arranger, and leaving behind the motion graphics and lighting elements.

Let’s get this out of the way: the OP-XY is expensive. Not that this is weird: people pay a lot for really nice instruments and outboard gear, but it inevitably informs how you evaluate this device. I was particularly interested to know if the OP-XY has the

power to replace a dedicated sequencer in a desktop rig. You might want to know if it can get close to, say, an MPC level of DAWless production. Either way, most of us are probably looking at the XY lustfully, so let’s see if I can help justify an investment... or talk ourselves down.

Shades Of Grey The OP-XY is an eight-part, multi-engine synth and sampler, with 16 tracks of sequencing, split into two blocks of eight.

The first eight are tied to synth or MIDI parts.

The second eight (the ‘Aux’ tracks) access other features, like a CV/gate track, effects returns, performance effects and The Brain: an intelligent transposer.

Looks aside, this is a different approach to the OP-1, which has multiple instrument slots but can only play one at a time. And although the OP-1 has sequence players, the only way to capture anything is to record live into the four-track digital ‘tape’ machine. The XY, like the Z, is multitimbral, with 24 voices dynamically allocated

across eight parallel engines, powered by a dual-core CPU and DSP coprocessor.

The XY has the same gorgeous housing as the OP-1 and is roughly the size of a compact computer keyboard. The difference on the panel is that the natural keys have been shrunk to the same size as the accidentals to make room for a new row of step triggers. Visually the OP-XY is differentiated with a greyscale design and an ombré gradient across the trig keys.

This is mirrored by the high-res screen, which, though capable of full colour, stays resolutely monochrome apart from some sparing use of red. Red means recording.

Connectivity comprises dedicated audio in/out and MIDI input on TRS mini‑jacks.

Bluetooth MIDI and clock are available, but not Wi-Fi. USB-C provides charging of the internal battery, plus MIDI, audio and data interfacing. Then there’s an unusual multifunction output port that can be switched between audio, MIDI, CV and sync modes.

The keys feel like a computer keyboard but are velocity sensitive. The encoders are also buttons and have a clicky action, which is great for working with lists but makes synth tweaking inherently steppy.

The Samplers New projects start with two drum tracks and a selection of instruments, but you can reassign the tracks freely. The drum sampler feels familiar from the OP-1 Field, with 24 one-shot sample slots that can be triggered or gated. Having a polyphonic kit of sounds taking up just one track offers a lot of scope for your productions.

A major improvement is that the drum sampler uses individual samples for each key instead of a single sliced file. You can load — and sample — into each key directly, and the XY has a proper sample manager on board. Like the OP-Z, each track has a filter, and two sends to global effects.

There are three low-pass filter modes and a high-pass to choose from. Apart from the whistly resonance on the ladder they all sound similar to me. Comparing them isn’t straightforward as, annoyingly, changing mode resets all parameters. The filter gets its own envelope. Usefully, there’s an additional static high-pass filter tucked away in the instrument settings.

All the synths have four primary parameters and an animated display.

A significant limitation in the drum sampler is that, although each slot has its own pitch, mode and trim settings, all sounds share envelope and filter settings. This seems strange, as each voice has its own filter and VCA that follow the envelopes independently — it’s not a paraphonic situation. Hopefully this might be tweaked in the future.

The other thing that falls short compared to other premium sampling workstations is the lack of chopping/slicing functionality. To play chops from a sampled passage you need to fall back on copying the same sample to multiple keys and assigning different start points manually.

There’s no option to time‑stretch for tempo‑matching chops or loops.

Two further sample engines are available. Sampler is the same as the OP-1 sampler, providing chromatic playback of a single sample. Multisampler offers multiple key zones (but not velocity zones). The factory library showcases this with some lovely presets including strings and a piano.

When building patches you can sample or load to a key and the XY will automatically create zones. When importing it will also set root pitches based on file names.

Crossfaded loop points can be set, or read if there’s loop metadata.

schemes, but are preconfigured in particular ways and feature just four adjustable parameters with fun, and sometimes esoteric, animated graphics as feedback.

Despite limited controls, the synths reach a range of results, moreso than the handful of sweet spots on the OP-1 synths.

(The OP-1 synths often give up controls to a filter and effect, which is not necessary here). I particularly like the Simple engine, with its charming but essentially irrelevant animation of descending slinkies, and the Wavetable engine, with its more informative oscilloscope. Wavetable features eight sweepable tables, with a Warp (symmetry/PWM) control and modulation. The Drift control goes up to sync’ed audio rate level, producing awesome phase modulation reminiscent of Native Instruments’ Massive. It would make a great standalone Eurorack module.

Modulation is less restrictive than on the OP-1, as you don’t have to decide between MIDI and internal modulation sources. It’s still a single mapping within each engine from a source like the filter envelope, LFO, or the built-in accelerometer, but having velocity as a second mod is great, and if you are playing the unit from an external keyboard you can add aftertouch, mod wheel and pitch-bend to the mix.

The Synths

Sequencing & Step Components

The eight synths are new to the OP-XY, and are uniquely crafted sonic toys that defy easy definition. They use familiar synthesis

Each of the 16 tracks has a sequencer with independent length (up to 64 steps), scale (speed), gate length, quantise amount and groove depth. Note entry follows familiar conventions: you can record in real time, step record in stop, or select notes and drop them onto steps. Drum tracks show trigs from all sound lanes at once and there’s External sockets are found at the side and include a USB‑C port and 3.5mm sockets for audio I/O, MIDI in and a multi-function MIDI/CV/sync/audio out port.

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no on-screen pattern display, but holding a note and tapping the Record button temporarily hides other lanes.

Parameter automation is captured during recording and it’s possible to write parameter locks by holding a step and setting a value. Lock points are treated as vectors, so automation glides between surrounding values. To achieve a more Elektron-style lock you’d need to also anchor null values on the adjacent steps.

What’s really exciting about the OP-XY sequencer is its step component system.

I first encountered this idea on the OP-Z, but have since found similar features on my main hardware sequencer (the Five12 Vector) as well as the Polyend Play, Intellijel Metropolix and Sequentix Cirklon. Step components are actions that can be added to specific trigs to modify their behaviour.

An example of something like this on many sequencers is probability: setting a chance for a given trigger to fire that’s evaluated each cycle.

Ironically, the XY doesn’t do probability, perhaps to emphasise that there are more fun and musical ways to introduce variation. There are 14 step component types, each with 10 variations.

The first three components provide the functionality I highlighted recently when reviewing Five12’s Vector, and that’s central to the RYK M185 and related Intellijel sequencers. These Pulse (repeat), Hold or Multiply (ratchet) steps. They can be used in combination (any step can have all 14 components active) and the first two push the length of your sequence longer than its nominal step count. Uniquely, you can set the value of these components to random, generating a new rhythm every cycle.

Pitch can be randomised within defined limits, or transposed, but my favourites

by far are the ramp up/down operations.

These are what the Metropolix and Cirklon call ‘accumulators’, walking a pitch up or down by a set amount each cycle.

Portamento is also a step component, as is Bend which offers several pitch articulations.

Tonality can be set to transpose by interval, in scale, or to ignore chord changes (more on that later).

Jump Step can send the sequence to a different place, pause it on the current step indefinitely, or (cleverly) realign the track when it’s gone out of phase. Another component sets a trig to play intermittently.

Some components, you might think, seem destructive. Why not just change the pitch instead of transposing a fixed amount? And doesn’t setting a component to jump to an earlier step simply shorten the sequence?

can arrange a string of Scenes. Simply typing in numbers writes out the Scene sequence from the active step. Using the +/- buttons you can cue up a position in the Song during playback, and in this case the change waits until the end of the Scene. You can in fact create up to 10 different Song sequences within a project, and recall them from the bottom row of keys. You can set a Song running then switch to other views to play along, trigger punch effects and so on, although there’s no persistent indication of Song position.

Aux Tracks & The Multiport Let’s jump back a bit as we skipped over the Aux tracks on the way to Arrange mode. I’m going to save Aux tracks 1 and 2 for a moment. Aux 3 is a dedicated MIDI track for playing, sequencing and controlling an external instrument. The four option pages provide channel, bank and program select, two sets of four CC controls, and a CC mod LFO. At this point we can’t get much further without talking about the multiport. This is a 3.5mm TRS port that operates in a number of modes. Set to MIDI, it becomes a TRS MIDI output. Set to CV, it carries a combined CV and gate signal which you can access with a splitter cable. It also has modes for various flavours of analogue and Pocket Operator sync, and an audio mode. Aux 4 is the source used when the port is in CV mode. You can sequence CV here along with step components, although glide and pitch mods don’t work.

Aux 5 is an audio input. While testing the CV/gate output, I connected my external synth to the audio input and routed it through this track. This brought it into the mix and gave me access to the send effects, as well as a drive control. You can also use it for an external send loop through an effects unit or pedal. If you set the multiport as an audio output, all the instrument tracks get a third send that goes out via the multiport, then back in via Aux 5.

Aux 6 is Tape: not a tape recorder like on the OP-1, but a sampling tape loop effect, familiar from the OP-Z. This is particularly useful as a stutter and glitch effect, looping quantised slices of the sampled range triggered by the note keys. You can also do experimental stuff like warping pitch and time, and applying an LFO to parameters.

“I had endless fun setting up short, simple sequences, loading them up with components like the pitch ramps, mapping out a Brain chord progression, then seasoning with Punch FX.” This is where the Component Skip component comes in, allowing you to set other operations on a step to only trigger every other time, or every third time, etc.

Bigger Picture

To progress your ideas there’s a structure of Patterns, Scenes and Songs. Arrange mode presents a horizontal row of clips representing the Patterns in your tracks.

Multiple Patterns in tracks are shown as stacks, which fan out into a column when the track is focused. You can then use an encoder to flip between Patterns.

Shift and the numeric keys recall Scenes (up to 96), which store a combination of active Patterns, track mute and Brain states, and even which engine is assigned. Changes update the current scene destructively.

Swapping Scenes, or Patterns within Scenes, is instant and playback continues from the current step, which is exactly how I like it.

Finally, Song mode: a 32‑slot grid where you A suite of mix tools and MIDI effects make the XY eminently performable.

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The vertical portion of the cross to the right-hand side is a level meter, and the barely visible protrusion at the bottom left is a pitch-bend pad.

Individual tracks can be included or excluded from the tape loop.

The last two auxes are effects returns.

Although these default to delay and reverb, either can be swapped out for chorus, lo‑fi, distortion or phaser. It’s particularly effective to replace the delay with distortion or lo‑fi, as the OP-XY can sound pretty clean. This lends itself to beautiful ambient tracks, but if you want something that will shake the floor a bit you can grunge things up here.

The Brain & Punch FX Looping back to Aux 1 and 2, we have some OP-Z-style magic with The Brain, and Punch FX. These complete the workflow picture for the OP-XY, providing performable (and sequenceable) real‑time progressions, variations and fills.

The Brain is a transposition engine that detects the key and scale of your project then lets you manipulate it in musically coherent ways. For example, it may decide that your piece conforms to C major. The Middle C key will become the home position for playback in The Brain. If you hold the E key above Middle C, your piece will play back two tones higher diatonically; that is to say, still using the notes in the C major scale.

It was slightly confusing figuring this out as The Brain expresses what it’s doing in terms of modes. So in the above example, the Brain will say it’s moved to E Phrygian, which is a fancy way of saying the notes of the C major scale starting from E. It’s easier to spot if you move to A, as it shows this as a transposition to A minor, which shares the same notes as C major.

You can be more prescriptive about what Brain does. First you can specify the key and scale instead of letting Brain guess. Second, if you hold a chord instead of a note, you

can force progressions that break out of the base scale, for example moving from major to minor on the same root.

Brain provides a new yet oldschool way of working that I really got into. You can capture some simple sequences like a bass line and a single held chord, then create a chord progression in Brain. Just turn the scale (speed) of Brain’s sequence track right down and sketch out song sections on a single page. Then string together multiple Patterns into a song, all in one track. You can choose which tracks are included: Tracks 1 and 2 default to off, for example, as they get preloaded with Drum Samplers. One large omission is that the Aux MIDI and CV tracks are not affected by The Brain, which limits this workflow when using external instruments. However, it does work over MIDI sent from the main tracks.

If you’ve played with the OP-Z, Pocket Operators or the EP-133 you’ll know how great Punch FX are. The Aux 2 track on OP-XY is dedicated to these momentary performance modifiers. Every note key holds one of the effects, which range from stops and stutters to rhythmic and melodic variations. There are 12 effects, repeated across the two octaves of the keyboard, allowing the Aux 2 track to apply Punch FX to melodic and percussive tracks as separate groups. (The distinction is also used by the master section of the mixer to balance drums against other instruments).

It’s possible to apply (and sequence) the Punch FX to any track individually while it’s focused.

Conclusion The OP-XY is deep, and I’ve not even covered some key features like players, grooves and tuning schemes. It doesn’t

have the level of core beatmaking, production and fine editing potential as an MPC or Maschine, but at the same time its dynamic, modular-inspired sequencing makes these classic workstations look staid. I was excited to see how much the sequencer could be used for external devices, and was disappointed to learn that the multiport can only do one thing at a time. It’s great that it can do CV, and incorporate external processing, and MIDI, but only one of these at a time. This is mitigated by being able to use any track for MIDI, and as I discovered, you can connect a USB thru device like the Retrokits RK-006 to break out to multiple externals.

Unless a classic linear recording workflow appeals to you, the XY is way more accessible and productive for creating songs and performances than the OP-1. The Brain and step components make the OP-XY both a generative sequencer and songwriting assistant.

I had endless fun setting up short, simple sequences, loading them up with components like the pitch ramps, mapping out a Brain chord progression, then seasoning with Punch FX. It is however a closed environment at the moment in terms of progressing a project. There’s no way to export or bounce stems, and no multi‑channel USB output.

So, the OP-XY doesn’t qualify as the mythical ‘one device to rule them all’ if that’s how you’re hoping to fund it. But it must be the most functional dedicated device of its size, it’s a beautiful thing, and it certainly inspires you to make music. $ $2299 W teenage.engineering www.soundonsound.com / April 2025

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USB Audio Interface

PreSonus Quantum HD8 SAM INGLIS

W

hen PreSonus first attached the name Quantum to a 1U audio interface, it was not short of unique selling points. Back in 2017, Quantums (or should that be Quanta?) connected to the host computer over Thunderbolt, and took full advantage of that protocol’s special powers. So much so, in fact, that even eight years later, the Thunderbolt Quantum interfaces are still unmatched for low-latency performance, and nothing since has come close to the Quantum 4848 for huge I/O counts on a project-studio budget. When the time came to refresh the Quantum range last year, however, PreSonus nevertheless gave up these advantages, switching to USB connectivity in pursuit of wider compatibility and lower cost.

There are obvious parallels here to what Focusrite have done with the Clarett range. In the first generation of Thunderbolt Clarett interfaces, the focus was on low-latency performance; but when the Clarett+ refresh took place in 2021, Thunderbolt was dropped, and the current models are all USB. Clarett+

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With their latest Quantum interfaces, PreSonus have swapped Thunderbolt for USB. Do they still maintain the same advantages? became the ‘premium’ alternative to Focusrite’s bestselling Scarlett line, offering additional features and better specs rather than a fundamentally different platform. PreSonus’ move with the Quantum interfaces looks very similar, positioning them as upmarket alternatives to the Studio C range. So, what do the USB Quantums offer that’s special, and why should you buy one of these rather than its cheaper Studio C counterpart?

Four Front There is a total of four models in the USB Quantum range (and the Thunderbolt Quantum 2626 is still a current product, too). The ES2 and ES4 occupy a wedge-shaped, desktop form factor, whilst the HD2 and HD8 are 1U half- and full-rack devices respectively. PreSonus sent all four models for review: I’ll focus mainly on the flagship HD8 and note any relevant differences where applicable.

Build quality throughout the range is reassuringly robust, with stout metal

shells and chunky faceplates to the fore.

Everything is finished in a smart and businesslike black, though some of the legending on the rackmount models is quite small and hard to read at a distance.

All four models use a new mic preamp design dubbed Max-HD, and I can’t discover any feature with the initials ES, so the thinking behind PreSonus’ naming conventions is a little obscure. The main functional differences are as follows. The ES models are USB powered, whilst the HD2 requires an external DC supply and the HD8 accepts an IEC mains cable. The ES variants have mini-jack MIDI in and out, whilst the HD Quantums have a DB9 connector for a flying cable assembly

PreSonus Quantum HD8 $799 pros • Excellent audio specifications, with particularly impressive mic preamps.

• Very good low-latency performance.

• Intuitive mixer and control panel utility.

• Efficient auto-gain function.

• HD models have dedicated re-amp outputs.

cons • Limited monitor control and ‘master section’ features.

• Re-amp outputs can only be addressed from your DAW.

summary The new Quantums offer a potent combination of sound quality, slick design and excellent low-latency performance.

complemented on the HD models by the intriguing addition of a pair of front-panel jack outputs specifically designed for re-amping. The remaining analogue outputs are all on balanced jacks, of which the ES2 has two, the HD2 and ES4 four, and the HD no fewer than 10.

Universal Credit

that also features coaxial S/PDIF in and out. Both HD models can access up to 16 channels of ADAT Lightpipe digital I/O on two pairs of optical Toslink connectors; the HD8 also has word clock in and out, whilst the ES units have no digital audio expansion. The HD8 can also be used without a computer connection as a standalone ADAT expander.

On the analogue front, the ES2 and HD2 both have two mic/line inputs on combi XLR/jack sockets, while the ES4 has four and the HD8 eight. The first input on the ES2 and the first pair on the others also have dedicated high-impedance jack sockets on the front panel, which override the rear-panel combi sockets when a guitar or bass is plugged in. These are

It’s obvious from the moment you power up one of the new Quantums that PreSonus have gone all-in on digital control. There is nary an analogue pot, button or switch to be found anywhere.

Preamps, cue mixing, monitor control, metering and even headphone level are set digitally, using a single master encoder. This delivers visual feedback using a ring of coloured LEDs, and has a push action to confirm parameter settings. It is accompanied by a small but detailed colour display, which does multiple duty as a meterbridge, numerical readout and so on. The entire state of a Quantum interface, including all preamp settings, can be stored and recalled as a scene in the companion Universal Control software. This is available for both macOS and Windows, but note that versions of macOS prior to 10.12 are not supported.

The back panel includes an IEC power input, optical ports, a breakout port for MIDI and S/PDIF sockets, eight quarter-inch line outputs, a pair of main outputs, and eight combi jack/XLR audio inputs.

After a slow start, the vast majority of audio interface designers now understand that the principles of good user interface design apply to control panel software just as much as they do in DAWs and other applications. Universal Control is an excellent example, presenting a tidy, friendly and intuitive interface whilst offering a surprising degree of control and flexibility. The vast majority of your time will be spent in the mixer view, which for some reason isn’t the default, but is a pleasure to use once discovered. Nice touches abound, such as the Logic-style graphical icons that can be used in addition to names on each channel, but it never looks cluttered or intimidating.

(And, talking of Logic, Quantum preamp parameters can be adjusted directly from the Logic mixer.)

Rather than present separate mixer panes for each output pair, PreSonus have chosen a single-window model, with a channel for each physical input. The main mix output is hardwired to the main monitor output pair, but each channel can have up to 14 aux sends, allowing it to be distributed both to other physical outputs and to two stereo loopback busses.

Adjacent to the main output fader you’ll also find a strip labelled Returns; here, you can select an output pair and hit Fader Flip, whereupon the channel faders will turn green and reposition themselves to reflect the level of the appropriate aux send. Digital and loopback channels can be hidden, as can sends to the ADAT channels. The Quantums’ headphone outputs aren’t independently addressable from your DAW, but can be switched to mirror either the main mixer output or any of the other output pairs.

As with most audio interfaces, there are enough monitor control features to be useful, but not enough that you won’t sometimes wish for a dedicated monitor controller. A Mute button can be repurposed as a dim if you prefer, and there are some software-switched options as to what the main encoder should control. It defaults to adjusting the level only at the main monitor outputs, but it’s possible to have it simultaneously

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All settings can be stored with the Universal Control software.

adjust the level of outputs 1-4, 1-6, 1-8, or of all outputs including the ADAT outs.

This would allow basic level control over a surround speaker system, though any calibration would have to be carried out elsewhere. Alternatively, you can engage a speaker switching mode, in which outputs 3+4 are assigned to a second pair of monitors; there’s no option to apply a trim offset to compensate for level differences, though.

The Quantums also lack other common ‘master section’ features such as talkback and mono fold-down.

which is about as quiet as preamps get, and an absolutely colossal gain range.

The digital gain control for each input spans 0 to 75 dB in 1dB steps, and on top of this, there’s also a switchable -20dB pad, so the difference between

gain is matched for stereo recording, but this can’t be done from the front panel.

The Quantums’ line-level I/O is aligned such that 0dBFS corresponds to an analogue level of +18dBu. This is a reasonably common standard, though you may have an easier time working with professional outboard if your interface supports +24dBu line-up levels. Modular aficionados will be pleased to learn that the line outputs are DC-coupled, and hence capable of transmitting control voltages; the same is, however, not true of the inputs. The headphone outputs, meanwhile, are specified for +11dBu into 50Ω, which is a relatively conservative level. Their output impedance of 33Ω is also higher than the norm these days, when most manufacturers seek to present an impedance as close to zero as possible in order to drive low-impedance headphones without loading them too much. Perhaps it was my imagination, but I found the headphone amps a little brighter than I was expecting with some cans.

The most unusual feature of the Quantum HD models is the provision of dedicated re-amping outputs. These are, as you’d expect, unbalanced, with a 100Ω output impedance, and are conveniently located on the front panel directly below the instrument inputs for channels 1+2. Slightly to my surprise, though, there is no way to mirror what’s coming in at those inputs on the re-amp outputs. They don’t appear as destinations in the internal mixer, so you can’t create a real-time, low-latency send from, say, input 1 to re-amp output 1. The only way to address them is from your DAW, so if you want to simultaneously record the DI input and a re-amped output, you’ll have to set up the appropriate routing in software and manage any timing offsets caused by latency after the fact.

Even so, it’s a whole lot better than not having dedicated re-amp outputs, as is the case with the vast majority of rival interfaces. These ones sounded good to me, and worked well.

“The second-generation Quantums should be judged not against their Thunderbolt predecessors, but against rival products.

And they stand up very well indeed.”

To The Max One of the most important differentiators for ‘premium’ interfaces is sound quality, at least as it is reflected in measurements and specifications. When even relatively affordable interfaces like the Focusrite Scarlett 4th Gen models offer specs that would have been considered state of the art a few years ago, it’s a challenge to stay ahead of the pack, but the new Quantums deliver some impressive numbers. Dynamic range on the line inputs and outputs is 118 and 123 dB (A-weighted) respectively, but the most eye-catching measurements concern PreSonus’ new Max-HD preamp design.

This has an A-weighted Equivalent Input Noise of -130dBu at maximum gain,

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minimum and maximum input gain is effectively 95dB. I’m not aware of any standalone mic preamps that span such a huge range, never mind those built into audio interfaces.

The icing on this particularly well-risen cake is that, like UA, Focusrite and Audient, PreSonus have also implemented auto-gain. This is easy to activate from the front panel: press Auto, use the numeric buttons to select which preamps you want to include, and press Auto again. The LED ring around the main encoder will then turn red, whereupon you have about eight seconds to pick up your chosen instrument and make a suitable racket. As with some rival implementations, this makes it hard to use if for example you need to go and sit at a drum kit. Adjacent odd/even input pairs can be linked in Universal Control so that

Short Trips Talking of latency, the Thunderbolt Quantums offered such blazingly fast performance that PreSonus didn’t feel it necessary to include an internal mixer.

With round-trip latencies of under 2ms

achievable even at base sample rates, cue mixing could be handled perfectly well in your DAW. In fact, as long as you avoided using plug-ins with look‑ahead, round-trip delay times through the DAW could be lower than was possible using the ‘low-latency’ mixers in some interfaces, including PreSonus’ own Studio 192.

Given that the new Quantums have abandoned Thunderbolt for USB, and that they do have an internal mixer, I was fully prepared to report a backward step on the latency front. Nearly all USB interfaces nowadays use Apple’s built-in Core Audio driver; with a 32-sample buffer, this usually delivers round‑trip figures of around 7ms at 44.1kHz, plus whatever is added by the internal mixer. That’s unproblematic in most circumstances, at least as long as your DAW session leaves enough CPU resources free to handle the processing overhead of running with the lowest buffer size, but it’s still significantly less good.

I was expecting the new Quantums to offer comparable performance, but I was pleasantly surprised to discover that Reaper reported input and output latencies

of just 1.4 and 1.8 milliseconds respectively for a 32-sample buffer. Re-recording an audio click showed that although this reported 3.2ms round trip latency was not perfectly accurate, it was only out by about 14 samples, this presumably representing the delay through the mixer. A round-trip latency of about 3.5ms is thus achievable on macOS even at base sample rates.

This may not be quite as good as the Thunderbolt Quantums, but it equals or betters pretty much every other interface I’ve tested, including other Thunderbolt models. Full marks to PreSonus’ engineers for not being content with ‘good enough’ on this front.

In practice, then, I think the only retrograde step from the Thunderbolt Quantums concerns I/O count. Thanks to its dual ADAT ports and S/PDIF I/O, the HD8 offers a total of 26 physical inputs at base sample rates. That’s as good as all of its direct rivals, and more than enough for most purposes; moreover, I’m told that if you expand an HD8 using another HD8 over ADAT, the second unit can be remote controlled from within Universal Control.

It’s a shame to see the Quantum 4848’s market niche stand empty, but I’m not going to view PreSonus harshly for this. Practically every manufacturer has abandoned Thunderbolt at this price point, and factors such as component costs and certification headaches are out of their control.

Ultimately, the second-generation Quantums should be judged not against their Thunderbolt predecessors, but against rival products. And they stand up very well indeed, with particular highlights being the Max-HD preamp design, the unusually good low-latency performance, the HD models’ dedicated re-amping outputs — and their highly competitive pricing. You also get a full year’s worth of PreSonus’ Studio One Professional thrown in for free.

Rather than striking out on their own, PreSonus are now taking other interface manufacturers on at their own game, and making a very good fist of it. $ $799.99 W www.presonus.com

Ag

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Polyverse Music Filterverse Filter Plug-in

JOHN WALDEN

A

filter is far from the only element to define a synth’s sound-design potential, but I’m sure most experienced synthologists will have expressed the ‘nice synth, shame about

Polyverse Music Filterverse $149 pros • A whole universe of filter types in a single plug-in.

• Powerful but easy‑to‑use modulation options.

• Fabulous and inspiring collection of presets included.

cons • Not a casual purchase (but a fair price for what’s on offer).

summary With an impressive array of filter types, a powerful modulation system, and a slick UI, in this universe at least, Filterverse is about as good as it gets for creative sound design.

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There’s creative potential lurking in every filter plug-in, but in this case the possibilities are almost boundless!

the filter’ sentiment on occasion! And since filters have creative uses for all sorts of sound sources, it’s just as much of a shame when you discover that the powerful filter inside your favourite synth isn’t available to process your guitar, drum or bass tracks. All of which means there’s a lot to be said for having a good, dedicated filter in your plug-in effects folder.

As Supermodal (reviewed in SOS May 2023: https://sosm. ag/polyverse-supermodal) ably demonstrated, Polyverse already knew a fair bit about designing creative filter plug-ins, but recently they launched a new multi-filter plug-in called Filterverse, which supports all the major plug-in formats for Windows and macOS.

Three From Many As shown in the first screen, the Filterverse GUI is divided into four main areas. The strip at the base provides access to the included presets, global Dry/Wet and Main Output controls (these can be locked to ensure that

levels don’t jump out of hand when you switch presets), and a simple but effective clipper that can be set to on, off or soft clip.

The top strip provides slots for up to three filter choices, and a routing tab offers five different routing schemes for audio to pass through these three slots.

For each slot, there’s an impressive choice of 26 filter types, and you’re free to use these in any combination, including multiples of a particular type.

I won’t waste column inches discussing all the filter options available here, but this is a good place to note that Polyverse have done an impressive job with their PDF manual for Filterverse. It provides a full reference for each filter type, along with some interesting and educational background (for example, noting that a specific filter topology exists in a well-known hardware synth).

The filter collection includes a number of classic filter designs, of course, and the majority of these exhibit non-linear behaviour, saturating when ‘pushed’.

Equally, many of them are state-variable,

with controls to blend between states (for example, between low-pass, band-pass and high-pass types), through direct manipulation of the controls, automation, or via built‑in modulation options. These classic filter types are given names such as Swiss Army Knife, Bread & Butter, Evil, or Clear Glass — a useful hint as to their characteristics for those in the know, but if that’s not you don’t worry, as the manual has all the details.

There’s a further category of multi-peak filters capable of more complex processing. Names such as Dueling Bands (two band-pass filters, polarity inverted), Ripples (multiple narrow filters) and Robo Comb (nested comb filters) suggest what’s on offer, but there are some pretty cool options within this group, including some experimental ones.

Particularly interesting to me were Vowel, which models how mouth shape and tongue position influences vowel sounds; and Strings, which uses waveguide modelling to simulate three interacting strings.

Both of these did some very interesting things on a variety of sound sources during the review period... although I must confess that I’m not entirely sure I understood how they were doing it!

The final filter group contains the ‘non-filters’. These essentially transition Filterverse into multi-effects territory as they include options such as Space (a creative reverb filter), Decimate (providing sample rate and sample & hold options), Cabinator (a surprisingly effective guitar amp/cab effect) and delay-based options such as Tempolay, Tempong, TemPitch and yaleD opmeT (a type of reverse delay).

The upper half of the GUI shows the controls for the currently selected filter slot. Aside from providing access to the control set for the specific filter within

Filterverse offers a number of different routing options for its three filter slots.

that slot, the Cutoff control includes a neat graphical representation of the filter behaviour and serves as a visual guide while you familiarise yourself with the characteristics of each filter type.

The Filter In/Out, Dry/Wet, Pan, Cutoff and Resonance controls are present in all the filters. The additional sliders vary in number (up to a maximum of four) depending upon the filter type selected.

assigning to a rotary hardware controller using MIDI Learn, for hands-on control.

Note also that if your DAW supports the required routing then Filterverse can receive both MIDI note/controller data and (via a side-chain input) CV data.

Pretty much every parameter within the three filter sections can be designated as a target for modulation, targets can be subjected to multiple modulation sources, modulation sources can target multiple parameters and, yes, modulation sources can target other modulation sources (although, obviously, not themselves!).

There’s bags of potential here and, thankfully, it’s easily realised — the GUI makes the whole process of selecting modulation sources, adjusting their individual characteristics, and linking sources to targets wonderfully intuitive. Polyverse really have done a great job here.

A simple drop-down menu allows you to select a modulation source type for each of the eight slots. The slots themselves use a fixed colour-coding (slot 1 is always red, slot 2 is always orange, and so on) and, as we’ll see, this helps you keep track of which modulation sources are linked to which target parameters. By default, a compact control layout is available across all slots, but clicking on the horizontal dots at the base of a specific

“I’ve always been a fan of envelope followers, and Filterverse’s is easy to configure and very effective.”

It’s also true that, for some of the less ‘filtery’ of the filter types, the controls may actually represent parameters that are not conventionally found on a filter. Usefully, the labels adjust themselves to help you find your way around.

Modulation Sensation In the lower half of the GUI, Filterverse boasts some equally impressive modulation features. Here, you get up to eight slots to fill with any combination of the available modulation sources, of which there are also eight options: ADSR envelope, envelope/step sequencer, oscillator, envelope follower (yay!), random value generator, pitch detector (ooooh!), MIDI/CV input and Meta Knob.

The Meta Knob makes a great target for

With classic filters, multi-peak filters and even ‘non-filter’ filters, there are plenty of tools to get creative with.

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Eight modulation source slots provide plenty of ways to inject some movement into your sounds.

slot provides a wider view that allows access to any additional settings.

There’s an impressive level of control here across all the modulation source types. The most straightforward is the ADSR envelope, but the sequencer/ LFO envelope has plenty to offer, with manual or tempo sync options, a useful selection of preset curves/step patterns, variable step counts up to 16 steps, options to shape the transitions between steps, and swing. You can, of course, create your own modulation curves/ step sequences. I’ve always been a fan of envelope followers, and Filterverse’s is easy to configure and very effective.

It’s a great choice for letting your audio signal’s dynamics drive the modulation on one or more filter parameters, and the Attack and Release controls here allow for some interesting options when used on sounds with obvious transients.

Other modulators, for example the Pitch Detect, Random and Oscillator, perhaps require a little more experimentation but they undoubtedly present some intriguing possibilities to be explored.

Once you’ve configured one or more of the modulation slots, clicking on any of the small modulation icons (towards the base of almost every parameter) pops open a panel letting you link any of the

Configuration of a target parameter to follow a modulation source is all done graphically.

modulation sources to that parameter and to set the modulation strength and direction (positive or negative). It’s here that the colour-coding comes in handy and, once a parameter has an active modulation link, the modulation icon changes to display the active modulation colours. Yes, it’s rather a compact icon, but it’s a helpful touch.

Kid In An Ear Candy Shop Clicking on the preset name in the bottom strip switches the lower half of the display to the preset browser, which is nicely done. There’s tag-based searching options on the left, and presets identified

Filterverse ships with some impressive presets to demonstrate its capabilities.

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by any tag selection shown in the rest of the window. You can create your own presets, identify favourite presets and create additional tags, so it’s easy to keep your growing preset collection organised.

The supplied presets are well worth exploring. Indeed, on first use with a couple of basic sound sources (a dull synth pad and a basic drum loop), I was smiling within a couple of minutes before losing myself in ear candy land for a couple of hours! In short, all that sound-design potential suggested by the technical features that I outlined above very quickly becomes a rather wonderful reality: put simply, Filterverse is awesome!

This is the kind of creative sound transformation that can make almost any sound the star of your mix. As you might expect, it can work wonders on almost any pad-style synth sound, whether that’s simply adding more interesting tonal movement, or transforming it into a cool rhythmic element. It’s also brilliant on electronic or acoustic drum parts, either for transforming the sounds themselves or, thanks to the various ‘non-filter’ reverb or delay effects, adding all sorts of rhythmic interest. Two levels of dry/ wet (global or on a per-filter basis) let you easily dial the processing from subtle to ‘notice me!’.

I’ve provided a few audio examples on the SOS website (https://sosm.ag/polyverse-filterverse) to illustrate some of the possibilities but, if you work in electronic music, pop, modern metal or modern film/TV soundtracks, all of which embrace the kinds of processed sound elements that Filterverse can so easily deliver, you could get value from the plug-in even if all you did was tweak the presets.

However, if you are happy to go DIY, there’s an almost endless set of creative opportunities to explore and, thankfully, the design and GUI make this both fun to do and relatively straightforward.

WE MAKE MIDI WORK FOR YOU

This Be The Filterverse There’s so much to explore here that it’s difficult to shine a light on every corner of the Filterverse universe in a review, but I ought to at least mention some of the other intriguing elements that lurk beneath the hood. For example, you’ll find Mid-Sides processing options. Both the Meta Knob and Random Generator modulators offer musical scale-based quantise too — if you want your filter frequency to ‘scale snap’, it’s easily achieved. Presets can be changed using MIDI CC 119 or a MIDI note, and the latter is particularly fun to explore because you can customise the MIDI note associated with any preset, effectively creating a set of keyswitches, and then from a suitable MIDI track you can trigger (or sequence) preset changes in real time. Preset changing seems instant, and while you do need to be mindful of the level changes that different processing chains might generate (and tweak the in/out settings accordingly), this is a fabulous way to ‘play’ with Filerverse’s creative potential, whether in a studio or live context.

It’s also worth noting that you can adjust Filterverse’s ‘CPU/Quality’. There are low, mid, and high settings — it defaults to mid, which certainly sounded great on my host system. Cubase reported only 0.2ms latency due to a Filterverse instance, so whatever processing is going on inside the plug-in, Polyverse appear to have made it very efficient.

