SOUND ON SOUND [UK] September 2024

SOUND ON SOUND THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE

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

MOOG MUSE E X C L U S I V E

SEPTEMBER 2024

The inspirational polysynth

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Zoom H4e Reinventing the field recorder

VOLUME 39 • ISSUE 11

How sampled instruments work

Ken Scott & Emre Ramazanoglu

ELECTRO-VOICE

Session discipline

WO R T H £ 1 0 5 5

Ziggy Stardust then and now

Lessons from tape for a digital world

EVERSE 12 PA www.soundonsound.com

ON TEST: ICON / MCDSP / KORG / DIRTYWAVE / SOUNDTHEORY / SPL / TEKNOSIGN / BEEZNEES / TOFT / VSL / WALDORF

TECHNIQUE: COMPOSING FOR COMMERCIALS / LOGIC’S SESSION PLAYERS

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LE ADER

MAGICAL THINKING Arthur C Clarke famously opined that “Any sufficiently advanced technology is indistinguishable from magic.” In my experience, though, the wow factor soon disappears, and we start to take incredible technological achievements for granted. When Clarke wrote his words in 1962, video calls were the stuff of science fiction. Today we FaceTime each other without a second thought.

If we ever feel anything towards the technology, it’s not so much wonderment as frustration.

In the studio, meanwhile, it’s hard to recall the heady days of the early ’80s when sampling was a novel technology, and a Fairlight orchestral stab briefly sounded like something from another planet. Magic is terrifying as well as dazzling, and the awe that early samplers inspired was freighted with fear. Would this technology make real musicians and studios redundant? We soon realised that the answer was ‘no’, and learned to complain about lack of memory instead of sitting there open-mouthed when we heard an aerosol spray used as a hi-hat.

I suspect that the same process of disillusionment is going on with machine learning right now. What initially seemed miraculous is beginning to feel commonplace. When I used AI to transcribe the interview with Ken Scott that you can read in this issue, my reaction was not to gasp “How is this even possible?”, but to wonder

ALLIA BUSINESS CENTRE KING’S HEDGES ROAD CAMBRIDGE CB4 2HY T +44 (0)1223 851658 sos@soundonsound.com www.soundonsound.com

how it could possibly have misheard his first job at Trident Studios as mixing ‘Give Pizza Chance’.

But I think there is a value in reminding ourselves just how remarkable the tools available to us are. A studio local to me hosts birthday parties for young children. Without exception, the thing that most fascinates them is talkback. The fact that a person can press a button in the control room to talk to another person in the live room is still magic to the six-year-old mind.

So when we get frustrated because the high C in a violin patch is slightly out of tune, or the legato bassoon doesn’t speak in quite the way we wanted, it’s perhaps worth stepping back and thinking exactly how extraordinary it is that a generic laptop can produce these sounds at all.

Sampling hasn’t put real musicians out of a job, as many once feared, but it has certainly made it hard to tell what’s sequenced and what’s live.

Clarke’s idea is sometimes understood to mean that technology appears magical only inasmuch as we can’t understand it, but I don’t think that has to be true. Anyone who reads Chris Korff’s eloquent explanation in this issue will understand very clearly how modern sampled instruments work — but won’t there always be something magical about pressing a key and hearing an entire symphony orchestra bursting from our speakers?

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TECHNOLOGY

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52 THE TAPE MINDSET

IN THIS ISSUE

www.soundonsound.com

September 2024 / issue 11 / volume 39

FEATURES 32 The World Of

Commercial Composition

WIN

ELECTRO-VOICE

EVERSE 12

Composing for adverts and music libraries can be rewarding, challenging and creative — and did we mention that you get paid?

42 How I Got That Sound David Lord explains how he combined a Mellotron with a Radio Shack amp and a metal bin on XTC’s ‘All You Pretty Girls’.

WORTH £1050

52 The Tape Mindset: An Analogue

Approach To Our Digital World

Working with tape imposed a discipline on recording sessions that was hugely beneficial — and which our DAWs have made optional.

PAGE 86

62 Ziggy Stardust In Atmos Ken Scott and Emre Ramazanoglu explain how they reinvented David Bowie’s classic album in three dimensions.

104 Inside Track: Julian Bunetta

80 How Virtual Instruments Work

Taking the decision to follow his instincts has elevated Julian Bunetta to the top rank of songwriters and producers.

We cut through the jargon to explain how software instruments achieve their incredible realism.

120 Spotlight: Kick Drum Microphones

88 Talkback: Ramera Abraham

We’re spoilt for choice when it comes to dedicated kick drum mics. We check out some of the best.

Producer and engineer Ramera Abraham on dissecting Beyoncé and why she’d love to sit in on an Ariana Grande session.

144 Q&A

92 Modular

Your studio and recording questions answered.

Cycle Instruments’ Josh Wilkinson talks about modules and reinventing the sequencer.

146 Why I Love... Minimal Synth Setups Dave Gale explains why he craves the simple life.

116 ORCHESTRAL TOOLS BENJAMIN WALLFISCH STRINGS

ON TEST 10

74 70

AIXDSP DrumEQ

125

BeesNeez Tribute 1 Cycle Instruments Tetrachords Eurorack Module

Dirtywave M8 Model:02 Portable Hardware Tracker

127

Drawmer 1972

Sample Library

108

128

Sample Library

24

McDSP APB Tape Moog Muse

Superior Drummer 3 Expansion

126

Channel Strip Plug-in

96

Sampleson Drumbada Sample Library

94

Sonic Charge Synplant 2 Software Synthesizer

129

Soniccouture Pan Drums II Sample Library

18

Soundtheory Kraftur Multiband Saturation Plug-in

72

SPL Phonitor 3 Headphone Amplifier

VSL MIR Pro 3D

& Vienna Power House Immersive Audio Reverb Suite

26

Waldorf Iridium Core Synthesizer

36

Zoom H4 essential Portable Recorder

Sample Library

128

Universal Audio LA-6176

Orchestral Tools

& Cali76 Bass

Toontrack

The Jazz Sessions SDX

Malcolm Toft Equate

124 Origin Effects Cali76 FET

Impact Soundworks Rosette Mandolin

Audio Processors

Sample Libraries

IK Multimedia Tonex One Amp Emulation Pedal

Make Believe Studios MixHead

Benjamin Wallfisch Strings

Icon Pro Audio/Harrison 32Ci USB Audio Interface

126

116

Teknosign Box Line Double Width

Polyphonic Synthesizer

Grace Design Rex & Roxi Microphone Preamps

58

44

Glasgow Synth Guild Oct Tone Eurorack Module

28

Eight-channel Equaliser

Emergence Audio Infinite Upright

91

Time-alignment Plug-in

Tape Emulation Plug-in For APB

Dual-channel Microphone Preamplifier

129

16 8

C OV E R

110

Lauten LS-408 & LS-508

Synchro Arts VocAlign Pro 6

Drum Microphones

Tape Emulation Plug-in

Valve Microphone

90

12

Drum, Percussion & Sampler Pad

Equaliser & Dynamics Plug-in

22

Korg MPS-10

20

Zynaptic Morph3 Pro Audio Morphing Plug-in

WORKSHOPS 130 134 138 140 142

Digital Performer Logic Pro Tools Cubase Studio One

ON TE ST

McDSP APB Tape

McDSP’s analogue plug-in platform can now replicate the complex dynamic behaviour of tape.

SAM INGLIS

M

cDSP’s Analog Processing Box is perhaps the ultimate hybrid device. It processes audio in the analogue domain, but its processing is presented within your DAW in exactly the same way as conventional digital plug-ins are. The upshot is that you get true analogue processing with full recall, automation and the other ergonomic pluses of software.

Most of the APB’s capabilities centre on dynamics processing, and the initial suite of plug-ins included several compressors and limiters. Since launch McDSP have added such treats as a valve-style mixer, a multiband compressor, a mastering EQ and a dedicated saturation processor, all of which are free to APB owners. The latest addition to the range is a tape emulator.

