SOUND ON SOUND [USA] December 2024 Issue

1985 — 2024

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LE ADER

TAPE IS BACK! Y

es, you read that right. Revox have introduced a brand‑new version of their classic B77 reel-to-reel recorder. For the trifling sum of €15,995, you, too, can learn to perform razorblade editing, azimuth adjustment and head demagnetisation, in pursuit of the glittering prize of over 20 minutes’ worth of two-track recording per reel. Or you can simply relax and enjoy one of the small but growing number of album releases currently available on quarter-inch tape, each of which will currently set you back several hundred pounds.

It’s pretty obvious that musicians and producers are not the main market here.

Revox’s press shots display the B77 MkIII artfully placed in a minimalist Scandi-style living room, conveniently omitting the cables, amp and loudspeakers you’d need to actually get any sound out of it. It would be easy to dismiss it as a lifestyle product for people with more money than sense, and the cynic might suggest Revox have missed a trick by not giving it a USB socket and a Bluetooth chip. But they haven’t, and whether or not working musicians have the money or the inclination to mix to tape, I think we should at least give Revox credit for trying to do things properly.

A few of these new MkIIIs will find their way into studios, where mastering engineers

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in particular will be glad of a brand-new, high-spec tape recorder that has no maintenance issues. But is it a problem if the other 95 percent end up in Brooklyn loft apartments or Mayfair townhouses, providing the soundtrack to financiers’ dinner parties? People who buy these for domestic use might have obscene amounts of disposable income, but at least they’re serious about listening to music. They’re also helping to keep alive the whole ecosystem of skills that surrounds tape machine manufacture, which are at serious risk of extinction. And if they end up spending five-figure sums in order to play back mixes that were probably done using Pro Tools and plug-ins, well, whatever makes people happy can’t be all bad.

The resurrection of tape as a consumer format might be strange, but I’m sure Revox are happy to sell their product to anyone who values it, and I wonder if that’s an attitude that music-makers could learn from. Although we’d all rather fill dance floors at underground clubs than end up on cringe YouTube videos, we can’t police access to our output. And, tape or no, we can’t control whether they hear it on high-end hi-fi or through their mobile phone speakers. But if there’s a worse outcome than the wrong people enjoying our music the wrong way, it’s for no-one to hear it at all.

“It would be easy to dismiss it as a lifestyle product for people with more money than sense...”

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www.soundonsound.com /December 2024

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124 INSIDE TRACK

IN THIS ISSUE

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December 2024 / issue 2 / volume 40

FEATURES 94 Modular Sebsongs main man Sebastian Hastrup talks modules and the importance of the DIY scene in Eurorack.

100 Musical Mixing:

The Art & The Science

All mix engineers need technical skills — but what’s most important is our ability to bring out the emotion in the music.

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112 Spotlight:

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In the age of mixerless studios, monitor controllers are on the rise!

116 Gear Of The Year The SOS team pick their favourite hardware and software from the last 12 months.

124 Inside Track:

Sean Cook & Nevin Sastry

The biggest hit of the decade so far took Shaboozey and his team half an hour to write — and weeks to perfect.

PAGE 38 130 How I Got That Sound Garth Richardson on the quest to find the perfect bass sound for Mudvayne’s ‘Dig’.

134 Q&A Your studio and recording questions answered.

138 Why I Love... Effects Pedals Paul White explains why, in an age of plug-in plenty, he still loves his stompboxes.

42 PWM MANTIS

132 10 Phantom Rooms

Low End Toys Sample Library

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Arturia KeyLab MkIII

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Boss IR-2

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Boredbrain Xcelon

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Modular Microphone System

Eurorack Module

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DPW Design P1 Pedal I/O

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Dreamtonics Vocoflex Voice Transformer Plug-in

64 Drum Workshop DW

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132 Native Instruments Alicia’s

Electric Keys Sample Library

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Console & gravitas MDS Dolby Atmos Mastering Software & Processor

MNTRA Pyramids

Soyuz 013 Ambisonic Ambisonic Microphone

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Tascam Model 2400 24-Track Recording Console

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Toontrack Alchemy Of Pop EZX EZdrummer 3/Superior Drummer 3 Expansion

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Warm Audio Pedal76 Compressor Pedal

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Waves Curves Equator Resonance Suppressor Plug-in

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Wes Audio ngTube EQ Dual-channel Equaliser

Sample Library

Soundworks 32 Fiedler Audio Mastering

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ModeAudio Airspace Reverb Plug-in

Eurorack Module

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Microtech Gefell Generation 4

Sennheiser MD 421 Kompakt Dynamic Microphone

Lunatic Audio Rocket Fuel Creative Effects Plug-in

Amp & Cab Emulation Pedal

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Kii Seven Active Monitors

Controller Keyboard

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54 Interspace Industries

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ADDAC System ADDAC310 ACS Custom Emotion Dynamic

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Mixing & Mastering Plug-ins

Eurorack Module

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Native Instruments Kontakt 8 Software Sampler

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Neural DSP Parallax X Bass Processing Plug-in

WORKSHOPS 104 106 108 110

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ON TE ST

Sennheiser MD 421 Kompakt Dynamic Microphone

Sennheiser have shrunk down their classic 421 — but have they retained everything that made it so popular?

NEIL ROGERS

I

f we’re comfortable using the term ‘legendary’ to describe a microphone, then I think it’s fair to include the Sennheiser MD 421 in that category.

With over half a million sold since its introduction in 1960, the 421 is a microphone most engineers will be

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familiar with, and one that can be found in the mic lockers of most commercial recording studios. I don’t like using the term ‘industry standard’, but if you ask any engineer to name the most commonly used mic for recording toms on a drum kit, I expect most to name this well-known dynamic mic. It’s not just toms, though: the 421’s ability to capture a ‘full’

sound in high‑SPL settings has made it a long-established choice for close miking guitar and bass cabinets, brass and vocals — and in any role where a reliable, unfussy dynamic mic is a good option.

What’s New I asked a few fellow engineers what they might like to improve about the 421, and

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ON TE ST

S E N N H E I S E R M D 4 21 K O M P A K T

we all without hesitation said “The clip!”

In part because of the angles it ends up in when used around a drum kit, you only have to lightly press the release switch on the original 421 clip and gravity will fire the mic forwards — often spilling onto the tom below and perhaps then bouncing off onto the floor. It’s a testament to how rugged these mics are that they can handle this abuse, but both of the 421s in my collection now have electrical tape permanently holding them to their clips.

It’s great, then, to see that the new 421 Kompakt now features a self-contained clip, which means this mic isn’t going anywhere!

The clue to the other significant practical improvement is in the name: this new Kompakt version is just over half the length of the original model, making it ideal for getting into position around a busy drum kit. Embracing the application it has become most widely known for, the Kompakt comes with an optional clip, the MZH, for attaching the mic directly to the edge of a drum.

This worked nicely in the studio and would, I imagine, be even more useful for live engineers.

It appears that Sennheiser have been careful to retain the sound of the original — a relatively ‘full range’ dynamic mic, with a frequency response of 30Hz-17kHz. The low-end response is achieved in part by an internal ‘bass tube’ port, which the designers claim to have retained in this more diminutive new version. Before I report on how the Kompakt compared with the two 421s I have in my studio, the only other difference that I can tell, apart from the price, is that the Kompakt omits the larger version’s bass roll-off control, which allows users to counter excessive proximity effect when using the mic up close. Like many, I suspect, I have never touched this control in well over a decade of using my 421s, so all in all the Kompakt seems like a sensibly repackaged 421 for studio and live engineers — providing it sounds the same...

flip side of this midrange peak is that cymbal bleed into the tom mics can sometimes be very unpleasant-sounding.

The Kompakt is just like the original in all of these respects.

Moving on from tom miking, the 421 Kompakt did an excellent job when used for close‑miking a saxophone player in a live session, handling what can be a surprisingly loud instrument with ease. It also fared well in front of a guitar cabinet (many engineers like the way the presence peak can present guitars), and it sounded full and satisfying in front of a very loud bass cab. Thanks to its compact size, I also found this new mic to be an excellent alternative to an Shure SM57 or beyerdynamic M201 for close snare miking.

The 421 Kompakt is only 122mm long, compared to the original’s 215mm.

isolated from the drum itself. I found no issue in that regard, and when I retired to the control room for a listen, I was impressed with the clear, full-sounding rack tom being reproduced through my monitors. The clip gives you just enough control over the mic’s angle but you can, of course, use a stand if you prefer.

Summing Up There’s nothing groundbreaking about the Sennheiser MD 421 Kompakt, but the practical changes the designers have made are significant and well thought-out.

There’s an awful lot of choice on offer when buying a dynamic mic in this price range, especially if you also consider the second‑hand market, though I’ve learned the hard way to be cautious about buying ‘vintage’ models in particular. The capsules in dynamic mics can lose their ability to reproduce low frequencies over time, and it’s often the case that they are unrepairable, or at least not cost-effective to repait.

I mention this as the new MD 421 Kompakt comes in at a similar price to many vintage Sennheiser 421s I’ve seen for sale — but with this new model, you can be confident you’re getting a mic that works perfectly. It’s also significantly cheaper than a brand-new ‘full size’ MD 421, and if you’re a studio or live engineer who specifically wants a 421, I struggle to see what you would miss compared to the original.

“I used the 421 Kompakt on several drum recording sessions during the review period and I’m happy to report it sounded indistinguishable from my original 421s...”

In Use It had to be toms as a first test for this new release, especially as I was also keen to see how I got on with the MZH drum rim mount. Studio engineers can be a bit suspicious of clip-on mics on drums, with concerns about the mic not being

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The smaller size of the Kompakt is a significant benefit when multi-miking a busy drum kit, and the ability to easily get the mic into position on a floor tom whilst avoiding an overhanging ride cymbal was much appreciated.

I used the 421 Kompakt on several drum recording sessions during the review period and I’m happy to report it sounded indistinguishable from my original 421s on both rack and floor tom recordings.

One thing that has always made the 421 a double-edged sword on drums is how it handles sources with a lot of upper‑midrange information. There’s a pretty substantial presence peak at around 4-5kHz; this can bring out clarity on toms, and alongside its extended low-frequency response, is what has made the 421 the go-to tom mic for generations of engineers. However, the

summary The MD 421 Kompakt is a streamlined, smaller and more affordable version of Sennheiser’s classic dymamic mic. It’s cheaper than both new and used originals, its smaller size makes miking drums easier, it sounds identical to the full-size version — and they fixed the clip!

$ $279; with MZH clip $299.

T Sennheiser USA +1 877 736 6434 W www.sennheiser.com

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ON TE ST

Vocoflex imposes the timbre of one or more reference voices onto another — so just how convincing are the results?

PAUL WHITE

W

e’ve had the ability to manipulate the pitch and formants of vocal parts for decades now, but while these processors have their applications I’ve always been a little disappointed when trying to obtain really natural-sounding results. Supporting all the mainstream plug-in formats for Mac and Windows hosts, Dreamtonics’ new Vocoflex, which also works standalone, tackles the same problem from a different angle, and I’ve been genuinely impressed by what it can do.

Basic Operation Essentially, Vocoflex applies the characteristics of one or more separate voice recordings to another voice recording: it can load multiple ‘reference’ voice samples and then morph between their characteristics, before applying them to the input signal.

So there’s scope for creating a wide range of vocal timbres.

Reference voice samples in WAV format, which need only be 10 to 20 seconds long (but should be solo voices, free from processing such as reverb or delay), can be dragged and dropped directly onto the Vocoflex GUI, at which point they are analysed and displayed as a series of points on a curve. Vocoflex breaks these segments down into shorter segments, and analyses the formant profile for each one (because vocal formants can change during a performance); the points relate to how the sample has been broken into segments.

Higher‑pitched or female-sounding voices are generally placed to the right of the screen and lower‑pitched, male-sounding voices are arranged on the left.

For the process to produce a natural sound, the reference voice should, ideally, be in the same pitch range as the vocal you’re working on, but where there’s a discrepancy, clicking a tuning fork icon opens a small display next to the displayed curve and a slider below the tuning fork. Pitch‑shifting can then be used to match the input voice as closely as possible to the reference voice. This step is particularly important

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The main Vocoflex screen.

Dreamtonics Vocoflex

Voice Transformer Plug-in

when changing a male voice to a female one with a naturally different register, or vice versa.

As you play the voice track, it’s possible to influence the result by moving a cursor on screen. This is joined by thin lines to the points on the reference voice curve and as you move it the contribution of the different points is changed. The differences can be quite subtle, as the reference formants may not change much from point to point, but things change completely when you drag in another reference voice. If you then place the cursor between the two reference curves, the two voice characters are morphed and the blend changes as you move the cursor. If you drag the cursor behind one of the curves, the display shows a subtle shadow being cast on the other curve and in that scenario, Vocoflex shapes the sound in the opposite direction to the shadowed curve — effectively the character of the shadowed curve is subtracted from the composite timbre. While that might not quite sound entirely intuitive, in practice it’s just a matter of moving the cursor around until you hear something you like.

Processing isn’t limited to two reference voices: you can drag in several of them, each with different characteristics. Once analysed, these are displayed as new curves, each assigned a different colour.

Again the various voice types are arranged automatically across the screen, so that similar voice types are placed close together and in general terms male and female

timbres are spread from left to right, respectively. Moving the cursor around the screen then creates a smooth voice morph, with the lines from the curves running to the cursor to show which references are influencing the source.

Discrete cursor positions can be stored as ‘waypoints’ and assigned for control using MIDI. For example, a keyboard or foot pedal can be useful when using this software live, but conventional DAW automation can also be used to control the cursor position. Waypoint mode is entered using the icon at the lower right of the window, after which a new waypoint is added at the current cursor position when you press the ‘+’ button. A Real-time button allows for low-latency ‘live’ use, with the processing time of Vocoflex going as low as 35ms. If this is overly taxing for

ON TE ST

DREAMTONICS VOCOFLEX

its vocal output can be transformed using Vocoflex.

A free light version of Synthesizer V Studio with a couple included voice types is available from the Dreamtonics website.

A database of 40 voices is included with Vocoflex, and additional ones can be purchased if desired.

Putting It Into Practice

your computer’s CPU, lower quality modes that ask less of your CPU are available.

No Voices? No Problem!

In a creative environment, where you’re looking for quick results and don’t have a huge library of reference voice samples, Vocoflex offers you another option. To the right of the window, around halfway down, are two icons, one for the ‘curves’ mode I’ve just discussed, and another resembling a dot within a circle. This brings up what Dreamtonics call the Voice Generator, which shows up as a coloured square in the centre of the window.

The left/right axis effects a male/female transformation while the vertical axis adjusts tonality. Below the coloured square are buttons for Random and Apply, as well as a code for the current colour. The actual vocal timbre is mapped onto the colour of the square. Clicking Random brings up a new colour and a new voice character, though you can also enter your own colour codes. Clicking Apply adds the new timbre to the main curves view, where it appears as a single point. Add one or more reference voice samples and this new timbre can be morphed with them to create further variations. The rabbit hole of timbral tweaking can get pretty deep, of course, but the process is simple and intuitive from the user’s standpoint.

Vocoflex can also be used in conjunction with Dreamtonics’ Synthesizer V Studio virtual instrument plug-in, with which you can synthesise surprisingly realistic-sounding pitched lyrics. Synthesizer V Studio itself can only be used with voice databases licensed from Dreamtonics (so as to avoid copyright issues), but interestingly

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There are all sorts of reasons The Voice Generator can help why you might want to turn when you don’t have one voice into another, but suitable reference the big question, really, is how voices to hand. effective Vocoflex is at doing that job. I have to say that the transformations can be immensely useful in giving you a more commercial-sounding voice that leans in a specific direction — assuming of course that you start off with a proficient vocal track in the first place.

If the voice you want to transform has a similar pitch range to your reference samples, then you can expect to achieve natural-sounding results fairly easily.

Inevitably there are some complex voices, especially those that include a lot of throat or lips noise, that create occasional audible artefacts when processed, but most cleanly recorded vocals work fine. Even transforming male voices to female ones or vice versa worked rather better than I expected — you just have to remember to use the onboard pitch‑shifter to match the source voice to the reference one.

This software also raises some interesting possibilities and ethical/legal questions. For example, if copyright weren’t an issue, and you used an isolated snippet of a commercial recording as your reference, could you use Vocoflex to make you sound like Elvis, or make Bob Dylan sound like Kate Bush? The answer would have to be no. Why? Simply because there’s a lot more to a singer’s sonic identity than their vocal timbre: phrasing, regional accent, the way a singer shapes their note transitions and other aspects of delivery... All these things remain unique to individuals.

The way that the source vocal timbre changes during performance also has a bearing on this.

As for going the other way, Dreamtonics tell us they apply additional security, such as ID verification and file watermarking to avoid the misuse of copyright vocal recordings. While I’m unsure as to the legality of extracting somebody else’s formants without permission, I suspect there could well be serious legal problems if you were to take somebody else’s vocal track and try to disguise it using Vocoflex!

When used as intended though, with reference voices you’ve recorded or to which you own the copyright, or with Dreamtonics’ voice databases, Vocoflex is undoubtedly a powerful tool that can produce natural-sounding vocal transformations that are genuinely useful in a musical context.

Out of interest, I also experimented with processing some sources other than voices and, most notably, I found that I could give an eBow guitar recording an interesting vocal character, even if there were a few octave pitch jumps in places where Vocoflex struggled with this new ‘voice’.

That is perhaps not surprising, given that the eBow can generate very strong harmonics in a very non-vocalist kind of way. But my point is that experimentation of this kind can occasionally produce very worthwhile ‘sound-design’ results. As a further experiment, I took some dialogue recording from a podcast I’d done with SOS Technical Editor Hugh Robjohns, and processed his voice using a sample of mine as the reference. The end result didn’t sound remotely like me — Hugh has a different accent and a different style of delivery — but importantly, what came out still sounded extremely natural and, to my ear, quite unlike either of us.

Verdict The more I worked with Vocoflex, the more impressed I became. Simple jobs like giving the same singer different characters for creating layered backing vocals worked beautifully, but the process is clean enough that in most instances it can be applied to a main vocal without ever giving the game away, and the results really can sound exceptional. If you work with vocals a lot, or perhaps if you need alternative voices for your songwriting demos, then you should find plenty of uses for Vocoflex.

summary A practical and easy-to-use voice-morphing tool that sounds way more natural than simple formant shifters.

$ $199.

W https://dreamtonics.com

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ON TE ST

Microtech Gefell Generation 4

Modular Microphone System Drawing on their established 300-series pencil mics, Microtech Gefell have created a small but potent modular range.

SAM INGLIS

W

hen his Berlin factory was damaged in an air raid in 1943, Georg Neumann set up a new plant in the small Thuringian town of Gefell. At the end of the Second World War, this area became part of the communist German Democratic Republic.

Neumann himself moved back to Berlin, where the Neumann company remain to

Microtech Gefell Generation 4 pros • Very good specifications and subjective sound quality.

• High SPL handling without the need for a pad.

• Beautifully made and extremely compact.

• Consistent tonality across all three directional capsules.

cons • Full range of accessories not yet available.

• Quite a bit more expensive than the fixed-capsule 300-series mics.

summary Microtech Gefell’s miniature modular mics are a joy to use and sound consistently excellent.

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this day, while the Gefell factory became a leading supplier of audio equipment to the Soviet bloc. Since returning to private ownership after German reunification, Microtech Gefell have remained one of the most important manufacturers of studio and measurement mics in the world, with a truly vast range of products.

Despite their heritage and their undoubted quality, though, Gefell mics still seem to fly under the radar, and don’t always get the attention that’s accorded to names such as Neumann or Shure.

In typically understated fashion, the company managed to launch an entire new range at this year’s AES show in New York without many people noticing! But now that they’re here, the Generation 4 series looks to offer a compelling alternative to modular mics from the likes of Schoeps, Neumann and DPA.

Mix & Match Microtech Gefell have always offered modular microphone systems, and for many years the workhorse of their line was the MV692 body, which could accept both large‑ and small-diaphragm capsules.

That is long discontinued, but for the last 30 years or so, their extensive range of small-diaphragm mics has included the SMS2000 modular system, with separate

capsules and preamp bodies. This modular range has coexisted peacefully with the somewhat more affordable M300, M310, M320 and M330 fixed-capsule cardioid, supercardioid, omnidirectional and subcardioid pencil mics, as well as a selection of omni mics with diaphragms made from rolled nickel, a technology that is now pretty much unique to Microtech Gefell.

The M3xx models remain current products, but the SMS2000 has been discontinued in favour of a new and much more compact modular system. At launch, the Generation 4 range comprises six core products. The MV400X and MV400L are transformerless, phantom-powered, solid-state impedance converters. The two are electrically and functionally identical, but whereas the X is equipped with a conventional XLR socket, the L has a Lemo connector. The main advantage of this is miniaturisation: the MV400L is a mere 46.5 x 21mm, and the cable itself has a very unobtrusive visual profile. Though not quite as small, the MV400X too is impressively compact, at 57 x 21mm; for reference, the M300 is 130mm long and an SMS2000 body plus capsule 166mm. The bodies are complemented by four capsules designated the M40, 41, 42 and 43. These are, respectively, cardioid, supercardioid, omnidirectional and subcardioid or ‘wide

cardioid’, and are screwed on in the usual way, with electrical contact made by two sprung gold contacts. All are externally polarised, small-diaphragm capsules, and they use a conventional gold-sputtered Mylar diaphragm.

As usual with Gefell, the core products are also supported by a range of accessories. These currently include three different clips, two windshields and a “weather protected case” which will hold a body and clip with capsule attached, along with up to three further capsules. At the time of writing, none of the available clips offers any shock protection, but a shockmount is currently in development. For review, I was loaned two MV400X bodies and two full sets of capsules, with a pair of MC4.01 cases to keep them all safe.

Back To Black The Generation 4 mics are available only in matte black, which further helps to keep a low visual profile. Despite their diminutive size, there’s a pleasing solidity to the construction, and the finish is absolutely flawless. The threading is

precise and robust, and I had no difficulty changing capsules.

Full specifications are available on the Gefell website along with frequency response and polar pattern charts, and all are impressive. Self-noise ranges from 13dBA for the omni capsule up to 15dBA for the wide cardioid, and sensitivity likewise falls within a narrow range, from 11mV/Pa for the wide cardioid to 15mV/Pa for the omni.

All of the capsules are specified for at least 145dB SPL before 0.5% THD is reached, which explains why there are no switchable or inline pads. There are also currently no options for high-pass filtering, which some SMS2000 users might miss.

The frequency response plots for the three directional capsules illustrate minor variations on a theme. All are flat through the midrange up to about 4kHz, with a gentle high-frequency emphasis culminating in peaks at 7kHz and 12kHz. The relative strength of these two peaks varies slightly, and the M41 supercardioid capsule exhibits a steeper roll-off at either end of the spectrum compared with the M40 cardioid and M43 subcardioid. The

M42 omni capsule, meanwhile, is equalised for diffuse-field use, with a ruler-flat response through the bass and midrange complemented by a broad 8dB lift centred around 9kHz. It was the M42/MV400X combo that I tried first, as an A‑B pair on a choral recording. My usual first choice for this role is a pair of MBHO 603A preamps with KA100DK omni capsules. These have always delivered excellent results, but I have to say the Generation 4 mics were even better. The two review samples seemed perfectly matched, and although the sound of the two systems was broadly similar, the Gefells provided better definition and a slightly more present upper midrange, without ever sounding harsh or strained.

Returning to the studio and swapping to the cardioid capsules, I turned to my trusty Neumann KM84 as a point of reference. In this sort of test, you can sometimes hear that another mic is theoretically superior — perhaps with a more extended frequency response, lower noise, or greater headroom — yet nevertheless prefer the subjective sound of the KM84. There’s something

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MI C ROT EC H GE FE LL GE N E R ATION 4

fundamentally ‘right’ about the sound of the Neumann classic that explains why it’s still so sought-after today, and other pencil mics tend either to sound sharp and edgy in comparison, or neutral but boring. Not so the M40/MV400X combination. It’s audibly cleaner than the KM84, without the slight coloration that is presumably introduced by the latter’s transformer. It’s also a touch brighter. But it is, for want of a better word, equally musical. The additional clarity doesn’t come across as sterile or cold, and harmonically rich sources such as cymbals never sound harsh or over-prominent. In fact, the most obvious difference compared with my KM84 was that the Gefell combo had much greater low-frequency extension, to the point where I think you could probably get away without a dedicated bass drum mic on a minimal kit recording.

I didn’t run into any real-world situations during the review period where the supercardioid or subcardioid capsules were useful, but I ran numerous comparisons between them and the cardioid. What impressed me most about the results was the consistency of subjective on-axis tonality. If you do find yourself in that situation where you’ve put up cardioid mics but decide that a narrower or broader pickup is called for, you could swap these out without really affecting the core tone of whatever it is that you’re capturing. The subcardioid pattern in particular is one that I’m not sure I would ever use enough to justify buying, but the beauty of a modular system is that you only need to pay for the bits you’re going to use!

That said, this particular modular system is still new, and is not yet as comprehensive as equivalents from other manufacturers. At the other extreme, Schoeps’ long-established Colette range now encompasses more than 10 different capsules, half a dozen bodies and a broad range of accessories. If you want figure-8, shotgun or free-field omni capsules, for example, the Generation 4 range currently cannot satisfy you.

Three Or Four?

One key comparison I wasn’t able to make during the review period was with other Gefell microphones such as the preceding SMS2000 models, or indeed the still-current 300 series. Although these are older designs, they nevertheless offer very good specifications, and in terms of directionality, there’s a 300-series mic to match every MV400X-plus-capsule combination from the Generation 4 series. As with the 300 series,

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The MC4.01 case houses a fully assembled mic plus three additional capsules.

there’s a stereo bar available, but not all of the other planned accessories have yet been announced. And if you do choose to go down the modular route, you’ll currently pay quite a large premium compared with the 300 series. In the UK, for example, the cardioid M300 retails for around 60 percent of the joint cost of the MV400X body and M40 capsule.

Now it’s true that for all I know the two might sound completely different, but in practice I’d be surprised if this was the case.

Assuming it’s not, then for music recording applications where visual impact is not an issue and flexibility of patterns is not required, the 300 series may offer more bang for the buck. But the Generation 4 fully exploits the technological possibilities of 2024, and its extremely compact dimensions aren’t only important in terms of visibility: they serve to make these some of the most lightweight and easily portable mics around, which will be an important factor for anyone travelling to inaccessible

locations, using them on boom arms or flying them over an orchestra. A fully kitted out MC4.01 case weighs next to nothing, fits in a large pocket and houses the means to make state-of-the-art recordings.

It took a good deal of self-restraint to send the review units back!

Pricing The full price list for the Generation 4 mics takes up a side of closely spaced A4 paper, since every single element is sold separately and there are also several bundled options. Pricing for the core items, however, is relatively straightforward.

The MV400X and 400L bodies each retail at $760, whilst the capsules are all $830 each. So, without clips or cases, a complete Generation 4 mic comes in at just under $1600.

$ See Pricing box.

W www.microtechgefell.de

ON TE ST

Rocket Fuel’s X/Y pad can act as a macro control, modulating multiple parameters, and can itself be modulated by the LFOs.

Lunatic Audio Creative Effects Rocket Fuel Plug-in

Looking for a new plug-in to fuel your creative endeavours?

JOHN WALDEN

L

unatic Audio obviously have a thing for creative multi-effects. I reviewed their impressive Narcotic plug-in back in SOS May 2022 (https:// sosm.ag/lunatic-audio-narcotic) and I’ve made regular use of it since, whenever I’ve needed to add a bit of sonic magic to a sound. Launched at a lower price and with what, at first sight, appears to be a somewhat more straightforward GUI, Rocket Fuel covers similar territory: you get a user-configurable multi-effects chain and a huge range of parameter modulation options that combine to provide almost endless creative ways to process a sound source.

Three Screens Rocket Fuel doesn’t hide its processing potential from the user, but Lunatic Audio have done a pretty good job of organising what’s quite a deep

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feature set into three main screens: an X/Y pad page, an effects chain/modulators page, and a modulation matrix page. These are accessed using three buttons at the top of the GUI and whichever is selected fills the bulk of the (resizeable) window, while the ever-present panel on the right contains the preset system, macro controls for the X/Y pad, and the main input, output, mix (wet/ dry balance) and filter controls. The last of these targets Rocket Fuel’s processing to a specific frequency range; all other frequencies pass through unprocessed.

Rocket Fuel’s X/Y pad is a familiar modulation concept — having assigned parameters to the X or Y axis of the pad elsewhere in the GUI, you can manually apply modulation by moving the node control. You can also modulate the X and Y axis values around the node’s position using any of four LFO sources, of which more below.

The FX Chain & Modulators page is where the bulk of the sound design takes place. The chain is constructed by combining any of Rocket Fuel’s nine effects modules (Chorus, Compressor, Delay, Distortion, Filter, Limiter, Phaser, Reverb and Utility) in the upper part of the page. Note that most of these effects can only be used once in a given chain, but you do get two instances of the Filter and Utility modules. The module order can be changed by dragging and dropping.

Each module has a compact control set that provides plenty of flexibility, and every control can

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be modulated if you wish. There’s little point providing a full run-down here, but there are a few observations worth highlighting. For example, the Utility module provides pan, gain and stereo width control — modulating pan and volume strikes me as being a more ‘creative’ endeavour than the term ‘utility’ might suggest, and it does these jobs with ease. All the modules are well specified, but the Filter and Distortion modules are worthy of particular attention. In addition to their more obvious controls, the former offers a huge number of underlying filter types (chosen using a drop-down menu in the mini filter curve display), while the latter includes an impressive range of distortion types, spanning everything from soft or tube-driven overdrive, through to full-on gnarly fuzz and bit-crushing. The Delay is also very flexible, with a filter, ping-pong options, and full control over left/right timing.

Lunatic have done a great job of easing you into the sound-design process by offering preset systems that work at every level of the plug-in.

As well as global presets, effects chain presets and modulation presets (all available via the right-side panel), every effects module also has its own presets and, as I’ll explore in more detail shortly, the four LFOs also have a dedicated preset system of envelopes/ curves. Perhaps my only minor gripe (which I imagine could be addressed pretty easily) is the lack of some instrument-based categories among the global presets. This is something that’s present in Narcotic, and I’ve found them really useful when auditioning presets for a particular type of source sound, such as drums, bass or vocals.

All Mod Cons Not only can pretty much every parameter in a Rocket Fuel signal chain can be modulated, but multiple parameters can be linked to a single modulation source, and individual parameters can be modulated by multiple sources. So whether you want the most subtle of sonic motions or full-on audio mayhem, the plug-in’s engine lets you create it.

After direct modulation through the X/Y pad, the FX Chain & Modulators page’s four colour-coded LFO modulators will

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LU N ATI C AU DIO RO CK E T F U E L

probably cater for the bulk of your initial modulation assignments. To link an LFO to a parameter, you simply drag from the colour-coded ‘+’ button and drop it on the target parameter in the upper effects chain panel.

A modulation Depth control (also colour-coded) pops open above the target parameter to allow you do dial in the required modulation relationship between the LFO source and the target parameter; multiple Depth controls will appear if the parameter is the target of more than one modulation source. If you move away from the parameter, this pop-up control disappears but will reappear again whenever you hover the mouse over it. You can also drop an LFO assignment onto the controls of the right-side panel, including the X/Y axis controls. All in all, it’s a delightfully simple and very intuitive approach to modulation assignment.

The four LFOs are identically specified, each coming with an individual Rate control that can be toggled between specifying Hertz or tempo-sync’ed timing. You can draw your own curves using a number of different brush types, and can specify both a horizontal and vertical grid resolution to assist in that process. The LFO

preset collection is well stocked, categorised into useful types such as Basic (waveforms), Envelopes, Gated and Grooves. You can add your own categories and user-created curves.

Finally, if you wish to get even more detailed when customising your modulation settings, the Modulation Matrix page provides a full list of all the active modulation assignments in the current Rocket Fuel instance. As this follows a fairly conventional format, it’s relatively easy to navigate and provides plenty of options for deep adjustment over how any modulation source/target relationship is configured.

Playing The Fuel Applying a little (or a lot!) of Rocket Fuel to a range of audio sources is a lot of fun. The most obvious candidates are sustained synth-based sounds, and the potential to bring even the most timid pad sound out of its sonic shell is considerable. Maybe some cool filter-based processing or a dollop of gritty drive, or perhaps some tempo-sync’ed volume modulation to create rhythmic or pulsing effects... there are lots of ways to bring such sounds alive.

With more complex starting points, such as a bass line, a rhythmic guitar part, a sung vocal or a drum loop, it’s possible to have too much of a good thing!

In these cases, unless you’re seeking to create an obvious spot-effect, I’d suggest you have a better chance of achieving satisfying results if you confine yourself to a signal chain with just one or two modules (there are some suitable starting points in the FX Chain’s Simple preset category), lower the Mix value, or focus the processing on a specific frequency range using the filter controls in the right-side panel.

Indeed, it’s worth stating that while Rocket Fuel can do ‘complex and creative’ with ease, if you narrow your options down a little in these ways, it’s also very capable of subtlety. If all you need to do is gently coax a little extra movement from a sound, Rocket Fuel can do just that.

Multi-effects, Multi-choice!

When it comes to creative multi-effects, we’re utterly spoiled for choice today. In SOS we’ve reviewed The FX Chain & Modulators page lets you build your effects chain and easily link modulation targets to sources.

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The Modulation Matrix page provides deeper editing options for your modulation settings.

various options in recent years, including (to name just a few) Output’s Movement, Sugar Bytes’ Turnado and Effectrix, Baby Audio’s Transit, and UJAM’s Finisher Dynamo, and they all provide viable alternatives to Rocket Fuel. Each has specific strengths and they range considerably in price and complexity, so there should be something among them that will be a good fit for almost any type of user. In terms of the creative processing options it provides, Rocket Fuel can be considered a strong competitor for all of the above.

But Rocket Fuel’s biggest direct competition is arguably Lunatic Audio’s other flagship product, Narcotic. Narcotic is a creative marvel that offers a wider selection of effects types. It’s completely bonkers (in a good way!) and very addictive. In contrast, Rocket Fuel boasts more modulation sources and perhaps supports more logical (conventional?) workflows. It too is very creative, but in a more ‘considered’ fashion that might appeal to different users — indeed, if this is your first foray into the world of creative multi-effects plug-ins, it would probably be the more accessible starting point. Either way, both deliver great results. Whether you’re a pop producer or electronic musician looking to up your ear-candy quotient, or a media composer looking for a fresh sound-design tool, you could do a lot worse than power those ambitions with a little Rocket Fuel.

summary Rocket Fuel’s well-designed UI and workflow deliver powerful creative sound design in an accessible format.

$ $129 W www.lunaticaudio.com

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JOHN WALDEN

A

lchemy Of Pop is a great demonstration of just how good drum tracks created within EZdrummer (or Superior Drummer) can sound.

Recorded at Canterbury Music Company studio in Toronto by Hill Kourkoutis and played by stellar drummer Skye Polson, the EZX features four distinct drum kits, some additional snares and a complementary collection of electronic sounds (drum machine kicks and snare, claps, shakers, tambourines, other percussion and various sound effects). Sticks, brushes and mallets were all used, and a couple of the kits were captured with damping applied.

The end result is a broad sound palette with a hefty dollop of vintage vibe but that can also easily bring a modern edge.

Four On The Floor The four kits were chosen specifically to cover different sonic ground. Each was also given a name to reflect the producer’s thoughts on its primary musical application.

For example, the AER kit (made by Pearl in the 1960s), intended for alternative, edgy-indie and pop/rock, provides a big punchy sound. It sounds great dry but you can, of course, dial in more room or artificial ambience within EZdrummer as required. The sticks version sounds great, while the damped variant adds a more restrained option that would suit more subdued musical contexts. In contrast, the Terra kit (a Slingerland Radio King) is more suited to pop, folk, country and roots genres, with sticks, mallets and brushes options to deliver some beautifully subtle sounds. The AQUA kit (a Rogers Big R) is intended for modern or vintage R&B, soul and

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Toontrack Alchemy Of Pop EZX

EZdrummer 3/Superior Drummer 3 Expansion Toontrack’s latest EZX expansion brings classic and contemporary pop drums to EZdrummer and Superior Drummer. hip‑hop and is provided both open and damped. Finally, the IGNIS kit (a Ludwig Thermogloss from the 1970s) is more rock and blues orientated. The Valhalla preset — which is my absolute favourite from the whole EZX — is based on this kit. It features plenty of room sound with some interesting delay on the snare and it absolutely shrieks ‘character’.

It’s also a great example of just how well thought out and valuable the collection of additional acoustic/electronic sounds are. These are mapped to the additional nine-pad trigger represented within the main GUI. The additional percussions sounds are all very usable, but it was the option to layer electronic kicks and snares with the main kit’s acoustic options that I was most impressed by. It really does broaden the palette and allows you to add a modern edge to what are, in essence, classic vintage acoustic drum sounds. The collection of MIDI grooves supplied with the EZX, which are always a valuable part of these libraries, makes really good use of these additional sounds, ably demonstrating the extra dimension they can bring.

Conclusion The bottom line here is that Alchemy Of Pop EZX sounds wonderful. The acoustic kits sound great on their own and I’m sure the standard of these libraries is constantly improving as the big

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brains at Toontrack simply get better at what they do (and they are already very good at it) with each release. No, it perhaps would not be my choice for straight-up rock, heavy rock or metal, but for a broad range of pop, R&B, roots, country, indie and alternative styles — vintage or modern — Alchemy Of Pop is a class act.

summary For those working in modern or classic pop, acoustic singer/ songwriter or roots-style Americana, Alchemy Of Pop provides a wonderful array of drum sounds that are full of character.

$ $89 W www.toontrack.com

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ON TE ST

ACS Custom Emotion Dynamic In-ear Monitors

IEMs have transformed the on-stage experience for many performers, but they aren’t always ideal for bass players. Could a new hybrid design be the answer?

ACS Custom Emotion Dynamic $1200 pros • A fundamentally more familiar and satisfying IEM sound for bass players.

• Isolation level options.

• Ambient venting without LF loss.

• Acoustic imaging that seems to extend outside the head.

cons • Pro-level price bracket.

summary A set of IEMs designed to make bass players happier may actually turn out to be just as helpful to players of other instruments. Not cheap, especially in custom-moulded form, but the tech may soon be available in affordable universal models too.

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DAV E LO C K WO O D

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alanced-armature drivers are widely used for in-ear monitors (IEMs), being smaller and lighter than other types of drivers whilst offering a detailed, accurate sound, as well as a degree of efficiency — something that is particularly important in the battery-powered world of hearing aids for which this type of sub-miniature balanced armature was first developed. But balanced armatures work best when they are only asked to reproduce a specific range of frequencies, which means that for high-quality, full-range audio applications they generally have to be used in multiples, This introduces issues around crossovers, filters, integration and tuning. And, small as they are, there

is a limit to how many you can get inside something designed to fit in your ear!

The dynamic drivers also used in some IEMs, by contrast, are effectively just miniature loudspeakers. With the weight of a wire ‘voice coil’, albeit a small one, on the back of the diaphragm, they can’t be as ‘fast’ as a balanced armature, but given its larger surface area, a dynamic driver is able to ‘move more air’, which is something you generally want at low frequencies. The latter, of course, is something of a generalisation, as you can build a balanced armature specifically to ‘move more air’ by giving it a more powerful motor and a bigger diaphragm, but then it is no longer as small and efficient... Logically enough, balanced armatures and dynamic drivers have been used together, utilising

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ACS CUSTOM EMOTION DYNAMIC

the strengths of each technology, in conventional headphones for some time now, before inevitably spreading into the world of IEMs. It may sound simple enough, but combining two very different technologies within the tight confines of an IEM is actually anything but simple, and the fact that many IEMs use the same drivers and yet end up sounding significantly different perhaps tells us that there is more than a degree of expertise and artistry involved in their design and tuning.

IEMs & Bass Players Which brings us to the point of this review. I’ve been a custom-moulded-IEM user since 1997, with my first and all subsequent sets being made by one of the pioneers in this field, the British company ACS. I’ve had different sets over the years, as the technology has advanced but also because our ears don’t stay the same shape as we age, so it is necessary to have new scans taken from time to time to make sure that your custom moulds do actually achieve a proper seal against outside sound. My current pair is a multiple balanced armature set that I have been very happy with — at least until I was faced with the prospect of playing bass at a silent-stage venue. A few experiments soon revealed that whilst I felt I could hear the bass perfectly clearly in a mix, it didn’t ‘feel’ particularly comfortable or satisfying. More compression seemed to make it worse, and whilst EQ on individual monitor feeds is readily available in most digital mixers, the sort of EQ that might correct this issue is going to eat up headroom and potentially run into distortion. What I felt was missing, even when I was using high‑quality amp and speaker modelling, was the familiar sense of ‘weight’ and resonance in the bass sound.

IEM-equipped bass players have sought out a number of innovative solutions to this problem, including vibrating

platforms or seats, body-worn vibration packs and, more recently... hybrid IEMs. The latter combine dynamic drivers and balanced armatures, with the cone driver’s ability to move air making the difference in terms of the perceived weight or impact in the low frequencies.

Ultimate Ears, one of the leading USA-based IEM suppliers, have hybrid sets, and as I learned at the recent Sound On Sound GearFest UK show in London, so too do ACS. After verifying that my current models were still a perfect fit, ACS were able to use my existing ear scans to send me a custom-mould set of their new Emotion Dynamic hybrid in-ear monitors.

But there’s a further complication here, in that I’ve been using IEMs equipped with ACS’s passive ambient filters. These allow a controlled amount of outside sound to leak in to facilitate communication, without my having to use ambient mics or hastily remove custom moulds whenever anyone tries to talk to me in anything other than sign language. Having now had that facility, I would never want to be without it, but good low-frequency IEM performance is usually said to rely on achieving a good seal in the ear. Keeping this passive ambient filter working without potentially undermining the extra low-frequency punch gained by incorporating the dynamic drivers looked like it might be a real design challenge.

Depth & Resonance My new custom moulds arrived in clear, medical-grade silicone, allowing me to see the full intricacy of the internal construction. The dynamic driver, made to a custom specification for ACS, is rear-vented with a tuning filter and has its own bore, being joined along the way by the bore from the ambient filter. A second bore acts as the outlet for the two balanced armatures that take care of the mids and highs, augmented by an extra balanced-armature ‘super-tweeter’.

Specifications Frequency range: 10Hz-22kHz. Input sensitivity: 108dB SPL/mW @ 1kHz. Impedance: 28Ω @ 1kHz. Noise isolation: tuned vented port with 10, 17 or 26 dB reduction of ambient noise.

Drivers: Custom dynamic transducer, three Knowles balanced armatures (mid, high and super-high).

Casing: Lacquered 40-Shore medical‑grade silicone.

Cable: detachable 1.3m, with two-pin connection and memory ear hook.

Connector: 3.5mm jack with included 1/4‑inch adaptor.

All very impressive looking, but does it work as intended? Yes, I think it does.

The bass has the same basic tonality as it did before, but now there’s more of what I can only describe as resonance — the kind of cabinet-formant-tone resonance that is the difference between playing with a really clean DI and an actual speaker. It just feels a bit more like I’m playing bass through an amp behind me, even without any of the very specific physical sensation that brings. It is tempting to imagine that there might be some small amount of physical coupling from this design, but if there is, then it is nothing that I can discern without scientific vibration instrumentation. On balance, I think this extra depth is just being done with sound.

The very low end does seem like it is sensibly curated without being too conservative — the low ‘B’ on my Stingray 5 doesn’t threaten to detach my fillings, but there is still some heft to it all the way down.

So, job done on that count, but, of course, you don’t just listen to yourself in an in-ears mix: the rest of the balance has to sound right, too. And by ‘right’, I mean the sound that gives you what you need in a live-performance situation. I’m sure it is different in the world of IEMs for hi-fi listening, but

What Is A Balanced Armature?

The ‘armature’ bit is just a small metal strip or ‘reed’ that is pivoted in the centre to hold it in the middle of a coil of wire that sits within the field of a permanent magnet. We use the term ‘balanced’ because the magnetic force on the reed is precisely equal from both poles of the magnet. So, it just sits there, until you apply an alternating current to the coil, inducing a magnetic force in the armature that pushes and pulls it between the opposite poles of the permanent magnet, corresponding with the polarity of the electrical signal. So far, still no actual sound, of course: the sound we hear comes from

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a small, lightweight diaphragm, mounted within the same casing, driven into vibration by a physical connection from one end of the armature and vented to the outside world through a small hole.

