Issuu

1985 — 2024

THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

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

TASCAM MODEL 2400 Mixer, recorder, interface and more

2024’s best studio kit

Kii Seven

“An experience genuinely unlike any other!”

‘A Bar Song’ How Shaboozey’s mega-hit was made

WIN

AUDIX MICS

WORTH $1 800 www.soundonsound.com

ON TEST: WES AUDIO / IK / LUNATIC AUDIO / NI / FIEDLER AUDIO / SENNHEISER / ACS / GEFELL / DW / SOYUZ / ARTURIA / PRESONUS

TECHNIQUE: MIXING MUSICALLY / STEM SPLITTING IN LOGIC

USA $11.99 / Canada C$12.75

124 INSIDE TRACK

IN THIS ISSUE

www.soundonsound.com

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.

WIN

AUDIX

MICS WORTH $1800

112 Spotlight:

Desktop Monitor Controllers

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

50 133

Arturia KeyLab MkIII

Distortion & Effects Plug-in

Boss IR-2

Boredbrain Xcelon

Modular Microphone System

Eurorack Module

DPW Design P1 Pedal I/O

Dreamtonics Vocoflex Voice Transformer Plug-in

64 Drum Workshop DW

133

132 Native Instruments Alicia’s

Electric Keys Sample Library

Virtual Drum Instrument

Console & gravitas MDS Dolby Atmos Mastering Software & Processor

MNTRA Pyramids

Soyuz 013 Ambisonic Ambisonic Microphone

Tascam Model 2400 24-Track Recording Console

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

Warm Audio Pedal76 Compressor Pedal

Waves Curves Equator Resonance Suppressor Plug-in

Wes Audio ngTube EQ Dual-channel Equaliser

Sample Library

Soundworks 32 Fiedler Audio Mastering

ModeAudio Airspace Reverb Plug-in

Eurorack Module

Microtech Gefell Generation 4

Sennheiser MD 421 Kompakt Dynamic Microphone

Lunatic Audio Rocket Fuel Creative Effects Plug-in

Amp & Cab Emulation Pedal

Kii Seven Active Monitors

Controller Keyboard

Audio Cable System

JMC Cyberdrive

PWM Mantis Analogue/Digital Synthesizer

Audio Puck

In-ear Monitors

In Session Audio Deep Pool

54 Interspace Industries

PreSonus Studio One 7 DAW Software

Sample Library

ADDAC System ADDAC310 ACS Custom Emotion Dynamic

Mixing & Mastering Plug-ins

Eurorack Module

IK Multimedia T-RackS 6

C O V E R

ON TEST

Native Instruments Kontakt 8 Software Sampler

Neural DSP Parallax X Bass Processing Plug-in

WORKSHOPS 104 106 108 110

Pro Tools Live Cubase Logic

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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

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

December 2024 / www.soundonsound.com

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

m908. Immersive Monitoring, Perfected. The m908 monitor controller gets you working in any format quickly and easily, expertly managing any speaker system from stereo to Dolby™ Atmos 9.1.6. Firmware 2.0 includes our new web-based control platform which lets you operate and configure the entire system from any web browser (desktop or mobile). Additionally, the m908’s room correction EQ capability provides 12 bands on all 24 channels at all sample rates. With unrivaled audio performance and mechanical elegance, the m908 is simply the finest all-in-one monitoring tool for modern music production.

“Grace is renown for incredible sonics, however the m908 is an entirely different machine. Not only does it sound phenomenal, but the functionality and routing is on another level. As the centerpiece of our Dolby Atmos room, it handles hundreds of I/O, multiple speaker sets, room correction, and so much more. I can’t say enough great things about it! ”

www.gracedesign.com

Glenn A Tabor III Multiple Grammy™ winning producer / engineer, Gat3 Studio (www.gat3.com)

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

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

December 2024 / www.soundonsound.com

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

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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

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.

December 2024 / www.soundonsound.com

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

www.soundonsound.com / December 2024

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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

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

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

December 2024 / www.soundonsound.com

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

The revolution continues. Revolution 14x16: The truly high-end recording interface with 2 Auteur preamps with Cinemag® transformers

The Revolution 14x16 is a class-compliant, 14-in / 16-out USB-C audio interface. Its built-in Cinemag transformer-driven Auteur preamps let you add weight and color to your tracks, and the headphone mixer lets you tailor a headphone mix from both hardware inputs and software playback. When your needs grow, Revolution 14x16 grows with you — add eight channels of input and output over ADAT; add external hardware processors via 4 switchable send/ returns; and sync your hardware devices via our patented Macro-MMC clock; treating your music to world-class clocking and low-noise conversion. Visit www.blacklionaudio.com to learn how to Revolutionize your studio.

Expand your I/O with the Revolution EXP: 8x8 AD/DA Converter

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JOHN WALDEN

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

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

December 2024 / www.soundonsound.com

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

Broadcast it ...