APB Tape has a straightforward but comprehensive parameter set. The amount of saturation or dynamic processing is adjusted using input and output gain controls, plus a dial labelled Comp, which seems to act as a ratio setting.

You can switch between two emulated tape formulations, and there are controls for mimicking the low-frequency ‘head bump’ and associated sub-bass roll-off that are characteristic of tape machines.

When used in stereo, you also have the option of operating APB Tape in L‑R or M‑S modes, and there are two well

thought‑out control linking schema that help to make setting these up easier.

There’s also a nice seleciton of presets categorised by instrument or application.

Talking Tape Some of the sonic artefacts of tape recording were undesirable, such as wow and flutter, high-end loss and hiss, but others are remembered more fondly.

Most of these relate to the complex dynamic changes introduced by saturating the medium, and there have been many previous attempts to replicate these in the analogue domain as well as in digital plug-ins. However, if the tape emulation plug-ins in my folder are anything to go by, no-one can quite agree what the good qualities of tape actually were!

Tape Emulation Plug-in For APB

there’s a noticeable bite to the sound, but you have to really cane it before obvious compression occurs. The 456 formulation, by contrast, is a lot more spongy and soft.

Either way, if you get the gain-reduction needle moving, the dynamic effect is very cool. It’s a little reminiscent of parallel compression, and is capable of bringing real excitement to drum-led mixes. Slate Digital’s VTM will do something of the sort when pushed, but APB Tape goes much further. I doubt you’d want to use the most extreme settings across an entire mix, but for something like a drum room mic or DI’ed funk guitar, they could be just the ticket. As you’d expect, the various elements of the processing are interactive, so whereas the ‘head bump’ sounds like a straightforward EQ curve at low gain levels, it responds to the dynamic changes at higher gain levels in quite a complex and interesting way.

I suspect that the sound of tape is a moving target, and I wouldn’t like to say whether APB Tape gets closer to it than other plug-ins. But it certainly stands out from the crowd, with an ability to inject energy into a mix and make everything ‘bounce’ that I haven’t heard in any purely digital tape emulation — or indeed in conventional compressor plug-ins. APB Tape further demonstrates the versatility and sonic potential of McDSP’s unique processing platform.

“If the tape emulation plug-ins in my folder are anything to go by, no-one can quite agree what the good qualities of tape actually were!”

8

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APB Tape adds another colour to the palette. It doesn’t introduce noise, pitch instability or other unwanted side-effects of tape recording, and unless pushed hard, it doesn’t drastically alter the timbre of the source. With the input gain backed off, in fact, it rivals Softube’s Tape in the “is it actually on?” levels of subtlety. As you raise the input level, though, you’ll encounter a wide range of variations on the ‘mix glue’ theme; and the hotter things get, the more the differences between the two emulated tape formulations become apparent. GP9 is, for want of a better word, quite ‘stiff’. As you pile on the signal, the upper midrange fills out nicely and

summary McDSP’s analogue tape emulation plug-in produces exciting and distinctive dynamic effects.

£ Free to APB owners.

W www.sxpro.co.uk W www.mcdsp.com

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ON TE ST

AIXDSP DrumEQ Equaliser & Dynamics Plug-in

Could this clever, labour-saving plug-in change the way you mix drums?

M AT T H O U G H TO N

A

IXDSP have been around for a few years now but when I first tried out their debut plug-ins, although I found them impressive, they didn’t feel quite finished, so I decided to await updates before reviewing them. Their offering has since matured considerably, and they now have a range of thoughtfully conceived EQ, dynamics and reverb plug-ins, including several that are aimed primarily (though not exclusively) at drum processing. These include a multiband gate, a multiband compressor and a dynamic EQ, but while each has some commendable features, by far their most impressive offering to me is DrumEQ, which is the focus of this review and is available separately and, along with Multiband Gate and Intuition Compressor, as part of their Drum Producer Pack.

Overview At heart, there’s some fairly conventional technology behind DrumEQ and, technically, it’s possible to achieve pretty much anything you can with DrumEQ using conventional EQs and dynamics processors. But the way in which these technologies have been combined in this implementation makes DrumEQ a very different proposition to any other plug-in I’ve used.

In addition to separate sweepable, fixed-slope high‑ and low-pass filters, there are 16 bands of fully parametric EQ here, each with gain, Q and frequency controls. But unlike in most EQs, these are organised in complementary pairs.

You can see one pair at a time in the main part of the GUI, along with all the controls for both bands. The band on the left is described as an Octave Filter and the other, on the right, as a Resonant

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September 2024 / www.soundonsound.com

Note Filter. Each band also has a dynamics section, with Threshold, Ratio, Attack, Hold and Release controls, and this can go from expansion to, effectively, limiting.

You get an overview of all the bands in a spectrum analyser in the upper pane, with Octave Filters represented as vertical blue lines, and the Resonant Note Filters as green ones.

Here, you see the EQ curve along with the input spectrum, and can toggle the output spectrum on/off as an overlay so you can quickly see what you’re doing to the signal in real time. Click on any octave in the spectrum analyser, and that octave’s filter pair is selected in the lower pane.

On the left of the GUI are three global control knobs: Tune Frequency, Master Transposition and View Filters.

Turning Tune Frequency shifts the frequency of all the parametric bands simultaneously (ie. everything but the high‑/low‑pass filters), while turning Master Transposition shifts all of the Resonant Note Filters, effectively introducing an offset between these and the Octave Filters. Meanwhile, View Filters cycles the GUI view through the different filter pairs.

There are also buttons to engage and solo each of the two visible EQ bands and to engage/bypass the dynamics processing, as well as the ability to load factory presets, or save/load your own.

Hit Maker?

If this all sounds conceptually simple, well it is... but that’s kinda the point here! Drums can be incredibly complex instruments, with multiple resonances and overtones contributing to each drum’s unique character. Processing them to your satisfaction can take time, and it’s easy to take the processing too far — such that it’s often as easy to zero everything and start again as it is to trace your steps and fix what you got wrong. With DrumEQ, it’s super easy to dial in just as much snap or girth from a drum as you want, while keeping nasty side‑effects at bay.

The manual suggests what I agree is the best way to work: listen to your drum to hear where there’s a resonant peak that’s contributing something useful

or unwanted, and loop that section.

Playing the loop, turn the global Tune Frequency knob until you sit a blue line (ie. an Octave Filter) on that resonance, and then start twisting the controls in that octave, pulling the resonance and its harmonics up or down as you please to emphasise the character. Then bring the Resonant Note filters into play to shape the resulting sound, perhaps cutting ugly resonances after a boost. Finally, you can fine tune the individual filters if required, and if you want to, bring the dynamics section into play.

In practice it’s an extremely intuitive and effective tool for tailoring your close drum mics. I’m normally one for using ears before eyes, but using the spectrum analyser for the initial filter placement works very well — you can get a long way very quickly, before fine-tuning by ear. My only note of caution is that those global filter positioning controls override anything you’ve set up differently in the individual bands, so you really do have to position those first, and if you make changes to the frequency of any individual bands, you might want to save a ‘safety preset’ for ease of recall.

The bottom line is that DrumEQ makes dialling in a huge and boomy or tight and snappy sound from your kick, snare and toms super easy and super fast. A great time-saver that will probably lead you to more satisfying results!

summary A novel and bloomin’ useful take on traditional EQ and dynamics processing, DrumEQ really could change the way you approach mixing multi-miked drums.

£ DrumEQ $59.99. Drum Producer Pack $149.99.

W https://aixdsp.com

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ON TE ST

Synchro Arts VocAlign Pro 6 Time-alignment Plug-in Along with various other improvements, version 6 of this labour-saving plug-in brings ARA2 support to Pro Tools users.