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Filter Tips Although Polyverse have done a fabulous job in terms of the user experience, there’s a lot to take in when you get started with Filterverse. Fortunately, both the PDF manual I mentioned above and Polyverse’s website are well populated with useful ideas, background and tips to help you along the learning curve. While I’d still consider myself to be on the lower slopes of that curve, I’ve thoroughly enjoyed my time with Filterverse, and if I were to offer one tip of my own it would simply be to download and explore the time-limited demo. There may be plenty of alternative contenders in the creative filter plug-in marketplace, but if you want a single plug-in that offers a whole universe of filter and modulation options, Polyverse’s Filterverse is very hard to beat. If you like to get creative with the sounds in your projects, you really should check it out.

What a great sound-design tool! $ $149.

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NuPrime DAC-9X D-A Converter & Headphone Amplifier Hitherto better known in hi-fi circles, NuPrime make converters of a quality that could appeal to pro audio users.

HUGH ROBJOHNS

A

merican company NuPrime are probably better known in the hi-fi market than in pro-audio circles, but their lineage dates back to 2005 in the form of a company called NuForce that was founded by Jason Lim, a man whose previous career path involved designing microprocessors and Internet technologies for the likes of Motorola, Sun Microsystems, Oracle and others. Lim served as the CEO of NuForce for eight years, during which time the company became a leading light in Class-D amplifier technology, amongst other things. In 2014 he founded NuPrime, which acquired the high-end audio division of NuForce and started creating high‑quality

NuPrime DAC-9X €1425 pros • Mastering-quality converter with very good technical performance.

• Powerful headphone amplifier.

• Supports wide variety of digital audio formats, including DSD and MQA.

cons • No direct-access buttons for H1/H2 sources on remote.

• Lacks common digital connections.

summary This is a very high-quality stereo D-A converter that can also serve as a digital preamp and a powerful headphone amplifier.

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products for the hi-fi, home-theatre and gaming markets. In their second year, NuPrime won four Product of the Year awards, and they’ve gathered many more accolades since.

The company currently make seven different D-A converter (DAC) products, but the NuPrime DAC-9X has been designed specifically for “studio professionals”, which is why this one has made its way to us at SOS for review. In essence, it’s a reference-grade DAC with a variety of digital and analogue inputs and outputs, plus a standard quarter-inch stereo headphone output. So it could conceivably serve in the studio or on location as a straight stereo D-A converter, a source selector and preamp, and as a headphone amplifier.

Technology Unusually, NuPrime’s DAC-9X is built with a dual-mono signal path, based around a pair of ESS Sabre ES9028Q2M digital-to-analogue converter chips. The 9028Q2M is a very high-performance stereo D-A converter that uses the manufacturer’s highly-regarded 32-bit Hyperstream technology, which is potentially capable of delivering a dynamic range of up to 129dB. It supports PCM sample rates up to 384kHz, as well as the DSD format up to DSD512 (22.6MHz).

Although designed as a stereo D-A converter chip, it can be configured as a mono converter, with the two

channels working together in a balanced configuration. This arrangement provides a further 3dB improvement in the noise performance, and is how and presumably why it is being used here — hence there being two of them. The ESS converter includes a variety of onboard DSP functions for different reconstruction filter responses, de-emphasis filtering, and volume control, but none are available to the end user; all are preset in the DAC-9X by the manufacturer.

Separately, additional DSP functionality (courtesy of an XMOS processor) recognises and decodes MQA-formatted audio files (for more on MQA see our SOS August 2016 article: https://sosm. ag/mqa-format), as well as supporting DSD-encoded audio via either the ASIO 2.1 (over USB) or DoP (DSD Over PCM) formats.

Although there is a digital volume control included within the D-A converter chips, NuPrime have chosen to use a separate, digitally controlled, 99-step analogue switched resistor network for the DAC-9X, to avoid noise and distortion artefacts.

Internal construction is to a high standard, with the toroidal transformer and power regulators towards the front of the unit, and all the converter, analogue output and control circuitry on a large circuit board at the rear of the unit, constructed using SMD components.

Connections All connections except for the front-panel headphone socket are at the rear, starting with an IEC mains inlet with integral power on/off switch. The voltage markings on the rear panel might be mistaken as suggesting that a universal switch-mode power supply is employed here, but that’s actually not the case. The DAC-9X contains a traditional linear power supply, which means the mains voltage (100-130 or 200-240 Volts AC) must be selected manually. Due to the

limited rear-panel space NuPrime have placed the voltage selector on the bottom of the case and put the mains fuse inside, in an in-line fuse holder on the mains wiring to the transformer. I’m told that all units are shipped with the switch set in the 230V position (which prevents inadvertent damage whatever territory you’re in) and that new units have a sticker to indicate the factory-set voltage. Ideally, though, I’d prefer a more permanent explicit warning on the rear panel or lid near the IEC inlet (stickers are easily removed, as on this review unit), and for this power-supply voltage arrangement to be given greater prominence in the user manual (currently, it isn’t mentioned until page 17 of 22).

Analogue outputs are provided on pairs of balanced XLRs and unbalanced RCA phono sockets, with maximum output levels of +20.1dBu at the XLRs and 6dB lower (4V RMS) for the unbalanced output — although it should be noted that this is an unusually ‘hot’ unbalanced signal. There are two further outputs. First, there’s an optical S/ PDIF output which appears to provide a loop-through of the selected digital input, with limited format support (only PCM rates up to 192kHz and DSD64; the user manual provides no other information). Second, there’s a 12V Trigger output for switching associated equipment on/off.

On the input side, a pair of RCA phono sockets accept an unbalanced analogue stereo input (maximum 1V RMS, +2dBu, 0dBV). Two S/PDIF digital inputs are catered for, in coaxial form via another RCA phono socket, and in optical form via the usual TOSlink port. A USB 2.0 B-type (square) socket accepts audio data from a computer (or NAS) in the usual way (a Windows Driver is available on the company’s website).

There’s also a USB A-type socket on the rear panel, but this one is non-standard, and intended for the exclusive use of NuPrime’s optional Omnia WR-1, BTR-HD, Stream Mini, or Stream Mini DAC Bluetooth and Wi-Fi streamer hardware. Another entirely proprietary connection format employs a pair of HDMI sockets that accept I2S and

DSD data streams in a bespoke format used by NuPrime’s own CD transports and streamers — these are not compatible with standard HDMI video sources.

As might be expected, these different digital connections have different capabilities in terms of the maximum digital resolution they support. For example, the USB2.0 interface supports PCM data up to 384kHz and 24‑bit from all computer platforms. Mac and Linux machines can also convey DSD64 or DSD128 via the DoP format, while Windows machines go further up to native DSD256, using the ASIO format.

The two I2S inputs have the greatest capability, supporting 32-bit audio data with sample rates up to 384kHz, as well as DSD formats up to DSD256. The coaxial S/PDIF input supports PCM up to 384kHz/24-bit

colour to indicate different MQA modes, with magenta indicating the MQA ‘original sample rate’ (ORFS) format, blue for the MQA Studio and green for MQA Authentic formats. Yellow is shown for DSD. A toggle switch on the front allows the output level to be fixed (at maximum volume) or variable, set using the right-hand rotary encoder, with the current volume indicated on the LED display (between 00 and 99, each step being 0.5dB). Pressing the volume knob mutes the output, and when unmuted the level ramps up over a few seconds to avoid surprises.

Plugging headphones into the front‑panel socket automatically mutes the rear-panel outputs, but if the volume control is switched to the ‘fixed level’ mode the headphones remain muted, which is a sensible safety-conscious feature. I was slightly surprised that there is no provision here for ‘balanced’ headphones, which is rapidly becoming a ‘standard’ feature these days, even in professional circles. The maximum headphone power is obviously dependent on the headphone impedance, but ranges between 1700mW for 32Ω and 135mW for 600Ω. I tried a selection of headphones, including the Sony MDR-7509 (24Ω, 107dB/ mW), AKG K702 (62Ω, 94dB/mW) and Sennheiser HD600 (300Ω, 98dB/mW), and all were able to deliver far more volume than I’d ever need.

A small handheld IR remote control is included with dedicated buttons at the top to switch between on/standby, and to change the display mode. A ‘ring dial’ controls the volume, cycles through the inputs, and mutes the outputs. Alternatively, a quincunx of buttons allows the first five sources to be selected directly, but as the two I2S inputs (H1, H2) don’t have dedicated buttons of their own, the analogue input (A5) button has to be pressed repeatedly to cycle around the A5, H1 and H2 sources.

“If the volume control is switched to the ‘fixed level’ mode the headphones remain muted, which is a sensible safety-conscious feature.” and DSD DoP 64 and 128 formats, while the optical S/PDIF input (and optical output) can only manage PCM up to 192kHz/24‑bit and DoP 64. And finally, the proprietary USB‑A socket for streaming extensions only accepts PCM data up to 192kHz/24‑bit.

Controls The DAC-9X’s front panel is pleasingly simple, with two rotary encoders on the outer edges. A quarter-inch headphone socket, a miniature toggle switch, and a pin-prick LED display (plus remote-control IR receiver) occupy the centre of the panel.

Both rotary controls have press functions.

On the left knob this activates the unit or puts it into a standby mode when held for three seconds, while rotating the knob selects input sources, which are displayed as a two-character code on the LED matrix (C1 for coax S/PDIF, for example).

The display can be switched to show the current sample rate (44.1 up to 384 kHz for PCM formats and a choice of 64, 128 or 256 DSD formats). A miniature LED hidden in the Q of the MQA logo changes

Test Bench Results I performed my usual range of technical measurements on the DAC-9X using an Audio Precision test set. Most

The DAC-9X features a range of analogue and digital input options, including up to 24-bit 384kHz PCM support over USB, but lacks direct support for some common professional formats such as AES3.

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Inside the box, the conversion is based around a pair of ESS Sabre ES9028Q2M chips.

measurements agreed with the published specifications, although the best THD+N figure I achieved was only 0.00075% instead of the claimed 0.0005%. Balancing that bitter disappointment [for comically daft legal reasons, SOS would like to stress that those words constitute a joke! — Ed.], the published signal-to-noise ratio is 110dB, whereas my measured was nearly twice as good at 115dB (or 119dB A-weighted).

Similarly, the specs give a ±0.3dB tolerance for the frequency response, but I found it flat within -0.1dB from 4Hz to 22kHz, dipping to just below -6dB at the Nyquist limit (at 48kHz sample rate).

Regular readers will know that my favoured DAC ‘overall quality check’ indicator is the AES17 dynamic range measurement — to achieve the best scores, every facet of a product’s design, including the power supply, needs to be exemplary

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— and that delivered a result of 120.5dB A-weighted. This performance ranks the DAC-9X in ninth position in the league table of converters that I have personally benchtested, equalling the original Lynx Hilo’s score, and sitting between the RME AIO-Pro PCIe card interface and the Cranborne Audio 500 R8 rack interface. Although it doesn’t quite attain what the converter chip manufacturers claim is possible, this is still an impressive performance that is comfortably of high-end professional mastering quality.

Prime Time?

Given the number of proprietary digital input connections used and that there’s no explicit support for AES3 at all (the coaxial S/PDIF input will accept AES3 data, of course), the ‘designed for studio professionals’ claim is perhaps a bit of

ALTERNATIVES In terms of technical performance and cost, the RME ADI-2 DAC FS is probably the closest alternative. The Benchmark DAC3 HGC and Lynx Hilo 2 are both technically better and offer similar facilities, but they are far more expensive.

a stretch, but I have no qualms about the quality of conversion on offer: the DAC-9X is an impressive product that achieves a very high technical standard and delivers useful facilities. It is nicely constructed, looks elegant, is easy to use, works well, and boasts a powerful headphone amplifier and balanced analogue outputs operating at (European) professional levels. $ $1425 W https://nuprimeaudio.com

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ON TE ST

Music Thing Modular Workshop System

Modular Music Thing’s Workshop System is Synthesizer a modular toolkit that’s designed to inspire.

WILLIAM STOKES

I

f, like me, you’re generally more interested in just using the thing than you are in building it, you’ll be pleased to know that Tom Whitwell of Music Thing Modular opted to send me a constructed Workshop System to review, though currently it’s solely available as a DIY kit.

Music Thing Modular are of course one of the most established names in DIY modular; one of the prime movers in the rise of the great Thonk over a decade ago and the creators of the much lauded Turing Machine random looping sequencer.

Its developer’s most expansive project to date, the Workshop System was originally designed, as you may remember Tom explaining in January’s Modular Profile, for participants in the Cornwall-based Dyski Sound Maps Residency in April of last year.

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A standalone (but case-mountable) unit in 42HP, it hosts 10 circuits (give or take), comes in its own flightcase (that’s right), is USB-powered (I know!) and is analogue everywhere that matters. I’ll say it now: if DIY is your thing, the Workshop System is bound to be a rewarding and enjoyable build. If not, see it as a necessary hurdle you will not regret overcoming.

Case Study From left to right, the Workshop System presents the mysteriously named Computer (which I’ll return to), then a pair of analogue oscillators offering sine or square wave shapes. It’s worth pointing out now that the three fundamental circuits of the Workshop System (that is, the oscillators, the filters and the function generators) come in near-identical pairs, laterally splitting the panel in half. This is

a very nice piece of design, since it means that they can be treated as two discrete synth voices or chained any which way.

If you were so inclined, you could just patch in a CV keyboard and more or less play the whole thing as a Minimoog-style dual‑oscillator synth.

The triangle-core oscillators themselves leave little to be desired: as Music Thing Modular themselves volunteer, these aren’t flawlessly precise but will track fine over three or four octaves once calibrated.

I like the binary choice of square or sine, which covers multitudes of low‑frequency or audio-rate functions, and in one of the System’s few patchless connections (which are of course editable, it being DIY) the sine wave outputs of both are normalled to each other, making some quite complex FM sounds quickly achievable.

Next to the oscillators is a raft of diminutive circuits: these consist of a simple stereo line input, a ring modulator, an external ‘Stompbox’ effects loop, an ‘Amplifier’ section offering another external signal input or even a contact microphone, and finally a four-button interface called ‘4 Voltages’.

All of these are generously imbued with creative functionality: the effects loop, for instance, has a Feedback knob for sending the wet signal back into the external unit, instantly potentiating some interesting

ways of using whatever hardware is patched into it (I’ve rather been enjoying self-oscillating reverbs lately). The Amplifier isn’t just a line input: you can either drive an input signal into tasty transistor-based distortion, or with its switch set to Mic, physically tap, stroke and scrape the Workshop System to create sonic source material. Go gently on that last one — or not, it’s your prerogative.

The 4 Voltages section is an elegant little quasi-random voltage generator, ostensibly spitting out static random values from its four outputs when one of its four buttons are pressed, though you can return to a previous set of values by pressing the previous button, which is key. Its primary role, in this sense, is simply to shake things up, and I’m a fan.

Next along is a pair of ‘Humpback’ analogue filters, offering low-pass and switchable band-pass or high-pass responses. These are ’70s-inspired — designed by Philip Goulding of Berlin-based developer God’s Box, no less — and very characterful: at high resonances they don’t whistle smoothly like a ladder filter, rather they drive right into enjoyably unruly distortion, not a far cry from the beloved likes of the Korg MS-20. As with the oscillators there’s some minimal normalling here: unpatched, the top filter’s output is connected to the bottom filter’s input, which is handy.

To the left of the filters is a brace of function generators; loosely based, we’re told, on the Serge VCS circuit. These are subtly different from one another in that the top slope is linear in response, while the bottom one is exponential.

Each has a switch for cycling the function (so you could, for instance, use one as

Music Thing Modular Workshop System £375 pros • Packed with useful and fun circuits.

• Ships in its own foam-lined hard case.

• Characterful oscillators and filters.

• Brilliantly designed multifunctional Computer module.

• An opportunity to test your DIY skills.

cons • A mult or VCA would be a welcome addition.

summary A triumph of a compact system, the Workshop System is powerful and highly flexible— and is excellent value for money.

an envelope and one as an LFO, à la Make Noise Maths), but again, there’s a nice performative touch here, with the switches’ lower ‘Blip’ position representing a momentary manual trigger.

Furthest to the right is a fairly simple four-channel mixer, channels 1 and 2 of which offer panning. There are also not one but two headphone outputs, the second of which to me immediately screams either aux send Or feedback patching — but that might just say more about how many friends I have (presumably, in that original workshop context, the second output would enable a supervisor to plug in their own headphones to listen to works in progress).

This mixer isn’t the most intuitively laid out, but in this real estate to have a mixer at all is wonderful, especially one with a nice big master volume knob. For that matter, all the knobs here are very nicely chosen and make for a perfectly scaled workflow.

OK Computer To the Computer, then: this in many ways is the crowning glory of the Workshop System. A multifunctional circuit, its jacks and knobs are generically labelled to fulfil a purpose according to which microSD card-sized program card is inserted in the slot at the bottom. It ships with a cute little collection of cards on a keyring: with these it can be a Turing Machine (yay!), a USB MIDI interface, a reverb unit or any number of things. I say ‘any number’ because it’s possible — encouraged, even — to program your own, and indeed many third-party coded functions are already appearing on GitHub. At the time of writing these include a Mutable Instruments-inspired wavetable oscillator, a quad LFO, USB audio output and more. It’s really very exciting; almost like a screenless, slightly downscaled version of the Expert Sleepers Disting.

As for those pre-programmed cards, they demonstrate aptly what the Computer is capable of. For the uninitiated: the Turing Machine really is an excellent design. In short, it randomises stepped quantised voltages according to a scale (which, alongside other parameters, is editable via a web app). This can then be adjusted in all manner of ways, with phrases of variable length able to be locked, looped and clocked in interesting ways thanks to the excellently named ‘Diviply’ control.

The Turing Machine’s most distinguishing function, though, is that its randomness can be adjusted across a continuum, in

this case with the main knob at the top of the Computer panel. At 12 o’clock things are completely random; at one extreme it locks into a repeating pattern; at the other, another repeating pattern but of double the length. Moving between these, things start to ‘slip’ in various ways, with patterns being more or less predictable but with occasional randomness creeping in.

I say all this because it’s the ideal control source for a system like this: it’s intuitive and impulsive, and if you’re in need of melodic patterns but don’t want to lug an external sequencer around or rely on MIDI from your laptop it’s ideal.

MIA If there’s one thing I found myself wishing for on more than one occasion, it’s a mult.

Granted, the panel here is packed, and perhaps utilities (a VCA also comes to mind, though technically the ring mod circuit can double as such) would simply spoil the fun. But in this way the Workshop System is simply a victim of its own excellence — I just wanted to keep patching! And there are compact systems out there that ship with a passive mult, for instance the 2HP Picnic Basket. But let’s be honest: with things like a bespoke foam-lined hard case in the picture, I’m hardly going to start complaining about everything the Workshop System doesn’t ship with, especially when you consider the price.

Conclusion Hopefully by now you’ve gathered that the Workshop System is not only a capable portable synthesizer system, but also that it can be the very centre of a setup — portable or otherwise.

Feed a dynamic mic into the Amplifier, your laptop into the stereo input, a pedalboard into the Stompbox circuit, the list goes on. In my experience, sometimes these sorts of instruments, while attractive, can quickly lead to patching ruts with their reliance on simple waveforms for source audio and filters for audio processors — even if they can accommodate external signals.

Here, though, we have control over signals, effects, ring modulation, a piezo microphone — not to mention the legions made possible by the Computer. It’s as much a playground for processing sounds as it is for generating them from scratch.

Fabulous in almost every way. $ £375 W www.musicthing.co.uk www.soundonsound.com / April 2025

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Speaker-cab innovators Barefaced add a dedicated power amp option to their full-range speaker designs.

DAV E LO C K WO O D

R

egular readers may recall my enthusiasm for the innovative Barefaced Reformer 1x12 guitar speaker cabinet back in 2018. Barefaced’s AVD (Augmented Vent Diffractor) design acts as both a tuned port for greater low-frequency extension and a diffuser for an amount of rear-firing sound. The result is an unfeasibly small and lightweight speaker cab that just sounds so much bigger than you’d imagine it possibly could. Barefaced have a lot of other cab designs, too, and started out making bass cabinets that were significantly lighter in weight than other bass cabs without this adversely affecting their performance, but it was apparent from the outset that their AVD cabs might also make a good platform for a full-range system. Barefaced soon launched their Reality series, marrying Celestion’s two-way, coaxial F12-X200 ‘Full Range Live Response’ 12-inch driver with an enhanced version of their AVD design, designated AVD8. The enhancement was mainly in the addition of an Eminence APT80 rear-facing high-frequency driver that coherently combines with the AVD’s rearward output to create a figure-8 polar response for the cabinet as a whole, the audible effect of which is a far more consistent tone across a wide range of listening positions. My long-term experience with the F12-X200 is that its low-end performance seems to be quite dependent on its enclosure, but the AVD design certainly seems to bring out the best in it.

Powering Up Whereas almost all full-range PA-style cabs likely to be used with digital amp modellers these days are self-powered, using a modeller with a passive Reality cab still leaves you needing to find a suitable power amp. The Barefaced team had apparently been working on a design for a high-end bass amplifier since 2017, but seeing a gap in the guitar market, they directed their efforts towards the project that became the Activier range: Activier 1 with a single input, gain control, and external Speakon for an extension speaker; and Activier 2, with two identical channels. The amplifiers can be fully integrated into even

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Barefaced Activier

Guitar Amplifier

the smallest AVD cabs, with each Activier channel able to deliver 200 Watts RMS into a 4Ω load, 100W into 8Ω or 50W into 16Ω, whilst adding less than one and a half pounds to the already remarkably light weight of a small AVD cab.

The Activier employs the latest ICEpower Class-D power-amp module, in combination with a custom-designed FET preamp offering 30dB gain with very low noise and distortion. It’s also passively cooled, so there’s no fan noise to worry

about. Even though it will most often be seeing a balanced line-level source from a modeller or effects unit, the input is unbalanced; a balanced line input on XLR or TRS would have been nice, given that most modellers will have a balanced output and the modeller might need to be located some distance from the speaker in a stage setup. Input impedance is 1MΩ, so you could even connect a guitar straight in if you only wanted a very clean sound. Testing with a variety of digital amp and speaker

Stereo Imaging The review model, an Activier 2, makes for a highly compact, easy‑to‑set‑up stereo rig with its second channel hooked up to a passive Reality cab. Personally, I tend not to risk the variables of sending a stereo feed to FOH — it’s often not helpful in trying to create a PA balance that works for the whole room — but stereo effects that are audible to you in your playing position on stage are always nice. Alternatively, you could use the dual channels for an on-stage wet/dry split.

modellers, sending at line level, I found I could easily get the Activier/Reality 112FR combination up to stage performance levels whilst maintaining the feeling of there being both plenty of headroom and more power in reserve.

Reality Check I have previously checked out the passive version of Barefaced’s Reality cab, and found it to be a significant step up from using a self-powered PA-style speaker as an on-stage monitor. The large-cone driver of the F12-X200 keeps it closer to the feel of a conventional guitar speaker, whilst the coaxial tweeter integrates much better, to my ears, than any system in which there is physical separation between the tweeter and the LF driver. The Reality’s rear-firing tweeter is the icing on the cake, giving the AVD cabs a lovely enveloping rear output that you can tune to the environment. It’s a bit like the effect of using an open-back combo cab with a reflective surface behind, but more, and better!

In a Reality cab with an integral Activier amp, the resistive tweeter level control

employed in the passive version is replaced by two mini switches, one offering 3dB attenuation, the other 6dB, and 9dB when combined. I generally preferred a setting of -6dB for loud electric guitar, but I could imagine going to -9dB in somewhere really bright-sounding.

Where I certainly would use the full rear tweeter output, however, is when using an AVD cab with an amplified acoustic instrument — an acoustic guitar with a natural-sounding pickup system sounds simply superb through one of these. It is like playing an actual acoustic instrument, only louder, with the sound seeming to come from everywhere around the instrument rather than a single point.

Better Than A PA Cab? There is a significant number of manufacturers now addressing the ‘dedicated powered speaker for modelling amps’ market, and that simply wouldn’t be the case if just using powered PA speakers

Barefaced Activier From $1535 pros • Great-sounding full-range speaker system for guitar modellers.

• All analogue.

• Very lightweight.

• Excellent dispersion.

• Doesn’t require fan cooling.

cons • Unbalanced input.

summary A powerful, flexible and lightweight FRFR amp/cab combo for digitally modelled guitar rigs. It looks like the real deal — and sounds like it too.

response, and the familiar-feeling dynamic response of a conventional guitar cabinet.

Even with only a nominal 200 Watts maximum rated output, compared to the quoted 1000 Watts or so in some other systems, it’s got ‘thump’ in the way that a plastic PA speaker never has. Barefaced’s ‘conventional guitar speaker’ aesthetic, too, will doubtless help this design appeal to those who just prefer to maintain a more traditional look to their stage setup even when using a modelling solution. The Activier amp module, perhaps thanks to the curated response and very low noise floor of its FET preamp design, is a cut above typical Class-D performance, in my experience, and I rather hope to see it made available as a standalone ‘pedalboard’ amp module as well, before too long.

“The Activier/Reality 112FR combination offers an incomparably compact and lightweight, self-powered, all‑analogue solution for the modelling-amp guitar player...” worked entirely satisfactorily for everyone.

I can’t say that I’ve tried them all, but with a good modeller source, I can confirm that I found I could use the Activier/Reality 112FR combination behind me on a small gig like a normal guitar amp, using it as the only guitar sound source for the room, and I can’t say I have ever really experienced that with a PA-type speaker. As always, you’ll need to dial in your tones at performance volume to make sure you have enough midrange — there’s no getting round the Fletcher-Munson effect!

The Activier/Reality 112FR combination offers an incomparably compact and lightweight, self-powered, all‑analogue solution for the modelling-amp guitar player, with excellent off-axis frequency The Activier Class-D amp module can be fitted to any Barefaced AVD guitar cabinet, and comes in one- or two-channel versions.

ALTERNATIVES There are some good standalone power amps if you want to use a modeller with a conventional guitar cab, such as Seymour Duncan’s Powerstage models, but there are significantly more options now for those wanting the convenience of an FRFR ‘combo solution’, including Fender’s FR range, Atomic’s CLR, and models from Laney, Headrush, Mission Engineering and doubtless one or two more. Personally, I’d take any of them over a plastic PA cab, and would certainly rate the Activier/Reality 112FR combination as a serious rival to the best of them.

$ Activier 1 & Reality 112FR

£1218 (approximately $1535), Activier 2 & Reality 112FR £1398 ($1761).

Prices exclude shipping. See website for other specification options.

W www.barefacedaudio.com

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Gainlab Audio GLA-OC1 Governor This clever dual-channel, dual-slope compressor is different from the rest...

BOB THOMAS

B

udapest-based Gainlab Audio are no strangers to the pages of SOS, and the company’s last release, the Bishop channel strip, greatly impressed Matt Houghton in SOS April 2024 (https://sosm.ag/gainlab-bishop).

That impressive valve-based, mono mic/ line/instrument preamplifier featured a unique Teletronix LA-2A-influenced optical compressor that, interestingly, featured a ‘dual-slope’, and Matt found it to be both intuitive to operate and capable of delivering a great-sounding performance. He wasn’t the only user or reviewer to come to that conclusion, so it’s perhaps not surprising that Gainlab decided to created a standalone version of the Bishop’s compressor. The result is the Governor, a dual-channel optical compressor with

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valve amplification and an enhanced suite of controls, features and facilities.

Grand Tour Gainlab’s Governor sits in a 2U, 19-inch rackmount chassis and is capable of stereo or dual-mono operation. To make that possible, there are two identical control sets, laid out in horizontal strips that are divided, both visually and functionally, into three sections.

The leftmost section of each channel begins with two three-position metal toggle switches, the first of which selects the compression type (Soft, Medium or Hard) and the second its Tube (valve) Boost frequency profile (Full, Off or Air). As with all the Governor’s toggle switches, yellow LEDs light to indicate the active selection. Sitting next to these is a line of three rotary controls. The first two of these set the thresholds for the dual compression slopes, about which more later. These controls are continuously variable and range from High to Low, with lines to aid setting recall but no

Valve Optical Compressor

other values shown. The third rotary control turns an 11-position output gain switch, which ranges from 0-22 dB. Nestling between the four threshold controls is a Link Off switch, its position reflecting the fact that it affects both channels. This toggles the Governor between stereo and dual-mono modes: when linked, the lower channel is controlled by the upper control set, with the exception of the side-chain filter frequency and output level selection, which must be set manually.

The centre section’s black background makes the two channels’ trio of horizontal meters easy to see. Within each trio, the upper two meters (GR1 and GR2) each show the amount of gain reduction (0 to 12 dB) being applied by one of the two slopes, whilst the third displays the output level (-20 to +20 dBu) with a VU-like ballistic and includes a true peak hold indicator. To the right of the meters is a pair of rotary switches that set each channel’s side-chain high-pass filter (off, 40, 70, 180 or 250 Hz) or select an external side-chain signal, which is connected using the dedicated quarter-inch jack inputs on the rear.

The right-hand section carries, for each channel, a compressor in/bypass toggle switch and the two six-position rotary switches that are used to select the attack and release times. The attack times are named rather than numbered, with a programme-dependent Auto setting followed by Fastest, Faster, Fast, Slow and Slowest, but the timings are given in the manual (6, 8, 10, 15 and 40 ms respectively). There’s no auto setting for the release, and this allows an additional Slower position to be inserted, giving you a choice of 80, 100, 200, 450, 600 and 750 ms. A final control is one that’s always good to see on a front panel these days: the power on/off toggle switch.

Each channel has an electronically balanced input and a transformer-balanced output, these being presented on XLR connectors on the rear along with the side-chain input jacks, a ground-lift switch and a fused and switched IEC mains power inlet that can accept 110 or 230 Volts AC. Internally, the Governor displays a very high quality of design, construction and attention to detail, and the compressor circuitry occupies three PCBs that sit behind the Governor’s

front panel. After the toroidal mains transformer in its screening enclosure, the largest components on display are the channels’ twin-triode ECC88 valve pairs and Gainlab’s proprietary GLA-AT3 output transformers. The unit’s stated specifications are just as impressive, with a ±0.5dB frequency response of 20Hz-20kHz, THD at 0.04% (1kHz, 0dBu) and 0.32% (1kHz, +20dBu), maximum input and output levels of +26dBu and +28dBu, respectively, and a signal-to-noise ratio of 105dB (+23dBu, <1%THD).

Dual-slope Compression Each channel on the Governor delivers gain reduction based on two compression slopes, and the nature of these slopes depends on the compressor type you’ve selected using the toggle switches on the left of the front panel. A ‘compression

“There’s significant creative sound-shaping potential in the Governor’s dual-slope architecture, but it always remains intuitive and easy to use.” slope’ defines the amount by which an audio signal will be compressed once it exceeds the threshold level — so you might think of these as slopes as ratio presets. (A ‘compression curve’, on the other hand, is created by the interaction between the threshold setting, the knee, the compression ratio, and the attack and release times.)

It’s a clever system that gives you a similar degree of control to using

Gainlab Audio GLA-OC1 Governor $2990 pros • Dual-slope compression delivers flexible shaping of transients and sustain.

• Multiple‑frequency side‑chain filtering with external input.

• Superb overall performance and sound quality.

cons • None.

summary A stereo/dual-mono valve optical compressor whose intuitive dual-threshold compression delivers a superb overall performance and exemplary sound quality.

two compressors arranged in series, but there’s actually only a single gain reduction element (per channel) acting on the signal. In the side-chain circuitry, Slope 1 precedes Slope 2, which means that Slope 2 acts on the output of Slope 1’s curve, rather than directly on the gain-reduction stage’s feedback signal.

It’s this interaction between the two compression curves that generates the final control signal for the LDR (light-dependent resistor) gain cell in each channel.

In the Governor’s Soft compression type, the compression curve for Slope 1 has a wider, softer knee and a lower compression ratio than that of Slope 2. While this differential is maintained across the type shift from Soft through Medium to Hard, the actual difference between the two slopes shrinks as their knees narrow and harden, with the rise in compression ratios. Slope 1’s softer knee overall means that gain reduction actually begins, albeit gently, below the threshold, and rises gradually to reach its preset ratio after the threshold has been exceeded. In contrast, Slope 2’s harder

The metering shows both the output level and the gain reduction being delivered by each slope.

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GAINL AB AUDIO GL A-OC1 GOVERNOR

There’s a quarter-inch jack side-chain input for each channel, and this can be selected using a control on the front.

knee means that gain reduction begins to occur later, and reaches its preset ratio earlier than in Slope 1.

For example, imagine you start with Slope 1’s threshold set so 6dB is indicated on gain reduction meter one (GR1) and Slope 2’s threshold is set at High, and no gain reduction is displayed on GR2. If you then lower the threshold of Slope 2, so it starts to drop below Slope 1’s threshold, you’ll see the gain reduction level begin to increase in GR2 and decrease in GR1.

Carry on lowering Slope 2’s threshold, and eventually GR1 will show no gain reduction at all. What you’ll hear while you’re doing this is that the transients that survived Slope 1 increasingly come under the control of Slope 2’s curve, until it takes over entirely.

I discovered that I could exploit this inter-threshold interaction by setting Slope 1’s threshold to just above the point that gain reduction starts to appear on GR1, at which point I was able to access Slope 2’s

compression on its own. Another bonus was that, in dual-mono mode, I could chain the Governor’s channels in series to give me not only the option of bringing in a third slope should I want to, but also an additional three lower-ratio/higher-ratio combinations, using Slope 1 from the Soft and Medium compression types from the first channel and Slope 2 (Medium and Hard) from the second channel. What all this boils down to is that the interaction between these two compression curves combines with the attack and release options to provide the user with a level of control over the sound and character of compression that I have never before experienced on a hardware device.

Tube Boost The Tube Boost circuit uses the output from a dedicated secondary winding on the channel’s output transformer to increase the gain in its valve output stage. As its name implies, the Full

option delivers a full-frequency boost, leading to an increase in valve‑ and transformer-related harmonic distortion and a subjective sense of enhanced warmth and overall weight. The Air option is essentially the Full Tube Boost again, but this time filtered so that the boost adds saturation more to the high frequencies. To my ears, Air opens up the top end of the compressed audio, creating a sense of added clarity and presence that helps counteract the sense of high-frequency loss that can often occur during compression.

In Use There’s significant creative sound-shaping potential in the Governor’s dual-slope architecture, then, but it always remains intuitive and easy to use. For me, the simplest way to set up single-slope compression was to start with the combination of Auto attack and Slow

Session Recall

The Governor template running in Session Recall.

Session Recall is a third-party software application that makes it easy to record the carefully crafted front-panel control settings of your analogue gear on graphic representations of the hardware. These are stored in what Session Recall call Sessions, which can be saved, recalled and

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edited within the application. The application itself, which now covers the products of some 300 pro-audio manufacturers, is free to download, and you then pay a small fee for each device panel that you wish to add — but Gainlab products now come with a free Session Recall code!

(450ms) release, then set a threshold and move between compressor types and slopes to find the compression curve that got me into my desired ballpark. After selecting a side-chain filter frequency to finesse the response, as I went along I’d then shape and fine-tune the transients and sustain by adjusting the threshold, attack, release and output gain controls.

Other than reading Matt’s review of the Bishop, I’d previously never encountered a dual-slope/twin‑threshold hardware compressor, so it was an interesting learning curve. To my mind, there’s no set way of approaching it, other than using your ears to gain an understanding of how the two slopes interact with different source dynamics. In general, I followed Matt’s lead: set up the compression on Slope 1, and then gradually bring in Slope 2 to taste. Having said that, if it was a source that I was intending to compress hard — as I might, for example, with drum room mics — I’d often open proceedings with Slope 2 and then bring in Slope 1.

As with all optical compressors, the Governor sounds very forgiving, and once I’d got my head around the dual-slope approach, its intuitive workflow led me to develop great-sounding results very quickly. It offers a relaxed, yet somehow also gently firm control in both the lower-ratio compression types, and this impressed me on a variety of sources, including vocals, guitars and basses (both acoustic and electric), acoustic piano, bowed strings and percussion. Although we’re dealing with multi-millisecond rather than microsecond attack times, which means a Governor wouldn’t be my desert island drum compressor, the Medium slopes were certainly well-suited to bringing cohesion to a drum kit, and its Hard setting could easily become my go-to solution for crushing drum room mics and shaping their sustain envelopes for parallel compression effects, especially with some full-range Tube Boost dialled in to lend the sound some additional weight and warmth.