Why bother making sound this way? Well, it can all be extremely small, as the coil only has to be big enough to induce movement in the tiny, lightweight armature, as opposed to the voice coil in a conventional speaker, where the coil has to move itself in the magnetic gap, in order to move the cone it is attached to.

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ACS CUSTOM EMOTION DYNAMIC

I find it is no great advantage for IEMs for live performance to be ‘flat’ in their frequency response. Having sound squirted into your ear canals at close range is an inherently unnatural situation anyway, and what really matters in my experience is that you can clearly hear transients rather than texture. The leading edges of things are where the timing is, and whilst it is common to get your own mix in your IEMs in the era of digital mixers, and probably your own overall EQ as well, you are not going to get individual sources EQ’ed just for you, so a bit of extra clarity in the presence region of your IEMs never goes amiss, and there’s plenty of that here.

Sensitivity in the Emotion Dynamics is a little lower than my previous all-balanced-armature set, but the nominal 28Ω impedance puts them very much in the easy-to‑drive zone. I had no problem getting more than enough level from a variety of headphone amps — an important consideration when sources are likely to be battery powered, as wireless belt packs need to be.

The standard ambient filter supplied gives 17dB of reduction, which I find to be just right, but you can opt for 10dB or indeed 26dB to better match the environment in which you will be working.

I’m not sure how much difference that

would make to the sound, but clearly these IEMs are a finely tuned overall system: I experimented with blocking the ambient filters whilst listening to playback, resulting in a much soggier bass, intruding on the lower midrange, and also a collapse of the nicely ‘out of the head’ soundfield into something much more just between the ears.

Supplied, as usual, with both carrying case and portable electronic dryer and sanitiser, these are cutting-edge custom

to give you a positive experience.

You must have enough isolation so you are actually hearing your IEMs; the mix in your ears must be the one you actually need to perform properly; if you are using wireless, that part of the system must be of decent quality and reliable; and finally, you have to have learned how to be comfortable with the very different experience of performing with IEMs. The actual sound quality of the units themselves shouldn’t really be the limiting factor in the outcome, but variations in the subjective quality of the sound do very much exist, sometimes even between units from different manufacturers using the same complement of drive units.

I recognise here that I have no exact equivalent point of comparison for the subjective aspects of this review, as I’ve never had hybrid custom moulds from another manufacturer — there is, of course, little incentive to go to the trouble of getting new scans from someone else unless you are unhappy with what you already have.

But I do have a lot of experience with other IEM models in universal form that I would contend lends validity to my conclusions. Leaving aside the initial goal of adding some apparent ‘weight’ for bass players, and surely drummers, too, these Emotion Dynamic hybrids are the most effective live‑sound monitoring IEM set I have so far experienced. ACS reference their patent-pending IRIS technology (Intelligent Resonance In Silicone), as a construction method that allows very precise acoustic tuning, with further fine-tuning in the form of their SSI Sound Stage Imaging. The latter, I think, may be using the ambient filters to achieve a wider soundfield extending outside the head. Whatever they’ve done, it results in not just great-sounding IEMs, but also, thanks to the openness of the ambient filters, something far less challenging for healthy long-term wearing and listening. Maybe silent-stage gigs aren’t going to be so bad after all!

“These Emotion Dynamic hybrids are the most effective live‑sound monitoring IEM set I have so far experienced.” moulds and by no means cheap, but it seems the ACS hybrid technology itself will very soon be available in their universal-fit range as well at a more affordable price point. Universals can work well for some people, depending on physical variables like the shape of your ears, and also on the amount of trouble you take to find exactly the right size and shape of tip to get a good fit.

Final Thoughts In real-world use, IEMs are always part of a whole system, every part of which must work reasonably well

Who Needs Ambient IEMs? There can be no doubt that the development and widespread use of in-ear monitors has revolutionised the live-performance experience of many musicians and sound engineers. For the latter, the elimination or significant reduction of wedges and side-fill speakers can clean up the front-of-house mix like nothing else, and for many performers, especially singers, moving to in-ears was the first time they had ever been able to really hear themselves properly on stage!

Using accurately fitted IEMs in both ears, so they achieve a good seal against outside noise, makes your working volume level entirely independent of your environment, whereas earbuds or ill-fitting universal IEMs will often exclude so little of the backline and ambient sound that you then need to have your in-ear signal uncomfortably loud to be able to hear it — a worse outcome than not using IEMs at all.

But there is no doubt that using a fully ‘isolating’ set of in-ear monitors can make you feel slightly disconnected from your audience and indeed your fellow performers. The latter is especially a problem in rehearsal, where what would once have been a few quick words can

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turn into a bit of a pantomime of lip-reading, gesticulation and reluctant IEM removal and re-fitting. That is, unless everyone maintains the discipline of only speaking one at a time, and into a microphone that they know is in everyone else’s IEM mix... and that never happens.

There are other drawbacks to fully sealed IEMs, moreover, especially for singers, which you can experience for yourself just by speaking or singing with your fingers in your ears — you’ll hear a kind of droning head resonance know as ‘occlusion effect’. Also, sealing off your ears from normal air flow for long periods of time is also not particularly healthy, as they will get both hot and moist.

The precise workings of ACS’s patented ‘tuned vented port’ designs are somewhat beyond my grasp of the physics of miniaturised acoustic systems, but I can say that they work spectacularly well in my experience.

Instinctively, the expected trade-off is some kind of compromised bass response, but the incorporation of the venting in the new ambient hybrids with their impressive bass response seems to prove that doesn’t have to be the case.

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ON TE ST

F IE DLE R AU DIO M A S T E RING CONS OLE & GR AV ITA S M DS

Fiedler Audio Mastering Console & gravitas MDS

Dolby Atmos Mastering Software & Processor This combination of software and multi‑channel plug-in overcomes some frustrating hurdles for those working in Atmos.

PHIL WARD

B

ack in SOS September 2023, I reviewed Fiedler Audio’s unique and innovative Dolby Atmos Composer plug-in suite (https://sosm. ag/DolbyAtmosComposer), and now, with the launch of their Dolby Atmos Mastering Console and associated gravitas MDS module, Fiedler have again come up with something that can’t easily be found anywhere else in the Dolby Atmos landscape. Before I describe what Fiedler’s Mastering Console and gravitas MDS plug-in can do, it might help if I spend a paragraph or two describing what a typical Dolby Atmos workflow can’t do — or at least what it struggles with.

Atmos Challenges With stereo mix work, we take it for granted that, at the end of the DAW,

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there’s always the stereo master bus on which we can apply some global EQ, compression, limiting or whatever takes our fancy, to give a final polish to our mix (even though we know we probably ought to be leaving at least some of that to a separate mastering stage!). Things are very different in the Atmos world. Firstly, an Atmos mix typically comprises beds and objects. Beds are conventionally panned elements spread, typically, over a 7.1.2 channel width, and objects are mono elements (or stereo elements effectively treated as two independent mono signals) that are panned according to dynamic metadata that defines their intended instantaneous X, Y and Z position in space. Notwithstanding that multi-channel EQ, compressor or limiter plug-ins that can be inserted on a 7.1.2 bed bus are somewhat niche, even if you had and used one, the plug-in wouldn’t ‘see’ the object channels because they’re separately routed to the Dolby Atmos Renderer (the discrete application that compiles the beds, objects and metadata into a Dolby Atmos output package called an ADM). So you can’t easily apply an output bus compressor or EQ to an Atmos mix. This is where Fiedler Audio’s Dolby

Screen 1: The Mastering Console Options window.

Atmos Mastering Console and associated gravitas plug-in come in...

A Dolby Atmos ADM file is actually a multi-channel WAV package that, when decoded by an Atmos-compatible output device, is ‘intelligently’ rendered to the available speakers. Those speakers might be in a consumer Atmos home theatre system or a professional Atmos mix or mastering system. The beds are rendered using conventional panning, whereas the objects are rendered using

Fiedler Audio Mastering Console & gravitas MDS From $99 pros • Uniquely useful.

• Versatile.

• Hugely configurable.

cons • None.

summary Mastering Console and gravitas MDS are really well conceived and valuable contributions to the Atmos landscape, and definitely worth a look if Atmos is your thing.

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F IE DLE R AU DIO M A S T E RING CONS OLE & GR AV ITA S M DS

Screen 2: The Mastering Console Monitoring window.

the embedded positional metadata.

In an Atmos mix or mastering studio environment, the Dolby Atmos Renderer application is not only able dynamically to render an Atmos mix coming in from the attached DAW, but it can also play out a previously rendered ADM package — and, essentially, that’s what the Fiedler Dolby Atmos Mastering Console does too. You load an ADM package into the Mastering Console, specify a sound output device, press Play, and the ADM plays over your Atmos monitoring or via a binaural re-render.

The Mastering Console

a sound output device. Many of the preferences in the Options window define how a mastered Atmos package will be downmixed if, as is very likely in the consumer space, it comes across a non-Atmos playback system. The Lo/ Ro (Left Only/Right Only) preferences, for example, refer to selection of the downmix algorithms that help optimise the stereo and mono compatibility of Atmos material that’s been downmixed to 5.1 (Dolby Pro Logic). The preferences of the Options window are pretty much a copy of those found in the Dolby Atmos Renderer app, so it may well be that when you open an ADM package they will already have been configured appropriately. The Options window also provides an Export function that enables the mastered ADM to be exported (to result, obviously, in several million Apple Music streams).

Moving on to the Monitoring window, illustrated in Screen 2, this is where the monitoring hardware and format is configured so that, on playback, Mastering Console renders the ADM package appropriately for the Atmos monitoring that’s present. Or, if you want to experience the Atmos package on an alternative speaker array, it can downmix here as needed. If you have 5.1.4 Atmos monitoring, for example, you also inherently have Pro Logic 5.1 monitoring — selecting that option in the Monitoring window will downmix the monitoring output appropriately (using the parameters defined in the Options window). The Monitoring window also provides access to an Audio Device Setup window, illustrated in Screen 3,

On opening Mastering Console, you’re presented with an empty window that asks that you import an ADM package.

So, to get started properly, you click on Import and browse for an appropriate file (one you’ve previously exported/ bounced from Dolby Atmos Renderer or Fiedler’s Dolby Atmos Composer). With an ADM package loaded, the Mastering Console springs to life with its Options window, shown in Screen 1.

Fundamentally, the Mastering Console comprises four windows and I’ll cover the Options window first. As suggested by its name, this window enables a range of mastering preferences and tweaks to be applied to the imported ADM package and, like all the Mastering Console windows, it includes transport controls that enable the file to be played through Screen 3: The Mastering Console Audio Device Setup window.

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that enables the monitoring output routing to be defined and a stereo or binaural downmix output to be configured, with or without HRTF (head related transfer function) data applied, and bussed to specified channels of the output interface.

The Audio Device Setup window also enables EQ and delay to be configured and applied to each live output channel.

Finally, the Monitoring window displays the activity of bed and object channels and enables them to be muted or soloed: a useful check to ensure that what you think you’ve defined as beds or objects have been so in reality.

The Channel Configuration window (Screen 4) enables binaural downmix settings to be adjusted individually for the beds and objects. The Dolby Atmos binaural render algorithm allows ‘Mid’, ‘Near’ or ‘Far’ parameters to be assigned individually to bed and object channels, and these parameters are pretty significant in defining how the binaural render will work on headphones. As you might expect, the Mid, Near and Far options define how distant an object or bed channel will appear to be from the listener. I’ve found the distance options to be really useful for adjusting how an Atmos mix presents when rendered binaurally, and although they can often be defined at the mix stage, they are I think the kind of thing that is quite useful to revisit with the kind of perspective that a mastering process tends to offer.

gravitas MDS So far, you might think that Mastering Console doesn’t really offer much that isn’t available from the Dolby Atmos

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F IE DLE R AU DIO M A S T E RING CONS OLE & GR AV ITA S M DS

Gravitas is a hugely configurable and versatile dynamics processor that demands quite some understanding, effort and commitment if you’re to become confident and familiar with it.

So it came as some relief to me that its range of factory presets offer some really effective starting points! I really liked, for example, what the ‘Atmos — Organic Glue’ preset did to an Atmos mix of a live eight-piece band recording I’m working on. To my ears it did exactly the job we expect of a touch of mastering dynamics processing, pulling all the mix elements together and helping them sit together somehow more naturally.

Screen 4: The Mastering Console Channel Configuration window.

Screen 5: The Mastering Console Master Channel window.

And Finally...

Finally, before I wrap up, there’s one Renderer. But its final Master Channel more really useful element of the signal, and subsequently processes it. window (Screen 5) sets it apart. In fact, as Mastering Console Master Channel There’s more sophistication on offer in far as I’m aware, it makes it unique. This window: volume analysis. Just as it is the detector section than it’s worth me is because the Master Channel window with stereo mix and mastering work, explaining here, but in general terms enables processing modules — plug-ins LUFS volume targets are a thing in it offers band-pass detector filtering, effectively — to be inserted in the Atmos world, but the the Mastering Console and to lack of an Atmos master process the ADM export. mix bus makes hitting The first available module is targets somewhat more gravitas MDS, a multi-channel problematic. The combination, dynamics processor, but Fiedler however, of volume analysis say an EQ module is on the in the Mastering Console way too. And when I write ‘multi-channel’, and the gravitas module with its detector timing (Attack, Hold, Release, remember that an Atmos ADM package make-up gain control literally makes Rise Time, Look Ahead and Detector can include up to 128 objects along with that problem go away. I’d have the Gain), a 3DET mode (this offers three perhaps multiple 7.1.2 (probably) beds — Mastering Console and gravitas MDS independently configurable detectors, it’s multi-channel, and then some! Screen for that capability alone. an idea Fiedler say they borrowed 6 shows that gravitas is fundamentally from an Amek mastering compressor), a dynamics processing plug-in that offers a running gain display showing the $ Mastering Console $99. all the configuration and variables you’d effect of signal processing, and gravitas MDS $199. expect of such a beast, plus a few extras a configurable channel-linking control.

W https://fiedler-audio.com aimed at Atmos mastering.

The top line of gravitas’ configuration controls offers preset selection, redo and undo, A/B comparison and selectable parameter linking. The middle section offers the functionality you’d expect of a dynamics processor plus an optional and adjustable (slopes and bandwidth) band‑pass filter to define the band over which the processor operates, and an option independently to adjust the make-up gain of the LFE channel. If the processor is working just in stereo, a Mid-Sides mode is also provided.

The bottom section of gravitas provides an expanded range of signal-detection control that allows the user to refine how gravitas responds to the incoming Screen 6: Gravitas is the first in a promised range of Atmos-capable processors for Mastering Console.

“I’ve found the distance options to be really useful for adjusting how an Atmos mix presents when rendered binaurally.”

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December 2024 / www.soundonsound.com

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Warm Audio Pedal76

Compressor Pedal This pedal-format compressor doubles up as a serious studio tool.

DAV E LO C K WO O D

P

edal-style versions of UREI’s classic 1176 solid-state peak limiter have proved popular with gigging guitarists since Origin Effects’ original Cali 76 model first made the FET-based circuit available in a compact, footswitchable form. Many others have followed, not least Origin themselves, with ever more compact versions, and, of course, Universal Audio (who still make the 1176 rackmount hardware) with their own digital emulations in pedal form. They’re joined now by Warm Audio, with an all-analogue, relatively large-format, footswitchable unit designated the Pedal76 Limiting Amplifier.

Overview Of course, Warm Audio are no strangers to the 1176-style circuit, having produced one of the most popular reproductions over the last decade in the form of their highly regarded WA‑76 rackmount compressor, and they’ve recently announced more 1176-inspired outboard too. The Pedal76 is said to utilise the same high-grade components and custom Cinemag output transformer to offer the same level of audio performance in a ‘pedalboard-friendly format’. It is actually more compact than the original Cali76 model, but it would still use up a bit of space on a smaller board, with dimensions of 160 x 115mm, and a casing height of 58mm, or 78mm to the top of the control knobs — it wouldn’t actually fit under the lid of my gigging pedalboard. Ironically, however, the very characteristics that make it perhaps a slightly awkward pedalboard companion make it a great desktop recording unit. The larger dimensions allow

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for larger control knobs somewhat in the style of the original hardware, and with some size differentiation of function that just make it easy to use for creative on-the-fly adjustments, whilst still being small enough to keep close at hand, especially compared to something that you have to rackmount.

The control line-up is the usual Input, Output, Attack and Release setup, but with a rotary control for Ratio rather than the switch bank of the original design. The Ratio control has the expected 4, 8, 12 and 20:1 options, with further settings at either extreme that I take to be 1:1 or Off (but still passing signal) at fully anticlockwise, and at the other end perhaps something to replicate the infamous ‘all-buttons-in’ setting? Although not identified as such, it certainly seems to exhibit something of the same program-dependent distortion, transient punch and idiosyncratic two-stage release of the ‘nuke’ setting. The Attack and Release controls have Fast and Slow idents rather than the arbitrary numbers of the original hardware, which should help those less familiar with 1176s to remember that the controls give you faster operation as you turn the controls up (clockwise) as opposed to setting a longer attack or release time by turning them up.

Dominating the front panel is an illuminated VU meter for gain reduction.

I can, if necessary, get by with just an LED ladder, or even a single LED that changes colour to indicate GR, but in my experience there is nothing that conveys what a compressor is doing — when and how much it is working — quite so well as a real mechanical meter deflecting left from the 0VU position. Full marks to Warm Audio for including one in a pedal version. The

illumination goes out to indicate that the pedal is bypassed via the footswitch, using either buffered or true mechanical bypass, switchable from the rear panel.

Other options on the back include an input sensitivity adjustment, slightly inconveniently rendered as a recessed, screwdriver-adjustable trim pot, plus mic or line output level and high and low drive settings to determine how much you want to saturate the transformer.

Instrument input and amplifier output are on unbalanced jacks, with an additional balanced TRS output jack with switchable ground lift also available. Input impedance is 1.1MΩ, which will help preserve the tone of directly connected instruments with passive pickups.

There’s the possibility of powering the Pedal76 from an internal 9V battery if you need to, but it’s important to be aware that you won’t be getting the full performance from the unit if you do (I should note that

Warm Audio Pedal76 $269 pros • A great-sounding 1176-style circuit.

• Cinemag output transformer.

• Can be a clean comp, a ‘dirty’ DI or anything in between.

cons • A bit large for some pedalboards?

summary This analogue, transformer-balanced, single-channel FET compressor is footswitchable but also ideal for desktop use. With all the expected sonic characteristics of the classic 1176 circuit, it loses nothing in comparison to dedicated studio units.

Warm Audio are very clear that this is the case). Using an external DC supply (minimum current capacity 125mA) automatically activates a DC-DC voltage converter upping the internal supply to 18V, thereby increasing headroom — highly recommended when working at line level for recording or with high-output active pickups on an instrument.

Sound Opinions When the 1176 was launched in 1967, using FETs in place of the tube-based circuitry of its predecessor model 176, it was notable for its very fast attack and release times, and the fact that the latter were both to some extent program dependent.

The circuit’s threshold value is nominally ‘fixed’, with the input gain determining how far above it you want to push the signal, although, in reality, the threshold changes a little with the ratio setting. The result of all these program and setting interdependencies is that it is quite hard to make an 1176 sound ‘wrong’ to any significant degree, with the subjective impression of transients somehow always being preserved. Even using the fastest attack and release will just distort a bass in a musically useful way or give you a nice aggressive edge to a vocal. Although always characterful, an 1176 is not really an ‘effect’ compressor, which is perhaps why people sometimes resorted to using the ‘all-buttons-in’ setting never envisaged by the designer!

Warm Audio have given us a very nice-sounding rendition of the classic 1176 circuitry, with enough level flexibility to operate as a pedal for a guitar player, a clean or saturated compressor/DI for a bass guitar, or indeed an insert effect for a vocal or instrument in a recording setup. The unit has very low self-noise even with the sensitivity jacked up whilst offering fine-resolution control over the amount of saturation and character generated within. There’s no onboard facility for mixing in a parallel ‘dry’ path, but as an all-analogue unit, that is not too hard to rig for yourself using a simple signal split. In common with the ‘real thing’, there’s also no side-chain access, which is understandable given the primary applications envisaged.

Priced competitively with other premium pedal compressors and with an audio performance to match expectations, I think the Pedal76 has something a little extra to offer with its format — I have rack compressors, mostly just out of reach when I need them, and pedal compressors that really belong under my feet, whilst this just might be the perfect ‘in-between’ to sit on the desktop. I think this one has $ $269.

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ON TE ST

PWM Mantis

Analogue/Digital Synthesizer

PWM’s sophomore outing is an intriguing combination of old and new.

WILLIAM STOKES

I

must say, I am a fan of how Paul Whittington’s still-nascent company PWM seems thus far to be characterised by the decision to celebrate, collaborate and co-create. Their output to date comprises two instruments that aim to push things forward while clearly endeavouring to preserve some classic design philosophies. In the case of PWM’s debut, the Malevolent, this involved collaborating with Bristol’s most excellent Future Sound Systems, and if I remember correctly the term ‘Korg MS-20’ was thrown about liberally as a key inspiration.

In the case of the sophomore Mantis, it’s the late, great Chris Huggett on the ticket, the man responsible for icons like the Wasp and the OSCar, with a number of very intriguing design features to boot.

A little history to help explain the Mantis: before Huggett’s tragic passing in 2020, he and Paul Whittington had worked together, namely designing numerous instruments for Novation such as the MiniNova, Bass Station II and Peak. The Mantis, which would be Huggett’s last synth project, incorporates the oscillator design of the Wasp and unites it with the filter design of the OSCar, doubling both into a fully duophonic instrument. It draws from the worlds of digital and analogue, as well as from the East and West Coast schools, to create something that’s at once a little bit classic, quite unique and phenomenally versatile. A good microcosm of this is its envelope design.

These are based around a conventional East Coast design with attack, decay, sustain and release, but as well as adding an extra ‘sustain fall’ stage, they can be set to cycle, like a Buchla-esque function generator. Envelope 1 is normalled to the main VCA, though both are assignable to up to six modulation destinations.

More on that later.

Player Ready In my review of the Malevolent back in 2022, I took umbrage with its mini keyboard. Side note: unless they are truly miniature, á la the OP-1, I think mini keyboards are, generally speaking, a waste of time. More often than not

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they simply add expense and take up space, all while those who do need keys are likely to connect a bigger and more playable keyboard anyway.

The Mantis, thankfully, has 37 lovely full-size, semi-weighted keys (that’s three octaves), and a pleasant-feeling keybed. Velocity, aftertouch, it’s all here.

Wonderful. Similar to the Malevolent, the Mantis’ side panels snap on and off magnetically; something I’m still waiting to see a good reason for but which I hope means we’ll see some nice accessories in due course. Considering the weight of the Mantis, these can detach at any moment when picking the thing up, which is a bit of a hazard. So be sure to lift from the bottom.

On the subject of those mysterious side panels, this is also as good a moment as any to point out that at the time of writing PWM have not published a full manual for the Mantis, nor have they still for the Malevolent. Much as I love the precocious appetite for moving onto new designs, I do — ahem — look forward to seeing both of those soon. These are powerful and intricate instruments after all, and aside from the user experience, Whittington and team deserve to show off their work right down to the last Shift function. As of now, I’d imagine there are many complaining about there being too much guesswork, and I can’t but sympathise. Suffice to say, it feels like this issue is somewhat endemic to the industry in 2024: with websites revisable at any time and firmware updates available instantly, far too many units seem to be hitting shelves before all ducks are in a row, to the point that it’s almost as if many of them are still in beta.

The Mantis’ hardware, however, is very much ready, with substantial build quality that I’ll say is becoming of its price. As I’ve mentioned, it is a weighty synth — around five kilos — with a matte‑black metal and plastic enclosure, firm rubberised buttons and sturdy knobs. These do wobble a little in use, I should point out, but not to a hugely concerning degree. They also behave rather like encoders, in that when turning a knob quickly there’s an ever-so-slight lag perceivable between its physical movement and that of its parameter. Joyfully, there’s no screen to speak of, and little more than one layer of panel functionality. Tick.

Considering how much is here, that’s quite an achievement. Alongside a robust pair of synth voices, with a sub‑oscillator to boot, there’s an onboard ring modulator, a noise generator with adjustable density, a host of expression and modulation options centred around two LFOs, and an arpeggiator with adjustable bpm and tap tempo, clock division, swing, direction and octave range. The filter section boasts an impressive eight filter modes along

PWM Mantis $1349 pros • Great WYSIWYG panel design.

• Replete with ‘bonus’ features.

• Excellent‑sounding signal path combining digital and analogue.

cons • Purpose of the side panels is still mysterious...

• Tantalisingly close to full four-voice polyphony, which can feel frustrating.

summary The Mantis has its idiosyncrasies but is without doubt a lovingly designed, highly powerful and brilliant-sounding synth.

steps if you accidentally overwrite something. A lot of thought has gone into the design of the Mantis, and it shows.

Singing Voices The Mantis’ musical architecture begins with its two DCOs, both offering sine, triangle, sawtooth and square waves — which all sound quite excellent and leave little to be desired. There are also two other, more curious options. The first of these is symbolised on the panel as a trio of organ pipes, and really does sound like a church organ, spreading what sound like subtly squared-off sine waves spread across an octave or two. This spread is psychoacoustically rather interesting; ascend a scale on the keyboard and it’ll feel like you’ve arrived where you started, not unlike a Shepard tone. Of course, add some modulation and filtering into the mix and not only are lush, Hammond-esque sounds readily available, this sound can be squashed and twisted in numerous different ways to make it, well, not organ-like at all. The second ‘extra wave’ option is labelled WT, and this takes the form of a fairly simple wavetable moving through seven or so

“A synth with buckets of character, it requires a particular approach and certainly rewards the time invested.”

At the back of the Mantis we find a power switch, USB‑C port, MIDI in, out and thru sockets, and a selection of quarter-inch jack sockets for pedals, audio output and headphones.

with adjustable pre-filter drive, and to cap it all off there are two digital bus effects on offer: reverb and mode-adjustable chorus, which can be saved as part of a preset. On the subject of presets, the Mantis has a capacious 200 slots, and these are all instantly accessible, menu-free, via two elegant columns of buttons laid out as two banks of 10 groups, each storing 10 patches. Simply hit bank A or B (the former containing factory presets that can be protected or overwritten as desired), hit a ‘10’ button and then a number. Done. Saving is equally quick and intuitive, and there’s even a handy Undo function for retracing one’s steps if needed — including save

www.soundonsound.com / December 2024

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PWM MANTIS

The Mantis measures 620 x 320 x 110mm and weighs 5.3kg.

different harmonically rich shapes. As of now, user wavetables can’t be imported to the Mantis, but I didn’t find myself missing this: the pre‑loaded wavetable posits itself more as a ‘bonus’ option that works beautifully with other aspects of the Mantis’ architecture, and can achieve great results very quickly indeed.

The Shape knob has been deployed to good effect for all of the above. Since the oscillators are digital, there are options for it to influence wave behaviour beyond the conventional pulse width for the square wave. For the wavetable, for instance, it moves through waveshapes.

For sine and organ, Shape adds negative or positive harmonics into the picture (de facto drawbars in the case of the organ — something Paul Whittington has described as another nod to the OSCar), resulting in a pleasing range of ‘cheat’ FM sounds. It’s another example of how the Mantis really does optimise its real estate, as well as why DCO sceptics should think twice before discounting it in favour of analogue for analogue’s sake.

Oscillator 2 has all of the above, plus one particular additional feature that ends up considerably expanding the scope of the Mantis’ overall voice.

Density and Density Rate knobs pertain to what is essentially a third oscillator, in that oscillator 2 is in fact two identical oscillators working together. Dialling up the Density alters both the mix and frequency of the third, ‘hidden’ oscillator,

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while Density Rate pertains to the maximum amount of detuning. It means that those detuned, super‑saw sounds can be achieved while still miraculously leaving oscillator 1 to take on other duties — or, if you wish, can be used with oscillator 2 to create truly massive super-super‑saw sounds. Or super-super-any-wave sounds, for that matter.

Two Plus Two As mentioned, the Mantis’ dual analogue filters offer a huge amount of flexibility. I was expecting them to sound good (they’re descended from one of the most famous filter designs around, after all) and I wasn’t disappointed.

Musical and decently throaty at higher resonance settings, they offer low-pass, band-pass and high-pass options, (generously available with 12dB or 24dB/octave slopes, depending on the selected mode), and once again two extra modes that make creative use of their configuration. Wide Band places the two filters in series and renders one high-pass and one low-pass. With the Width knob expanding or contracting the resulting bandwidth, it makes for one very flexible tone-shaping tool. Wide Notch, on the other hand, places the two in parallel and swaps them around to create a notch filter, which is once again adjustable with the Width control.

This is the point at which the Mantis’ voice structure gets slightly

convoluted. At its most basic level it is indeed a fully-fledged duophonic synth, with two discrete voice paths with their own VCAs and VCFs, each running their respective pair of oscillators (or, if we’re splitting hairs, four oscillators including the sub and DCO 2’s Density setting). While it’s not possible to stack everything into one super-unison voice, it is possible to further spread things across four voices in a Quad mode. Since the Mantis only has two filters, it essentially takes on a ‘dual-paraphonic’ structure when you play with four fingers, with the filters applied only to the first two notes played. But it’s not really paraphonic, strictly speaking: it would be if the two oscillators per voice were separated and split, but they’re not: in fact they’re replicated, just as they would be in a four-voice polyphonic synth, with each note given its own VCA envelope. So in practice, that’s two filters but four VCA paths, so the Mantis can feel like a proper four-voice polysynth... so long as you don’t require the filter envelopes too much. If you want to get the most from its filters, you’re better off sticking to monophonic or duophonic modes.

“Why not just give it four filters and make it four-voice, then?” I hear you ask. It’s a good question, and one I’m not sure of the answer to. What I can tell you is that the Mantis’ design did not set out to be polyphonic, so once again

this is essentially a very handy bonus feature, though one that shouldn’t be relied upon for the backbone of its sound. I don’t mind this. In my opinion a little limitation is healthy, even if it does feel like polyphony is being dangled tantalisingly close. Unlike many polysynths out there, it seems to me the Mantis is inviting — requiring, even — its users to think carefully about how they’re designing their sounds, how they’re optimising its available features and what they’re choosing to prioritise. For that I can’t in good conscience disparage it.

Extra, Extra Beyond its fundamentals, the Mantis has a number of extra tools and little bonus features that significantly augment its design. Its modulation matrix is very well thought out, with three simple sets of six buttons for Control (that is, destination), Source and Scale. These can be used to compound a control source, like aftertouch or an external expression pedal, with a modulation source such as an LFO. It’s visual, quick and creative;

while six modulation destinations sounds slim, in practice it feels like plenty since there are numerous alternative means of generating movement and character beyond these ‘usual’ means of modulation routing.

There’s an Osc Drift knob, for instance, which imparts some pleasantly tape‑flutter-like instability into the DCOs. The LFOs each have a bipolar Fade knob for having the wave grow or decay with each note, essentially replacing the need to route an envelope to them.

The envelopes have adjustable velocity response and, as mentioned, add an extra stage onto the conventional ADSR stages: sustain fall, which shapes the movement of a sound while the gate remains high, and can be set to cycle. As on the Malevolent, there’s a very useful Hold function for latching a drone, engaged by simply flicking the joystick down. The onboard effects are well chosen, with four different algorithms of lush, wide-sounding chorus choose from (in addition to the stereo Voice Spread knob) and reverb.

The Mantis’ Schroeder-type reverb is nice and flexible, with its own filtering and time control, though frustratingly it’s in mono, so in many ways dialling it up results in the Mantis’ output getting narrower and smaller, not bigger and wider.

Conclusion If you’re looking for a polyphonic all-rounder for two-handed keys playing — even in a four-voice capacity — I daresay the Mantis isn’t for you.

If, however, you’re hunting down an extraordinarily versatile sound-designready synth, with chordal capabilities if needed, I recommend it very highly. Much like the Malevolent, the Mantis is not an instrument that will simply slot into your setup as part of the furniture, and nor is it trying to be — quite the opposite, in fact. A synth with buckets of character, it requires a particular approach and certainly rewards the time invested. $ $1349 W www.1pwm.com

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Tascam Model 2400

Mixer, Multitrack Recorder & Audio Interface Tascam’s new recording mixer is the most flexible Model‑series console yet.

M AT T H O U G H TO N

I

reviewed Tascam’s Model 24 in SOS February 2019 (https://sosm.ag/ tascam-model-24) and the smaller Model 12 in SOS December 2020 (https:// sosm.ag/tascam-model-12). With new and improved features, the latter addressed many of the issues I raised when assessing the Model 24, and the Model 16 then added more channels. Now, Tascam have released the flagship mixer in this range: the Model 2400 includes almost all the Model 12’s improvements and introduces

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more features. Larger and heavier than the Model 24, it has more channels and more analogue I/O, but it remains portable enough that it’s easy to stick in the car.

Overview The Model 2400 is, at heart, an analogue mixer, with one-knob compression on 12 of the 22 input channels, and a three-band EQ on all of them. There are four stereo subgroups, five aux sends (three pre-fader, one post-fader, one switchable pre/post), a stereo digital effects processor to which you can send from any track, and a separate

stereo digital EQ and compressor for the master bus. There are insert points for 12 input channels and the master bus, and an array of other facilities that I’ll cover below. But what makes the Model 2400 different from most mixers is its 24/22 (in/ out) channels of 16‑/24-bit and 44.1/48 kHz A-D/D-A conversion. This makes possible recording standalone to an SDXC card (up to 512GB) or, over USB, to Mac, Windows or iOS/iPadOS devices.

For Mac and iOS you don’t need drivers for USB interfacing, but there’s a useful ModelMixer app whose main purpose is to

Click Tracks The onboard click can be routed to the control room and headphones outputs, or to the dedicated click output. It can’t be routed directly to the aux outputs, which surprised me, but you can plug a jack patch cable from the click out into an input channel, and from there can route the sound to the auxes. As with the delay effect, the click supports tap tempo, accessed either from a button on the top panel or using an attached momentary footswitch. As described in the main text, there are various time signatures to choose from, but I didn’t find a way to change time signature or (other than tap) the tempo within a song. For such projects, though, you could always prepare a session in your DAW, render the click or more detailed backing track, and import that to the Model 2400.

provide more extensive metering than the hardware. Windows users also have this app but need the ASIO/WDM driver; both these drivers can access the hardware, so two apps using the same sample rate can communicate with the mixer simultaneously.

With five-pin DIN MIDI out and in sockets on the rear, the Model 2400 can act as a USB MIDI interface. It can also output MIDI Time Code and Clock. The USB connection also allows you to transfer files to and from the SD card, to back up data or load stored projects.

Recording

Tascam Model 2400 $1999 pros • Sounds decent.

• More inserts, more subgroups, more auxes, more outputs...

• Can record pre- or post-processing.

• Click track makes computerless recording a breeze.

cons • No polarity inversion.

summary Tascam’s impressive Model series of recording mixers continues to evolve, and this flagship model offers more inserts, more subgroups, more auxes and, thus, many more potential applications than any of the other mixers in the range.

Recording standalone or to a computer, the 24 inputs are for the mixer’s 22 input channels plus the stereo mix, while the 22 outputs are routed back to the main input channels, just before their insert point (or for those without one, the compressor). As with the Model 12 and 16 (not the 24), a button above the channel compressor/EQ sets the point in the signal path sent to the A-D converter: before or after the processing stages, by which I mean anything patched into the insert, as well as the onboard compressor and EQ. A simple touch, but it makes the mixer more versatile than the Model 24: you could record a clean signal post the preamps and HPF, while using the inserts, compressors and EQs for a cue mix or live performance; or use the processing and commit what you hear to ‘tape’.

Whichever’s the case, you can break a mix back out onto the mixer and use the insert

and processing stages again. This welcome A-D routing flexibility is as on the Model 12, but many more input channels have an insert point (12, compared with the Model 12 and 16’s two), so integrating outboard is a much more realistic proposition.

The standalone recording process is also pretty much as on the Model 12. You can create a song, set up a click track, and change the tempo and time signature. Each track has a record-arm button so you can record anything from a single track to all of them. Some more advanced functions make good use of the screen, the Multi Jog push/ twist encoder and buttons, such as manual or automatic punching in and markers — if you fluff a note or two, you can flag that up, or sing along and drop in a better part rather than retake the whole performance or rely on edits and pitch-shifting later on.

Since you can also import files from the SD card, you can prepare backing tracks on a computer for use on standalone sessions (or as live backing tracks). Computers make so much possible, but when I don’t have to look at a screen while making music I prefer not to. And the simple fact of having so many tracks with inserts available makes working this way much more attractive.

Still, some prefer to use a DAW, and as well as being an interface the Model 2400 can serve as a basic HUI or MCU controller. For example, the mixer’s transport buttons control your DAW transport, the Multi Jog becomes a scrub wheel, you can create and navigate to markers, and the 16 record‑arm buttons arm your DAW tracks. Notably, there’s no fader control, but you can still use all the channels of the USB audio and MIDI interface, and there’s enough control to be useful. Tascam have created dedicated profiles for various DAWs, and this is detailed in the DAW Control PDF manual on Tascam’s website.

Bus Processing & Effects I’ve mentioned insert points already and, importantly, it’s not just the input channels

The Model 2400 has far greater analogue connectivity than previous models in the series, with dedicated subgroup and aux outputs, master-bus insert points on the rear, and 12 input channels featuring insert points on the top panel.

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The built‑in master bus compressor ranges from gentle compression to, effectively, limiting and has dedicated physical controls — but the master bus also now has an insert point, so if you already have a favourite bus compressor you can use it.

that have them. Unlike previous Model mixers the 2400 also features them on the master stereo bus; another simple touch that constitutes a big improvement. The four stereo subgroups don’t have inserts, which I’d have liked (even if it increased the cost), but they do have direct outputs on the rear, which is another new feature.

If you have unused input channels, you could, say, feed a drum subgroup’s outputs to a compressor (making sure the group isn’t routed to the mix bus), and route the compressor outputs back to the desk, routing that channel only to the master bus.

You don’t have to rely on outboard gear for master-bus processing, though. You also have the built-in digital stereo compressor and EQ. The compressor has physical controls above the screen. A button switches this master processing in/out of the signal path, and there are dedicated knobs to adjust the threshold, tweak the ratio (1:1 to limiting), set the attack (2-200 milliseconds) and release (10 milliseconds to 1 second), and apply up to 20dB of make‑up gain. You can set the compressor

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by ear, but to see what’s going on in terms of levels and gain reduction, you need to access the compressor window on the screen.

Thankfully, menu-diving on the Model mixers using the function buttons and Multi Jog wheel is quick and easy. The digital EQ offers you four ±12dB bands. The outer two are shelves and the inner pair parametric. As it’s all controlled using the Multi Jog and function buttons, this is only tweakable when the EQ screen is active.

Both the master bus insert point and the digital processor come, by default, after the signal splits to feed the headphone and control room outputs. Depending on what you’re doing, that could well make sense: if you want to apply EQ to the main output to compensate for a venue’s acoustics, you can hear the result on the PA and won’t want to hear it in your headphones. But in the studio you won’t hear the effect of that nice mix-bus compressor you patched in — to hear that you need to engage the master bus’ AFL (after-fade listen) button.

The digital send effects engine is, as far as I can discern without having the units side by side in front of me, as it is on earlier Models. You have the choice of workmanlike (decent-sounding but nothing special) reverbs and delays, a chorus, a flanger and some combination effects that pair different hall reverbs with delay or chorus. Again, these are all editable using the screen, Multi Jog and function buttons, and while you don’t get too many parameters to play with, I usually had enough control to set up ‘comfort’ reverbs and delay effects and would be happy using them for live sound, not least because the delays can follow tap tempo (using the top-panel button or an attached footswitch). The output of this effects engine can be assigned to the main mix, aux outputs 1+2 or 3+4, or any of the subgroups.

If you want more effects or would like to plumb better ones in, you can use the aux sends; these are also as on previous models

but as you have more of them, mixing with several reverbs/delays is possible. All five auxes have global level controls and mute/ AFL buttons. Aux 5 defaults to feeding the internal effects engine but also has a physical output.

Other than for the internal effects engine, there are no dedicated aux returns, so you’ll need to reserve input channels for that.

Input Channels

Input channel 1, one of two to have a high-impedance instrument input option, is otherwise identical to the next 11 channel strips.

Speaking of input channels, the first 12 are almost identical, except that as well as accepting mic and line sources plugged into the combo XLR/jack socket at the top of the channel strip, channels 1 and 2 can also accept instrument signals (still through the combo socket’s central jack, but there’s a latching button to switch the mode). So there are fewer instrument inputs than on the Model 12 (which had this option on eight channels), but the same as on the Models 24 and 16. It’s not a big loss, in my view: I rarely want to capture more than two DI’ed instruments at once, and if I do then DI boxes aren’t expensive.

The Ultra-HDDA mic preamp is the same as that introduced on the Model 12 (an upgrade on the Model 24). As befits a general-purpose mixer, it’s clean, quiet and unfussy.

It offers modest gain (0-50 dB) but it’s ample in most scenarios, and there’s plenty more gain available elsewhere in the signal path. Not a preamp you’ll to want to drive hard for character, perhaps, but it’s of decent quality — I just never really had to think about them throughout my review tests.

Phantom power is switchable in banks of four, as on the Model 24; it’s global on the other models.

Contrary to popular myth, phantom power won’t usually damage ribbon mics unless they’re very old or faulty, but it’s nice that you can plug/unplug phantom powered mics without switching them all on/ off, and it might prevent accidental damage to any line-level gear that’s accidentally been patched into the XLR input. Speaking of line-level signals, the gain range for these is -10 to +60 dB.

After the preamp is a switchable 100Hz, 18dB/octave high-pass filter, but sadly, as on previous Models, there’s no channel with a polarity invert button. This will surely be missed by anyone wanting to put one mic on a drum’s batter head and another on the resonant head (particularly common on snare drums), and I hope Tascam consider adding this to a couple of channels in the future; the instrument buttons on channels 1 and 2 and the layout of the Model 12 suggest there’s space to accommodate them. Still, this time there’s a simple workaround: plug an inverter cable or box into a channel insert point.

Below the filter, a slide switch selects one of three input signals: that from the preamp stage; the multitrack return from a computer; or playback from the SD recorder.

Another switch determines the point from which the signal is tapped to feed the A-D converter. Then we have the one-knob VCA compressor, followed by a three-band EQ with sweepable mid; these two stages are treated as one and share a bypass button. Each EQ band offers ±15dB of gain.

The top and bottom bands are high (10kHz) and low (60Hz) shelves, while the fixed‑Q (0.5) mid ranges from 100Hz to 8kHz. Channels 13 to 20 (eight channels) occupy only four physical control strips.

These can function as mono mic/line channels (using the combi XLR input) or stereo line channels using the combi’s jack

for the left and an adjacent TRS jack for the right. The facilities are otherwise the same as on the lower-numbered channels, except there’s no compressor and the EQ’s mid band is fixed at 2.5kHz — an area of the spectrum we’re particularly sensitive to, and that’s important for vocal intelligibility.

Another channel strip can be a mono (channel 21) or stereo (channels 21 and 22) line input, can monitor the stereo USB or multitrack return, or receive a signal from the built‑in Bluetooth receiver. For Bluetooth, you set the selector in the line position and, in the dedicated Bluetooth area of the panel, route the Bluetooth

kid record with it, and she grasped most things pretty quickly, really enjoying the intuitive, hands-on nature of it. Working with her also confirmed that the dual headphone outputs were very handy.

The preamps seem very decent for a device of this price, and while I might not look to the one-knob compressors and EQ for my own studio recordings, they’re competent, do a fine job for cue mixes and will tick most people’s boxes for live sound. I love that this Model has four subgroups and allows me to integrate my studio outboard gear — in fact, for me that’s perhaps the biggest plus point compared with its older siblings; it’s so much more versatile.

The Model 2400 also makes for a good audio interface if you’re content with working at 44.1 or 48 kHz. There are plenty of channels, the analogue mixer makes setting up cue mixes a piece of cake, and an artist can hear their compressed, EQ’ed performance with reverb but no latency. Though fairly basic, I found the DAW control functions more helpful than I’d anticipated too.

But what I love most is just how much you can do on this thing without a computer!

There are plenty of tracks to accommodate full bands, decent-sized synth and drum machine setups, or subgroup processors and effects returns. There are enough auxes to create different cue mixes for several band members, and I found that the punch-in facilities encouraged a more immediate, performance-focused approach that I too often forget to adopt when in front of the DAW. There are a few small compromises, as you’d expect on a multi-function device at this price. The biggest for me is the lack of polarity inversion. Insert points on the subgroups would have been nice, as would separate level controls for the two headphone outs. But on the whole I’ve been seriously impressed with the Model 2400.