LUKE WOOD

O

riginally conceived for the AV post-production world, VocAlign soon became popular in music production because its ability to automatically align vocal and instrument takes could save hours of manual editing. In recent years, it’s been given more music-friendly features, derived from Synchro Arts’ flagship software Revoice Pro, including the ability to match the tuning of tracks as well as their timing. Sam Inglis’ review of the last major version (SOS February 2021: sosm.ag/synchro-arts-vocalign-ultra) details VocAlign’s core functionality, and rather than repeat that I’ll focus here on the new features.

With the release of version 6, VocAlign Project and Ultra are now known as Standard and Pro, respectively, but the key differences between them remain. The former, the entry-level version, is capable of automatically applying time‑alignment across multiple audio tracks. The latter also offers pitch-matching, along with formant-shifting and transposition effects.

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September 2024 / www.soundonsound.com

Both sport a newly designed ‘dark theme’ user interface, and the ARA2 version of the plug-in is now compatible with the latest version of Avid Pro Tools (2024.6). They’ve also both gained undo and redo buttons, and a convenient Arm Capture All function that makes it possible to capture all a session’s tracks with a single click. New features exclusive to Pro include Process Groups, that allow you to quickly apply identical settings to groups of tracks; SmartPitch, a tool that defines how the plug-in handles unison and non-unison parts; and some improvements to the accuracy of the Sync Points.

Sync’ing Feeling Shortly after VocAlign 6 was announced, an artist with whom I work regularly informed me they were recording a song that featured a piano and lead vocal, accompanied by a 21-track backing vocal ensemble. The

ensemble included a selection of doubles of the main vocal, octave-down parts, harmonies and an array of choral ‘ohhs’ and ‘ahhs’, so it was a perfect candidate to test out the plug-in’s new capabilities. And, as a Pro Tools user, I was particularly keen to check out the new ARA2 compatibility, so installed the latest Pro Tools update, loaded the multitracks, and then set about seeing what VocAlign 6 had to offer.

An ARA2 instance of VocAlign can be added to individual clips (using their right-click menu or Pro Tools’ Clip menu) or to an entire track (via the Elastic Audio/ARA menu, the File / Track menu, or the track’s right-click menu). Once loaded, VocAlign appears docked in Pro Tools’ Edit window. It can be opened and closed using a VocAlign 6 tab at the bottom of the screen, and it’s possible to undock it so that it can be freely moved and resized.

The general layout remains as in the previous version, with three lanes displaying the Guide, Dub and Output tracks, and the plug-in’s Match Timing, Match Pitch and Other control panels housed to the right. There are three new icons that allow these panels to be shown or hidden, which reduces visual clutter and saves screen

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ON TE ST

SYNCHRO ARTS VOCALIGN PRO 6

space when you’re not using them. Once happy with the settings, you right-click on your clip(s), select Render from the VocAlign menu, and the edited parts are committed to the timeline.

Thanks to ARA2, audio can be captured from the Pro Tools tracks to the Guide and Dub lanes instantly, and is routed directly back to the appropriate track. Not only does this make it possible to control playback of VocAlign using the DAW transport controls, but it also means that the mixer’s mute and solo controls can be used for detailed auditioning of the corrected parts. This was already possible in other ARA2-compatible DAWs, but for those working in Pro Tools, it’s a welcome addition that makes VocAlign feel much more integrated and intuitive.

Something that’s new for everyone is the Process Groups function, which has made its way over from Revoice Pro. A new option placed above the Capture button in the Dub lane allows multiple tracks to be added to a new group, so they can all be processed as a single operation. I put this to work on a section that had a chorus of nine ‘ohhs’ and ‘ahhs’, made up of three tripled-up parts that all shared timing but with a mixture of different harmonies. I was able to quickly line them all up by defining one part as the Guide and applying the same Match Time settings to all three groups, while using different Match Pitch settings for each. It’s worth keeping an eye on the mode that’s selected here, as well as the current Pitch Target and Target Mode settings. The more parts involved, the more you’ll appreciate it!

Another feature borrowed from Revoice Pro is SmartPitch, accessed in a new drop-down menu in VocAlign Pro’s Match Pitch panel. Three modes govern how the module applies its pitch processing. With the default setting (Match All To Guide), the pitch of all Dub parts will be aligned to that of the Guide. Match Unison Only aligns parts that are close to the Guide but ignores obviously different parts, such as harmonies.

Finally, Match Unison & Tune Non-Unison aligns closely matching parts to the Guide, and applies nearest-note tuning correction to everything else. It’s worth keeping an eye on the mode you’ve selected here, as well as the Pitch Target setting that determines whether parts are tuned to the Guide or Dub; I had a bit of a shock when switching to editing a new part, hitting play and hearing the plug-in desperately trying to pull octave-down parts up to the pitch of the Guide!

The plug-in has been designed to carry out alignment work automatically, of course,

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The ARA2 VocAlign Pro 6 plug-in allows more seamless integration with the latest version of Pro Tools, in which the GUI can be docked (as shown) or floating.

but there are some scenarios where it might not get things 100 percent right first time, so Synchro Arts have also included a couple of tools that allow you to nudge its processing in the right direction. The first, Protected Areas, makes it possible to exclude areas of a track from alignment or tuning. This is ideal for parts that double only some words or phrases, or contain signals that throw off the automatic detection and cause undesired ‘corrections’. The second is Sync Points, which are user-defined alignment points in the Guide and Dub tracks. It’s important to appreciate that, while Sync Points resemble the direct, ‘warp’ style controls many of us have in our DAWs, they are actually used to influence the automatic alignment. So until you grow used to working with them, a bit of experimentation is required if you’re to achieve the desired results — they require a different approach than tools like Elastic Audio or Flex Time.

I have to say that VocAlign’s automatic detection was more than accurate enough for just about everything I loaded in. There were only a couple of instances where I felt the urge to create and move Sync Points, but when that need arose, they worked well. But it was good to know the new undo button was there, should I have needed to extract myself from any mess I created while learning the ropes!

Make Mine Align?

All in all, Synchro Arts have taken an already very capable tool, and added some genuinely useful features that make it even quicker and easier to use. The ARA2 integration will be all that’s needed

to convince many Pro Tools users to invest, but there are plenty of noteworthy changes that should make upgrading worthwhile for others too. In particular, Process Groups and SmartPitch are both significant improvements that will make VocAlign 6 Pro very tempting for anyone working with complex vocal stacks.

There’s also the more expensive Revoice Pro 5 (reviewed in SOS March 2024: sosm.ag/synchro-arts-revoice-pro-5) to consider. VocAlign Pro’s latest features definitely narrow the gap, but that remains the do-everything product in Synchro Arts’ range. The key distinction now lies in the pitch-correction: VocAlign is capable of matching tuning between tracks but provides no control over the tuning itself, other than how closely it’s matched to the Guide; Revoice Pro is equipped with a powerful set of pitch-correction tools, including the ability to adjust pitch modulations such as vibrato in detail. If that sort of thing is less important to you, then VocAlign Pro may now do everything that you need for a smaller outlay.

summary A worthwhile update to the best-in-class time-alignment plug-in, Pro Tool users in particular have reason to cheer now they have support for the ARA2 version.

£ VocAlign Standard £130.80. VocAlign

Pro £282. Prices include VAT. Discounts apply for crossgrades and updates from previous versions.

E sales@synchroarts.com W www.synchroarts.com

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ON TE ST

Malcolm Toft Equate Eight-channel Equaliser

Ditched your console but still have a thirst for analogue EQ? Maybe the Equate can quench it...

NEIL ROGERS

W

ell known for the highly respected Trident consoles of the ’70s and ’80s, Malcolm Toft is still designing gear, most recently releasing a range of niche products under his own name. I really enjoyed his 500-series Punishr analogue distortion (which I reviewed in SOS June 2023: https://sosm.ag/malcolm-toft-punishr) so was keen to check out his latest creation, which sees him return to territory for which he is perhaps best known: equalisation.

Design & Performance The Equate is a convenient solution for engineers who still like to use several channels of analogue EQ but don’t have or want a full console. A 3U rackmount device, it resembles a loaded 500-series rack but this is not a modular system.