ALTERNATIVES It’s not directly equivalent to the Governor, but Gyraf Audio’s more expensive Gyratec XXIV stereo compressor (reviewed in SOS October 2019: https://sosm.ag/GyrafG24) offers two fully-featured side-chains, which can act either separately or blended together on the stereo signal.

Slope 2 acts after Slope 1 has done its thing, which means the Governor can deliver results similar to using two compressors in series — and on top of that, you can use the two channels in series to give you even more control over a mono signal!

Alongside their in-line roles, the two Tube Boost options are very effective and, indeed, can become very addictive, when used as parallel distortion effects. Full mode, coupled with some judicious EQ in the parallel path, delivered great-sounding results on acoustic instruments and percussion, and was particularly effective on vocals.

The Air setting can also be used very successfully on all of these sources, delivering an enhanced clarity and presence that doesn’t require EQ on the source.

As anticipated, the Governor excelled as a bus compressor, a role in which both its gentle action and its ability to add warmth and weight whilst controlling dynamics and bringing a mix (or submix) into focus was seriously impressive. The Air boost turned out to be a bit of a star in this environment, bringing definition to a submix’s high frequencies in a way that often felt more natural than conventional EQ. For me personally, getting the best out of the Governor is less about thinking in terms of threshold levels, ratios and

attack and release time constants, and more about getting my hands on the controls and listening. Its intuitive operation encourages experimentation, and really that’s necessary if your to explore the ways in which the twin slopes and their individual thresholds interact with each other, with the source, and with the side-chain filtering, the attack and release controls, the three compression types, and the Tube and Air boosts!

Verdict At this price, there’s plenty of worthy compression competition for the Governor out there, but I’m not aware of any other two-channel valve optical compressor anywhere near its price point that can match it for its ease of use, its versatile compression capabilities and its superb overall performance. It also has an attractive asking price, given the exemplary sound quality and creative capabilities on offer. Highly recommended. $ $2990.

W https://gainlabaudio.com www.soundonsound.com / April 2025

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Safari Pedals Zebra Clipper & Bull Sub Machine pros • Quick and intuitive • Neither too many options nor too few • Bull Sub arguably best in class!

cons • Easy to overdo it.

summary Safari Pedals’ plug-ins are all colourful, fun, easy-to-use and affordable tools, and Zebra Limiter and Bull Sub are no exception. Don’t let the slimmed‑down control sets fool you either — they’re capable of excellent results, and can help you get there quickly.

Safari Pedals Zebra Clipper & Bull Sub Machine

Limiter & Low-end Enhancement Plug-ins Powerful, creative tools lurk beneath these plug-ins’ playful interfaces...

NEIL ROGERS

S

afari Pedals were founded by Tel Avi-based mixing and mastering engineer Noah Levinberg, who wanted to create audio tools that gave musicians and producers the creative feel of using guitar pedals when producing and mixing. In a relatively short time, Safari’s range of plug-ins has grown into a full-blown menagerie, with plug-ins that combine a bold ‘wild animal’ aesthetic with a simple feature set, focused on ease of use and creativity. I’ve had access to their catalogue for a few months now and have enjoyed throwing these colourful tools onto my mixes with abandon.

Dirty Animals This is the first time we’ve looked at Safari’s plug-ins, so I’ll quickly introduce the range

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(currently 17 plug-ins) before reviewing the two newest releases, Zebra Clipper and Bull Sub Machine (from here on Bull Sub). Each is competitively priced, and bundles even more so. Most are similar in their styling and controls, and resemble stompboxes.

Yak Delay and the Ladybug and DirtyDog reverbs offer creative and colourful spatial effects, and of these DirtyDog (created in collaboration with Joe Chiccarelli) really stood out: it combines saturation with reverb and is especially good for spicing up dry-sounding drums. Alongside these obviously creative tools are some slightly more conventional offerings, with, for example, Owl Control providing four modes of compression, and Dragon EQ. A few are styled differently, like the Rhino Reverb, which is based on the Pioneer SR202 spring reverb, and there’s a Roland 501-inspired Fox Echo Chorus. The Cobra Fuzz, looking like a larger vintage-style pedal, is unapologetically unsubtle in its attempts to mangle and distort your audio.

I enjoyed using these three a great deal,

especially the Cobra Fuzz, which offered me a distortion flavour I didn’t already have — it sounded great on drums and could lend a nice ‘edge’ to a vocal.

Some mixing jobs require specific tools, but I also like having plug-ins I can throw onto a track to change a part’s feel, even if I’m not quite sure what I’m looking for, and this is where Safari’s ‘pedals’ excel. One of Noah’s first creations, Gorilla Drive, is a great example of what these plug-ins do well: you have input and output controls to add drive and extra weight to your tracks, along with a single tone control and bypass and polarity flip switches. Time Machine, meanwhile, allows you to quickly audition different filters and compression styles loosely based on sonic eras: the ’50s setting sounds like much of the high frequencies are rolled off and allows a source to feel more ‘set back’ in a mix. And while I’m rarely a fan of developers building in authentic noise and hiss, I often found myself blending in a little of the tasteful ‘crackle’ offered by some of these plug-ins.

Such vintage/analogue-esque flavour is a common theme throughout the range but the simple, well-thought-out controls mean you always feel in control and can take it precisely how far you want to go.

I’ll conclude this whistle-stop tour here, but hopefully you get the idea: there are numerous tools and flavours to choose from and I’d say it’s well worth trying the fully functional 14-day free trials. So let’s now move on to the newest releases...

Zebra Clipper Clippers have become increasingly popular in recent years and rather than turn down the peaks that exceed a threshold, as a limiter would, they

‘shave off’ those peaks. Zebra Clipper is intended to sit somewhere between a conventional clipper and a more soft-clipping style of compressor. Unlike many clippers, Zebra offers no visual cues to indicate how much you’re clipping your audio, and I suspect that’s deliberate: it directs you to make decisions with your ears. Its three ‘style’ controls offer distinctly different tonal options, with Analog sounding softer at the top end and Modern sounding more bright and defined. A Round option sits somewhere in between, and often ended up as my preference. As with the rest of the Safari collection, the additional controls are few but tastefully judged: Mojo ramps up the sense of saturation and Sub Boost helps to fill out the low end (which is often lost when you squash your audio like this). At the time of writing the Zebra was very much my plug-in of the moment, and it works superbly on drums for adding a sense of solidity and vibe. The only down side is that it is very easy to overdo it!

Bull Sub Machine Bull Sub is a collaboration with mix engineer Leslie Brathwaite, who is probably best known for helping shape the sound of bass-heavy productions for the likes of Cardi B, Pharrell and Lil Uzi Vert. As the name suggests, it’s focused exclusively on the low end: it’s intended to beef up your kick drums and bass parts, and to suggest creative ways of blending your parts into your mix or production.

And as always, there are few controls, but they’re really well judged.

There are plenty of plug-ins for adding sub-harmonic content to your tracks, but I find that you often have to use them in conjunction with other plug-ins to get their output to really gel with your tracks — it’s often as easy to ruin a mix as improve it! Bull Sub addresses this challenge superbly, with just the right amount of control for dialling in the frequency range you want to add or exaggerate, along with a Drive control to add some saturation and grit that can help your bass parts cut through a busy mix or be audible on smaller speakers. The Resonance control is an excellent addition that makes it easy to extend or shorten the length of kick drums — a sort of onboard transient designer. Lastly, we have Punch, the details of which are scant but it’s modelled on Leslie’s analogue chain and is intended to offer a little extra ‘oomph’ when that’s needed!

You do need to tread a little carefully with Bull Sub. It’s very addictive and easy to overdo, especially on less accurate monitoring setups. But Bull Sub is hands-down the best tool of this style that I’ve played with. I found using the blend control the key technique for getting it to work in a mix, but it took mere seconds to beef up a limp-sounding kick drum I was working with! $ Zebra Clipper $89, discounted

to $49 when going to press.

Bull Sub Machine $79, discounted to $29. Everything bundle $1080, discounted to $129.

W https://safaripedals.com

There’s much more to be found on this Safari, and the full bundle represents great value for money.

ISO-STAND ISO-155 HEIGHT ADJUSTMENT

TILT ADJUSTMENT UP TO 6.5 DEGREES REDUCES UNWANTED VIBRATIONS

14 TILT/HEIGHT VARIATIONS

OPTIMIZE SPEAKER PLACEMENT & FOCUS REDUCES INTERNAL VIBRATIONS

SCAN HERE FIND YOUR PERFECT

ISOLATION STAND

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USB Audio Interface

Audient iD48 Audient’s iD48 makes it easier than ever to integrate outboard into a computer-based recording setup.

SAM INGLIS

O

ur small island has produced many great mixer manufacturers, and Audient are at the forefront.

Their ASP desks make a powerful case for analogue in an increasingly digital world, and have proved enduringly popular, offering the feature set we expect from a large-format console at a surprisingly affordable price level. Audient have also spun off technology from these mixers

Audient iD48 $999 pros • A versatile audio interface offering eight high-quality mic preamps, each with an insert point, and up to 16 channels of digital I/O. • Insert points can be repurposed as hardware inserts at mixdown, without repatching.

• Unusually good monitor control and master section features.

• Excellent audio specifications and performance.

cons • Signal routing and I/O naming relating to hardware inserts can be confusing.

• The use of generic drivers means low-latency performance is OK but not stellar.

summary The iD48 is a well-specified USB audio interface with a unique killer feature: switchable insert points that can be used to integrate outboard both during tracking and mixdown.

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into rackmounting mic preamps and the iD range of USB audio interfaces, and their latest product is the iD48: a device that, in some ways, brings these two lines together.

The Ins & Outs All of the previous iD models have been desktop devices, but the iD48 is a 1U rackmounting unit. Cosmetically, it inherits the smart dark grey colour scheme of the smaller iDs, but in terms of front-panel layout, it’s more obviously a cousin of the ASP880 eight-channel preamp. Either way, build quality is faultless. In its default state, however, it shows no sign of life, and it’s only when you press a button or run some signal through it that anything lights up to indicate that it’s actually switched on.

The iD48 has eight mic preamps, all accessed through combi XLR/jack sockets on the rear panel. The first pair also have auto-sensing high-impedance inputs on the front panel for guitars.

Unlike on Audient’s more affordable EVO range, gain is set using analogue potentiometers, and phantom power activated (on a per-channel basis) using physical switches. As a result, gain is not a recallable setting, and EVO-style auto-gain is not available. By contrast, the input pad, polarity reversal and 100Hz high-pass filter are toggled digitally, either using front-panel buttons or from the associated iD software. A switchable 10dB boost operates in the digital domain

and is only accessible from software.

Input metering comprises just two LEDs per channel.

On the output front, the iD48 has two pairs of dedicated monitor outs on balanced jacks, along with two front-panel headphone outputs. The monitor outputs are usually under the control of the large front-panel level encoder (with push action) and four associated buttons, which together handle ‘master section’ features such as speaker switching, talkback, mute, dim and mono fold-down. Two of the buttons are assignable in software, reflecting the fact that the iD48’s monitor control functions are appreciably more sophisticated than those found in most audio interfaces. For example, both the dim level and the trim offset for the alternate speakers can be adjusted, you can flip the polarity of one channel to audition the Sides signal in mono, and best of all, the mono fold-down can be routed to a single speaker rather than both. Bravo. There is no built-in talkback mic, but the iD48 can make use of a separately connected USB mic for this purpose, so you don’t need to sacrifice an input channel in order to berate the talent.

As well as the eight combi inputs and four jack outputs, the rear panel

also sports two pairs of Toslink optical sockets, which are independently switchable between ADAT Lightpipe and S/PDIF format, plus word clock in and out on BNC connectors (with 75Ω termination switched in software) and an IEC inlet for mains power. But the iD48’s most interesting feature is the pair of DB25 connectors in the centre of the panel. These are carried over from the ASP880, where they provide switchable insert points on each of the eight preamp channels. They still do this in the context of a USB audio interface, but their powers have grown.

Console Class Like all the products in the iD and ASP ranges, the iD48 uses Audient’s “console preamp design” developed for the ASP mixers. The implementation of this circuit varies, and the iD48 lacks the ASP880’s impedance options and sweepable high-pass filter. In the main, though, the specifications are broadly similar, with an impressive unweighted EIN figure of -127dB, which would correspond to about -130dB A-weighted. The pot has a gain range of 58dB, augmented by that 10dB digital boost at one end and the -10dB pad at the other. The headphone outputs can supply 565mW into a 60Ω load, which should be more than enough for most use cases. On the digital side, the iD48 offers an A-weighted dynamic range of 120dB at the A-D converters and a massive 126.5dB on the D-A side. Both sets of converters are aligned such that 0dBFS corresponds to an analogue level of +18dBu. This is the lowest of the standards in widespread professional use, and some might miss the ability to operate at +20 or +24 dBu reference levels.

Equally, if you’re using its insert points to patch in guitar pedals and the like, you might wish for the ability to trim the alignment levels in the other direction.

Like the other iD interfaces, the iD48 uses generic audio drivers. On macOS, that means Apple’s own Core Audio driver, and at the lowest 32-sample

The Routing panel in the iD control software offers considerable flexibility, but some potential for confusion.

The eight Line Outputs shown here are key to the unit’s insert switching functionality.

buffer setting, this delivers a round-trip latency of just under 8ms at 44.1kHz. That’s adequate for most purposes, but less good than, for example, PreSonus have achieved with their new Quantum USB interfaces.

The iD48 can be configured to act as a standalone preamp, but in its main role as a USB interface, needs to interact with Audient’s iD software. The first time I opened this, I wondered why everything was greyed out, but it turns out that’s just how it looks. It’s generally clear if seen straight on, but I’d welcome a brightness control, and the grey lines disappear into the grey background when viewed at an angle.

Some control panel utilities provide a single mixer pane, with auxiliary sends used to establish alternate balances for cue mixes and so on. Others ditch the aux sends in favour of providing multiple, simpler mixer panes, and this is the route Audient have gone down, allowing you to set up a Main Mix plus up to four Cue Mixes. The available sources are the eight mic inputs, the optical ADAT or S/PDIF inputs, and five stereo DAW return channels.

This is where things can get slightly confusing, because the outputs available to your DAW are named as if hardwired to specific physical outputs — and although they can be routed like that, they usually aren’t. By default, for example, the DAW output named Analogue 1-2 is actually routed to the first stereo DAW return

channel in the iD mixer. This, in turn, feeds into the Main and Cue Mixes, and it’s these mixes that would typically feed physical outputs such as the monitor and headphone outs. In other words, the naming conventions don’t take into account the fact that there’s normally a mixer and routing matrix sitting in between the most-used DAW outputs and the physical outputs.

What source feeds what output is determined in the Routing panel of the iD software, where you’ll see an additional set of eight D-A converters labelled Line Outputs. These sit in the insert path accessed through the DB25 outputs, and can’t be addressed directly from the DAW. Line outputs can be set to carry the Main or any of the Cue Mixes, which would be useful for adding external headphone amps, but the most important options are the settings labelled DAW Analogue and DAW ADAT. These set a given line output to mirror the corresponding DAW output, drawn either from the first or second bank of eight. For instance, setting line output 5 to DAW Analogue will feed it from the output your DAW addresses as Analogue 5, which also feeds DAW return channel 5 in the iD mixer. Setting the same line output to DAW ADAT mode, alternatively, will cause it to mirror DAW output 13, which emerges as the fifth

A crowded back panel features an IEC power socket, Toslink optical I/O ports, BNC word clock connectors, four quarter-inch monitor outputs, the all‑important pair of DB25 connectors, and eight combi XLR/jack inputs.

www.soundonsound.com / April 2025

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The iD48’s internal mixer has separate, very grey panes for the Main Mix and four Cue Mixes.

channel in the first bank of ADAT outputs, and doesn’t appear in the mixer.

À La Mode These options exist in order to support a software-switched ADC Path setting that is available on the eight analogue inputs. With ADC Path in Mic Insert mode, the input signal passes first through the mic preamp, then through whatever hardware you have connected via the DB25 connectors, then to the A-D converter, allowing you to print a processed signal during tracking. In Mic mode, input signals are routed directly to the A-D converter, with the corresponding DB25 output freed up to act as an additional line out, subject to the routing described above. The major innovation of the iD48, however, is ADC Direct mode.

In this configuration, the preamp signal goes nowhere, and the insert loop is fed from the line output, thus permitting your outboard to process a DAW return signal rather than an input signal.

The benefits of this are obvious. Let’s say you have a compressor patched into channel 5 of the DB25 loop. By setting input 5 to Mic Insert mode, you can route a mic coming into that input through the compressor during tracking. Then, by switching it to ADC Direct mode, you can employ that same compressor as a hardware insert at mixdown, without re‑patching. Alternatively, you could connect a bunch of hardware synths to the DB25 return, and toggle Mic and Mic Insert modes to switch between tracking mics and synths, again with no need to unplug anything. And as all eight input channels have insert points, you can connect up to eight outboard devices or chains of devices to use in this way

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(at which point you’d probably want to consider adding a patchbay too!).

As described above, the DB25 line outputs always mirror other DAW outputs when set to Mic and ADC Direct modes.

This introduces a potential source of complication, because you can’t use the same DAW output for two purposes at once. By default, for example, your DAW’s master channel will probably address analogue outputs 1-2. This precludes you from simultaneously setting line outputs 1-2 to DAW Analogue mode and using them for hardware inserts.

In most cases, DAW ADAT mode provides a simple solution to this dilemma. What your DAW sees as digital outputs 11 and upwards don’t have DAW return channels in the mixer and won’t typically be used for monitoring, so it’s not a problem to have the line outputs mirror them when you want to use hardware inserts. However, DAW ADAT mode won’t work in Pro Tools, which insists that hardware inserts use the same-numbered input and output.

Since the inserts are always returned on inputs 1-8, they must send from outputs

Each of the iD48’s eight analogue inputs can be switched between three insert routing configurations.

1-8, and hence Pro Tools users are tied to DAW Analogue mode, with any outputs used as conventional DAW return paths unavailable for hardware insert use. If you want to use eight hardware inserts simultaneously in Pro Tools, your only option is to use outputs 9-10, nominally the first pair of digital outs, as your main monitor outputs, feeding the last DAW return channel in the mixer.

I did wonder if Audient could solve this by allowing the user to change the order in which inputs and outputs appear to the DAW, as is possible with Thunderbolt interfaces such as the UA Apollo and Focusrite Red models. Ultimately, however, it isn’t Audient’s fault, and the best solution would be for Avid to change how hardware inserts work.

The iD48’s routing options can be a bit hard to get your head around, in part because of the terminology, and in part because the otherwise excellent manual needs more detail on this specific topic. At the end of the day, though, the routing is logical, and if there’s a little brain-ache involved in understanding it, the unique possibilities opened up by the iD48’s insert switching more than justify the effort.

Identity Parade The Audient iD48 brings together technologies that Audient have spent a long time perfecting in the iD and ASP ranges: a proven mic preamp design, a slick software control panel utility, uncommonly well thought-out monitor control features and more. To this powerful combination, it adds the latest and greatest A-D and D-A converters and an outstanding innovation in the form of the insert switching. The result is a product that feels mature and classy, whilst doing something very important that nothing else can. I can’t think of any comparable interface that offers insert points on all eight analogue inputs, let alone the ability to switch them into line output paths for use at mixdown. The iD48 is a compelling proposition even if you never use its insert points, thanks to its superb audio specs, comprehensive feature set and highly competitive pricing.

And if you do want to integrate hardware into both your tracking and your mixing approaches without investing in a mixing console, it’s in a field of its own. $ $999.99 W www.audient.com

ON TE ST

Vienna Symphonic Library Sample Synchron Prime Edition Library This all-in-one symphonic package offers an easy way to create orchestral arrangements.

DAV E ST E WA RT

V

SL’s Synchron Prime Edition gathers together all the essential instruments and articulations of the orchestra along with some tasty extras in one easy-to-use 31.4GB package. The samples are distilled from seven large VSL Synchron Series libraries comprising a 46-player string ensemble, chamber strings, brass, woodwinds, ethnic winds, percussion and harp, all performed by hand-picked members of the Synchron Stage Orchestra. The library runs exclusively on the Vienna Synchron Player, included free with all Synchron products.

You can check out the library with a free 30-day demo, and upgrade and crossgrade prices are available to and from the big Synchron collections.

It’s worth noting that while VSL’s iLok licence management system can be used either on the iLok Cloud or with a physical iLok 2 or 3 USB key, it doesn’t work with the old iLok 1 key. Once you’ve activated your iLok licence, the excellent free Vienna Assistant software takes the strain out of downloading and installation.

A major update for Synchron Prime Edition is Flow View, an alternative user interface designed to streamline your workflow by presenting just the essential controls for each instrument.

Also included is a new Universal

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playing mode which tracks your performance and switches articulations in real time (see box for details).

Downsizing In order to squeeze a complete orchestra into such a small package, VSL omitted some performance styles (for example, runs and trills) and restricted microphone signals to two active channels. Every instrument has a stereo room mix of all mics used in the bigger Synchron Series collections, augmented by a ‘mid’ position for the ensembles and close‑miking for the solo instruments. The larger libraries’ dynamic layers are faithfully preserved for all instruments except the percussion, which now has fewer than the 16 velocity layers and eight round-robin repetitions used in Synchron Percussion.

Sound quality has not been sacrificed: the combination of the room mix’s large-hall ambience and the drier mid or close signals covers most requirements, and if you require additional mic positions you can port your arrangements across to the big libraries using their ‘Prime compatible’ presets.

Strings & Harp The Prime Edition’s strings come in two sizes: a large 14/10/8/8/6 ensemble lifted from VSL Synchron Strings Pro, and a smaller 6/5/4/4/3 line-up taken from VSL Synchron Elite Strings. These

collections were reviewed in SOS May 2021 and August 2021 respectively.

If you want a big symphonic sound, the 46-player ensemble is the way to go. Played with just the right amount of vibrato, the 24 first and second violins’ musically perfect legatos sound fabulous on soaring themes, and their Bollywood-esque portamento slides are a delight. The short staccatos are ideal for propelling rhythmic passages, a tight, brisk delivery which sounds great from the violins down to the cellos and basses.

Performed by a selection of Vienna’s finest string players, the chamber-sized Elite Strings’ more transparent sound works well for intimate ballads and sensitive film music. The large and small ensemble perform an identical articulation menu of silky-smooth legatos, long notes with a choice of attack styles, staccato, tremolo, marcato, sforzato and regular and snap pizzicato. Both also include handy full-range ‘tutti’ presets with the instruments mapped over a B0‑E7 span.

VSL’s Synchron Harp (reviewed here in December 2022) is a superb collection performed by the Synchron Stage Orchestra’s harpist Tina Žerdin. The pared-down Prime version features straight plucks and an ethereal set of harmonics — both sound absolutely gorgeous. (User tip:

Vienna Symphonic Library Synchron Prime Edition €595 pros • Contains all the essential instruments and articulations of the orchestra in a single modestly-sized package.

• Beautifully played and perfectly recorded in VSL’s large Synchron hall.

• The intuitive Flow View interface puts the fun back into playing.

cons • Switching between Flow View and the old GUI causes both to return to their default state with no preset loaded.

summary

Synchron Prime Edition’s new Flow GUI is designed to streamline your workflow by presenting just the essential controls for each instrument. It coexists with the old GUI within the Synchron Player with no loss of the latter’s functionality.

adding reverb to the harp’s close mics sounds very nice indeed.)

Woodwinds & World Winds Also featured in the above mentioned review was Synchron Woodwinds. All but one of its 13 solo instruments have been collated into the Prime Edition, the sole absentee being the alto flute.

(A pity. Still, as the old joke goes, ‘You can’t have everything — where would you put it?’) The consolation is there are two examples of flute, oboe, Bb clarinet and bassoon, which together with the superbly playable English horn give you a wide range of melodic woodwind timbres.

All instruments perform precise, rhythmically consistent staccato and portato short notes, long notes, legatos and sforzatos, and all but the piccolo add emphatic accented sforzatissimos.

Most incorporate no-vibrato and vibrato sustains which you can crossfade using the MIDI CC20 controller. The Bb clarinets and superb bass clarinet use no vibrato, but their beautifully smooth, rounded tone is nonetheless highly expressive.

Three of the four included world winds are bamboo flutes: the Indian bansuri has a lovely soft breathy tone, while the higher-pitched Chinese dizi is brighter and more penetrating. More commonly found in sample libraries, the plaintive Japanese shakuhachi is often used in modern film scores, and the good old tin whistle (a common fixture in UK high

street music shop windows) epitomises the lively sound of traditional Irish music.

These instruments’ expressive grace notes and pitch‑bends will add a colourful dimension to your arrangements.

Brass Derived from Synchron Brass (reviewed here in November 2021), the Prime Edition’s brass covers all the orchestral essentials. Its two Bb trumpets are played with and without vibrato, producing magnificent melodic results for both pop and classical styles. Benefitting from

Synchron Prime Edition represents a super new chapter in VSL’s long history of starter editions. Distilled from seven large Synchron collections with no loss in sound quality, the library comprises two string ensembles, brass, woodwinds, ethnic winds, percussion and harp playing all the essential orchestral articulations. The new Flow View interface is intuitive and fun to play, and contains links to VSL’s educational Academy instrumentology.

Recommended for all those taking their first steps in virtual orchestration.

VSL’s trademark true legato mode, the two solo horns and tenor trombone also sound convincing playing lead lines and supporting melodies.

While the bass trombone’s loud staccatos are ideal for oom-pah bass lines, its quiet sustains also work nicely for chord pads; the bass tuba is

Flow’s simplified control panel uses a maximum of five switchable playing styles, one of which is the new Universal playing mode, which tracks your performance and automatically switches articulations in real time.

www.soundonsound.com / April 2025

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Flow View

The library retains the traditional Synchron Player GUI, now referred to as Precision View.

equally versatile, providing sonorous low pedal notes for brass chords and occasionally contributing its own plaintive bass melodies.

For motoring rhythmic ostinatos, all instruments and ensembles perform excellent short staccatos and the solo instruments have an additional portato short-note style. All come equipped with sforzato and sforzatissimo artics, the latter rasping, blasting delivery the brass player’s equivalent of turning up to 11.

If you want a big symphonic brass sound, the four trumpets and six horns ensembles will deliver epic fanfares and soaring regal themes, supported by four trombones in the lower register.

For instant cinematic grandiosity, the 15-player tutti ensemble combines these three sections with the bass tuba, providing a playable C2-D6 setup with built-in low‑end octaves — an impressive and inspirational symphonic brass texture.

Percussion Reviewed in May 2021 alongside Synchron Strings Pro, Synchron Percussion I was VSL’s debut Synchron Stage release. Many of its samples have now been rounded up and herded into the Prime Edition, sounding as powerful as ever and providing a decent spread of tuned and unpitched instruments.

A highlight are the ringing, ultra-clean timpani. Played with medium mallets and spanning a two-octave range, their performances comprise booming single hits, looped sustained rolls and walloping

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octave strikes. A 36-inch orchestral bass drum, a snare drum and three concert toms also play hits and rolls, which you can accompany with piatti cymbals, suspended cymbal and tam tam crashes.

In the tuned department we have a pretty celeste (which needs some additional high-end brightness IMO), a sparkling glockenspiel, an impeccably sampled xylophone and ecclesiastical-sounding tubular bells.

The collection is rounded out by a nice set of wood blocks and an ethereal, ear-catching triangle.

Conclusion Don’t tell the UK Flood Defences Committee, but the Flow is spreading rapidly — VSL have released a Flow version of Synchron Duality Strings called Duality Strings Essentials, which contains the same articulations as the Prime Edition strings with an option to blend between its simultaneously recorded 52-player and 27-player ensembles. Sounds like fun! You can read about the original library in my January 2024 SOS review.

Meanwhile, Synchron Prime Edition serves as an excellent primer for orchestral sample beginners, a perfect tool for sketching or live playing, and a resource-saving compact orchestra you can run on a laptop while travelling (as someone implausibly said) “on the red-eye from NY to LA”, or more likely on the delayed Central Line service from Epping to West Ruislip (which probably takes a bit longer).

While the Synchron Player is jam-packed with useful features, its level of detail can be intimidating. With this in mind, VSL created an additional GUI that aims to make life simpler for its users. Enter Flow View, which coexists with the old GUI within the Synchron Player with no loss of the latter’s functionality.

The new Flow interface opens with a large virtual orchestral seating plan — clicking on a section reveals its list of available instruments, which load very quickly with a single click. The control screen has three large panels: Instrument contains dynamics, humanisation, timbre softening and expression settings; Sound comprises mic balance, reverb effect send, mix preset selection and master volume; while the all-important Articulation panel shows a maximum of five keyswitchable playing styles.

The new Universal artic reacts to your playing velocity, switching automatically between (for example) regular, sfz, sffz and staccato deliveries or soft/normal/ marcato note attacks, depending on the instrument. The strings can also be switched between legato and portamento with the sustain pedal.

Though Flow is spacious, beautifully designed, intuitive and fun to play, it comes with a few wrinkles: if you switch back to the old GUI (now called Precision View), any edits you made in Flow will be lost and both GUIs dump their loaded presets. Not ideal. Your Flow edits can’t be saved as user presets; instead, you have to save the song and your DAW will remember the edit when you reload the project.

Thirdly, although setups created using the old GUI can be replicated using the big Synchron libraries, there is no simple way to port Flow-based arrangements over to the bigger collections. VSL say the compatibility presets are really only for the Precision View presets, and advise that if you intend to end up using the bigger collections, it makes sense to start your arrangement with the old GUI.

A brilliant, comprehensive and intuitively playable collection of pristine orchestral instruments available at less than 20 percent of the total price of its source libraries, Synchron Prime Edition is the ideal starter package for anyone wishing to enter the world of virtual orchestration.

Suggested further reading: The Sampled Orchestra www.soundonsound.com/ series/sampled-orchestra $ €595 W www.vsl.co.at

Bridging in-room and personal headphone monitoring Our UNIOTM Ecosystem gives the ability to optimise, align and seamlessly navigate your monitoring options, for the flexibility to work anywhere, anytime – from stereo to immersive.

Now, with the introduction of Personal Reference Monitoring – our pioneering new active monitoring headphone solution – UNIO offers the very highest level of headphone accuracy. genelec.com/unio

Explore UNIO

ON TE ST

Qu-Bit Stardust Eurorack Module

T

here’s a magic to tape. Not just the ‘ooh, that sounds lovely’ magic of tape saturation, but the compositional magic of Frippertronics or sound on sound looping. The magic of interacting with a machine and never being entirely sure what’s going to happen next.

There are worlds of experimentation to explore with tape, but as anyone who has mucked about with the magnetic brown stuff will know, it’s not without its drawbacks; price, maintenance, capacitors going pop, thousands of feet of the stuff unexpectedly spooling onto the floor...

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So imagine all of that potential for sound exploration contained within one convenient Eurorack module, all controllable by CV. Well, many people have, perhaps most notably Make Noise, but the latest device to take up the challenge of ‘tape without tape’ is the Stardust, which Qu-Bit describe as a ‘cosmic tape looper’. The cosmic part is open to interpretation, but the tape looper part is fairly straightforward, and straightforward is what the Stardust is all about.

That’s not to say it can’t produce complex results — complex and surprising results — but that Qu-Bit have done their best to make things simpler.

Not just simpler than working with tape — that could hardly be more complicated — but simpler than other tape-inspired modules. To this end the front‑panel functions and controls are clearly labelled and easy to understand. The only slightly cryptic element is the cosmos graphic that surrounds the central Varispeed encoder, the changing colours of which denote different states — useful if you can manage to remember what the colours mean. Everything else is fairly self explanatory. There are stereo inputs with level control, and stereo outputs with a wet/ dry control; buttons for record, play, reset and reverse; loop start and size knobs... All of which do what you would expect.

Slightly less expected, the Slice control not only slices the loop into increasing divisions as you turn it, but also randomly skips between slices (or splices in Stardust parlance); and Skip doesn’t skip between slices as you might expect but introduces ‘splice transformations’. This is probably where you’ll reach for the manual for the first time, but not understanding exactly what’s going on is no barrier to experimentation, and that’s what we’re here for, right?

There are built-in effects

including tape-style hiss and flutter, digital degradation, high- and low-pass filtering and a reverb that’s respectable enough that you could pair the Stardust with a single sound source and get an ambient album out of it.

Central to the Stardust, both physically and philosophically, is the Varispeed control. Like tape, changing the speed changes the pitch and, also like tape, if you, say, record with the Varispeed at half speed, what you’ve recorded will play back at twice the pitch when you return to normal speed (which you can do by pressing the encoder). This, and the fact that you’re only ever working with one loop (no Reels here) is the Stardust’s main qualification as a ‘tape looper’. Next to the Varispeed control is the Inertia knob, which controls how quickly Varispeed changes take effect, creating greater tape-style slowdown and stop effects as you turn it clockwise.

One area in which the Stardust comprehensively beats tape is in CV control. Everything bar Loop Mode and Effect Type has a dedicated CV input and there’s a clock input with which you can control Record, Reset, Slice, Skip and Freeze in a rhythmic fashion. Revox never thought of that, did they?

Lastly, but by no means leastly, the Stardust has four Loop Modes, the default being Sound On Sound, which, to quote the Orb, is for layering different sounds on top of each other.

There’s also Replace, which replaces existing audio when you hit record, and Resample, which resamples the current loop along with any modifications you’ve made to it. Things can get delightfully messy here, but you can return to the original loop with the Undo button. Finally, Frippertronics mode mimics the method first popularised by Terry Riley as Time Lag Accumulation and later rediscovered and rechristened by Brian Eno and Robert Fripp. In the original version the tape runs between

two machines, the first of which records and the second of which plays back, creating a delay, the length of which is determined by the distance between the machines. The playback signal is routed back to the first machine to be recorded again, making a loop, and the level of the returned signal dictates how quickly the loop fades. Get this right and the loop will slowly fade out over a number of repetitions, degrading charmingly as it does so. Get it wrong and the loops will get louder with each repetition and eventually (or sometimes really quite quickly) feed back, not that that’s always a bad thing.

With Qu-Bit’s version you set the loop length with your initial recording, and from then on the Stardust stays in record mode, gradually fading older loops as new ones are added. What you can’t do is control the amount of time it takes for loops to fade, but nor can you accidentally provoke it into ever‑increasing feedback, so swings and roundabouts. You can, however, stop recording to maintain the loop’s current level, which you certainly can’t do with tape.

As a Frippertronics system makes recordings of recordings, with tape the fidelity will gradually decrease with each loop, often deteriorating into a bed of pleasingly fuzzy noise before disappearing completely. The Stardust’s approach is more clinical, but what you lose in character you certainly make up for in practicality.

That last sentence pretty much sums things up: the Stardust doesn’t sound like tape, but it will take you into the same experimental territory, whilst being much more practical. More practical because you’re not wrestling with a temperamental piece of 50-year-old recording equipment that may have ideas of its own, but also because you can do things that whilst technically possible with tape would take so long you’d never consider doing them. It’s also true that the impracticalities inherent with tape are enough to put most people off trying to use it in the first place, which makes the Stardust infinitely more practical. So yes, you sacrifice character with a module like the Stardust, but you also gain a lot of things you might never get to try otherwise, all of them controllable by CV. The Stardust doesn’t have mesmerically turning reels or the welcoming warmth of tape saturation, but it does deliver some of the same magic as a tool for composition or sound design. You can take a simple piece of audio and surprise yourself with what it turns into, and that’s pretty magic.

David Glasper $

$499

W

www.qubitelectronix.com

Winter Plankton DassBrum Eurorack Module

I

am often fascinated by the amount of control and possibility that gets loaded onto a module designed to make one particular sound. In a regular synthesizer or drum machine the kick drum is pretty well defined, and if you’re lucky, you might get a decay or tone control. In Eurorack, you seem to get every control you can think of, which makes me wonder whether anyone is happy with their kick drum. Do we need to be constantly tweaking it into something unique, or at least something else?