I can well imagine it as the centrepiece of a songwriter or band setup, with a few mics, instruments and processors/effects semi-permanently plumbed in. You could pretty much just switch on and be recording as soon as inspiration strikes.

“When I don’t have to look at a screen while making music I prefer not to. And the simple fact of having so many tracks with inserts available makes working this way much more attractive.” signal to 21+22. That switch also allows you to turn Bluetooth off, or assign it directly to the main mix, freeing up 21+22 for other things. Finally, though it has no ‘strip’ as such, there’s also a dedicated talkback mic input, located next to the pair of headphone outputs at the top right of the mixer.

Impressions I very much enjoyed using the Model 2400. It sounds very pleasing to my ears, and if you have any knowledge of how a mixer works it’s super easy to operate. I did the usual test of letting my

The ModelMixer app provides more metering options than can be found on the hardware — and on a bigger screen too!

$ $1999.

T TEAC America Inc. +1 323 726 0303.

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The T-RackS 6 Mastering Console.

JOHN WALDEN

F

aced with an absolute plethora of plug-in choices, there can be something really appealing about the idea of buying one ‘do it all’ bundle from a single manufacturer that can cover most bases. IK Multimedia aren’t the only company to offer such a bundle, but their T-RackS range is well established, well stocked, well regarded, and available in a single-purchase, perpetual-licence format, which will no doubt appeal to those who are uncomfortable with subscription-based models. IK recently released T-RackS 6, which adds new processors and new functionality to the package. So, if you’re tempted by the idea of a plug-in suite, could T-RackS 6 be a suitable contender?

Back T-Racking We’ve reviewed previous versions of T-RackS on a number of occasions, most recently T-RackS 5 Max in SOS March 2019 (https://sosm.ag/ik-multimedia-t-racks-5). As a brief reminder, T-RackS has two different forms: there’s a standalone application called the Mastering Console, that’s aimed primarily at mastering duties and lets you build custom signal chains from the suite’s individual processors; but all the individual processors can also be used as separate DAW plug-ins (AU, VST3 and AAX, all in 64-bit), so you’re free to use them for mixing

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IK Multimedia T-RackS 6 Mixing & Mastering Plug-ins

Can one bundle really meet all your mixing and mastering plug-in needs? or mastering duties in any suitable host. As I’ll explain below, though, the new release makes possible a third approach.

As before, T-RackS is available in multiple ‘tiers’, whose prices and feature sets are designed to appeal to different users. In v6, these tiers are called Intro (a free taster version with only three plug-ins), the standard version (just called T-RackS 6, this includes 19 processing options), Pro (40 processing options) and Max (60 processing options). IK’s website displays a comprehensive list of the processors/plug-ins offered in each version, but it’s worth pointing out that all of the processors are available for individual purchase too, so should you want to add just one or two extras to a specific tier you can do that. Whichever (paid) package you opt for, you’ll get an interesting combination

of IK’s emulations of some classic hardware designs (inspired by outboard from the likes of SSL, Neve, Teletronix and Pultec) and their own original creations (for example, the Classic Compressor, Equalizer, Clipper and Multi-band Limiter).

The latter group sometimes also take their inspiration from hardware, but they also provide modern features made possible by the software format.

Interestingly, since my v5 review, the pricing has been significantly restructured, presumably because the plug-in market is becoming so competitive, and the revised pricing makes each of the paid tiers far more accessible for first-time buyers.

Rack ’Em Up All the old favourites are still present and correct, of course, and still sound good, but

given these have been covered in past SOS reviews, I’ll focus more on the highlights among the new and improved offerings of this latest release. This includes not just new processors, but also a revamp of the standalone Mastering Console. The broad look and workflow of this remain intact, but a new Module Master Match X lets you match the tonal balance and dynamics of your tracks Manager panel lets you customise to a specific audio file or genre reference. which plug-ins and plug-in formats are edition gets Master Match X (a considerably visible — so, if there are things you don’t revamped version of the earlier Master use, you can keep them out of your way.

Match) and Channel Strip X. Pro then adds When mastering, the lower panel now Delay Lab, LO-FI Punch and Filter Fusion. includes a Master Bus view alongside the On top of all these, Max gets Bass ONE, Waveform or Chain (the signal chain applied Dual Spring and Pusher. to the currently selected audio file) options.

Master Match X offers a function The Master Bus lets you add up to four that’s similar in nature to iZotope Ozone’s ‘global’ processors that can be applied to Assistant View, in that it uses genre-based every audio track in your current project. references (or reference Both the Master Bus and Chain views tracks you have selected now also show input/output level meters yourself) as targets to between every processor in your chain, and ‘match’ in terms of tonal so make gain-staging much easier. balance and dynamics.

The Metering panel can now be opened After a brief analysis as a customisable floating window that period, Master Match can display an impressive combination of X will suggest some loudness, dynamic range, spectral data and initial settings and you stereo image information to help you hit can then use a compact whatever technical specifications you might set of macro-style have in mind. Both the Album Assembly controls to make further panel (where you can sequence and edit adjustments, including multiple tracks into an album-style project) detailed EQ/tonal balance and the Preset window can now also pop refinements in the upper open as floating windows. frequency-spectrum I find the Mastering Console very panel. Master Match impressive. Rather like the Montage X is a doddle to use approach in Steinberg’s WaveLab, it and, while you can provides a very intuitive framework for undoubtedly push things mastering multiple songs in one project. far enough to make As all paid versions also include ONE (IK’s your master a disaster, excellent macro-control all-in-one mastering if used sensibly it’s a very processor) and the new Master Match X effective tool that almost processor (more on this shortly), T-RackS 6 anyone could use. is also a very accessible starting point for Like the other T-RackS those looking to get into DIY mastering.

‘X’ processors, Channel

In With The New A total of eight new processors have been added to the T-RackS family. The standard

IK Multimedia T-RackS 6 From $132 pros • All tiers are good value for money.

• Options include both emulated classics and modern processors.

• Some excellent new additions, notably Master Match X, Bass ONE and Filter Fusion.

cons • Lacks pitch-correction and creative modulation effects.

summary As a bundle of mastering and mixing processors, T-RackS 6 provides a great combination of emulated classics and modern interpretations.

It offers great performance and very good value in every tier.

Strip X avoids any attempt to resemble classic hardware, instead offering a compact set of macro-style controls that are intended to give you all the EQ and dynamics options you need for routine processing of a single sound source. My only quibble might be that a few more Channel Strip X presets would have been nice, perhaps including some aimed at bus-level processing. That said, it’s

LO-FI Punch, Bass ONE and Channel Strip X are among the most significant new additions.

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super easy to use, so it’s pretty quick to dial in something suitable for sweetening, say, a drum or guitar bus.

Of the two new ambience processors, Delay Lab and Dual Spring, the former is undoubtedly the highlight. I remarked in the T-RackS 5 review that the collection needed a workhorse delay processor — Delay Lab ticks that box and more. The more conventional delay control set is complemented by a very useful visual display of the delay treatment, while the bottom-most panel provides additional

options to add more character. It can sound great, and is especially effective with vocals or melodic lines. Dual Spring sounds authentically spring-like, with the pleasant twist of letting you blend two different spring reverb treatments in a single processor.

Bass ONE may well be the pick of the new additions for many users. It takes the ‘do it all in one plug-in’ mastering tool concept of ONE and applies it to handling your mix’s low end. The macro-style control set lets you refine the EQ, compression and stereo width, and add styles of saturation/ distortion but, with simple filter controls located top left, you target just the low-end elements.

I thought it excellent. For example, with separate instances applied to drum and bass busses, it gave me all sorts of interesting ways to finesse the kick/ bass relationship.

As with ONE, there are enough options to play with that there’s a bit of a learning curve, but it remains accessible even to less experienced users.

Before giving LO-FI Punch a spin, I was halfexpecting it to be IK’s answer to XLN Audio’s RC-20 Retro Color but that’s not really the case — the focus is perhaps more on grit and characterful compression. Yes, with a suitable wet/dry mix, the Noise, Wow, Flutter and Pitch controls allow you to dial in some subtle sonic charm for those classic From top: Pusher, Delay Lab, Dual Spring and Filter Fusion. lo-fi relaxation

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vibes, but a harder edge is also attainable here, and it can work really well if your musical needs are darker or more sinister.

Pusher is a sonically and visually colourful four-band saturation processor.

With five saturation modes per band to choose from, and options to drive the saturation with envelope followers, it offers a tremendous amount of flexibility: it can easily span subtle warmth through to flatout distorted destruction. The adjustable bands can be individually toggled on and off, so if you want to add the effect only to a specific frequency range, that’s easy to do.

It’s great on a drum bus, provides all sorts of different ways to make a standard electric bass sound more assertive or aggressive, and is controllable enough to provide subtle thickening to either male or female vocals.

Filter Fusion was, to me, the biggest surprise among the new arrivals.

A conventional set of filter types and slopes are offered, but there’s additional coolness in its three modulation sources: an envelope follower, a step sequencer and LFO can be combined and blended in real time using the triangular ‘pad’ on the top right of its GUI. Whether you use it to add an electronic element to a standard acoustic drum loop, spice up a static synth sound or add a filter groove to a rhythm guitar part, Filter Fusion is super easy to use and, most importantly a lot of fun!

Best Of The Rest Among T-RackS 6’s other improvements there are two personal highlights that I must mention. First, you now get the T-RackS 6 plug-in, which puts the Mastering Console’s signal-chain functionality into your DAW, to give us that ‘third workflow’ I mentioned above.

In a single DAW insert slot and plug-in window, you now have access to full T-RackS-based processor chains on instrument tracks or busses while mixing.

The plug-in ships with an impressive list of presets to get you off to a flying start in constructing a suitable chain for most common mixing tasks too. But whether you value a little assistance with your plug-in choices, or simply need to work quickly, I reckon this is a very useful enhancement.

The second highlight is that IK have now added external side-chain support for the various dynamics processors when they’re used as individual plug-ins. This obviously has to be configured in your host DAW in the usual fashion (which might vary from DAW to DAW) but there’s also a small activation button in the plug-in header strip

that has to be enabled (top right, beside the A/B/C/D snapshot buttons — and this could perhaps be better documented in the PDF manual!). Side-chain options are now pretty much ubiquitous when it comes to dynamics plug-ins, so you might argue that this is a case of IK playing catch-up — but catch up they have, and it’s a very welcome addition that will make T-Racks 6 more useful to a lot of users.

A Full Rack?

As I hinted at the outset, there is a lot of competition when it comes to plug-in bundles, but in terms of quality and choice IK are up there and deserve your consideration. With T-RackS 6 you can achieve great results, whether mixing or mastering, and the three differently priced tiers of T-RackS 6 each provide some great options, covering the essential mixing and mastering tasks of EQ, dynamics, saturation and ambience. Your choices in each regard and in terms of the depth of control increase as you move up the range, of course, and by the time you get to the Max edition, you have access to a pretty

The new T-RackS plug-in includes a good collection of preset signal chains aimed at many common mixing tasks.

comprehensive bundle of tools. There are a few things you’ll find in other bundles/ subscriptions that you won’t find here: there’s no pitch‑correction or, other than the modulated filter, creative modulation options. Equally, T-RackS doesn’t include guitar rig simulation or instruments but, of course, IK offer those separately, not least in the form of AmpliTube and Tonex. It will be interesting to see if IK’s Total Studio packages (which have combined T-RackS, AmpliTube and some

of their virtual instruments) get updated to include T-RackS 6. If they do (and they may well have been updated by the time you read this) then I have to say that these bundles represent astonishingly good value for money. $ T-RackS 6 €119.99 (about $132),

Pro €239.99 ($265), Max €359.99 ($397).

Upgrade pricing and discounts available to owners of IK Multimedia products.

W www.ikmultimedia.com

SOLO SERIES SOLO MPA MULTIVOICE TUBE PREAMPLIFIER

SOLO VLA MUTLIVOICE TUBE/OPTO COMPRESSOR

SOLO STUDIO MULTIVOICE CHANNEL STRIP

All Analog. Real Tubes. Shockingly Affordable.

ARTPROAUDIO.COM

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The Audio Puck control unit and loom, pictured here with the forthcoming TAP-CAT box that supports audio over Cat6/7 cables.

Interspace Industries Audio Puck

analogue audio who has a regular Audio Cable with need to interconnect devices that aren’t directly compatible.

System

Does this box of tricks live up to its billing as the Swiss Army Knife of audio cables?

HUGH ROBJOHNS

W

e’ve all needed some form of special ‘bodge cable’ to link one device to another because of non-standard connections or incompatible interfacing, and most of us will have built up an assortment of adaptors and special cables. Interspace Industries aim to tackle this perennial issue with their new Audio Puck, which they claim can be configured into more than 90 different connection formats.

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Achieved with just one compact switch box and a detachable connector loom, it also has options for balanced-to-unbalanced interfacing, stereo splitting, insert breakout, phantom-power blocking, ground lifting, and more besides. It’s entirely passive, with no transformers: just wires, switches, and connectors. The Audio Puck was designed by a live-sound engineer primarily for solving everyday live-sound interfacing problems, but it’s obviously of use to anyone working

Overview

Housed in a strong metal case, the Audio Puck control unit measures roughly 82 x 79 x 42mm and weighs 200 grams, making it smaller and lighter than a typical DI box. The front panel carries a pair of combi XLRs and a pair of RCA phono connectors, while the rear has a 25-pin D-Sub connector for attaching the supplied cable loom (see ‘Loom Wiring Arrangements’ box).

A recessed trough on the top protects a trio of switches and a 3.5mm mini-jack socket. Two of these switches are miniature toggles and select an Iso/ Mono mode (of which more later) and

a ground-lift function. A larger, three-way toggle configures the system for balanced, unbalanced, or stereo/insert modes. Most of the switch labelling is black on a white background, for easy legibility in dark situations, while the sockets and Iso switch have colour‑coded markings that indicate which cable groups and sockets carry which signals.

Supplied with the Audio Puck is a short (about 45cm) cable loom weighing around 460 grams. There are a lot of connectors in this loom, most being Neutrik types, and the control box end is terminated robustly in a 25-pin male D-Sub connector. The loom itself fans out into nine separate tails, each with a different connector, and with three different cable colours. There are two tails in the blue group, two in the green, and five in the red. The D-Sub is wired to the AES59 (Tascam) standard, with each tail’s connection on its own channel. The only exception is that the left and right RCA phono connectors in the red group share a channel (there’s more about that in the box).

All connectors in the same colour group are essentially wired in parallel, but the different groups are linked (or not) to each other in various ways, depending on the mode setting on the control unit. The blue cable group is designated as carrying the left-channel or insert-send signals, while the green group conveys right-channel or insert-return signals. The corresponding combi XLR sockets on the control unit have the same coloured labels. In most modes, the larger

red group effectively serves as the input or output connection set.

Obviously, colour coding is essential in identifying the different connector groups, and the designer told me he assigned the red, green and blue colours for “no rhyme or reason”. I’m sure regular use will bring familiarity but it has the potential to confuse some users: in the consumer world, the CEA standard uses white for the left channel and red for right, and my BBC training means I always associate red with left and green with right (as with navigation lights on planes and ships), so coping with blue and green felt a little unnatural. The control unit sockets being colour coded helps a lot, though, and if all else fails then a QR code printed on the base of the unit links directly to the full user manual online, which is a very thoughtful touch.

While we’re on the subject of the manual, it has 18 pages in total, though only seven are needed to explain the multitude of configuration options and functions — and that’s quite an

Interspace Industries Audio Puck $560 pros • Amazingly versatile problem solving for different audio connectors and formats.

• Most unbalanced connections are protected from phantom power.

• Compact and robust solution.

• Easy to use and configure.

• All-in-one system easier to find than a specific bodge cable in a big box!

cons • No transformers for galvanic isolation or true balancing/unbalancing.

summary A clever everything-to-everything interface for versatile and rapid problem solving, with every option in one place and easy to configure.

worth noting. The left RCA phono socket signal connection is wired to the blue (left/send) hot line (pin 2/tip), while the right RCA phono socket signal connection is wired to the green (right/ return) hot line. Both RCA phono signal lines are protected with phantom-power blocking capacitors, which is very useful.

With multiple loom connectors wired in parallel and many with exposed contact points, there’s a real risk of signals being lost should a stray connector accidentally touch something it shouldn’t.

To combat this, most of the loom’s plug connectors are provided with natty little isolating ‘jackette’ covers, secured to their cables by wire tethers so they can’t be lost. These were designed and 3D-printed by the manufacturer and fit snugly over their associated plug type — it’s a very neat solution.

“Most of the loom’s plug connectors are provided with natty little isolating ‘jackette’ covers, secured to their cables by wire tethers so they can’t be lost.”

Loom Wiring Arrangements The Audio Puck’s connector loom is terminated with a D-Sub using the AES59 (Tascam) standard wiring arrangement.

Consequently, the loom itself can be used as an emergency connecting cable for interfacing eight-channel analogue equipment using AES59 connectors. The various colour group connectors are wired to the eight balanced AES59 channels as follows, with the channel numbers in bold: Red 1. RCA phono L/R 2. Mini-jack plug 3. TRS 4. XLR

Green 5. TRS 6. XLR

Blue 7. TRS 8. XLR

achievement given that there are over 90 variations! There are also clear and useful block diagrams to illustrate the signal routing in different modes, for those who prefer images to words.

Blue & Green Group Connections In terms of loom connectivity, the two blue (left/send) and two green (right/ return) tails each terminate in a Neutrik ConvertCon and a quarter-inch TRS plug. For anyone unfamiliar with the ConvertCon, it’s an ingenious XLR connector which transforms from female to male forms just by sliding the outer sleeve forward or back.

When connected to the control unit, the XLR and TRS connectors in each colour group are wired directly in parallel, along with their corresponding combi XLRs and RCA phono plugs on the control unit. Both RCA phono sockets are wired with their sleeves connected to the green (right/return) channel only — rarely of significance in practice, but

Red Group Connections The red group has five separate tails.

The first two are terminated in an XLR ConvertCon and a quarter-inch TRS plug, as with the green and blue groups.

Once again, these two plugs are wired in parallel via the control box, as well as being linked directly to the 3.5mm TRS socket in the control box switch recess. The third tail carries a 3.5mm TRS mini‑jack plug, while the fourth and fifth have RCA phono plugs. The left RCA plug is wired to the pin 2/tip hot line, while the right RCA plug is wired to the pin 3/ring cold line. All three are again www.soundonsound.com / December 2024

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INTERSPACE INDUSTRIES AUDIO PUCK

The control unit’s configuration switches are located in a protective recess, while the loom attaches securely to it using a DB25 D-Sub connector.

protected by DC-blocking capacitors to protect from phantom power, and all have protective jackettes.

Another point worth noting is that while the red group mini‑jack plug is protected against phantom power, the 3.5mm socket on the control box is not.

This is to allow the passage of plug-in power to electret microphones and the like if an alternative cable is being used.

Operating Modes The simplest configuration mode is Balanced, where all three cable groups are wired directly in parallel, essentially with the red group as the input and the blue and green groups as the outputs.

Consequently, this mode allows balanced XLR-jack conversions in either direction, signal splits or mults, and even balanced connections over mini-jacks. For example, by plugging the red quarter-inch TRS plug

into a balanced output, a signal can be shared across combinations of three male XLRs, five female XLRs, two more jack plugs, or two jack sockets. Or, if a signal is plugged into one of the combi XLR quarter-inch sockets on the control unit, it can be split to feed three male XLRs or four females XLRs, and three more jack plugs. It’s also possible to create gender reversers with female to female or male to male XLR connections.

By operating the ground-lift switch, the sleeve/pin 1 connection is broken between the three different colour groups — but the grounding is maintained within each colour group; all of the red connectors remain grounded to each other, but that ground is not shared with the green or blue group connectors.

Selecting the Unbalanced mode shares signals between connectors in the same way, except that the ‘cold’ side

of each balanced connector is shorted to the screen/ground within each colour group. That’s to say, XLR pin 3 is linked to pin 1, and the TRS ring to the sleeve.

This allows unbalanced signals to be passed over balanced connectors and cabling in a straightforward way, as well as for connecting unbalanced outputs to balanced inputs or balanced outputs to unbalanced inputs.

Although shorting the cold side to ground works in many situations, it can cause problems with some types of balanced output. It works correctly with transformer outputs and cross-coupled electronic outputs, but it can sometimes cause problems with simpler electronic balanced outputs. In these cases, the cold-side output device might not like being shorted to ground, and can introduce peak or transient distortion as a result; if in doubt, always check

TAP-CAT The use of Cat6/7 Ethernet cable as a low-cost, four-pair, analogue multicore is growing in popularity and a dedicated adaptor module called the TAP-CAT is in development. This allows the Audio Puck control box and its loom to send or receive four separate channels of audio over Ethernet cable. I tested a fully functional prototype of the TAP-CAT, which is fully compatible with all known Ethernet analogue connection systems, and the production version should be available by the time you read this review.

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The TAP-CAT box is slightly smaller than the Audio Puck, with a male D-Sub connector at the front which plugs into the D-Sub connector at the rear of the control box, creating a larger control unit with two extra connectors on the sides.

The standard cable loom plugs into a female D-Sub at the rear of the TAP-CAT adaptor box. EtherCon sockets are fitted on each side of the TAP-CAT adaptor, allowing one or two Ethernet cables to be connected, and the four balanced analogue audio channels can then be accessed with the blue

December 2024 / www.soundonsound.com

and green combi XLRs on the control box itself carrying channels 1 and 2, respectively, while the blue and green cable groups in the loom carry channels 3 and 4, respectively. In other words, channels 1 and 2 appear on female XLRs or TRS sockets on the control unit, while channels 3 and 4 appear on the ConvertCon XLR and TRS plugs on the loom.

The TAP-CAT can also be used on its own as a simple passive EtherCon cable joiner.

The prototype TAP-CAT sent along with the review unit allows the Audio Puck to cater for audio over Cat6/7 cable.

the manufacturer’s recommendation when unbalancing a balanced output!

Instigating the Stereo/Insert mode assigns the red cable group as the unbalanced stereo or insert connector, routing two unbalanced audio channels separately to/from the blue and green groups. So, for example, if the red TRS plug was inserted into a stereo headphone output or a mixer’s insert socket, the blue group connectors will carry the unbalanced left channel or send signal, while the green group connectors carry the unbalanced right or return signals. This mode also allows conversion from stereo mini-jack to dual-XLR/TRS, or dual-RCAs to dual-XLR/TRS, for example.

The final switch selects Iso/Mono modes. In the stereo/insert mode, activating the switch means the blue and green groups both receive an unbalanced mono sum from the (red group) mini‑jack plug’s left and right channels, mixed via 100Ω resistors. In balanced mode, the Iso switch disconnects the blue and green groups from each other, allowing the blue group to handle one signal, and the

green group another. The blue and green group grounds are still shared unless the ground‑lift switch is operated. (When using this Iso mode in the balanced or unbalanced configurations, the red group is not used.)

Looming Large As you can see, there’s lots of capability and flexibility in the Audio Puck, and the claim of it being configurable “into over 90 different cable types” is genuine: there are 93 explicitly different setups!

In practical applications, all of the most commonly required configurations are quick and easy to set up: balanced signal splitting/distribution in mono or stereo, ground lifting, unbalanced insert or stereo breakouts, and balanced/ unbalanced connections. Moreover, all of the most common connector types are available here, too — and, importantly, most of the unbalanced ones are protected from phantom power, which is a very nice feature.

On the down side, with no isolating transformers in the Audio Puck it’s

not possible to create a balanced signal from an unbalanced source, or to provide genuine galvanic isolation. The shorted-cold side issue I mentioned earlier wouldn’t happen with transformer unbalancing, either, and it’s partly for these reasons that I’ve relied heavily on ART Pro DTI transformer isolation boxes for a lot of my interfacing needs — and these boxes feature paralleled XLR, TRS and RCA phono connections. However, the Audio Puck can do things my DTI boxes can’t and it’s genuinely worthy of the Swiss Army Knife moniker: it’s a tool for fast and efficient problem solving in the vast majority of commonly encountered situations. It works very well, is easy to set up and use, and is a nicely integrated solution. $ Audio Puck and standard Loom $560.

(At the time of going to press, the TAP-CAT was not yet available and pricing was yet to be announced).

T Interspace Industries +1 203 814 1879.

E moreinfo@i2llc.com W https://i2llc.com

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Soyuz 013 Ambisonic Ambisonic Microphone

Will Soyuz’s beautifully crafted four-capsule mic help turn Ambisonics from a niche interest to a mainstream studio technology?

playback. In practice, this is typically manufactured by panning mono and stereo sources around a speaker array.

Ambisonics, by contrast, is a spatial recording format, and an Ambisonic microphone is designed to capture the 3D soundfield that impinges on it.

The Age Of Ambisonics SAM INGLIS

T

he brainchild of pioneering mathematicians Michael Gerzon and Peter Craven, Ambisonics extends the idea of Mid-Sides stereo into three dimensions. And given how much popular confusion there is about Mid-Sides stereo, it’s hardly surprising that Ambisonics is widely misunderstood.

In particular, I think many people view it as something that’s only relevant if you’re working in surround — but nothing could be further from the truth.

An Ambisonic signal simply describes the 3D soundfield at a particular point in space, and the beauty of it is that this soundfield can be mapped onto any playback system you like, from a mono speaker to a vast Atmos array.

Formats such as Dolby Atmos are ‘spatial’ in the sense that they generate a three-dimensional soundscape on

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In conjunction with Ken Farrer at Calrec, Gerzon and Craven created the first commercial Ambisonic mic, the Soundfield, back in the late ’70s. Now part of the Rode group, Soundfield still make high-end Ambisonic mics such as the ST450 MkII and SPS200, and were instrumental in the design of the impressive Rode NT-SF1 back in 2018. All Soundfield mics are first-order Ambisonic mics, which employ four cardioid or subcardioid capsules in a tetrahedral array. The same is true of models such as the Sennheiser Ambeo, Brahma Studio 4 and Core Sound Tetramic, and you can now buy second-order Ambisonic mics with eight capsules such as the Voyage Audio Spatial Mic, Core Sound Octomic and Brahma Studio 8. In principle, a second-order mic offers sharper localisation and thus the potential for greater realism, at the expense of greater complexity and cost.

Before anything useful can be done with the raw sound from the capsules of any Ambisonic mic, it must be processed. In the analogue domain, closely matched gain settings are used to preamplify each capsule’s output by exactly the same amount. Filters are then used to matrix the raw four- or eight-channel signal into a universal pattern known as the B-format. It’s this B-format signal that can be thought of as the 3D extension of Mid-Sides, and which can then be decoded into mono, stereo, or any surround format. If you’re working in stereo, this decoding can mimic any coincident stereo miking array, pointing in any direction you like. In other words, an Ambisonic mic can be thought of a stereo mic that can be reoriented after the fact.

(It can do a whole lot else, too.)

These days, the filtering and decoding is typically handled digitally, and for good reason. Since the Ambisonic mic is by definition a coincident array, directional information is represented purely as amplitude differences between the channels, and a high degree of precision and consistency is important in order to maintain the integrity of this information.

This is much easier to achieve in the digital domain, albeit that no plug-in is quite as impressive to look at as the

analogue decoder box from a Soundfield MkIII! Whereas some of the Soundfield models still come with a custom-made preamp and transcoder, most other Ambisonic mics supply only the means to capture the signal from each capsule directly. It’s up to the user to provide four or eight matched preamp channels and A-D conversion, whereupon digital processing is used to extract the B-format magic from the raw (A-format) signal. As a consequence, mics like the NT-SF1, Ambeo, Tetramic and Studio 4 are more affordable to buy than the high-end Soundfield models.

Purity Test It’s probably fair to say that all the Ambisonic mics that have existed up to now have been marketed primarily as tools for capturing sound in an uncoloured, natural, lifelike fashion.

They’re not conceived as ‘character’ mics in the way that, say, a valve mic or a large-diaphragm capacitor mic might be, and they’re not designed to add their own contribution to what we hear.

An Ambisonic mic is typically intended to provide a completely transparent — and uniquely versatile — window on the audio world.

That’s not quite so true of the latest first-order Ambisonic mic to reach the market. Like most such designs, Soyuz’s 013 Ambisonic employs four directional capsules in a tetrahedral array. But unlike any other current Ambisonic mic, it has a transformer-balanced output stage for each capsule, employing Soyuz’s custom toroidal transformers to add “additional depth with a subtle character that complements any source”. And, whereas many Ambisonic mics use miniature electret capsules, especially at the affordable end of the scale, the 013 Ambisonic features Soyuz’s own, externally polarised (‘true capacitor’) 013 capsules. At 24mm across, these are somewhat larger than those used in many pencil mics, or indeed in other Ambisonic models. Consequently, it’s rather more costly than the NT-SF1 and its rivals, but is still more affordable than the flagship Soundfield variants.

The 013 Ambisonic’s unique positioning as a vibey alternative to traditional Ambisonic mics is evident from its packaging. Whereas the Soundfield ST450 MkII and its accessories ship in a bulletproof but unglamorous Peli-type hard plastic case, the 013 comes in

a gorgeous wooden box. The mic itself is also very nicely made, and again, its luxuriant curves contrast with the more utilitarian style of mics like the ST450.

Whereas the entire capsule assembly on the ST450 is housed within a protective cage, the 013 follows the design of the NT-SF1 in giving each capsule its own grille and so on. The capsules can actually be unscrewed from the ball-like hub that holds them in place, though I can’t imagine there’s much mileage in a ‘modular’ Ambisonic mic with swappable capsules! The mic as a whole is a couple of inches longer than the ST450, and although it’s only fractionally wider, the spacing between the four capsules is quite a bit bigger.

A good-quality shockmount is included, as is a basic standmount, both of which are held in place by a ring that screws onto the base of the mic. One popular application for Ambisonic mics is ambience recording, but if you plan to use one outdoors, you’ll need a good purpose-built windscreen. Nothing is included with the 013, and Soyuz themselves don’t yet offer any optional add-ons, but they say that compatible offerings are available from third parties.

At 579g, the 013 is almost exactly twice

Soyuz 013 Ambisonic $2799 pros • Beautifully made.

• An interesting alternative to clinical-sounding Ambisonic mics, with a colourful character and presence.

• Can accept very high sound pressure levels.

• Good-quality shockmount and case.

cons • Transcoding plug-in only available to registered owners of the mic.

• No windshield included.

summary Soyuz have made an interesting and ambitious attempt to create a characterful Ambisonic mic that is particularly suited for music recording in the studio.

as heavy as the Soundfield ST450, which could be an issue for use on a boom.

Like most other current Ambisonic mics, the 013 doesn’t include a hardware preamp or decoder. Instead, it presents four conventional mono outputs, which need to be fed into four phantom-powered preamp channels with closely matched gain settings. Supplied with the mic are a multicore cable with

Shockmount, standmount and cable are included, but there’s no wind protection as standard.

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12-pin connectors at either end, and a short breakout loom that terminates in four colour-coded male XLRs. The review mic came with a rather inadequate 3m multicore, but production mics should come with a more usable 5m cable, and longer ones will be available from Soyuz.

The plug and socket at either end are held in place using locking rings, and resemble the connectors that are used on some valve mics, such as the Neumann M147. To my mind, they are not as confidence-inspiring as the LEMO connectors that Soundfield use on the ST450. Soyuz say that they have been rigorously tested and thrown up no problems, but you’ll want to make sure they are out of the way of clumsy feet in the middle of the studio floor.

formats, though the Audio Units version is designed specifically for Logic; if you use another Mac DAW, you should load the AAX or VST versions.

These plug-ins are installed through a somewhat intrusive plug-in and licence manager called Pulse, which I’ve not encountered before. In order to download and activate them, you’ll need to register your 013 Ambisonic at the Soyuz website and use the licence key that is emailed to

Documentation for the 013 plug-ins is also a bit sketchy. There’s no built-in help, nor any manual as such, so you are reduced to scouring YouTube videos or searching on the Audio Brewers website for vital information. In particular, the 013 Transcoder plug-in gives you the option of outputting B-format in either FuMa or AmbiX formats; it turns out that the Decoder only accepts the latter, but this is not apparent from the user interface.

(Quite why the Ambisonics world requires two incompatible B-formats that differ only in the order of the channels, I’ve never managed to explain.) Two transcoding modes are available: a low-CPU, zero-latency one for real-time monitoring, and a high-quality one for mixdown. The difference in sound quality between the two is drastic, but the high-quality mode incurs 4096 samples of latency, which equates to a stonking 93ms at 44.1kHz. When it comes to decoding, as a long-term Soundfield user, I’m accustomed to the ‘virtual microphone’ paradigm employed in that company’s Surroundzone and Soundfield By Rode plug-ins. The Audio Brewers decoder uses an alternative ‘virtual speaker’ paradigm, which probably makes more sense if you’re decoding for surround or binaural playback, though it is arguably less versatile for stereo use.

In both cases, you can rotate the virtual soundfield in three dimensions; the Audio Brewers plug-in also has a Dominance control that allows you to ‘zoom in’, essentially increasing the directivity.

Unlike the Soundfield plug-ins, it offers a binaural output option as well as stereo and numerous surround formats. At the end of the day, it’s often possible to come up with settings in both that sound quite similar; and if it isn’t, Soundfield

By Rode is a free plug-in that doesn’t require registration or a licence key, so the only thing that stands in the way of using it instead of the AB Decoder is the fact that there’s currently no Apple Silicon-compatible version.

“What the 013 does offer that you won’t get from any other Ambisonic mic is that emphasis on character.”

From A To B The raw A-format output from the 013 Ambisonic’s four capsules would typically be recorded to a four-channel track in your DAW, before being matrixed to B-format and then decoded. To make this possible, Soyuz have teamed up with plug-in developers Audio Brewers to supply two separate plug-ins named 013 Transcoder and 013 Decoder. These are variants of the same company’s generic Ambisonics plug-ins adapted for the 013, and are available in all major native

you. This is fine as long as you are mixing the material you record, not so fine if you want to share your Ambisonic recordings in their raw state. For example, during the review period I was hired to make some location recordings for a TV production.

I used this as an opportunity to compare the 013 with my own Soundfield ST450 MkII, but if I’d wanted to make the 013 recordings useful to the mix engineer, I would have had to bounce every single take through the 013 Transcoder plug-in before delivery. I would urge Soyuz to make this plug-in at least freely downloadable without registration.

In The Field

The 013 Transcoder plug-in is an essential part of the signal chain, transforming the 013’s A-format output to a B-format signal.

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Providing meaningful specifications for an Ambisonic mic is a challenge, because properties such as frequency response and effective self-noise are dependent on the decoder settings. Soyuz quote a sensitivity of 20mV/Pa, a maximum SPL of 143dB and an A-weighted equivalent

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noise of 16dB; these are similar to the specs of the standard 013 FET pencil mic, and Soyuz confirmed that they relate to the performance of the individual capsules rather than the mic as a whole.

Inasmuch as a direct comparison is possible, they pretty much match the specs of the more affordable Rode NT-SF1, and improve on it as regards SPL handling. It’s still a couple of dB less quiet than the Soundfield ST450, but in practice, is unlikely to be significantly noisier than a typical small-diaphragm mic, and self-noise was a non-issue in all the recordings I made with it.

The ST450 MkII is still the gold standard against which other first-order Ambisonic mics are measured, so I was intrigued to see how the 013 Ambisonic would compare. Among other things, I used the two together on a choral recording, to document band rehearsals, to capture an upright piano and as drum kit overheads, and in each case, two things struck me about the respective sounds of the mics.

First, as I’ve already mentioned, frequency response is a bit of a moving target in Ambisonics, but when I matched decoder settings across the two mics, there was a clear difference in their sound character. The 013 has a livelier and more present timbre, with a lot more going on in the 5-10 kHz area than the Soundfield. At the same time, it seemed less extended at the very top end, above 12kHz or so, and was noticeably less weighty in the lower midrange.

Whether or not this was a good thing was quite context-dependent. For example, when I put both mics up in the middle of a rehearsal room, the presence and sparkle of the 013 were clearly preferable to the Soundfield’s more subdued tonality, which sounded muddy in comparison. But on the choral session, the 013 felt a little aggressive and thin, while the Soundfield remained rich and smooth even when the trebles were belting out their high notes. On drum kit, the Soundfield delivered meatier snare and tom tones, while the 013 offered more rock & roll thrill.

The other difference that was apparent in all of my tests related to stereo presentation. Assuming you choose to decode it that way, the Soundfield presents an effortlessly wide stereo picture. With a suitable source such as a choir, orchestra or other large ensemble, you can spread the

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The 013 Decoder plug-in uses a ‘virtual speaker’ paradigm rather than the ‘virtual mic’ model employed by some other decoders.

signal out across the entire stereo field, with an expansive quality that you might not expect from a coincident stereo mic.

In comparison, recordings made with the 013 always seemed to occupy a slightly narrower window, almost as though someone had moved the speakers a little closer together; and this wasn’t something that could be rectified by modifying the decoder settings.

Localisation of individual sounds within the stereo field was also subtly but consistently better on the Soundfield.

Grand Ambitions Looking back at my 2018 review of the Rode NT-SF1, I find I made almost identical comments about that mic, too. Like the 013, it has a slightly less coincident capsule spacing than the ST450, and I wonder whether this affects its ability to present such a wide stereo image. I also recall the NT-SF1 having a slightly leaner and more forward sound than the Soundfield, though perhaps not to the same extent as the 013; whether this too is a function of the different geometry, or simply reflects the fact that the mics use different capsules, who knows.

In comparing the two, it must be remembered that the 013 Ambisonic is around half the price of the ST450 MkII,

but also that the 450 includes its own hardware preamp and transcoder. If you don’t already own a preamp that can offer matched gain across four channels, buying one will add significantly to the 013’s cost. Even so, the 013 will likely be the more affordable option — or perhaps I should say the less unaffordable option — but it in turn is a lot more costly than the NT-SF1 and other first-order rivals.

What the 013 does offer that you won’t get from any other Ambisonic mic is that emphasis on character.

I don’t know to what extent Soyuz’s transformers are responsible for its presence lift in the 5-10 kHz region, but it’s a distinctive sound and one that suggests a slightly different intended role compared with the NT-SF1s of this world. While they’re out there capturing forest soundscapes and market ambiences, the 013 Ambisonic seems more of a studio-oriented beast, adding its characteristic sparkle to drum kits, pianos, acoustic instruments and anything that demands versatile stereo recording. If you have this sort of money to spend on a high-quality studio mic, it will offer you something you won’t get from yet another U47 clone! $ $2799 W www.soyuzmicrophones.com

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An equalizer is probably the tool you use most while mixing and mastering, so you need the best of the best! With FabFilter Pro-Q 3, you get the highest possible sound quality and a gorgeous, innovative interface with unrivalled ease of use.

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To find a dealer visit www.musicmarketing.ca

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Drum Workshop DW Soundworks Drum Workshop bring their

Virtual Drum considerable experience of drum Instrument hardware to the virtual world.

MARK GORDON

D

W — or Drum Workshop, to give them their full title — are one of the leading manufacturers of acoustic drums. For many years there has been speculation that they’ve been developing electronic drum technology and, in a surprising move, at the end of 2022 the Roland Corporation bought DW. In the documentation that was released as part of the acquisition, Roland explained that they were “buying game-changing technology that allows us to accelerate the introduction of the next generation of

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hybrid drums.” Jump forward two years and DW have released the DWe kit, a groundbreaking combination of high-end acoustic drums and cable-free, Wi-Fi triggering. The kit is unique in the world of electronic drums in that it doesn’t ship with a sound module, all of the sounds instead being generated by Soundworks, a virtual instrument for Mac and Windows that is also available to purchase as a standalone application and a VST/AU plug-in.

The DWe kit itself is probably a little too deep into ‘drummer’ territory for a full Sound On Sound review, as it actually is an acoustic kit (see ‘The DWe Hardware’ box),

Selecting a drum or cymbal in the Kit Designer reveals a host of information, from the size and material of the drum to which heads were fitted and the choice of microphone used to capture it.

but the Soundworks virtual instrument is absolutely fair game, so here we go!

In The Works In the only obvious link with Roland, the Soundworks application and kit libraries are downloaded through the Roland Cloud. Installation is very simple, although personally I find navigating the Roland Cloud quite confusing. I’ll put that down to a lack of familiarity on my part.

Soundworks ships with six included libraries — DW 50th Anniversary, DW Maple Mahogany, DW Pure Almond, DW Pure Maple, Gretsch USA Custom and Slingerland Vintage — each weighing in at around 8GB. Fortunately, DW own the iconic Gretsch and Slingerland brands (hence their inclusion), which allows for the varied range of kits within the package. Currently, the only additional ‘kit’ available for purchase is the DW True-Cast Snare, which features the DW Bell-Bronze True Cast Snare and will set you back an additional $29.99.

Visually, the software uses a familiar VI format, with three main windows: Drum, Mixer and Grooves. The Drum window is where you’ll probably spend most of your time and features a graphic representation of a drum kit in the centre, a channel strip to the left, and editing and instrument selection windows to the right and below. In addition to the kit view, there’s also a ‘Cell View’ option that displays each element of the kit in a grid pattern and allows you to audition elements dynamically by clicking in different areas of the drum. The Mixer and Grooves screens are accessed via tabs across the top of the main display, along with a drop-down browser menu for loading and saving preset kits, transport controls and a processor activity meter.

All kits are presented in a seven-drum format, comprising five toms, a kick and a snare. Cymbal configurations change subtly with each kit, but are based around a hi-hat, crash and ride combination, plus various splashes and china/effect cymbals.

As you might expect from a drum company, the Soundworks preset kits sound exceptional. Clicking on the Kit Browser opens a drop-down menu listing all the kits in alphabetical order. Each library includes between eight and 10 presets, except the DW 50th Anniversary library, which features 20 presets. In addition to the preset name, Genre and Ambience tags are provided to help identify and categorise the kits, but it doesn’t seem possible to sort the kits based on any of these criteria. Although you can ‘favourite’ kits by clicking the star icon, it would be useful to be able to sort by genre or ambience. You can also add

a kit to a SetList, a feature designed for using Soundworks in live performance (see ‘The Dwe Hardware’ box).

Clicking on the small play button next to a preset’s name plays a short snippet of the drums in action, which is a handy way to audition a kit within the browser before you load it. In terms of styles and genres, the DW kits span the majority of rock and metal categories, and the Gretsch and Slingerland kits offer great options for a poppier or more jazzy sound.

Get Your Kit On Soundworks takes a fairly conventional approach to kit editing but adds some nice touches. With a preset loaded, clicking on any element within the kit displays the details of that drum or cymbal in the Kit Designer window located directly below the main screen. An info button brings up additional information, such as which heads were fitted and what microphones were used to capture the direct, overhead and ambience signals. The available libraries and drum categories (toms, kicks, snares and so on) are displayed to the left, which makes selecting alternative drums very quick and easy. Once you’ve selected a drum or cymbal, you can audition it, before pressing the Assign button to swap it into the kit setup. Cymbals are surprisingly well catered for, with a range of Paiste, Zildjian, Sabian, Meinl and Istanbul examples to choose from.

I was a little disappointed to see that the detailed graphic representations of the drums, with their coloured wraps or natural wood finishes, as shown in the Kit Designer

Drum Workshop DW Soundworks $199 pros • Excellent-sounding libraries.

• Studio-style EQ and compression.

• Comprehensive multi-mic drum samples.

cons • No extensive library available... yet.

• Groove composition tools limited.

summary DW Soundworks offers a large selection of well-recorded drums using a range of multi-mic configurations.

With the help of its comprehensive EQ, compression and reverb facilities, it’s possible to achieve natural‑sounding drum parts that will work in a wide range of styles.

window, were not duplicated into the main overview window. Regardless of what kit you load, or which elements you change, the main kit remains as the default red DW configuration. Of course, this doesn’t have any bearing on the drum sounds, but it would be a nice touch and would give some useful visual feedback as to which drums were being used.

In addition to the drums and cymbals, Soundworks includes 10 ‘X-Pads’. These are extra ‘locations’, to which you can add drums and cymbals that can be triggered from external MIDI inputs, such as pads or a MIDI keyboard. All in all, Soundworks allows up to 30 active instruments at any one time.

Detailed manipulation of the drum sounds is achieved via the two panes either side of the kit overview. The window to the right displays the individual articulations, such as centre, edge and rim, for each drum, and their respective levels. As you audition each element, you can clearly hear the multiple round-robin samples that add to the realism of each drum. A mini-menu gives access to Advanced MIDI note mapping, MIDI CC settings and Playback options, where you can edit how articulations are assigned to incoming MIDI data from your keyboard or electronic drum kit.