There are eight identical channels of EQ, along with an internal power supply and, conveniently, both TRS jack and DB25 D-Sub input and output connections.

Unsurprisingly, the Equate leans on the heritage of the Toft’s Trident 80B EQ design, each channel featuring two sweepable ‘peaking’ midrange bands as well as high‑ and low-frequency shelves.

Unlike the 80B console EQ, the Equate sports sweepable high- and low-pass filters, to give us six bands in total.

I received the Equate during a busy period of tracking and mixing and, after

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a little rummaging around in my cable boxes, patched it in to sit as inserts on the first eight channels of my Audient ASP8024 console. Given Malcolm’s background in large-format mixers, it wasn’t a surprise to find that it seemed completely at home in that setting!

In busy recording sessions, I’m largely looking for an easy‑to‑use EQ that can give me a helping hand in shaping sources on the way in, so that they’ll need less work come mixing time. The two midrange ‘peaking’ style bands on the Equate EQ are superb for this, and the low-mid band was probably the control I found myself using the most, especially to remove ugly low-midrange build-up or resonances.

With its range extending from 100Hz up to 1.5kHz it was also handy for boosting audibility from 800Hz upwards on bass guitars or synths. The upper mid band also covers lots of ground (1-15 kHz) and, generally, I was impressed with how smooth and forgiving it sounded when boosting for presence and clarity. There’s no Q control with this style of EQ, but the bandwidth sharpens/relaxes according to the amount of cut or boost applied, so it’s pretty forgiving.

The low and high shelves of my console are often the most used section for gentle shaping while recording, and the equivalent options on the Equate performed the same role with aplomb.

Centred at 80Hz, the low band can be used to dial in plenty of weight on bass guitars and kick drums, and it soon became

a favourite technique to add a generous amount of heft with the low shelf whilst using the high-pass filter to keep things contained. Centred around 8kHz, the high-shelf was similarly useful: great for easing off the top end of drum room mics or opening up the highs of an acoustic guitar or piano recording.

I’ve touched on how I liked to use the high-pass filter section, but it was great to have a variable low-pass filter here too. It allows you to ‘bracket’ a sound to tuck it into the mix, but rolling off the high end can also make things sound warmer (I think this is one of the things about tape emulation products that makes them so appealing!), or push them ‘back’ in the mix.

Verdict There are plenty of engineers who still like to record with multiple analogue EQs but who don’t necessarily need or have space for a console, and for anyone running such a ‘hybrid’ setup, the Equate could be a great option, perhaps sitting below a rack of 500-series preamps or one of the many multi-channel preamps now available. It might even prompt some who still mix on an analogue console to weigh up the pros and cons of keeping their desk. The bottom line is that the Equate is a great-sounding, all-round tracking EQ, with more than enough flexibility and creative options for most situations — and it’s better value for money than eight channels of 500-series EQ too.

summary The Equate features eight channels of great‑sounding, flexible, console-style EQ in a convenient 500‑series-sized rack.

£ £1699 including VAT. T KMR Audio +44 (0)20 8445 2446 E sales@kmraudio.com W www.malcolmtoft.com

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ON TE ST

Soundtheory Kraftur

Multiband Saturation Plug-in

Want to add power, energy and excitement to a source? Perhaps Kraftur can help...

PAUL WHITE

S

oundtheory’s Kraftur is named for an Icelandic word for strength, and it’s a multiband saturation plug-in with some really nice touches. Used in the right way, multiband saturation — a technique whose potential I really began to discover back in the ’90s when using Drawmer’s Masterflow hardware — allows you to add a sense of power to a sound in a really focused way. For instance, you can put more energy and edge into the highs without increasing peak levels.

Similarly, you can often add weight to the lows, without clouding the all-important midrange, or keep the lows clean while ‘inflating’ the mids. There’s another benefit to this multiband approach too, relative to conventional full-band saturation: while there’s obviously plenty of harmonic distortion going on, the process causes less intermodulation distortion, which means it often delivers more ‘natural’ or ‘effortless’ results.

Kraftur is authorised using an iLok account, and supports all the common plug-in formats for macOS and Windows hosts, including AAX. It can run at any sample rate up to 384kHz and oversampling (between 392 and 768 kHz) is used internally to minimise aliasing artefacts. A switchable Match function keeps the output at the same level as the input to make evaluating the effect easier.

Orientation Kraftur adds to the multiband saturation concept in a couple of useful ways. First, it offers control over the intensity of saturation in each band.

Second, rather than a conventional wet/dry mix control, it provides a very practical, intuitive triangular blend pad, that allows the user to control the contribution of three different audio paths: the dry sound, full-band saturation and multiband saturation.

For the multiband path, Kraftur splits the signal into three separate bands, with user-adjustable crossover frequencies. You have control over each band’s individual distortion characteristics. To the left of the screen are controls that affect the saturation, and Drive determines the signal level that’s fed into both the single-band and multiband processors. While the sound is generally pleasing, I did find that if Drive was set too high it could introduce unwanted distortion, so you do need to listen carefully as you set that.

The shape of the transfer curve is affected by both Offset and Knee controls. The Knee does what you’d usually expect in a dynamics processor, while offset allows you to create a lower‑reaching, gentler curve before the knee, for a smooth transition. The position of the curves for each band (low/mid/high) is controlled by three Shift sliders that can either be operated independently or, if you prefer, as one,

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courtesy of a link control. Bands can also be soloed to help you fine-tune things when setting up each one, though obviously the result in the context of the recombined sound is all-important. Finally, a soft‑clipping stage, with adjustable headroom, comes after the output gain control, the idea being that it can be engaged at the output to catch any ‘overs’ that result when the bands’ signals are recombined.

The level meters display both RMS and peak levels as dBFS (decibels relative to digital ‘full scale’). Peak shows whichever channel is highest and a peak hold line shows the peak values for two seconds, while RMS levels are averaged across all channels. That leaves the large central display, which shows both the split and full-band curves along with histograms showing peak levels. It makes excellent use of colour for clear visual feedback, with the colours becoming more saturated to reflect what’s happening to the audio. There’s also a choice of several alternative colour schemes if you don’t like the default.

Sat Tests Some might find it unusual that Kraftur doesn’t come with any presets, but it’s not an oversight — the results depend so much on both the level and dynamics of the incoming signal, along with its spectral content, that presets would be of little use.

But even beginners shouldn’t let that put them off: Kraftur is really easy to operate.

What matters most, of course, is the sound. I have to say that Kraftur scores well

in this department generally.

For processing overall mixes or submixes, I usually found that subtle settings worked best — it tended to be a case of using little or no processing on the mids, and then adding targeted enhancement to the highs and lows, depending on what I wanted to achieve.

Things like drum busses and rhythmic loops can be treated more aggressively, though, to add punch or attitude, and fairly assertive settings can also help to make bass sounds seem more dense or more present in a busy mix.

In fact, anywhere you might be tempted to use a conventional saturation plug-in, you can try Kraftur, and you’ll probably find that it affords you much more control over the results.

There’s a free trial period so give it a shot.

summary A slick take on the concept of multiband saturation, Kraftur seems able to inject a sense of power or energy into almost any source, and is really easy to use.

£ £79 (discounted to £56.46

when going to press).

Prices include VAT. W www.soundtheory.com

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ON TE ST

Zynaptiq Morph 3 Pro Audio Morphing Plug-in Ever wanted to blend the characteristics of two different sounds? Morph Pro makes that easy.

M AT T H O U G H TO N

A

vailable for the usual plug-in formats on Mac and Windows, Morph Pro takes two audio signals and computes a third signal, called the Morph, based on their characteristics. Add in some simple controls to manipulate the result, a few onboard effects and a mixer to blend the various signals, and what you have is a clever, powerful and unique sound-design tool.