Winter Plankton seem to be well aware of what you could turn a kick or bass drum into. So much so that they’ve amusingly changed the name around as if to prove that this isn’t just any old bass drum; no no, this is the DassBrum.

The DassBrum starts with an analogue sine wave pumped out of the VCO and playable over five octaves. Stick a sequence into the Pitch input and, provided you trigger the internal analogue envelope, you have a tune. It has a single Release control, giving anything up to 12 seconds. There’s no Attack, and the leading edge has a satisfying thump to it, which is probably exactly what you’re looking for in a percussion module.

To further its sonic versatility DassBrum has four wavefolding options: Hard clipping, Sergestyle waveshaping, Custom and Hard + Serge. These amount to pushing the sine into a square‑ish shape, two flavours of gentle folding and then harder clipping. These can be set post or pre the VCA, which offers an almost discernible difference.

The combination makes for a really perky synth voice. The simplicity of the Release and Fold sliders makes it very playable, and adding some modulation to the Fold brings in a wonderfully wobbly bit of PWM style.

There are a couple of small annoyances to mention. Firstly, the Fold modes are represented by the colour of the button, but the difference between ‘soft orange’ for pre VCA and ‘orange’ for post VCA is so slight that you’d never know by looking at it and are constantly having to cycle through all the modes to remind yourself which you are on.

Secondly, the Fold CV input can control either the Fold amount or the Fold mode but, for some reason, they’ve decided to put the switch for this on the back in the tiniest form I’ve ever seen — or rather could only see once I’d donned my DIY magnifying glasses. Switching between the two modes could be a useful variation tactic so having it on the front would be fabulous, but hey.

Right, interesting synth voice aside, we’re here for that kick drum. And for that, you need to look no further than the two knobs in the middle.

These handle the Decay and Amount of the digitally controlled pitch envelope. I’m certain there’s some sort of scientific equation of Decay-to-Amount ratio that forms the perfect 808-style analogue kick, but you can find plenty of candidates just by fiddling with those two knobs. If you unpatch the Pitch, Fold, Accent and other modulations you can get some really solid sounds.

This is enhanced further by the Sweet Spot button, which restricts the range of the Decay and Amount knobs by removing the extremes and keeping the full sweep of the knob in the most kick‑drum-friendly area. It’s a neat idea.

When you have everything happening at once — folding, accents, sequences and modulation — it does a fantastic job of being a funky and fascinating synth and bass-drum party. Having DassBrum as purely your kick drum would be to underutilise its versatility massively.

Robin Vincent $

$199.99

W

www.winter-modular.com

W

www.planktonelectronics.com

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for that because you don’t overthink it.

Digital is of course super convenient and can be much more precise. So I think using the best of both worlds is a great option that we have now.

Modular Profile: Julien Guillot

S

ynthesist and producer Julien Guillot started his project Stazma the Junglechrist back in 2008, in his words at “the super violent and punk end” of the breakcore scene, before breaking out into more experimental territory. Alongside his prolific output as an artist and a busy schedule as a mixing and mastering engineer at his own Electric Voyage Studio, he also teaches sound synthesis at Lyon’s venerable Les Escales Buissonnières. You can follow Julien’s adventures on YouTube at www.youtube.com/@Stazma.

On his entry into modular I was always intrigued by these giant beasts filled with cables that I would see in album artwork, or inside CD or vinyl booklets. I got into Eurorack in early 2016 with only an Intellijel Rubicon, which I still have, that I sequenced and filtered with my Elektron Analog Four. I started adding mostly Make Noise modules to this, as I was really drawn to their dadaist interface— I blame Richard Devine’s videos for that! They also looked and sounded very different to the classic Moog-style synthesis I was used to.

After I filled my first case I got more into a ‘one case, one manufacturer’ thing and built quite a few dedicated instruments like that. I like to dive into the design philosophy of one person and commit to it as an instrument, maybe because I was a bass guitar player at first. It has since grown way out of proportion, especially since I do the YouTube thing.

On his go-to modules That’s a hard one! I have so many now [laughs]. I’m still deeply in love with playing and sequencing with the Make Noise René — the old one — and Pressure Points together. That combo really feels like a perfect instrument interface to me. I also still consider [Make Noise] Maths as one of the best envelope generators ever. Being a sucker for the vintage Roland sound, I’m often going to my Intellijel Dr Octature II and System80 810 [3-in-1 Subtractive Voice Module] and 860 [filter] for acid bleeps and rubbery bass lines.

Even before I started working with Schlappi Engineering, I was super into

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the Angle Grinder [oscillator or filter and distortion]. I could have a full rack of just this thing! The Serge VCFQ [filter] is also one of my favourite things on this planet, not to mention other Serge designs like the DUSG [slope generator] and TKB [touch keyboard and sequencer].

And of course I’ve been enjoying anything Buchla-inspired, such as the Verbos range— top mentions to the Harmonic Oscillator and Multi-Delay Processor — and the recent Tiptop/ Buchla 200t series — and I’ll give a special mention to the 296t. I’ll also add a shoutout to the Shakmat crew for their Bard Quartet, which I think is the best quantiser ever made!

On Electric Voyage Studio I’ve been doing mastering and a bit of mixing for more than 15 years now, even though the official ‘studio’ is quite a recent thing. Sound engineering is the only thing I went to school for, all the rest I’m just an obsessive experimenter and manual-reader! I love mastering as a job: to me it’s exactly the same kind of super-precise, finicky process as patching a modular synth. I love taking time, listening to my client’s music and the process of making it sound like a cohesive thing and not just a collection of tracks that have been put together. I use both digital and analogue; I feel it’s important to really listen and ‘feel’ the way you alter the sound. Analogue is always better

On teaching modular synthesis at Les Escales Buissonnières This is something of a dream job that I got thanks to my friend Stéphane Bernard, aka Uzul, who was already teaching Ableton Live and Dub technics at Les Escales Buissonnières before he got me in. I already had a big modular collection, so I could have a system-per-student, and I loved the idea of teaching and sharing my knowledge with others. I can answer the same questions over and over without getting mad, which is key! [laughs] I’ve met some amazing people during these teaching sessions, and I myself also learn new stuff every time.

I usually write patches on a white board, with the audio path and control path colour-coded to quickly show how it is ‘supposed’ to sound, and I let everyone try it on headphones, then debug their patches and answer questions. We go further and further; we end up sync’ing systems together, then jamming and recording things. It’s a lot of fun — it feels more like a paid holiday to me than a job!

On the culture of modular Coming from one of the most niche musical scenes there is, breakcore, I find a lot of similarity in the kind of punk, DIY, ‘we do our stuff and we don’t care’ attitude of the Eurorack scene. There are so many weird ideas being put into circuits, and the fact that there are still new ideas to be had is baffling. I also feel that the people making modules are super open, nerdy — in a good way — and fun to talk to. Superbooth last year was my first and it was great meeting so many of them at the same time.

I also love the retro side of it, the fact you can find all the pieces of your favourite cult synths and mix and match them together, even though I still prefer single-manufacturer systems for myself.

I’m not super happy with this trend that every module has to be as small as possible though: I like big panels and big knobs. I feel super small modules are often super hard to use, especially the digital ones with tons of functions.

I prefer big things that can be ‘patch programmed’ — in the spirit of Serge.

William Stokes

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ON TE ST

PYE 4061S PAUL WHITE

T

ed Fletcher has been involved in a number of audio companies over the years. He designed the original Alice consoles, for example, and was behind both the original JoeMeek and TFPro brands, whose gear earned plaudits. Today, Ted’s again designing for Alice, who recently acquired the PYE brand name, and he was invited to design and build a high-end stereo compressor that would be worthy of the old name. One of his aims was to keep the operation as simple as possible, in line with the ethos of hardware used in the 1960s, but to improve on the noise performance of those vintage devices. The result of his endeavours is the PYE 4061S,

PYE 4061S $2000 pros • Simple operation and intuitive metering.

• Transparent sound.

• Handles anything from fast limiting to slower levelling.

• Stereo width control.

cons • No external side-chain input.

summary A classy sounding, high‑quality stereo compressor, the 4061S can act very quickly and is a good option where you need transparent control, whether you’re limiting or levelling. The stereo width control is a pleasant bonus.

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Analogue Compressor Ted Fletcher breathes new life into this venerable old name.

and while it may have little in common with the original PYE compressor’s circuitry, it is nonetheless impressive.

Design & Operation Described as a 21st Century analogue levelling amplifier, the PYE 4061S is a stereo feed-forward compressor that’s designed to sound as transparent as possible, both when controlling peak levels and when used for longer-term levelling.

It’s billed as being suitable in the recording chain for tracking, for track processing during mixing, and for mastering. The circuit employ’s Ted’s previously proven ‘sum and difference’ mode to eliminate stereo image shift when processing stereo signals.

The ‘feed-forward’ description relates to the way the gain‑control circuitry is configured. Most compressors work by sensing the output level from the compression stage and then feeding it back to the side-chain and gain-control stage, which might be typically a FET, VCA or opto-electronic circuit. By contrast, a feed-forward compressor monitors the input signal’s envelope and then sends this information on to the gain-control element, which in this instance uses a transconductance amplifier in the feedback path of a high‑performance BA4580 op-amp. Ted says this feed-forward analogue limiter/ compressor design adjusts its compression ratio in a way similar to the old Altec compressor, but without introducing low-frequency bumps and without the overshoots of the slower valve circuitry.

By the way, there’s no external side-chain access, as the topography of this circuit doesn’t have what you’d recognise as a conventional side-chain.

The 4061S is housed in a 2U rackmount case, with a reassuringly thick, white-painted front panel. On the front, a pair of illuminated moving-coil meters monitors the levels and gain reduction.

Levels of up to +28dBu can be handled by the output stage, which employs circuitry built around UK-built Stevens & Billington transformers. Two illuminated buttons bypass the compressor and switch the metering from showing either the left and right levels, or showing the mono level on the leftmost meter and gain reduction on the right.

The controls are set out clearly, with an input gain knob allowing for up to 20dB of gain, using balancing transformers configured in Ted’s favoured current mode, which improves the low‑frequency performance. It’s by adjusting this input gain control that you set the amount of compression — there’s no separate compression or threshold knob — and the output level is then adjusted to taste (or to suit the next device in the chain) using a gain make‑up control. As with typical soft-knee compressors, the higher the input level the greater the compression ratio, and when pushed hard the compressor ratio gets very close to pure limiting.

In addition, there are controls for attack (an eight-way switch that includes an Off option) and release. When attack is set to Off, the compressor no longer applies gain

reduction but the signal does still pass through the transformer input and output stages, and both the input and output gain controls still function. The Stereo Width control is made possible because of Ted’s aforementioned sum-and-difference approach to compression, and is essentially a Mid-Sides balance knob, enabling a stereo source to be ‘dialled back’ to mono or widened up to 150 percent of its initial spread. The left‑right channel balance can also be adjusted using a Balance control.

All the pots have a reassuringly smooth action and everything is clearly labelled.

The transient response can be set very fast — the attack time can be set from 0.2ms to 25ms — while the release is variable from 100ms to four seconds. At the fastest attack, which is under a quarter of a millisecond, the brief overloads that can sneak through a compressor circuit should be too short to be audible. The wide range of release times makes this device well suited to a range of different stereo compression and limiting duties.

On the rear panel, the balanced inputs and outputs are presented both on XLRs and quarter-inch TRS jacks, making connection to other gear convenient. Also on the rear, you’ll find the IEC mains AC power inlet, and most common international voltages can be accommodated. The total power draw is under 6W.

only game in town, and once you’d learned to work with them you came to appreciate how intuitive they actually are: for most sources, they relate pretty well to the dynamics of human hearing perception.

I get the impression that Ted designed the PYE 4061S to be used more as a polishing tool rather than adopting the ‘angle grinder’ philosophy of things like the Empirical Labs Distressor, and that’s consistent with his previous designs (which I’ve enjoyed). Obvious ‘character’ compression is not the main focus, and there may well be other compressors that are more appropriate if your aim is simply to add some attitude to a track or

has to keep in mind when treating very low‑frequency sounds.

A common challenge in mixing that every engineer faces is that of a vocalist with an excessive dynamic range or imperfect mic technique. The usual solutions involve using both level automation and compression, and in this role, the PYE 4061S does an excellent job of mitigating excessive peaks without making vocal sounds appear obviously compressed. At longer release times, it acts well as a leveller, smoothing out sounds that fluctuate in level over a longer time period and, because the compression ratio adjusts automatically with the input level, high-level peaks get pushed down harder while more well-behaved sections are treated more lightly. Again, that’s helpful in maintaining a sense of transparency. Those longer release settings can also work well when processing complete mixes or submixes, as they can help ‘glue’ the parts together. But as every job is a little different, it all comes down to using your ears when deciding how much compression to add and what attack and release times work best.

With the attack control set to its Off position, the signal can be treated to a trip through the circuity and transformer stages without any compression being applied.

While the overall aesthetic is clean and transparent, as with any well-designed transformer there’s an audible effect that’s subtle but still somehow flattering, adding just a hint of warmth and depth.

“The 4061S seems to be ‘kinder’ to the high frequencies, such as the cymbals in a drum track, than some other compressors I’ve tried.”

Easy As PYE? Those more used to plug-ins with fast faux-LED metering will need to get used to the more restrained ballistics of moving‑coil meters, which is especially noticeable when processing things like drums with a fast release time — in that scenario, the gain reduction meter may barely have time to blip before the sound has passed, and in such situations you must rely on your ears rather than ‘going by the numbers’, the more so because of how fast this compressor can operate. Of course, the engineers of yesteryear all worked that way, because moving‑coil meters were the

mix. Having said that, though, if you push it hard it will start to slam in a very audible way when processing something like a drum track.

Still, to me it seems best-suited to dealing with level-balancing situations without making its presence too obvious — even when it is being used quite assertively. Also, to my ears, the 4061S seems to be ‘kinder’ to the high frequencies, such as the cymbals in a drum track, than some other compressors I’ve tried, with the result sounding somehow pleasing and natural. It works very well for tightening up bass tracks, although as the attack and release times can be set very fast, there’s some skill involved in setting it up — it’s important not to set the release time to be too short if used in combination with a fast attack, as the compressor can then respond so quickly that it reacts to the individual cycles of a low‑frequency input, rather than to the overall envelope, and that manifests itself as distortion. This isn’t a fault — just a possibility with any very fast compressor, and something the user

Conclusion In summary then, the PYE 4061S is a high-quality compressor that keeps operation simple, and has proven itself very effective for tackling a wide range of limiting, levelling and general compression jobs, just as it was conceived to be. It’s one I’d recommend reaching for more where you want control while still preserving the original character of the source sound, rather than using a compressor to beat and bend sounds into a different shape. But it’s capable of very fast attack and release times, so it can still get very assertive if you want that. The Mid-Sides widening is a very useful tool to have to hand when compressing stereo sources too.

$ £1650 (about $2000) excluding taxes.

T Alice +44 (0)1442 863494 E info@alice.co.uk The XLR inputs and outputs are duplicated on TRS jacks, so you should have no trouble finding a suitable cable! W www.alice.co.uk www.soundonsound.com / April 2025

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Polyend Synth Synthesizer It’s a synth called Synth!

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RORY DOW

I

n recent years, Polyend have become known for their sequencing and tracker products, namely Play and Tracker. In my recent review of the Tracker+, Polyend had added some new digital synthesizers to their tracker system. At the same time, they were busy developing several more algorithms and, in a move not rivalled since my mate Brian named his dog ‘Dog’, have packed them into ‘Synth’.

Synths, plural, might have been a better choice because this is a three-part multitimbral device, where each part uses one of eight different synth

engines. Instead of a traditional keyboard, the Synth uses a 5x12 grid of silicone pads, which can be configured to play all three synth parts together in rather clever ways.

The idea is that the three different sounds become one playable, interactive performance grid.

Throw in effects, arpeggiation and sequencing into the mix, and you have a surprisingly configurable synthesis playground.

Synth Er Size The Synth is a desktop unit measuring 282 x 207 x 35mm and largely follows the design philosophy of Polyend’s other desktop products like the Play and Tracker+. A small but informative colour LCD screen is used for editing along with touch-sensitive encoders to change values and buttons to navigate. There are three dedicated encoders for master volume, tempo and pitch-bend, three macro encoders, and nine soft encoders that follow whatever is shown on screen. Some dual-use functions are accessed with a Shift key, but it is intuitive overall with some unavoidable menu-diving.

The Synth is three-part multitimbral and each part can be assigned one of eight synthesizers. These parts can then be mixed, panned and sent to three effects: chorus, delay and reverb. Let’s take a look at each of the synth engines.

The Synth is 100 percent digital, but the synthesis engines cover virtual analogue, wavetable, FM, phase distortion, granular, sample playback and physical modelling.

On the analogue side, there is ACD, which is inspired by 101s and 303s based around sawtooth and square oscillators, a sub-oscillator, noise, pulse-width modulation and a choice of low-pass filters. FAT is more Minimoog-inspired, with three oscillators, noise and ladder filters.

VAP is a Prophet-style polysynth with dual oscillators, soft-sync and multi-mode filters.

On the more digital side, we have WAVS, a wavetable synth with dual wavetables, morphing, warping and multi-mode filters. WTFM is a 2-op FM synthesizer with multiple feedback paths and waveshapes. PMD is a physical modelling engine capable of strings, bells, plucks, drums, breathy sounds and more. PHZ is a modern phase-distortion engine with some unique waveshaping options. And finally, GRAIN is a granular sample-playback engine capable of tempo-sync rhythmic textures and the usual cloud-based atmospheres. All in all, there is a lot of synth to choose from and plenty of excellent presets to get you started.

The synthesis engines generally share some common features. Envelopes and LFOs are often the same, although some engines offer more than others. The filter sections are often the same, too. Some engines share the same low-pass designs, and some share the same multi-mode designs. Each part also has an optional 64-step sequencer or arpeggiator.

Most (although not all) of the engines also have a modulation matrix with six slots and varying source and destination lists, depending on which synth you have loaded. Furthermore, each preset can assign three macros with dedicated front‑panel encoders to change up to six parameters simultaneously. All the factory presets have macros assigned and sensibly named. Needless to say, there are plenty of opportunities to automate sounds.

The Scenic Route Your choice of three synth parts is packaged together in a Scene. Scenes are global presets that save the entire state of the Synth. In addition to the synth sounds, sequencer, mixer and effects settings, they also save the grid setup and voice allocation settings.

Voice allocation allows you to spread eight available voices across all three synth parts. Eight voices isn’t a huge number for a multitimbral device like this, and I would have preferred 16 or even 32.

Polyend Synth $549 pros • Three synths in one.

• Virtual analogue, FM, phase modulation, physical modelling and granular synthesis all under one roof.

• The chord follow system is a great way to generate song ideas.

cons • Three synths but only one stereo audio output.

• Only eight voices to be shared.

summary The Polyend Synth is a fun performance-orientated device combining multiple synthesis methods with a unique pad grid that allows for some inspiring ways to play three synth sounds at once.

But in general, eight is manageable. You might have one for a bass sound, five for a polysynth, and two for a special effect. It does preclude you from layering up three ambient pad sounds with long releases, but that’s a limitation we must live with.

One exception is that the GRAIN (granular) engine is paraphonic, allowing you to play theoretically unlimited ‘voices’.

My favourite aspect of the Synth, however, is the grid. This innocuous grid of pads hides some neat tricks. Firstly, you choose one of a dozen grid layouts. These change the ‘split’ of the grid so that each of the three synth parts occupies a different area. The backlit pads light up blue, yellow and purple to show you which area each sound is assigned to.

In addition, you can set a scale and root note by which all areas abide. The grid will show this root note as a different LED brightness. Individual areas can also be assigned a root octave number to ensure they play the correct range.

It gets really interesting when you assign one area to Chord mode. You can assign a single chord that will be transposed (in or out of scale) or a ‘chord pack’ — a curated set of up to a dozen chords. The Synth ships with around 50 chord packs, but you can also create your own using the onboard chord pack editor.

With one part playing chords, either of the remaining parts can be assigned to Follow mode. Whenever you play a chord, these areas will reconfigure themselves so that you can play notes only from that chord. If a part has a sequence or arpeggiator assigned, it will also follow the current chord. This keeps performances www.soundonsound.com / April 2025

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On the rear, we have a single stereo output and MIDI in and out (Type B) on 3.5mm jacks, a microSD card slot, an emergency reset switch, USB-C for power and MIDI, and an on/off button.

dynamic and often inspires new and interesting melodies. The pads are velocity and poly-pressure sensitive too, which opens up the performance possibilities.

Another nice aspect of the grid is that it will output MIDI on three separate channels on either USB or mini-jack so that you can play external synthesizers and benefit from the sequencer, arpeggiator and smart-chord features.

In Use The Synth comes with plenty of Scene and Engine presets, but if you want to create your own Scenes (and you will) the general workflow is like this. Firstly, choose a layout for the grid. I suspect many people will settle on a favourite that they reuse often.

Sometimes, you’ll want to play a single sound across the entire grid, which also involves choosing a grid layout. It would be nice if there were a quick shortcut to put the grid into Blue, Yellow or Purple mode instead of having to menu‑dive, as it’s something I found myself wanting to do often during sound design sessions, then reverting to one of the split modes afterwards.

Then, it’s a case of populating the grid with three sounds and choosing the playback mode for each grid section.

Choosing sounds might be selecting a preset or choosing an engine, initialising the patch, and then using the navigation buttons and 3x3 grid of encoders to create a sound.

The experience of sound design is OK but not amazing. That’s not the fault of the Synth, but rather that there is little muscle memory that can develop when using it. Because there are eight different synth engines, and the knobs are generic controllers rather than having fixed functions, you have to constantly look at the screen whilst sound designing. I’m not sure Polyend could do anything about this, but it’s worth bearing in mind if you’re more of a ‘one knob per function’ person.

Once you’ve got sounds for each part, assign modulation in the modulation

SD Card Contents The Synth comes with a 16GB microSD card. It stores various data the Synth uses, including settings, synth patches, Scene presets and chord packs. It also comes with a library of WAV files used for the granular engine, and you can easily add your own, although they must be 16-bit mono (any sample rate). Lastly, there is a folder for wavetables. These are also in WAV format, and you can add your own using almost any window size, as long as it’s written in the filename (eg. “Harmonic Sweep-2048.wav”) so importing from your favourite wavetable synthesizer should be a breeze.

matrix, assign macros, then mix and send to the three effects busses.

The effects are nice. The chorus is rich, the reverb has a nice ‘bloom’ character that suits synths well, and the delay is more than functional. It’s a shame there’s only a single stereo output on the rear of the unit, as it would be nice to externally process each part separately.

It is a nice side-effect of the multitimbral setup that effects are part of the Scene and not part of the individual synth preset because you can browse for new sounds without changing the effect send settings. Say you want a sound that

on all the engines. I cannot say why; perhaps it’s the digital‑to‑analogue converters or some CPU optimisations, but it’s not something you find on all digital synths. My ancient (and cheap) Yamaha TX81z is both digital and can alias terribly, but it doesn’t sound harsh in those higher frequencies. Another problem that I had too often was internal clipping. Even with the master output reduced, I often had to reduce the gain at the synth level to eliminate some internal clipping.

These are minor complaints, however, and the overriding experience the Synth delivers is fun. It encourages you to be playful in a completely different way from a traditional piano keyboard. I would love to see future updates to the grid performance feature because, as good as the synthesis engines are, the grid is often the source of the most fun. Even if you bought a Synth purely to use as a MIDI controller for other synthesizers, it would still be a great product. Creating a custom chord pack of all the chords I was working on for a particular song and then using this as the basis for a new Scene was incredibly rewarding.

Already, the Polyend Synth is a hit. I see a lot of positive feedback online, and it’s well deserved. Polyend continue to make affordable products that offer something unique, and the Synth is no exception.

It’s fun, musical, and can generate more sounds than it has any right to do at this price. Highly recommended.

“Polyend continue to make affordable products that offer something unique, and the Synth is no exception.”

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delays or has some reverb to push it back in the mix, that will remain constant while you browse for a new preset.

Once your sound design is done, it’s time to play. It is particularly fun to have one part playing a chord pack whilst other parts ‘follow’. Throw in a little arpeggiation or sequence, and new song ideas tumble forth. You can even set parts to hold so a sequence plays constantly and follows your chord changes.

Conclusion In terms of sound and scope, the Synth is impressive. The eight different synth engines, whilst not class-leading, are feature-rich and distinct enough to allow the Synth to create a larger scope of sounds than your average synth. In that way, the Synth is like a Waldorf Iridium, with many synthesis types.

The sound itself is good. I did detect a certain harshness to the high frequencies

$ $549 W www.polyend.com

OMX-E

Expanded reach for varied mic technique

OMX-M OMX-T

Medium-tight control with mid-high presence

Tight control for close vocals

MAKE THE OMX-E SOUND CHOICE OMX-T Expanded reach for varied mic technique

Tight control for close vocals

MAKE THE SOUND CHOICE

OMX-M Medium-tight control with mid-high presence

ON TE ST

HOFA-Plugins Colour Delay Delay Plug-in Not sure which type of delay might best suit your song? German plug-in developers HOFA might be able to help you solve that dilemma, as their Colour Delay not only offers a choice of Tape, Bucket Brigade, Digital and Destruction delay types, but also allows these styles to be combined before further treatment by some onboard effects. It supports all the common plug-in formats for Mac and Windows machines, and can be authorised to run on two computers.

The delay types are depicted as blocks, with one or two controls relating to that specific delay type. For example, the Bucket Brigade has adjustable noise, while Tape offers drive and wow/flutter.

Bucket Brigade also offers a choice of four types, ranging from vintage to modern.

Digital has adjustable digital word length (ie. ‘bit depth’) and sample rate, which allows it to emulate early 12- or 8-bit delays — but with a minimum resolution of three bits, it has the range to take you into seriously lo-fi territory (the maximum is 16 bits). Destruction emulates tube distortion and also has an adjustable clipper.

The delay time, feedback, modulation and delay polarity (Inv) controls apply globally, and there’s a small selection of ping‑pong options for when using the stereo version of the plug-in. The colour-coded delay style blocks have on/ off buttons and blocks may be dragged into any order — this can deliver noticeably different results when using two or more blocks. The effects again offer only one or two controls each, along with tempo-sync options where available and, as with the delay type blocks, they can be dragged into any order. Variable diffusion adds an almost reverb-like atmosphere to the delays, then there’s a phaser, chorus, auto-panner and ducker to explore. Wet level and stereo width controls are displayed as horizontal sliders at the top of the interface, along with an overall output level control. The delay

PSP Audioware EasyVerb 2 Reverb Plug-in EasyVerb 2 has a very different look from PSP’s original EasyVerb, along with an enhanced feature set.

It supports AU, VST, VST3 and AAX formats on macOS and Windows, and uses iLok protection (hardware iLok

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time and auto-pan rates can be set freely or tempo-sync’ed, with a maximum delay time of five seconds. A Tap button also allows for entering the delay rate by repeated clicking.

An upper display section shows the position of the delays over time, using vertical bars that get smaller as the repeats decay, and an audio waveform is visible in blue behind them. Here, the dry/wet levels, delay time and delay feedback can be adjusted by click-dragging. The display’s lower section hosts a spectrum analyser and an EQ curve, with up to four bands of fully parametric EQ. Points on the curve can be dragged to adjust the frequency and cut/boost, at which point a parameter box pops providing access to all three parametric values. A number of shelving, cut/boost and band-pass filter shapes are available. Note that the diffusion, phaser, chorus, auto‑pan and ducker affect the wet signal after the feedback loop, providing overall colour and shaping.

There’s lots going on here, then, but how is HOFA’s Colour Delay to use in practice? Well, for starters, it’s extremely simple and intuitive, and capable of convincing emulations of the various delay types. But it also gives you the ability to push things much further, since any combination of delay types affect the signal being fed back to create the repeats. This allows you to make things very colourful when needed — hence the name! While it is simple to control, the effects work very well for adding interest to the delay sound, so it takes very little effort to move from convincing emulations of the main delay types, via various style of analogue and digital lo-fi, to some useful ambient washiness. It’s well worth grabbing the free time-limited demo to try it for yourself.

Paul White

or machine-based). Once you have a PSP account set up, you can also avail yourself of the free PSP Chamber plug-in, which has a red rather than blue interface — this runs only the chamber algorithm, but is otherwise identical to EasyVerb2. Owners of the original PSP EasyVerb can upgrade to this new version for a modest $19.

Along with the stylish new interface, EasyVerb delivers 10 algorithms that cover ambience, rooms and springs up to clubs, chambers, plates, arenas, cathedrals and halls, as well as a useful reverse reverb offering. Each algorithm can be switched to small, medium or large size, with a further choice of three variations: Legacy, Alt 1 and Alt 2, the latter changing

$ €85 (about $89).

W https://hofa-plugins.de

the tuning of the algorithms to offer three different tonal characters. In effect then each algorithm now has nine options.

The user is given direct control over key parameters for each algorithm: decay time, damping, and pre-delay.

The last of those is set using a large horizontal slider, while up/down buttons below the display access the size, algorithm and variation options.

If you want to recreate the exact same sound as the original EasyVerb, you can set a medium size running the Legacy variation. There’s also a new three-band semi-parametric EQ, with upper and lower shelving filters and a tunable mid band, which is welcome — it seems you’d set this flat should you want to recreate the original EasyVerb sound. At the bottom of the window are controls for width, mix and output level, as well as a bypass button that kills the feed to the reverb engine.

The plug-in’s name could lead you to believe that you are getting some sort of ‘stripped down’, or ‘entry-level’ reverb

but that’s far from the case! Each of the algorithms captures the essence of the space that it emulates, and having two new variations on each as well as a choice of three sizes means there’s a lot of scope

to shape the space. The reverb tails are smooth, and they sit well with the dry sound rather than appearing to be ‘grafted on’. The smaller reverb sizes are particularly telling, as that’s where the illusion of reality falls down with some less sophisticated designs, but here, the small rooms and ambiences really do invoke the sense of an intimate space.

If you were to press me on the subject of reverb quality, I’d say that EasyVerb 2 sits between the classic digital reverbs of yesteryear and the more sophisticated algorithmic reverbs available today — most of which are considerably more expensive that this plug-in. More important, however, is that the reverbs on offer are inherently musical-sounding, and can be dialled in easily to suit vocals, drums and other instruments.

If a reverb as easy as this can sound so good, why would you choose a difficult one? Paul White $ $99.

W www.pspaudioware.com

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Denise Audio Perfect Room 2 Reverb Plug-in Denise Audio say that their Perfect Room 2 takes a new approach to algorithmic reverb, with the aim of delivering a natural, transparent character. It’s essentially an update of their original Perfect Room plug-in. They’ve improved the clarity, while also offering zero‑latency performance, and this new version has some useful creative options, such as changing the shape of the reverb tail and reversing it. That can produce some very musical treatments that I already know will come in useful in my own ambient compositions! Also useful is that key parameters can be tempo-sync’ed, specifically the pre-delay time, attack time and reverb length, and the last of those is particularly helpful when using reversed reverb tails.

This isn’t one of those conventional reverbs with separate early and late reflections controls, though. Its simplified control set belies its ability to integrate with the real sound in a very believable way, conjuring up the illusion of a real space, whether that be a small room, a cathedral or somewhere in between.

An EQ, with five fully parametric EQ nodes, sits over a spectrum analyser display. The large Reverb Time knob has a sync button and a reverse button below it. To the left we have controls for pre-delay, attack time, pitch mode and Halo, the last extending the decay time at either the high- or low-frequency

end, depending in which way it is moved from its mid position.

Denise describe this as adding a lingering spectral effect — and in my book, you can never have too much of those!

Bottom centre of the GUI, you’ll find a control for stereo width, ranging from mono to ‘wide’, and to the right are controls for Frequency Profile, which is essentially a tilt EQ; Decay Profile, which changes the shape of the reverb decay tail; and a ducker with adjustable release time. The ducker essentially compresses the reverb tail while the input signal comes in and then allows it to ‘bloom’ once the initial sound has passed. At the bottom of the screen are sliders for mix and output level, as well as a bypass button, and a helpful Lock button that retains the mix setting when you change presets.

There’s no choice of algorithms here, but the Decay Profile control is instrumental in adjusting the perceived character of the reverb tail decay, and the reverb effects on offer cover shimmery plate-like treatments, small and medium rooms and large indoor spaces. Throughout the range, the result comes over as natural, and the wet and dry sounds combine in a very believable way. Flip to Reverse mode and there’s a world of textures and delay/reverb hybrid effects to explore — they somehow always remain very musical-sounding and

usable — while the tempo‑sync options keep delay and reverse type effects in time with the DAW. I found Perfect Room 2 to be very flattering to piano and guitar in particular, but it also works well on drums and percussion loops, as it retains their presence in the mix rather than burying them. It’s a simple enough plug-in perhaps, but sometimes simple is what you want — and there’s just enough of a departure from the norm to make it usefully different when you need it to be.

Paul White $ $79. Discounted to $39 when going to press.

W www.deniseaudio.com

Fast‑forward to 2020, and we were reviewing an analogue equivalent of the Masterflow called the Drawmer 1976

(https://sosm.ag/drawmer-1976). This 1U rackmounting product offers three bands of saturation, again with control

Softube Drawmer 1976 Saturation & Stereo Widener Plug-in My first encounter with three-band saturation was when reviewing the Drawmer Masterflow 2476 hardware processor, way back in SOS October 1999 (https://sosm. ag/drawmer-masterflow-dc2476).

That processor that might best be thought of as a TC Finalizer with better manners!

Not only did it allow the user to decide which parts of the spectrum to saturate and by how much, but it also enabled the stereo width of each band to be adjusted, and I felt that important. It mean you could keep bass frequencies at or close to mono, helping to avoid any phase issues when playing back over a mono system, while at the same time adding flattering width higher up the spectrum.

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over the stereo width in each band, and now Softube have modelled it to create the Drawmer 1976 plug-in. Unlike the hardware, this plug-in reincarnation has a GUI that appears 2U high, to accommodate the additional features and more precise metering that software makes possible. Saturators are generally used to add musically pleasing harmonics, of course, after the style of tape or circuity incorporating valves and transformers rather than really obvious overdrive, but if you deliberately push this 1976 plug-in, it can, like the hardware, make its presence known more assertively!

The plug-in, which can be used in mono or stereo, adheres to the classic Drawmer aesthetic, with yellow and black knobs, a clear layout and clean labelling.

Its lower section closely resembles the original hardware, with identical control sets for the three bands covering Saturate, Width and Level plus buttons for Bypass and Mute. Each band also hosts a bar-graph meter showing how much saturation is being added. Between the bands, the two crossover controls are adjustable from 70Hz to 1.5kHz for the low/mid boundary, and 800Hz to 15kHz for the mid/highs.

At the top of the window are controls for the input and output levels (which are placed on the left and right of the hardware), along with some controls not found on the hardware: a wet/dry mix control is joined by a pair of global low‑cut and high-cut filters with adjustable frequencies. The low-cut filter comes before the saturation stage in the signal path, and has a range from 21Hz to 1kHz. The high-cut filter comes post saturation, so it can tame any harmonics created that way, and ranges from 1kHz to 2.2kHz. Large meters show the L-R and M-S output levels for both channels, in 16 steps. A button for Width mode selects either Analogue or Digital: analogue emulates the circuitry of the original hardware, whereas Digital is Softube’s interpretation of an optimum response.

There’s also a handy Mono button, providing a fast way to check whether the stereo widening settings are causing any phase problems.

Alongside the Softube logo are buttons that open the input, output and lower panels common to all Softube plugins, showing advanced metering, levels, global bypass, and gain matching either to the input level or to a 0dB output level.

There’s also a phase adjustment slider.

Although operation is very straightforward, the input level obviously affects the amount of saturation that’s applied, so if you want to browse the factory presets, you may have to adjust the input level to get the most out of them; the saturation meters are a great help but these things are so subjective, and your ears really do need to tell you what sounds right.

When processing drums or drum loops, adding more saturation to the lows can add an addictive punch to kick drums, while using saturation in the upper mids and highs can lift out snares and hi-hats. A little high saturation can also add presence to vocals or acoustic guitars. For mastering I like to leave the

mids relatively untouched while adding a little enhancement to both the lows and highs. Those width control can help when mastering too, where my default approach is to keep the lows mono, the mids unchanged and with perhaps just a little more width added to the highs.