... your way Craft your sound with Stealth Broadcast, the award-winning microphone that covers all the angles. With its four mic voicings and a built-in Class A preamp, Stealth combines outstanding versatility with world-class performance, for all professional broadcast and studio recording applications. The bundle includes the fully-articulating Stealth Side Mount, for complete control over positioning, while the mic’s superb off-axis rejection ensures clean and accurate focus on the targeted sound source. Read more on Stealth Broadcast and Stealth Side Mount (also available as an accessory for existing Stealth owners) and find your nearest retailer at astonmics.com

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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.

December 2024 / www.soundonsound.com

DAV E LO C K WO O D

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

_KeyLab mk3 Universal MIDI controller Whether sculpting a studio masterpiece or channeling the raw energy of a live performance, KeyLab mk3 is an advanced MIDI controller that will inspire your very best composition. From its design, to its pro-grade aftertouch keybed and seamless software control KeyLab mk3 leaves no stone unturned.

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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

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,

December 2024 / www.soundonsound.com

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.

Introducing the MKII

Elevated Performance Fuel Your Creativity The original SSL 2 and SSL 2+ interfaces changed the game, delivering a level of performance never seen before in its class. The SSL 2 MKII and SSL 2+ MKII build upon the success of its predecessor, with a series of significant enhancements and feature additions. Designed for creatives who demand the highest standards in sound quality, the MKII interfaces provide unmatched audio performance and user-friendly design that caters for both those starting their journey and seasoned professionals. Class-leading, low-noise microphone preamps | Next-Gen 32-bit 192 kHz converters | Legacy 4K Analog Enhancement | 2 x Auto-detect instrument (DI) inputs | 2 x High-Current Headphone Outs (independent on SSL 2+ MKII) Zero-latency monitoring | MIDI I/O (SSL 2+ MKII only) | Stereo Loopback | Bundled with SSL Production Pack

2 IN/2 OUT USB AUDIO INTERFACE

#WhatWillYouCreate www.solidstatelogic.com

2 IN/4 OUT USB AUDIO INTERFACE

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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

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

December 2024 / www.soundonsound.com

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. W www.warmaudio.com found a new home.

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www.soundonsound.com / December 2024

31/05/2024 11:04

ON TE ST

PWM Mantis

Analogue/Digital Synthesizer

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

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

ON TE ST

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

www.soundonsound.com / December 2024

ON TE ST

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

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

December 2024 / www.soundonsound.com

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.

www.soundonsound.com / December 2024

ON TE ST

TA S C A M MODE L 24 0 0

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

December 2024 / www.soundonsound.com

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. W https://tascam.com/us www.soundonsound.com / December 2024

ON TE ST

The T-RackS 6 Mastering Console.

JOHN WALDEN

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

December 2024 / www.soundonsound.com

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.

www.soundonsound.com / December 2024

ON TE ST

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

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.

December 2024 / www.soundonsound.com

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

ON TE ST

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

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

December 2024 / www.soundonsound.com

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.

www.soundonsound.com / December 2024

ON TE ST

Drum Workshop DW Soundworks Drum Workshop bring their

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

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

December 2024 / www.soundonsound.com

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.

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

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.

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

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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Kii Seven Active Monitors

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

PHIL WARD

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

December 2024 / www.soundonsound.com

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

THE FUTURE OF GLOBAL RF

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

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.

December 2024 / www.soundonsound.com

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

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

ON TE ST

Boredbrain Xcelon Eurorack Module

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

December 2024 / www.soundonsound.com

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

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

www.boredbrainmusic.com

$399

www.addacsystem.com

DPW Design P1 Pedal I/O Eurorack Module

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

TECHNIQUE

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

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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TECHNIQUE

Pro Tools

We get reacquainted with some of Pro Tools’ stock plug-ins. JULIAN RODGERS

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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Logic Logic

TECHNIQUE

We revisit some old tape masters with the help of Logic’s new Stem Splitter. PAUL WHITE

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

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SPOTLIGHT

SPL MTC Mk2

Desktop Monitor Controllers In the age of mixer-less studios, monitor controllers are on the rise! LUKE WOOD

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

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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FE ATURE

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

w w w. s o u n d o n s o u n d . c o m / December 2024

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INSIDE TRACK

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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FE ATURE

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

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.”

Hear What You've Been Missing

The Nuance Select is a monitor controller designed with the belief that nothing should stand in the way of the sound that reaches your speakers. With exceptional sound quality, it sets new standards for audio reproduction, letting you hear the true, uncolored sound of your monitors with astonishing clarity. Reveal subtle details that might previously have been obscured, and enjoy a more authentic, immersive stereo field. By giving you the full picture, the Nuance Select empowers you to take control and make every mix decision with confidence.

Nuance select

Studio Monitor Controller

Hear the True Sound of your Monitors

www.radialeng.com

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.

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? SOS Facebook message

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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

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. And they come in such pretty colours!

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SOUND ON SOUND [USA] December 2024 Issue