The panel to the left displays a single channel strip, representing the currently selected drum or cymbal. Identical to its The Mixer window gives access to the multi-mic configurations used to record the various drums and cymbals. In addition, studio style EQ, compression, saturation and reverb modules allow for further processing of the audio.

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DRUM WORKSHOP DW SOUNDWORKS

its six drum libraries features two groove styles, spanning everything from metal and funk to pop, Afro‑Cuban and jazz. Within these styles are verses, choruses, bridge sections and fills that can be auditioned and dragged into a timeline at the bottom of the screen. However, the composition tools are quite basic, and the lack of any bar numbers on the timeline makes navigation a little tricky. Although there is no facility to edit the grooves within Soundworks, you can export your ‘song’ as a MIDI file or stereo WAV file, or drag and drop the MIDI data directly into your DAW.

Design A Drum

Included grooves can be dragged into a timeline to create a song structure. Although there is no facility to edit the grooves within Soundworks, you can export your ‘song’ as a MIDI file or stereo WAV file.

counterpart in the Mixing Board view (only in a more compact form), this gives you access to almost all editing parameters, while still allowing you to see the overview and Kit Designer displays.

Hit & Mix The Mixing Board displays all kit elements, X-Pads, aux busses and send levels, in blocks of 10 faders. Faders for Overhead and Ambience mic levels plus a master fader always remain visible on the right of the mixer display.

Each instrument channel strip comprises a gain knob, mute and solo buttons and a pitch slider allowing you to adjust overall gain structure and pitch of all mic sources (Direct, Overhead and Ambience) simultaneously. Below this are the Direct, Overhead and Ambience level knobs, each with their own solo and mute facility.

Clicking on a small breakout button gives access to level controls for the individual mics within that group. For example, the snare channel includes a top dynamic mic, top capacitor mic, bottom and side mics. Similarly, the Overhead offers stereo and mono mic levels and the Ambience features Close, Mid, Mid Mono, Far and Far ORTF (a near-coincident stereo recording technique using two cardioid microphones). This is a very generous selection of mic options, comparable The Instrument Designer allows you to add your own kit elements, with up to eight round‑robin samples, 19 dynamic layers, 29 articulations and five microphone sources for each of the three mic groups.

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with the likes of Superior Drummer 3, and allows for a great deal of flexibility.

Buttons to the right of each fader open individual effects modules, including comprehensive EQ and dynamics processing, as well as saturation and reverb effects. The design of these modules gives the mixer the familiar feel of an analogue console, making it easy to navigate.

Groove Armada The majority of drum VIs come with a selection of ready-made grooves that you can use in your own productions, and Soundworks is no exception. Each of

In the unlikely event that the available drums and cymbals don’t provide exactly what you’re looking for, the Soundworks Instrument Designer lets you add your own samples to enhance or augment an existing kit. Located at the bottom of the main display, the Instrument Designer becomes ‘active’ when you select a kit piece that is not populated (ie. has no instrument assigned to it). The preset kits, in addition to the standard toms, bass drum and snare drum configuration, include an empty cowbell and auxiliary snare, and these locations are typically where you would add your own samples, but you can also clear the sound from an existing kit piece or use the X-Pads as a starting point.

It’s worth noting that creating an instrument using an X-Pad enables you to save it independently and load it into any preset kit, while instruments created using the cowbell or auxiliary snare locations are only saved within the kit they were created in.

The DWe Hardware part of this article — to generate The DWe kit has caused quite the sounds, rather than the more a stir in the world of electronic usual dedicated sound module. drums. Based around DW’s A compact audio interface, the top‑of‑the‑range Collectors shell DWe RTAP, equipped with two pack, it’s as much at home fitted quarter-inch jack outputs and with acoustic heads as it is with a 3.5mm headphone and auxiliary the DW DrumLink trigger system input, is used to route the sound and accompanying mesh heads. to speakers, although you could The cymbals are made of the usual potentially use any audio interface rubber and plastic combination if additional discrete audio outputs but are metal coated, giving were required. The same is true of them a realistic, if rather brittle, the Soundworks software — you quality. The standard kit includes can potentially use any virtual two 16-inch crash cymbals, an instrument with the DWe kit, and 18-inch ride and a 14-inch hi-hat. the DWe Control software (used All are three-zone designs, offer to configure the triggers) comes 360-degree triggering and include with a large selection of MIDI maps the ability to choke by way of an to integrate the kit with Superior FSR (Force Sensitive Resistor) Drummer 3, Steven Slate Drums and edge sensor.

The SetList feature is ideal for live use, allowing you to compile a list of songs many more drum instruments.

Using larger, acoustic-size drums and assign each one a preset kit, bpm setting and time signature.

The Soundworks software has for electronic kits has become very several features that pertain primarily to using it with the DWe kit. In addition to popular recently, with the Roland VAD series leading the field, but those kits creating a Favourites list of preset kits, you can also add kits to a SetList. Here are not using premium shells and don’t encourage you to convert them to full acoustic kits. However convincing the ‘virtual’ acoustic kit looks, they still require you can compile a list of songs and assign each a preset kit, bpm setting and time signature. To the left of the SetList window are a metronome and buttons reams of cables to connect each drum and cymbal to the sound module. to step through the kits and songs in order. As you step through your set, each What sets the DWe kit apart is the use of an entirely wireless set of triggers, kit loads in turn. The effectiveness of the SetList feature is dependent on how including those for the cymbals and hi-hat. Wireless drum triggers have been quickly kits will load on your particular computer, but it is a great way of using around for a few years and the DW system is based on the Verstrigger system the Soundworks software in a live context. developed in 2016. It has, as you might expect, been significantly improved The DWe kit, in conjunction with Soundworks and accompanying hardware, upon since then, and offers an incredibly accurate and reliable triggering is an incredibly impressive package. A cost of £6500 to £10,500 for a full kit system that communicates with a dedicated DrumLink hub using Wi-Fi. DW are (depending on the configuration) means it isn’t for the faint‑hearted or financially at pains to point out that this is not Bluetooth or any other low-level form of challenged. However, you get an amazing package for this not inconsiderable communication, and in the demonstrations I’ve seen it clearly works very well. sum: a top-of-the-range DW acoustic kit and some of the most groundbreaking The receiver plugs into your Mac or PC via USB and, in another radical electronic drum technology to hit the market in a very long time. move, DW use a virtual instrument — DW Soundworks, as reviewed in the main

The Instrument Designer offers the same level of detail as any of the preset kit elements. Keep it simple by creating a single shot, or pull out all the stops with up to eight round-robin samples, 19 dynamic layers, 29 articulations and five microphone sources in each of the Direct, Overhead and Ambience groups — or anything in between. Bringing samples into the Instrument Designer window is achieved by clicking the ‘+’ symbol that corresponds to each velocity layer and selecting the required sample from the desired location on your computer hard drive. You can add multiple samples by Shift-clicking, which speeds up the process of adding a set of round-robin samples to each layer.

Banging Drums?

Soundworks is easy to use, well-featured, and as much at home as a virtual instrument working within a DAW as it is in its guise as the sound-generating engine of the DWe kit.

Overall, it has a more raw and natural sound than some other drum VIs I’ve

used, but this may be due to its function as a sound source for a very drummer-focused electronic kit, whose users will be looking for a more realistic playing experience. As this is a DW-branded product, I imagine it’s unlikely we’ll see additional libraries that include other drum brands (as in the case of Toontrack, for example). However, the included libraries certainly cover enough styles and genres for most applications, and the DW website does suggest that additional kits will become available.

In terms of audio processing, the EQ and compressor, coupled with the reverb and saturation modules, allow a significant level of manipulation that should enable users to achieve the precise sound they’re looking for, and the impressive multiple mic options also offer the potential to achieve a huge range of variation in sound from the available kits.

The built-in grooves cover a wide range of styles and are very usable, but I found the composition facilities a little too basic. The lack of bar lines makes

navigation difficult, and although the majority of drum VIs actually don’t allow editing of grooves at all, EZ Drummer 3 offers far more functionality in this area.

Having said that, the ability to export as a WAV or MIDI file does mean you can edit grooves in a DAW. In terms of processing, Soundworks is up there with Toontrack’s flagship Superior Drummer 3, with a user interface that is logically laid out and easy to navigate. It certainly stands up as an instrument in its own right, regardless of whether you’re using the DWe kit with it or not.

If you’re looking for a drum VI with a really extensive library of kits and grooves, Soundworks may not be for you.

If, however, you favour a wide selection of well-recorded, high-end drums, with studio-style processing and a natural feel, you should certainly take a look. $ $199 W www.dwdrums.com W www.roland.com www.soundonsound.com / December 2024

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Arturia KeyLab MkIII Controller Keyboard

‘One keyboard to rule them all’ is very much the order of the day with Arturia’s new KeyLab. ROBIN VINCENT

I

n August 2023, I reviewed the Arturia KeyLab Essential 61 MkIII. It was a decent, lightweight but fully featured MIDI controller with some new and well‑implemented plug-in and DAW integration. Over a year later Arturia are bringing their premium KeyLab up to speed with a fresh MkIII overhaul.

It makes me wonder whether the new KeyLab 61 I have here is just playing catch-up or if it has more to offer than its cut-price sibling.

Before I even got it out of the box I could tell from its sheer weight that this is a premium machine. You could fling the KE61 MkIII around with one hand, whereas this new KeyLab is a definite two-hander. The underbelly is made of metal, with well-placed feet, and so sits resolutely on your desk: unmoving, unbending and really rather handsome.

Compared to the MkII, the refinement of the design is so classy that it looks like it comes from a different league. It feels more like a stage piano or synthesizer than a humble MIDI controller. So often, MIDI controllers are rather chunky bits of plastic; this is the opposite.

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The one I have in for review is the 61-keyed black version and it’s gorgeous.

The off-centred colour screen is set in a square of shiny fingerprint-attracting plastic, surrounded by eight buttons with a lovely silver hi-fi‑style encoder just to the right. Arturia have shaved off a bit of the depth, which helps it feel neater and slimmer, but at the cost of four pads and a row of what were probably useful buttons. On the left side, you have function buttons, transport controls and the remaining 12 pads. On the 61-key version, the pitch and mod wheels are pushed up above the keyboard, which helps keep the width under control.

The right side houses a row of nine encoders and nine faders. These are featureless and sink back into a black background making them almost invisible in low light as if designed by Hotblack Desiato himself. Thankfully, they are touch-sensitive and so the display will tell you which one your finger is on. I feared I might have to resort to adding some Tipex to the encoders and fader caps just so I can see them more clearly.

Out the back, we have proper in and out DIN MIDI ports, three pedal/ footswitch ports and a USB-C connection

Arturia KeyLab MkIII From $499 pros • Great‑feeling premium keyboard.

• Stylish design.

• Very creative arp and Chord system.

• Fantastic DAW and native device control.

• Near perfect integration with Analog Lab. • Great with other Arturia instruments up to a point.

• Comes with a lot of great sounds and instruments.

cons • Seamless control of multiple Arturia instruments is not there yet.

• Interaction within Analog Lab not quite finished.

• Still no control over Arturia effects.

• No polyphonic aftertouch.

summary Arturia ace DAW and device control with the Keylab MkIII while not quite mastering the control of their own instruments. It takes the shine off an otherwise superbly premium MIDI controller keyboard.

that also provides power. There’s also a DC power input if you want to power it differently, but there’s no adaptor in the box. I’m disappointed not to see any CV/ gate outputs.

Turning to the keyboard itself, this is a new design, custom-made by Arturia, with a semi-weighted, synth-action feel.

It’s soft, bouncy and convincing under your fingers. It makes very little noise and feels much nicer than anything else within reach at the moment. Sadly, Arturia haven’t pulled in the polyphonic aftertouch from the PolyBrute 12, which feels like a missed opportunity for their premium controller. But we do have channel aftertouch, which is well implemented, and you don’t tend to set it off by accident. It also has release velocity, but while I understand the concept, I can’t find any way to use it, nor any mention of it in the manual.

Three Mighty Modes The KeyLab has three layers of control. You have Arturia mode, which is custom-designed for Analog Lab and the other Arturia software instruments. You have DAW mode for controlling the mixer and native devices within a supported DAW. And then you have User mode, where you can create custom MIDI configurations to control whatever you like.

If you just want MIDI control over third-party plug-ins that can MIDI Learn, then a User mode patch will do all that by default. In fact, the Arturia mode will do so too, but the display won’t show anything to do with the instrument you’re playing. For more nuanced control, the MIDI Control Centre software provides the tools you need to customise every control on the KeyLab. You can set up and name controls for hardware synths and other bits of gear that are easily recallable. You can also edit it all on the KeyLab itself, without a computer, if you don’t mind digging around in menus.

Arturia Standalone Arturia Mode is where all the magic lies, and in particular that means Analog Lab. The KeyLab comes with Analog Lab Pro and offers a simplified interface over hundreds and hundreds of presets taken from Arturia’s whole range of virtual instruments. The interface is grey and uninviting and likes nothing more than being a list of patches. However, the KeyLab brings some joy to the experience by throwing a small picture of the synth in question onto the display along with the preset name and any parameters you are currently touching.

All the sounds are pre-mapped to the encoder/fader array and ready for your tweaking. The first four knobs are always mapped to the Brightness, Timbre, Time and Movement macros and the remaining knobs and sliders are mapped to other appropriate controls. They can vary

for every preset and every instrument.

Using the buttons around the display and master encoder, you can browse and drill down through the entire library without ever lifting your head to look at the computer. You can combine two sounds into one patch, either layered or split across the keyboard, and then switch between them for tweaking.

The experience really is as close to hardware as it gets. We’ve seen Arturia take this idea further with the AstroLab stage keyboard, where you can store the virtual instrument in the actual keyboard.

The KeyLab MkIII can give a very similar feel, provided you don’t mind taking the computer along.

Analog Lab does have its limitations.

You can’t, for instance, remap any of the controls: you are stuck with whatever Arturia have set up for you. I’ve found that if I don’t have the actual instrument installed, then Analog Lab will complain when I try to open the GUI for it within Analog Lab. It’s like it’s expecting me to already own the V Collection, which is a little presumptuous. The preset editor window doesn’t offer any hardware control other than the EQ and master level, which leaves a row of stompbox effects out of the loop. Arturia have evidently put a lot of work into curating these Analog Lab presets and programming the control connection to the KeyLab. They’ve achieved a pretty seamless level of interaction which is very satisfying but feels a little unfinished in some places.

If you own any other Arturia instruments, you can use Arturia mode directly on them too, although this is where I encountered a bit of a difference of opinion over who should be in charge of what I want to control. As with the curated control in Analog Lab, the first four encoders of every Arturia synth I’ve tried are mapped to the four macros.

Now, this may just be me, but I don’t want this. When I load up a synthesizer Round the back we find three quarter-inch sockets for aux in and expression and sustain pedals, full‑size MIDI I/O ports, an external power supply input and a USB‑C port.

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ARTURIA KEYL AB MKIII

The KeyLab MkIII is also available in white...

like the Mini V or the Jun-6 V, I want to control the filter, but I can’t because those first two knobs are mapped to Brightness and Timbre. Those two macros might well be controlling the filter, but they tend to be scaled and are often actually mapped to something else or a combination of things. In the right context, the macro might be perfect, but more often than not I just want the filter cutoff and resonance on those two first knobs, with the full range of movement and nothing else — please.

I could remap them, which sounds sensible, but I ran into a problem where the filter behaved strangely if I mapped it to the first two encoders. Any other controls map without any issues. So there’s something odd at play. It’s probably just a bug that Arturia haven’t noticed, because why would anyone not want to use the beautifully curated mappings that they’ve given us?

In every instance a Mixer option throws level and panning control on the faders and encoders. All the transport controls work properly, and it feels like a solid, regular, basic DAW controller.

It’s when you activate ‘Plug‑in’ mode (also known as Device mode in some DAWs) that things start to get interesting.

All the Devices in Live and Bitwig, whether instruments or effects, and all the included plug-ins in FL Studio, were automatically mapped and ready to go. It’s all there, with no fuss, labelled correctly on the screen. It’s a remarkably

that you can’t map anything while in DAW mode, and this is the same for all the DAWs I’ve tried. The Plug‑in or Device mode that gives excellent control over in-built plug-ins doesn’t offer any sort of MIDI control over anything else.

It’s really counter‑intuitive when you’re controlling an FL Studio plug-in or a Bitwig Device to have to switch out to a completely different mode to control an Arturia or third-party instrument.

And then the control over Arturia instruments was surprisingly inconsistent.

I found that you could load up one Arturia synth and it would work fine; you could move from track to track, switch to Arturia mode and the GUI would come up with everything perfectly mapped for control. If you loaded up the same instrument on another track, say multiple Analog Labs, they would all work flawlessly. However, if you loaded up a different Arturia instrument the KeyLab would get hung up on the first one and not reflect the move to another synth on the display. It’s like the display had crashed. I tried this out on multiple DAWs, different computers and even phoned a couple of friends, and they all reported similar outcomes.

You can absolutely see what Arturia are aiming for, and when it works, it’s superb, but it’s just not coming off reliably enough, which makes it difficult to fully trust and you end up using a User mode to map controls the old-fashioned way. Hopefully, these things will get ironed out in time

“The Keylab MkIII feels like a beautifully made MIDI controller with a premium keyboard and some really useful features.”

DAW Control Scripted DAW Control was introduced with the KE MkIII and did a good job of offering more in-depth and specific control over the familiar Mackie/ HUI control you get with most MIDI controllers. The KeyLab MkIII supports the same list of DAWs, including Logic Pro, Cubase, Ableton Live, Bitwig Studio and FL Studio. For this review, I knocked around in those last three.

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slick and seamless instrument-controlling experience. One of my criticisms of the KeyLab Essential was that you couldn’t skip to the next Device in a chain, but now you can by simply holding a button and turning the encoder. This is how to do DAW control.

In Bitwig and Ableton Live the 4x3 pads become clip launchers, in full colour, and have a genuine ability to useful in their limited view and range. In FL Studio they come into play as steps and pattern selectors for the step sequencer, which is a nice touch. Otherwise they are decent as drum pads with a good feel and velocity sensitivity.

However, it’s not all plain sailing.

One thing you realise pretty quickly is

because they undermine what is otherwise a sublime controlling experience.

Arp Arpeggiators are very useful but often very standard affairs, and so my expectations here were fairly low. I’m thrilled to report that the KeyLab has a bunch of really cool features that could make it unexpectedly fun and interesting.

It starts off looking normal enough, with a handful of uninspiring modes such as time division, octave range, swing and bpm. But if you press the Random button, you then get to play with the probabilistic nature behind the curtain. You can dial down the likelihood of each note triggering, you can increase the possibility of notes being skipped, you can mess about with the division, flow out and into octaves, push the gate length about, screw in some ratchets and float the velocity about the place. Suddenly, you’ve got a whole wave of interesting things happening and if you’re not sure what to do next, just hit the dice for instant semi-controlled mayhem.

Chords & Scales Chord and Scale modes are very common in controllers, but the KeyLab makes good use of the display to give you a visual representation of what’s going on. You can select the root note and type of scale, and the result is shown on the display across two octaves of keys. There’s a good selection of scales, and you can create your own by turning keys on and off.

When you play, any notes outside the scale are pulled into line, so you never play a wrong note.

Chord mode is similarly solid with lots of types, added bass notes, revoicing and a really nice strumming feature. With the strumming, you can chose a direction and speed, like with an arpeggiator, but you can also add ‘Spread’, which will do interesting things

like increase the range of notes based on velocity or aftertouch. You can create your own chords, and that can be as simple as holding the Chord button and playing what you’re after.

The only thing missing would be a quicker way to move between chord types, like maybe having a handful of types mapped to the mod wheel or a spare encoder so you could move between them in performance.

Conclusion Putting aside the Arturia mode for a moment, the KeyLab MkIII feels like a beautifully made MIDI controller with a premium keyboard and some really useful features. The Chord and Scale modes are nicely done, the arp is really great and fun to use, and the display, knobs and sliders are all top-notch. Working within the three DAWs I tried, the mixer and transport control is solid and the Device/Plug‑in mapping to the DAWs’ own plug-ins is brilliant. It’s an almost seamless experience. What would make this side of the equation perfect for me would be to have a third button within the DAW mode that lets you MIDI map to any other synth. It seems really weird to me that you can’t do that without dropping out of DAW mode and into a User mode.

Arturia’s vision for a seamless virtual instrument-driven experience is almost there, but unfortunately it currently misses the mark. When it works then the KeyLab-to-Analog Lab connection is fantastic and feels like a hardware synthesizer. It’s good too on other Arturia instruments, provided you stay within accepted parameters and do as you’re told. If Arturia can sort out the issues of remapping the filter and switching between their own synths within a DAW, then they’ll have a wonderfully integrated system.

I think because other companies like Native Instruments, Novation and Akai Pro have similar systems of integrated control in place, our expectations of how well that works are much higher than they used to be. Arturia have shown that they have the potential to be all over this; they just have to squash the bugs. $ KeyLab MkIII 49 $499, KeyLab MkIII 61 $549.

W www.arturia.com

www.soundonsound.com / December 2024

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Wes Audio ngTube EQ

Dual-channel Equaliser

Have Wes Audio achieved the perfect union of analogue processing and digital control?

M AT T H O U G H TO N

W

es Audio’s ng (‘next gen’) products always manage to impress me: it seems that every time one of these devices has come to my studio for evaluation there’s been a few interesting new features that make their digitally controlled analogue circuitry that much more convenient to use in a modern studio setting. In recent years it’s been

Wes Audio ngTube EQ $5999 pros • Sounds sublime.

• Immaculate construction.

• Switchable character options.

• The best implementation of digitally controlled analogue gear I’ve yet encountered.

• Lots of hidden functions mean versatility, but never detract from ease of use.

cons • None whatsoever.

summary A four-band, dual-channel analogue EQ with digital control, the ngTube EQ’s sound is as sublime as its feature set is impressive.

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a case of evolution more than revolution, but for their latest release, the ngTube EQ, they’ve made some significant strides, and created what I can say with confidence is the most appealing analogue processor I’ve used to date. If money were no object it would definitely remain in my studio.

Since I last reviewed one of the ng‑series devices, not only have Wes further developed the plug-in remote control side of things (more on that below), but they’ve also progressed the front-panel user controls, taking advantage of touch-responsive knobs, short and long button presses, onboard memory and a miniature but clear and crisp LCD screen, to deliver some incredibly cool features that I’ve not previously seen on analogue outboard. There’s way more to this thing than the ability to save and recall settings with your DAW session, even if that remains a big part of the appeal.

Tubey Or Not Tubey That really is the question! As a dual-channel, passive, four-band, inductor-based parametric EQ, the analogue circuitry that lies at the heart of the ngTube EQ is not dissimilar in concept to that in Manley’s famous Massive Passive. Despite its name, though, the ‘tube’ (valve) stages

in this device are optional: you have, at the push of a button, the choice of a ‘clean and clinical’ electronically balanced output, or a more characterful valve-plus-transformer output stage. In the latter configuration, the Output knob controls amplification based around JJ-Tesla ECC81 (12AT7) and ECC82 (12AU7) valves, which means it determines how hard the signal hits the Carnhill output transformers, and thus how much character is imparted to the signal by transformer saturation. The pad is placed post-transformer, allowing you to set the final output signal level. Cleverly, the current output mode is indicated on the hardware by a change in the colour of the backlit controls (green for clean, red for warmth) so you’re never in any doubt about which setting is selected.

Before the signal reaches those output stages, it first flows through Wes’s familiar THD processor, which can be bypassed or set between 0 and 100 percent, to give you yet more control over the signal coloration.

It then passes through bypassable active high- and low-pass filter stages and the passive inductor-based four-band parametric EQ, which I’ll explore in more detail below. At the time of writing, the otherwise excellent and hugely informative PDF manual offered conflicting advice about

the order in which these sections appear: a signal‑flow diagram suggests the filters are between the THD stage and the parametric bands, while some words elsewhere suggest they come after the EQ bands.

The THD section, which definitely comes first, has been thoughtfully integrated: as you turn this knob clockwise to add harmonics, the signal level flowing into the EQ stages inherently increases, so it also influences the sonic contribution of the inductors. But the output is then automatically attenuated to compensate for that increase in level, so the user experience is simply one of turning the knob one way for more ‘grit’, and turning it the other way to dial things down. Also worth noting is that the THD knob is a touch-sensitive push-turn encoder, and is used to access some secondary functions; more on those below.

First Among Equalisers The high-pass filter can be set to 12 or 24 dB/oct slopes and almost anywhere from 12 to 300 Hz; being a digital control it’s not technically continuous, but with about 80 steps, the resolution is fine enough that you perceive it as such. The low-pass filter offers the same slopes and steps, and the turnover frequency can be set from way down at 6kHz to a stratospheric 50kHz, so all bases are covered. Bypassing the filter is a simple matter of pressing and holding the push-turn frequency knob.

Moving on to the parametric EQ, each of the four bands is identical, other than for their centre frequencies. They can each be switched individually to operate either as a bell or a shelving band, and

The ngTube EQ can connect to a computer for DAW plug-in remote control over USB, or directly or via a LAN using the Ethernet port.

have gain, frequency and Q (bandwidth) knobs. As with all the knobs on this device, these turn continuous rotary encoders rather than pots, and their status is indicated with a circle of LEDs — a particularly elegant arrangement similar to that described in my SOS November 2021 review of the ngBusComp (https://sosm.ag/ wes-ngbuscomp), whereby LED brightness, not just position, is used to give a very precise indication of the setting.

These bands have two (globally switchable per channel) gain modes, ±15dB and ±5dB, which is a fantastic touch: for traditional EQs of this type, manufacturers sometimes offer a dedicated ‘mastering edition’ with a smaller gain range for finer control and, depending on the model, perhaps other features such as switches in place of pots. With Wes’s approach, there’s simply no need for separate mixing and mastering versions: you can access either configuration in the same product. In ±15dB mode, you already have very fine individual gain steps of 0.25dB, which is still fine enough to be automated by your DAW without audible artefacts, but allows you to make bigger changes on the hardware quickly and easily.

Switch to ±5dB and each step becomes an alternating 0.08 or 0.09 dB — marginally better than the 0.1dB claimed in the manual, and more precise than any stepped analogue EQ I’ve encountered previously.

The gain knobs have secondary and tertiary functions too. A double-push on any band’s gain knob inverts its gain, turning a boost into a cut or vice versa. This makes it easy to sweep to identify a problem frequency and then nix it. Neatly, each band can also be bypassed with a single push of its gain knob, and pushing it in and holding for a couple of seconds resets it to unity gain. These ‘hidden’ functions are clearly described in the manual, and while it can take a short time to grow accustomed to them it’s a really neat way to accommodate so much control on an analogue device,

without the front panel feeling cluttered or too different in use from a traditional analogue device.

The bandwidth for each band can individually be set to Q values (in the default constant-Q mode) from a tight-ish 4.0 to a gentle 0.5, while pushing the Q knob toggles it between bell and shelving types. The Q control remains active for the shelves, and with all bands being switchable in this way, you have generous scope to fine-tune the overall EQ curve.

In bell mode, the bands are switchable between constant Q and proportional Q. In the latter configuration, the bell narrows dynamically as more boost/cut is applied.

I should point out that this is a per-channel setting, rather than a per‑band one, and that this mode isn’t (yet) perfectly reflected on the plug-in GUI. Particularly in the dual-channel and Mid-Sides modes, where you can use different Q types on different signals, this feature extends the unit’s utility considerably.

Each of the four bands can be set to one of 12 frequencies and, in combination, the four frequency controls span a generous 22Hz to 28kHz. There’s useful overlap in the individual ranges, which are as follows: low band 22Hz‑1kHz; low-mid 90Hz‑4.8kHz; high-mid 260Hz‑11kHz; high 600Hz‑28kHz. As you can see, there’s even a little overlap between the outermost bands, never mind adjacent ones, and it lets you really dig into the midrange if you need to.

Channel Linking & Presets One thing that’s great about all dual-channel Wes Audio processors is that they can be configured to operate as L-R stereo, Mid-Sides or dual‑mono devices, and the ngTube EQ is no exception. On the hardware, this mode is determined using buttons in the central section between the two channels. When it’s stereo, the left-channel controls operate on both channels. A minor frustration (not a particularly big deal) was that the www.soundonsound.com / December 2024

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right-channel controls in always easy to read. It defaults this mode are inactive. to showing horizontal bar meters Ideally I’d have liked to be for the input and output levels able to use whichever set (ie. two bars per channel) but was closest to hand, but touch any control (no need to there’s a workaround for move it) and, after the briefest of that as I’ll explain shortly. waits, the screen displays that The Mid-Sides encoder control’s parameter and status. comes before the THD For example, the LED rings stage, and the decoder around the Q control already between the EQ and give you a decent indication output stages, so the of the current setting, but rest left control set operates your fingers on the control and on the Mid and the right you’ll see the precise setting on set on the Sides. But the screen. Turn the control and note that the output gain you’ll see the on-screen value and valve/transformer change accordingly. coloration apply only to The touch-responsive the left and right channels. controls aren’t just about In dual-channel mode, information, though; they’re the role of the controls also used to access hidden will, I hope, be obvious. functions. Perhaps the simplest But there’s a channel illustration (and a very practical Parameter Link button application) is the inverse linking at the top of the central of the output gain and Iron A couple of the hardware’s ‘hidden’ features: toggling between the ±5 and ±15 dB gain section and when this is Pad controls. Rest a finger on modes; and the preset menu, accessed by a long press of the Parameter Link button. engaged and you’re in the Iron Pad knob while using either M-S or dual-channel The red backlit buttons denote that the valve output stage is engaged, while they light your other hand to turn the orange in preset mode and (not shown) green if the electronically balanced output is active. mode, the controls for the adjacent output gain control either side of the screen, are used to two channels are linked, such that when (to drive the valve-plus-transformer stage navigate through the presets. There’s you turn, say, the gain or frequency knob on harder) and the pad value will increase also a preset bank feature, so that you the left channel, the right channel’s control accordingly, to deliver attenuation that can recall different banks of A, B, and moves accordingly. This provides that results in approximately unity gain. It’s just C button presets. workaround for using either set of controls about possible to do that one-handed too, It’s superbly well thought out. Since the in stereo. But since engaging parameter by resting a finger on one control and using procedure for storage and recall of presets linking doesn’t override either channel’s a thumb and finger to turn the other. As long and banks is clearly described in the manual settings, it’s more useful than that. (You can as you’re touching both controls, you can I won’t detail it here, but I will say that it’s copy the setting from one channel to the turn either one and achieve the same thing. very quick and intuitive in use. It’s great for other, by the way). Any offsets between the While inverse parameter linking is a neat those who prefer hands-on control and in two channels’ settings are preserved so, trick, it’s perhaps not unique. But I bet particular those working on sessions likely for example, if there’s a low-mid dip on the you’ve not seen anything like the channel to require multiple different setting recalls Sides and a low-mid boost on the Mid, you and band solo functions on an analogue — album mastering sessions, for example, can hit parameter link and adjust the gains EQ before! Just pressing the THD button with different settings for each track. But to of both relative to their starting position with and touching a gain button simultaneously my mind it also makes the hardware that either channel’s gain control. will solo the channel when in dual‑mono or bit more futureproof. If (perish the thought!)

The Parameter Link button also has M-S modes. Or press THD and touch the Wes were one day to cease operating a handy secondary function: a press and frequency knob, and you solo only the band and plug-in support ended, then whatever two-second hold accesses a preset menu — something Wes achieve by temporarily Apple or Microsoft or the DAW developers on the central display. Presets aren’t new to changing the setting of the high- and decided to do in the future, the hardware the ng series, but traditionally the hardware low-pass filters. You can adjust any of the would still support quick and accurate recall has offered only A, B and C buttons (still band’s controls while you’re doing this too, of many settings. Top marks! available here) for quick comparison, and so that ‘sweep and invert’ trick becomes has relied on the remote‑control plug-in incredibly powerful. It’s so quick and easy Advanced Stuff for more extensive preset management. to identify precisely where you want to cut This time, you have 100 onboard memory and with what Q, and it’s the sort of feature I haven’t mentioned the plug-in side of slots, any of whose contents can be loaded I’ve seen only on modern EQ plug-ins like things much yet, but there are still more to the A, B or C buttons. When you’re in FabFilter’s Pro-Q 3. Frankly, it’s astonishing hardware features to explore, including preset mode you’ll know it, because as to see this on an analogue processor, some I believe are unique. The use of well as the screen displaying the preset accessed by front-panel controls. touch-responsive controls allow good use information, all the backlit buttons turn I’m sure I’ve overlooked some other to be made of the LCD screen. Though tiny, orange. The channel bypass buttons, secondary control functions in this write-up; the screen’s resolution is good and it’s

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The first ever USB-C Audio Interfaces with 73 style Class A preamps

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hardware owners

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WES AUDIO NGTUBE EQ

A stand-out feature is the implementation of the valve amp and output transformer stage.

The hardware’s touch-responsive controls allow the valve gain and post-transformer pad to be inversely linked — and the plug-in, of course, achieves the same thing with a single control.

there’s simply too much of this stuff to convey everything in minute detail in a review. But, again, it’s all explained very clearly in the 68-page PDF manual, and if you’re interested in learning more about this side of things, I’d recommend you download it from the Wes Audio website. In fact, if you plan on visiting a store or demo’ing this device somewhere, it’d be well worth browsing through it in advance to make the most of your auditioning time.

GCon control software that runs in the background. The software (for Windows or Mac) was a 440MB download for my Mac, and includes installers for the control plug-ins of Wes’ entire ng range. You get

operating the hardware controls and using the plug-in intuitive and seamless.

But it might also be viewed as a missed opportunity because, free from the design constraints of hardware, plug-in developers have made great strides in GUI design in recent years, and in EQs you can now usually visualise the EQ curve and manipulate it with a mouse, using nodes placed directly on that curve. Wes have taken the bold decision to nudge their control plug-in GUIs in this direction too but, importantly, they’ve not thrown the baby out with the bathwater: they’ve retained the hardware-style controls too.

For the most part, it’s brilliant: in a hybrid setup, I was able to treat it almost like a regular plug-in, with no need to touch or even look at the hardware.

The communication is bidirectional: turn a hardware control and the plug-in updates instantly; adjust a plug-in parameter and the hardware follows suit. Some functions are easier to use in software too: it requires no exceptional dexterity to tweak the linked output and Iron Pad controls on the plug-in! Of course, because of the nature

“It’s the sort of feature I’ve seen only on modern EQ plug-ins like FabFilter’s Pro-Q 3. Frankly, it’s astonishing to see this on an analogue processor.”

Plug-in Power On the rear panel, alongside the analogue I/O (on XLRs) and IEC mains power inlet (which feeds an internal, vertically mounted toroidal transformer), you’ll find both an RJ45 Ethernet socket and a USB-B port. You can connect directly to a computer using either, or to a LAN using Ethernet, and once connected you can update the firmware and/or connect to the remote control plug-in. I had to update the firmware to the latest version for this review, and that side of things was automatic, quick and pain-free.

To use the plug-in you must first have installed it along with Wes Audio’s

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to choose which ones and which versions (VST3/AAX/AU) you install, though. Once the ngTube EQ is connected to your computer and you’ve instantiated the plug-in in a DAW session, you must select the attached hardware — it’s recognised automatically, but it’s possible to have more than one device attached so you must choose it. This setting will thereafter be saved and recalled with your project.

The plug-in’s GUI is strikingly different from the version I last used with other ng hardware. Previously, the interfaces have always been skeuomorphic, strongly resembling the hardware. That’s an approach that makes switching between

of the hardware, with each band having a set menu of frequency options, some things don’t translate quite so smoothly to this paradigm but the generous overlap between bands means it still works pretty well, and if you prefer not to use the nodes, you have the virtual hardware controls.

One tiny limitation I noticed compared with the hardware is that the Listen Mode (band solo) control is only active while clicking on it; release the mouse and this function turns off. It’s not a problem as such, but it does mean you can’t tweak other parameters while in solo/listen mode, as you can on the hardware. I wonder whether a more conventional solo on/off button (or a keyboard modifier for latching control?) might be a tiny improvement. Of course, while holding down the plug-in’s listen control with the mouse you can still adjust any settings using the hardware controls, a MIDI controller or the DAW’s automation

facilities. Another minor issue (and which Wes are aware of) is that the proportional-Q settings aren’t yet reflected on the GUI curve, which seems to be calculated using the constant-Q settings. Neither is a major issue, though, and on the whole the ‘new GUI’ approach works very well; a definite improvement on what went before.

Just Like Heaven?

As you can probably tell, I’ve been absolutely blown away by the ngTube EQ’s feature set, but what about the sound of this thing? In short, it was an absolute joy to hear it in action. The overall character was everything I’d expect to hear in this style of EQ. Smooth and gently flattering in the way that only a good inductor EQ can be.

Even something as simple as lifting the top end of a picked acoustic guitar part, with the cleaner electronically balanced output stage selected, was a delight: an effortlessly

With a frequency curve and node controls to accompany the virtual hardware controls, and a resizeable window that adapts its view according to the channel mode, the plug-in GUI is a significant step forward.

smooth brightness. The same familiar character shone through on pretty much any source I put through it, in fact, and for mastering it was wonderful to have access to such fine settings and the Mid‑Sides mode. But this is more than an EQ, as there’s plenty colour on tap too. If a vocal needs just a bit more edge, the THD control can help. If a bass guitar would benefit from tubes and transformer thickening, that’s also to hand. And on the stereo bus and for stereo mastering, it offered everything I could want. Plug-ins just don’t yet sound this good to me! Pair this with the convenience of plug-in remote control, recall and automation, and the features like the band solo mode, and what you have is a very special processor indeed.

Forget About The Price Tag Of course, dual-channel, four-band valve inductor equalisers don’t come cheap, and neither, usually, does plug-in remote‑controlled analogue gear. So it’s hardly surprising that many of us, particularly the hobbyists out there, will consider the price Wes ask for the ngTube EQ to be an eye-watering amount of money. You’d certainly have to be making good money as an audio professional or be a very well-off amateur to consider acquiring one. But setting aside whether or not it is realistically within your grasp, I don’t think it’s unreasonable to ask this much.

Not only is there a lot of hardware here, but a huge amount of development time must have gone into creating this device.

There’s its convenience and sublime sonic performance to consider. And then throw the price of the broadly comparable analogue competition into the equation (devices such as the Manley Massive Passive or the SPL PassEQ, which all lack the digital remote control and recall functions, offer fewer features and are less versatile), and for those who can afford it, it might look like surprisingly decent value.

This EQ is unique. It’s utterly, amazingly wonderful. In fact, I’m put in mind of the words uttered in awe by Juba in the film Gladiator, on first sighting the Colosseum: “I didn’t know men could build such things.”

Well, now we know they can! Hopefully the fruit of Wes’ labours will start to filter down into products more of us can afford... $ $5999.

T Music Max Distribution +1 614 897 0007 E sales@musicmaxdistribution.com W https://musicmaxdistribution.com W https://wesaudio.com www.soundonsound.com / December 2024

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Kii Seven Active Monitors

Kii’s impressive new cardioid monitors prove there’s still room for innovation in loudspeaker design.

PHIL WARD

I

t was back in the January 2017 issue that we reviewed the then-new and remarkable Kii Three active cardioid monitor. For a significant number of professional engineers and consumer audiophiles, the Kii Three set the benchmark for what is possible from a pair of loudspeakers. And now, after a somewhat extended gestation period, Kii have launched a smaller sibling: the Kii Seven.

I mentioned both the professional and consumer markets, and that’s because Kii have been unusually successful in both sectors with the Three. Kii are one of very few speaker manufacturers who have pulled off that trick. Dynaudio and PMC, for example, have successful consumer

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audio product lines alongside their professional monitors, but not with the same speakers. A Kii Three sold as a studio monitor is the same as a Kii Three sold as a consumer hi-fi speaker. Now, this not only demonstrates, as I’ve always believed, that a good loudspeaker is a good loudspeaker however it’s used, it has also impacted on the design of the Kii Seven. Because, while this review is going to concentrate on its performance and suitability for nearfield monitoring, there’s a lot about the Seven that is aimed at consumer applications. For a start, it is to be sold in singles, for use as a mono, stand-alone audio player. It’s an expensive way of playing Spotify from your kitchen table, but that’s never hampered Bang & Olufsen, for example. It’s perhaps fortunate that this review will primarily concern professional applications, because

at the time of writing some of the consumer streaming features of the Seven aren’t fully implemented. The reasons for this, I suspect, are not so much technical as commercial, because navigating the licence demands required to implement multiple streaming services on the same product is far from trivial. However, Kii say that by the time these words are published, Spotify and Tidal streaming will very likely be available or imminently so, and that Apple Airplay, Qobuz and ROON Ready will follow as soon as the third‑party certification processes are complete.

Going Pro Let’s leave all that consumer streaming malarky behind and concentrate on the Kii Seven in a studio environment. If you’re familiar with the appearance and design of

Kii Seven $9995 pros • Extraordinary combination of bandwidth, low coloration and low distortion.

• Cardioid character means more monitor and less room.

• Reveals mix detail like nothing else.

• Stereo imaging to die for.

cons • None (apart from the cost, obviously).

summary The Kii Seven might have taken a while to arrive, but it was so worth the wait.

It’s an extraordinary monitor.

the Kii Three, the look of the Seven won’t come as a great surprise. It sports a very similar aesthetic and is the same width as the Three, but it’s significantly less tall and less deep. It’s dimensionally better suited to compact studio spaces than the Kii Three – especially in its reduced depth.

The Seven’s enclosure construction is of the same foamed structural polyurethane as the Three, with the same black‑finished aluminium trim panels. It comes in either white or dark grey finishes, and Kii say that the custom colour options of the Kii Three will not be available for the Seven.

Like the Three, the Seven has bass drivers located in each side panel, but it does without the Three’s additional two rear‑panel bass drivers. This perhaps makes for easier installation because, where the Three demands at least 8cm of free space to the rear, the Seven can be used with its rear panel hard up against the wall. In fact, Kii recommend that if a pair

A cutaway render of the tweeter waveguide and lens. The lens broadens the tweeter’s directivity at the top end of its frequency range, while the waveguide narrows it at the lower end.

of Sevens is to be used in close proximity to a rear wall, they are best located right up against it. The Seven does come with one positioning constraint, though, and that concerns space either side to allow the side-mounted bass driver room to ‘breathe’. Kii Designer Bruno Putzeys put it to me this way: “An obstacle should be further away than the size of the Kii Seven or the obstacle, whichever is greatest.” This of course means that you might need to avoid additional monitors sharing a shelf with Kii Sevens, but without wishing to spill the beans before getting to the end of the

review, considering the presence of other monitors if you have a pair of Kii Sevens might be a moot point.

Beam Me Up The side‑panel drivers are nominally 140mm‑diameter, aluminium-diaphragm units — the same as those employed in the Kii Three. And while they nominally work in parallel up to around 125Hz, where they hand over to the midrange driver, things aren’t quite that simple, because the secondary job of the side-mounted bass drivers is to help create the Seven’s

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A touch panel on the rear allows for input selection, LF and HF EQ setting, and phase compensation mode selection.

cardioid characteristics, by overlapping with output from the midrange driver to modify its directionality. This overlap works all the way up to around 900Hz, where the cardioid character that results from the Kii Seven’s enclosure dimensions begins to take effect. In blithely writing that the LF and MF driver outputs “overlap” to yield a cardioid radiation pattern, I’ve simplified something that’s actually been achieved through the combination of deep electro-acoustic understanding and some complex DSP. Kii describe the technique as “beam-forming”, and it involves a beam-forming filter fed from the output of the midrange driver’s band-pass filter. The filter’s output is then routed to the bass driver feed after the output of its low-pass filter. The bass drivers are consequently employed in the overlap region, in parallel with their low‑frequency duties, to manipulate the summed mid and bass driver output to create the cardioid radiation pattern.

The schematic of Diagram 1 illustrates the general arrangement of the Kii Seven signal flow and filtering.

In their conventional role, the bass drivers work with closed-box loading up to a fourth-order (24dB/octave) low-pass roll-off at 125Hz. In contrast to the Kii Three, where cardioid radiation extends all the way down to 50Hz thanks to the combination of both rear‑ and

side‑firing bass drivers, the Kii Seven displays cardioid radiation only from about 125Hz upwards. This might appear to constitute a compromise, but with Kii Sevens likely to be installed much closer to a rear wall than the Kii Threes, it is in practice probably not a particularly

material and volume ask too much of the drivers. The protection-sensing algorithm of the Kii Seven is, according to Kii, much improved on that used on the Three.