There are so many plug-ins that you try once and might be vaguely impressed by, but then end up not using very much in the real world. This one, I suspect, won’t be one of those, as Softube have done a fantastic job. Their 1976 sounds to my ears just like the Drawmer hardware!

Paul White $ €149 (about $150) W www.softube.com

DDK4000

Drum Microphone Kit DPA’s selection of mics for the drum kit provides the most natural and precise sound possible, giving you complete control.

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Modular Contributors

The SOS Guide To Gigging With Modular Thinking of taking your rack on the road? We have some expert advice...

ROBIN VINCENT

A

s musicians, we suffer from that inescapable desire to project our music out into the world.

With the rise in EMOM (Electronic Music Open Mic) gatherings and consistent interest in experimental electronic music, there’s more opportunity than ever to perform live electronically. However, if your chosen vessel of expression is the modular synthesizer, how do you transform your kitchen table experiment in patch cable origami into something performable? In order to bring a broader range of usefulness to the table, I reached out to a number of modular artists and musicians (see the ‘Modular Contributors’ box for the full run down)

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to get some wider insight into how to approach the challenges of performing with modular.

Portability Modular synthesis is somewhat unique in its versatility and complexity. When you grab a guitar and head out to perform as a singer-songwriter, you only have to decide what songs to sing and perhaps which pedal to take. With modular, you have to start by building your instrument. When playing at home, patches tend to bleed out into multiple cases, odd racks, other bits of gear, effects, mixers and anything else a patch cable can reach. So, your first consideration needs to be around the logistics of successfully getting some modular to the venue. It will probably be unrealistic to take everything, so you might need to consider getting a compact case that will travel well. Intellijel and Befaco both do 7U cases designed for this purpose. Perhaps you need two

Alex Theakston, aka Mylar Melodies: Modular and synthesizer YouTuber, podcaster, educator and performer.

Julia Bondar: Electronic music artist and co-founder of Endorphin.es Gaz Williams: Bass player, music producer, modular improviser and co-founder of Rack Records that publishes “spontaneous electronic conversations”.

Steve Davis: DJ, modular performer with the Utopia Strong and co-founder of Rack Records. Used to play a bit of snooker.

Nicola Lancerotti: Double bass player, jazz musician and modular performer.

Martha Bahr, aka Panic Girl: Composer, synthesizer sound designer, nerd.

Loula Yorke: Composer, improviser and performer of modular synths.

cases, but how long would you need to set up the patch between them? I’ve used a pair of Arturia RackBrute cases that work together in a clamshell configuration and has a bag designed to hold them.

If you use compact patch cables, you can pack it away completely patched.

However, it’s cumbersome to carry.

The trick then becomes fitting the sounds you want to make within the confines of the case or cases you can reasonably transport to the gig, and

Martha Bahr, aka Panic Girl.

set up at the other end. There doesn’t appear to be an ideal solution to this. I’ve seen Mylar Melodies and Julia Bondar pull off amazing sets with a single case, and I’ve enjoyed Loula Yorke and Panic Girl exploring the possibilities in two cases and more. I find that the building of the instrument and the choice of modules are actually part of the modular experience and as unique to the player as the music they’re making.

During my first live performance at an EMOM in Norwich, I blundered naïvely into these issues. I was able to push on through, but it made the experience far more stressful than it needed to be.

Another lesson I learned on that first outing was to patch loosely. Just before I packed my case into a bag to take to the gig, I decided to replace any dangly long cables with nice, short and tidy ones. The case looked very neat, but when it came to the performance, I struggled to get my fingers between the tight cables to turn the knobs. And as I already couldn’t see or hear anything, it added yet another level of difficulty.

Some of the struggles our panel of experts highlighted revolve around

Sight, Sound & Safety Nets But what about the business of actually performing? First off, pretty much everyone I’ve spoken to can agree on a couple of universal factors.

Improvisation I asked the panel how pre-patched and pre-programmed they are and how much room there is for improvisation. What’s interesting about that question is that they are not as related as you’d think.

Most of the panel arrived pre-patched

Photo: Victoria Wai

• I t will sound different. You may not be able to hear the things you thought you would; the monitoring may be mono or so muddy you can’t really distinguish sounds or match them to what you’re doing. On the one hand, this can be exciting, on the other, it can knock your confidence and leave you flailing.

• You won’t be able to see anything.

The lighting will probably be terrible, and your modular will be at the wrong height. The darkness, coupled with the brightness of your racks LEDs, will make it impossible to read any labels.

The height problem is less easy to solve as the table provided for your performance could be at any height.

Having an adjustable desktop stand can be helpful. Gaz takes a Stay Music stand with him so that his rig is always at the right height and in the best position for him to play.

logistics, such as having time to set up your patch, warm up your oscillators, and keep them in tune. Other thoughts were around having a big enough rig to cover a long set and being able to transition smoothly from one piece to another. Other struggles were around the nature of the performance itself.

Nicola has had difficulty finding the same level of expression that he’s used to with acoustic instruments, whereas Gaz has pushed back against the pressure to be virtuosic, where he suggests that “more notes and ideas do not necessarily make for better music”. Alex threw in the most devastating comment to imply the ridiculous notion that sometimes what is needed is more practice, not more modules!

All sorts of things can happen on the night, from the gig being completely empty to module failure, power cycling and dodgy patch cables. One tip from Martha is that she always takes an iPad as a safety net that she can run a track from while she’s doing an emergency restart. I’ve seen Loula and Steve use modules like the Music Thing Radio Music with spoken word samples or recorded oddities to fill in time while repatching. It’s an excellent way to mask a transition.

Loula Yorke.

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and ready to play, and some said they would never change the patch in a performance. And yet they nearly all improvise to one degree or another

The Hardest Thing About Performing With Modular Gaz: Feeling the need to constantly innovate and generate significantly different musical ideas that have some cohesion. Keeping certain modules in tune too.

Nicola: Coming from mostly improvised acoustic music, I struggle a lot with expression. It’s harder to translate the dynamics one can achieve with a traditional instrument onto the modular.

Martha: To patch an entire set lasting about 30, 45 or even 60 minutes using just one case. Ideally, the case shouldn’t be too large, as I need to carry it around a lot for live shows. My overall goal is to perform a 45-60 minute set with a minimal amount of gear on stage.

Loula: At this point, it’s actually the logistics.

You will need to check table size and suitability in advance, explain clearly you can’t unplug and move anything, that your oscillators need time to warm up, that you have to have time to set up your patches (but, you know, nicely!). I also suffer mad amounts of anxiety about damaging my gear in transit.

Steve: I struggle with working out how to prepare for the next piece of music without any major hold up in proceedings. Early on, we (the Utopia Strong) did individual pieces, but now we are playing one continuous piece and that makes it even more important to be seamless.

within that patch. There were a couple of exceptions: Nicola would patch up just before going on and improvise from there. Julia arrives fully patched and works everything from the Endorphin.es Ground Control sequencer, recreating songs and album tracks with little or no intentional improvisation.

Overall, there’s a bias towards having something pre-prepared as a starting point, at least some known sequences and chosen sound sources, and then improvising from there. Martha had an interesting approach that brought in some additional instruments to improvise around a more defined modular case: “I need a fair amount of pre-programmed ‘safety’ patches and sounds that will work 100 percent on stage, but I also love to improvise in the moment with certain elements of the tracks. I usually like to use my Cocoquantus for that purpose, along with the OP-1, a touch controller module combined with the Polygogo, or whatever sound source I feel like using at that moment, as well as the Keystep from Arturia.”

Gaz has also started toying with something a bit more prepared after doing a lot of fully improvised performances. “So far it’s been 100

percent improvised but now with the addition of the Oxi One sequencer I’m starting to pre-program certain conditions.

This will allow for more diverse sets without compromising the improvised nature of it. For instance on the Oxi you can store in a pattern parameters that pertain to random note generation. This feels like a way to still embrace free improv but with a certain amount of control of the results.”

From my perspective, I find I have enough room in my case to create three or four different sound generators.

That might be a couple of sequenced oscillator-to-filter synth voices, some percussion and perhaps something that can be played as well as effects that can be fiddled with. For a 15-minute EMOM performance, you can get away with dropping those elements in and out without having to rework anything or improvise too much. However, once you are booked to play for 30 minutes or more you are going to have to either improvise or pack in more sounds, options and variations. On the question of whether to pre-program or improvise, Alex, who always improvises, suggests, “The crowd has no idea, so why bother?

That’s a question. Ultimately, what matters

“Performing with modular has some unique challenges, but they are offset by how engaging the whole process is.”

Julia: The hardest part is getting everything to work together as expected. There are so many technical details to learn, which can distract from focusing on the actual music.

Luckily, I have the most genius person by my side who helps me set everything up properly — we work as a team.

Alex: Improvising completely and making that entertaining. As well as knowing when what is needed is more practice, not more (or different) modules.

Julia Bondar.

Gaz Williams.

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Alex Theakston, aka Mylar Melodies, and Steve Davis.

Nicola Lancerotti.

is what comes out of the speakers and whether you are happy with it.”

Practicalities With all the challenges that modular faces, wouldn’t it be less stressful to perform with easier gear? There’s certainly a sense that the logistics could be easier, and part of developing your set is finding more efficient pathways to reduce the size of your rig. But no one wants to lose the joy, excitement and possibility that performing with modular brings. Steve summed this up nicely: “Yeah, sometimes I’ve thought that it would be far easier just whacking everything into a sampler and just rolling up with a Black Box and a few effects, but then I realise that most of the magic moments we have had have arisen from the unplanned stuff and then just running with them. I’m not sure I’d be so sure if I was a solo performer, but within the structure of a three-person band, there is a lot of time to reflect on that old adage that less is more. I’m pretty sure some of our best music has been when I’ve been scratching my head, holding a patch wire, trying to work out where it was supposed to go!”

On a more practical note, I asked the panel for the most useful piece of gear to take with them. We got several votes for a tuner, and that headphones are essential. For more specific module recommendations, Gaz relies heavily on the ALM Pam’s Pro Workout as his primary clock source, Nicola keeps a Mutable Instruments Beads and Music Thing Radio Music patched together as an instant single-knob-turn safety net, and Steve considers his Korg SQ-1 a personal friend in a crisis.

What becomes clear is that performing with modular has some unique challenges, but they are offset by how thoroughly engaging the whole process is. Modular is a fascinating way to express yourself, from designing and continuing to massage the contents of the case(s), to the practice and learning of the instrument through to the sheer excitement and unpredictability of performance.

I am very grateful to the insight of the panel and to summarise their top tips, practice is always good, and so is more practice, although don’t be afraid to mess up and consider the idea that an unstable performance can be an exciting one. Bring more cables than you think you’ll need, don’t forget the lights and mixers can be useful.

But above all, have fun.

Top Tips Alex: Practice half an hour a day for two weeks, and remember to go more slowly than you think you should when you’re on stage.

Gaz: To accept that with improvised music there will be natural peaks and troughs and to embrace this. Personally, seeing someone play with a laptop, for instance, and have a perfect-sounding set is boring, dull and so old-fashioned now. Having an eclectic and unstable performance makes for a more exciting live experience. I’d strongly urge the performer to angle themselves on stage to allow the audience to see the process as much as possible.

Nicola: Technically, bring always more cables than you need (not just patch cables, but USB, 6.3mm, etc) and have a compressor/limiter out.

Before the gig, prepare: play the instrument, the patch, the composition a lot. Then, on the gig, just step back and listen. Time runs differently for you and the audience. I have played gigs where I thought I was boring the audience and felt like stopping to hear them afterwards say, “Why did you stop so early?”.

Actually, these are all tips for myself too.

Loula: Get a little clip-on LED light. Label important patch cables with a tiny bit of tape. Rehearse, rehearse a lot. I do a full run-through twice a day in the week leading up to a gig, finishing on a rest day where you’re packing down, labelling stuff, making a checklist for loading in and out so you don’t forget anything, and writing a set list with prompts if you need them.

Martha: The most important thing is to rehearse as much as possible, know your gear inside out, and record your rehearsals. Listen to them and analyse what you could improve! Music often feels different when you play live compared to when you listen back to it afterwards; it’s surprising how much it can change. Prepare some safety nets in case something goes wrong, like background tracks that you can play from your iPad. And make a list so you don’t forget anything, like your power supply, tuner, or whatever else you need to make it work. And don’t forget to have fun on stage!

Steve: Get a light; it’s essential! Consider a mixer with the ability to pre-hear what potential crap you are just about to unleash on an unsuspecting audience. There’s nothing worse than opening up a fader and immediately realising that you need to get rid ASAP!

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TECHNIQUE

How To Create Depth Of Field

Any reverb plug-in can be used to create the perception of depth — and your mixes will be more engaging and three-dimensional as a result.

C H R I S T O P H E R ‘J S C H L O M O ’ H E I M E R

M

ixing engineers frequently talk about “depth of field”, and placing instruments in a space.

That sounds grand and complicated, but often it simply means using two or three different reverbs. In this article, I want to share a way of creating a really realistic depth of field that can be used to glue your entire mix, with three instances of just one reverb.

To be clear, there’s absolutely nothing wrong with using different reverbs. In fact, every engineer I know does that, myself included. But doing so does not usually create a realistic perception of depth of field. Rather, it’s about the sound and feel of the specific reverbs used, which is why I call them ‘special sauce’ reverbs. So what is a realistic depth of field?

Points Of View Imagine you’re standing on a hill overlooking a small town, and beyond that, you can see all the way to the open sea. In effect, you’ll see three planes of distance. The houses nearest to you will appear quite sharp, and you’ll be able to spot some details. Beyond that, things start to get a little blurry: you can still see the individual houses, but details soon get lost with distance. And further still, everything kind of blends together, with the waterfront and horizon being the only clear things you can distinguish.

If we transfer this basic concept to audio, we end up with three basic planes: near, mid, and far. It is not by coincidence that you can spot these terms in many

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In our simplified scheme, the room is divided into three zones.

engineers’ reverb setups. Often they are used more or less interchangeably with the terms short, medium and long. What this usually means is that, whatever three reverbs are used and whatever they are called, one is set for a small ambience, one is set for something bigger such as a room, chamber, plate or hall, and one is set for something really large, which could be anything from a hall to a plate or cathedral. This is all fine and valid, and I’m not saying you shouldn’t do this. This is how hit records have been mixed for decades. What I’m saying is that using different verbs with different lengths for the different planes does not deliver realistic depth of field perception. So let’s jump right in and see how you can do this.

I said at the beginning that you only need one reverb. That’s true, and it can be pretty much any reverb as long as you can set up the following basic parameters: pre-delay, high cut, and the balance of early reflections and reverb tail. Of course, in the rare case that your favourite reverb doesn’t feature pre-delay, you could just insert a delay plug-in before the reverb.

I find algorithmic reverbs are preferable for the task, but feel free to use a convolution reverb if you prefer. I use Softube’s classic TSAR Reverb for this myself because it offers all the controls needed, sounds good and works well.

Before setting up any parameters, let’s think about the theory behind it. Imagine you’re not standing on a hill overlooking the town, but at one end of the live room

Three instances of Softube’s TSAR reverb configured for close, mid and far distance settings. You can use almost any algorithmic reverb for similar purposes.

in a recording studio. What actually defines the three planes within a rather small, confined space? It’s the proximity to a wall: not only does a close-by wall give the most obvious early reflections, it’s the boundaries that define the size and sound of a closed space. So the near, mid and far planes in the live room correspond to close to the wall, further into the room, and finally deep in the centre of the room. You could also think of them as progressing from ‘sharp’ to ‘blurry’. What we want to do is set up these positions in the reverb, but all based on the same room algorithm.

(I’m aware, of course, that this is a simplification of reality. Every room has a lot of boundaries, and they are usually made of different materials, especially in a recording studio. The goal of the whole approach is to provide a very convincing and effective illusion of the real world that’s very easy to use.)

Inner Visions First, we have to decide on the actual patch. This is completely up to you, but since the goal is not to swamp your tracks in verb, the actual reverb time probably shouldn’t be too long.

And because we are aiming for depth and ‘glue’ rather than obvious reverb,

the artificial room should sound fairly controlled, just like a good studio live room. This is where a damping control on the reverb plug-in may come in handy.

After you have decided on a nice patch, set up three stereo sends, all with the same patch loaded on the same reverb plug-in, and label them something like AMB A, AMB B and AMB C. Route their fader returns to a bus or group called something like AMB Master, which will let us hear and control our virtual room as one.

AMB A will be our position closest to the wall, so let’s start with that. Leave your pre-delay at 0ms, set the blend of early reflections and reverb to around 75 percent in favor of the early reflections, and your high cut at 10kHz. This tames the sharpest highs but keeps the effect very up-front, which is what we want. AMB B will be our position further into the room, so we will introduce some pre-delay. Start with about 25ms, set the ER/tail blend to 50 percent and the high cut to 5kHz. Moving away from the wall is mimicked by introducing the pre-delay and rolling off more highs, in line with real-life physics of sound: the farther away you get, the more the high frequencies are attenuated (and lows for that matter, but wait). AMB C, you guessed it, will be the position far away in

the room, so we double up the pre-delay to about 50ms, set the blend to 25 percent early reflections only, and further cut the highs, setting the high cut at 3kHz. Take a look at the illustration to visualise the concept again. As I said, it’s a simplification, because we are not taking the ceiling or the rear wall into account. Doing so would make for a far more complex setup, without sounding significantly more realistic. By contrast, what does make this simplified approach more convincing is narrowing the stereo width of the returns progressively: keep AMB A wide, pan AMB B in at about two thirds on each side and AMB C about one third. That way, you are really “looking into the three-dimensional room” as on the picture. Feel free to experiment with this.

Level Pegging So far, so simple, right? Now comes the fun part. In my experience it’s super important to establish a trusted ‘unity’ reference post-fader send level. This is completely dependent on the actual reverb plug-in you use. Sending too little dry signal into the verbs will not produce enough ambience, while sending too much will sound nasty and unnatural. With TSAR I found that -9dB works well on the sends, with the actual www.soundonsound.com / April 2025

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C R E ATING DE P T H OF F IE LD

Compressing The Reverb

The realism of the three-reverb setup can be enhanced by gently compressing the output and smoothing out any resonances with a plug-in like Oeksound’s Soothe 2.

reverb returns or the AMB Master at -6dB. These are just arbitrary numbers I ended up with after listening.

Let’s say we have a rock/pop band mix, because that’s where this approach works best. The drums are usually positioned rather close to the wall, so they get sent to AMB A. And with drums, I mean the whole drum kit. Of course, overheads and rooms will already have significant room information baked in, but we want the whole kit to be placed in our virtual room for the approach to work properly. Of course, feel free to experiment. I tend to send all drum mics at reference level to unify the kit, with only the kick being drier.

Same with strummed acoustic guitars: we typically want those rather dry, so I usually start with sending a little less. Close-miked electric guitar amps definitely benefit from a little more room, so start by sending them to AMB B at a little less than your reference level. For your pianos, keyboards, strings or horns, start with sending them to AMB

B at reference level and adjust from there. Finally send your lead vocals and lead instruments to AMB C. These get the most depth from the room position, but since that has the longest pre-delay, the dry portion stays up front.

In my experience, one reason why any virtual room tends to lack realism is related to dynamics. Imagine actually being in the live room with a band playing. The combined volume of the instruments within a tight and controlled room will make your ears compress pretty much immediately. So I usually compress the AMB Master bus described in the main text, and I like to use an SSL channel for that because I can do my EQ here, too. In the screenshot, you can see I’m low-passing the the whole AMB Master at 6.6kHz to keep it nicely tucked behind the dry sounds.

My preferred approach with any SSL channel compressor tends to work exceptionally well here: threshold all the way down, to ‘dig in’, then raise the ratio from zero until the first yellow light is constantly moving. Here I favour fast attack and medium release to smooth out the transients and add some thickness. A good tape emulation works very well in this spot, too, as does Oeksound’s Soothe 2 plug-in for taming any spiky resonances.

the acoustic guitars barely noticeable, the electric guitars blasting away into the room, and everything else somewhere in between, with the vocals sitting nicely in the music. Unsolo the AMB Master and turn it all the way down. Now slowly raise the fader until you can really hear your mix opening up and losing its two-dimensional dryness. Note that position and raise further until you can really hear the effect and the mix starts to sound ‘wet’.

Note that position, too. Return to the first position and listen for a while. Mute the AMB Master. The mix itself shouldn’t sound different now, but flat. Unmute the AMB Master and lower the fader until you feel you can’t hear the effect. Note that position again. Now you have a window from ‘almost dry’ to ‘rather too much’, which you can work within for different parts of the song.

It takes some practice, but when done right, it’s astonishing how loud you can actually make the AMB Master without your mix becoming obscured or overly wet. Instead the result feels not unlike using a parallel bus compressor that fills the space between instruments and glues your mix together. I’ve prepared a couple of audio files, so have a listen yourself. After you’ve

“There’s absolutely nothing wrong with using different reverbs... every engineer I know does that, myself included.

But doing so does not usually create a realistic perception of depth of field.”

Audio Examples To hear the audio examples that accompany this column, visit the online version of this article at: https://sosm.ag/depth-of-field.

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The beauty of this simple approach is that you can realise not only those three positions, but intermediate positions as well. This works pretty well for background vocals, for example.

By sending them to AMB A, they get some tight room information surrounding them, and by also sending them to AMB B or AMB C you can introduce some of the size of the room, too.

Diving In Now it’s time to listen to the AMB Master in solo. You should hear a nice, three-dimensional ambience that gives you the illusion of standing in the live room, surrounded by the instruments. The drums should sound rather loud and tight,

got the hang of it, there’s still room for further tweaking. Try cutting some lows on the returns (either with a high-pass or with a shelf): start with 100Hz on AMB A, 160Hz on AMB B and 220Hz on AMB C. Try EQ’ing the AMB Master: sometimes some midrange frequencies can get harsh, and in a busy mix, it can help to further roll off lows and highs here. Something like Oeksound’s Soothe 2 also works wonders here to keep rogue frequencies under control.

Also, try panning the sends for your widespread guitars to the opposite side, for a nice sense of width and better mono compatibility.

As an extension of this idea, you can also try using mono reverbs instead of stereo instances. For AMB A, set up three instances hard left, right and centre. For AMB B set up two instances and pan them half left and right. And for AMB C, try just one dead centre, or keep the stereo instance, but narrow the stereo field in your DAW to spread it just slightly. The mono setup is a little more cumbersome in use, but can give you

Send Level Cheat Sheet As a ‘quick start’ reference, here are the settings I usually begin with on a mix, no matter the genre.

Note that ‘unity’ refers to my reference send level of -9dB; if you’re using a different reverb or gain structure, yours will vary. An asterisk means that the send is optional. Panning follows main panner except where noted, but feel free to get creative and really place the instruments where you want.

I tend to stick to a simple rule: with stereo instruments, don’t mess with the panning. With mono instruments, the wider they are panned, the more it can work to pan the sends opposite. I tend to let AMB A always follow the main panner, but no one says you can’t pan those opposite, too. Just listen!

Instrument

AMB A

AMB B

AMB C

Drums (usually less/no kick) Percussion Acoustic guitars (strummed) Acoustic guitars (main) Electric guitars Piano/keys Strings Horns Lead vocals/instruments Background vocals

unity unity unity -3 unity unity -6* unity -6* unity -6 unity none unity -3

none unity -9 unity -6* none unity -3 unity unity -3 unity -6 none none

none none none unity -6 none none none none unity unity -6

a really nice and articulate separation and a super-realistic depth perception.

As I said at the start, this technique is not meant to replace your favourite reverbs. By all means, keep using your

Panning B opposite for panned mono percussion

B opposite for wide mono guitars

favourite vocal or snare reverb as well.

But I find that with this technique you’ll need less of those ‘special sauce’ verbs — and the ones you do use can really shine. I call that a win-win.

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Impact XT is now right at home in the MIDI editor.

ROBIN VINCENT

W

ith Studio One v7, PreSonus decided to pull the interface of their Impact drum machine into the Piano Roll/Drum/Note editor window.

No more having to deal with floating windows when making beats or fiddling with samples. I wonder if this concept will spread to other instruments, because it is indeed very tidy. With the 7.1 update, PreSonus have doubled down on the Impact integration and put a few more bits and pieces within reach without having to load the full GUI. Just to recap, Impact XT is an MPC-style pad-based sample player designed for drum kits and percussive elements. It has eight banks of 16 pads that you can fill with samples, all within a single patch.

These can be single one-shot hits, loops

Studio One

From the Note, Drum or Pattern editor, clicking on the little pads icon will open up a streamlined Impact XT GUI, in which you can access most of Impact’s functions. or instrument sounds, and you have shaping controls dynamically scales and rearranges itself to over the pitch, filtering and amplitude of suit your view. If you make it really small, each pad. The sample display window you’ll eventually lose the pads entirely and gives you a certain amount of editing in be left with just the sample display and terms of start and end points, normalisation parameters. A pair of buttons in the top and reversing. There’s lots you can do with right corner lets you swap between sample layering, round robins, parameters and view and pad view. You have no real output routing, and there is plenty of depth control over what the layout does, so you to explore, but for this workshop, we’re just have to resize the editor until you get going to keep things relatively simple and what you need. focus on the new integration.

The design of the front end is slightly different to the main GUI, but you will find Brace For Impact almost all of the features and controls if you dig around. The main details panel, The magic button to absorb Impact into the usually to the right of the sample display, Note editor is the button to the far right of is now in a rather untidily arranged space the toolbar that looks like two rows of four directly underneath. All the options are drum pads. It’s visible whether you are in there for setting the sample type, quantise, the Piano Roll editor, Drum editor or Pattern tempo settings, choke and offsets; they’re editor. (The score view, it seems, is not just not as neatly presented. You can interested.) Impact appears as if squashed edit the sample in the same way, apply over to the right of the editor, but you normalisation, reverse, set the start and can resize it by dragging the dividing bar. end points (although there’s no Start/End In fact, the layout is very adaptable, and

With the integrated Impact GUI open, you can drag and drop samples directly onto the pads, or onto any of the notes in the Note editor.

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value display) and add a crossfade to loops. The pitch, filter and amp controls are all present, although the envelope settings are hidden behind a button.

The only really noticeable omissions are the mute and solo buttons on each pad. I can’t find these anywhere other than in the original GUI. The output selection control is also missing from the pad, but you’ll find it shifted up into the details panel.

Workflow Now that we have Impact XT nicely nestled into the editor, let’s go through some basics so we can see how well this works. If we start with the default empty Impact patch on a fresh new track, we can now build a kit by pulling samples from the Browser directly into the Note editor window.

Create a clip, open the editor and turn on the integrated Impact. There are two ways you can populate the Impact pads with samples: you can drag and drop each sample directly onto the pad that is now right there in the editor, or you can drop it onto the note in the editor that corresponds to the pad. As you do so, all the other notes disappear, making the kit building less cluttered. You can add further samples underneath the note, to build up the kit across the pads. If you have a bunch of samples that are already in the right order, you can select them all in the browser and drag them across together. Before you drop them, hold Shift and they will spread themselves across the pads automatically.

Just a note on undo/redo with Impact.

If you drop a sample onto the wrong pad, your instinct might be to hit undo and try again. Unfortunately, that doesn’t work in this particular case. Undo applies to moving parameters, not actions like the dropping of samples. Your only option is to drop another sample over the top to replace it, or right-click the pad and select Clear Pad.

Loop slicing in Impact XT will automatically chop up a loop and place the slices onto individual pads. In the full version, but not the integrated Impact, you’ll also get a new MIDI part, which will trigger the pads in order to recreate the original loop.

update it will also generate a MIDI part on the instrument track that will play all the slices back in the rhythm of the loop.

However, that doesn’t appear to work with the integrated version, or when you drop the loop onto the Note editor. You have to use the full Impact GUI, and then it only works if the last thing you did was click on the arrangement before dragging the sample. I think there’s something odd going on, as whether the MIDI part appears

can’t drag it around like you can in the standard Impact. With the full GUI pads, you can hold Ctrl (Windows) or Cmd (Mac) to drag a copy of one pad to another, hold Ctrl+Shift (Cmd+Shift) to swap the samples and Ctrl+Alt (Cmd+Option) to copy just the parameters from one pad to another.

You also can’t select multiple pads in the integrated version, which is easy to do in the main GUI by holding Shift.

We also hit the same problem when we want to pull samples across into SampleOne, as you can’t pull anything off the integrated version’s pads, regardless of how many keyboard combos you try. However, with the standard Impact open and pinned, you can pull samples or slices across into SampleOne and turn them into an instant instrument.

By contrast, you can pull samples from SampleOne into the integrated Impact without any bother. It’s a really neat level of cross-instrument integration that needs to make its way into the integrated version.

I’m sure it will get there, but at the moment it’s a bit frustrating having to open up the full Impact for a few quite everyday tasks.

“Studio One 7 introduces a new feature called the Instrument Bus, which is going to sort out the console chaos.”

Going Loopy Working with loops was helpfully made easier in version 7, but this is also where we start to see the discrepancies between the integrated and the full Impact GUI. If you hold Shift while dragging a loop into Impact, it will automatically slice the loop across multiple pads. It’s a really easy way to pull a drum kit out of a loop, or do some sample recycling. With the new

or not seems to be a bit inconsistent, but the sure fire way seems to be to ensure the arrangement is in focus before doing it.

The auto-slicing is normally very good, but if the results are not exactly what you were after, you can edit the start and end points of each slice. Click on the pad that holds the slice you want to edit. In the display, you can zoom in and out and move the start and end markers to precisely where you want them to be. They should automatically land on zero crossing points to minimise the possibility of clicks, and the new crossfade parameter can help to smooth out any jerkiness.

Pad Behaviour The integrated Impact lacks some of its usual facilities when it comes to moving samples. It appears that once a sample is on the pad in the integrated GUI, you

Instrument Bus PreSonus provide some powerful mixing tools for use with their own instruments.

This has always been very evident in Impact XT; whenever you open it, you seem to see a couple of dozen extra channels appear in your mixer. Each pad can then be routed to an individual channel, giving you godlike powers of level control and audio processing. However, www.soundonsound.com / April 2025

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As of version 7, loading a new instance of Impact automatically creates an Instrument Bus, giving you all the routing flexibility you’ll ever need but with the ability to hide the instrument’s sub-channels when not needed.

a lot of the time, those extra channels were just getting in the way, and you found yourself having to scroll past them to get to what you were actually interested in.

Studio One 7 introduces a new feature called the Instrument Bus, which is going to sort out the console chaos and give us a lot less scrolling. When you load a kit into Impact and then open the console, you’ll find just the one Impact channel strip. At

the bottom of the fader, there’s a tiny folder icon with three vertical lines. Click the folder, and all the sub-channels will appear.

Click it again, and they all neatly slide back into the folder, so you don’t have to worry about them unless you want to.

Note Editor One last highlight from the new version of Studio One is designed for melodic parts

You can greatly speed up your drum programming by showing only the notes that are in use.

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but is actually very relevant to Impact XT and drum programming in general: with the new Scale options in the Note editor, you can select different scales and reduce the familiar piano keyboard down to show only the notes in a given scale. OK, that’s not relevant to percussion, but the next bit is... There’s also an option to show only the notes on the piano that are being ‘used’. So, if you were writing a drum part that just used kick, snare and hi-hats, you could lose all the other notes and just focus on that pattern. Weirdly, this feature only appears in the Note editor for instrument parts, and doesn’t work in the Drum editor or Pattern editor views, which feels like a missed opportunity to me. Maybe its usefulness in pattern writing will make itself known, and PreSonus will update Studio One accordingly.

We last looked at Impact XT back in 2019, and sadly not a lot has changed with the instrument itself, though the handful of adjustments are definitely worth looking into. However, the new integration into the MIDI editing windows is a really great move from a workflow point of view, helping to keep those floating windows under control and laying the foundations of a really powerful beat‑making system.

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Pro Tools Live

TECHNIQUE

Transform your MIDI with Live 12’s new tools.

Arpeggiate will transform chords into arpeggios.

Ornament creates flams and trills.

OLI FREKE

I

n the August 2024 column we covered the stimulating world of Live 12’s MIDI note generators. And now that we’ve generated all these notes, what are we going to do with them? Transform them, of course!

To save you, the humble electronic music genius, from having to manipulate these notes manually, why not get the computer to do it? The new (MIDI) Transformation Tools are your new (MIDI) robots in disguise!

Live 12 came with a raft of intriguing and useful MIDI Transformations, which can manipulate notes in a MIDI clip. Some reduce the time it takes to create frequently used effects, and some can create melodic and rhythmic variations on existing material, potentially re-stimulating the creative juices.

You will find these 12 new options in a MIDI clip’s MIDI Transformations tab. What follows is a brief overview of each and their most likely applications.

be edited further on the grid, ensuring the results meet your needs. In keeping with Live 12’s new features, the Distance setting can be set to Chromatic or Scale Aware, a nice touch.

Something to note is that if you trigger or edit a different MIDI clip from the one you are working on the arpeggiation notes on the piano roll will become permanent, and you won’t be able to toggle back to the originating chord sequence again.

If I could request one feature here, it would be great to have Euclidean-style rhythms available in the presets, or the ability to mute notes in the pattern to give more rhythmic variety.

Strum As you might guess, Strum emulates the natural feel of strummed chords on a guitar or similar stringed instrument.

Instead of all the notes in a chord

Arpeggiate The Arpeggiate Transformer builds on the Arpeggiator device and contains a subset of its options (Styles, Steps, Distance, Rate and Gate). Given that it acts on selected notes in a MIDI clip it’s no surprise you can see the notes change in real time as you change the parameters, which is, of course, extremely handy. They can then

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Strum emulates the sound of strumming a guitar.

triggering simultaneously, Strum introduces a timing offset between each note.

This is beautifully implemented and easy to use: simply drag the bottom of the ‘pendulum’ left or right, and the selected chords will shift according to that slope.

The slope can be curved using the Tension control, and the top of the pendulum can also be shifted using the High control (which accordingly moves the top note of a chord around). It’s simple, fun and effective. To get the best results I would recommend taking the time to give each chord its own strum setting, which will help keep things feeling organic and varied.

Something to be aware of is that if you have a chord starting on beat one of the first bar (a pretty normal thing to have!), moving either the top or bottom of the strum start point to the left will move one or more of the chord’s notes to before the start of the bar, so you won’t hear it. A way of locking notes to not move beyond the start of a clip would be helpful here. (Or the ancient Emagic Notator feature that could trigger notes before the start of the clip could be one to revive...)

Time Warp Time Warp applies non-linear time stretching or compression to MIDI clips, creating rhythmic effects like accelerandos, ritardandos and tempo fluctuations.

This is possibly the most fun of the bunch. The simplest use case is to take a snare roll, and apply a subtle — or not so subtle — accelerando or decelerando.

It’s also easy to create the ‘ticky hats’ of grime, trap and drill.

Applying the effect to a full drum part could be useful as a slowing up effect on a last repeat, but can sound a bit chaotic on a looping beat (depending on how much of a fan of Autechre you are, perhaps!).

Ticking Quantize keeps notes on the grid, and in that mode can generate some interesting ideas and variations.

For more traditional film and TV scoring, or more melodic (less beat‑driven) music,

Time Warp applies time-stretching to notes.

the ability to gently tweak the tempo can help create a more human feel. This kind of effect can take a long time to implement manually, especially if you want to try some different options, so this definitely comes under the category of time saver as well as a time manipulator!

Ornament Ornament adds decorative notes or embellishments to existing MIDI notes, creating trills, flams and grace notes. And it does exactly what it says on the tin, adding a little flair to melodies, albeit in a somewhat baroque manner, which is probably due to these being actual Baroque ornamentations.

Adding a multi-note trill is fine, but they are so fast most instruments just make them an indistinct blur. It’s usually necessary to slow them down by use of the Position control each time. Also, if the trills are set to occur before the start of the original note, the previous note is not shortened to accommodate, so a polyphonic sound can result in a messy overlap. Not critical issues as they are easily worked around, but maybe a couple of opportunities for tightening up the feature in future.