Morphology The resizable (50-200%) GUI has input source controls on the left, the mixer and effects on the right and an X/Y pad to adjust the Morph in the middle. The track on which you insert Morph Pro is the Main In, and the side-chain input is the Aux In. New to v3 is the Modeller, which not only plays user or factory samples, as an alternative to the Aux In, but also offers a Style Transfer function — effectively a granular processor that attempts to reconstruct the Main input using granular slices of the Modeller. It has several modes (Loop, RMS, Peak, Spectral and a Custom option), a sampler-style pop-up interface, a multi-zone mapping editor and additional analysis features so there’s plenty of control here. Also worth noting is that you can adjust the loop’s start and end points during playback, which can create some interesting sounds and effects.

You can morph only the Aux or the Modeller with the Main signal, but all the input sources are available in the mixer, which has level sliders, mute/solo buttons and high‑/low‑pass filters, so you can create and shape a blend of any of the inputs and the Morph.

On one axis of the X/Y control matrix, the engine starts with the Main In signal at point A and blends in the character of the other signal as you move towards B. On the other, it’s the other way around, but as we’re not mixing audio here, but rather creating a new signal, the results on each axis could be broadly similar or wildly different, depending on the sounds you’re working with. You can also choose from 11 morphing algorithms. Six are new to v3, and both Fusion and Sonance are unique to the Pro version. There’s ‘method behind the madness’, but I can’t claim to truly understand what these do — yet they deliver different

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results and I tended to just audition them and choose the best-sounding one. Either side of the algorithm selector are an algorithm-dependent knob and a Detail control; both help to finesse your Morph.

Finally, there’s a global output limiter and a handful of effects that apply only to the Morph: a transient processor, a ±2 octave formant shifter and a fairly basic (in terms of controls) but appealing and huge-sounding reverb.

Play Time I had a whale of a time feeding signals (sometimes very similar sounds, sometimes strikingly different) into Morph 3 Pro. Initially, the results ranged from woeful to wonderful, but after a morning’s experimentation I found I could consistently achieve interesting and engaging sounds. The results are always a bit unpredictable (it really is all about experimentation) but you soon get a feel for what sounds might work well and how to shape the result.

With similar sounds (two synth pads, say, or vocal oohs/ aahs and an evolving pad), you tend to be greeted with a wider range of musically useful results, and I found that the formant shifter was very

handy: blending a Morph I’d ‘deepened’ with the formant shifter with the Main signal often rewarded me with a dark, moody and interesting texture. Dragging the dot around the X/Y pad then generally gave me some useful control over the timbre.

With more contrast between the sources (eg. pads with percussion, voices with samples of engine noises, or drones with ambient chatter) the results were less predictable, the useable area in the X/Y pad usually smaller, and the sonic contribution of the secondary signal more obvious. But I found I could create some incredible soundscapes that way or, for example, liven up a repetitive drone or pad to sustain interest over time — some creations were vaguely reminiscent of the Orb’s Adventures Beyond The Ultraworld but that’s probably a story for another time!

Used with subtlety, such combinations can be a great route to soundtrack-friendly moods and tension.

Hopefully, I’ve given a sense of what this uniquely creative plug-in is all about.

There’s way more to it, but so much depends on the source sounds, so you’d be better off experimenting with the free 30-day demo than reading more of my words! Oh, and if the price is an obstacle, note that there’s a more afordable non-Pro version based on the same tech.

summary This innovative sound-design tool offers a good dollop of creative inspiration and, after an initial learning curve, is really easy to use.

£ Morph 3 €179. Morph 3 Pro €309.

W www.zynaptiq.com

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ON TE ST

BeesNeez Tribute 1 Valve Microphone available today. BeesNeez say that they have managed to track down a source of this material to use it in their M7 recreation. They have also designed and built a faithful replica of the original mic’s BV-08 transformer.

Unsurprisingly, though, they haven’t managed to recreate its unique VF14 valve, so like many high-end U47 recreations, the Tribute 1 uses an EF12 as standard; BeesNeez can fit or supply an alternative option if required.

In Use

The Neumann U47 is legendary for a reason — as this homage makes clear.

NEIL ROGERS

S

ince starting production of their own mics back in 2002, Australian pro audio company BeesNeez have built up an impressive catalogue. I reviewed the Lulu Tube from their budget-friendly Studio range (www.soundonsound. com/reviews/beesneez-lulu-tube) last month, and was impressed to find a high-quality valve SDC mic that would be affordable to a small studio owner like myself. For review this month we have another valve mic, but this time from their professionally priced Tribute range. The Tribute 1 is BeesNeez’s take on the classic Neumann U47.

There are already plenty of other options out there for engineers seeking a U47-inspired mic, so BeesNeez are keen to highlight the lengths they’ve gone to to reproduce key aspects of the original design and assembly, with the all-important capsule being a great example. The M7 capsule used in early U47 and M49 microphones was, apparently, made from gun-barrel material, which is not readily

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The powder-coated, matte grey finish to the Tribute 1 is very tasteful, and I like how minimal the mic itself looks — and how it doesn’t try to pass as a direct clone of its inspiration. Pattern switching is more versatile than on the original, running from omni to figure‑8 in nine steps through a switch on the power supply.

The build quality seems immaculate, and getting the mic up and running with the swivel‑mount connector and cable felt effortless.

I’ve crossed paths with a few different U47-type mics in recent years, and when I come across a really good one, I’m always reminded how tricky it can be to describe the sonic magic they impart without reverting to pro audio cliché. Sometimes there’s no alternative, however! On my first vocal session using the Tribute 1, I was impressed with how the upper midrange of a female vocalist, who I have recorded with for a few years, suddenly sounded more

‘present’ and ‘like a proper record’. And that’s U47s in a nutshell: there’s a reason these mics are so revered. They don’t suit every singer, but when they work, they can frame a voice in a way that is very familiar to our ears and helps it sit beautifully against the instrumentation.

The Tribute 1 sounded superb in the all vocal sessions where I used it, whether capturing silky vocal tones or with the singer leaning into the capsule for more authoritative, low-end‑heavy results.

I also had great success using the mic on instruments, and when used above a drum kit or acoustic guitar in its omni pattern, it produced a lovely ‘open’ sound whilst adding a sense of depth — especially when in a good-sounding recording space.

Final Thoughts I had high expectations for this mic, and they were comfortably met in all the sessions when I used it in my studio. As well as ticking the boxes I would want from an all-round large‑diaphragm studio mic, I loved how this mic gently pushes forward the ‘presence’ frequencies of most voices I tried it on. This is not a cheap mic by any means, but the exchange rate in the UK at the time of writing pitches the Tribute 1 pretty squarely price-wise against at least two of the better‑known U47-inspired mics currently available. With this in mind, If you’re in the market for a mic in this style, you should definitely put the Tribute 1 on your shortlist for auditioning.

summary A boutique, handmade reimagining of one of the most famous mics of all time, the Tribute 1 is both a faithful recreation and a beautiful mic in its own right.

£ £4165 including VAT. T Funky Junk +44 (0)207 281 4478 E sales@funky-junk.com W www.funky-junk.com W www.beesneezproaudio.com

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ON TE ST

JOHN WALDEN

W

hile Toontrack have certainly ventured into jazz drum sounds in the past, their latest expansion library for Superior Drummer 3, The Jazz Sessions, is undoubtedly their most ambitious project aimed at that genre.

Recorded in collaboration with James Farber — whose recording credits read like a Who’s Who? of jazz greats from the last 40 years — the expansion features seven kits selected to span the history of jazz drum sounds and styles. These were recorded in four different rooms within New York’s Power Station, James’ studio of choice. A full range of performance articulations are provided, and the library includes options for sticks, brushes, rods, mallets and hands. The full library comes in at 161GB, but you can opt for a more compact install by forgoing some mic options (for example, the surround mics).

Delicious Hot, Disgusting Cold Once installed, SD3 divides the kits based upon the four rooms used for the recording.

Each of these offers a distinct acoustic space, with the production team making kit selections for each room based upon the type of sound they were seeking. The kits themselves were chosen to span multiple eras, from the big band sounds of the 1920s to the jazz fusion sounds of the 1970s. That means that the available kits include both large and smaller kick drums, simple and more complex tom setups, and both darker and brighter cymbal choices.