Move On Up The Seven’s midrange driver is the same as the one fitted to the Kii Three. It’s a 150mm paper‑diaphragm driver that covers the band from 125Hz to 2kHz, with fourth‑order filters at either end. With its relatively low high-pass crossover filter, the Kii Seven actually works its midrange driver relatively vigorously (more so than in the Kii Three), but with its role likely being nearfield monitoring in relatively smaller spaces, this is unlikely ever to become an issue in practice.

One area in which the Kii Seven offers a significant advance on the Three is in its tweeter technology. The Kii Three employs a tweeter that is fundamentally an off-the-shelf OEM unit, and while the guts (magnet, voice-coil and aluminium/ magnesium dome) of the Seven tweeter are the same, it differs significantly in a couple of respects. Firstly, the Seven tweeter employs ferrofluid in the magnet gap for higher power handling and lower thermal compression, and secondly and most importantly, it employs in-house‑developed waveguide and ‘lens’ components. Designer Bruno Putzeys kindly provided us with a CAD cutaway render of the tweeter waveguide and lens (co-developed by himself and Lars Risbo).

The waveguide and lens work in tandem to overcome an inherent weakness of tweeter waveguides working alone.

While waveguides usefully narrow tweeter dispersion at the lower end of the band so that it better matches dispersion at the top end of the mid band (and simultaneously increase tweeter sensitivity), they do

“Listening to the Kii Seven is an experience genuinely unlike any other...” big deal. And anyway, it’s the cardioid character in the low midrange, where the musical fundamentals of mix elements such as voices reside, that I think really makes the difference.

The likelihood of Sevens being used in close proximity to room boundaries has resulted in Kii including some boundary EQ options within the setup menus of the Kii Control hardware (which I’ll describe further down the page). The boundary EQ operates simply through bass driver gain adjustment, and is set to -6dB by default.

This means that when you first switch on and listen to a pair of Kii Sevens, if they’re not located close to the wall, chances are they’ll sound bass-light.

Like the Kii Three, the Seven employs a dynamic, look-ahead bass driver protection algorithm that dials back the bass level should the programme

The Kii Control unit adds additional inputs to the Kii Seven (coaxial and optical S/PDIF, USB and Bluetooth), and also provides more detailed control over the speaker EQ options. It can also perform some monitor control functions, such as mono summing and Mid-Sides auditioning.

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nothing to aid the less desirable narrowing of tweeter dispersion at higher frequencies (above, say, 8kHz). So the aim of the Kii Seven tweeter lens is to pre-condition the tweeter radiation wavefront so that, while the waveguide narrows its dispersion at lower frequencies, the lens widens dispersion at higher frequencies. The result is constant tweeter dispersion right up to 15kHz. Bruno Putzeys believes that “the extremely engaging sound of the Seven owes a lot to this innovation.”

Amps Leaving drivers behind, the Kii Seven amplification comprises Putzeys‑designed Purifi Eigentakt Class-D amplifiers for the midrange driver and tweeter, with Pascal S-PRO2 amplifier modules for the bass drivers. The total amplification power available is 600W, shared among the drivers. A neat amplification trick of the Kii Three, one that I really like, is repeated on the Kii Seven. The trick is known as ‘current drive’ and it’s one I’m continually surprised other active monitor manufacturers don’t embrace (I only know for certain of one other major manufacturer regularly using it).

Current drive was first proposed

by Dr Malcolm Hawksford of Essex University in a paper presented to the Audio Engineering Society in 1989. The technique involves placing a small resistor in series with the driver and using the voltage measured across it (which will be proportional to the current flowing through the voice coil) as a feedback signal. In

Diagram 1: A signal flow diagram of the crossover network and beam-forming filter. Note that the beam-forming signal is derived from the mid band, but sent to the LF drivers.

a conventional amplification system, the amplifier output voltage follows the input voltage and delivers current simply depending on the instantaneous driver impedance. A current-drive amplifier, by contrast, will compare its input voltage with the measured voice-coil current and adjust the output voltage gain such that they track. So now, the voice-coil current depends on the input voltage and not on the driver impedance. The amplifier delivers any voltage required to make the voice-coil current track the input voltage, which is significant because a driver’s acoustic output is proportional to the current flowing in its voice coil, not the voltage across its terminals.

Channel One Mk

The primary advantage of current drive is reduced distortion, especially that caused by hysteresis nonlinearities in a driver’s magnet system. But it also fixes thermal compression, because the increase in impedance (and loss of output level) that results from a rise in voice-coil temperature is ‘seen’ by the current feedback as an error that results in correction by increasing the output voltage.

Panel Show On the rear panel, along the lower edge, are mains power, audio input, control and network sockets. The audio inputs comprise an analogue combi socket for

The perfect front-end for the modern producer.

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Compact and competent The next generation of Channel-Strips – For more information: www.soundonsound.com www.spl.audio / December 2024

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KII SEVEN

XLR or jack, and an XLR for AES digital.

The control sockets comprise in and thru RJ45 sockets for connection to the Kii Control hardware. The in socket connects to the Kii Control itself and the thru connects to the second speaker of the pair (Kii Sevens can be further daisy-chained for multi‑channel systems). Above the control sockets are twin network sockets for connection of Kii Sevens to a Dante or AES67 audio‑over‑IP network.

Above the connection is a touch control panel that offers input selection, standby, low- and high‑frequency EQ, reset, and latency options. The latency options engage or disengage the Kii Seven’s wide‑band phase compensation.

With Optimum latency engaged, the Kii Seven is minimum‑phase compensated over the entire audible band, resulting in around 70ms of overall latency. With phase compensation disengaged, latency falls to around 7ms. The argument for phase compensation is that it corrects for the various phenomena within both the electronic and electro-acoustic domains of speakers that result in time‑domain errors, and as time is a vitally important element of the way we perceive sound, a speaker that’s free of such errors ought to offer subjective benefits.

Diagram 2: The axial frequency response of the Kii Seven at 80dB SPL at 1m (red trace). Second- and third-harmonic distortion are shown in green and blue, respectively.

Diagram 3: As Diagram 2, but measured at 90dB SPL.

Kii Control I’ve referred to the Kii Control hardware a couple of times now, so it’s probably about time I described it more fully.

The Kii Control is a USB bus‑powered compact desktop device that primarily offers input selection and volume control for connected Kii monitors. It also itself provides S/PDIF, digital optical, USB and Bluetooth inputs. As described earlier, the Kii Control connects to Kii monitors via an Ethernet-like RJ45‑terminated cable. In addition to its volume and input functions, the Kii Control provides access to far more comprehensive setup and configuration options than are available from the Kii Seven’s rear‑panel buttons.

For example, the simple LF and HF EQ control of the rear panel is extended to include turnover frequencies, and an additional advanced EQ option offers a further eight bands of parametric EQ. The boundary EQ described earlier can also be adjusted using the Kii Control setup menu.

Additional options include the ability to set a reference level, configure and save setup presets, adjust lip sync delay, perform mono or Mid/Sides auditioning,

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Diagram 4: Comparing the on-axis frequency response (red trace) with measurements taken 20, 40, 60 and 80 degrees horizontally off axis (blue, green and orange traces, respectively). The divergence of the traces above around 100Hz is a direct result of the Kii Seven’s cardioid response.

indicate limiter action and set the standby behaviour. While it’s perfectly possible to install and use a pair of Kii Sevens without a Kii Control, its extra cost is relatively modest, and my feeling is that the additional configuration facilities provided make it all but obligatory, particularly for professional monitoring applications.

Graphic Detail Since I tested the Kii Three back in 2017, I’ve found a new and much more effective speaker measuring space, so I was very much looking forward to measuring the Kii Seven there. Diagrams 2 and 3 illustrate the axial frequency response and harmonic distortion at 80dB and 90dB (at 1m) SPL. The first thing to note

about the axial response is that the default -6dB boundary compensation is obviously apparent. The second thing to note is that the amplitude response is generally nicely flat and that the harmonic distortion performance is, as expected, exceptional. At 80dB SPL the third harmonic effectively bumps along the measurement noise floor, at around -65dB from a couple of hundred Hertz upwards.

Now, -65dB is equivalent to about 0.05%, and for a speaker, that’s extraordinarily low.

At 90dB SPL the distortion clearly rises a little, but it’s still remarkably low.

Diagram 4 illustrates where the Kii Seven differs from the majority of monitors, and that’s in its horizontal dispersion. The curves illustrate the

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of depth within it, is perhaps the best I’ve ever heard in my own listening space (this is undoubtedly aided by the lack of room influence). I was particularly struck for example by how well very fine details, right on the edge of audibility, retained their image focus and position. Presenting layers of detail deep, deep into a mix is a Kii Seven speciality and it’s an incredibly useful quality for a monitoring tool.

So the Kii Seven delivers all that I hoped it would, but there are a couple of minor subjective head-scratches. Firstly, further out into the room, in the reverberant field, I felt the very low end of the Kii Seven was a little overcooked. This of course is likely to be a room‑dependent phenomenon, and it could quite possibly be fixed using the EQ facilities of the Kiii Control, but it was something I’ve not encountered in my room before to the same degree. It didn’t impact on the Seven’s primary job of nearfield monitoring, so is more intriguing than significant. Secondly, I initially found the default balance of the Kii Seven to benefit from a broad (Q=0.7) 1.5dB or so boost in the 200Hz region, just to warm the low midrange a little, and an HF cut of 1dB or so from 3kHz up (I found the default balance just a little bright). I suspect my low‑mid boost preference might well be a result of less side‑ and rear‑wall energy arriving at the nearfield listening position than I am used to. However, if I had started mixing from cold on the Kii Seven, their default balance might potentially have led me astray a little.

I guess, though, with all monitors there’s a period of acclimatisation to go through; it just might take a little bit longer with the Kii Seven. But to be honest, any time spent listening to the Kii Seven is time well spent. $ $9995 including Kii Control.

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Listening to the Kii Seven is an experience genuinely unlike any other, except for listening to the Kii Three perhaps. I write this not just because, by any subjective assessment, it is an extraordinarily impressive monitoring tool, but because its cardioid nature drives the listening space differently.

As a result, its subjective character is inherently unlike that of a conventional monitor. Specifically, the Kii Seven pushes the transition in the room from the nearfield to the reverberant field further back. So, if you’re using the Kii Seven as a nearfield monitor, at a typical nearfield monitoring listening distance, you hear rather less room and rather more monitor. In particular, significantly less rear‑ and side-wall reflection energy arrives at the listening position. And the implications of that for critical monitoring are that you have a better chance of hearing midrange details further down into the mix, because they are less confused and masked by the delayed arrival of side‑ and rear‑wall reflections. At least, that’s what I think is going on.

So a significant element of Kii Seven’s performance derives from the fact that it is less coloured by the room, but that could be a double-edged sword, because it means that if the Sevens were at all flawed, there’d be nowhere to hide. Luckily (actually, I suspect no luck at all was involved) the Kii Seven is very far from flawed.

I listened almost exclusively in minimum phase mode (I found it added a little extra focus to stereo imaging), and from the first moment of listening to the first track (‘Inside Out’ from Athena Andreadis’ first album) the Kii Seven revealed itself to be extraordinary.

The midrange sings with a beautifully neutral and explicit clarity, high frequencies are unimaginably detailed yet completely natural, and the bass is far more extended than the enclosure size would suggest, yet simultaneously dynamic and accurately pitched. And all this is available at pretty much any sensible volume level from quiet to very loud. Stereo imaging, and the resolution

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axial frequency response again, along with overlays for 20, 40 and 60 degrees off-axis. The cardioid nature of the Kii Seven, from not much above 100Hz, is clearly apparent.

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Peluso Microphone Lab www.soundonsound.com / December 2024

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Native Instruments Kontakt 8

Software Sampler

NI’s long-established software sampler gains a raft of new features and other improvements.

JOHN WALDEN

O

ver the last 20 or more years Kontakt has become an essential tool for countless musicians and producers. A new version is, therefore, something that both the user and developer communities will look forward to with anticipation and, perhaps, just a little trepidation. So, with the release of Kontakt 8, what have Native Instruments done to refine and improve upon one of the world’s most widely used pieces of music software?

First Kontakt As before, Kontakt 8 is available in both full (paid‑for) and streamlined Player (free to use but with some limitations) versions.

Both can run standalone or in a plug-in (VST3, AU and AAX) format. As well as an individual product, both Kontakt versions are also available as part of NI’s various Komplete bundles. These bundles have also been updated and a separate SOS review of Komplete 15 will follow shortly.

For existing Kontakt users, the reworking of the UI will be the most immediately obvious change in v8.

This exploits some significant graphical enhancements under the hood but,

Native Instruments Kontakt 8 $299 pros • Conflux brings hybrid sample/synth technology to Kontakt.

• Chords, Phrases and Leap add plenty of new creative options.

• Lots of interesting under-the-hood changes for developers to exploit.

cons • Not every Kontakt user might see the new creative tools as essential to their personal workflow?

summary Kontakt 8 brings a slick UI revamp and some impressive new creative options.

It will be very interesting to see where the Kontakt development community take some of the underlying technology.

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Kontakt 8 features a UI refresh with several new content categories available within the main Browser.

if you prefer the previous look, just opt for Classic View from the View menu and you can switch back to the earlier design.

There are a number of notable changes in the redesign. For example, on the Library page, there are now six main tabs — Instrument, Combined, Tools, Leap, Loops and One-shots — and I’ll touch on all of these below. Once you have selected and loaded a specific preset from a library, as before, the main instrument UI dominates the Kontakt window, but you now get the Side Pane (it can be toggled on/off as preferred) on the left. This contains sections for Tools (more on these in a minute) and Instruments but, very usefully, it provides a compact version of the browser, giving you instant access to your entire Kontakt content including the presets for any loaded instrument.

Conflux Instrument The full version of Kontakt includes a significant new instrument called Conflux. NI describe this as a something of a first in that it blends synthesis with Kontakt’s usual sample-based sound creation in a hybrid design. This is provided by a complete reworking of

Kontakt’s existing wavetable synthesis engine, adding FM, phase modulation and ring modulation wavetable oscillators to the existing PPG wavetables.

Conflux’s control set is laid out across three pages; Play, Edit and Settings.

The first of these provides easy access to the presets and a set of macro-style controls but the Edit page is where the full potential of the synth engine can be accessed. Here, the Source panel provides access to the Wavetable and Sample oscillator choices, and the ability to blend and adjust the balance of these two elements. The Audio Mod panel adds an additional sound source, offering three modes; Audio Mod, Shaper and Oscillator. The first two provide additional manipulation of the wavetable source but the third provides an independent virtual‑analogue oscillator that can be blended into the overall sound.

The engine offers a filter, two LFOs, two envelopes, two flexible effects slots, delay, reverb and a step animator.

Alongside the six macro controls (each of which can be used to modulate multiple parameters), this makes for plenty of modulation sources and lots of suitable modulation targets. Linking sources to

targets is all done graphically and is easy to configure, although a MIDI Learn option for linking the six macro knobs to an external hardware controller would be a nice thing to see. That said, hardware control was easy enough to map using the features in my DAW host.

Sonically, Conflux is impressive and involving. It perhaps won’t be an obvious pick for your bread-and-butter pop or EDM sounds, but for media composers or the more experimentally minded looking for some quirky sonic elements or evolving otherworldly soundscapes, it’s a great choice.

New Tool Box Kontakt 8 includes a new Tools concept.

This first iteration sees two such Tools introduced; Chords and Phrases. I’ll explore these below, but two things are worth noting. First, these Tools can be used with every Kontakt instrument.

As shown in the screenshots, the side pane includes a Tools section at the top, allowing you to add Tools that might then be utilised with whatever instruments are also present.

Second, and quite impressively, these new Tools are included in both the full (paid) and Player (free) versions of Kontakt with no restrictions; every Kontakt user gets access. NI have suggested that there will be further Tools to come so watch this space.

Chords Systems for triggering full chords via a single MIDI note are a familiar concept and many DAWs or virtual instruments have such options built into their feature set. However, the new Chords Tool embeds chord triggering directly into

Kontakt and is usable with any Kontakt instrument. The Chords UI shows the seven chords with a chord set as concentric circle graphics that represent the chord’s notes. The tag-based browser lets you easily search the impressive selection of presets to find either a new chord set or an individual chord, while dice icons allow you to randomly change chord sets or individual chords. There are also plenty of options to adjust or adapt the presets. For example, at the top right you can adjust the key/scale used and the individual chords within the currently loaded set will adapt themselves automatically. If you click on an individual chord circle, you can access the ‘record’ option and manually create a chord, including its full voicing, and you can save your own chord set presets (just right-click on the Chords entry in the side pane to access the appropriate drop-down menu).

Chords does, of course, let you create a full chord performance (including velocity sensitivity) with a single finger, and chord labels can be displayed as you trigger them.

You can also apply a little Humanize (including velocity/ timing variation and note skipping) or Strum (up or down with control over speed) while performing.

If you record your performance, it’s Conflux’s Edit page provides plenty of sound design options for those wanting to go beyond the included presets.

The new Chords Tool embeds chord triggering directly within Kontakt for use with any of your installed libraries.

the trigger notes that get recorded (and then re-triggered on playback) rather than the full chord voicing. You can drag-and-drop MIDI data for individual chords, or the full chord set, to your DAW, but you would then have to build a performance from these MIDI notes. It would be a nice addition if Chords had an ‘export full MIDI note data’ option built into its UI. I did think Kontakt’s Send MIDI To Outside World setting (found in the Options panel) might make this possible, but I couldn’t seem to make it function as I expected. That comment aside though, Chords is both very easy to use and the presets contain lots of interesting chord set combinations that — paired with the right sound — can be instantly inspiring.

Phrases With a similar UI, Phrases offers you a set of seven melodic phrases that can be triggered via the white keys. Again, a circular display provides a graphical indication of the note pattern within each phrase. Key and scale can be adjusted, and the phrases will adjust to fit.

Phrase triggering behaviour can also be adjusted (including a Latch option) and, via the Edit page (accessed from the small pencil icon located top right), you can adjust a whole range of parameters such as note length, quantise, velocity, phrase length and playback direction, either globally (all phrases in a set) or to a single phrase. Equally, things like the Rotate and Invert knobs, which both create interesting changes in the performance of a specific phrase, can be automated in your DAW to add further performance variations on the fly. The www.soundonsound.com / December 2024

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N ATI V E INS T RU M E N T S KON TA K T 8

now drag and drop content from any installed Expansion packs (those aimed at Maschine) directly into Leap. Equally, you can drag and drop other sample content from your OS browser onto a Leap key slot. All of Leap’s tools are then available to edit or adjust these samples, letting you repurpose that content to create your own Leap presets.

If loops and one-shots are a regular part of your music production workflow, Leap brings that directly into Kontakt in a way that’s both fun and flexible. For those prepared to dig in, there is some very impressive functionality here.

Multi Out, Combo In The Phrases Tool provides instant melodic part inspiration when used with any suitable sound source, as shown here with the new Piano Uno instrument that’s included with both the full and Player versions of Kontakt 8.

large Randomize button randomises many of the time, pitch and velocity parameters, and the mod wheel acts as a global dynamics control, letting you add further performance variation. As with Chords, you can save your own Phrases presets.

While you can drag and drop Phrases MIDI to your DAW for further editing, Phrases doesn’t currently let you manually edit or create phrases of your own. If you record a Phrases performance, you get the trigger notes rather than the full melodic performance.

That said, Phrases is a great source of inspiration to get an idea started.

Leap Kontakt’s other major new addition is Leap, which NI describe as a ‘sampler within a sampler’. Leap’s UI is divided into four main pages; Play, Edit, Send FX and Perform FX. Users of Output’s Arcade might have a sense of the familiar with Leap’s Play page. This provides 16 keys/slots for triggering sample loops/one-shots mapped to a block of white keys. Across the associated black keys are a series of performance effects — stutters, speed changes, pitch‑shift and so on — that let you process the triggered sample playback in real time. You can also switch the engine to Single mode, which maps a selected sample across the keyboard range so it can be played as a melodic instrument. It’s worth noting that, as with Chords and Phrases, Leap is included in both Kontakt and Kontakt Player.

The Edit page lets you change how a specific sample is used on triggering (loop, one-shot, etc.) and

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provides settings to adjust the sample length, pitch, tempo, filtering and basic amplitude envelope controls. The Send FX section offers delay and reverb options, while the Perform FX page lets you customise which of the performance effects are associated with each black key and tweak how they behave. The UI also includes eight macro knobs for real-time control of further effects options.

Leap also has its own Expansions format and 12 of these are included with the full version of Kontakt (just one with the Player version). I’m sure NI will not be slow to make additional Leap Expansions available via their online shop. However, rather wonderfully, via the Loops and One-shots pages of Kontakt’s new Browser, you can

While not a revolution, it’s worth noting that Kontakt’s multi format is referred to as the Combined format. Yes, it’s much the same thing, allowing you to load a preset that contains a combination of instruments, but Combined presets now also accommodate Tools and many of the new Combined presets supplied with Kontakt 8 are based around an instance of Chords or Phrases and one of the sounds from the Factory Library as demonstrations of this new functionality. You can, of course, create your own Combined presets and, as far as I can tell, under the hood, it’s still the NKM file format that’s being used so existing multi presets load as before.

Deep Dive Development While the vast majority of SOS readers will fall into the Kontakt user community, NI obviously have

Leap adds a very creative loop and one-shot performance instrument and, as well as its own Expansion format, you can import your own samples via drag-and-drop to build your own Leap presets.

to encourage their developer community too. As such, Kontakt 8 brings a new development toolkit — Complete Instrument Building Toolkit 1.0 — allowing them to fully exploit all the new elements of Kontakt 8 for improved/expanded scripting and enhanced UI building.

In addition, developers have access to the new elements of the wavetable synth engine included within Conflux, an SDK for Tools and an SDK for Leap expansions.

The development community will undoubtedly be keen to offer us users some tempting products that exploit all these new options.

Watch this space.

As suggested

by Leap’s Edit page, the instrument features plenty of options for those looking to get creative with their loop and one-shot samples.

Make Contact With Kontakt 8?

As a new purchase, Kontakt is obviously a significant investment for many but this platform — and the huge range of libraries it supports — has provided the sonic power behind countless song and film score hits. As a software sampler, it is the industry standard and v8 just refines and expands upon the software’s already established capabilities.

Personally, I’m excited to see how developers might exploit the new elements within Kontakt’s engine. However, while the UI and under-the-hood enhancements are undoubtedly welcome, for individual users, the decision to upgrade may rest upon whether the highlight additions — Conflux, Chords, Phrases or Leap — enhance their personal workflow or creativity. That might not be the case for everyone, but it’s easy to see how all of these new features might appeal to specific sub-sets of Kontakt users, and the upgrade pricing is relatively modest.

NI’s software sampler is the very definition of an ‘industry standard’. The latest release does nothing to undermine that position and, if Kontakt is already part of your daily workflow, Kontakt 8 has the potential to enhance that workflow with a collection of new and inspiring creative options. $ $299. Upgrade pricing available.

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PreSonus Studio One Pro 7 DAW Software Studio One Pro 7 is a solid upgrade with something for almost everyone.

ROBIN VINCENT

I

n their typically understated manner, PreSonus have declared that Studio One Pro 7 is the “One DAW for All”. It sounds ambitious, but with the new version I have before me, that claim is almost but not entirely unreasonable. Except that the tagline actually refers to the fact that PreSonus have killed off Studio One Prime and Artist, so now it’s all about Studio One Pro. With this adjustment comes a new pricing structure, giving you a perpetual licence and year of updates for $199. You can upgrade that every year for $149 or opt for the Studio One Pro + subscription for $179 per annum, which gets you a lot of added extras and includes the perpetual licence. PreSonus are also trying to stick to a more predictable feature release schedule.

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Significant updates will come three or four times a year, rather than holding things back for a single major update every two years, as has been the case up until now.

For this review, I’m looking at the regular Studio One Pro 7 without any additional thrills or spills, but the first thing you notice is that it comes with many of them anyway. Pro 7 includes the Batch Converter, Presence XT Editor, Lead Architect and Deep Flight One, all of which were additional extras for version 6. So, it’s less money for more stuff.

I like this update already.

The top new features in version 7 include AI-powered stem separation, Splice integration, integrated Launcher, Global Transpose and a new mode for Impact, all of which I’ll dip into. But there are more things twinkling away in this shiny new version that are worth a look, not least of which is the unexpected CV Instrument casually hanging out in the Instruments folder.

AI-powered Stem Separation Stem separation is all the rage these days, and Studio One is not one to be

left behind for long. On the website, PreSonus say that the technology is still in its infancy, so further performance/ quality improvements are planned.

I confess to sharing their restrained level of scepticism. That is, until I came up with something to use it for, and now I’m having far too much fun. I have a lot of old songs knocking around drives that are no longer attached to multitrack projects, either because of carelessness, misadventure or because software is no longer available.

Using stem separation, I can regain a certain amount of control, get back into the mix and breathe new life into these tunes.

The process is very simple. Drag your mixed music file onto a track, right-click, and you’ll find Separate Stems under the Audio menu. It throws up a window where you can select which of the four stems you want to extract: Vocals, Drums, Bass and Other. There’s also a check box you can tick if you want to remix it directly back into a single file again. So, for instance, you can create a mix without vocals with a single click. That’s a neat little feature.

Otherwise, the four stems are thrown into a folder underneath the original track, which has now been disabled.

Does it work? Yes and no. Yes, because this really is remarkable technology, the process is smooth and the results are largely clean and well separated. But also no, for more or less the same reasons. I found when removing vocals that there were often ghostly voices in reverb tails and sort of squashed artefacts in the mid frequencies where the singer used to be.

When soloing the vocals, there’s often a lingering sense and sometimes a definite bleed of other instruments. I guess that is the stem separation conundrum; the results tend to be ‘not bad’ or ‘pretty good’ rather than drop-dead fantastic.

However, for reworking old mixes, isolating reasonably clean samples, and for working up new ideas from existing material, it has found itself to be useful.

I ran some comparisons with the stem separation in Image Line’s FL Studio.

The results were largely similar, with FL Studio occasionally doing a slightly better job of isolating vocals, but they all suffer from the same separation artefacts.

It can vary a lot depending on the source material, and it makes you wish there were some settings you could adjust to tweak the process in some way.

Tempo Map Detection One particular stem separation scenario wraps in nicely with the new ability to generate a tempo map from a recorded performance. You can imagine receiving a mixed demo from someone, maybe voice and guitar or piano, and you’d like to flesh out a project around that performance.

Pulling the voice and accompaniment apart gives you much more room to develop the music around it. However, it probably wasn’t recorded to a click track, so the timing naturally ebbs and flows a bit.

Fitting it into the Studio One grid can be a challenge, and while there are all sorts of tools, from groove extractions for quantising to time-stretching transients to get things under control, it doesn’t always feel right.

With Tempo Map Detection, you can make the grid follow the changing tempo of the performance rather than forcing the performance to fit the grid.

Under the Audio menu you’ll find an option to ‘extract to the tempo track’, and all the variations in tempo will appear as a wiggly line. You can now add patterns and loops and have them all follow the tempo of the original piece.

Splice Integration Third-party integrations tend to make me feel suspicious. More often than not, they are trying to sell me something.

Some, like the TuneCore integration, save you a few clicks but require you to sign on with the distribution service. Others, like Melodyne, give you fantastic tools along with the suggestion that things could be even better if you upgraded.

So, I was undecided how excited I should be about the integration into the browser of sample‑library powerhouse Splice.

If you are already a subscriber, then this is all gravy, but if you’re not, then the question is, will it bring awesome things to your project, or is it a ruse to opt you into an unexpected subscription?

In between the Loops and Files tabs in the browser we now have ‘Splice’. Click on it, and you find a search box, some

PreSonus Studio One Pro 7 $199 pros • The Launcher is a solid development.

• Pricing and included content are good value for money.

• Polyrhythmic melodic patterns.

• CV Instrument.

• Splice integration is amazing for subscribers.

• The inevitable stem separation does all right.

cons • You need an extra subscription to use Splice.

• Launcher features are quite rudimentary.

• Nothing new for mastering engineers or live performers.

• No new synths or plug-ins.

summary Studio One Pro 7 keeps itself in the game with stem separation and a clip launcher, while introducing attractive pricing and update schedules. It remains a solid and mature platform that can serve the needs of a wide range of users with genuine style and substance.

categories to browse and a big box with the words ‘Search With Sound’ on it. The idea is that you drag a clip or two, a chorus, verse or anything up to eight bars, into the box, and its AI algorithms will go off and retrieve a bunch of loops that fit with what you’re doing. It’s pretty much on the money every time. The loops are tempo-matched and in the right key, ready to be dropped into your project — this is brilliant. It beats scrolling through old sample folders looking for something that might but usually won’t fit.

Initially it offers up around 50 possibilities taken from a spread of categories covering

We all love separating stems — well now you can do it in Studio One.

www.soundonsound.com / December 2024

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PRESONUS STUDIO ONE PRO 7

Tempo Map Detection adjusts the grid to a track rather than the track to a grid.

percussion, vocals, synths, instruments and effects. When you start drilling down into specifics, you discover that there are hundreds of thousands of samples in here that are (hopefully) going to make your track sound amazing. There are things you can do to tweak the key and time stretching.

You can ‘like’ samples to build up a bunch of favourites, and you can see how this is going to pair up really well with the Clip Launcher that we’ll come to in a minute.

Then, all you need to do is drag the sample into your project. And that’s when we discover the limitations.

You can’t use the majority of the content unless you have a subscription with Splice.

If you attempt to drag a premium sample into your project, you get a pop-up asking you to subscribe, and you don’t know if something is premium until you try to drag it. PreSonus tell us that there’s loads of free content; it’s just that there’s no way to filter it to just the free content when using the Search With Sound feature. If you’re doing regular searching or browsing, you can filter the results by clicking an ‘Included’ tag to remove any premium content. But, for some reason, that filter isn’t available through Search With Sound. In order to try this out, I’ve had to create a Splice account and start the seven-day trial. After that, it’s $12.99 a month. Search With Sound is a fantastically useful feature, but without a subscription, it becomes almost impossible to find

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a sample you can use. Splice then becomes another folder of loops, and that’s not anywhere near as impressive.

Integrated Launcher Launching loops, clips and scenes is a major part of music production to which around half of the most popular DAWs sign up. Studio One has decided to lean into loop-based production with The Launcher. It’s an ideal companion to the Splice integration but also gives you a brilliant way to reinterpret your music without having to export a DAW project to Bitwig. This feels like a good development that draws solid nods of approval rather than being a stunningly innovative move.

The Launcher can sit alongside your arrangement, occupying the same space as a Scratch Pad, or you can lose the Arranger window and focus exclusively on the Launcher. It’s more in the style of Logic Pro than Ableton Live, where your tracks remain horizontal and your scenes are stacked vertically. The interplay is really smooth.

You can pull across individual clips and drop them on to cells; you can pull in a whole chorus from the Arranger Track and it will rack it all up for you. If the clip is playing, the track in the arrangement is muted, and vice versa, so you only get one or other.

However, you can have a mix of linear tracks and Search With Sound is great for finding samples, but you need a Splice subscription to make the most of it.

Launcher clip tracks all playing together, and it’s easy to swap them on the fly.

You can record directly into a cell, although it’s not particularly sophisticated.

For instance, you can’t set the size of a clip; it just keeps expanding until you click Stop, which makes any jamming-style loop layering a bit cumbersome. There is support for the PreSonus Atom pad controller for clip and scene launching, but I couldn’t find a way of getting that to work with something like the Novation Launchkey. However, I did find key command options for starting and stopping cells, which effectively meant you could set a cell recording and then set it to play with a key press. I also couldn’t find a way of recording automation on top of what I’d recorded. I can capture it at the time of recording or add it in the modulation lanes of the piano roll, but I can’t overdub it with a knob twiddle. However, if you are dragging an event in from the timeline, it will bring any existing automation with it.

One nice feature is the scene playlist.

It reflects some of the Arranger Track functionality, where you can make a list of scenes and then play them back in any order you choose. The difference here is that the scene order doesn’t rearrange the arrangement, and you can specify how many times you want each scene to loop.

You can build whole songs this way.

If the Launcher is not really your thing, then there’s also a new looping feature in the regular timeline. In the right-click context menu for any event, you can tick the Loop box to turn it into a draggable, repeatable loop. This is how Patterns have been working, where they repeat down the timeline rather than having to duplicate or copy/paste them. It now works with any type of event.

New Instruments As I’ve said many times in every Studio One review since version 3, there are no new instruments. What about Lead Architect and Deep Flight One, I hear you ask? Yes, Lead Architect was introduced as an extra

Launcher introduces a non-linear clip-based approach to audio.

optional instrument with version 6.6 and Deep Flight One came along a bit under the radar, but both are essentially clever reskins of Presence XT. They have some great sounds, and I applaud the fact that they are now included for free, but the lack of development in the virtual instrument department is the one area where Studio One drags behind other DAWs. However, a couple of new instrument-related things in version 7 are worth mentioning. Firstly, Presence XT now comes with the editor built in for free, so you can build your own instruments, and that’s pretty great. Secondly, while functionally unchanged, Impact can now nest itself into the piano roll rather than being a floating instrument window. It certainly makes working with samples and building kits a lot easier. Thirdly, Studio One now supports the open‑source CLAP plug-in format. And fourthly, there’s a CV Instrument.

I’ve not seen much mention of this in the marketing material or promotional videos, but sitting under the PreSonus Instrument tab is an unassuming little plug-in called CV Instrument. The idea is that the CV Instrument can generate control voltages in response to MIDI notes, giving you the

ability to sequence your external analogue or modular gear. Currently, it’s quite basic, with a single CV pitch out and a CV gate output. That’s all you need for sequencing or drum triggering, but it would even more exciting to see some modulation capability for using looped automation or some such.

Unfortunately, I couldn’t get it to calibrate properly and so I never managed to sequence anything very tunefully, but the potential is there, and it’s very likely that I’m doing it wrong or the audio interface I’m using isn’t DC-coupled.

It can be a bit complicated to configure, so there’s plenty of room for messing it up and the documentation is very factual rather than instructional. I will look into this a lot deeper in a Studio One workshop because, for modular musicians, this is a really exciting development.

Odds & Ends

Version 6 brought in some major interface improvements, but with version 7 PreSonus appear to be largely happy with how it is. There’s some roundness to the edges of selection boxes, and they’ve been given a bit more density so they are easier to see in a large project.

There’s a new Global Transpose option, which is one of those features you sort of assumed was already in there, but now everything that has a defined key can be transposed together — that’s great.

User scales have been introduced to the piano roll, where you can define exactly what notes you want to use or display. It’s like The new CV Instrument let’s you generate control voltages PreSonus have expanded some and gates from a Studio One MIDI track. Useful!

of the cool things about drum pattern editing into the piano roll, and that’s a really interesting approach. It brings individual step counts and resolutions to notes so that your previously linear melody becomes a fabulous jumble of polyrhythmic note patterns. There’s also a rather neat Auto Zoom button that focuses in on the active notes.

Conclusion It feels like there are some important new features in Studio One Pro 7, but that can largely depend on what sort of user you are.

The stem separation is decent and quite good fun for everyone, but it’s one of those tools that’s looking for a way to be helpful even if it doesn’t know what that is yet.

The Splice integration is amazing if you’re a subscriber, but unless that’s your thing, it has limited use. The Launcher, on the other hand, is a great development that every sort of user can find a reason to dip into. I also really appreciate the pricing change, the versatility of the payment options and the extra bits and pieces they’ve folded in as standard. The new release schedule should keep us on our toes and gives me hope that fledgling features like the Launcher and CV Instrument will see steady development and growth going forward. So, it’s not bursting with new things; there’s nothing new for the Song Mastering or Show Page, as far as I know, but it’s building on a solid foundation and a good understanding of how to appeal to a very wide range of users. $ $199.99.

Subscription pricing also available.

W www.presonus.com

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Boredbrain Xcelon Eurorack Module

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f someone asked you to design an uncompromising analogue Eurorack mixer, it would probably look like the Xcelon. It would have mono or stereo channels, decent faders, good metering, adjustable gain, proper stereo effect sends and returns, solid clickless mute buttons, switchable outputs and CV control over everything. You’d want proper-sized knobs, useful LEDs and for it all to fit into a compact space. The Xcelon is almost all these things and more and may even tempt those using external consoles to consider pulling the mixing into the rack. All it asks is that you embrace the density of patch cables that inevitably overwhelms it from all sides.

In 48HP, Boredbrain have squeezed in only six channels, but over a third of the space is used up by 50 patch points, over half of which are CV inputs. Each channel is nicely spaced, with half the height given over to the 45mm fader alongside the LED meters, mute button, destination switch and XFADE button, which we’ll come to in a moment. You’ve then got two large knobs for the effects sends and smaller knobs for gain and balance.

All the controls fit well and feel good; there’s just enough room for it not to feel cluttered or cramped, which is one benefit of moving all the inputs to the side.

For connections, on the right side, you have two assignable mix outputs and two stereo effects sends and returns, which are also assignable and have knobs to

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control the return level. But it’s the input section that’s a lot more interesting. Each channel has a collection of six inputs. There’s a stereo input, which can also be a mono input or dual mono inputs, and this is where the XFADE comes in. With XFADE engaged, the balance knob becomes a crossfade between two mono sources going into that channel. This is a very neat way of increasing your channel count for things that don’t need panning. The remaining four patch points give you CV control over the volume and balance, as well as two effects send amounts.

This opens up several interesting modulation possibilities and also gives us the possibility of using channels as VCAs. While pushing all the patch points to the sides does make the Xcelon a bit bloated, the benefits of keeping the interface uncluttered should not be underestimated. The gain control is really well implemented, with lots of gain for matching wayward modules all the way to line level, plus a useful peak LED. The faders are particularly fingerable, and I like the way the glow increases as you push it up.

This also comes into play when modulated with CV. There’s an A/B output switch on the effect returns, meaning that you can quickly cut effects out of the signal chain, and I appreciate the larger effects send knobs that lean into effects as being part of a playable interface.

And then there’s the CV control that just wants to make you pump in a load of movement.

These things along with the

space and clarity make the Xcelon a thoroughly enjoyable and comprehensive mixing experience that doesn’t seem to suffer from compromises made on many other Eurorack mixers.

However, the Xcelon is merely the central hub of a whole system that Boredbrain are developing around it. On the back, alongside some jumpers that change the behaviour of the meters, effects sends and volume CV, are two expansion slots. You can add an XP3 three-channel expansion module to each. These add three extra channels, just like the original six, that route to the same master outputs and effects.

They have their own bank of CV control and can effectively turn it into a nine‑ or 12-channel mixer.

Sadly, they don’t fit perfectly, as whatever side of the Xcelon you place it there will always be a patchbay in the way.

There are also expansions for direct outputs, letting you route channels for multitrack recording.

The TRS module gives you a dozen balanced outputs or six stereo outputs on quarter-inch jacks. The DB25 module has three eight-channel D-Sub connectors for 12 stereo or 24 balanced audio outputs. Lastly, there’s the I/O module that has 30 channels of input or output that can be used to adapt to the TRS or DB25 modules or provide Eurorack outputs for the Xcelon.

What’s missing, perhaps, is a master section and EQ, but Boredbrain have already thought of that with their Monitr and EQx5 modules. The Monitr provides two stereo channel

output controls that pairs perfectly with the Xcelon, and you also get a headphone output with channel cueing.

The EQx5 is a five-band CV‑controllable equaliser that would need to be patched in before the Xcelon as there are no insert points. Six channels of EQx5 would get a bit unwieldy in your rack, and I wonder whether a simplified version with rear connections would make for a great future expansion. All in all, if you add the two expansions, the Monitr output and the DB25 to cover all the direct outputs, you would be eating up 110HP and wouldn’t get much change from £2400, but there’s nothing quite

as comprehensive, versatile or as extraordinary as this system in Eurorack right now.

There is a lot going on in terms of cabling and connectivity, which can threaten to drown out the sublime mixing experience. However, provided you keep your spaghetti under control and can afford the investment in money and HP, the Xcelon has the space to provide a solid interface for pro-level recording and a playground for live performance while feasting on the modular mayhem of CV control. Robin Vincent

ADDAC System ADDAC310

and Roland — have historically gone to great lengths to create controllers befitting of even the best wind instrumentalists.

While the ADDAC310 indeed seeks to bring that style of expression into Eurorack, it behaves in no way like a conventional wind instrument, primarily because its mouthpiece and wind pipe has no real airflow, in that there’s nowhere for any air to escape. This, say ADDAC, is to allow even the subtlest of changes in air pressure to have an impact on the module’s output and thus reducing the need to take many breaths. I found that it made blowing rather hard work, though with practice I did begin to get used to it, and was soon patching it to filters to create swelling chords, triggering drums with beatbox-style bursts of air and modulating an array of different parameters in both positive and negative directions. I could even ‘blow’ reverb into a signal, which was a thrill. Prepare for a learning curve, but if wind is your thing, the ADDAC310 will be too. William Stokes

Eurorack Module

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f it wasn’t clear already, ADDAC System have made it their business with the ADDAC300 series to create as many unconventionally expressive methods of interfacing with your modular system as possible. The latest is the ADDAC310 Pressure To CV, a similarly conceived module that uses your breath as its control source.

This it does by way of an oesophageal-looking melodica windpipe, which connects to the module panel via a curious looking air filter. Two identical but independent channels of CV are available; these can therefore be used either to send CV to completely different destinations or to variously affect different facets of the same destination. Fair to say one couldn’t ask for more functionality than ADDAC have included here: in addition to the linear wind-to-CV function, each channel offers a gate output with adjustable threshold, a variable response curve (that is, between exponential and logarithmic) and even an envelope-style slew limiter which can slow the attack of your wind-generated CV or add decay to it after you’ve stopped blowing.

There’s an inverted CV output for creating counter-functions and also a gate-controllable Hold button, which in effect turns the ADDAC310 into a sample & hold module — only it’s your breath as the voltage source.

Of course, many developers — not least the heavy-hitting likes of Yamaha

$

$899

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www.boredbrainmusic.com

$

$399

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www.addacsystem.com

DPW Design P1 Pedal I/O Eurorack Module

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t some point, all of us spill out of our racks into a motley assortment of pedals and effects boxes. I’m not sure what triggers it, but maybe you found yourself deep into a patch wishing you had a flanger module, and it occurred to you that you had an Electro-Harmonix Deluxe Electric Mistress sitting over in your guitar case. Before you know it, you’ve stolen some headphone adaptors, messily wired it into your patch, and are left fighting with distorted and deliciously wobbling audio, wondering why you never did this before and how you can do it better.

Into the chaos of mismatched levels and dodgy adaptors steps the P1 from DPW Designs. The P1 is a beautifully designed, two-channel pedal interface that sets the level and gives you the right connections to and from your pedals.

There’s no setup and no tweaking. The surface is devoid of knobs or controls, so there’s nothing you can do to mess it up.

You might pause briefly to give your brain time to comprehend the meaning of the weird yellow hieroglyphs, but other than that, you’re good to go.

The P1 is designed to be sympathetic to guitar tone and do all it can to make the transition from Eurorack levels to instrument levels as smooth as possible.

The P1 is AC‑coupled, keeping out any flat voltages or offsets. It has more than 100kHz bandwidth in both directions.

The inputs are Hi-Z and include soft, controlled, symmetrical clipping to reign in wayward signals with some very friendly colour.

Compared to guitar level, Eurorack is around 18dB hotter and so the job of the P1 is to sort out that disparity. When interfacing to pedals, a 5V Eurorack signal (+14dBV) will give you -4dBV at the pedal side. When going the other way, guitars don’t tend to be quite as well-behaved and often have edges and transients that push beyond the expected -4dBV. The soft‑limiting starts www.soundonsound.com / December 2024

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to kick in at this point and reaches full compression and symmetrical brickwall limiting at +6dBV to a maximum of ±10V on the Eurorack side.

In use, the P1 sets up a perfectly good effects loop or two through whatever pedals you want to play with.

The lack of fuss is quite refreshing, and all you have to contend with is having four chunky guitar leads poking

out of your rack. I suggest putting it at the edge of a case and using some right-angled ones. It’s a brilliantly useful way to bring some alternative vibes into your modular. Going the other way lets you use your modular as a massive guitar effects processor. This is where you start to appreciate the soft‑limiting and tone of the P1. My twin‑humbucker semi-acoustic really appreciates the

fuzzy edges, my acoustic is clean, and my Telecaster sounds like a guitar should sound before it hits the chaos of CV-modulated effects.