Recombine Recombine rearranges the order, pitch, velocity or duration of notes in a MIDI clip, creating new melodic and rhythmic variations. I found this one to be entirely unpredictable; if one has a chord sequence the notes are arbitrarily swapped around in a most disconcerting way and a melody can have its notes ‘rotated’, which means shifting them along in relation to their starting position, which is visually intuitive when there’s only a couple of notes, but easy to lose track of with more complex progression.

Useful for throwing the cards up in the air and seeing where they land, I suppose!

I dropped the Live team a line to see if I was missing anything obvious, but they acknowledged it can seem a bit complicated and have recently been

making efforts to simplify it for those wanting ‘intentional outcomes’, but that yes it can just be used for ‘card shuffling’ too!

Connect Connect will generate notes in between existing ones. I had fun creating some anchor melody notes alongside a chord sequence and then using Connect to create a ‘fill-in’ melody between those points. It’s scale aware when Scale Mode is on, and chromatic when off.

It was equally interesting when used on a standard multi-note drum part.

New drums popped up in interesting syncopated places. And using smaller grid values in the Rate setting lends itself to glitch styles. More generally, this tool seems like another very usable option for generating rhythmic variations and melodic options you might not think of otherwise.

Chop

Pressure. (Like Glissando, this only works with MPE enabled instruments).

Span Span focuses on the lengths of notes. It can automatically create legato notes, or uniformly short, staccato notes. The difference between tenuto and the other options (legato and staccato) can be imperceptible with some settings, but apparent with others. The Variation control is neat, re-randomising the lengths of each note to avoid too much uniformity.

Velocity Shaper Velocity Shaper is a quick and handy way of applying a velocity curve to a group of notes. Not only is it easy to create crescendo/diminuendo slopes, but any arbitrary shape can be applied with minimum fuss.

Transformed

Though simple in concept, Chop allows for some surprising variety by merely slicing a note into multiple parts. Playing with the Variation percentage is where things get interesting — the individual parts are moved away from their rigid grid locations into free‑floating non-grid space. Could be useful for creating interesting percussion parts and other melodic interventions.

In summary then, these MIDI Transformations vary between the expected and obviously useful to the novel and somewhat unpredictable.

The integration of MPE and the additional ability to import your own custom or third‑party Max For Live effects is potentially very powerful too.

It didn’t take me long to try out each Transformation tool, get the gist of what Glissando (MPE) it did, and to start thinking of how I might find it useful in the long run. And then, All Live 12 instruments are MPE capable without, even realising so this rather pleasing it, I’d started reaching Transformer will work unconsciously for on any of them, as Ornament when I wanted well as the increasing a drum flam — it was just number of third-party quicker to use than to start MPE‑capable instruments. duplicating and dragging It’s more fun to play with round tiny, fiddly notes on that you think it will be! the grid. I think this shows LFO that these new MIDI tools can embed themselves A handy MIDI-based into your ways of working LFO that can control an as their true utility reveals instrument’s pitch or the itself over time.

LFO is an MPE‑generated LFO. MPE parameters Slide and www.soundonsound.com / April 2025

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JULIAN RODGERS

T

here are many ways to finesse a drum recording at the mix stage, from manual editing to audio quantisation tools like Elastic Audio and Beat Detective. And as virtual drum instruments have evolved, many producers now prioritise the control and flexibility of MIDI-triggered drums over traditional audio-based editing. However, replacing or augmenting real drum performances with third-party samples from a virtual instrument can compromise the individuality of the drummer’s sound. A hybrid approach, using samples derived from the original drum recording, offers the best of both worlds: the control that virtual instruments offer, without changing the core drum sound itself. That’s what we’re looking at this month.

The advantages of this are threefold.

It retains the character of the original performance, it avoids the potential for sonic mismatches between the original and replacement drum sounds, and it provides the kind of forensic control over timing and dynamics that users of drum virtual instruments are accustomed to. So how do you do it?

Sample Recording Creating a library of drum samples can be time-consuming if you’re doing it properly.

There’s a lot of editing and file management involved. If you only need to fix a couple of snare hits this might be overkill. But if you want full control while staying faithful to the sound of the original performance, here are some tips to get you started.

Firstly, while it is possible to create a library from a performance, if you anticipate wanting to go deep into sampling, it’s well worth putting some time aside for capturing some clean samples at the tracking stage. In the same way that a film crew’s sound recordist captures clean ‘room tone’ to help with edits later on, spend some time with the drummer capturing a variety of clean hits at varying velocities and with all the articulations you anticipate needing.

Often this isn’t possible, but identifying clean hits from the performance can usually still give perfectly usable results — you just might not have the luxury of clean tails, and so may need to edit the samples a little tighter. You’ll need several alternative hits at each velocity, and while you might think that getting a lot of variety in your hits will add life and authenticity, in my experience, the most consistent hits give the best results. Hits that sound identical

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There are many great virtual drum instruments around, but sometimes it’s best to create your own!

Here, a snare top and bottom mic sample are being bounced together via a bus, so that the final sample contains the complete snare sound.

inevitably aren’t, but if you use only a single sample, you’ll immediately hear the artificial ‘machine-gunning’ effect of the same sample being repeated too close together.

Top & Tail Editing hits relies on familiar tools like Tab to Transients, Strip Silence and Separate at Transients, which speed up the process significantly. To ensure consistency, drag alternative samples onto the timeline with Timeline Drop Order set to Top to Bottom and trim them to equal lengths.

Once on the timeline, check that sample starts are all tight with the transient, then select across all of the samples and use ‘S’ to trim them to length. Use Option+Shift+3 to consolidate clips, and right-click them in the Clips List to reveal in Finder for subsequent file management.

If you are simply capturing individual close mics as samples, then things are reasonably straightforward.

However, if you have multiple mics on a kit piece — for example, a top and bottom mic on a snare — Groove Cell is great for quick sampling jobs, but has its limitations.

For example, you can’t implement both velocity layering and round robin playback at the same time.

then you’ll need to pre-mix those samples.

To do this, make an Edit Selection across the hit, soloing the channels you need, right-click on your drum bus, and select Bounce to create a complete snare sample.

This includes any plug‑in processing on the bus if desired, and you can incorporate overheads or room mics too if you like.

Care should be taken to check edits and apply fades to keep samples clean.

Once complete, you should choose a naming convention and stick to it. I use the following: Project, Kit Piece, Articulation (if necessary), Velocity, Sample Number.

For example: “ADB snare rimshot hard 1”.

Finally, create a sample folder and keep things organised!

Go Big How detailed your sampling needs to be depends on your project. I visited a friend’s studio recently where they had spent

AIR’s Structure may be dated, but offers all the tools you need to create a fully featured drum library.

many hours preparing a fully playable library of TCI files using Slate Trigger 2 for a project. It had multiple velocity layers, with each having multiple round robins of well-matched samples. (A round robin is a set of multiple alternative samples within a single velocity zone, set up such that a different sample plays each time that zone is triggered.) It was a lot of work.

At the other end of the spectrum, I often record my band’s rehearsals using very minimal miking on the drums — usually a single overhead and a mic on kick and snare. I’ll augment this at the mixing stage with some tom samples, which I’ll capture by temporarily moving the overhead mic onto each tom to record some clean hits.

This is preferable to using a third-party sample because it matches the tom sound in the overhead. Both use the same techniques, but the scope of the libraries is very different.

Tools For The Job While Pro Tools’ built-in samplers have limitations — such as GrooveCell’s inability to combine round robins with velocity layers, which you would ideally use at the same time — there are still ways to get useful results. For simpler tasks, GrooveCell is a quick and effective option. For more detailed work, premium sampling products like Kontakt offer the deeper functionality necessary for really in-depth library creation.

AIR’s ageing Structure Free is too limited for complex sampling, but I’ve been a user of the full version for many years, and it’s still ideal for this kind of work. Pro Tools 2024.10 introduced a partnership with Native Instruments, and users with an

update plan or subscription get access to some curated content for Kontakt 8 Player.

Kontakt Player itself is free, and while you can sample into it, this is restricted to creating custom Leap presets. This is a great feature for looping, but for detailed sampling like this, you’ll need the full version. A few years ago, Avid partnered with UVI and bundled Falcon with subscriptions and update plans. Falcon is a synthesizer and also a very capable sampler, but is no longer included with Pro Tools.

Whatever software sampler you favour, the good news is that with the 2024.10 release, a long-missed function of Pro Tools returned: drag and drop of clips from the timeline into sampler plug‑ins. The workflow of editing clips on the timeline and dropping them straight into the sampler is super-fast, especially for simple sampling jobs.

original drums and third-party replacements can be an issue when using drum libraries, and using your own samples avoids this.

The friend I mentioned earlier had a good example: they wanted to use only the room mics in a section of the song, which would have made the use of third‑party samples very obvious. However, because he was using his own custom samples, it sounded consistent with the original recording.

A common challenge when using custom samples from the original kit is phase cancellation. Since the samples are closely related to the original drum tracks, they may interfere sonically. Using a phase alignment plug‑in is advisable, but at the very least, check polarity to ensure that the real drums and samples are reinforcing each other in the mix.

By combining the flexibility of MIDI with the unique character of your artist’s live drum performance, rolling your own samples offers a powerful alternative to traditional editing or sample replacement.

Whether you’re subtly reinforcing a live kit or creating a fully MIDI-controlled drum track, taking the time to capture and curate your own drum samples ensures that the final result stays true to the original performance while gaining the benefits of modern production tools.

Trigger Warning Samples loaded into a plug-in can be triggered either using MIDI notes, or directly from the audio on a drum track using third-party tools such as Slate Trigger 2, WaveMachine Labs Drumagog and Toontrack Superior Drummer’s Tracker feature. If you’re taking the former route, the Audio to MIDI function in Pro Tools does a great job if you have clear audio to generate triggers from. If you find your preferred mic placement on a particular kit piece suffers from spill, use a gate on a duplicate track (not routed anywhere you’ll hear it) so you can use extreme gating settings without affecting the mix.

You can also use the Select / Split Notes feature in the Event Operations window to quickly remove low-level false MIDI triggers caused by spill.

Once you have a set of MIDI triggers and suitable samples in a sampler, the benefits of all this work come to the fore. With MIDI controlling the performance, it’s easy to alter the drums subtly or dramatically.

Mismatches in tone and tuning between the

If you want to trigger your samples from an existing audio performance, the Event Operations window can help to clean up the trigger signal.

www.soundonsound.com / April 2025

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Pro Tools Cubase

TECHNIQUE

Want to awaken those sleepy synth pads? Then give Cubase 14’s Modulators a try!

JOHN WALDEN

I

f existing Cubase users needed a single excuse to upgrade to Pro 14, the new Modulator options might well provide it, especially if your music makes use of synth-based elements.

While lots of modern virtual synths have modulation options on board, those only let you modulate parameters inside that instrument. In contrast, Cubase’s new Modulators have track-wide reach — they grant you access to any parameter on the selected track, whether in the instrument itself, or in an insert effect or a parameter of the channel hosting it.

The creative sound‑design options the Modulator system offers are almost limitless but let’s start an exploration of the potential with three (from many) different options the Modulator system provides for tackling a common task: banishing the ‘boring’ from a simple synth pad.

Inside & Out There are contexts where a static pad sound, such as the Retrologue sound I’ve started with in the audio examples (available on the SOS website: https:// sosm.ag/cubase-0425), can be just what the musical mood requires. But if you need to add some sonic movement, parameter modulation is a great place to start. Retrologue has its own well-featured modulation system, of

Spicing up our static pad with the new Modulator options by targeting parameters both in Retrologue itself and insert effects on the Retrologue channel.

course, but in this example the preset I’ve created makes no use of that.

Instead, I’ve opened the new Modulators panel in the Project window’s Lower Zone and, as shown in the first screenshot, used its modulation options instead.

The new Modulator system provides six different modulator types: LFO, Envelope Follower, Shaper, Macro Knob, Step Modulator and ModScripter. Any combination of these can be added in the eight available slots on the selected track/channel. To start things fairly simply, this first example just uses two instances of the LFO Modulator type and one of the Shaper. The LFO instances are both sine waves, but are set to different Note (speed) values, and with some Shape and Phase changes applied to the second instance. These target parameters in the synth’s engine, including Filter Cutoff and Osc 3 Fine pitch (LFO 1), and Filter Resonance and Osc 2 Fine pitch (LFO 2). The system for adding targets to a Modulator is beautifully implemented — in the majority of cases, you simply hit the large ‘+’ button, then click the desired target parameter.

In this example, all the Modulation Depth values are set quite modestly, so the changes to both filter and pitch are subtle, while the Note settings mean the modulation also occurs quite slowly.

Even with just these two Modulators

applied, the pad benefits from some nice tonal and pitch movement.

Up to this point, these modulation moves could have been created in Retrologue’s own modulation matrix. The Shaper modulator, though, demonstrates one of the key advantages of the new Modulator system, because it’s targeting parameters outside Retrologue. In this case, the targets reside in two insert effects, Destroyer and Chopper, but they could have been any parameter in the channel that’s hosting Retrologue.

What you have, therefore, is an integrated, channel-level modulation system, where individual Modulators can provide synchronised modulation of both instrument and channel/effects parameters. This can make for some very cool sound-design possibilities!

Having added my own tweaks to Shaper’s modulation curve, the modulation is targeting Destroyer’s Drive control (changing the amount of distortion added) and Chopper’s Mix control (changing the depth of the volume ‘pulse’ Chopper is adding).

Whether in the plug-ins themselves, or in any of the three Modulators, there are already plenty of adjustments you could make to finesse the result. But the sound has gone from a static pad to something with tonal, pitch, distortion and volume modulation and, for the purposes of our demonstration at least, is sonically more interesting.

Up The Intensity

Don’t let ModScripter’s JavaScript coding possibilities put you off exploring: there’s an interesting selection of presets provided, including the Intensity option shown here.

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The Modulator system allows one Modulator to modulate another, so there are plenty of creative possibilities, but there’s one caveat. In this first iteration, at least, you can only use a Modulator in one slot to modulate the controls in the

main panel of another modulation slot; the parameters in the target parameter sub-panels, including the Modulation Depth controls, cannot themselves be selected as modulation targets, and in terms of our ‘pad enhancer’ task, that’s a bit of shame...

For instance, in our previous example, we could have added an instance of the Macro Knob in an additional slot and used this to target the Modulation Depth controls in other modulators, thereby intensifying the resulting sonic changes. Unfortunately, that’s currently not possible. Thankfully, by combining the Macro Knob modulator with another highlight of the new Modulator system, the ModScripter modulator, we can create a workaround that achieves a similar type of end result.

The second screenshot shows a possible configuration. ModScripter is perhaps the least intuitive (most intimidating?) of the modulation types. As the name suggests, it includes a scripting panel that lets you code your own modulation scripts using JavaScript.

Thankfully, for those of us who are here to make music rather than code, ModScripter comes with some very useful presets, accessed via the drop-down menu located top right. There are some intriguing options amongst this selection, including the rather wonderful Apply Randomness, but for this example I selected the Intensity preset.

ModScripter’s control set includes a Modulator Input control (the slider beneath the waveform display) and this particular preset script also provides an Intensity control. In our example, Slot 1’s Step Modulator is sending its step-based modulation pattern to ModScripter’s Modulator Input; that slider is controlled by the Step Modulator. However, the Intensity control can then be used to modify the (yup!) intensity of that modulation signal, essentially scaling it from 0 to 100 percent. This modified (modulated) modulation pattern in ModScripter is then used to target both the Filter Cutoff and Filter Resonance, but could potentially control any other parameters on the Retrologue channel.

Finally, in Slot 3, an instance of Macro Knob is configured to target the ModScripter’s Intensity control, and as I have Macro Knob linked to my MIDI controller’s mod wheel (courtesy of the

If you use a multitimbral instrument such as HALion Sonic, blending between pad sounds on different layers can be easily configured on a single Cubase channel.

MIDI Remote system), I have hands-on control: moving the mod wheel adjusts the Macro Knob, which in turn adjusts ModScripter’s Intensity, which in turn scales the Step Modulator modulation data transmitted via ModScripter to Retrologue’s filter. Got all that? Phew! Yes, it can at first seem like data-modulation spaghetti, but for our boring pad, it does provide a way to manually control the intensity of any modulation being applied to the sound. It’s not exactly the

I’ve set up two (boring!) pad presets in HALion Sonic’s first two sound slots, but have configured both slots to respond to MIDI channel 1. So whatever notes I play will trigger both sounds. I’ve then configured a couple of Modulators to spice up each of the individual pad layers, much as I did for the Retrologue example earlier.

The blending fun really begins when you use a single Modulator to target both pad layers. Slot 3, using a Macro Knob, shows this at its most simple.

I’ve assigned the volume of each layer to the HS Quick Controls, and then targeted these using the Macro Knob, but with the Modulation Depth set in opposite directions for each layer. As you rotate the knob, one layer will get louder as the other gets quieter, meaning you can adjust the sound blend as required.

You can also do exactly the same sort of thing with other parameter targets, and suitable candidates include pan, filter cutoff and pitch. There are all sorts of possibilities here, and to give you a feel for what you can do, I’ve provided a couple of additional options in the audio examples on the SOS website.

It’s so easy to configure the modulation for all of the layers because it can be done in a single location, using the new Modulators panel.

“The Modulator system allows one Modulator to modulate another, so there are plenty of creative possibilities.” same result you’d get by adjusting the Modulation Depth sliders, but it’s certainly in the same ball park!

Multitimbral Modulation Another strategy for banishing bland from pad sounds is to layer and/or blend two sound sources together. This could, of course, be combined with applying modulation to the two (or more?) individual layers. However, as the new Modulator features work at the track level, if you use two separate virtual instrument tracks for your blending, you can’t easily ‘share’ Modulators between the two layers; each requires their own Modulator configuration...

Except that you can! In fact, there are a couple of options. One of them exploits the audio side-chain support that’s offered by some of the Modulators — these have a whole range of applications, by the way, and I plan to explore them in more detail in another workshop. The second option is much more pragmatic: simply use a multitimbral virtual instrument! In Cubase’s HALion Sonic (and the full version of HALion), Steinberg have an excellent candidate for this. In my final example, therefore,

All Mods, No Cons!

I have to say that the Modulators are a brilliant addition to Cubase Pro, and absolutely brimming with potential.

I’ve barely scratched the thinnest of surfaces here in terms of what’s possible. My experiments to date suggest that the side-chain audio input mentioned earlier is well worth further exploration, and the Envelope Follower modulator also seems to hold plenty of promise. So watch this space — this is undoubtedly a topic we will return to in a future column or three! www.soundonsound.com / April 2025

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SPOTLIGHT

Battery-powered PA Speakers From busking to barbecues, gigging with no mains power is now an affordable reality. We check out the current options.

LUKE WOOD

T

here’s no shortage of portable PA gear these days, and thanks to efficient amplifiers and clever cabinet designs, even light, compact systems can pack a serious punch.

However, it doesn’t matter how easy a rig is to transport if there’s nowhere to plug it in! Thankfully, there are now plenty of options available that can run entirely on batteries, offering a great solution for outdoor events or those awkward rooms where there are no spare power sockets to be found. They come in all shapes and sizes, from tiny units designed for rehearsing or providing light reinforcement to much larger boxes that can be linked together to create a serious mains-free alternative to traditional PA systems. So, as you might have guessed, this month’s SOS Spotlight is focused on battery-powered PA speakers equipped with built-in mixing capabilities.

Alto Busker

Alto’s aptly named Busker houses a 6.5-inch woofer and one-inch tweeter with up to 200W of power in a compact and lightweight cabinet that can be transported easily, but is durable enough to stand up to some serious use. Its angled cabinet design allows it to be tilted up in both horizontal and vertical positions, and there’s a built‑in mount for attaching it to a speaker stand. For those wanting to run a stereo setup, it’s possible to link two Buskers wirelessly via Bluetooth. An integrated three-channel mixer offers a pair of XLR/TRS combo

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Alto Busker

sockets for mic and line-level sources, with both wired and wireless aux connections provided by 3.5mm and Bluetooth connectivity; you also get a selection of built-in Alesis reverb, delay and modulation effects, along with Mix and Record outputs (on XLR and RCA respectively) and a USB port for charging a mobile device. As well as being able to run on mains power, the Busker has an onboard battery that can offer up to 24 hours of operation thanks to a Battery Eco Mode setting. Just as we were about to go to press, Alto announced the new TX4 Series, which includes the TX412B, a 350W 12-inch cabinet that features a built-in two-channel mixer with mic, line-level and Bluetooth connectivity. $ Busker $349, TX412B $349.

W www.altoprofessional.com

that offer XLR/TRS combi sockets along with built-in wireless support for the included mic, while their stereo channel offers both 3.5mm TRS and Bluetooth connections; the MPA100BT can be switched between Music and Speech modes, and the MPA200BT includes some onboard processing courtesy of Klark Teknik. For larger events there’s the C210B, a 160W compact line array that pairs an eight-inch sub and four 2.75-inch drivers with a three‑channel mixer that accepts a mixture of mic, line-level and instrument sources. It also boasts an onboard media player that will play tracks from a USB stick or SD card, as well as wirelessly over Bluetooth. $ From $209.

W www.behringer.com

Behringer MPA Series & C210B You’ll find a number of options in Behringer’s MPA range, starting with the MPA40BT and MPA40BT-PRO, a pair of 40W systems with a built-in three-channel mixer that offers two mic/instrument channels and a stereo channel equipped with RCA and wireless Bluetooth inputs; both units feature a master two-band EQ and offer support for Behringer’s range of wireless mics. The PRO variant occupies a slightly larger cabinet and features integrated wheels and an extendable handle. Next up are the MPA100BT and MPA200BT, 100W and 200W systems that each come bundled with a wireless mic.

Both feature a pair of mic/line channels

Behringer MPA200BT

Bose S1 Pro+

Bose S1 Pro+

Bose are a popular name in the portable PA world, and their battery-powered offering comes in the form of the S1 Pro+, a compact unit that can be either freestanding (horizontal or vertical tilting is possible) or standmounted. A pair of mic and line-level inputs are provided by two channels equipped with XLR/TRS combi inputs, with a third channel offering wired (quarter-inch and 3.5mm TRS) and wireless (Bluetooth) connectivity for playback devices. There’s also a line-level output that can be used to feed a second device. A rotary encoder on each channel offers control over level and a two-band EQ, with the two mic/line channels also benefitting from a built-in reverb and a range of ToneMatch EQ presets — the ToneMatch library can be managed using the Bose Music mobile app, which also offers full control over the S1 Pro+.

At lower playback levels, the system is capable of running for up to 11 hours when powered by its built-in battery, and Bose say users performing indoors to an audience of up to 100 people can expect around five hours of operation. For those looking to go completely cable-free, the company also offer a range of mic and instrument RF transmitters that can be stored in, and charged by, an S1 Pro+. $ $699.

W www.bose.com

E‑V Everse 8 & Everse 12 E‑V say that the aim of their Everse range is to deliver a step up in sound quality for the

battery-powered PA market, while still delivering high SPLs and full-bandwidth playback. There are two models available: the Everse 8 and Everse 12, which, as their names suggest, are equipped with eight- and 12-inch drivers respectively.

Both are capable of operating for up to 12 hours on a single charge, with the Everse 12 offering a higher maximum SPL and extended low-end response — 126dB and -3dB at 55Hz compared to the Everse 8’s 121dB and 60Hz figures.

An onboard four-channel mixer features a pair of XLR/TRS combi sockets for mic and line-level signals, along with 3.5mm TRS and Bluetooth connectivity for playback devices, and there’s a 12V DC output that can be used to power E-V’s wireless mic systems. If you’re looking to put together a larger system, it’s possible to link up to six Everse units wirelessly using Bluetooth. Some processing and effects are provided courtesy of sibling brand Dynacord, and in addition to the local controls, the mixer can be operated remotely from a mobile device using the E‑V QuickSmart app. Both systems boast a compact yet durable design, and are even IP43-rated, meaning they’re resistant to splashes of water and light rain. $ Everse 8 $749, Everse 12 $999.

W www.soundonsound.com/reviews/ electro-voice-everse-8

W www.soundonsound.com/reviews/ electro-voice-everse-12

W www.electrovoice.com

Fishman Loudbox Mini Charge

Fishman’s Loudbox range comprises five amplifiers that are primarily aimed at acoustic guitarists. The Loudbox Mini Charge takes the popular 60W Mini variant and adds a built-in battery that allows for up to 18 hours of operation at low levels, or four hours at full volume. The first of its two input channels is dedicated to instruments, with a quarter-inch TS input designed to accept a signal from passive or active pickups followed by a polarity button, three-band EQ section and a pair of knobs that E‑V Everse 12

Fishman Loudbox Mini Charge

control the level of the amp’s onboard reverb and chorus effects. The second channel features an XLR input for connecting a microphone, and includes a two-band EQ along with a reverb level control. $ $499.95.

W www.fishman.com

HK Audio Move 8

Described by HK Audio as “the battery-powered all-rounder”, the Move 8 provides users with a versatile all-in-one system capable of tackling everything from background music playback to live performances. A part of the company’s Premium PR:O family, the system promises to deliver transparent, full-range sound, and will operate for up to 11 hours when fitted with two fully charged batteries. There’s a four-channel mixer built in, with the first two channels accepting mic and line-level signals on XLR/TRS combi sockets and offering a two-band EQ and reverb effect; the third channel boasts an instrument

HK Audio Move 8

www.soundonsound.com / April 2025

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SPOTLIGHT

B AT T E R Y- P O W E R E D PA S P E A K E R S

input with switchable Celestion cabinet modelling, EQ and reverb, and the fourth is equipped with 3.5mm TRS and Bluetooth connections. A free Move 8 Remote mobile app not only provides remote control over the system, but also unlocks some additional features including a two-band EQ for the aux/Bluetooth channel, cabinet simulation and effects selections for the instrument channel, and a built-in karaoke function. $ $849.

W www.hkaudio.com

JBL Eon One Compact & Eon One Mk2 JBL’s Eon One Compact speaker is small and light enough to carry with one hand, but still promises to deliver a sound big enough JBL Eon One Mk2 to fill a room. It’ll reach 112dB SPL, and despite its tiny footprint, boasts a low-frequency response that extends down to 37.5Hz. There are four channels on offer: the first two accept mic and line-level sources on XLR/TRS combi sockets, the third is equipped with a high‑impedance instrument input, and the fourth accommodates wired and wireless playback via 3.5mm TRS and Bluetooth connections. A quarter-inch TRS socket allows a signal to be passed on to additional units, and there’s also a headphone output and a pair of USB

ports for charging mobile devices. The JBL Compact Connect mobile app offers remote control over the likes of level, EQ, effects and preset management, as well as allowing audio to be streamed directly to up to four Eon One speakers. As for battery life, the system is capable of running for up to 12 hours, its batteries can be swapped out in seconds without the need for any tools, and it can be quick-charged in 2.5 hours — you can also charge the system while its being used. If you need something with more power to hand, then JBL’s Eon One Mk2 compact line array can also be run on battery power, and comes equipped with a built-in five-channel mixer with three mic/line-level inputs, two dedicated line-level channels and support for Bluetooth streaming. $ EON ONE Compact $549,

the ranges, but all are kitted out with at least one mic/line-level channel and a dedicated input for music playback devices; many also feature an instrument input, along with link outputs that can be used to combine multiple units. $ From $399.99.

W www.ld-systems.com

EON ONE MK2 $1299.

W www.soundonsound.com/reviews/ jbl-eon-one-compact

W www.jblpro.com

LD Systems Road, Maui & Anny LD Systems have quite the range of battery-powered systems, many of which can be purchased as bundles that combine them with the company’s various wireless microphone systems.

There are three ranges to choose from: the Road, Maui and Anny Series, each of which is available in a variety of sizes and configurations. Both the Road Series and Anny Series feature traditional cabinet designs fitted with eight- or 10-inch drivers and a one-inch tweeter, while the Maui takes the form of a compact line array that combines an eight-inch subwoofer with a detachable column loaded with three-inch drivers. The amount of I/O varies across

Mackie ShowBox

Mackie Thump Go, Thrash212 Go & ShowBox Mackie are one of the best-known names in the small-format PA world, so it’s no surprise to see a range of battery-powered options in their current line-up. First up is the Thump Go, an eight-inch speaker with 200W of power, an onboard two-channel mixer and a range of voicing modes. Both channels accept mic and line-level signals on XLR/TRS combi sockets, and the second also offers a 3.5mm TRS connection and wireless streaming over Bluetooth.

The Thrash212 pushes the power up to 300W, pairing a 12-inch driver with a one-inch tweeter;

LD Systems Maui 5 Go JBL Eon One Compact

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SPOTLIGHT

B AT T E R Y- P O W E R E D PA S P E A K E R S

Mackie ShowBox Controller

it offers the same connectivity as the Thump Go, but without the voicing modes. The ShowBox steps things up a gear with a built-in six-channel mixer that will accept a mixture of mic, instrument and line-level signals along with Bluetooth streaming, with each channel benefitting from three-band EQ and one-knob compression. There’s a collection of onboard effects aimed at vocals and guitars, along with rear-panel send and return points for hooking up external processors, and the unit even includes a built-in looper and tuner.

Helpfully, the ShowBox’s control panel can be removed and mounted to a mic stand, placing all of its key controls within easy reach during a performance. $ Thump Go $449.99, Thrash212 Go $499.99, ShowBox $799.99.

W www.mackie.com

Mackie Thump Go

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Positive Grid Spark Live

Positive Grid Spark Live Building on the success of their popular Spark amps, Positive Grid’s Spark Live offers an all-in-one PA solution that’s capable of handling the likes of guitars, basses, vocals and keyboards. The first of its four channels is dedicated to guitars, with a row of front-panel controls providing amp-style gain and EQ, while the second channel is equipped with an XLR mic input. Both benefit from a selection of eight customisable presets. A pair of inputs for connecting stereo line-level sources follows, with wireless connections supported thanks to Bluetooth connectivity; a pair of line-level outputs allows a stereo signal to be passed to another device. The company offer a choice of two optional wireless foot controllers that allow users to toggle

Boss Cube Street II

effects and switch between presets, and there’s also MIDI I/O on five-pin DIN sockets for connecting third-party controllers. Given the unit’s heritage, it’s no surprise to see a wealth of modelling capabilities built in, and connecting to the Spark Live’s companion mobile app offers access to 33 amp models and 43 effects, in addition to in-depth control of the device’s mixer functions. The system can be placed horizontally or vertically, as well as upright or tilted, and is fitted with a sensor that detects the orientation and can automatically apply the appropriate corrective EQ. With its optional battery pack fitted, the Spark Live will operate for up to eight hours. $ $549; battery $79.

W www.soundonsound.com/reviews/ positive-grid-spark-live

W www.positivegrid.com

Roland Cube Street EX

Roland Cube Street II / EX

Yamaha Stagepas 200BTR

Roland’s Cube series has long offered a versatile selection of portable amps and PA speakers, and the two current Street variants will both run on eight AA batteries.

Branded under the Boss name, the Cube Street II features dual 6.5-inch drivers with 10W of power, and is capable of operating in stereo. Its first channel will accept mic or instrument signals, and includes a three-band EQ along with reverb and an automatic harmony function with three operation modes: unison, high and higher. The second channel will also accept mic and instrument signals, but focuses on guitar, offering a three-band EQ, chorus, delay, reverb and a set of nine amp simulations. The larger Cube Street EX opts for a pair of eight-inch drivers and two-inch tweeters, ups the power to 50W — there are selectable 25 and 10W options intended to extend battery life — and adds Audio In and Line In channels, with the former offering a 3.5mm TRS connection and the latter sporting a pair of quarter-inch TRS sockets. Both models integrate with Roland’s i-CUBE Link and CUBE JAM iPhone/iPad apps, which provide a range of rehearsal, jamming and recording features. The Cube Street II also benefits from a CUBE Street II Editor app, but requires an optional Bluetooth adaptor for iPhone/ iPad integration. $ Cube Street II $349, Cube Street EX $679.

W www.boss.info

Yamaha’s Stagepas 200BTR packs some of the technology behind their high-end audio gear into a small, portable unit that can run for up to 10 hours without any mains power. There’s 180W of power on offer, with the compact cabinet housing an eight-inch driver and 1.4-inch tweeter that deliver a maximum SPL of 125dB and reproduce frequencies as low as 60Hz. An onboard five-channel digital mixer features three mic/line-level inputs on XLR/TRS combo sockets — channels two and three will also accept Hi-Z instrument signals — while channels four and five sport a quarter-inch TRS socket dedicated Yamaha Stagepas 200BTR to stereo (or summed mono) line-level signals, processor and control over the unit’s along with wireless streaming via reverb type and time settings. The Bluetooth. There’s also a link output Stagepas 200BTR is supplied with that can be used to feed a monitor or a stand that allows it to be positioned second PA speaker. Top-panel level at a 30- or 60-degree angle (it can and EQ preset controls are available also be mounted on a speaker stand), across the board, while the mic/line and Yamaha offer a dedicated carrying channels benefit from built-in reverb, case as an optional extra. delay and chorus effects; a Main $ $879.98. output section then provides Speech, W www.soundonsound.com/reviews/ Music and Club operating modes, yamaha-stagepas-200 W www.yamaha.com along with a Feedback Suppressor www.soundonsound.com / April 2025

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INSIDE TRACK

Koreless

FKA Twigs has worked with many big-name producers, but it was experimental electronic artist Koreless who guided the making of her new hit album Eusexua.

PAUL TINGEN

“T

he fact that FKA Twigs’ last album went to number three in the UK is huge for me,” says Koreless, aka Lewis Roberts. “I tend to think that I make comparatively obscure

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electronic music, so it is nice to see it resonate with so many people. I don’t know if I also speak for Twigs, but we did not try to make a hit album. My feeling is that if you try to do that, it never works.

The best thing you can do is just make the best music you can. And people love

Twigs for being Twigs. Just doing what you think is right is always a good place to be.”

Eusexua is the third and most successful full-length album in a career that has seen FKA Twigs, aka Tahliah Barnett, collaborating with artists like A$AP Rocky, Kanye West, Skrillex, Jorja Smith, Fred again.., the Weeknd, the 1975, and more. She has also co-produced and co-written with big names such as Emile Haynie, Paul Epworth, Jack

Eusexua is FKA Twigs’ third full-length album.

Antonoff, Sounwave, Cirkut, Mike Dean and Benny Blanco, alongside more obscure producers. Eusexua continues this trend, involving both heavyweight pop producers like Jeff Bhasker, Stuart Price, Ojivolta and Stargate and relative unknowns such as AOD, Tic, Eartheater, Felix Joseph and Koreless.

Photo: Eloise Parryfree

Youth Policy Koreless was Twigs’ closest collaborator during the making of the album, much of which consists of abstract, futuristic electronic soundscapes. Both artists are signed to the same label, Young (formerly Young Turks), who initially brought them together. “I was brought in to work on her second album, Magdalene [2019], initially just to do some vocal effects. We ended up getting on quite well and we made a few songs together for the album. From there on we have been working together more and more, including on her mixtape Caprisongs [2022]. For Eusexua we spent a lot of time together.”

Roberts and Twigs are the only constants in the credits for every track on Eusexua, with Roberts the featured artist and sole producer on one, ‘Drums Of Death’.

“Much of Eusexua came into being through writing camps. We did one in Ibiza, one in Big Sur in California, and some smaller ones in London. We’d invite a really mixed bag of collaborators, from really established songwriters to other artists. For example, Twigs found this guy on Instagram called Matteo Santini [@xquisite_korpse] who is a wizard on the Elektron Machines. We might do a long improvised jam with him and Twigs and maybe someone else from a totally different world, and we might cut a few songs out of that. We’d just write all day, really fast. We were working really, really

fast in general. During the songwriting camps there’d be five or six ideas a day, for a week or two.

“I think the concentrated writing camps were a new process for Twigs. It certainly was a new process for me. If we just sat down together to write songs it could have become quite predictable, and we wanted to get out of that. She always wants to stir the pot and try interesting things. Writing camps are a good way of doing that. There’s definitely something great about two people sitting down and writing a song, and Twigs and I have written many songs like that. But many of the best moments on Eusexua came from crazy songwriting camps, with three or four people having live jams.”