For those more familiar with using SD3 for pop or rock drums, prepare yourself for some new mixing situations, as once these kits are triggered, there are some big kicks and splashy cymbals to manage. That’s in no way a criticism, though; the sounds are big, beautiful, detailed and totally in keeping with the genre. There are a huge range of sounds available across the seven kits and an impressive collection of presets created by both James Farber and the Toontrack team that ably show off the breadth and depth of the sounds. James comments on Toontrack’s website that the room mics are where he really starts when recording jazz drums, with the spot mics being used to simply reinforce elements as needed. It’s therefore interesting to solo the overhead mics in the presets as these present the kits beautifully within their respective acoustic spaces.

As ever, the SDX includes a dedicated MIDI grooves collection, all played by

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Toontrack The Jazz Sessions SDX

Superior Drummer 3 Expansion

Toontracks’ SD3 Expansion has drum sounds suitable for almost any jazz project you can imagine.

Toontrack’s Norman Garschke. These cover all the main eras of jazz and provide straight and swung options in both 4/4 and 3/4 and across a range of tempos. They also include specific performances with the various tools including sticks, brushes (these sound very cool), mallets, rods and hands. I’d be surprised if additional drum MIDI packs aimed directly at The Jazz Sessions SDX were not on the horizon but, during testing, some of Toontrack’s older jazz-themed MIDI drum grooves also worked very well with these kits. If your own jazz drumming chops are still a work in progress, the supplied MIDI content is a valuable element within the overall package.

If you happen to also be an EZkeys and EZbass (and perhaps the Upright EBX) user, SD3 and The Jazz Sessions is an ideal source of drum sounds to round out your virtual jazz trio. Indeed, writing this review took me way longer that it should have done because I got suitably distracted with this experiment myself.

It’s pretty amazing how quickly you can flesh out an idea when these three virtual musicians talk to each other via their respective Bandmate and Tracker options.

Nice...

Hats off to James Farber and the Toontrack team — The Jazz Sessions SDX is a bit of a triumph. It offers an impressive range of jazz drum flavours and gives you a huge amount of control over how you craft the sounds into your mix. The detailed nature of the sounds makes them super convincing. If you need to drop some world‑class drums into your next jazz composition, then look no further: The Jazz Sessions SDX is the perfect choice.

summary The Jazz Sessions provides a top-class collection of jazz drum sounds that would grace any recording context.

£ €179 including VAT. W www.toontrack.com

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ON TE ST

Waldorf Iridium Core Synthesizer

Waldorf’s all-encompassing Iridium engine is now available in its smallest form yet.

RORY DOW

A

s you may already know, the Iridium is not just any digital synthesizer. It’s Waldorf’s flagship.

With a sprawling synthesis engine embracing virtual analogue, wavetable, frequency modulation, phase modulation, granular, multisampling and physical modelling, the Iridium is a true all-rounder.

While being everything to everyone may seem impossible, the Iridium’s sheer scope is undeniably impressive.

The Core is the latest option in an expanding line-up and the smallest Iridium thus far. It measures 346 x 200 x 64mm and weighs 2.2kg, so it will fit comfortably on your desktop. It retains all of the features of its bigger siblings but with a reduced voice count of 12

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(16 in the bigger models), a few changes to the inputs and outputs on the back, no keyboard, and fewer dedicated encoders.

Atomic Number 77 Like the previous models, the Iridium Core revolves around a large 1024x600‑pixel colour touchscreen. Most dedicated controls for synthesis parameters found on the larger keyboard and desktop models have been sacrificed for a smaller form factor, but you still get the top row of Section buttons for navigating, Macro buttons that can be freely assigned, the six contextual endless stepped encoders, eight rubberised pads for triggering notes and chords, and a selection of other encoders and buttons for quick access and navigation. One addition that helps with the reduction in dedicated controls is the four assignable Control knobs located underneath the screen.

By default, they adjust filter cutoff, resonance, effects send and patch volume, but they can also be customised per patch,

giving you control over the most important parameters for any given preset.

Iridium’s synthesis capabilities remain identical to those of its older siblings.

Patches between all the Iridium models can be shared and loaded on any other model, which is easy thanks to the microSD card slot. The Iridium Core has 2.6GB flash storage for samples, pre‑loaded with a 2GB library of factory samples. There are over 1700 patches, with space to save several thousand more.

The synth engine uses two layers, which can be stacked, key-split and panned and assigned a custom number of voices per layer. Each layer uses a voice per note, so if you stack two polyphonic layers, you get a maximum of six voices.

Layers can also be played bi-timbrally from different MIDI channels. Each layer is essentially a full synth patch, with three oscillators, three filters, six envelopes, six LFOs, a Komplex modulator and up to five effects. Layers are always saved together as a ‘Multi’ but can be loaded individually.

The power of the Iridium lies in its many and varied synthesis methods.

These include classic wavetable synthesis that Waldorf are well known for, virtual analogue including eight-voice unison, multisample and granular (combined into what they call ‘Particle’ oscillators),

physical modelling, a speech synthesizer and even a highly customisable modular environment called Kernels, which allows you to combine amplitude modulation, frequency modulation, phase modulation, ring modulation and phase distortion in many different ways.

For filters, each layer has up to three multi-mode filters with different models that cover the full gamut of clean to filthy and can be routed in different ways. The filters are digital (unlike the analogue filters on the original Waldorf Quantum) and operate in stereo, which is great given that many of the oscillator types are stereo.

All this synthesis power is backed up by oodles of modulation: six LFOs (per layer), six envelopes with adjustable curves, and the aforementioned Komplex modulator, which combines two multi-stage waveforms to create a complex, looping, synchronised modulation source. Each layer has a 40-slot modulation matrix to help you make the most of all these sources.

In terms of effects, there are phaser, chorus, flanger, delay, reverb, EQ, overdrive and compressor. You can choose up to five per layer, and most parameters can be modulated. In addition, there is a global one-knob compressor and a bass boost option.

A 32-step sequencer and arpeggiator help keep things moving with advanced features like scale quantisation, rhythm patterns, swing and even MIDI output. The touch screen can be used as an X/Y pad modulation source too. In terms of MIDI, the Iridium engine supports poly aftertouch and MPE for whatever form of polyphonic expression you prefer. The Iridium Core has even kept its bigger siblings’ CV capabilities, albeit in a slightly reduced

Waldorf Iridium Core £1349 pros • It is the most affordable version of the Iridium yet with very few compromises.

cons • Reduced polyphony of 12.

• Mini‑jacks for MIDI I/O should not be a thing at this price.

summary The Iridium contains 35 years of Waldorf’s expertise in digital synthesis, and the Iridium Core is the most affordable version yet. Despite a slightly reduced voice count, it retains all the synthesis capabilities of the bigger Keyboard and Desktop versions.

form, with clock in and out and two CV inputs that can be used to play the synth monophonically or as modulation sources.

We’ve covered the finer details of the synthesis engine in previous reviews, so if you want the full breakdown of anything mentioned above, I refer you to our previous reviews of the Waldorf Quantum (Iridium’s digital/analogue predecessor) and the Iridium Keyboard in the April 2019 and September 2022 issues, respectively. I’ll summarise by saying that the Iridium engine is a powerful beast that encompasses many synthesis types and has the potential to keep you in new and exciting sounds for many years.

Choices, Choices Waldorf are aiming to sell the Iridium Core to people who cannot afford a flagship synthesizer that costs thousands. And they’ve done an excellent job of making it compact and more affordable, although you still cannot describe it as cheap.

Have they made the right decisions on what to cut from the bigger flagships?

Clearly, they did not want to compromise on the engine. Patches are identical to the bigger siblings, and that makes total sense. I assume that the Iridium Core is running on a slightly less powerful chip, hence the reduction in the number of voices from 16 to 12. But that is the only software downgrade.