The P1 is a simple, thoughtful module that’ll bridge the gap between your modular and guitar worlds, making both all the richer for it. Robin Vincent $

$129

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www.dpw.se

Modular Profile: Sebastian Hastrup

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ponymous founder of Swedish company Sebsongs, Malmö’s Sebastian Hastrup has burst onto the Eurorack scene in recent years with a slew of gorgeously minimal, modestly sized and functionally named modules, from the wildly creative 12-bit digital Sampler to the brilliantly designed Polyseq flexible step sequencer. His latest module, the Bread & Butter, takes the hallowed sounds of General MIDI and implements them into your modular system with equal parts irony and sincerity.

On his entry into modular I’ve had an interest in music and electronics for as long as I can remember — my first serious DIY projects were stripboard guitar pedals based on any designs and schematics I could find online.

When I eventually discovered modular synths it seemed obvious that it would be my thing as well. I started building DIY YuSynth 5U modules back in 2013. I had circuit boards made and sourced all components manually.

I instantly fell in love with the process! The idea for my module Polyseq was actually born already back in 2014 and that was also my first venture into the Eurorack format, as I realised that the Polyseq could be a commercially viable module and Eurorack

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was the most popular format already back then.

On his go-to modules (aside from his own!)

One of my early module purchases was Mutable Instruments’ Plaits and I still use that module in many of my patches — most of the time for kick drums, but it’s so versatile and comes in handy in a lot of different scenarios.

Tom Whitwell’s [Music Thing Modular] Radio Music is also a frequent part of my patches.

It’s such a simple and brilliant design. I also love the Doboz modules and I’ve built both the TSNM II and a couple of his VCO340s, the latter being one of the most complete and compact analogue oscillators that I’ve used.

On the Bread & Butter I had a lot of fun developing the Bread & Butter! I wanted to release a synth voice as my next module after the Sampler and went through many different ideas and designs before settling on the General MIDI polyphonic synth voice concept. The module is based around a cool little audio encoding chip that happens to have a really nice‑sounding GM synth engine on board.

My main goals for the module were to keep it simple, small and fun to use, and it seems to have resonated well with people so far. I really like how easy it is to create a lush and

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wide polyphonic soundscape and it’s become an integral part of my own patches.

On making DIY kits After a fire back in 2019 I lost my 5U system and when the pandemic hit I wanted to start building a modular system again. So I started designing Eurorack modules as a recreational and therapeutic thing, just for myself. After having designed a bunch of modules and seeing them work well together in my own system, I started thinking about somehow sharing them with the world. In late 2021 I reached out to Thonk and they were very enthusiastic about my designs, so we started a collaboration.

To this day I design and prototype the modules, and the fine folks at Thonk assemble and distribute the DIY kits. As I’m a DIY-er myself, putting my modules on the market as DIY kits made a lot of sense to me. I’ve really enjoyed being part of the DIY scene myself for the last 10 years, and contributing to the DIY culture is something I feel strongly about.

On the culture of modular In my experience, the modular scene has been very embracing and open to people like myself, especially the synth DIY scene. I joined the Electro-Music forum early on, posted updates on my 5U system and got a lot of encouragement and help from that community. Since joining the modular scene as one of the Eurorack module brands and having conversations with customers from many corners of the world, it has really hit me how diverse and joyful the modular culture can be. I like to make an effort contributing to the DIY synth culture by offering good support and by encouraging newcomers to learn how to build DIY modules themselves. More broadly, modular culture seems to continue to grow and I see modular systems in many different scenarios: from contemporary music composition and film scoring to experimental art installations and much more.

I’m definitely proud to be a part of this culture and I look forward to what the future might bring. William Stokes

GEORGE MASSENBURG SUPERIOR DRUMMER 3

Audio industry pioneer, inventor and the man behind the core library sounds of Superior Drummer 3.

AL SCHMITT

FRANK FILIPETTI

ELLIOT SCHEINER

STORIES SDX

STATE OF THE ART SDX

From pop, rock and metal to original cast Broadway shows and major studio Hollywood films – Frank has done it all!

One of the most awarded and, hands down, absolute best in the history of the business.

BOB ROCK

EDDIE KRAMER

DECADES SDX

THE ROCK FOUNDRY SDX

LEGACY OF ROCK SDX

The embodiment of craftsmanship.

We are proud that a part of his legacy lives on through this SDX.

The man who’s shaped the sound of some of the best-selling hard rock and metal albums of all time.

With Led Zeppelin, Jimi Hendrix and many more, Eddie helped define the sound of an era.

HUGH PADGHAM

JAMES FARBER

HITMAKER SDX

THE JAZZ SESSIONS SDX

Still as relevant today as when his inventive techniques defined the sound of the 1980s.

A true master of the craft and an icon in the world of jazz recording.

LEGENDARY DRUM SOUNDS.

For Superior Drummer 3 and our universe of SDX expansions, we’ve had the honor of working with some of the most legendary engineers of our time. Here are just some of them. Meet the rest and listen to their sounds on our site.

WWW.TOONTRACK.COM

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Boss IR-2 Amp & Cab Emulation Pedal Roland and Boss have a long history of developing guitar amp modelling, but somehow they’ve managed to cram their latest technology into a Boss compact pedal format, making the new IR-2 one of the most compact amp modeller pedals out there.

Even for a digital pedal, that’s no mean feat, as the case houses six quarter-inch jacks, a headphone mini‑jack output, a USB-C port and a power inlet, as well as three dual-concentric knobs and one rotary switch.

There are 11 amp models to play with, and each one loads with an appropriate Celestion Digital IR, but you can also load your own IRs over USB, using the free IR-2 Loader app (for Mac and Windows).

Looking at those connections first, on the right there’s the input, a socket for a channel-select footswitch (if connected, the integral footswitch just works as a bypass) and a TRS return for an internal effects loop. The loop supports stereo operation when needed, sending stereo effects to the IR-2’s audio output jacks, which are on the left along with the FX Send jack.

On the rear, the USB-C port facilitates connection to a computer — the pedal can operate as an audio interface as well as load user IRs using the app —and there’s also the headphone jack. As with other Boss compact pedals, there’s a battery compartment under the pedal, but using a power supply is recommended, as the digital circuitry can be a touch power hungry; this pedal takes 160mA. We’re informed that the DSP utilises 32-bit floating-point processing at a 96kHz sample rate.

The centre sections of the three dual-concentric knobs control Ambience, Level and Gain, with the right‑most knob being the model selector. The outside rings of the knobs controls Bass, Middle and Treble. Ranging from clean to super hot, the 11 amp types are: Clean, Twin, Tweed, Diamond, Crunch, Brit, Hi-Gain, SLDN, Brown, Modded and Rfier. The Ambience knob adds some natural roominess to the sound.

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The in‑built footswitch alternates between two channels of amp settings or, when held down, engages bypass. There’s no ‘save’ routine: you just set the knobs as you like, press the footswitch (which toggles the LED between red and green), then dial in new settings for your second ‘channel’, and then wait a couple of seconds before changing channels to allow time for the parameters to be updated. The amp and cabinet sections can be turned on or off independently, so you can bypass the cab when plugging into a regular guitar amp.

“While there’s virtually no learning curve for using the IR-2, a downloadable manual details more advanced features.”

While there’s virtually no learning curve for using the IR-2, a downloadable manual details more advanced features, many of which are accessed through specific power-up routines. These include bypassing the amp or cab sections, and it’s also possible to change the ambience from Room to Hall or Plate, and to configure the return jack for mono or stereo use.

The USB-C port will work without drivers in Generic mode but defaults to Vendor, which needs the downloadable

Boss drivers. The Generic driver setting allows it to work with iOS devices. There are settings to match the sound to different types of amp (some Boss and Roland models, plus a few generic ones). If you connect to the IR-2 Loader app, there’s an Editor tab that will let you switch the amp and cab sections on or off, select the ambience type, select the output mode and switch the return to mono or stereo. You can also switch from Generic to Vendor driver modes, activate a loopback function and adjust the effects out, mix in and loopback levels.

On the IR page, you get to see what drivers and mic IRs are used for each amp type, with the option to locate a third-party IR and load that into the amplifier type slot instead. There’s also a Menu page that allows for data backup and accessing the manual, if you haven’t downloaded it already.

Sonically, this pedal sounds much like its ‘grown-up’ Boss counterparts, though as there are no onboard effects other than the Ambience options you may wish to combine it with other pedals or plug-ins; the effects send and returns are a useful addition in that respect. The sounds respond well to playing dynamics and those tricky on-the-edge breakup sounds come over convincingly. Clean sounds have a comfortable playing feel and don’t sound at all sterile, while more assertive rock sounds are great fun and cut through nicely without getting fizzy. DI’ed into my DAW and adding just a hint of reverb produced plenty of amp timbres I’d be happy to record with when miking an amp is impractical.

Whether you need a pedal that can conjure up 11 amp types only you can say, but the amp types cover just about any genre and you can have a different amp type with different control settings loaded into each channel too. For the live musician, an IR-2 on a pedalboard could provide a good direct-to-PA solution, be used as an overdrive, or just live in your gig bag as a backup option. Given its impressive sound quality, ease of use, modest cost and five-year warranty, the IR-2 looks destined to become very popular. Paul White $ $199.99 W www.boss.info

ModeAudio Airspace Reverb Plug-in It might get filed under the general category of reverb, but ModeAudio’s Airspace could more accurately be described as a sound‑design tool. This plug-in, which supports all the common formats for Mac and Windows hosts, comprises two separate, identical IR engines with a delay section placed between them. The first IR section is designated Colour, and as that implies it’s normally used to add colour to the sound rather than obvious reverb. The IRs designed for this coloration are generally short recordings of things such as oven door slams, drum machines, synths, string scrapes and so on. There are also guitar amps, found sounds and a huge range of other weird and wonderful things. From here, the signal feeds into the delay section and thence into the Space IR engine. That one would normally be loaded with a more spatial type of IR captured from a real space or a physical reverb device. That’s the intended workflow, but actually both IR

Neural DSP Parallax X Bass Processing Plug-in Evolved from Neural DSP’s earlier Parallax, Parallax X is a multiband processor designed to enhance bass sounds, using separate levels of high‑ and mid-band distortion combined with low-band compression and the ability to transpose pitch. The plug-in supports all the popular formats for Mac and Windows, and is authorised through an iLok account.

At the heart of the plug-in is a three-band crossover, and the crossover frequencies can be adjusted by dragging points in the centre section of the GUI, which depicts the crossover curves. The levels of the three sections may be adjusted by using the rotary control below or again by dragging the points on the crossover display.

sections are identical and have access to the full range of IRs — so you could easily break a few rules in the name of creativity by, for example, chaining two colour IRs or two long reverbs.

The GUI is well organised, with the Colour section to the left, the Delay section in the centre with its modulation panel below, and the Space section to the right. Each section has its own level control and bypass button. The waveform of the IRs loaded into the two sections is displayed on screen, and there’s a manageable number of controls to adjust the IR duration, wet/dry mix, pre-delay and reverb envelope. Clicking EQ below the waveform display replaces it with a four-band EQ, which can be activated using the EQ On button. The delay can be set differently in the left and right channels, with many note subdivisions to choose from if you want to lock it to your DAW tempo. The two channels can be linked if needed and there’s also a crossfeed option. The feedback controls in the delay section are linked to the adjustable high- and low-pass filters.

The modulation section offers multiple LFO waveforms including random, with separate sections for Time Mod and Pan Mod. In the centre are controls for Pitch Shift (±2 octaves) and Shift Mix. An overall wet/dry slider is available at the top right of the GUI. In fact, really the only missed trick I can think of is that there’s no option here to reverse the IRs. What makes Airspace such a powerful sound-design tool is the sheer number of Colour IRs, which in many cases render the original sound source unidentifiable.

For example, a concert grand piano can be transformed into anything from a granular synth pad to a haunting wind chime. If creating ear candy is on your wish list, there are plenty of tools here to achieve your ends. The spatial reverbs also sound good and vary from springs, plates and digital reverb units to real spaces of all sizes. As expected, there are plenty of presets, and these are thoughtfully divided into Ambient, Blur, Cosmos, Reflect, Transform and Warp categories.

Exploring these gives a good idea of the creative potential of this plug-in. The range of effects is extraordinary but most retain the musical integrity of the original sound, so that you end up with a playable instrument rather than an abstract noise.

I’m definitely going to have a lot of fun exploring this plug-in further — there are so many permutations of IRs and control options that there will always be something new to discover, and unique results every time. Paul White

“The range of effects is extraordinary but most retain the musical integrity of the original, so that you end up with a playable instrument rather than an abstract noise.”

$ $79 W https://modeaudio.com

At the top of the screen are controls for a noise gate and a pitch transposer with a ±12 semitone range. This works extremely smoothly on bass sounds, so it’s just the thing if, for example, you just need to get to a bottom D when recording a bass guitar part. Mono or stereo input mode can be selected, and there’s a generous selection of presets (many by specific artists) that show off the scope of the processing. For live performance, there’s also a standalone mode.

At the bottom of the screen are three main control sections: Low Compression, Mid Distortion and High Distortion. There’s also a six-band graphic equaliser. There’s no master wet/dry control here, but if you need that facility some DAWs offer it per insert slot, and it’s easy enough to use Parallax X as a send rather than as an insert effect.

Clicking on the concentric circles logo at the top of the screen switches the display to cabinet emulation with a choice of microphone types, mic position and mic distance. There’s a choice of three dynamic mics, one capacitor mic or two ribbon models and the position can be adjusted www.soundonsound.com / December 2024

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using a rotary control or by dragging in the display. When used in stereo mode there are separate cab emulations for the two channels (affecting only the mid and high bands) with independent level, pan and mic position controls. There’s also the option to load custom IRs. MIDI parameter control is catered for and there’s a built-in tuner that can be set to mute the audio while tuning.

“You might think it’s aimed mainly at metal-style bass sounds, and it does those exceptionally well, but it has a more subtle side too.”

If you’ve seen any online demos of Parallax X, you might think it’s aimed mainly at metal-style bass sounds, and it does those exceptionally well, but it has a more subtle side too. For example, compressing the lows while adding just a little dirt to the mids can help an electric bass pop out of a mix without sounding obviously dirty, and there’s lot of tonal territory covered simply by swapping out mic types and positions. Compressing just the low end adds fatness without having too much effect on the mids and highs.

I also found Parallax X to be a useful tool for sculpting bass synth sounds, whether to add harmonic complexity or maybe just some low-end weight, and there are also applications in spicing up drum parts. There are many competing ‘add dirt’ products available today, but for treating bass sounds specifically, Parallax X offers a great balance between sound, features and ease of use — I’d definitely recommend giving the free 14-day demo a try and seeing if it works for you. Paul White $ €99 (about $109) W https://neuraldsp.com

Parallax X’s cabinet emulation page.

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Waves Curves Equator Resonance Suppressor Plug-in Curves Equator is a ‘smart’ dynamic equaliser plug-in that’s similar in concept to Oeksound’s well-respected Soothe. As with that tool, Curves Equator tracks and attenuates resonances dynamically, to achieve a more balanced sound that sits comfortably in a mix, and it supports side-chain operation so that the plug-in can work as a kind of spectral ducker, carving out space by pulling back only those frequencies that conflict with the side-chain signal.

There are a few operational differences, though, and each offers the odd tweak that the other doesn’t. Notably, Equator’s Learn feature is available on both the input and side-chain curves, each doing a specific job.

Basic processing is straightforward.

A threshold curve determines the level above which processing takes place, and adjustments determine how sounds above the threshold are treated. The default is mode is Adaptive mode, which uses a default threshold curve (or one of the factory preset curves), but you can adjust these to suit the audio being processed.

For example, if you feel some parts of the spectrum need processing more than others, there are tools for that. Alternatively, the Learn feature can analyse a section of audio (ideally a good‑sounding section that’s ‘typical’ of the track), auto-generate a threshold curve, and aim to keep the rest of the track sounding consistent. If using the side-chain option, the side-chain signal can also be learned. If your content needs overall spectral balancing or adjustments, it’s worth running a signal through Curves Equator with the default settings, adjusting the main knob along with the various Shape and Tilt parameters — this is generally pretty effective in identifying and pulling back resonances across the whole spectrum.

The main view shows the threshold curve, the input spectrum and, in blue, the

“Curves Equator does an effective, surprisingly transparent job of pulling down annoying resonances.”

spectrum of what’s being tamed. You can often get perfectly fine results with the basic controls, but I’ve tended to stay in Advanced view, which doesn’t give you too many controls to contend with. Three variable crossover points split the processing into four bands. Knobs adjust Sensitivity, Shape, Tilt and Mix, and there are buttons for Limiter and Dynamic Auto Gain Make‑up.

The larger main knob is used as a fader to control the threshold level, and a Tooltips feature explains the various other controls as you mouse over them. At the bottom of the GUI a row of smaller knobs adjusts the Attack, Release and Precision of the processing, as well as a frequency Tilt (this tilts the threshold curve to favour the highs or lows), and Smoothing and Balance parameters, which can be set to L-R or Mid and Sides.

The levels in each band can be adjusted — so you have a basic EQ — and the crossovers can be moved to define frequency ranges to which you want to apply more/less processing, with a range of 0 (no processing) to 200 percent.

The processing can be more precisely targeted by adding and adjusting up to eight nodes on the curve too: if there’s any part of the spectrum you’d prefer were left alone, you just increase the threshold level in that region, or for more assertive processing pull it down.

On individual tracks, Curves Equator does an effective, surprisingly transparent job of pulling down annoying resonances.

I did my first tests on an acoustic guitar recording that I felt was a bit bottom heavy and also boomy in parts, and rather than just pull down all the lows as would happen with a multiband compressor, Curves Equator generated a dynamic shape that responded to the resonance contours of the input,

pulling down just the offending frequencies whenever they became excessive. It worked well to tame excessive booming and boxy body resonances. It could tame vocal resonances and harshness too. But really, any source that shows signs of unwanted resonances can be improved upon, including drums. The side-chain support was a pleasant bonus, as the spectral ducking feature can be so useful.

Most of the time, its judgement is pretty good, but should you decide that you like particular resonances (it can sometimes get overenthusiastic on snare hits, for example), it’s easy to tweak the threshold

JMC Cyberdrive Distortion & Effects Plug-in JMC’s Cyberdrive, which supports VST2/3, AAX and AU hosts on Mac and Windows machines, is one of those plug-ins that’s designed to break your audio in interesting and creative ways.

At its heart are three distortion engines, which are connected in series, and each can be set to one of 64 different distortion types. The distortion types encompass all manner of analogue emulations, plus some seriously disturbing digital tools that go way beyond simple bit reduction and wave folding. They are arranged as Classic, Modern, Pixel, Amp, Pedal, Doom and Freak, the last of these being where you can access more experimental distortion techniques such as ring mod, feedback, FM and spectral processing. Adjustable HP and LP filtering (6-96 dB/octave) is provided for each distortion section. There’s also a feedback section with tempo sync and a Follow knob that controls the depth of the level follower modulation. Other modules can be placed inside the feedback loop, though, so this can go much further than simple repeat delays.

Six processors at the bottom of the GUI offer Dynamics, Tone 1, Tone 2, Motion, Profile and Space options, each of which has its own sub‑menu of variants.

For example, Dynamics offers a choice of Modern Compressor, Classic Compressor, Maximiser, Gate, Transient, Hyper or Limiter.

Motion serves up all the classic Flanger, Chorus, Vibrato, Tremolo and Phaser effects but also adds in some lo-fi wobble.

Profile provides a range of speaker cab

“Mayhem has never been so intuitive!”

nodes to reduce processing in those areas.

I found the side-chain mode particularly valuable: with the right routing set up in your DAW, if you put Curves Equator on a bus, this feature makes it possible to unmask important elements in the mix.

The Learn mode generally worked well, whether on the main or side-chain signal, even if the results sometimes required fine-tuning — I found it easy to make precise adjustments by adding and adjusting nodes. I should point out that it can often be tempting to over-process resonances, only to find that your track no longer pops out of the mix quite so strongly;

emulations plus a selection of metallic-sounding IRs, while Space is devoted to reverbs and delays. Those two tone sections offer a range of useful EQ types including notch filters.

One particularly interesting filter mode is Multi, which is essentially a crossover — everything above and below the set frequencies passes through unprocessed.

As if this weren’t enough to play with, there’s also a section at the very bottom edge of the window that lets you drag the various processing modules into a different order, and that includes the ability to place individual distortion engines between the other tone-processing blocks. Mayhem has never been so intuitive! You also get a Dice button for randomising your patches, with a menu allowing you to decide which elements get randomised and which stay the same, and as usual there are presets to get you started. AGC (Automatic Gain Conrol) is available for each distortion section, along with a clip warning LED and an extra control that changes function depending on what type of distortion is being used. All sections have their own bypass buttons. Overall master buttons activate oversampling, master AGC, drive level adjustment, stereo width adjustment and the wet/dry mix balance.

as with any powerful tool, you should use Curves Equator with care, and evaluate its results in the context of a mix.

The overall concept may not be original, but in its execution Curves Equator is effective. It’s straightforward to use at a basic level, but more advanced controls are there when they’re needed. Use it judiciously, and you can achieve impressive results — and it’s priced to be affordable to the home studio operator too. Worth checking out. Paul White $ $79 (discounted to $39 when going to press)

W www.waves.com

The outcome? With Cyberdrive, you can damage your audio in superbly creative ways that run from a dusting of analogue-style saturation to full-on, snarling digital chaos. There are combinations that sound like a howler monkey has swallowed an electronic speak and spell game, and others that sound like something really bad happening in a scrap yard. There are phantom tones produced by the more radical bit-wrecking options — some are reminiscent of an FM synth fighting it out with a ring modulator — while other settings might put you in mind of an electrical storm on an alien world. Drum loops are obvious targets for this type of processing, as such material can be changed beyond recognition, but really you can use it to tear apart pretty much any source, from bass lines to vocals, pianos to orchestras.

If cruelty to audio is your thing, you have just got to get Cyberdrive — destruction rarely gets this creative! Paul White $ $87 W https://unitedplugins.com www.soundonsound.com / December 2024

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The Art & Science Of Mixing All mix engineers need technical skills — but what’s most important is our ability to bring out the emotion in the music.

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 is a balancing act.

We have to fulfil the artist’s musical vision, while meeting a technical standard. That sounds easier than it actually is, and the reason may be that we sometimes focus too much on the technical aspect.

In fact, I know a lot of engineers who approach mixing from a purely technical standpoint. When they start a mix, they open up the session, hit Play and immediately start adding processing — because that’s what mixing is all about, right? I’m not being judgemental here, nor do I want to tell anyone how they should

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mix. What matters is that the mix comes out great and the client is happy. But in my experience, more often than not it’s these ‘technical’ engineers who struggle to get a great mix in a reasonable amount of time. And that’s because they tend to skip the most crucial part of mixing: actually listening to the production and ‘feeling’ what really needs to be done.

While this may sound obvious, it cannot be overstated, because it’s so easy to forget: mixing is all about the music! Without the song, there’s no mix. And the hard truth is that we are rarely hired to change it all and apply wicked mixing skills just because we can. We are hired because the artist or

label trusts us to make the best possible decisions – which can mean anything from doing hundreds of bold moves to only touching four tracks.

There’s tons of great mixing tutorials around, often featuring the very people who worked on our favourite records.

This is great, but what that information often lacks is context.

Diamonds In The Rough Of course, there is a technical dimension to mixing, which often involves correcting and fixing technical issues that ideally should have already been taken care of during the production. But all this is just a small piece of the puzzle. The essential skill all great mixers have in common is their ability to combine the roles of music listener and audio engineer, by translating what they are hearing into an emotional response that fits the song.

This makes them artists in their own right.

If you don’t take my word for it, this is what Manny Marroquin, one of the most in-demand mixers for over two decades now, has to say about it: “I know the emotion I got when I turned faders and EQs. I can change the emotion of the song. That still blows my mind to this day. So, to say that I love what I do is an understatement because I can change emotions based on frequencies. It sounds

obvious, but it’s powerful when you utilise it the right way. And that’s what we’re here for. We’re here to play with people’s emotions and emphasise.”

This is why it’s so important to really listen to the music you’re about to mix.

Because if you don’t, how could you possibly know which emotions you want to emphasise and play with? You could start with a blank slate, but most likely there will be a rough mix the artist took home after the recording. This could be anything from just a flat balance bounce off the board or DAW session to a quick and dirty mastered listening copy. Whatever it is, make sure you have it in the session, level matched and in sync with the actual tracks (a plug-in like Metric AB comes in super handy for this).

Personally, I always request a rough mix, and highly encourage you to do so, too.

The artist will most likely have listened to this rough mix a hundred times. They left the studio full of endorphins and excited about their newest masterpiece.

This is how it sounds to them, and with a good tracking engineer and producer,

the balance on the rough will be far from arbitrary. To radically change it, you will have to have a good reason.

When it’s time to hit play and listen, I like turning up the volume, to really evaluate the physical and emotional response I get, but of course that’s up to you. The vocals in the rough may be too loud, but don’t let that distract you. Pay

attention to how they panned out the instruments and their relative balances.

Pay attention to what grabs you: a certain guitar part, a driving percussion rhythm, anything you think is making the song interesting and worth featuring. Pay extra attention to the transitions between different song parts, and whether the dynamic you want to feel is already there

The Benefits Of Attended Mixes Many mixers don’t like the artist to be present while they work. But although it can be very time-consuming and distracting, I have learned very valuable things from artists sitting in on a mixing session, and learning their language has definitely sharpened my skills. Because when an artist says “I think we need more power in this part,” this can mean very different things depending not only on the artist’s individual response to the music, but also on the genre and arrangement. Maybe more low end is needed to establish the felt sense of power — or maybe it’s more punch in the drums, or louder guitars.

Usually, though, people will equate power to overall loudness. This is why I like to work with

The SERIES SlatFusor

different layers of compression and parallel paths, because I have different options to gain more perceived loudness without having to just rely on riding the master fader. If the artist wants more “aggression”, this could mean more distortion, more compression or simply boosting an aggressive midrange frequency. What’s even more important is their body language. If they move their body to the music, that’s a good sign.

If they raise an eyebrow at a certain point, make sure you pay attention to this. Getting good at sensing what the artist feels or means in that moment and being able to make that happen has given me a lot confidence. And without confidence there’s rarely a great mix.

Modern Elegance Superior Acous�c Panels & Bass Traps

www.soundonsound.com / December 2024

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and just needs to be emphasised, or will need a lot of help. Pay special attention to your physical reaction, and whether the rough already makes you move a certain way or leaves you frozen.

And, of course, pay attention to the lyrics. The words may very well deserve a listening pass on their own. What is the song about? What emotional response do you have, and how does that translate to sound? Should the mix feel loose or stiff?

Should it put you at ease or on alert? Do you imagine a band playing in a particular space, such that the song needs to sound raw, real and intimate? Does it call for extra-wide and artificial with lots of effects because it feels larger than life?

Do the drums need to sound huge and open, or compact and characterful? What sound does the vocal need to convey the message: clear and breathy highs, or warm and crunchy mids? Bone-dry, with a haunting slap echo, or drenched in a lush reverb? All this helps to answer the key question: how do you, as the listener, want this to sound? Making this happen without losing the integrity of the rough mix is your plot for the mix.

Because without a plot, how can you possibly mix this at all?

Feel First Michael Brauer, who is known for stressing the importance of focusing on emotion despite his notoriously complex mixing setup, has coined a term that really stuck with me: MAI = match and improve.

He always makes sure to first match the rough mix as closely as possible, and only then starts improving it where he feels necessary. I feel that’s a very good strategy, because it really makes you listen and focus on the important stuff.

Also, making sure your initial balance matches the rough not only makes you familiar with all the elements in the session, it can give you valuable information about why you feel something needs to be improved or changed. Sometimes, maybe raising the acoustic guitars one dB negates the need for making the move you thought you’d need on the electrics.

Sometimes, a simple panning move can be all it takes, because you intentionally bring the mix out of balance at a certain point to create tension.

Learning The Language

As a foundation for communicating with artists, I find it helpful to establish a catalogue of emotional and musical terms. If the artist asks why you did a particular mixing move, you’ll probably want to answer using this sort of language rather than technical jargon. The artist won’t be interested in exactly what frequencies you cut, or the sophisticated side-chain compression technique you use. They want their music to be felt the way it’s supposed to be. And so should you, too.

Thinking about terms you need to have in your arsenal, I always find it’s good practice to think in opposed pairs, because this gives each side

The ability to recognise what to keep and what to change, and to figure out the best ways to do that, is what gets top mixers

a perspective. Take dark vs bright, for example: you need to know bright to understand what’s dark. Or take raw/dirty vs polished/pretty; without polished, there’s no perspective on what’s raw and vice versa. All of these terms have both a musical and an emotional component. For example, the term ‘urgency’ clearly describes an emotional response, but one that can be evoked by different musical or mixing moves. Feeling urgent can mean something is compressed and saturated severely, but it can also mean it has to drive rhythmically. Both goals can be achieved using compression, but it’s your job to know which settings will deliver which type of urgency.

in every aspect? Remember the plot you had! If it does, feel the warm hug of confidence. If not, there’s no need to feel ashamed or defensive.

This happens to everyone, and a good rough is surprisingly hard to beat. If something doesn’t work, accept that and go back to the drawing board. The artist deserves the best possible mix you can do. They don’t pay you for applying clever processing if it doesn’t help their music. They pay you for making the listener want to sing along and hit repeat. They don’t care what you have to do — or not do — to make that happen.

“The words may very well deserve a listening pass on their own. What is the song about?

What emotional response do you have, and how does that translate to sound?”

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hired again and again. Knowing what and, especially, what not to do is one of the quintessential skills to develop as a professional mixer.

Another one is to be brutally honest with yourself. When you switch back and forth between your mix and the rough after two hours of mixing, does your mix sound better to you

Decapitator The Secret Weapon of Top Mix Engineers

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“Decapitator is instavibe-in-a-box.”

– Fabrice “Fab” Dupont (André 3000, Shakira, Jennifer Lopez, Marc Ronson, Isaac Hayes)

Free 30-Day Trial at www.soundtoys.com

TECHNIQUE

Pro Tools

We get reacquainted with some of Pro Tools’ stock plug-ins.

JULIAN RODGERS

I

f you install all the plug‑ins which come with Pro Tools you’ll be greeted with a generous array of processing options — tools that are often overshadowed by the allure of third-party alternatives. Yet, hidden among these are some real gems that can be just as effective, whether you’re new to the DAW or a seasoned professional. In this article, we’ll explore some of these overlooked plug‑ins and share practical tips.

The exact set of plug-ins you have access to depends on which Pro Tools tier you’re using — whether it’s Intro, Artist, Studio, or Ultimate. Some premium Avid plug-ins require you to be ‘on plan’ with a subscription or support plan, but even without it, the stock plug-in library is substantial.

There are also some third‑party plug-ins that come as part of Avid’s Inner Circle scheme, which adds value over time for on‑plan customers. But the total available to a new user is significantly boosted by legacy plug-ins, some of which have been around a really long time.

Although they are ‘legacy’, Avid ensure that they’re kept up to date so that users can open older sessions without issues. You might be surprised at how well these older plug-ins still hold up. Also, nearly all Avid plug-ins are available in AAX DSP versions — an increasingly rare feature.

If you haven’t checked out the stock plug-ins as thoroughly as you might have, or if you’ve never even installed

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The venerable EQ III has a neat feature whereby you can temporarily turn any band into a band-pass filter, all of them, there’s probably a reason for allowing you to home in on your desired frequency. that. Maybe you have a subscription with a third‑party plug-in brand that already and then sweeping it. If you’re cutting and provides you with more plug-ins you can you want to double-check you’ve got the shake a stick at? Whatever the reason, let’s right frequency, Shift‑clicking the control get into some specific examples which point will invert the gain to a boost (or vice might encourage you to look again at these plug-ins. Switch on the preference versa). Get these under your fingers and to display plug-ins by manufacturer as you’ll really get the value of EQ III. well as by type, and have a browse under Moving onto compressors, the 1176 is ‘Avid’ to see what’s already installed. one of the best known, and is of course much imitated. BF76 holds its own EQ & Compression against premium offerings, in spite of the ancient UI with its less than smooth meter.

I have to mention the stock EQ III 7-Band.

Dialling in some moderate compression There are third‑party alternatives that offer on a drum room mic, I was able to get more features and flexibility, but as a utility a sound which, while not identical, was equaliser, EQ III is a solid choice. If you very close to a premium alternative. There don’t yet know about the band‑pass were differences, but in a blind test I’m not feature in EQ III, try holding Control+Shift sure I’d have chosen the premium plug-in on a Mac or Start+Shift on Windows and as better. Under extreme settings the BF76 grabbing a control point on the curve or was nastier (in a good way), and given that the frequency aggressive sounds are often the intention knob. This gives when breaking out an 1176, it might be your a momentary preferred choice. To test it out set up your band‑pass favourite 1176 emulation against BF76, set behaviour, up one in slot A-E and the other in slot F-J, allowing you match by ear and A/B between them using to quickly find Shift+2/3. the offending frequency Going Gritty without resorting to setting up I’m not alone in really rating the Lo-Fi a narrow boost plug-in. The secret as far as I’m concerned is to ignore all the controls apart from BF76 might be Distortion, and only use tiny amounts of old, but it still holds that. Dial in some Distortion on a DI bass its own against guitar. Use a setting under 1 and follow it modern third-party with a Trim plug-in to match the apparent 1176 emulations.

levels. You’ll hear a nice saturation effect that works really well on transient-rich material like drums if you take care not to go too far; it can really warm up a bass guitar, too. If I’m looking to put some subtle fur onto a sound rather than introducing a more audible ‘trash’, I’ll sometimes use the stock SansAmp PSA-1.

Pitch II A great way to thicken or widen sounds is to use a detune effect, much like one would when detuning oscillators against each other in an analogue synth. Rather than using modulated delays to achieve this effect, try a static pitch‑shift of a few cents in each channel of a stereo or a mono-to-stereo Pitch II plug-in.

Instantiate a mono‑to-stereo Pitch II on a mono guitar track and tune down by up to five cents on one side and up by a few cents on the other. Mix to taste using the Mix sliders for some thickening and width. The beating caused by the detuning will change its speed depending on the settings, and if you want to try creating a more complex effect, stack multiple Pitch II plug-ins, with the Mix controls adjusted to blend the effect.

Reverb & Delay Don’t write off the humble D-Verb. It’s been used on more big-selling records than most of us realise. However, you do need to be aware of something important about it: the way it modulates pitch.

Set up a D-Verb as an insert on a simple sustaining part played on a pitched

instrument — a piano is ideal.

Or, if you prefer, just set up a Signal Generator playing a tone (something not too piercing is recommended!).

I used a square wave playing at 100Hz. Now listen to how the Hall algorithm affects the pitch: it modulates up and down in a clearly cyclical way like a slow vibrato. Now listen to Church. It’s a much wider modulation. Next listen to Plate... you’ll hear it’s now modulating in this rather obvious and rather out‑of‑tune way. Keep this in mind when choosing which algorithm to use with pitched sources.

While the Avid Mod Delay is a flexible delay plug-in, if you’re looking for something a little more characterful, don’t forget the Tape Echo and BBD Delay plug-ins. These feature the distinctive bottom panel of dark grey buttons and knobs, which reveal their roots in the plug-ins originally developed for the Eleven Rack guitar interface. These plug-ins all respond to MIDI control in a way most AAX audio plug-ins do not, and so can be controlled from a standard MIDI controller.

If you don’t have a dedicated control surface and want to play your delays in real time for dubby delay throws that would be difficult to perform using a mouse, instantiate the Tape Delay on an aux, create a MIDI track, assign the output to the plug-in and use MIDI Learn to assign controls to parameters by right‑clicking the parameter and moving the encoder.

AIR Stereo Width

D-Verb can be very effective, but it does apply varying degrees of pitch modulation depending on which algorithm you’ve selected.

Unlike most AAX plug-ins, Tape Delay can be controlled using MIDI — great for dub delay throws!

mono. To toggle to stereo just bypass it. This feels the wrong way round to me but it’s easy enough to control without having to open the plug-in UI. Just remember to Command‑click the plug-in to toggle its bypass state.

Blue for bypassed equals stereo.

If you don’t have all the stock plug-ins installed, then it’s worth knowing that since Pro Tools 2024.3 you no longer need to restart Pro Tools to use newly installed plug-ins. For all plug-ins other than ARA ones, Pro Tools should find them while it is running. My experience is that some third‑party plug-ins don’t do this, but if you want to try any of the tips listed here and don’t have the necessary plug-ins installed, head over to your Avid account and download them. You can try them even if you have Pro Tools open right now.

So why not dive back into your Pro Tools plug-in library? Whether you’re a newcomer or a seasoned user, there’s plenty to explore in the stock plug-ins.

You might discover that the tools you already have are more powerful than you think.

While I don’t use most of the audio plug-ins in the AIR Effects Bundle, there is an exception.

AIR Stereo Width can be used as a workaround for a feature that otherwise is missing from Pro Tools: the ability to quickly mono check stereo material.

To do this, instantiate a Stereo Width plug-in on a stereo track. Use the first mode, Adjust, and set the Width all the way counterclockwise to 0 percent. Now with the plug-in active By dialling the Width right down, the AIR Stereo Width you are summing to plug-in can be used as a simple mono-sum button. www.soundonsound.com / December 2024

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Pro Tools Live

TECHNIQUE

Introduced in Live 12, Roar is much more than just a saturation plug-in...

OLI FREKE

R

oar is another of the amazing new devices that arrived with the major Live 12 update earlier this year. At first glance, it would appear to ‘just’ be another saturation and distortion effect, but look again... Roar combines analogue modelling techniques with extensive modulation and delay capabilities to create something genuinely different and deceptively powerful. As Marco Fink, lead developer of Roar, says: “The saturation curve in the analogue domain is never static.” And that idea of dynamism is something the team have deliberately put at the heart of Roar and extended to its logical limit.

Understanding how Roar works is the key to getting the best from it, and whilst it’s perhaps not completely intuitive at first glance — especially the feedback/ compressor stage — it doesn’t take long to grasp what each of the sections do and to start getting amazing results. Let’s dive in.

Input Stage Before you even apply the Shapers (which is what the distortion algorithms are called), Drive and Tone can be adjusted. Given the power of what’s to follow, this might almost seem superfluous, but one reason it’s there is to

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ensure good gain staging as indicated by the yellow Drive LED. The Tone knob is a kind of tilt EQ that can emphasise either the highs or lows in the input signal. This can be useful to remove some of the low end, which, having more energy, can otherwise cause heavier distortion more quickly. Below the Tone control is a frequency slider, which sets the frequency around which the tone control ‘pivots’.

Now that we have prepared our signal, the next most important decision is to choose which mode this instance of Roar will run in.

Single is the simplest mode and uses just one Shaper. In fact, it’s the best way to hear the basic raw (roar?) material. It’s highly instructive to use a sine wave to play a simple riff (generated by one of the new MIDI Generators, of course — see the October Live column) and step through each of the Shapers to get a feel for each of their characteristics.

Serial/Parallel/Mid-Sides are the different modes in which two Shaper modules can be arranged, Mid-Sides being a particularly interesting inclusion for creating spatial effects.

Processing the Sides channel to your heart’s content while keeping the Mid channel relatively Shaper’s 12 algorithms are a mixture of digital and analogue‑modelling options.

Opening Roar’s Expanded View allows you to see all stages at once and quickly make changes to any parameter.

unprocessed is one obvious — and powerful — use case.

Multiband provides a third Shaper for multiband processing — a standard option for a subgroup or drums mix.

Distortion effects often benefit from the low end being processed separately so as not to overwhelm the higher frequencies. Unless that’s what you want of course — distortion is a broad (wide-band?) church!

Feedback is a mode where a single Shaper and a Feedback Loop (also including a Shaper) can run in parallel.

More on Feedback later...

Distortion Stages The drop‑down Shaper menu is where you choose the distortion algorithms that will work on the source signal. There are two categories of Shaper, the first comprising fairly standard analogue and digital models: Soft Sine, Diode Clipper, Tube Pre-Amp, Half Wave Rectifier, Full Wave Rectifier, Digital Clip and Bit Crusher. All of which are fairly familiar (but great‑sounding) distortion algorithms.

The second type are novel saturators that provide a different kind of digital non-linear distortion. These are: Polynomial, Fractal, Tri Fold, Noise Injection and Shards. The distortion from these Shapers can often sound more like wavefolding than out-and-out fuzz (especially Tri Fold, which actually is a wavefolding algorithm!), but the main point is they are not standard analogue saturators, and provide an extended set of sound‑sculpting options.

In each of the Stage modules, there are a few simple controls: an Amount

knob to control how much distortion is applied; a Bias control for increasingly asymmetric clipping (which in the analogue-modelling types often leads to the sound itself breaking up, but with the digital models can sound more like a wavetable sweep through the distortion curve), and a Filter, which can be inserted pre- or post-Shaper.

There are eight filters: Low-pass, Band-pass, High-pass, Notch, Peak, Morph, Comb and Resampling. This last emulates a digital machine with a low sampling rate. So you could recreate the general sound of a 12-bit/12kHz sampler like the Akai MPC60 by selecting Bit

Roar’s eight filter options include a low-sample-rate emulator.

If you’re wondering where all the modulation destinations are when you first open the mod matrix, never fear: move a control in Roar and it will appear automatically in the grid. It’s a clever feature, but I feel that sometimes built-in plug-in prompts would help discovery with these kinds of hidden features, even if it disrupts the otherwise ergonomically clean display.

The modulation matrix is one of the key sound‑design areas and you should spend lots of time here! You could either very intentionally decide to modulate

“Modulating the amount of distortion, compression and delay feedback is a kind of sonic Three-Body Problem where all kinds of interacting effects start to take over as these different circuits push and pull against each other.”

Crush for the Shaper and Resampling for the filter! Though it’s not possible to dial precise values, you can certainly get lo-fi sampler crunch this way.

Modulation Section Modulation is really what sets Roar apart.

As stated above, this is a self-proclaimed dynamic saturator, and keeping settings static is very much not what the designers of Roar think you should do!

But it goes much further than just offering a gentle flux in general saturation to model an analogue circuit.

No, like many other Ableton plug‑ins, Roar features a full‑blown modulation panel, allowing you to add movement and change to practically any setting within Roar by virtue of two LFOs, an envelope follower and four noise sources. This is a major difference compared with other saturation plug-ins, where settings are usually just a case of ‘set and forget’.

The options are endless: you could choose to control the amount of distortion with the envelope follower. Or Bias amount could be changed randomly by the noise source. Or how about setting up two different Shapers in parallel mode and using the LFO to morph between them in time with the track?

parameter X with control source Y for a particular and desired result; or you could just experiment and be inspired by what happens: what Marco Tonni, the team lead for the Roar project, calls “controlled chaos”. And it really is fun with Roar. It’s impossible not to come up with unexpected sounds and really usable results through experimental whim and chance.

Feedback & Compression: If Modulation wasn’t enough, another key source of dynamic change is the Feedback section. The Amount control sets how much of the output of Roar is fed back into the input of Roar, which can, of course, lead to a simple positive feedback loop of pure distorted noise.

But this feedback loop has been given the extra superpower of delay as well. This means that depending on the amount of actual distortion in the chain, it can also function as a simple delay effect, with the Amount knob then also controlling the delay feedback.

The delay time can be set from 1 to 500 Hz in free Time mode, or set to musically significant chunks using 16th, Dotted or Triplet modes. Note mode sets a specific delay time to create that very short delay comb-filter effect, only in this case, it can be precisely set to be in tune with your track.

If you increase the amount of distortion within the Shaper stage, then things can get very wild as that distortion is fed back in a positive feedback loop to potential howl-round.

But this is where the final stage, Compressor, comes in to suppress the level of the feedback itself, preventing complete collapse into pure noise even at the very highest feedback levels.

Modulating the amount of distortion, compression and delay feedback is a kind of sonic Three-Body Problem where all kinds of interacting effects start to take over as these different circuits push and pull against each other. Which is really the summit (and point) of this extraordinary plug-in: creating effects that no ordinary distortion plug-in could ever do. Not to mention the changes you can experience as you play an instrument in real time through the effect, with Roar in turn affecting what you might choose to play in response.

I’d love to write more, but as you can hopefully tell, the addition of modulation and delay to what might otherwise have been ‘just another saturator’ (albeit with some novel digital programs) transforms this into a creative tool of some considerable depth.

I’ve barely scratched the sonic surface in the writing of this article and can’t wait to dig into it even more!