Country Life Roberts develops his ideas from his studio in rural Wales. “I’m always working here in Wales, always making stuff, not always full songs, just elements that I’m cataloguing and reassembling later. I like working on just one small sound or loop quite intensely before starting to think about turning it into a song. Sometimes giving

these small elements attention means that they need less support in the context of a full arrangement. I have a big folder of these small elements, to be used as starting points, either on my own records or with other people.

“My studio is in an annexe to the house. It’s a bit cobbled together, and it’s not the best room, to be honest. I like a really dead room, with monitors very close. I like to feel like I’m inside a pair of headphones. The walls are heavily treated acoustically, so it sounds good to me in here now. My monitors are ATC SCM 45As. I feel they’re too big for me, but they’re cool. I was getting a lot of weird ringy stuff bouncing off the back wall, so I tried placing them on window ledges and boxing them off into a makeshift pseudo-soffit to stop the back wall reflections and that seems to work.

They’re not properly soffit-mounted for speaker boundary stuff, but the response is flatter and the stereo image is cleaner.

They feel closer and more like a pair of headphones. I just got the Antelope Orion 32 as an interface. I like that it has lots of inputs, and it sounds good. I‘ve also

Lewis Roberts’ studio in rural Wales is set up to give him the feeling of listening on a gigantic pair of headphones.

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recently got a few racks of outboard, an old AMX 1580 pitch-shifter and delay that I absolutely love, the Alan Smart C2 compressor, some Thermionic Culture units, like the Rooster tube mic pre/EQ, and more.”

After trying his hands at Ableton in his early years, Roberts abandoned the DAW, only to reconnect with it later on. “I left Ableton for a while because it didn’t do a lot of the things I wanted it to do. I had a brief detour using Reaper, and then moved to Pro Tools. Ableton has rectified many of the issues I had with it in more recent updates, so I’m using it again. And I still use Pro Tools.

“I like writing in Ableton because of Max/MSP and the MIDI/MPE editor. I’m pretty fast with it now. I often have really long crazy effect chains, and smash everything together through compressors.

My sessions can get really chaotic and busy, so I like to combine things in blocks, turn them into WAV files, and then transfer

At the ideas stage, Lewis Roberts works mainly in Ableton Live.

This screen shows the arrangement view for ‘Drums Of Death’.

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them to Pro Tools, where I chop them up. It’s mentally easier to deal with than thinking about a sound as consisting of hundreds of layers. In Pro Tools I try to think of it as just one layer, and treat it like a sample. I also like to record audio in Pro Tools, and finishing tracks in Pro Tools feels a little more professional. I seem to be able to make more sensible mix decisions in Pro Tools than in Ableton.”

Building Sounds “In general, my process is to start with sounds,” says Roberts. “As soon as I have something interesting, I go from there.

Once I have a sound I really like, and also often a sequence, the whole idea takes shape quite quickly. So I spend a lot of time programming. Too much time!

I’m programming patches in different instruments, and in Max, but I found that I was spending so much time with Max that I’ve banned myself from using that at the moment!”

Roberts’ ideas were the foundation of three of the songs that appear on Eusexua: ‘Room Of Fools’, ‘Drums Of Death’ and ‘24hr Dog’. “When Twigs and I were finishing the album, we felt like we were missing something obnoxious and hard. I was flicking through a big folder of music that I’ve made, and came across ‘Drums Of Death’, and Twigs really liked it. I’d kind of forgotten about it really. We just cut it on the day. An artist called Tintin Freeman came in and helped write some of the poetry stuff that’s in there.

“My original demo was based on a sample of a vocal by G-Dragon, who is a huge South Korean rapper. I sampled that randomly. I don’t know why. I guess the sonic quality of the voice was interesting. Normally when you’re cutting an audio file, you try and do a nice crossfade and make it nice and smooth.

But with this, it was about trying to make all those pops and clicks as loud as possible. So I was cutting the waveforms

at their highest peaks, and then adding only the slightest of crossfades, so that the vocal had a huge amount of transients to it. In context of that song, all of those pops and clicks sound like percussion. They all got smashed together through some saturation, which glued everything together. Chopping up the vocals at the highest points of the waveform is very ‘101 of things you shouldn’t do’!

“We kept some of that sample, and then we redid a lot of it with Twigs’ vocal. So we sat and re-recorded every syllable 20 to 30 times just to get exactly the right diction.

We did the chops on the grid in Ableton’s arrangement window, so we did not use a sampler or anything. For the rest, there are no synths in that demo, just a drum sample going through the Waves CLA 76, with 25dB of compression. It’s absolutely slammed, the needle barely moves out of its maximum position. That was it. It’s really simple. Just a lot of editing by hand. “‘24hr Dog’ was another instrumental idea that I had that Twigs really liked. When I played it for her, she immediately started singing on top of it, and then we were like, ‘OK, we can use that.’ Twigs of course in general comes up with many of the vocal and lyric ideas. We have some posh mics to record her final vocals, but actually, we often use the Shure SM7B, because it adds quite a lot of

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low end to her voice, which is cool as her voice is quite high. The mic adds some nice depth, and it’s really good for writing, because you can use it in a room without it feeding back badly. We end up keeping a lot of the SM7B recordings. “‘Room Of Fools’ took a long time.

Twigs and I really changed that song a lot in the studio, adding all the pianos and so on. I had just the seed of the idea. We spent ages trying different things at her studio, which is very well equipped with ATCs and so on. It’s where we home in on stuff for long, long days. Marius [de Vries] added six bars of live orchestra at the end of the song. He was in Berlin recording something with a string section, and while we Facetimed he said, ‘OK, let’s just record it now.’ Marius also added the tablas.”

Party Island As Roberts explained earlier, most of the tracks on Eusexua grew out of writing camps. “The album’s title track was a classic one. It was started in Ibiza. We were in the north of the island, at a yuppie hotel with an atmosphere similar to in

White Lotus, the TV show. The hotel has a really well-equipped studio in its basement, with big Quested monitors and so on.

“We brought in different people for a few days. Eartheater came to Ibiza to join us and played some amazing guitar chords and vocal ideas. We wrote ‘Eusexua’ together as a live jam, the result of loads of people throwing ideas down.

That was a really difficult song to make.

On the surface, it doesn’t sound like there’s too much going on, but there are many layers to it. Twigs and I were trying to balance two different worlds. We were looking for real toughness, but we still wanted it to be open and soft and quite ambient. We were looking for the drums to feel really aggressive and hard, but also like they were far away and throbbing and pulsing, and not an immediate threat.

The drums needed to feel like something further, further away, imposing, but not too loud and harsh. “‘Girl Feels Good’ was another one that started out as a live jam, during a writing camp, in London. AOD was playing the guitar chords, I had the acid-like loop,

One advantage of living in the middle of nowhere is that there are no neighbours to annoy!

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and Ojivolta did the drums. This was all happening on different computers, going into our big Pro Tools rig, and then Twigs got the vocal down. Often we were still flicking through basic ideas on people’s computers, and then all of a sudden Twigs is signalling to have the mic. And then, before we had a chance to set up properly, we’re in it! We may not even have the sessions running properly yet, and there are already ideas flying around.

“Sometimes the lyrics would just be gobbledygook, and we’d have just a really rough idea of a song with some promise, but the feeling would be there. Sometimes we had something fully written, but the music was bad. There were many different things, and we’d mash them together and take a little bit from here and a little bit from there and put it all together into a larger whole. The writing camps were really fast and furious and chaotic, with ideas flowing everywhere. It was a bit crazy, but also very productive and fun!”

To keep track of the sessions, FKA Twigs and Roberts relied a lot on engineer Jonny Leslie. “He is also a great producer.

He was there at the writing camps, often on the main computer, recording everything. Other producers would AirDrop stereo tracks to that computer.

Sometimes we’d track into Pro Tools, sometimes into Ableton. Jonny was a big part of keeping track of everything and making sure things were well organised.

This was quite a mammoth task at times.

“We tried to get about 80 percent of the song finished, and in between and after the songwriting camps, Twigs and I would come back to her studio, where we had the time and freedom to try out what worked and what didn’t.

We really carefully put things together and rearranged them and changed the balance of things, and so on. She loves the Dave Smith Tempest drum machine.

She’s really good at that, and she generates a lot of cool sounds on that that we added to the mix. So we’re just sitting and piecing everything together.

“We spent a long time at her studio chipping away at and rearranging the songs. Throughout this whole process, we’d have people come in to do something specific. If we knew someone good at a certain type of drums, they would come in to do that. Or they might work an arrangement thing, or do some additional processing. We worked a long time and had several people come and do different things. Sometimes we’d

Digital Rules As an electronic musician, Lewis Roberts’ work often involves synthesizers, but he has some unusual perspectives. “I tend to prefer early digital synths, because they have more interesting mechanics and they’re really flexible. The Nord Modular is one of my favourite synths, it’s from 1997. I love all the old Nord stuff. I also got a UDO Audio Super 6 recently, which is cool. And I really like the Roland SH-101. I use mostly these synths at the moment.

“The old 101 is smooth and silky, and predictable and tidy, but many of the big old analogue synths that people love are not for me. They’re too unpredictable and boisterous. People love analogue synths because they have that bit of ‘oomph’, but you can get that back with digital synths if you just stick them through a nice preamp. I use the Thermionic Rooster for this. I run my digital synths through that, and I re-amp soft synths through the Rooster as well. There’s something about the aesthetic of tidy, clean digital synths being driven gently afterwards that I really love. It’s very precise and feels like clockwork.

“The soft synths I have are not particularly interesting. The simpler the better, for me. With some of the soft synths I find there’s too much going on. I’ve also made some synths in Max/MSP that I really like.

But instead of using synths, I like to work with bare waveforms a lot of the time. I’ll record just one long note and treat it in the Ableton arrangement view. I may move it and resample it and chop and/or reverse it. You can have better control over the envelope of notes and so on. The arrangement window is the most powerful tool of them all. Instead of having one synth, I like to put that one note on several different tracks that each have different effects. I think that’s more interesting.

“For effects I like spectral stuff, for example from the GRM plug-ins.

They make spectral effects that sound great. I use a collection of old plug-ins that I’ve been gathering from the early 2000s, that still just about work on my new Mac, for example some of the Michael Norris stuff. He made some interesting plug-ins. It concerns me that a lot of my favourite plug-ins and gear are from a long time ago. Like my Nord Modular, which is software-controlled. The software still just about works, but it’s a bit janky and I’m worried about the future.

“When I’m on a keyboard, I just noodle around. I’m not a particularly great keyboard player, and when I do use a keyboard, I often gravitate towards the same chords and ideas. One of the reasons I like the Nord Modular is that I can program something semi-random, and it will generate something I might not think of. I may start with a musical idea, and use the Nord to generate sequences based on a chord or something, and then I’ll pick the best bits of that. I like to add elements of chaos!”

send a whole song to someone and they’d do a different version, and we’d take some bits of that and put it back into the previous version. We were trying everything.”

Searching Trying everything also involved trips to LA to work with the aforementioned Marius de Vries, an eminence grise of the electronic pop world, who has worked with Björk, Madonna, Massive Attack, David Bowie and many more. “Twigs and I felt that there was something missing on the album, and we thought Marius might have the answer, and he did.

He is responsible for many of the very beautiful organic live-sounding parts on the record. He did some amazing string arrangements. We first met him in LA, and

The left of the two equipment racks at Koreless’ studio houses his synths (top to bottom): Behringer Model D, Miditech MIDI interface, Oberheim Matrix 1000, Roland D‑550 and Nord Modular. The right-hand rack contains processors and effects: Alesis Quadraverb reverb, Thermionic Culture Vulture valve processor, AMS DMX 15-80 delay, Smart C2 and Aphex 661 compressors, Thermionic Rooster preamp and Roland SBF-325 flanger.

then he came to London for a bit, and later we went to LA again to meet him.

He has a co-production credit on ‘24hr Dog’ because he took many things out of the arrangement, which turned out to be genius. It made the song much better.

“At the end of this production process, Twigs and I would be rough mixing, and getting the sessions ready to send to the two mixers, Manny Marroquin and Jon Castelli. Both of the mixers did a really great job of making sense of quite dense productions. The arrangements are not always straightforward and often quite abstract, and they tidied everything up and made everything quite clear and so on.

“Working in a computer is not like recording on a multitrack. A lot of what I do is processing and stuff. I wasn’t thinking, ‘Oh, someone’s going to mix

this at the end, so I can just leave it.’ I was trying to get it as close as possible. But you don’t want to do too much, because you want to leave some room for the mixers, especially with compression and EQ. I like to give them more options.

They’re going to do a better job than me, with fresh ears and better skills. So I like to keep things alive and try not to control everything too much in the production process.”

When the question “What’s next?” is to put to Roberts, it turns out that he approaches his career with a similar attitude of not trying to control things too much. With a shrug of his shoulders, Roberts says, “I’m not even sure how my week is going to pan out! Doing the Twigs stuff was a surprise to me. So I’m open to more surprises!” www.soundonsound.com / April 2025

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Ela Minus WILLIAM STOKES

C

olombian-born Ela Minus was studying jazz performance as a drummer at Berklee College of Music when she got a taste for electronic music. “I loved synths,” the artist and producer says. “I saw there was another major called Music Synthesis. But I never really wanted to produce, to be honest, but I loved synthesis, so I did a double major.” She subsequently landed a job building synthesizers with Critter & Guitari, where she contributed to the design of the celebrated Organelle synth.

“I was always playing drums in parallel, and it got to a stage where I had sounds in my head — the music that I wanted to make — but I couldn’t communicate it to my bandmates,” she continues. “And I realised that learning how to produce was probably going to be easier than trying to explain what I wanted to other people. I just taught myself to do it, really.”

Fast‑forward to the present, and latest album DÍA has drawn critical acclaim from all sides.

At the moment I can’t stop listening to One would be Nala Sinephro, Endlessness. But also I can’t stop listening to Nicolas Jaar Piedras 1 and Piedras 2, obsessively. Both of them. I’ve been a fan of Nicolas Jaar for forever. I would say he’s probably the only contemporary musician that I admire in every aspect: the production, the musicality, the way he’s dealt with his career... When you see him, you can tell he stands for something, and I really admire that. I guess I look to that as a reference. The combination of these two records, which he released together, one being very experimental, kind of sound candy, and the other one being kind of a pop record, really, in a lo-fi way. The songs pretty much have a pop structure, they have choruses, and it’s so interesting for me.

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The artist I’d most like to collaborate with For this one I’ll say Nala Sinephro! You know, I love jazz and I love jazz musicians.

I think the combination of jazz musicians and electronic producers is an incredibly rich and prolific source of music; obviously a lot of people have already tapped into it, like Floating Points, but there’s still so much more we can do. Nala Sinephro’s musicality, the way she mixes acoustic instruments with synthesizers — that’s something that I really love, which I would love to do more of.

The first thing I look for in a studio Again, I’m going to answer with two things. The first one is the vibe. Like, if it’s got really bright lights, like you’re in like in a hospital, I’ll go get a candle. Or if it’s messy, it just makes me want to organise it! It just doesn’t make me want to make music. It makes me think, “This needs to be fixed!” That stresses me out. So first of all, it just needs to be an inspiring environment.

And the second, obviously, is the synths. Before this record, I would never really go to studios, because I had my own home studio. I just didn’t have the need, before the pandemic. And then in the pandemic I had to leave my home.

This new record I made completely in rented studios. I think the one thing I instantly looked for, even in the gear list before I went in, is which synths they have. The weirder, the better, obviously! Eurorack systems are very cool because they’re so unique — even if they have certain modules that are quite mainstream, the way someone chooses to build it will always be different.

One studio I have been working in has an ARP 2600 that was very old, as they all are. But this one they had clearly fixed, over and over. So it had a very unique sound to it. It was very messy, but it worked. That studio actually had a couple of synths I think the owner had fixed

himself. He also had a [Moog] Voyager that was very... uniquely fixed! It had real character.

The person I would consider my mentor Well, I think my drum teacher. Her name is Terri Lyne Carrington. She deserves all the shout-outs! She really changed my life. She came into my life at Berklee, really soon after I started, like, in my second semester. I was the only female drummer in my year. So to have a teacher who was also female, who could talk to me about it — I’d never met anybody who could talk about it with so much experience. She really taught me through drums. It expanded through to my life and my production and to everything I do. She taught me that while I needed to practise the instrument with a lot of discipline, the only reason to do that was to get to know my own voice as a musician as deeply as I could. That all that mattered was that I was able to connect to that, to something that we each have that nobody else has, and to be able to express that through a certain skill.

I had four years of her teaching me that.

She’s an incredible drummer. I do think she really changed my life.

My go-to reference track or album I think it depends on what I’m working on. I have a playlist of mix references, of master references, songwriting references. And it really is kind of all over the place! I can remember a couple off the top of my head, like for my song ‘Broken’, the reference was a song from the album Crash by Charlie XCX. It’s really poppy, but it sounds amazing. I really like that record.

Usually when I’m about to mix, that’s when I’m looking for references. I don’t want to sound like an arsehole, but the way I tried to mix this record, I couldn’t think of any references, and that was kind of the point.

I worked with Marta Salogni, who is the only person I mix with. We spent a lot of

planned for the Hollywood Bowl, so Shabaka played his parts, and even the live mixing of that was incredible!

I travelled to go to that because I love that record, and I knew it was going to be something special. And it was. Probably my favourite show of the past few years.

So, to witness that in the studio...

The producer I’d most like to work with El Guincho. He’s produced a lot of FKA Twigs and Björk stuff. He’s one of those producers who I think, again, clearly has a pop sensibility, but with unique production. He has a solo record maybe from 2010, called Pop Negro, which is like a cult favourite, I guess. It’s great.

I think he did another solo record, but he’s moved more into production. He produced ROSALÍA, El Mal Querer. Like, really big pop. But I do feel like a lot of what makes it really interesting is his production. I mean, that’s what makes it into interesting pop. His rhythms feel very fluid, the way he produces beats.

It feels like it’s coming from humans!

As if you were in a in a carnival in Brazil, or something. But it’s really electronic.

It has a very unique character.

time mixing, and the entire concept was: “This cannot sound like anything else we’ve ever heard!”

My secret weapon in the studio is It’ll change, but right now it’s a bit-crusher that exists both in hardware and as a plug-in. It’s called Biscuit, by Oto Machines. I’m obsessed, it’s my secret weapon. I use it a lot, and every time I use it, everybody is like, “What the fuck is that?” I’m very interested in distortion and bit-crushing, any way of deconstructing sound. Pushing sound to extremes, I guess. I was investigating other ways to distort. And obviously bit-crushing is a big one, but I was interested in a physical thing that wasn’t necessarily digital. And I could only find this one. I actually have the serial number #1! I bought it just as they were starting to make them. I’m very happy. It has eight buttons for each bit, and you can, you know, leave it clean, or invert the bit, or mute it. The inversion really fucks up the sound, very extremely.

But because you have eight of them, you can do things like invert one but mute another one. And it really makes interesting distortion. It has other modes too: you can double the waveform, like, add a sawtooth wave to whatever your sound source is and tune that. It’s really, really interesting. And they have a really good plug-in now that honestly works equally well. I was very surprised with the plug-in.

The studio session I wish I’d witnessed Floating Points, Promises. That’s the first thing I think of. Pharoah Sanders was such an incredible player. The album sounds so magical: the arrangements, the way they recorded it, how his saxophone sounds, how the symphony sounds, the mixing of the synths, everything.

Everything about that record sounds so unique and special to me. I was lucky to see it live — without Pharoah but with Shabaka [Hutchings]. When [Sanders] died, the show was already

The studio experience that taught me the most Just one? I’ll say: the process of making this entire last record. It really is the most I’ve worked in other, external studios, so the entire thing was a huge lesson for me. I was a producer who was attached to my hardware. And I had to go into all of these other studios with nothing but my laptop, and work from scratch with different hardware, with different systems, with different interfaces. And I was able to make a record out of it. The entire thing was a very big lesson.

The advice I’d give myself of 10 years ago I think, just dive in faster. With producing, I didn’t really think about it as a serious thing. I was like, “I’m a drummer and I have a job building synths! I’m just going to produce a couple of songs to, like, quiet the urge of this music I have in my head, but I’m not going to, you know, be a producer or do a solo project.”

I didn’t take it seriously at all. So my advice now would be: take it a bit more seriously.

Spend more time, do it faster. It took me a while to think, “Oh, actually, I think I can do this.” It took me years to become like that. So just spend more time doing it, from the get-go. www.soundonsound.com / April 2025

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FE ATURE

Bernard MacMahon & Nick Bergh: Becoming Led Zeppelin A hit movie about the world’s greatest rock band turns conventional ideas of audio restoration on their head.

SAM INGLIS

“I

n the early stages of making the film, we met with extraordinary resistance to the concept of playing whole songs,” says Bernard MacMahon. “People were just like, ‘No‑one’s going to be interested in that!’

But if you think that the story of this band is worth you sitting in a theatre for two hours, then I believe you’ve goddamn got to play a whole lot of that music. And if that isn’t a total thrill for the audience, why are you even doing this band in the first place?”

Not only does Becoming Led Zeppelin include full songs, but the team behind the movie also flew in the face of industry convention in their approach to sourcing and preparing this music.

Director MacMahon, sound supervisor Nick Bergh, writer, producer and music supervisor Allison McGourty and editor Dan Gitlin had the full cooperation of the band and Atlantic Records. They could therefore have accessed the master tapes, or any of the numerous digital remasters that have been released over the years. Instead, they used the original vinyl records.

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“We made a very conscious choice to go with the discs,” says MacMahon, “and the reason is that the tape is often so different from what the public heard.

Even if the master tapes survived of all the studio recordings, which a number don’t, they were always made with the mind that they were going to be significantly altered in bringing them to the public on a piece of plastic. The production master was altered from the quarter-inch that was made in the studio, then that production master was enormously goosed at the cutting stage by a Bob Ludwig.

“What the public heard on their record players is not what you would have heard played back in the studio.

So the truest thing is for you to hear the ultimate thing that millions of people bought, in the truest possible way.

That’s the philosophy of the film.”

Pressing Matters The first step was, therefore, to obtain the best possible vinyl copies of everything.

This was not as simple as it might appear, given the complex nature of record manufacture in the ’60s and ’70s. Taking Led Zeppelin’s 1969 debut album as an

example, “There were many different early pressings of Led Zeppelin I that differ widely in sound quality,” says Nick. “This wide disparity seemed to be largely due to record azimuth errors in the original tape and the various ways mastering engineers were correcting for it, or not correcting for it.”

“Atlantic were using four pressing plants,” continues Bernard, “and on album one, each of those plants had different lacquer cuts made. And the white label test pressings that are very desirable by collectors, that’s another cut as well. And each cutting engineer that’s making the lacquers for those different plants is maybe doing two or three or four lacquers. And each of those might have minor differences, as anything would in the real world. Each of those is different. On the RCA pressing, the channels are the wrong way around! The fans generally seem to think that’s a different mix.

It’s not, it just obviously doesn’t sound the same as what they’re familiar with.

“So what was amazing with Led Zeppelin I is that the pressings are radically different in terms of sound and quality. And the best was from George Piros, a classical cutting engineer who came from Mercury [Records].

He didn’t like rock music at all, but he was

Photo: Allison McGourty

Director Bernard MacMahon (left) and sound supervisor Nick Bergh (right) review Nick’s transfer of Led Zeppelin I at Endpoint Audio Labs.

essentially just getting what was on that tape on the record, and that’s why his is the best.”

So different are the various pressings, in fact, that MacMahon believes this was a factor in the dire press reviews of Zeppelin’s debut: the white label copies that were sent to journalists were from one of the worst cuts.

was a problem with super-loud records: they would skip because the needle on a bad hi-fi would be jumping so much it would bounce out of the grooves.

“So that record was recut. The first 200,000 or whatever copies were this

amazing version, and then everything else that was sold subsequently was this much more dialled-back master with a much less dramatic cut. When you hear the Ludwig cut, the music has this unbelievable excitement to it. It sounds so good that

The First Cut Is The Loudest

Photo: Ron Raffaelli

The film concludes with the triumphant release of Zeppelin’s second album, a disc which threw up different but equally confusing choices for the team. “When Led Zeppelin II was completed, it was sent to Bob Ludwig to cut,” reports MacMahon.

“Unlike on album one, Ludwig cut all the lacquers for North America. And that record sounds absolutely amazing, and it’s incredibly loud.

“And, as you know, volume on a record is created by the needle moving rapidly and dramatically in the groove — but the apocryphal story is that [Atlantic Records co-founder and President] Ahmet Ertegun’s daughter had a white label pressing of the record and was playing it on a cheap kids’ turntable and the record was skipping. This

Eddie Kramer (left) and Jimmy Page mixing Led Zeppelin II. Rather than return to the masters or remasters, the film-makers chose Bob Ludwig’s original vinyl cut as the definitive representation of the album.

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people will now pay upwards of three and a half, four thousand dollars for pristine copies, even though it’s not technically a rare record. Unlike on the first album, all the media would have heard that version.

The white labels were all that version, to my knowledge. And that record had a seismic impact in America.”

Transfer Windows Once the best version of each release had been identified, the next challenge was to track down the best possible copy of that version: ideally, one that has never been played.

“Even a single play can instantly damage the record,” warns Nick.

“The louder the cut, the more the stylus is moving in the groove. If you played it with less than ideal equipment, especially back then, the stylus can’t move fast enough. So essentially it’s gouging the groove as it’s being thrown around in the groove.”

What’s more, there can still be problems even with unplayed copies of the best pressings. “Atlantic were

notorious for using shoddy material in their vinyl,” says Bernard, “putting a lot of filler and junk in. And so I was buying sealed copies that were expensive even before the market had exploded, in the hope that there was a Ludwig in there.

A couple of times, I got lucky, and this one happened to be an unusually quiet press — a nice puck, you know, with very little detritus in it.”

The results more than justify Bernard and Nick’s unusual approach. Even the older, pre-Zeppelin mono recordings in the movie

best restoration is no restoration”. The goal is to achieve the best possible transfer from the best possible version of the source, and then to leave it alone.

“I couldn’t imagine mixing a film like this without personally doing every single transfer, because that’s really where the sound gets locked in. That’s where the cake is being baked. Everything else is just kind of frosting.

“One of the biggest problems with mono material is low expectations, when really the mono material can sound incredible. The challenge is getting that incredible sound translated to a newer system.

It’s primarily a translation issue more than anything else, because if you play mono material on an original great mono system, it can still be incredibly compelling.

“When people hear archival audio these days, it is typically multiple generations away from the source, or it’s gone through processing, and so It’s often hard to explain that ‘This

Nick Bergh: “One of the biggest problems with mono material is low expectations, when really the mono material can sound incredible.

The challenge is getting that incredible sound translated to a newer system.” sound fantastic, and it’s tempting to imagine that lots of digital post-processing must have been involved. Not so.

“There is no traditional restoration processing,” says Nick. “There is no broadband denoise anywhere in this film.” His philosophy, rather, is “the

Photo: Allison McGourty

Left to right: editor Dan Gitlin, sound supervisor Nick Bergh and director Bernard MacMahon at Endpoint Audio Labs. The gravity-fed lathe restored by Nick and featured in American Epic is visible against the wall to the left, along with part of its associated rack of power amplifiers.

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Photo: Nick Bergh

A pristine copy of Led Zeppelin I on Nick Bergh’s EMT 927 turntable.

stuff can sound fantastic — you just haven’t heard it correctly yet.’ Trying to just get that sound onto the screen is always the challenge, because any time you put it through processing or mess with it, it’s gonna start to chip away at that magical sound. Broadband denoising is essentially like taking your sound and dipping it into a piranha tank!”

To The End The all-important transfer process took place at Nick’s remarkable Endpoint Audio Labs facility, a working museum of film sound. Amongst many other fascinating items, Endpoint hosts Nick’s restoration of the gravity-fed lathe that was used by the likes of Ralph Peer to cut classic blues records in the ’30s, and which featured prominently in the film project that originally brought MacMahon and Bergh together. American Epic unearthed vast amounts of previously unseen archive material relating to early blues, country and roots music, and was instrumental in persuading the surviving members of Led Zeppelin to place their faith in MacMahon.

“Even though those records are very old, they’re very demanding as far as fidelity goes,” insists Nick Bergh of the American Epic material.

“Because there is no limiter, there’s no negative feedback and there are very fast transients, those old blues records from the ’30s are very demanding to get everything out of them.”

Following painstaking cleaning, Bergh’s transfer process involves an EMT 927 turntable with an SME arm, and custom electronics

by Vladimir Lamb, feeding a Lavry Gold A-D converter. “It’s not just playing the record through all the way through,” he explains. “With a swinging tone arm, the arm is only right in two places on the record, and it is set so it is averaged across the whole side. For this film, because time doesn’t matter, I can adjust it perfectly for one track and then adjust it again for the next track. Small things like that add up to get a great sound out of it.

“Something we experienced in American Epic was that it’s amazing how delicate a great transfer is, and how much it doesn’t ever want to be touched after that. Any EQ, anything you touch it with, you hear it. Even if the processing is helping in some ways, it seems to yell out: ‘Hey, you did something to this!’”

Out Of The Archive In most cases, no digital post-processing was applied at all, not even basic equalisation. The audio from Led Zeppelin I and II did not touch a single plug-in except for the panner. Any surface noise was painstakingly removed by manual waveform editing rather than plug-ins. (This manual declicking was undertaken by Peter Henderson, perhaps better known as co-producer of Supertramp’s Breakfast In America.)

However, not all of the audio in Becoming Led Zeppelin originated on vinyl. The movie includes numerous filmed live performances by Zeppelin and other acts, from many sources. These covered an amazing range of audio media that one may not expect for rock & roll, such as 78rpm records, 16mm optical, 35mm optical, 16mm mag, 35mm mag, one-inch and two-inch video, 45rpm singles, lacquer discs, and various more typical tape formats. Once again, the key to achieving the best possible sound quality was to have Nick Bergh personally make the transfers from the original sources — but when the original sources were priceless archival artefacts, this introduced a whole new level of difficulty.

“From my point of view,” says Bernard, “the number one thing is getting the material into Endpoint with Nick, so he’s doing the transfer and we have the master ourselves. And the most difficult thing in doing this film was: how do you do that? Because it is normally against all archival policy to have masters travelling around the world to some facility 5000 miles away. This is normally just not done.

THE FUTURE OF GLOBAL RF

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Archive material was sourced and freshly transferred from an incredible variety of original formats. Clockwise from top left: 16mm negative of the 1969 film Supershow, with Zeppelin performing ‘Dazed And Confused’; the Danish two-inch PAL SECAM tape from which ‘Communication Breakdown’ was taken; the crucial quarter-inch tape of John Bonham and Robert Plant being interviewed for Australian radio, seen through a magnetic viewer at Endpoint Audio Labs; and 35mm reel from the groundbreaking 1964 concert film TAMI Show, shot in Bill Sargent’s pioneering high-definition Electronovision format. Photos: Michael Rice, Dan Gitlin and Nick Bergh

“And in the early edits of the film, we planned to have an evocation of the Hindenburg’s flight over to New Jersey and its explosion [as featured on the cover of Led Zeppelin I]. That is, obviously, one of the most iconic historical pieces of archive ever.

And we wanted to use some of Herbert Morrison’s legendary commentary. So Allison spent a long time dealing with the National Archives, and she persuaded the National Archives to bring out to Endpoint the original lacquers that Herbert Morrison cut in 1937.

“When Nick transferred them, the fidelity of the sound of his voice was shockingly improved. It was mind-bogglingly better.

The lacquer had been damaged where the airship had exploded, and it would normally have been unplayable, but Nick was able to navigate the disc, so for the first time ever

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you could actually hear the real explosion of the craft, instead of the sound effect that was used on the newsreel. You could also hear the screaming of the people on the ground, and coming out of the craft.

“One of our archive editors said, ‘This is the most powerful introduction I’ve ever seen to any movie.’ But because of how shocking this audio was, it was literally horrifying to watch and listen to. And you couldn’t begin a film about Led Zeppelin after two minutes of that — but that’s the power of sound.”

“That disc had only been played a few times, maybe three times or so,” says Nick. “But unfortunately it was in the era of very gnarly heavy pickups. And any playback on a lacquer severely damages it. A lacquer is essentially just thick paint on aluminium. So that’s a really interesting

example, where if no‑one had played that thing in the ’30s, the audio would be essentially contemporary-sounding. That’s the quality of that audio on the record. But unfortunately it was severely damaged when they even played it a couple times on NBC. The particular microphone and electronics they were using was really high fidelity, and it’s still very compelling.”

Detective Work Although the Herbert Morrison material ultimately proved too strong a meat for a music documentary, it helped MacMahon and McGourty persuade other archives to trust them. Crucially, this meant they were able to access a previously unheard radio interview with drummer John Bonham. Led Zeppelin never did many interviews, and even when they did, Bonham tended not to

Led Zeppelin: The Musical?

As Hollywood’s leading specialist in vintage film sound, Nick Bergh has been responsible for preserving many classic movies, including great musicals like Oklahoma! Unlikely as it sounds, this fed directly into the making of Becoming Led Zeppelin. For one thing, it inspired the team to go against modern convention by avoiding any processing that would homogenise the sound of dialogue and music, or of recordings from different eras.

Photo: Michael Rice

talk, so including his voice was one of the film-makers’ biggest challenges. Bernard MacMahon takes up the story. “Sam Rapallo, who runs a Led Zeppelin website, turned me on to an obscure bootleg with this pretty crummy-sounding multi-generational copy of an interview with Robert Plant and John Bonham. But when Nick and I listened to it, we could hear straight away this had originally been recorded to a quarter-inch tape. I could also tell it was a radio journalist, because he was not talking over them and he was giving them space. All we could hear on the bit that was on this bootleg was that it was an Australian journalist.

“Fortunately, the University of Canberra had done an American Epic festival, and I called the guy that had organised it, and he eventually identified the voice and what radio station it was. We went to the radio station and said ‘Do you have the original tape?’ And they looked for a while and they didn’t have it. And then I said, ‘Have you ever sent any of your tapes anywhere?’ He said, ‘Well, we have bequeathed some of our old tapes to an archive.’ So I called them back and they looked through the archive and said, ‘No, we don’t have it.’

“And then I remembered from American Epic that a lot of the big archives will often have pallets of recordings that they haven’t had time to catalogue. So I said, ‘Do you have any uncatalogued tapes?’ And they literally said ‘Thousands!’ And because we’d been helpful with American Epic and they were really nice, they started to look at it. Months later, I got a phone call near midnight out here in LA and it was this guy Sean from the library. ‘I think I’ve got it. Go to your computer.’ And he’d sent an MP3 and there it was, the tape.

“After that we got the transfer of the whole thing in 24/96, and Nick went, ‘We need to do this ourselves.’ So I called them back and I said, ‘National Archives... Herbert Morrison...’ And, incredibly, they sent it to us, and we were able to transfer it, and that’s what you hear.”

Cleaning the rarest disc featured in the movie, a 1968 lacquer recorded

by Robert Plant and John Bonham’s pre-Zeppelin group Band Of Joy.

“One of the things I like about keeping archival music so original is the recordings all sound a little different,” says Nick, “because they’re all done in different studios and with different gear.

As soon as we start putting the same EQ and the same limiter on these things, some of that goes away.

“One of the most striking things about working on the old classic musical films from the 1950 and ’60s was how much care was typically taken with music fidelity. The musical numbers were often like structural pillars that everything else built up against.

The aesthetic with musicals has changed a bit now, where the mics and fidelity are intentionally chosen so they flow better with the production dialogue scenes. They can still sound good this way, but it is not like Julie Andrews singing through a U47 with minor tube compression and essentially nothing else.”

The influence of classic musicals also extended into the structure of the film itself.

“The film is constructed like a musical,” says Bernard. “All the songs are sequenced and chosen in terms of what’s being sung about relating to what’s going on. The first time you hear ‘Communication Breakdown’, they’re playing to an unreceptive European audience that are putting their fingers in their ears. As they’re going off to try and get a deal in America with Atlantic, we’re playing, ‘Your Time Is Gonna Come’. And we’re using the chorus of that song as a narrative device. Every single song is used to advance the story. We’re using musical techniques that we learned from being in here and hearing these great musicals being played back.”