The hardware is where the cuts have been made, most obviously by having fewer encoders and no keyboard. The side-effect of this is that the Iridium Core starts to feel a bit like a smart tablet with a few dedicated knobs and buttons attached. The lack of dedicated synthesis controls could impact your ability to use muscle memory to program the synth. I’m not sure that Waldorf could have avoided this, but it’s a fact nonetheless, and it’s worth bearing in mind that you can spend the exact same amount of money elsewhere and get a full hands-on synthesizer experience (just not the Iridium experience).

The price is higher than I expected, especially for a unit using mini-jacks for MIDI, and I’m not sure why Waldorf kept the rubberised pads. They don’t serve much purpose other than to trigger notes and chords, and they aren’t velocity or pressure sensitive, so they are just gated buttons. I’m sure the price could have been more competitive without them, or at least the MIDI in and out could have been upgraded to 5-pin DIN.

Round The Back The Iridium Core has many of the same connections as its siblings: a 12V DC power socket and switch, headphone output, stereo audio inputs and outputs on unbalanced TS quarter‑inch jacks, four CV connectors on 3.5mm jacks (reduced from eight on the Iridium Keyboard), MIDI in and out also on 3.5mm jacks, USB connectors for USB MIDI computer connection, and a USB host socket for class‑compliant controllers or USB storage. Also, there’s a microSD card slot for file transfers, backups and OS updates. There are no pedal inputs, which makes sense as these can be supplied via a MIDI controller.

All this being said, the Iridium is still a flagship synthesizer. It is still an unbelievably powerful synth that could keep you busy for years. As a digital polysynth, it’s hard to think of an alternative that can do more, and you could spend a lifetime getting the best from it. The Core represents the most cost-efficient way to own an Iridium, and many people will prefer the smaller design and appreciate getting access to the ‘full’ synthesis engine.

The Iridium software is full of nice little touches that make sound design more effective. The favourites system for patches is a lifesaver when you have thousands of sounds to go through. My personal favourite is the ability to save oscillator settings as presets. This allows you to save your favourite oscillators, including samples, or the much more complex Kernels configurations, so that you can reuse them in other patches. I point these things out as examples of how mature the Iridium engine is now. It’s six years old, has had many updates, and it builds upon 35 years of digital synthesis expertise.

I was sceptical that the Iridium Core may have been hacked away at too much, but after spending time with it, I think it’s a great package. The build quality is excellent. You get all the power of the Iridium engine and the only real things missing are four voices and some knobs and buttons. Waldorf are doing a great job of offering the Iridium/ Quantum engine in as many different forms as possible, and it feels like there’s an option for everyone. £ £1349 including VAT. W www.handinhand.uk.net W www.waldorfmusic.com

www.soundonsound.com / September 2024

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ON TE ST

Teknosign Box Line Double Width Audio Processors

With bold looks, original designs and affordable pricing, Teknosign’s preamp, EQ and compressor offer an intriguing alternative to better-known names.

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SAM INGLIS

I

t’s amazing how things can fly under the radar, even in this age of information.

I mean, if there was someone out there offering a large range of high-quality analogue audio equipment, mostly based on original designs and built entirely in

Europe, you’d expect to know about it, right? Especially if that equipment was being offered at very competitive prices?

Well, perhaps I’m just out of the loop, but I hadn’t heard of Teknosign until I ran across their booth at this year’s NAMM Show. And when I did, I was impressed.

Analogue outboard is now a choice rather than a necessity, and the market has changed as a result. There used to be numerous European and American manufacturers who specialised in developing affordable, functional, engineering-led designs, but quite a few have disappeared or changed their business models. The low end of the market is now dominated by Far Eastern manufacturing, and by copies of vintage outboard. A few companies successfully buck the trend — Drawmer, Radial Engineering, DAV and Lake People spring to mind — but increasingly it seems as though Western manufacturing and original designs are being squeezed out of the picture.

So it’s refreshing to encounter a European manufacturer who are doing things their own way. Based in Liguria, Italy, about halfway between Genoa and Pisa, Teknosign began producing audio equipment in 2011, building on the already extensive electronics manufacturing experience of company founders Riccardo Angeletti and Claudio Furno. Notably, Teknosign don’t just design equipment to be manufactured offshore: they build

Teknosign Box Line Double Width pros

• Good-sounding processors, with the features you’d expect from professional units.

• Pleasingly affordable, especially considering they are built entirely in the EU. • Versatile, well thought-out semi-modular system allows up to eight units to be powered from one PSU.

cons • Loose controls on review units.

• Panel legends under the knobs are hard to read.

summary Teknosign’s Box Line is an impressive range of European-made analogue outboard, containing a broad range of processors that are affordable to almost everyone, yet perfectly at home in professional recording studios.

everything in their own factory, and in fact their relatively affordable prices are made possible by their ability to make almost every component themselves. PCBs, cases, knobs, buttons, switches, transformers and more are all produced in-house, and the only items that have to be bought in are meters and audio connectors.

What’s In The Box Line?

New items are being added to the Teknosign range all the time, with the launches at this year’s NAMM Show including lunchbox-type guitar and bass amplifiers. There’s a pretty extensive selection of utility products, such as patchbays and power conditioners, and a handful of 1U rackmount processors including two summing mixers. But most of the company’s studio tools are currently offered as part of what they call the Box Line. These processors occupy one of two form factors, which can be used as freestanding desktop units or housed in a 2U rack enclosure that is sold separately. A single 2U rack housing can accommodate four standard Box Line processors or two Double Width units.

At the time of writing, there are nine products in the basic Box Line. Many of these are utility devices — there’s an active and a passive DI, a passive stereo mic splitter, a headphone amp, a powered USB hub and a phantom power supply, for example — but the Box Line also includes the single-channel SSMP or Solid State Microphone Preamplifier, and what must be the world’s most compact 16:2 summing mixer. The Double Width range, meanwhile, comprises three products. The DSMP is a two-channel version of the SSMP, the PEQ is a solid-state equaliser inspired by the classic Pultec EQP-1A, and the VMC is a rather interesting compressor, of which more presently. For the purposes of this review, Teknosign sent me these three units along with the one basic Box Line device I haven’t yet mentioned.

That device is called the RPS, and its diminutive front panel is home only to a fused IEC inlet and an on/off switch. Turn it around, though, and you’ll find eight DC power sockets. All powered Box Line and Double Width processors run on 24V DC, so a single RPS can be used to power up to eight units, with patch cables of different lengths included. It’s no different really from powering a guitar pedalboard, except that you don’t have to worry about devices with non-standard requirements, and the front-panel on/off switch on the RPS means

ALTERNATIVES The most obvious alternative I can think of for the DSMP, in terms of design and format, is DAV Electronics’ BG1. On the Pultec-style EQ front, both Warm Audio and Klark Teknik make keenly priced valve-based copies of the original EQP-1A. And if you’re looking for an affordable analogue compressor in a similar compact format, you could consider one of several options from Golden Age, though these are all vintage-inspired rather than original designs.

you don’t need to access the rear-panel on/ off buttons on the processors themselves.

Primary Colours With their contrasting blue front panels and red knobs, the Box Line products are eye-catching and very distinctive. I rather like the cosmetics, which come across as friendly and functional at the same time, and the only minor criticism I can level is that because the knobs protrude quite a long way from the cases, any legending underneath them is hard to see. The cases are made of aluminium, and seem robust.

The settings of switched functions such as phantom power on the mic preamp or bypass on the compressor are clearly indicated using LEDs. Some of the controls on the review units were a little wobbly and I’d have liked a bit more resistance from the pots, but in general there’s nothing to suggest they wouldn’t survive intensive use over a long period.

Audio I/O on all three of the Double Width units is exclusively on rear-panel XLRs, and is limited to the basic ins and outs. There’s no external side-chain input or stereo link function on the compressor, for example (although a 1U stereo version is in the works), and no high-impedance input on the mic preamp. All are designed to operate at ‘professional’ levels, with the compressor and equaliser accepting inputs of up to +20dBu and the metering configured so that 0VU=+4dBu. All three come with 24V external power supplies, so the RPS is very much an optional convenience rather than a necessity.