Right‑clicking a stage will give you the option of quickly copying settings from one stage to another.

www.soundonsound.com / December 2024

107

Pro Tools Cubase

TECHNIQUE

Organise your plug-in folder with VST Plug‑in Manager.

JOHN WALDEN

W

ith so many bundles, demos and tempting offers around, many of us probably have more plug-ins than is really manageable when we go to choose one in the middle of a busy mixing session; it’s all too easy to get lost among the possibilities. Thankfully, though, Cubase makes it easy to get your plug-in folder under control, thanks to its VST Plug‑in Manager. It offers plenty of options to customise which plug-ins are visible and how they’re arranged when you open a plug-in selection panel, and a little time spent here can repay itself many times over when your projects are in full swing.

Call The Manager When you first launch it, Cubase scans your computer and creates two default plug-in lists (Steinberg call these lists ‘collections’), one for VST Effects and another for VST Instruments. It also does a quick scan every time it is subsequently launched so it can detect any new plug-ins you’ve installed. By default, these collections are what’s shown to you whenever you go to select a plug-in effect or instrument.

The Studio menu (or a key command) access the VST Plug‑in Manager, in which you can view the contents of these two default collections. In its left window, you can view three different categories of plug-in that, between them, should include all of the plug-ins that are installed. These are: VST Effects, VST Instruments and the Blocklist.

The latter, which may well be empty, is simply a list of plug-ins that Cubase has scanned, but thinks might have compatibility issues — if you can’t find one you’ve installed it’s worth checking here; you can force a plug-in off the

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The VST Plug‑in Manager, with its multi-panel layout, lets you create custom collections of both VST Effects and VST Instruments to streamline your workflow within Cubase.

Blocklist if you are feeling brave, but I’d probably test that in a fresh project just in case! If you select an individual plug-in in any of these three panels, the bottom-left panel shows further useful information about it, including where on your machine it is installed.

The panel on the right shows the plug-ins in the current ‘collection’, depending upon which tab (VST Effects or VST Instruments) is selected in the left-side panel. These will, at first, be the Default collections (this name appears in the tab at the top of the panel). But you can move, remove, or add plug-ins to the current collection within this panel, including reordering (almost) any folder structure. For the default collections, via the top-right drop-down menu, you can choose to show the plug-ins organised

by Category (eg. EQ, Dynamics etc.) or Vendor; this choice is reflected in how the plug-ins are shown when you’re making selections in your Cubase project.

List Of Lists The VST Plug‑in Manger also lets you create your own custom collections, and there are a number of reasons why these might prove advantageous compared with the catch-all ‘default’ list. For example, you can compile a ‘favourites’ list — a compact selection of your current go-to plug‑ins that make it easy to find the right effect or instrument — or to hide expired demos, duplicate versions

of plug-ins or any plug-ins installed within a bundle to which you don’t actually have a licence.

Alternatively, it could be useful to create collections with only the plug-ins available to any collaborators you work with — a collection containing only Cubase Elements’ stock plug-ins, for example, the more comprehensive Cubase Pro options, or a specific third-party developer’s bundle. Another possibility is more streamlined collections designed for specific tasks, such as composing, tracking, mixing or mastering.

In each case, you can customise what’s visible so that you don’t get bogged down with certain plug-ins or plug-in formats that you don’t currently use or need to see (for example, mono versions, VST2 versions or older trial plug-ins).

Two points are worth emphasising.

First, adding or removing a plug-in from a particular collection doesn’t change its actual installation status on your computer, just its visibility in Cubase’s various plug-in selection panels. Second, whether in the default collections or new custom collections, your VST Effects and VST Instruments form separate lists.

New From Old Creating a new collection is easy. Select the VST Effects or VST Instruments tab in the VST Plug‑in Manager window, then simply choose one of the New Collection options from the drop-down menu

located top-right. Any of the three options (Empty, Add All Plug‑ins or Copy Current Collection) can be used, but if you’re currently using the Default collection, the last one is probably the most sensible choice, as it may take less time to move or remove plug-ins from an existing collection than to start from scratch.

Collections can be named, edited or deleted, and allow you to create custom folder structures that can be populated with your chosen plug-ins.

Let’s imagine we’re creating a ‘favourites’ VST effects collection.

First, create a New Collection from the menu using the Copy Current Collection option. This will prompt us to give our collection a suitable title (you can change it later). Once done, the new collection will become the active collection; its name will appear at the top of the VST Plug‑in Manager’s right-hand panel and the plug-ins within it will be what we see if, for example, we try to add a plug-in by clicking on a channel’s insert slot.

By default, our new ‘favourites’ VST Effects collection will contain folders based on Vendor. I prefer to organise my go‑to plug-in options by type, though, so a useful second step is simply to create a new folder (just click on the New Folder button located top right), name this folder ‘Vendors’, and drag and drop all the individual Vendor folders into this one. This folder will serve as a useful catch-all category in any collection; while it hides them in a subfolder, if your current ‘favourite’ selections don’t offer you quite what you need for a specific task then you can still easily seek something out here.

At the top level of this folder structure, alongside the Vendors folder,

you can now use the New Folder button as many times as required to create your own custom list of categories. For example, in my own VST Effects ‘favourites’ collection for mixing duties, I’ve got folders such as EQ, compression, limiting, saturation, amp modelling, pitch-correction, creative, vocals, drums, bass, reverb, delay and tools.

Finally, you can find those favourite plug-in choices and then drag and drop them into their appropriate place in the new custom folder structure. This drag and drop can be done in two ways. First, you can drag them from the catch-all folder you created (the Vendors folder in our example) or, second, if you want a particular plug-in to appear in multiple folders (for example, a channel strip plug-in might appear in a dedicated channel strip folder and also within an EQ folder), then you can simply find it within the main left-side panel and drag and drop it from there as many times as needed.

In my case, my selection of VST Effects favourites might only consist of three or four go-to plug-ins in each category.

When subsequently adding plug-ins to my project, this makes finding those go-to choices in the plug-in selection panels much faster. Creating a favourites collection for VST Instruments can be done in exactly the same fashion. And, of course, you can easily edit/update any collection as your tastes change or new plug-ins replace older favourites.

Switch & Find

Custom plug-in collections exist globally on your host system so, once created, they can be accessed in any Cubase project.

A further neat element of these custom plug-in collections is that you can quickly switch between them: just visit the VST Plug‑in Manager, select either the VST Effects or VST Instruments The drop-down menu located top right lets you create new plug-in collections tab, drop down and instantly switch between any of the existing collections.

With its custom folder structure and just a few ‘go‑to’ plug-ins within each category, a ‘favourites’ VST Effects plug-in collection lets you quickly pick what you need and get on with your mix.

the menu from the top left and you can then choose the required collection.

The change is applied instantly so, next time you go to select a new plug-in instance within your current project, you will see your plug-in choices as organised in the currently selected collection.

Finally, note that changing the collection doesn’t influence any plug-ins already inserted into your project — a plug-in that isn’t included in the current collection won’t suddenly disappear from your project; you simply won’t be able to ‘see’ it in the plug-in selector panel to insert a new instance.

Pick & Mix As soon as your plug-in collection starts to expand beyond the stock Cubase offerings, the workflow advantages of creating custom collections via the VST Plug-in Manager is an absolute no-brainer.

So don’t let your plug-in addiction slow you down! www.soundonsound.com / December 2024

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Logic Logic

TECHNIQUE

We revisit some old tape masters with the help of Logic’s new Stem Splitter.

PAUL WHITE

I

covered the new intelligent Session Players back in the September issue, but the other headline feature of Logic Pro 11 is a surprisingly capable stem splitter. The less good news is that the stem splitting facility is only available to those running on an Apple Silicon machine, presumably because it makes use of the AI technology in the M-series processors.

For those unfamiliar with stem splitting, the aim is to be able to extract individual parts from a complete mix, something that is accomplished on a spectral level and often achieved with the help of machine learning.

Unsurprisingly, the results are never quite perfect, as even the best-trained AI sometimes fails to separate the harmonics of one instrument from those of another, but the way the separation takes place means that if the individual stems are then added back together, the original audio file is recreated exactly — which means you can often usefully rebalance elements, even if the separation isn’t perfect.

Logic Pro’s Stem Splitter is to be found in the Functions menu and it can separate a track into a maximum of four parts, or ‘stems’, categorised as Vocals, Drums, Bass and Other: guitars, keyboards and so on fall into the Other category. If you have guitar and keyboards playing at the same time, you can’t extract the guitar track as you can with some of the more exotic (and more expensive) third-party stem splitters.

Time To Split To make this work, all you have to do is import a mix into Logic Pro 11, select it in the Main window and then go to Stem Splitter, which is at the top of the third section down in the Functions menu. If no audio track is selected, Stem Splitter will be greyed out.

A window opens showing the four stem options with a tick box by each one, so if, for example, you just want to separate the vocals, you’d only tick the Vocals box. If you want the full four-way split, tick all four boxes. Having done this, you click Split and in a surprisingly short period of time (typically half a minute for an average-length song) your stems are displayed as four separate tracks within a Track Stack.

In my own tests, examination of the individual tracks revealed that vocals are separated surprisingly well, though there are sometimes slight artefacts as the software puts some vocal information into

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the wrong stem. Drums also separate out very cleanly, with the elements most likely to suffer tonal changes being things like crash and ride cymbals — but it’s still pretty impressive, and these all come back cleanly when the stems are recombined. Likewise, you can mute the drum track and all traces of the drums vanish, leaving you free to add a new drum part if you want to. Bass guitar also splits out fairly cleanly, though on some occasions high notes or higher harmonics might end up in the Other section along with the regular guitars.

What Next?

If your plan is to extract vocals from a mix to use in your own compositions (assuming of course that you have the permission of the copyright holder), the results can vary depending on the character of the voice and on the type of mix you are extracting it from. If effects such as echo have been applied to the voice, those will also be extracted, often with a lesser fidelity than had the voice been reasonably clean.

On the flip side, if you mute the vocal track, you generally end up with a pretty clean backing track. However, for me that’s not the real power of stem splitting. As a test I found some recordings made by my own band dating from around 1978 to 1980, which had been copied onto cassette tape. All the original recordings and the machines that made them were long gone by now. These recordings were the result of my earlier adventures in home recording, the originals made on a Tascam four-track open‑reel machine with the only effects being tape echo and a spring reverb.

Back then, even mics such as the Shure SM58 seemed unaffordable, so I shudder to think what cheapies we pressed into service! Despite the ravages of age and the limitations of the gear available at the time, the recordings still sounded fairly clean, though the cheap mics lacked any decent bass response, so the tracks were definitely short on low-end punch.

Combine & Conquer As long as you plan to recombine all the tracks after processing, there’s a surprising amount that can be achieved without introducing noticeable side‑effects.

For the vocals I used some gentle pitch correction and then a compressor (all courtesy of Logic Pro’s own plug-ins), as we didn’t have anything so exotic as a compressor back then. I also added

Stem Splitter gives you the option to separate complete mixes into up to four parts: Vocals, Drums, Bass and Other.

a little tape delay, as none had been used on the track I chose to experiment on.

For the drums I used compression plus some low EQ boost, and a similar combination was employed on the bass guitar, as that too was sounding distinctly thin. The Other section, which comprised

my electric guitar and Roland GR‑300 guitar synth (sadly also long gone) was largely left alone, with just a little low-mid EQ to help fatten it up. All this was done fairly quickly under the guise of experimentation, but when the four parts were played together, the track sounded warmer and more solid without any of my cheating trickery being obvious. I was also able to lift the vocal track by a dB or two.

I then did a few further experiments using Logic Pro’s Stereo Spreader plug-in to add some width to the Other and Drums tracks, as the limitations of the time meant that pretty much everything was in mono. Though it has been around for quite a while, Logic Pro’s Stereo Spreader is actually quite a useful and often underappreciated tool, as you can change the position of the filter peaks by adjusting the frequency limits and the filter order, the latter determining how many peaks and troughs the filter will create. This allows the filters to be tuned by ear to a critical part of the sound being processed, for example the pitch of a kick drum. The process is

mono-compatible too, as whatever is done to one channel is applied in the inverse to the other, which is what creates the illusion of stereo width from a mono source. If the process is affecting the bass end too much, reducing the Lower Int fader to apply less processing at the low end of the spectrum is all that’s needed.

Summary The ability to extract elements from a mix to reuse in new projects seems to be the headline use for stem splitting, but for me, the technology is most valuable when used to improving existing archive mixes, by allowing you to processes the stems of a mix individually before recombining them. As my example shows, it is also possible to add more of a sense of stereo width to mixes originally made in mono, both by adding stereo ambience/ delay effects or by using artificial stereo widening techniques such as Stereo Spreader. Another valid application would be to, for example, remove the bass guitar parts and then record a new part to replace

it, or take out the drums to use what’s left as the basis of a remix.

On my test mix, muting the bass guitar track left no noticeable trace of the bass guitar when the remaining three stems were recombined. It might also be possible to remove the Other stem so that you can build a new mix using the original vocal, drums and bass parts, though in my experience this is when vocal stem artefacts tend to show up the most, as small parts of the vocal spectrum end up being wrongly shunted into the Others track.

The success of this type of operation varies depending on the source material, so give it a try and let your ears decide.

I still think that stem splitting is magic, though, and its side-effects are far less than I would have believed possible.

Listen Online You can check out the before and after audio files of this restoration/remix job, as well as the individual stems, at https://sosm.ag/logic-1224.

Once you’ve selected an audio track and invoked Stem Splitter, Logic will create a new Track Stack and populate it with the extracted stems.

www.soundonsound.com / December 2024

111

SPOTLIGHT

SPL MTC Mk2

Desktop Monitor Controllers In the age of mixer-less studios, monitor controllers are on the rise!

LUKE WOOD

T

he first thing that springs to mind when you think of analogue consoles is likely their front end — the mic preamps, filters, EQs, dynamics processors, and so on — but their monitoring sections also played a hugely important role in the traditional studio, providing a convenient way to switch between input sources and speaker systems, set monitoring levels, communicate with artists and quickly check a mix’s mono compatibility, among other things.

Many studios today have no console; we often look to modern audio interfaces, DAW software and plug-ins now to provide some of these facilities. But there still remain strong arguments for using a dedicated physical device. A hardware monitor controller can accommodate additional input sources and speaker outputs, without sacrificing interface I/O or even requiring you to boot up your computer. It can put dedicated controls for your most-used functions within arm’s reach. Importantly, it puts a hardware volume control between your interface and speakers too, and could potentially save your equipment or ears in the event of unexpected level jumps or full-scale noise caused by a computer or interface crash.

There’s no shortage of monitor controllers available now, and they come in various shapes and sizes. Some are API MC531

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passive, others use active electronics, and they range from simple stereo volume controls, through desktop devices with more features, up to very sophisticated rackmount units with remote controls that handle large amounts of I/O and incorporate cue-mixing and talkback facilities. If you want to learn more about monitor controllers in general, and why certain facilities might or might not be useful to you, check out Hugh Robjohns’ October 2021 article (https://sosm.ag/ monitor-controllers). This month, we’ll shine the SOS Spotlight on a selection of desktop monitor controllers intended for those working in stereo.

three stereo line-level outputs along with a mono output designed for feeding a subwoofer (all with independent level trims) and an output for driving an external meter, as well as a pair of independent headphone amps that can either mirror the control room signal or receive any of the five inputs. Along with level control, the MC531 offers mono, dim and mute (left, right and overall) functions, and also features a built-in talkback mic. $ $2495.

W https://apiaudio.com

API MC531

Designed to bring fully featured monitor control to DAW users, API’s MC531 is a standalone version of the centre section of their renowned analogue consoles.

Three stereo line-level inputs are joined by a switchable USB/AES3 digital input, while mobile devices are catered for by front- and rear-panel mini-jack inputs and Bluetooth connectivity. You get

Audient Nero

Audient Nero

Another manufacturer bringing their console expertise to the standalone controller world are Audient. The Nero combines a pair of line-level inputs with a dedicated cue input and an alt source that will accept either unbalanced line-level (via RCA or mini-jack) or digital (coaxial or optical S/PDIF) signals. There’s also a built-in talkback mic, as well as the option of connecting an external one. Three pairs of speaker outputs are joined by a mono subwoofer out, and there are four headphone outputs that each benefit from their

own top-panel level controls and source selection switches. The main level control utilises Audient’s Precision Matched Attenuation Technology to ensure perfectly matched volume in the left and right channels, and as well as dim, mute and mono buttons, the Nero includes a polarity switch, making it possible to audition the stereo difference (Sides) signal. Different monitoring configurations can be saved using a Smart Touchpoints system, which offers some handy features such as permanently setting an output to mono without preventing the mono switch from working on other sources. $ $579.

W https://sosm.ag/audient-nero

an unbalanced input on both mini-jack and RCA connectors, along with three stereo outputs, a subwoofer out and two headphone amps. All of the monitor outputs benefit from individual level trims, and there’s a built-in talkback mic that can be routed elsewhere, thanks to a rear-panel output. As for monitoring checks, the MC2.1 offers left, right and overall mutes along with dim, polarity and mono functions. Lastly, the MC3.1 expands connectivity even further with an AES (or S/PDIF) digital input and external talkback connectivity, as well as introducing a Cue section that can route any input to either headphone amp independently of the main output. A greater selection of monitoring checks are present, too, with low-, mid- and high-band solos joined by a left/right swap function. $ CMC2 $359, CMC3 $799, MC2.1 $799, MC3.1 $1249.

W https://sosm.ag/drawmer-cmc2 W https://sosm.ag/drawmer-mc2-1 W www.drawmer.com

Hayakumo Izarai Drawmer MC3.1

Drawmer CMC & MC Series

Of the seven monitor controllers in Drawmer’s range, you’ll find four desktop models aimed at stereo setups. The smallest, the CMC2, has a pair of balanced stereo inputs and an unbalanced mini-jack input, along with support for two pairs of speakers, a dedicated mono subwoofer output and a built-in headphone amp. Its larger sibling, the CMC3, adds a third stereo output and a second headphone amp, along with a digital S/PDIF input and both internal and external talkback options. Both are kitted out with mute, dim, mono and polarity switches, and the CMC3 adds individual left/right mutes, and the ability to store and recall a listening level. Next up is the MC2.1, which features two balanced stereo inputs along with Drawmer MC2.1

The Hayakumo Izarai is a compact controller that features a simple, passive design and employs high-quality Japanese-made components.

Rather than a potentiometer, the unit uses a 23-step rotary switch for level-control duties, allowing it to offer Hayakumo Izarai adjustment in fixed steps (2dB increments in the most-used range, and 5 or 10 dB steps at the upper and lower zones), as well as guaranteeing accurate stereo tracking. A pair of stereo inputs are joined by three stereo outputs, one of which can be switched in and out independently — a handy function to have when using the output to connect a subwoofer — and a thru output for feeding an external meter. You also get mono and mute functions, as well

as a dim switch that can be set to apply either 10 or 20 dB of attenuation. $ $530 W https://sosm.ag/hayakumo-izarai W https://en.hayakumo.tokyo

Heritage Audio RAM 1000

Heritage Audio RAM Series

As well as their large rackmount System 5000, Heritage Audio’s range of monitor controllers includes a trio of desktop units that cater for a variety of needs and budgets. First up is the Baby RAM, a compact passive unit equipped with a pair of stereo inputs and outputs, a 24-step rotary switch that offers level adjustment in 3dB steps, and a trio of buttons that provide mono, mute and dim functions.

The larger RAM 1000 adds a headphone amp, an input level meter and a third stereo input that offers convenient Bluetooth connectivity. Stepping things up a gear, the RAM System 2000 features a trio of analogue inputs and outputs along with subwoofer and mix outputs, and includes both S/PDIF and Bluetooth inputs. There’s also a second headphone amp and additional stereo I/O connections for routing cue mixes to artists, along with a built-in talkback mic with three routing options. A 64-step switch offers volume adjustment in 1dB increments, and it’s possible to store and recall two preset listening levels. $ Baby RAM $199, RAM 1000 $499, RAM System 2000 $999.

W https://sosm.ag/heritage-ram1000 W https://heritageaudio.com www.soundonsound.com / December 2024

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SPOTLIGHT

DESKTOP MONITOR CONTROLLERS

Lake People MC100

Lake People MC100

Lake People’s MC100 has a useful selection of analogue I/O, including two balanced stereo inputs, and unbalanced inputs on both mini-jack and RCA sockets. It also includes a USB port that allows it to function as a high-quality D-A converter, capable of handling PCM signals up to 32-bit/384kHz and DSD 556. The inputs can be activated individually or simultaneously, and can be routed to four outputs (three stereo and one mono subwoofer out) as well as a pair of slave outputs designed to feed external meters or recorders.

There’s also a built-in headphone amp with an adjustable crossfeed feature designed to emulate the experience of listening on loudspeakers. Level adjustment is handled by a 41-detent ALPS RK27 potentiometer, and there’s a healthy selection of function switches including left, right and overall mutes, left/right swap, polarity, dim and mono.

There’s also the MC100 Pro, which replaces the volume control used in the standard model with a 256-step relay-controlled attenuator. $ M100 €1899.89, M100 Pro €2399.90.

W https://sosm.ag/lake-people-mc100 W www.lake-people.de

Mackie Big Knob Passive

Although the larger models in Mackie’s popular Big Knob range have now evolved into audio interfaces, the series still includes a dedicated standalone monitor controller in the

Mackie Big Knob Passive

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form of the Big Knob Passive (not to be confused with the original Big Knob, which was active).

As its name suggests, this device has a passive signal path, while Source and Monitor Select buttons make it possible to switch between two input sources and two pairs of monitors. Along with its large volume control, the unit is equipped with mono, mute and dim functions. $ $69.99 W https://mackie.com

Palmer Monicon XL

Palmer’s Monicon XL has been designed to act as the centrepiece of a modern studio setup, combining input and speaker switching with talkback and cue-mix facilities. There are two balanced line-level inputs, and a third input that will accept either unbalanced analogue line-level signals through a pair of RCA phono connectors or a mini-jack input, or a digital S/PDIF on a separate RCA connector. The unit’s three speaker outputs all feature an adjustable level trim, and they’re joined by a summed mono output that’s intended for hooking up a subwoofer or dedicated mono reference speaker. As well as feeding the main monitoring path, any input can also serve as a cue mix, and there are two headphone amps that can be switched between main and cue mix paths — both signal paths can also be passed on to external gear using rear-panel outputs.

Volume, mute, mono and dim controls are provided and, as well as a stereo level meter, the Monicon XL also boasts a correlation meter that provides a visual indication of a mix’s mono compatibility. $ $399.99.

W www.palmer-germany.com

Palmer Monicon XL

PreSonus Monitor Station V2 & MicroStation-BT The latest iteration of PreSonus’ flagship desktop monitor controller aims to offer a convenient and affordable way of managing a studio’s monitoring system.

Two pairs of analogue stereo inputs are joined by a third channel that accepts either a digital S/PDIF signal on its own RCA connector, or unbalanced analogue via mini-jack or RCA. You get three pairs of monitor outputs with individual level trims, and four headphone amps that can monitor either the controller’s main

PreSonus Monitor Station V2

PreSonus MicroStation-BT

output or any of the three input signals.

There’s also a built-in talkback mic for communication with artists over their headphones, and the cue mix can be passed on to other devices thanks to a pair of rear-panel outputs. Along with the obligatory volume control, the Monitor Station V2 is equipped with mono, mute and dim buttons. If you’re after a more compact solution, PreSonus also offer the MicroStation-BT, which pairs a single stereo input and output with a subwoofer out and a wireless Bluetooth input. $ Monitor Station V2 $349.99, MicroStation BT $129.99.

W www.presonus.com

Radial Nuance Select

Radial devices are well known for their bulletproof construction and impressive technical performance, and their recently introduced monitor controller is no exception. The Nuance Select is housed in a rugged steel chassis with an aluminium faceplate, and employs all-analogue Class-A circuitry that achieves an outstandingly low THD figure of less than 0.00001%! The device is capable of switching between two input sources and two pairs of monitors, and includes a mono output designed to feed a subwoofer. There’s also a pair of headphone amps with independent level controls and source-selection switches, along with an unbalanced stereo output that allows signals to be passed on to other gear. As for monitoring functions, the Nuance Select’s large level control is joined by mono, mute and dim buttons. $ $699.99.

W https://sosm.ag/radial-nuance-select W www.radialeng.com www.radialeng.com

RCF MC-1

RCF MC-1

Housed in a compact, rugged steel chassis, RCF’s MC-1 is a simple passive controller capable of switching between two input sources and two pairs of speakers. The first input accepts balanced line-level signals via XLR/TRS combo sockets, while the second caters to both balanced and unbalanced devices with a pair of quarter-inch TRS connectors and a stereo mini-jack input — both outputs are provided on XLR sockets. On the top panel, you’ll find a pair of switches for selecting the active input and output, as well as a large volume control with mute, mono and dim buttons. $ $149.

W www.rcf.it

SPL MTC Mk2 & 2Control

features a pair of stereo inputs and outputs, along with a subwoofer out, a line output for passing the selected input to another device, a mono switch, and a built-in headphone amp with an adjustable crossfeed control based on SPL’s Phonitor Matrix. The 2Control then offers a pair of stereo inputs (which can both be active at once) with three stereo monitor outputs and a mono output for feeding a sub or single speaker, as well as adding a second headphone output.

Their flagship stereo desktop unit, the MTC Mk2, still features four outputs (three stereo and one mono), but doubles the input count and includes a built-in talkback mic. Additional cue mix, talkback and metering outputs are present, and one of the inputs can be configured as a dedicated artist cue mix. A healthy selection of monitoring function buttons is provided, including left/right swap, polarity, mono, dim and mute. $ Control One $599, 2Control $699, MTC Mk2 $1299.

W https://spl.audio

SPL 2Control

SPL’s range boasts some very serious, high-end mastering consoles and surround controllers, but they also offer several more compact desktop units that are designed for simpler stereo setups. Their simplest is the Control One, which

Radial Nuance Select

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The SOS team pick their favourite hardware and software from the last 12 months.

Cranborne Audio Carnaby HE2

Studio Hardware This year, Cranborne became one of only a small handful of companies to offer a plug-in remote-controlled analogue outboard processor, and their Carnaby HE2 is something very special indeed.

The digital control side of things is, from a user perspective, refreshingly unfussy — the plug-in works well, and the hardware ‘feels’ analogue — but there are also some ingenious extras. Importantly, the clever analogue circuitry combines EQ with per-band saturation to deliver effects I’ve not heard from any other manufacturer. The price isn’t ‘aspirational’ either. Definitely a winning combination! Matt Houghton https://sosm.ag/cranborne-carnaby-500 Sticking with the plug-in remote-controlled hardware theme, Wes Audio have achieved amazing things with their ‘ng’ processors.

Every time one comes to me for review, I conclude that Wes must have taken the concept about as far as it can go — yet, next time, they always push the concept forward in imaginative ways. Their ngTube

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Wes Audio ngTube EQ

EQ is, at heart, a four-band passive mastering EQ with a switchable valve and transformer output stage. It sounds divine.

But it also showcases several novel features, including a revamped approach to plug-in remote control, a context-sensitive screen, and touch-responsive controls that access many useful ‘hidden’ features. Will it be the pinnacle of Wes Audio’s achievements? Only time will tell, but when it comes to digitally controlled analogue outboard, everyone else is playing catch-up.

Matt Houghton https://sosm.ag/ngTubeEQ SSL’s PureDrive Octo rackmounting mic preamp provides superb audio specs, with an incredible

90dB gain range, and a blizzard of additional features including filtering, switchable ‘colour’ options, variable impedance, built-in A-D conversion and even USB interfacing. And at not much more than £200 per channel, it’s remarkable value for money. Sam Inglis https://sosm.ag/ssl-puredrive

Another EQ? Really? Yep. And it’s another one that sounds great and has features that make it stand out from the crowd. With only six knobs (three being dual-concentric) and a handful of toggle switches, the Undertone UTEQ-500 gives you access to pretty much any curve you can imagine, including the possibility of notch filters! Speaking of curves, while you can master basic console-style EQ’ing on this module pretty swiftly, if you’re to get the most out of it, there’s a bit of a learning curve. But get your head Undertone Audio UTEQ-500 around it and operation becomes

Teaching Machines Wellspring Atomic Analog Special OP-6

SSL B-Dyn 611B

AEA TRP3 & RPQ3

second nature. So much power in such a tiny package. Matt Houghton https://sosm.ag/uteq-500

sounds incredible — on mics, as a DI, and as a line-level saturation processor. Sam Inglis https://sosm.ag/aa-special-op6

The Teaching Machines Wellspring is an analogue spring reverb, but it’s also much more than just a reverb device.

With its analogue delay, modulation and filter sections, it’s a box that invites and encourages you to create unusual and original effects. It can be a superb‑sounding straight-up, luxurious spring reverb if needed, but the real fun is deciding if the world (or your client!) is ready for the sonic statement you’ve created with this excellent sounding all-analogue stereo effects device!

Neil Rogers https://sosm.ag/tm-wellspring

AEA’s TRP and RPQ preamps have always been special, but the new third generation raise their performance to the true pinnacle of engineering excellence. Although originally designed for passive ribbons, the current TRP3’s capabilities reach far wider, and the latest design offers phantom power and can be used with any microphone.

The unusually high input impedance minimises the load on dynamic and transformerless capacitor microphones, allowing them to deliver their best possible sound quality. And with up to 85dB of switch-selected gain, even the least sensitive mics and the quietest of sources can be raised to a useful signal level with ease.

The discrete TRP3 design boasts one of the lowest noise floors of any preamp, while also enjoying incredibly low harmonic distortion, a huge bandwidth, and generous headroom.

This all translates to a phenomenally detailed and accurate, yet tonally neutral preamp — the apocryphal ‘straight wire with gain’. The RPQ3 occupies 1U of rack space instead of the half-rack format of the TRP3, and provides a more versatile EQ facility plus a balanced insert option, but is otherwise identical in performance. Hugh Robjohns https://sosm.ag/trp3-rpq3

Plug-ins can be so good these days that it takes something special to make me want to route a signal ‘out of the box’ when mixing.

SSL’s B-Dyn 611B is one such device, and I was surprised by just how much I enjoyed putting it through its paces this year. Yes, it’s an SSL dynamics processor (there’s both a compressor and an expander/gate on board), and with that company’s reputation you’d expect it to be decent. But having used both the E-Dyn and plug-in versions of both the E and B type processors before, I wasn’t expecting this module to sound quite so distinctively punchy, aggressive and in-your-face as it does. Not your ‘use on everything’ compressor, but a very handy tool that I love. I’ve still not been able to dial in the same effects with plug-ins, or even get close quickly. Matt Houghton https://sosm.ag/ssl-b-dyn You don’t need an Atomic Analog Special OP-6. I don’t need an Atomic Analog Special OP-6. But having tried it, I couldn’t bear to part with it. This modern recreation of RCA’s venerable valve preamp just

Monitoring Monitor controllers may be functional rather than glamorous, but the right one can make a huge difference to any control room. In terms of features, ergonomics and sound quality, the Radial Nuance Select is pretty much the perfect fit for me and my setup. It’s also built to survive anything that studio life could throw at it. I bought one! Sam Inglis https://sosm.ag/radial-nuance-select

SSL PureDrive Octo

ADAM Audio A8H

The A8H marks ADAM Audio’s impressive entry into the compact three-way monitor market. Its new MLM bass and DCH midrange drivers, paired with ADAM’s signature X-ART pleated ribbon tweeter and a tri-power hybrid PWM and Class‑A/B amplification setup, produce a seemingly effortless wide, deep, detailed and solidly centred soundfield. ADAM’s updated A-Control software simplifies the process of configuring the AH8, and its integration with Sonarworks’ SoundID Reference room calibration software helps deliver a flat response at your listening position.

Its outstanding audio performance and attractive price point offer outstanding value for money — so much so that I bought a pair.

Bob Thomas https://sosm.ag/adam-a8h I can’t quite remember how I came across the AIAIAI Unit-4, but I’m really pleased that I did. The Unit-4 is a compact and portable battery‑powerable monitoring

Radial Nuance Select

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Microphones

AIAIAI Unit-4

system that also offers wireless connection (both between its left and right speakers and from a source device), and despite the limitations of its modest dimensions, drivers and amplification, it works well.

It’s perfectly suited to monitoring on the go in hotel rooms and in those remote Scandinavian beachside summer houses we’ll all find ourselves writing melancholy pad‑based music in one day. The thing I really loved about the Unit-4, though, is its approach to sustainability. Its enclosure is made from post-consumer plastic and it’s designed to be fully repairable.

The cabinet comes apart easily (no glue) and replacement parts can be purchased from the AIAIAI website. All other monitor manufacturers (yes, that means you), please take note. Phil Ward https://sosm.ag/aiaiai-unit4 Just getting its foot in the door of 2024 before it slams shut is the Kii Seven. Given the extraordinary monitoring impact of the Kii Three over the last five years or so, a more compact version of the same technology from the same Bruno Putzeys brain was likely to be significant. And the Kii Seven doesn’t disappoint. In some ways, it’s perhaps a stronger overall package than the Kii Three because it’s more suited to compact studio spaces, and it’s significantly less expensive (although still unaffordable for most of us). But it has the same otherwordly ability to make mix details audible that most mortal monitors pass by completely.

And the best thing, from my speaker engineer perspective, is that there’s no snake oil or magic involved in the Kii Seven’s performance, just brilliant and innovative electro-acoustic design and engineering. Phil Ward https://sosm.ag/kii-seven

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Warm Audio WA-19

Cloning vintage gear has always been Warm Audio’s core business, but the quality and accuracy of their reproductions has reached new heights in the last couple of years.

The WA-19 is the first ever attempt to replicate a classic moving-coil dynamic mic — the legendary AKG D19 — and the results are impressive. Sam Inglis https://sosm.ag/warm-wa19 The Lulu Tube was my first encounter with Australian mic manufacturers BeesNeez, and I was suitably impressed. Hand‑built small‑diaphragm valve capacitor mics can often come with a hefty price tag (especially vintage models!), so it’s great to have an option that’s within reach of a small‑studio owner like myself. Great on strings, acoustic guitar, drum overheads, and even vocals, these little mics can be detailed and clean when needed while offering a nice sense of subtle tube saturation when pushed a little harder. Neil Rogers https://sosm.ag/lulu-tube I probably own a dozen mics that I’d cheerfully use on snare drums or toms, but getting a decent sound from close-miked bass drums always seems like more of a challenge. The Earthworks DM6 solves that challenge at a stroke, presenting the perfect balance of low-end thump and transient punch with minimal processing required. Sam Inglis https://sosm.ag/earthworks-dm6 Latvian mic manufacturers JZ have built a great reputation in recent years for their distinctive-looking mics that offer excellent value for money. I got to try out the BT202, their first ever small-diaphragm capacitor mic, and found it to be one the best‑sounding mics I’ve tried on my studio upright piano.

With its detailed, balanced sound and excellent transient response, it’s become a regular fixture in my studio on any of the sources where you might typically use a pencil mic.

Neil Rogers https://sosm.ag/jz-bt202 Kii Seven

Earthworks DM6

BeesNeez Lulu Tube

JZ Microphones BT202

Audio Interfaces Among all the monitor reviews of the year, it was an interface that stood out.

Audient’s ORIA is the first reasonably affordable USB interface intended specifically for Atmos mix duties, and if any other interface manufacturers decide to go for the same market, they’re going to have quite a job improving on it. As well as creating an interface that looks and feels just right, Audient have equipped the ORIA with exactly what’s needed for a mid-sized studio Atmos workflow, and made it delightfully straightforward to configure and use. Slick integration with Sonarworks multi-channel room and monitor optimisation, and a properly sorted and usable control application, serve to make the ORIA proposition even more convincing.

Phil Ward https://sosm.ag/oria The Blackstar Polar 4 has a couple of useful features to help out vocalists and guitarists. The guitar input has a switchable saturation/limit mode that prevents clipping and adds warmth as the input level is raised, while the XLR inputs can be switched to provide a vintage-style preamp character.

Alternatively you can simply use it as a straightforward USB interface with those

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GEAR OF THE YEAR

Audient ORIA

RME UFX III Blackstar Polar 4

MOTU UltraLite Mk5

Lynx Hilo 2

features turned off. I like the Polar 4 in particular, because those extra ins and outs provide a means to use hardware effects to process the DAW sounds or to record small ensembles, and they also come in handy when re‑amping. There’s a very generous suite of included software, too, including the St James amplifier plug-ins, and the price is very attractive. Paul White https://sosm.ag/polar4 The original Lynx Hilo was introduced way back in 2012, and incredibly, despite the passage of a dozen years, few other devices have managed to match, let alone beat, its extraordinary technical performance. However, Lynx are not a company to rest on their laurels, and this year they released an updated version — the Hilo 2 — with the same unique styling and popular feature set, while introducing several key operational improvements and, most importantly, an impressive step up in the technical performance of its converters. The latter have reestablished the Hilo 2 on the top of my converters league table, and I think it’s fair to say that this update represents a phenomenal, benchmark-setting professional converter with stunning sound quality, superb features, great

Apple Logic Pro 11

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versatility, and even easier operation than its forebear. Hugh Robjohns https://sosm.ag/hilo2 RME’s UFX interfaces have held the top spot as the company’s flagship products since 2011, originally offering 60 channels of simultaneous I/O, which was astonishing for the time. Various updates and related models have been introduced since then, culminating in the latest star: the UFX III. For aficionados of RME’s product range, this newest model is functionally similar to the UFX+, but boasts new and seriously impressive converters, a quartet of stunningly clean and noise‑free mic preamps, a wealth of connectivity and features, and USB 3 computer connectivity.

With 12 analogue inputs (four mic and eight balanced line), and digital inputs spanning stereo AES3, dual ADAT and 64 channels of MADI, the UFX III accommodates a whopping 94 inputs in total. There are also 94 outputs, including digital, analogue and two stereo headphone feeds — 188 channels in a 1U rackmount box is an impressive feat! I couldn’t fault the UFX III in any way, and it’s impossible not to be impressed by its capabilities, sound quality, and massive connectivity. Hugh Robjohns https://sosm.ag/ufx-iii

Harrison Mixbus 10

To judge it by its I/O count, the MOTU UltraLite Mk5 would seem pretty unremarkable. But its technical performance is outstanding, while some of its features are borrowed from MOTU’s much more expensive devices. You can remotely control its DSP from your phone, you can drive modular synths with it through its DC-coupled line outs, and you can master through its top-tier converters. You can even link all 10 line outs to the front-panel monitor encoder, making it the smallest and cheapest Atmos-capable interface around.

Factor in MOTU’s excellent form for product support and driver updates, and the Mk5 should serve a lot of users very well, for a very long time. Chris Korff https://sosm.ag/ultralite-mk5

DAWs Updates to Logic are always welcome, and this time around, the jump to Logic Pro 11 has improved the auto drummer feature and also added automatic keyboard and bass players, along with new instruments for them to play. These newcomers can generate convincing parts automatically from a chord track populated by the user.

Although chords can’t yet be extracted from existing performances, that ability

PreSonus Studio One Pro 7

will hopefully be added some time soon.

There’s also a new stem splitter that does a fair job of splitting a mixed track into vocals, drums, bass and whatever else remains, though that particular feature only runs on Apple Silicon machines. Paul White https://sosm.ag/logic-11 Harrison’s Mixbus has built its reputation on its ability to deliver both the sound and the recording/mixing workflow of a classic hardware mixing console. It undoubtedly retains those console-style elements that have always made it unique, but it has evolved into a mature and very well-featured environment for the recording and mixing of music in almost any style, whether in stereo or — in Mixbus 10 Pro — in Dolby Atmos. If you are looking to recreate the sound and experience of the analogue studio world, but with the convenience of a software environment, Mixbus 10 absolutely nails it. John Walden https://sosm.ag/mixbus-10 PreSonus’ workstation has always felt to me like software for the modern age: fast, user-friendly, supremely flexible and ceaselessly innovative. Studio One Pro 7 raises the bar considerably. A new Launcher environment takes Ableton’s Session View paradigm but flips it, literally, for an exquisitely fluid relationship between linear and loop-based workflows. There’s groundbreaking integration with the vast Splice sound library, browsable in-app, incorporating near-instantaneous pitch and key recognition. All this, in addition to offline stem separation, cutting-edge Dolby Atmos mixing/rendering (introduced in version 6), and a beautifully rounded feature set generally, makes this the DAW to beat, I reckon. Robin Bigwood https://sosm.ag/studio-one-7

Synths, Keyboards & Controllers It’s rare that something truly innovative appears in the synthesizer world and, when it does, it’s not always

Native Instruments S88 Mk3

displayed automatically in the main display, making it easy to see what you’re tweaking.

It works very smoothly and provides direct access to all the commonly used channel functions, and features a coloured LED lighting scheme to help you navigate. It also looks pretty cool! Paul White https://sosm.ag/nektar-panorama-cs12

Arturia Polybrute 12

the success that its inventors had hoped.

But the FullTouch modes embodied in the Arturia PolyBrute 12 take us beyond aftertouch, poly-aftertouch and even MPE, and at least one of them offers a unique method of shaping notes that can extend players’ performances in ways that we’ve not heard before. I hope enough of us take the time to master FullTouch so that it’s adopted more widely (under licence, of course) and becomes one of our standard forms of keyboard expression. Gordon Reid https://sosm.ag/arturia-polybrute-12 I’m a fairly heavy Komplete/Kontakt user, and until recently my controller of choice was the original version of NI’s S61. I often found myself running out of keys, though, and although I’m certainly no pianist, was keen to switch to something with hammer-action keys. With Native Instruments’ S88 Mk3, browsing instruments and tweaking parameters is easier than ever thanks to its large built-in display, and both the browser system and Light Guide now work directly with Kontakt rather than relying on the Komplete Kontrol wrapper plug-in or application. The S88 is also a rare beast in that its hammer-action keybed boasts polyphonic aftertouch, so as well as doing a great job when paired with piano libraries, it’s also a very capable synth controller. Luke Wood https://sosm.ag/s88-mk3 The Nektar Panorama CS12 goes beyond most singe-fader surfaces by adding a panel of 12 rotary controls and four buttons to directly access Nektar plug-in parameters.

Panorama CS12 The parameters are

Now and again, you come across something that’s simply genius. Earlier in the year, FlatSix released an unassuming little one-octave button keyboard that fits snugly into a 1U Eurorack space. The Slight Of Hand was useful for quick demos and drone notes, and it earned a place in my case. A while later FlatSix released new firmware and a front panel that has radically transformed my approach to almost every patch I make. The Slight Of Hand became a most unusual arpeggiator: one which you add notes to, like an X0X sequencer, but can then shift to accommodate new notes and drop out old ones. It’s instant, interactive, fabulous for live performance and jamming something on the fly. It has multiple play modes, up to 16 notes, looks amazing and has the best name in modular: the ARP Of Darkness. Robin Vincent https://www.flatsixmodular.com

Software SpectraLayers Pro 11 has come a remarkable distance since Steinberg acquired it from Magix in 2019. This is, in part, linked to the rapid developments in AI, and many of the improvements in SpectraLayers are built on AI technology.

However, what’s just as impressive with SLP 11 is just how accessible it makes insanely complex audio editing tasks — including various types of audio unmixing — available to any level of user. Truly remarkable technology. John Walden https://sosm.ag/spectralayers-11 Best Service’s The Score By Sonuscore makes me smile every time I use it.

It manages to combine a really good underlying orchestral sample library with an amazing performance engine that lets you sketch out sophisticated orchestral

FlatSix ARP Of Darkness

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Steinberg SpectraLayers

Accentize dxRevive Pro

arrangements with a combination of just a few chords, a couple of keyswitches and a nudge or two of the mod wheel. Creating epic orchestral arrangements within your own DAW should not be as easy as this.

With an impressive ensemble sound, and a slick workflow, this is a hugely inspiring platform for composition. The Score scores on many levels. John Walden https://sosm.ag/the-score-sonuscore For many years I’ve been beseeching manufacturers for a softsynth that emulates the ARP Pro Soloist. Although there will be many who don’t understand why I keep banging on about this rickety old preset monosynth, it can sound wonderful, so I have to compliment Cherry Audio’s developers for their excellent job in recreating it. They’ve also extended their virtual Pro Soloist to do things far beyond the capabilities of any original ARP synthesizer. It’s eminently affordable, too.

Gordon Reid https://sosm.ag/cherry-pro-soloist Sound Particles’ inDelay leverages the company’s proprietary 3D surround panning within a delay plug-in that allows the repeats to be placed (or to wander around within) a surround space. Pretty much every surround format is supported right up to Dolby Atmos, but there’s also a binaural option for those who want to add a sense of space and movement to a stereo headphone mix. Up to 100 delays can be created, making it possible to set up whole clouds of delays within the virtual space.