It’s a highly effective device, allowing the viewer to feel caught up in and swept

away by the band’s growing success.

“That’s a construct of the pieces of music and the order that they’re placed in, to create that effect that you are going further and further forward. You start with this very simple skiffle sound, and Little Richard in this unbelievably powerful mono, and you end up with this enormously expansive sound with the the whole drum kit panned all the way across the picture. You go on the sonic journey that the world took over those two decades.”

Just a few weeks after release, Becoming Led Zeppelin is already the most successful IMAX music movie of all time, to the point where demand has forced Sony Pictures to bring it back for a second IMAX run across America.

And fans, many of whom are viewing it for the third or fourth time, would agree wholeheartedly with MacMahon and Bergh that one factor above all is fundamental to its success: the sound.

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

track/4dtThTihlGhxcggGCAfhed

W www.youtube.com/

watch?v=jA4JDTSi6RI

Stefan Boman • The Hellacopters ‘No Song Unheard’ J O E M AT E R A

S

tefan Boman is the house mix engineer at Atlantis Studios in Stockholm, Sweden. He began his career at ABBA’s famed Polar Studios in 1995, and has worked with the likes of Def Leppard, Alice Cooper, Backstreet Boys, Roxette, Avicii, Opeth and Ghost to name but a few. Asked to choose a favourite sound from a session he’s worked on, he nominates ‘No Song Unheard’ by the Hellacopters.

Back In The Room “Prior to this album, the Hellacopters were kind of lo-fi and all about distortion and attitude. But with this song and album, we really wanted to turn down all the gain and focus more on the subtle things like the room sounds. So the way we worked

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on capturing the room sound helped shift the group’s sound from one that was lo-fi to one that was hi-fi.

“We recorded the track at ABBA’s old studios, Polar Studios in Stockholm. We had the drums set up in a really dry room that had thick carpet on the floors, and all the ceilings were completely dead, and right next to it was a big marble room. We left the doors open to the big marble room, which had a Neumann KM84 mounted in the ceiling which was about 4m high. That way, you could have a super-dry close‑miked kit, but those room mics had just about no direct sound and only reflections. It gave some ambience on the side that you could later choose in the mix.

“We had the guitars in an all-wooden room, with the walls lined with flat panels of pine wood so it wasn’t overly diffuse.

Due to the good reflection from the wood walls, and also how we arranged the room that the guitars were in, it brought a subtleness to it instead of just being overpowered with aggression. We used a couple of Neumann U47 FETs as room

microphones, which, with all the wood reflection, gave the sound a real classy and nice tone.

“The tom sound was very important too.

We ended up taking the bottom skins off on the toms and placed a Sennheiser MD 421 in the tom. We had to add a bit of EQ and took away the frequency around 350Hz in order to get rid of that ‘beach ball’ sound that you get from miking the inside of a tom, and it worked really well. The vocals were later overdubbed but were recorded in a completely dry room as we wanted to have more of a slap-back kind of delay on the vocals instead of any natural reverb.”

Tape Treatment “In the mix I added a bit of compression, and that was it, as we managed to nail the recording perfectly, so the mix was just fine. It was more about just doing a bit of tweaking and balancing, pretty much.

I have to add that the whole album was recorded on a Studer A820, which helped polish off some of the transients on the drums and add ‘oomph’.”

Stefan Boman: “The way we worked on capturing the room sound helped shift the group’s sound from one that was lo-fi to one that was hi-fi.”

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INTER VIE W

Noah Lennox (left) overdubs a vocal at Josh Dibb (right)’s Rabbit Hole Studio in Baltimore.

Noah Lennox & Josh Dibb: Recording Sinister Grift The new album by Noah Lennox, aka Panda Bear, features familiar collaborators pulling in a very different musical direction.

WILLIAM STOKES

A

merican-born, Portugal-based artist Noah Lennox has found enduring popularity in the world of electronic music, both as part of Animal Collective and as a solo artist under the name Panda Bear. New album Sinister Grift is his sixth under the latter name, and sees him backed by something surprisingly close to conventional rock arrangements.

And although it’s very much a Panda Bear record, Sinister Grift features contributions from the other three members of Animal Collective: Dave Portner (Avey Tare), Brian Weitz (Geologist) and Josh Dibb (Deakin), and was co-produced and mixed by veteran Panda Bear and Animal Collective collaborator Josh Dibb.

Dibb, it turns out, had been enlisted

by Lennox before there was even an album to be mixed. “During the pandemic we decided to record a record remotely, which became [2022’s] Time Skiffs,” he remembers. “Noah had been recording his

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own drums. Everybody probably does this, but I will often start to ‘pre-mix’ things, just to help them make sense to me. Noah was responding really well to what I’d done, and he had even suggested he’d be up for me mixing the record. We all talked about it and ultimately the band felt like that wasn’t the right move. But he said, ‘Well, I want you to do my next record.’ And I was like, ‘Dude, thank you. But you have no record right now!’ But, lo and behold: years later, he stuck to it.”

Unbeknownst to Dibb, an album was indeed taking shape. “I got an offer to do a festival in Madrid, pre-pandemic,” explains Lennox. “I thought it would be exciting to play all-new music. So I wrote five, maybe six songs for that. Most of them appear on this album in some form.

“I had the idea to do some straight-ahead recordings of bass, drums, guitar and singing. I would play everything, and then we would, kind of, abstract those songs into something else. But after recording everything, we just really liked

them the way they were. So that idea faded away, and this sort of rock band thing kept going! That’s how I find a lot of stuff happens: there’ll be maybe 10 different ideas I’ll have that hover around for a while, and some of them will go away, some of them grow into other things. And maybe two or three you can trace from the very beginning all the way to the end.”

Camping Out The decision was made to record Sinister Grift at Lennox’s own Estudio Campo in Lisbon: by his own admission a modest space, but no less than what was needed.

“I just wanted a room,” he explains. “It has a big kind of booth, where you can put a drum kit — which we did — and a control room. I have a pair of ATC monitors, a UAD Apollo interface, a two-channel Neve preamp, a Drawmer compressor, and that’s pretty much it. I don’t have a lot of outboard stuff. This was the first kind of ‘big project’ that we did there. And Josh was the first person I ever started

recording stuff with, so it felt like a closing of the circle, in a way. I was really excited to do it with him.”

For Dibb, the decision to record in Portugal presented a daunting but exciting set of challenges. “I was trying to figure out how much stuff it would make sense for me to fly over with,” he remembers.

“Noah had some [Shure] SM57s there, an [Shure] SM7B...

I had also convinced him to get some Coles 4038s when we were tracking Time Skiffs. He is quite gear-resistant in that way! He’ll pick certain things now and again, though, like his ATC monitors in the studio. It didn’t seem cost-effective for me to fly over with multiple cases, so he sent me a list of mics and preamps and stuff that he could get if needed, and I gave him a wish list based on that. What we ended up with was great.

We made it work.”

Work began in November 2023, with a streamlined and light-footed approach to recording. “We would jump around, day to day,” says Dibb. “I’d have mics up on the kit, but then I’d have to move them to another room to do percussion, and then move them back to the kit... I don’t want to overemphasise the jankiness of it; most of the records that I’ve done, Dave [Portner]’s records, too, they’re often like that. They almost feel closer to home recording.

So I get into the vibe of that. There’s something beautiful, I think, in getting to the point where you’re like, ‘Well, these are the limitations of what we have to work with, and we’re going to make it work.’”

he wanted, like distortion: that played a bigger role in this record than it has in other records. He had an Elektron Analog Drive, usually on the front end of his guitar, but he didn’t have any amps. Initially I was like, ‘Hey, man. It’d be cool to have an amp...’ But that ended up becoming

guitar performance with three different signals: there’d be like a completely DI’ed signal, there’d be the mono drive signal, and then there would oftentimes be another stereo signal that could either be foundational to the song, or be more like a wash in the background. So there’s times in the record where it feels like we must have tracked like three things on top of each other, but it’s actually just one performance.”

Distortion and spatial effects would both come to play a key role in the album’s sound, be they boutique, vintage or modern.

“There’s one pedal company, Vongon, who make some really beautiful pedals,” Dibb continues. “They have a reverb and vibrato pedal called an Ultrasheer, and you can switch the order of the two effects. So sometimes the reverb goes through the vibrato, which is a very different sound.

Josh Dibb: “I’ve observed other engineers who really differentiate tracking from production from mixing; those are three very separate stages for them. I have a very hard time doing that. I’m looking for vibe.” a thing. We recorded all the bass and the guitars direct and would work it out later down the line.

“I did re-amp some stuff in the end, I have a [Fender] Twin in my studio and a Music Man. But honestly, for a lot of it, I just was shaping the direct sound. I’d split his guitar at the front end and track one

Guitar Music “I wanted to do something that was rock,” says Lennox, “but, really early rock. The [2015’s Panda Bear Meets The] Grim Reaper, [2019’s] Buoys era — I guess that was, like, computer music, really. For this, it was early rock; that’s the kind of stuff that was really resonating with me. Fender had given me an Acoustasonic. We used that a bit, and then also, actually, my dad’s old guitar. A red Stratocaster. I’m very much like, whatever is around: I’ll just use that! I’m not super picky.”

“A lot of the stuff that I brought was pedals,” remembers Josh Dibb. “I brought a pretty hefty case of pedals! Noah had told me some sort of touchstone sounds

The first sessions for Sinister Grift took place at Lennox’s own Estudio Campo.

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INTER VIE W

NOAH LENNOX & JOSH DIBB: RECORDING SINISTER GRIF T

Effects pedals, mostly supplied by Josh Dibb, were central to the sound of the album.

We also borrowed a bunch of different distortion and fuzz pedals from Pete [Kember, aka Sonic Boom] and we cycled through those to find the right tone for different songs. We used the [Hologram] Microcosm as well; although that pedal I have a very love-hate relationship with.

It’s almost too easy. You turn it on and it’s instantly such a vibe! Also the [Hologram] Chroma Console — that became a huge part of it, although more in the mixing phase, I think. They weren’t out yet when we were recording, but by the time I got back to the States, a buddy of mine told me to check it out. I think Noah is now using it for playing live.”

Noises On The contributions of Animal Collective’s other members were varied. “‘Ends Meet’ has a Dave [Portner aka Avey Tare] noise solo,” Lennox notes. “I have no idea how he made it. I just said, ‘Give me a noise solo here!’ Two days later, it came back.”

Meanwhile, Brian Weitz, aka Geologist, had compiled a bespoke sample pack at Lennox’s request. “I asked Brian to make a big sound pack,” he explains.

“I had that before we even went into the studio. There’s something called Playbox that Native Instruments make. It’s a sort of matrix that you can pump your own samples into, but they’re all divided up into different sections: synths, voices, basses, noises... maybe five or six categories.

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I asked him to do me packs of all those, because I was planning on using that, but then we just wound up using them as audio files. I would drag them in, and we would see if we could sort them out.

“A lot of his stuff appears on the songs, little bits here and there. There’s a song called ‘Elegy For Noah Lou’ towards the end of the album, which we wove a whole bunch of his sounds behind. That one is sort of like a live take, where I was playing guitar and Josh was tweaking a table of effects that the guitar was running through, and then we kind of wove Brian’s sounds around that take.”

Where, then, is the line between collaborating as individuals and collaborating as Animal Collective? “I don’t think there really is one,” replies Lennox.

“At least, I’d prefer there not to be. It really feels less interesting to completely separate the two. One thing always bleeds into the next, so they all kind of feel part of the same creative wave to me. So there is no line, for me. It might sound lofty or corny, but I try to make sure that whatever is best for the song is most important.

Everything else should be secondary to that. Like, the solo on ‘Ends Meet’. I knew I wouldn’t do it as well as Dave would, so I asked him to do it.”

Split & Recombine Having finished the lion’s share of tracking in Lisbon in late 2023, Dibb

and Lennox separated to continue working on the album individually, before reuniting the following March for further recording and post-production. This was to take place in Baltimore; firstly at Dibb’s own studio, Rabbit Hole, and subsequently at the larger, SSL-furnished Wright Way. “After we did the main recordings, I did kind of a whole run of stuff with a [Sequential] Prophet-5,” remembers Lennox, “just kind of going through every song. I spent a couple of weeks, just making little parts anywhere I thought a song needed something.”

Dibb was similarly diligent during the interim. “I spent something like two-and-a-half months working on things before Noah came to the States for further production,” he says. “But the overall shape of stuff was more or less established by the time we left the tracking session. I recorded some piano stuff, the horn arrangements on ‘Ferry Lady’... that was stuff I wrote while we were away, and would send to him to see what he thought. Some of the guitar effects were locked in when we were tracking, but some of them didn’t land, so I had to reimagine them or try different things. That’s where having that original DI performance was really helpful.

“It’s like a two steps forward, one step back kind of thing. You know, a mixing choice can become a production choice — you might decide to do some

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INTER VIE W

NOAH LENNOX & JOSH DIBB: RECORDING SINISTER GRIF T

The Singing Panda A distinctive element that consistently shines through Panda Bear’s output, and Noah Lennox’s numerous guest appearances, is his signature vocal delivery. Needless to say, the right choice of microphone was critical. “Most of the lead vocals we did through a Soyuz 017,” says Josh Dibb. “I do think Noah felt like he’d finally found his mic. I think he has gotten one for himself, too.

Hearing himself back, even while tracking, it just felt really good to him. And I agreed.

“Noah has such a beautiful voice, so much clarity. But at the same time there is something very particular that’s needed. I think that that mic just brought out the precision of his voice,

but with enough gentleness at the same time.

It’s so smooth.”

The 017 wasn’t the only mic in play, however.

“There were some songs where I knew I wanted to use the SM7B instead,” continues Dibb.

“Where I just wanted that kind of mid-forward, focused kind of sound. And then some songs where I knew I wanted the 017 for that more open sound. On one occasion I ended up using a blend of the of the two: I felt like there was something I could get from the push of the SM7B, that wasn’t quite coming through in the 017. That kind of crunch in the vocals, a bit of distress already in the mic signal.”

Sinister Grift is the sixth Panda Bear album.

further processing on some fundamental element — and it might shift the entire feeling of what’s going on. I’ve observed other engineers who really differentiate tracking from production from mixing; those are three very separate stages for them. I have a very hard time doing that.

I’m looking for vibe.

“For me, the moment something is tracked, it’s like, ‘Does this feel right?’ It’s usually pretty hard for me just to say, ‘Oh yeah, we’ve recorded the snare decently.

It’ll be fine.’ I want to make sure of the sound we’re getting: sometimes that is just about mics in the room and the way that it sounds, but oftentimes it’s about additional processing. In tracking, I was Mixing took place at Wright Way Studio.

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immediately starting to add reverbs, add effects, tremolos, distortion, delays, like all sorts of stuff to try and get things into the space that I knew we were aiming for.

And in a lot of cases, a lot of that stuff did hold all the way to the end.”

Finally, stems were mixed down by way of the SSL console at Baltimore’s Wright Way to achieve, in Lennox’s words, “a little grit, a little analogue dust”.

Stereo subgroups were stemmed through the console and selectively processed through the studio’s vintage outboard gear — a vintage API 560 EQ or two, a UREI 1178 compressor where needed.

“Paul Mercer at Wright Way, he is obsessed with transformers!” says

Dibb. “He had made a three-channel transformer bus box, which basically the entire mix can run through. And you can switch between different transformer types. That transformer box was a huge part of it. We would just flip through the different options until we found one we liked — that would usually put a curve on the low end. We had done so much finetuning of the mixes already at that point. It was really more about just stepping back and hearing the warmth, seeing how the low end would really come to life.”

Shared Excitement Thanks to the involvement of Josh Dibb and Lennox’s bandmates, Sinister Grift is the latest flowering of creative relationships that have stood the test of decades. “We started working with multitrack cassette stuff together when we were maybe 13 years old,” says Dibb.

“That’s how we spent our weekends, we would mess around. And this, in many ways, feels the same. Really, nothing has changed. You know, it sounds like a funny way to put it, but I’ve been really grateful for that. There were moments where I was like, ‘God, I could really fuck this up.’ With Noah, if something’s bothering him — or if there’s something he’s feeling really good about — he’ll let me know. But he allows a lot of space to mess around.”

Noah Lennox, in fact, says that creative gratification comes as much from generating enthusiasm in those around him as it does from within. “Part of the impetus for getting a lot of people involved is, I like to make other people feel excited about doing the thing,” he says.

“I get more excited to see somebody else get really jazzed up. I feel like that drives me, more than my own contentment. It feels more fun.”

PODCASTS ELECTRONIC

RECORDING

MUSIC Robert Aiki Aubrey Lowe AKA Litchens

Composer and Sound Designer Robert Aiki Aubrey Lowe talks about his preferred sound modules and working methodologies, plus shares details some of the film and TV projects that he has recently been collaborating on.

& MIXING Guitar & Amp Modelling

Paul White delves into the fun world of modelled guitars, amps and cabs, that cannot only make your songwriting and studio sessions more creative and unique, but can also greatly assist with streamlining live performances.

ELECTRONIC

PEOPLE &

MUSIC Tony Rolando

My Life In Modules Tony talks about his beginnings in the industry and how Make Noise has relaunched in 2025 with a new team and a brand new module release, which he then showcases in an exclusive performance alongside four additional modules.

MUSIC INDUSTRY

Dave McCracken - Producer The MixBus Interview

Dave McCracken talks about his impressive career as a songwriter and Producer, starting from his early years at Sarm Studios with Trevor Horn, where he worked with Ian Brown and Depeche Mode.

Follow our channels by subscribing to the shows on Apple Podcasts, Spotify, Amazon Music or wherever you get your podcasts.

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

Check out our website page for further details

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ON TE ST

e-instruments Slower Kontakt Instrument

HHHHH Tape‑emulation plug-ins are very much a thing at present, and there are plenty to choose from. However, in the creation process for e-instruments’ latest Kontakt-based virtual instrument, the source instruments (64 in total and including pianos, strings, woodwinds, brass, guitars, keys, various pitched percussion and some classic synths) were actually recorded to four different types of tape machine — large reel-to-reel, small reel-to-reel, standard cassette and micro cassette. Then, they were resampled from those recordings with the tape playback slowed down to half-speed. The intention was to enhance the character (including the imperfections) that each tape format offers and to bring out the lower frequencies within the original sources, to create a warm and organic sound. The end result — Slower — offers some 230 presets and, while undoubtedly pitched at media composers, may well appeal to anyone working in ambient, lo-fi, hip-hop or electronica.

If that process, while obviously an interesting experiment, sounds like it is a lot of effort when you could simply slap a tape emulation plug-in on any old virtual piano, string or brass sound, let me place the cat firmly out the bag by saying Slower sounds absolutely fantastic. Pretty much every preset you load is full of character, depth and warmth. You get the essence of the original instrument but (presumably due to the tape speed reduction) something that’s rather beautifully different. Yes, e-instruments have gone to the effort of using used real analogue tape so you don’t have to and the results are very impressive.

Oh, and then they have also wrapped all these sounds into an intuitive UI that provides you with plenty of additional sound design, modulation and effects options, letting you tweak the sounds in all sorts of additional directions. This includes the option to switch between the four tape formats (things get progressively more degraded as you move to the more compact) and, if you do want even more ‘tapeiness’ (that’s probably not a word, but I hope you know what I mean), you can add additional tape noise (actual tape noise from the recorders used) within the Sound page’s Special section; lo-fi

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fans will love it. The presets come with three macro controls and the mod wheel configured for modulation but, via the Modulation page, you can dig pretty deep should you wish. This includes a step-based modulator which can, if required, be used to ‘pulse’ the volume for more rhythmic results.

Finally, the FX page provides a five-slot customisable effects chain, with 27 effects to choose from including reverb and delay.

Hats off to the creative team at e-instruments for both having the original idea and then the determination to follow it though to its very impressive conclusion. If you like the idea of adding something with genuine character to your sample library collection, Slower is undoubtedly worth exploring. Beautiful sound, beautifully done.

John Walden €149 www.e-instruments.com

Sonora Cinematic Poiesis Cello 2 Kontakt Instrument

HHHH Back in 2020, Dave Stewart reviewed the original Poiesis Cello for Kontakt and he had a lot of very good things to say about it. Rather than straightforward cello notes, the library comprises atmospheric articulations performed by principal cellist Alan Black, so although the note might not change throughout the duration of a sample, the playing technique is often varied to give the samples an evolving, cinematic quality. Many of these character shifts differ from what you’d expect in a normal cello performance, including haunting harmonics, jittery bowing and horror-movie bowing effects that sound more like a waterphone, but at the same time many of them layer well with more conventional string sounds.

The original cello samples are unprocessed so the user has free rein over what effects to add. There are effects on board in addition to modulation facilities and filters.

Patches for Poiesis Cello 2 are arranged by category under Poiesis Cello, Textural, Ambiences, Dynamic Crossfade, Mixed

Techniques, Mixed Techniques Motion, Motion Engine and X/Y Assignments.

Articulations show up as a separate instrument and both instruments come with a generous selection of presets that are easy to edit.

The Poiesis Cello 2 instrument uses two layers and a mod wheel, or other MIDI CCs can be used to change the X/Y parameters of the control pad, which can be used to crossfade between the A/B layers during performance by dragging the puck in the horizontal axis. Effect and filter control can be linked to the vertical axis. If the Fade button associated with a layer is turned off, then that sound remains at a fixed level while the level of the other can still be varied. The X/Y puck movement can also be recorded so that it traces the same path whenever a new note is triggered.

While the original Poiesis Cello was impressive, Poiesis Cello 2 takes things further while retaining the ethos of the original and still remaining affordable. (The good news for existing owners is that the upgrade is free.) The presets show off the scope of the instrument, which can produce hauntingly beautiful tones at one extreme and disturbing horror-movie effects at the other. It is easy to make changes simply by swapping out the samples in the layers — and after that there are the effects and filters to play with so there’s endless scope for making the sounds your own. Some sample-based instruments can sound impressive without being particularly useful in a musical context but Sonora Cinematic have got it exactly right.

Poiesis Cello 2 comes with a multitude of musically useful and wonderfully expressive cello sounds and I can see this instrument getting a lot of use both in music-for-picture and general music production. Paul White £82.50 www.sonoracinematic.com

Noon Instruments Vessels Kontakt Instrument

HHHHH Vessels is a 26GB sound library for Kontakt (v7.72 or later, full version or free Player) and follows Noon Instruments’ moody and atmospheric Toska.

Recorded at Holy Mountain Studios in Hackney, London, it presents a broad range of hardware synth-based timbres in adjustable blends of clean captures alongside variously patinated and ‘distressed’ re-amped and re-recorded versions of those same sounds. Across three banks (called Convergent, Divergent, and Relics) the synths are played through an oil drum and thundersheet, guitar amps, fed into a Korg MS-20 and various effects pedals, and also recorded to tape. The same 220 instrument presets (divided into Arp, Atmosphere, Keys, Pad and Pulse categories) make up Convergent and Divergent, but have different treatments. Relics has just 20 more heavily processed patches.

Getting right to it, to my ear Vessels is a rare and wonderful meeting of timbral complexity and straightforward musical potential, with many sounds having real depth and character but also a sort of wide-open generic quality. There’s a distinct lack of cheesy or novelty sounds, or anything aligned with particular stylistic genres. Instead you get an understated, moody coolness, with occasional deliberate naïvety: a neo-analogue character (you might say) that works beautifully for structural beds and textures rather than goofily drawing attention to itself. I was frequently reminded of certain veins of grown-up electronica: Ultraísta, Boards Of Canada, Arovane and others. And hence there’s a lot that should be an instant fit for suspenseful, dark and dramatic music for picture.

Happily, too, Vessels is no mere preset player. The compact, minimalist interface hides it, but there’s an extensive synth and effects architecture, with dual low-/high-pass filters, twin ADSRs, and a pair of LFOs with multiple modulation destinations. A dedicated distortion section and a convolution reverb with some excellent, unique impulses feels very much part of the synth engine. Crisp decaying plucks can quickly become ominous drones, and vice versa.

There are a few quirks, such as no user control of velocity sensitivity: some presets

respond, and others don’t. There’s no aftertouch response at all (so forget MPE...) but the mod wheel controls level at all times. Another slight bother for me were the many presets that loaded with harmony voices sounding below the played note: unusual and inspiring sometimes, but on the whole less compositionally useful than those that sound above, like partials. A few presets were a sniff out of tune, or at the wrong basic pitch, and when editing some patches I often missed a filter tracking parameter.

Also, with its multiple layers, Vessels can consume a chunk of CPU, and muting layers does not reduce the voice count, sadly.

However, none of this really diminished Vessels’ countless delights. It’s one those sound libraries in which almost every preset could launch a creative journey, while offering the flexibility to quickly adapt to specific needs. For composers it’ll provide instant, subtle atmosphere, and for music producers a range of unique timbres that are immediately playable, engaging, and soulful. Hugely inspiring. Robin Bigwood £99 www.nooninstruments.com

Fracture Sounds Soft String Soloists Kontakt Instrument

HHHH With Soft String Soloists, Fracture Sounds provide an intimate solo violin, viola and cello recorded at Nave Studios and inspired by the Scandinavian sound.

Presented in the free Kontakt Player, the primary feature of this library is a fantastic soft legato, which includes standard and portamento transitions. Where other bright, virtuosic solo strings libraries cannot convincingly produce soft melodies, Soft String Soloists fills that gap. The sound of this articulation is gorgeous, and it is also pretty agile. There is no round-robin legato on the transitions, however there is a very realistic round-robin same‑note re‑bow. The ability to trigger a naturally performed slow attack/release (using a keyswitch or the pitch-bend wheel) is welcomed.

Each instrument also comes with sustains, harmonics, gestures (beautiful ‘placed’ short notes) and swells (not

tempo-sync’ed, but lovely). These fit nicely within the style, and as with the legato, the performances, recordings and editing are of a very high quality.

The close, mid and far mic positions provide great sculpting flexibility, and can quickly be mixed using the Perspective slider. The close mic is bitey, while the far gives a nice broad sound, but not extremely reverberant.

The library comes with a host of useful features, the most fun of which is the instrument placement option. In the vein of plug-ins like VSL’s MIR, Samplicity’s Berlin Studio and Spitfire Audio’s AIR Studios Reverb, this virtual positioning tool (based on impulse responses recorded in the same room and with the same mic positions as the samples) allows the user to place the instrument in one of 11 positions ranging from front to back and left to right. I had a lot of fun creating bands of five violinists, four violas, three cellos and so on, all seated in different layouts. The Centre Stereo option provides an additional stereo mic position to centre the instrument, and I was happy to see the other microphone options automatically purging themselves when this was enabled. The global latency control keeps latency consistent even when switching between articulations — a function I wish was present in all sample libraries.

A couple of minor nitpicks; the vibrato is not controllable, however this limitation comes with the benefit of a beautiful natural onset of vibrato (crossfading vibrato in solo libraries is difficult without phasing issues or compromising natural vibrato in favour of an LFO‑based option). The instruments don’t reach super high, but the developer explained that this is because their experimentation showed going beyond the chosen ranges did not achieve the desired soft tone.

Alongside being useful for media composition, Soft String Soloists could easily find its way into neoclassical and pop/folk productions. It fills a niche very well, and is housed in an easy‑to‑use interface with great quality-of-life features. Thomas Field £149 www.fracturesounds.com Audio examples of this month’s libraries are available at www.soundonsound.com.

www.soundonsound.com / April 2025

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Q

Why aren’t my rendered and live plug-in sounds nulling?

I use Reaper, and to check that some clip effects were rendering properly, I bounced them and put them on a clean track. I matched the fader levels and flipped the polarity of the new track.

I should hear nothing when both tracks play, right? But I don’t. Is it a problem with Reaper or is there something I’m missing?

Thanks in advance for any insight!

Anonymous, via email Matt Houghton, Reviews Editor There are a few possible explanations here, but you don’t mention what plug-ins you’re using — and the plug-in is the most likely culprit. Two signals will only cancel completely when one is inverted if they are both identical. Yet lots of plug-ins don’t do quite the same thing each time you play through your track. For example, if there’s modulation involved, the phase of that modulation won’t necessarily be linked to your DAW’s transport — in other words, the modulation may not trigger from the

Q

What’s a good first microphone?

My daughter has decided she wants to take GCSE [a high-school qualification in the UK] music next year and has asked if we can help her by buying a microphone, which she says will help her with coursework. We’re keen to help, and have a budget of £75-£100 (about $95-125 including taxes). She’ll be

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Some plug-ins might not behave in exactly the same way every time you play your track. In the case of PSP’s excellent Wobbler plug-in, for example, the chosen Mode setting dictates whether or not the phase of the modulation will be reset so that it starts from the same point in its cycle each time you start playback.

start of its cycle each time you hit Play. The effect might involve some randomisation element in the algorithm too. Similarly, a sample-based instrument or drum trigger plug-in might, by design, randomly play different samples to make it sound more realistic. In such cases, your rendered file won’t be identical to the plug-in’s output next time you play the track — that could well explain why you’re not hearing silence.

Another possibility is that there’s a level change or other process applied in the rendering signal path. For example, if your initial track is routed through a bus on which the gain or fader is at anything but unity and you print that level change when rendering, and the rendered track is routed through that same bus, then the level change is applied once to the live track, but twice to the rendered one. The

same goes for any bus compression or EQ. Again, in either case, the two signals won’t null because they’re not identical.

If the changes are small, then one more avenue to explore is the resolution at which you’re rendering: if you export at a different sample rate or word length (‘bit depth’) than the project, and then re‑import to the project, that could cause small differences. However, since you mention “clip effects” I believe these are automatically rendered at the project settings — which is what leads me to suspect that it’s simply a matter of the plug-in doing slightly different things each time you hit play. What matters most, of course, is whether things sound good when you bounce them.

If they do, then there’s really no point looking for problems.

“You don’t mention what plug-ins you’re using — and the plug-in is the most likely culprit.” using it mainly for vocals but might need to record other things too, including acoustic guitar. Is our budget realistic, and would you be able to recommend something suitable?

Hillary Roper via email Matt Houghton For that budget, and arguably for someone of that age too, there are a few things to consider when choosing

a general-purpose microphone. But buy wisely, and you could have a mic that remains useful throughout your daughter’s school years and beyond.

Before I get onto the mic itself, it’s important to understand that the mic is never the only cost in even the most basic recording setup! You need to plug the mic in to something (eg. a computer, iPad or audio interface), you’ll need some sort of cable between the two, and unless she’s

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Q U E S T ION S A N S W E R E D, P R OBL E M S S OLV E D

Above: a large-diaphragm capacitor mic that offers both a USB connection to a computer/phone and a professional XLR output for mixers and audio interfaces could be a good first mic on a budget.

Below: for vocals specifically, it’s hard to go wrong with a good dynamic mic such as the Shure SM58, but you’ll need to budget for an audio interface along with other accessories.

happy always using a handheld mic, you’ll need a mic stand, a clip/shockmount of some sort, and ideally a pop filter. She’ll need to hear what she’s recorded, so will need some headphones at least. So to some extent, the advice depends on whether your daughter has access to a recording device and other accessories, and how far you can push the budget!

If all you need is a vocal mic, there are plenty of handheld cardioid dynamic mics that should be up to the job for now.

These are really ‘stage’ mics, designed for the singer’s mouth to be close to the grille, but that could well be a benefit for recording at home, since the closer the singer is to the mic, the louder the voice will be relative to the unwanted ‘room sound’ (the sound of typical domestic rooms is rarely helpful on a recording!) and any background noise.

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A ‘classic’ option is the Shure SM58, which will set you back around £90 ($114) new, and just about leave you enough for the XLR cable you’d need to plug it into a mixer or audio interface. Plenty of good records have been recorded with these, and they tend to hold their value pretty well.

Should your daughter ever want to gig, it should serve her very well there too. There are similar models from lots of reputable pro-audio manufacturers, including Rode, Audio-Technica, AKG and Beyerdynamic, amongst others, and some of those may cost less. Note, though, if you’re tempted to go second hand, that there are lots of reports of counterfeits, and that’s a risk you must weigh up. And do please steer clear of ‘no name’ models from Amazon, eBay and the like — you might get lucky but it’s a lottery!

While it will capture an acoustic guitar, a dynamic mic isn’t likely to be

a great choice for that, because you won’t generally want to be so close (the sound emanates from more than one part of a guitar) and you’ll usually want a more ‘natural’ sound than it can deliver. A capacitor (or ‘condenser’) mic would be preferable, and if it were just for guitar, I’d probably choose a cardioid small-diaphragm model (a ‘pencil mic’, as they’re often known). As you plan on using it primarily for vocals, though, a large-diaphragm model would be my pick, and it will still serve well for acoustic guitar.

The Audio-Technica AT2020 is within your budget and has long been a popular budget choice at SOS, but there’s strong competition from the likes of sE Electronics, Aston, Rode, MXL, Lewitt and many more. Almost all such mics will require phantom power to be delivered by a mic preamp, such as on a mixer or a USB audio interface. The good news is that you can often pick something capable for around £60 ($75) second hand. And while I can’t guarantee it, the problem of counterfeits at this end of the capacitor mic market seems to be little to none.

If your budget isn’t enough to cover an audio interface, then consider a USB capacitor mic that can plug into a computer and be powered by it. Again, steer well clear of the cheap ‘no name’ models aimed at karaoke and so on, and Audio-Technica offer a suitable model, the AT2020USB+, for a shade under £100 ($125). Like other such mics, this doubles up as an audio interface into which you can plug your headphones, so between this, a computer and some DAW software your daughter will have all she needs to record her voice/ instrument and listen back.

Finally, if the budget can be stretched just a little further, or you’re happy going second hand, I’d recommend you look for one that has both a USB port and an XLR socket — that way, your daughter will be able to use it both directly with her computer at home and at school with their mixers and audio interfaces, and it should still serve her well if she goes on to acquire a better audio interface in the future.

Sticking with the Audio-Technica range for now, the AT2020USB-XP is currently about £149 ($188) new, but there will be equivalent models from some of the other brands I’ve listed above.

“It’s important to understand that the mic is never the only cost in even the most basic recording setup!”

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PROBLEM SOLVING MARK GORDON

P

roblem solving seems to find its way into so many of the things I do, and I’ve come to realise that rather than being a burden, it gives me a huge amount of pleasure. It’s not just the joy of reaching a solution, but the whole process of achieving it that I enjoy. From tackling musical challenges to navigating the complex world of hi-tech studio equipment, problem solving offers endless opportunities to innovate and think outside the box.

The most obvious scenario is in troubleshooting malfunctioning gear or software. I have provided technical support for several

companies over the years and although this can come with its frustrations, the opportunity to pick apart a scenario and break it down into its component parts to solve its mysteries is something I find incredibly satisfying.

In my ‘weekend job’ as a gigging drummer in a function band, I’ve taken enormous pleasure in compiling backing tracks and working out the most effective and creative ways in which they can be utilised, using the latest and greatest hi-tech kit I can lay my hands on.

Nothing beats the excitement of suddenly realising that if you do ‘that’, then you can send a signal to ‘there’, which would make ‘this’

work! The fun has extended to the process of physically constructing a compact playback rig, comprising digital mixer and IEM monitoring system, complete with colour coordinated cables and labelling, to make the load-in and setup process as quick and painless as possible.

The same is true when I indulge in any studio-based work. I’m not a songwriter and I don’t claim to be particularly talented in what might be considered traditional musicianship. My real pleasure comes from ‘problem solving’ other people’s songs — which, by any other name, might be considered producing. Imagine you have a melody and you need to complement it with

“Problem-solving, at its core, is about overcoming obstacles, and it requires both logical thinking and creative intuition.”

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the right harmony, rhythm, and texture. The solution requires problem-solving skills, experimenting with different instruments, and testing various combinations of sounds to see what works best. Each layer you add to the composition solves a musical problem, helping the song evolve and take shape.

Problem-solving, at its core, is about overcoming obstacles, and it requires both logical thinking and creative intuition. Whether I’m troubleshooting a piece of audio equipment or trying to fix a mix that’s not coming together, the process is the same: break it down, experiment, and try things until the desired result is achieved. It’s this mix of creativity and logic that makes problem-solving so enjoyable.

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