Paul White https://sosm.ag/sound-particles-indelay

Synchro Arts VocAlign Pro 6

Accentize’s dxRevive Pro has quite simply been the most useful time-saving tool in my studio this year. It’s a real-time plug-in (with a little latency) that cleans up dialogue, removing noise, reverb and common digital artefacts, as well as synthesizing missing high and low frequencies. This year’s update brought new algorithms that give you more options, which made it more usable on a wider range of material. Without a doubt, anyone who does any amount of work with dialogue captured in less than ideal conditions should own this plug-in.

(Now the technology’s there, if they’d only train it to do for acoustic guitar what it does for dialogue!) Matt Houghton https://sosm.ag/accentize-dxrevive-pro I’d never had really good results from formant‑shifting software until Dreamtonics’ Vocoflex ended up on my review to-do list. Not only does it minimise processing artefacts but it also enables the user to import one or more voice samples and then use their formant structure to modify the voice being processed. It is easy to use, and if you import multiple voice samples, the graphical interface makes it easy to change the influence each imported voice has.

There’s also a voice-changing mode that doesn’t rely on imported voices. Paul White https://sosm.ag/dreamtonics-vocoflex

Best Service The Score By Sonuscore

lined up, I hadn’t actually tried Synchro Arts’ other offerings until VocAlign Pro 6 arrived this year. The big news for Pro Tools users is ARA2 support, which speeds things up significantly and helps make VocAlign feel very much a part of the DAW itself rather than a plug-in. Equally impressive for me, though, is the introduction of Revoice’s Process Groups function. Being able to quickly create variations of settings to suit different parts and then apply them across groups of tracks has been a huge time-saver, and I’m already wondering how I put up with doing everything manually!

Luke Wood https://sosm.ag/synchro-arts-vocalign-6 It’s easy to become enamoured of a new plug-in when you’re reviewing it, but that doesn’t always mean it enters the core arsenal of processors you use on every project. Oeksound’s ‘make it sound better’ Bloom plug-in absolutely has, and my only reservation is that it sometimes feels like cheating! Sam Inglis https://sosm.ag/oeksound-bloom At a NAMM Show a few years back, I tried to persuade some well-known software makers to build a ‘spectral ducker’. While a conventional ducker pulls down the level of one signal to make room for another,

Quite a lot of the projects I mix involve big stacks of vocals, and although I’ve put Revoice Pro to work on a few occasions when getting takes Cherry Audio Pro Soloist

Oeksound Bloom

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Sound Particles inDelay

FireSonic FireSpacer

a spectral ducker only ducks conflicting frequencies — and that’s exactly how FireSonic’s FireSpacer works. The result sounds far more natural than a conventional ducker, and it is also very affordable.

Paul White https://sosm.ag/firesonic-firespacer

Guitar Technology I have a number of things that will do the sound of an 1176 very well, but sometimes the way you physically interact with a unit and the information that its control settings and displays give you is what determines how well you will work with it. Like many contemporary recreations of this classic, the Warm Audio Pedal76 is actually quieter than the ‘real thing’ in terms of self-noise, whilst still offering plenty of the original circuit’s character when pushed, in this case courtesy of a real output transformer.

But, nice as it is, it is not the sound that sets the Pedal76 apart for me. Although big by pedal standards, it is a lot smaller than even the half-rack size of my UREI LA-4, which means I can more easily keep it right by me when I’m recording. It might seem odd for a unit prominently featuring a footswitch, but it actually makes more sense to me placed on my desktop. Compact Warm Audio Pedal76 pedal compressors also tend not to have space for the full control set, whereas the Pedal76 not only has large, easily viewed controls, but also a real mechanical VU-meter-style gain reduction meter — simply the best way to read gain reduction. Dave Lockwood https://sosm.ag/URL

else on the market. I doubt there’s ever been a multi-amp modeller made that didn’t have a set of somewhat generic Marshall distortion tones, but a good non-master‑volume example from the classic years is so much more than that.

Marshalls can actually do gloriously rich, musical clean and semi-clean tones as well as distortion, and those are beautifully represented in the UA modelling. There’s probably nowhere other than a big festival stage where I could turn up the real thing to its sweet spot volume any more, but playing through a Lion ’68 is still like being reunited with an old friend. Dave Lockwood https://sosm.ag/warm-pedal76

Fender Tone Master Pro

When the world’s oldest, most storied guitar‑amplifier manufacturer launches a digital amp and effects modelling workstation with a Mac/Windows control app, you know that the guitar world has just shifted on its axis. The Fender Tone Master Pro’s current selection of amplifiers (both their own and from other brands), and wide variety of effects pedals and cabinet IRs, will give you more than enough to be going on with. However, its performance, competitive price point, technology, eight-core processor and the fact that it has had five updates since launch indicates that Fender are into modelling for the long haul.

Bob Thomas https://sosm.ag/fender-tone-master-pro

Digital modelling of guitar amps and speaker cabs has come a long way over the last few years such that even very affordable models are capable of a wide range of convincingly authentic tones. But If all that Two notes’ Genome plug-in points of difference remain, even if many could do is allow you to work with the of them are just in the realm of company’s proprietary amp player preference. Universal models and renowned DynIR Audio’s ‘one amp per pedal’ cabinet impulse responses range, with its one-size-fits-all within your DAW, then it hardware form factor, no would be a great product.

MIDI and limited presets, However, the plug-in is clearly didn’t suit everyone, also compatible with but to me, the sound and AI captures from NAM, feel — especially the dynamic Aida-X and Proteus, and touch‑responsiveness — using Two notes’ Codex Universal were a cut above anything engine, it can even Audio Lion ’68

Two notes Genome

manipulate these quite extensively — and that’s what could well catapult Genome into being the software hub of choice for those wishing to combine existing collections of plug-ins with the possibilities offered by AI-based model captures. Bob Thomas https://sosm.ag/two-notes-genome Positive Grid’s Spark range of phone-controlled modelling amps keeps growing, with more and bigger models seemingly launched every month. But it’s their smallest, the Spark Go, that I’ve been enjoying the most. Having recently had a son, my ‘studio’ is starting to resemble a crèche storeroom, but the tiny, battery-powered Spark Go lets me play electric guitar anywhere the baby isn’t sleeping, at any time and with minimal fuss. The tiny speaker punches way above its weight, while the DSP is capable of reproducing any tone you could wish for.

It’s a cute gizmo, and a very useful one too.

Chris Korff https://www.positivegrid.com Positive Grid Spark Go

Designed to work in tandem with IK Multimedia’s Tonex amp capture software, the Tonex One provides a means to take a favourite amp rig on the road (or into the studio), all packed in a mini stompbox format fitted with a large Drive knobs plus three small, dual‑function knobs for, EQ, compression and reverb adjustment. The idea is that you use the included Tonex software to set up your ideal amplifier, either by capturing an amp you like or by picking one of the factory options, then you transfer it to the pedal via IK Multimedia USB. Neat! Paul White https://sosm.ag/ik-tonex-one Tonex One w w w. s o u n d o n s o u n d . c o m / December 2024

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Sean Cook & Nevin Sastry: ‘A Bar Song (Tipsy)’

Along with Shaboozey himself, Sean Cook (left) and Nevin Sastry were the main production and writing team behind ‘A Bar Song’ and its parent album.

The biggest hit of the decade so far took Shaboozey and his team half an hour to write — and weeks to perfect.

PAUL TINGEN

“I

t was crazy,” remembers Sean Cook. “It wasn’t a planned thing.

It was wild. We were working on ‘Annabelle’, a song on the album, and we were like, ‘Let’s take a break.’

Nevin and I were joking around on guitar, playing whatever chords came to mind, and Shaboozey responded with some words and melodies. In that moment it just happened. We went with it, and boom, the song gods gave us

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this idea! We weren’t really thinking.

For me it was a blur.

“Shaboozey knew right away that we had something special. I also had a gut feeling, and put my iPhone on record and set it to the side. I recorded the whole song being written in a 30-minute voice memo. Afterwards, in about 15 minutes, we cut Shaboozey’s lead vocal. We later recut a couple lines, but 95 percent of his vocal is from that day, which is shocking because Boozey is very particular with vocals. But he was like, ‘Don’t change it!’”

‘A Bar Song (Tipsy)’ has enjoyed record-breaking success on the Billboard Hot 100.

“Sometimes when making an album,” adds Nevin Sastry, “you’re trying to fill a spot. This was absolutely not that. It was just us having fun, messing around. And it turned out great. In the clip that Sean has on his voice memo while we were writing it, you can hear Shaboozey saying, ‘This is my country number one. I know.’”

Magic Moment At that moment in November 2023, the three musicians in the room — Sean Cook, Nevin Sastry and Collins Chibueze,

aka Shaboozey — knew they had struck gold. Even so, none of them could have predicted what would follow. At the time of writing, Shaboozey’s ‘A Bar Song (Tipsy)’ has been number one on the US Billboard Hot 100 singles chart for 15 weeks. By the time you’re reading this, you will know whether ‘A Bar Song’ will have surpassed the current US record of 19 weeks at number one, set in 2019 by another crossover country song, Lil Nas X and Billy Ray Cyrus’ ‘Old Town Road’.

‘A Bar Song’ has made Shaboozey the first black male artist to top Billboard’s Hot Country Songs and Hot 100 charts at the same time, immediately after Beyoncé had paved the way by becoming the first black artist to do the same with ‘Texas Hold ’Em’. In fact, Beyoncé’s number one album Cowboy Carter, released in March 2024, helped introduce Shaboozey to a wider audience, thanks to guest performances that were in themselves inspired by ‘A Bar Song’. This was the result of a showcase Shaboozey did in January at Winston House in Venice Beach, LA, appearing on stage with Cook and Sastry.

“We were only showcasing four songs,” says Sastry, “and although we had not quite completed the production of ‘A Bar Song’, we decided that we had to perform it.

So the three of us performed the song with a backing track that was pretty close to what the world would eventually hear. The reactions were crazy. We saw everybody from everywhere in the bar run to the stage. Shaboozey was still a bit skeptical about the 808 that Sean and I had put at the end of the song, but when he saw the audience reaction when the 808s came in and rumbled the stage, he was like, ‘OK, I get it.’”

“I’ve never seen anything like it,” agrees Cook. “When the pre-chorus started, 20 people immediately walked in from the other room, asking, ‘What song is this?’ It’s the first time I’ve ever seen the energy shift instantly like that! Apparently, the show also led to Shaboozey being invited to appear on her album. That night and that song are such a launching pad for everything, it’s crazy.”

of production and writing work. “I’m an Indian kid from Ohio,” explains Nevin Sastry. “I started playing piano and guitar when I was 12, and did all the guitar shredding and so on, which led me to jazz. I then went to med school and worked as a doctor, but in 2016, when I was 24, I decided that music was my calling, and moved to LA. I’m a guitarist, writer and producer, and first met Shaboozey in 2016. He ended up moving into my apartment. He signed with Republic that year, and through that I signed with Savan Kotecha, a producer who worked with Max Martin.

“Having come from making beats in my bedroom, pure instinct, I got to learn from the Max Martin crowd! Savan got me to really cut my teeth. I worked on Shaboozey’s first album, Lady Wrangler [2018], but because it was hard to make money through placements, I also did a lot of music for sync. I did all the music for two seasons for a CBBC kids show called Almost Never. I had to write and produce and cut all these covers and

inclinations, despite the artist starting out as a rapper. Cook: “When I first met him, he said, ‘I’m from Virginia, and want to have a sound that really represents Virginia.’ He grew up listening to Willie Nelson, but also Young Thug, and wanted to combine both worlds.”

Crossing Boundaries In 2023, when Cook and Shaboozey were about to start work on the artist’s third album, Shaboozey suggested involving Sastry, who had contributed to one song on Cowboys Live Forever. Sastry recalls that Shaboozey was by now more determined than ever to integrate all his influences, especially country.

“When I first met him, he was wearing a Canadian tuxedo, ie. all denim, and the first thing I heard by him, unreleased, was based on a Led Zeppelin sample. He’s always been inspired by classic rock, country, guitar-driven things. He’s so good at rapping that people constantly wanted him to rap, but we made real progress when he decided that he could do whatever. He had an understandable fear of the market not understanding him. It took him a while to get to the point where he felt he could release these country-influenced records.

In the end, the timing worked out.”

The timing involved Shaboozey riding a crossover country wave spearheaded by black artists that was arguably started

by Lil Nas X in 2019, and led to the success of Beyoncé’s country album, and eventually ‘A Bar Song’. Sastry and Cook, meanwhile, brought their own country experiences to the table. “I grew up by corn fields close to Ohio, which is the Midwest,” says Sastry, “and I was working with country crossover artists right before Shaboozey.”

“I did not grow up with country,” admits Cook, “but in 2020-21, when Covid shut down the LA music industry, I stayed in Nashville, and was around a lot of country music, and was at many sessions and saw the inner workings of that world.”

Most of the songs on Where I’ve Been, Isn’t Where I’m Going were co-written

by Shaboozey, Cook and Sastry, with the latter two producing. Perhaps for this reason, the album has a consistent and unique sound, with songs that often have anthemic melodies, and production that is strongly country-influenced, but

Nevin Sastry: “Once you get to 30 acoustic guitar tracks, you really have to make sure they’re all glued together, otherwise the mix gets terrible.”

Getting To The Start As is always the case, the 30 minutes of magic that produced ‘A Bar Song’ were made possible by a long history

mix and master everything for TV. To learn more, I spent a lot of late nights watching YouTube videos and reading Sound On Sound!”

Sean Cook, meanwhile, grew up in Southern California, mostly Sevenoaks and Orange County. “My mum was a vocal teacher and my dad a guitar teacher, and they wanted me to play music in church. I learned to play guitar and bass, and was playing basketball competitively, but after an injury I went into making beats to pass time. My dad said, ‘These are actually pretty good,’ and I got an internship with a producer, which opened me up to the idea of it being a job. I played in a lot of bands in my 20s, but always with a producer brain, wanting to be in the studio writing and producing songs.”

Like Sastry, Cook eventually signed a publishing deal as a songwriter. He first met Shaboozey in 2020, and worked with the artist on his second album, Cowboys Live Forever, Outlaws Never Die (2022), a title that already revealed clear country

www.soundonsound.com / December 2024

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‘A BA R S ONG ( TIP S Y ) ’

The hitmaking trio at work in LA’s Playhaus Studio, with Nevin Sastry picking up the acoustic guitar that features prominently on Shaboozey’s album.

also, as Rolling Stone put it, “a full-on barrage of post-genre possibilities”.

In other words, ‘A Bar Song’ is not an outlier on the album.

“We never walked into a room saying ‘Let’s make a Nashville country song,’” insists Cook. “We just made Shaboozey songs. Nevin’s skill set and mine catered to that. When the three of us combined, it made the Boozey sound.”

“Definitely,” Sastry adds. “A lot of it comes from the writing and Boozey being authentic in what he appreciates about country music. He loves simplicity, directness, hitting the emotions. The first album he and I did had maybe a few country influences, but was more a pop-rap-trap album. The album he did with Sean had more country influences, but was still mostly trap, with maybe two fully organic songs. The new album is far, far more organic.”

High Contrast On the day in November 2023 when ‘A Bar Song’ came into being, Shaboozey, Sastry, and Cook were at the end of the writing and production process for the artist’s new album. They had occasionally collaborated with co-writers and co-producers, and some of the recordings had taken place in Nashville,

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but the two of them held the overall production vision, and were for the vast majority of time holed up in Cook’s studio, because there was building going on in Sastry’s house. Cook’s studio contains “Barefoot Footprints. I have an M3 MacBook Pro, and an Apollo 8 that I run everything through. I also have Yamaha HS3s. They’re crappy speakers but good for referencing. I have some toys, like some outboard and pedals and synths and stuff, but obviously we were more organic for this project.

“When we took that ear break one evening and Nevin and I came up with the guitar rhythm and chords, Shaboozey said, ‘I want to do a 2000s rap song flip.’

He loves that era of rap, and came up with lyric references to J-Kwon’s ‘Tipsy’, mainly ‘Everybody at the bar getting tipsy,’ and ‘One, here comes the two to the three to the four.’ He referenced the lyric and the cadence of the song, not really the beat. We were playing guitars as he was figuring out the idea, and then he made the cadence fit what we were doing.”

Sastry: “I remember Boozey fishing for melodies, and we were like, ‘Oh, that one!’ Then he came up with the opening lines, ‘My baby want a Birkin, she’s been tellin’ me all night long / gasoline and groceries, the list goes on and on,’ and

it’s really cool how he was blending rap elements and country elements. Birkin is rap and gasoline and groceries are country. We did this juxtaposition of rap and country with the entire song, the production, the lyrics. Everything is a contrast, which makes it interesting.”

Endless Guitars Being aware that they had struck gold, the trio spent somewhere close to two weeks in total on the arrangement and production, making sure they got them absolutely right. In so doing they expanded on some of the production elements that were already present in many of the other songs on the album: specifically, roomy, moody atmospheres, often with acoustic guitars, gang vocals and percussion elements, recorded with mics at a distance.

“A lot of that is to Boozey’s credit,” explains Cook. “He said to us, ‘I don’t want it to sound too clean, too pop, too in your face. I want there to be an element of roominess in it.’ ‘A Bar Song’ has many acoustic guitars. In the verses there’s my initial acoustic with a Nashville high-strung guitar [the three lower strings replaced with a lighter gauge and tuned an octave higher], and the choruses have a bunch

of guitars, played by Nevin, John Mark Nelson and myself. The main ones are Nevin and I each playing a double left-right of acoustic, and then a high-strung underneath. We ended up adding tons of layers of acoustic guitars, and also some electrics, and a banjo.”

“There always is a deceptive amount of guitars in our music,” agrees Sastry.

“With the acoustics in ‘A Bar Song’ we were using different distances from the

mics, to get different tones and different ambiences. We had a couple of days when we set up four different mics in Sean’s living room and were recording every possible permutation. We tried to get that Lumineers big room vibe playing 10 feet away from the microphone. We’d also throw a Valhalla Room verb on these guitars. I don’t know if it was 100 percent wet, but we tried to give them as much depth as possible.

“The challenge with acoustic guitar is that once it is in the box, you can’t do too much. You can time and chop and do whatever, but the main thing is how you perform it and how you mic it. We also added a lot of fingerpicking and more arpeggiated guitar patterns. A lot of the creativity in the production comes from layering voicings, using different players, different performances, and mics and distances and everything. And

Nevin Sastry works mainly in Ableton Live. This screen shows part of the initial production session for ‘A Bar Song’.

www.soundonsound.com / December 2024

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the creative mixing after that, so in the end, people think it’s just one guitar! It was really important that it felt as much as possible like Shaboozey singing over one strummed guitar!”

“We recorded five or six guitars up to 15 feet away from the mics, all in different tunings, mostly using my Mojave Audio MA-200 mic, Nevin’s Warm Audio WA-251 mic, the sE Electronics RNT Rupert Neve tube mic and a Royer ribbon,” adds Cook. “John Marks brought some Neumann KM184 mics. We used several preamps, but for the most part we went through my Hazelrigg VLC, which is amazing, and adds a wonderful colour. We panned the guitars, and added more reverb, all to give this air of them being in a room.”

In The Room While roomy acoustic guitars are the most immediately apparent element of the ‘A Bar Song’ production, there also are roomy claps, stomps, gang vocals and more, all contributing to the unique feel.

“We spent a lot of time doing the claps,” elaborates Cook. “There’s Nevin and I clapping in the living room with the mic in my hallway, really far away. There’s a hall clap like that in the verses, and in the pre-chorus we added two more, left and right, and in the chorus two gang claps.

So the claps are constantly growing throughout the song. There’s a lot of detail that gives the song a roomy feel, which makes you feel like you’re in a saloon, singing the song.”

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Halfway through production, ‘A Bar Song’ was transferred to Pro Tools for mixing. This screen capture shows the various handclaps that Nevin and Sastry recorded, making clear how they build throughout the song.

“We also did a lot of gang vocals,” adds Sastry, “again recorded with mics at a distance. We had all guys and then we were like ‘We need some female energy. Nobody wants to be in a bar with only guys!’ So we got some girls together as well. We also took parts of the audio from the video of the showcase to get the vibe of the crowd noises from the show. We added a crazy amount of details, which resulted in a huge amount of tracks. It’s very multi-layered.

Sean Cook: “We tried to keep it simple, but we didn’t really manage!”

“We also had a fiddle. Shaboozey came over and he and I found this Splice fiddle loop and we chopped it up and created a melody from that.

We then had a real fiddle player come in and play the line and we blended the loop and live fiddle together. She’s playing the main line in the post‑chorus, and also the fills, and really brought it to life. We used some creative panning in using them both, because there was some magic in that original loop.

It’s funny that the fiddle ended up becoming such a staple of the song.”

“The day we wrote the song, we had just guitars and vocal,” says Cook.

“We tried re-tracking them and we

tried different options later, but in the end, they didn’t have the same magic. So we ended up keeping the original vocal and guitars. We recorded Shaboozey’s vocals with the Mojave MA-200 going through the Hazelrigg.

I also added an 808 on the first day, but I muted it because I knew Boozey would go, ‘No, no, no, no, I want something that’s not been done before.’ Credit to Boozey, because he always wants to do something that’s different, and 808s are everywhere now, even in country. So during the month we produced the song, he said he did not want to do anything trap-related at all. I respected that, because it’s become a lazy thing to put trap drums and an 808 on a country track.

“The drums on many of the songs on the album had been played by my brother, who has a 12-mic drum setup in his studio. But after that first session, Shaboozey told us that he did not want any drums on ‘A Bar Song’.

Nevin and I were like, ‘This is the biggest song ever. We have to put on drums!’

So we bounced guitars and vocals of the first day, and Boozey sent us some references. Once we had all the guitars and claps and roominess and the more indie acoustic vibes to where Boozey liked them, we threw in that 808 at the end, which ties the production together.”

“We tried to keep it simple, but we didn’t really manage!” admits Cook. “There’s a lot of ear candy, and the drums really push the song.

a White Claw alcohol drink, and then pouring it into a pint glass with the sound of the fizz and everything. It’s right in the beginning, when we’re also talking and laughing. The talking and laughing goes on throughout the whole song and gradually gets louder and louder.

“We automated the rise in volume of these gang vocals. So the last hook is massive, with a lot of talking and laughing in the background. We aimed for the sound of everyone singing by the end of the song. These room fun noises under the whole track give it the feel of liveliness that it has. There’s a lot of those cinematic textures across the album.”

Glueing The Sound

Most of them are programmed, using samples that we recorded ourselves.

There’s a kick stomp that’s a hit on a guitar case. We layered in many of these stomps. And there are some live drum brushes, and a soft crash cymbal when the chorus hits. There’s also an upright bass and the gang vocals coming in, which gives that moment a lift. Plus there’s a piano on the second hook, and Farfisa and B3 organs in the last hook.

“Boozey likes to add cinematic live textures. We were trying to add that extra one percent of energy and fun. One example of this was when Nevin got on the mic and recorded himself opening

Sastry and Cook both stress that the distinctive sound of ‘A Bar Song’ was primarily down to experimenting with mic placements. “Looking back, most of the technical stuff I did on this project was to do with miking,” recalls the former. “I work in Ableton, and Sean works in Pro Tools. Most of what I did was record into Ableton, tracking with light EQ and compression, using mostly stock plug-ins, because my laptop was in bad shape. I’d say half the production was done in Ableton, and the other half in Pro Tools. And then we ended up rough mixing in Pro Tools. Eventually, things always end up in there, and with modern production you are also mixing.”

“During recording, we were being very creative, cranking up the Hazelrigg and trying different settings

to get different sounds,” agrees Cook.

“After recording we added plug-ins; there’s a lot of EQ on the guitars, and for the gang acoustics we did a bus with a room verb, to make it darker.

We used the FabFilter Pro-Q 3 and the Safari Pedals Flamingo Verb for that. Also, we sent all the acoustics to a mono bus that’s mixed in at a low level behind the panned guitars, because we have so many wide instruments. We gave it more clarity with some EQ. Nevin’s acoustics in the chorus were played harder and more aggressively, so we had to be careful how we blended them in.”

“Sean and I put as much love as possible into timing the acoustic strums, beyond even the performance, because once you get to 30 acoustic guitar tracks, you really have to make sure they’re all glued together, otherwise the mix gets terrible,” says Sastry. “Also, there’s not a ton of processing on Boozey’s vocal. There’s a basic Pro Tools EQ taking out low end, there’s the Waves CLA 76 doing some slight compression, and the Waves Rvox pushing it forwards a little. And then there’s a de-esser. From there it goes to a lead vocal bus, which goes through the [IK] T-RackS 6, and there’s a FabFilter Pro-MB, a mono Flamingo Verb, centred, and we also had a Valhalla reverb on his vocals. Shoutout also to Raul Lopez, who mixed this album. On ‘A Bar Song’ specifically he ran all the claps through the tape machine at his studio and back into Pro Tools.”

Framing Art

Stillwell Audio’s Event Horizon was used on the drum bus at the mix “with slight clipping, to give the drums grit and punch”.

“Boozey likes to throw a lot at the wall,” concludes Sean Cook, “and then we figure out what works. With this one, it was finding the right balance of the organic vibe that Boozey liked, but also keeping the hip-hop side. It’s such a big song, we didn’t want to shy away from that. We tried all these different things and then we reeled it in and muted what we needed to mute.”

“Boozey was quite involved, because he cares about the framing,” saids Nevin Sastry. “He cares a lot about his art. He’s a great artist. There’s really no other way to put it. He loves music and likes to be involved in all the details of it. That’s why he’s been successful. You’re hearing something unique and true to him, and we’re helping him bring that vision to life.” www.soundonsound.com / December 2024

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

track/7vrLphPq4yjaX7jiDdd2Nc

W www.youtube.com/

watch?v=YIqbdnaPcT8

Garth Richardson • Mudvayne ‘Dig’ J O E M AT E R A

C

anadian producer and engineer Garth ‘GGGarth’ Richardson has worked with some of music’s biggest acts, including the Red Hot Chili Peppers, Rage Against The Machine, Kiss, Alice Cooper, Ugly Kid Joe and Taylor Swift to name but a few. He currently works out of his own studio facility, The Farm, located in British Columbia. A sonic highlight from his vast discography is the bass sound from Mudvayne’s 2000 track ‘Dig’.

“It took us four days to achieve the distinctive slapping bass sound on this track. Mudvayne’s bassist Ryan Martinie tried all of the amps we had in the studio. We had an Ampeg SVT cabinet, a Mesa Boogie bass head, a Euphonic Audio bass cabinet and a Marshall 444 cab. We also put up foam that was one foot thick by four feet by eight feet between each cab to stop the bleeding

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between the mics. I did this as it takes the room sound out of the mics.

“I also used a great trick with cables that I learned from Michael Beinhorn when I worked with him, where we spent a few hours flipping guitar cables back and forth, as the cables sound better going one way. It always blows away every band when we show them.

The less resistance gets you a better bottom end. Everyone tells me I’m full of shit when I start telling them about this voodoo thing. Please try it. I was not a believer, but now I am!”

Persistence Pays Off “For two days straight, we battled to get Ryan’s sound, but he wasn’t happy with any of the sounds he was getting.

By day three, he was so frustrated that he couldn’t get the sound in his head to come out of the speakers.

He seemed like he was a beaten-down man, so we sent him back to the hotel for the evening.

“On day four, he came back and began slapping his strings on his five-string Warwick bass. He was excited and said ‘This is the sound I’m looking for!’ So, engineer, Dean Maher, myself and Ryan put headphones on and tracked inside the control room with all the amps going at the same time. We used an AKG C451 to mic the

strings because we had no shotgun mics in the studio. Finally, we captured the bass sound that he was after.

“We also used a [Neumann] U47 tube on the SVT, and Sennheiser MD 421s and a Neumann U87 on the other cabinets too, to get some different textures. I tend to use as few mics as possible because of phase issues.

At the end of the day, the intensity and challenge we went through made it all worthwhile. The end result speaks for itself. The bass sound made that record stand out and you can really hear it when it comes through any speaker.

“The entire sound of the band was driven for the most part due to Ryan finding his sound. And that slapping sound is so integral to the track with its distinctive tone. One fun part of the recording was that during the mixing session, I kept telling Andy Wallace to turn up the string mic — and Andy, being a bass player, had a huge smile on his face, which doesn’t often happen.”

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10 Phantom Rooms Low End Toys

Kontakt Instrument

HHHHH Low End Toys is 10 Phantom Rooms’ third release in their Low End series of Kontakt instruments. Like Low End Strings and Low End Modular, the focus is deliberately on the lower end of the frequency spectrum but, as suggested by the title, the sample base in this latest release is built from a collection of acoustic, mechanical, and electronic toy-like instruments including wooden and glass toys, toy robots, clocks, trains and cars, as well as toy pianos, guitars, flutes and synths.

This underlying +3GB sample set forms the basis of over 160 Kontakt Snapshot presets, each of which combines up to four sound layers. The Snapshots are organised into a number of useful categories — Bass Drones, Basses, Bells, Keys, Pads and SFX/Toy Stories — and while the emphasis is often of the lower note ranges, there are plenty of sounds here that can also be used to create melodic lines. In terms of sonic moods, I can imagine the sounds working in some more experimental and/ or avant-garde musical styles, but a more obvious role would be for film/media composers looking for some child-like sound elements for their next creepy horror score. In that context, Low End Toys would make a great starting point.

The presets sound really good, but the Kontakt front end provides plenty of options for DIY sound design. You can, of course, create your own blend from the underlying samples, and a sound browser provides some tag-based options when exploring these. However, what’s particularly impressive is the range of effects and modulation options, elements of which you can access from across the four main pages of the UI: Main, Mixer, Edit and Animate. By default, the mod wheel lets you blends between three ‘states’ of each Snapshot and, interestingly, you can tweak the preset’s settings and save these into one of the three states. You can also switch the mod wheel’s focus to the six macro-style controls — Animate, Filter, Space, Echo, Drift and Time — if you prefer. The latter provide a rapid means

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of adjusting some of the deeper sound‑design options across the other pages of the UI. These include velocity, LFO and step‑sequencer-based modulation of filter, reverb, echo and distortion. With options to also pitch-shift individual sound sources within your preset ±12 semitones, there is plenty of potential for those that want to fully capitalise on what is a very cool and quirky sample set.

OK, so a sample-based virtual instrument based upon toys is something of a niche product but, if you compose music in suitable styles, or have a more experimental streak, Low End Toys is an absolute joy to explore. It’s also very fairly priced. Low End Toys is no toy; it’s an ideal sound source for film composers needing to creep out their audience.

John Walden €99 www.10phantomrooms.com

Native Instruments Alicia’s Electric Keys

Kontakt Instrument

HHHH Almost a decade and a half after their first collaboration, titled Alicia’s Keys, Native Instruments and multiple Grammy-winner Alicia Keys return with a new offering — more keys from Alicia, only this time they’re electric. While the first virtual instrument sampled her Yamaha C3 Neo grand piano, the second release captures the sound of her personal Yamaha CP-70 electric grand piano, with an aim to cover “a vast range of tones that blend classic electric pianos with flexible modern soundscapes”.

Recorded at Jungle City Studios in New York, this Kontakt instrument provides highly detailed control of the sound from the get‑go. Starting with the instrument samples themselves, you can choose between two source signals — pickup and microphone — with the ability to blend between the two. Next is the

ability to adjust the dynamic range, attack, sub, overtones, and resonance of the sounds. For added character, tweak the Noise setting to dial in (or out) the sound of the hammer and pedal, add in a noise loop, and manage the stereo spread of both the DI and mic signal.

The original piano had 73 keys, but this instrument allows you to extend the key range to 88. This could come in handy, but note that there is a definite sweet spot for most of the presets where the instrument sounds best. In several patches, the notes at the furthest ends of the range feel different, tonally, from the character of that particular preset — if you closed your eyes and played across the full range, the spell would be broken when you hit the extremes.

There is nothing subtle about the way this library sounds or looks. Sonically, the in-your-face nature makes sense — the CP-70, after all, has a very distinct vibe and was made to be played on stage and be heard amidst all the other amplified instruments accompanying it.

As a result, if you’re looking to create understated, quiet arrangements, you might have to work a little to rein in the sound. Your friends here are the controls for Color (select how hard or soft the sound is) and Tonal Shift (adjust the playback speed of each sample, making the instrument sound sharper or darker, with less spiky transients).

When you audition the preset patches, it’s best if you turn off the audio preview or you could run into feedback issues that are positively terrifying when you have headphones on. Learn from the mistakes of others.

Alicia’s Electric Keys is the kind of instrument you would add to your collection if you already had the basics and were looking for more character-filled piano options. Or maybe you’re ready to go full retro and bring back the sounds of the ’70s and ’80s, when the

CP-70 reigned supreme. In either case, this instrument is a good place to start.

Sonal D’Silva $149 www.native-instruments.com

MNTRA Pyramids

Plug-in Instrument

HHHH MNTRA Pyramids runs on the free MNDALA 2 Player supporting all the common plug-in formats. The premise is a touch New Age-ish as the sounds come from crystalline quartz instruments fashioned in the outline shape of pyramids, rather like three-dimensional orchestral triangles. Their ratios were inspired by those of the Great Pyramids of Egypt.

How much that affects the sound I don’t know but to my ears the resulting structures sound somewhat like crystal bells when struck or like a crystal bowl when bowed. The samples used in this instrument total 5.2GB and include round robins to prevent repeated strikes sounding identical. All manner of tricks were used when creating the samples, including spinning the pyramids in front of the microphones. The samples were recorded at a sample rate of 384kHz at a 32-bit resolution.

While few of the sounds are obvious fits for pop music, there’s a lot here that works well in a more cinematic context, with obvious applications in relaxation and ambient music. MNDALA 2 includes sequencers, envelope generators and onboard effects to push the sampled sounds beyond their natural limits, allowing for the creation of shifting drones and pads as well as crystalline bell sounds.

Some of the sample sets are quite large so take a while to load. The presets are categorised as Atelier (a posh word for studio or workshop), Elements and Steffen Ki. Each of these has further sub‑categories. One of the hallmarks of the MNDALA 2 engine is that it offers three MIDI control of parameters designated X, Y and Z. These are set to control different aspects of the sound depending on which preset is chosen.

Sometimes the designations are a little vague — Bio Resonance, Plasma Charge and Inertia (relating to the Negentropics preset) — but in reality they tend to

control various effects parameters, with one of the functions being mapped to the mod wheel. There’s a Learn function to assign any convenient CCs to the three modulators, and these parameters can also be adjusted and therefore automated within the DAW by dragging on the graphics (an equals sign, a triangle and a round puck) in MNDALA 2’s Perform view.

Sonically there’s more variety than you might expect, from tuneful hits and harmonically rich bows and drones to jangly dissonance and evolving atmospheric textures. There are shifting bell swarms and deep resonances as well as distant tinklings that make perfect ear candy to drop into the background of ambient music. Some of the sounds exhibit very strong overtones, but that characteristic is an intrinsic part of the character of bell-like structures. If you make cinematic music you will almost certainly like Pyramids, but if your focus is more on relaxation and ambient music, you’ll absolutely love it. Paul White $49 www.mntra.io

In Session Audio Deep Pool

Kontakt Instrument

HHHHH In Session Audio don’t really do ‘me too!’ products, so there is always something intriguing to uncover. Their latest release — Deep Pool — comes with a tag line of Modern Downtempo Drums.

It’s a perfect description of what this Kontakt-based instrument is about. Built from just over 4GB of sample content, Deep Pool blends detailed sampling of acoustic drums, electronic drums and some sound‑designed elements to create a suite of very stylish drum and percussion sounds.

The underlying sounds are presented to the user within a well thought‑out performance engine that provides some 20 suites of drum, percussion and additional one-shot sounds each with a full complement of performance loops suitable for building full song-style

arrangements. These loops are mapped across the MIDI keyboard into multiple key ranges (these are neatly colour-coded within Kontakt’s UI), providing flexible trigger options so you can trigger the full loops, or mix and match between the low‑, mid‑ or high‑frequency elements of each loop. A further colour-coded keyboard range provides one-shot samples for additional performance customisation.

The loop content was originally designed for the 60 to 84 bpm range but will, of course, follow your host’s tempo. Additional sound‑design options expand the possibilities much further.

These operate at a number of levels.

For example, within the upper half of the UI, you can customise properties such as volume, pan, EQ and tuning for the currently selected individual loop or one-shot. The lower half of the UI offers three further panels with Global, Playable and Send processing options.

The first of these includes adjusting the velocity-based volume and filter response of the instrument, letting you coax a wide range of performance dynamics from the sounds. The Playable effects — which include Panner, Filter, Crush and Lo-Fi, among other options — provide spot effects that you can trigger in real time via a further mapped MIDI key range.

The Send options include delay and two different reverbs. These various sound‑design options combine beautifully and are easy to use, adding plenty of creative possibilities.

Songwriters or producers working in downtempo, chill or lo-fi styles will undoubtedly enjoy what’s on offer here, but I could easily see media composers making full use of the creative sound‑design tools — including the EQ and filters — to generate an underlying rhythmic pulse in a drama and tension cue. Yes, Deep Pool may be something of a niche product, but it brings a touch of sonic class to whatever project you drop it into. Downtempo fans should be encouraged to dip a toe into Deep Pool; well worth auditioning. John Walden $149 www.insessionaudio.com Audio examples of this month’s libraries are available at www.soundonsound.com.

www.soundonsound.com / December 2024

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Q

Why does my DI box have a third output?

I was given an LA Audio DI2 DI box, and as well as the link output (which I usually use to send the signal on to a guitar amp, for example) and regular XLR balanced output, it has an unbalanced TS jack output. I’ve not seen this kind of feature before. The manual says it’s “an unbalanced output suitable for connecting to unbalanced equipment inputs”. So, I guess the input and link are connected in parallel, but that third unbalanced jack seems a bit odd to me.

What would it be used for?

Also, something else in the manual that really intrigued me was a section headed Speaker Feed. It said: “Plug the output of the instrument amplifier into the input. Now the instrument feed from the balanced or unbalanced outputs of the DI2 includes all EQ, effects and amplifier distortion.

The pad switch should be in the pad position for this application.” I assume one has to use speaker gauge cable, not a regular instrument lead for amp out/speaker jacks — is that right?

SOS Forum Post Hugh Robjohns, Technical Editor The separate unbalanced output is not that unusual on upmarket active DI boxes, and it can be very useful.

It’s a second buffered output, usually with its own active buffer circuitry but sometimes derived from a second output winding on an output transformer.

Applications include feeding a local monitoring mixer, a guitar tuner pedal, or anything else you like, all while being electronically isolated from the source.

That means that whatever you connect won’t load the guitar’s pickups and change its tone in any way, and neither will it affect the balanced output.

Q

What exactly is half‑normalling?

I think I need a patchbay for my outboard but don’t really understand what normalling and half-normalling are — how do I know what kind of setup I need?

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134

Many DI boxes can also be used to derive a signal from a backline power amplifier, such as in a guitar amp. Sometimes this is a preferred technique because it captures any effects and tones created within the amp itself. But as the manual implies, this approach requires care because there are good and bad ways of doing it! Obviously, the signal level from a power amp is going to be very high, so at the very least you’ll need to use the pad option on the DI box’s input. Most DI boxes employ a 40dB pad for this purpose, and some have a dedicated speaker input socket, separate to the normal instrument input socket, specifically for working in this way.

When connecting a guitar (or other instrument) to a DI box, the usual approach is to plug the guitar into the input socket, and then use the link output to pass the signal directly on to the backline amp. Since the input and link sockets are wired together in parallel, the guitar is effectively connected directly to the amp, and the DI box is just ‘monitoring’ the signal on that connection as it passes through. Its very high input impedance means it doesn’t load the guitar at all, so it doesn’t affect the tone. It’s a convenient and practical way of working.

In theory, you could work the same way to hook up the signal passing between the power amp and speaker.

But this is a high-voltage signal with hefty currents and lots of power involved! To handle a power amp’s output safely you need chunky cables and thick circuit board tracks and, in reality, very few DI boxes are designed to tolerate such power — it’s really not a good idea to connect a DI box between amp and speaker using the input and link sockets unless the manufacturer specifically recommends it!

Matt Houghton, Reviews Editor Normalling just means there’s a ‘normal’ signal path when no cables are plugged into the front of the patchbay. When you plug cables in the front, you either interrupt (break) that connection or take a copy of it to be routed elsewhere.

December 2024 / www.soundonsound.com

If your DI box has a third output, you can use it to feed a monitor mixer, a guitar tuner or other device separately from the FOH mixer and guitar amp.

A much better, safer option is to connect the amp directly to the speaker as usual, and then access the speaker signal via either a second output socket on the amp or a second input/link socket on the speaker. For this connection, a standard instrument cable can be used to run that signal over to the appropriate DI box input (with pad selected!).

When connected in this way, there is no power flowing through the DI box, so there’s no need for chunky speaker cables. Instead, the DI box is just monitoring the (high) voltage signal at the amp or speaker. Again, the very high input impedance of the DI box ensures that there is no significant current flow, and no loading effect on the amp or speaker.

Just remember that you will need to engage the input pad switch as the signal will be very much bigger than a typical guitar signal!

Most budget jack patchbays default to a half-normalled setup: plugging into the top row copies the signal, whereas in a (fully) normalled setup it would break it. In either setup, plugging into the bottom row breaks the normal connection.

You can also configure a ‘thru’ connection, with no

normalling — the front and rear connectors are directly linked and go nowhere else until you patch cables in. For an accessible ‘deeper dive’ that should tell you what you need to know for your setup, check out Hugh Robjohns’ article (https://sosm.ag/ PatchbaysInTheModernStudio) in SOS December 2020.

Audeze Maxwell It’s marketed as a gaming headset, but with head‑tracking and Dolby Atmos support, the Audeze Maxwell could be even better suited to the studio.

Read our review in the next issue of Sound On Sound.

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137

EFFECTS PEDALS PAUL WHITE

Y

ou might wonder why I’m still drawn toward effects pedals when plug-ins can often do so much more, but in a way that answers the question. The majority of effects pedals have a manageable number of controls that enable the user to explore the full range of their capabilities in a short space of time. By contrast, some plug-in effects take you way down the rabbit hole of adjusting obscure parameters.

There are some pedals that take a little more exploration, especially some of the granular delay offerings out there, but in the main they do what they say on the tin.

Pedals tend to be free from any perceptible latency, which can be a big deal when it comes to playing feel, and

all you need to connect them to each other is a supply of short jack patch cables. In the studio a pedal power supply that can run multiple pedals is a good option, and one with fully isolated power outlets will ensure no additional noise or hum creeps in. Fully isolated outputs also allow you to connect those pedals that have non-standard, centre-pin‑positive connections simply by using a polarity inverting power cable. In my own studio I have a low table set up with a multi-outlet PSU installed and I have covered the tabletop with non-slip mat so that I can hook up different arrangements of pedals without the risk of them sliding around. This is a good system for feeding into my DAW as my setups don’t usually require me to

“In some ways, stringing together a load of pedals is the guitarist’s equivalent of a modular synthesizer...”

138

December 2024 / www.soundonsound.com

stomp on switches during recording, and if I need a volume pedal or expression pedal, I can still put that on the floor. In some ways, stringing together a load of pedals is the guitarist’s equivalent of a modular synthesizer, though that doesn’t exclude users of other electronic instruments from using them too!

Another reason to love pedals is that you don’t have to authorise them to get them to work and they won’t stop working when falling foul of an incompatible operating system upgrade.

What’s more, a pedal often costs less than a plug-in that tries to emulate it, and while you can run multiple instances of a plug-in, I tend to use pedals while recording rather than for mixing so I don’t need multiple instances. And, should you fall out of love with a pedal, you’ll find that they are easier to sell than plug-ins too, even where the plug-in has an owner transfer policy, which many don’t.

For those of us who still play live (I’ve suffered for my music so I make sure everybody else does too!), pedals provide a practical means to take effects on stage without the need for a computer to run plug-ins, though that is of course an option if you like to work that way. There’s a great deal of satisfaction to be had from setting up a pedalboard to meet specific live needs, and in my case I have three of them: one for the covers band I play with, one for duo gigs and yet another for solo ambient gigs where I need things like loopers, granular delays, long reverbs, envelope shapers and so on. You can of course buy amp sim/effects units that do everything in one box, but there’s something reassuring about being able to just reach out to a pedal and turn a knob if something needs adjusting mid performance.

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