Turning first to the mic preamp, this offers a 69dB gain range in 1dB steps, courtesy of coarse (10dB) and fine (1dB) rotary switches. Phantom power and polarity are individually switchable for each channel, as are not one but two filters. The high-pass filter has a 12dB/octave slope turning over at 80Hz, and there’s also the much more unusual option of a 6dB/octave low-pass

www.soundonsound.com / September 2024

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ON TE ST

TEKNOSIGN BOX LINE DOUBLE WIDTH

A single RPS power supply can power up to eight Box Line units.

you can overdrive it to good effect, though you need to be a little bit careful as the onset of saturation is quite sudden. Approached more conservatively, I’d be happy to use the DSMP for critical tasks such as stereo main pairs, and my measurements suggested that the two channels on the review unit were matched to within 0.1dB or so.

What, No Valves?

filter turning over at 10kHz. The circuit itself is a transformerless, solid-state design, and offers a huge frequency range of 15Hz to 300kHz. The most surprising specification, though, is the 600Ω input impedance.

Nowadays, most gear is designed to have comparatively high input impedances and low output impedances. The rule of thumb is that if the input impedance is at least 10 times the output impedance of the source device, loading effects will be minimal and the frequency content of the source will be preserved. Typically, therefore, modern mic preamps have an input impedance of at least 1.5kΩ and often much more, while mics are commonly specified with a 150Ω output impedance. The DSMP’s design could be seen as harking back to an earlier era when impedances were matched to maximise power transfer. Practically speaking, it’s likely to make little difference with electronically balanced mics, but could have a noticeable effect on the sound of transformer-balanced designs, especially vintage ribbon and dynamic models.

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That was borne out to an extent in my testing. The DSMP is certainly a very capable general-purpose mic preamp, and when confronted with a modern capacitor mic, does exactly what you’d hope, amplifying the signal without adding tonal changes, distortion or noise. But it sounded particularly good on, for example, the Warm Audio WA-44 I reviewed in last month’s issue. As a sort of joint test I recorded an entire song using nothing but the WA-44 through the DSMP (and, on occasion, the PEQ), and the results really jumped out of the speakers in a way that recordings made with ribbon mics don’t always do.

The choice of VU meters gives the DSMP vintage appeal, but means that if you want your output to register at a healthy mark on an A-D converter calibrated to professional levels, you’ll have to get used to the sight of the needle being pinned against the end stops. Not a problem in itself, but it would be nice if there was also some visual indication of overloads. Put an attenuating device in the signal chain after the DSMP — the VMC will do nicely — and

Of the three Double Width units in the Box Line range, the PEQ is the only one that’s obviously indebted to a vintage design. Its three EQ bands follow the template established by the classic Pultec EQP-1A, meaning that you can apply up to 15dB boost and/or cut in the low band, up to 15dB semi-parametric boost in the midrange and a separate high-frequency cut. The Pultec is a passive equaliser that uses a valve make-up gain stage, and hence is sometimes described as a valve EQ. Both the valve stage and the input and output transformers contribute to its distinctive thickness or colour, and engineers have been known to patch a Pultec in set flat, for these qualities alone. That’s not something you’d do with the PEQ, which is a transformerless device that uses solid-state circuitry in its active stages, but it’s still a pretty sweet-sounding EQ. With midrange boosts in particular, there’s an appealing ‘lightness’ to the sound that might be harder to extract from a full-fat Pultec, and of course you can still do the well-known trick of simultaneously boosting and cutting in the low band that is so characteristic of the Pultec design.

Compared with the original, Teknosign have added extra switched frequency options to the low band and the high‑cut band. The latter are all at potentially useful frequencies, and although I had my doubts about the value of the 20Hz and 25Hz settings on the former, their effect can be audible well into the midrange — and when you do the simultaneous cut/boost thing, each setting does sound distinctly different.

Selecting the 20Hz setting and turning both the Boost and Cut controls a long way up can do something very nice for bass instruments, and makes a surprising difference even on more midrange-y sources such as vocals.

And so to the compressor, which is perhaps the most unusual of the three units.

VMC stands for “variable-mu compressor”, a term that usually denotes the use of a valve as the gain-control device. But, like the other two Double Width units, the VMC is a defiantly solid-state affair. Little information is forthcoming about the actual operating principle, but unlike many actual valve compressors it has the full set of standard controls, with attack and release both running from 10ms to 500ms and six ratio settings including 1:1 at one end of the scale and infinity:1 (limiting) at the other. The VU meter can be set to display either gain reduction or input level, and there’s a hard bypass button as well as two further knobs.

On the review unit, these were rather misleadingly labelled Makeup and Master; the Makeup control is actually a wet/dry mix balance knob, while the Master applies ±20dB make-up gain following this.

Despite the absence of valves and transformers, the VMC certainly isn’t a boring utility compressor, but neither is it a one-trick pony. With longish time constant settings, it does the sort of gentle smoothing-out that I associate with some old-school designs: not transparent,

exactly, but instinctively sympathetic to the source, and seeming to maintain the natural dynamics of a voice or bass guitar even as it shaves 10dB off the louder bits. Turn the release control all the way anti‑clockwise and it gets somewhat more ‘grabby’, and can be made to pump on percussive sources, though it’ll never be as fast as something like an 1176. Initially I wasn’t convinced that the wet/dry control was going to be all that useful, but discovered that blending a small amount of the dry signal back in can be a really nice alternative to backing off the threshold when you’ve gone just that bit too far. As is often the case, I got the best results when I set the controls by ear rather than by eye; this is much easier to do on hardware than it is with plug-ins, and led me to use settings I wouldn’t have thought to try otherwise. The forthcoming stereo version will make for a really tasty bus processor at, one hopes, a very competitive price.

Box Clever All in all, I think there’s a lot to like about these cheerful and compact little boxes.

The competitive pricing may be apparent in the feel of the controls, but it certainly isn’t reflected in either the sound or the capabilities of these processors. They lend themselves equally well to desktop or rackmount use, and the fact that each Double Width unit comes with its own power supply also means that they’re self-contained, so you can get started without the need for a 500-series chassis or similar. Then, once you have a couple of Teknosign processors in your collection, you can think about adding the RPS to help tidy everything up.

As I mentioned at the start of this review, using analogue outboard in this day and age is a matter of choice rather than necessity. Teknosign’s comprehensive range opens up that choice to many new people without compromising on audio quality. £ DSMP £769; PEQ £529; VMC £609; RPS £262. Prices include VAT.

T Teknosign +39 0187 559992 E info@teknosign.it W www.teknosign.com

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www.soundonsound.com / September 2024

31

TECHNIQUE

Music For Money

Moby’s decision to license tracks from Play for use in commercials helped the album sell better, and made him a lot of money.

Photo: Bill Ebbesen

The World Of Commercial Composition Composing for adverts and music libraries can be rewarding, challenging and creative — and did we mention that you get paid?

SAM BOYDELL

T

he year is 1999. Moby has released the album Play to critical acclaim, only to go and do something unforgivable in the eyes of music critics: he has licensed the music for use in adverts.

How dare he?

As an educator and musician who’s watched endlessly talented people fail to cement a career in music, you’ll have to forgive me for starting by simply addressing the ridiculous notion that music and commerce cannot creatively coexist. In a world where music is basically free to the consumer, the only way we can survive is by exploiting other sources of income apart from direct sales. Currently, one main avenue is in commercial composing and sync licensing. It’s widely known that record labels make a significant portion of their income from the sync sector, but you don’t need to be signed to a major to do so.

Don’t be fooled into thinking that writing music for commercials can’t be creatively inspiring, either. I’ve been fortunate enough to work with some amazingly talented film-makers, many of whom are creating adverts that are more likely to pop up at Cannes than on TV. For instance, I’ve written music to gamechanging fundraising

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adverts, highlighting heart disease and poverty. I’ve also made music for museum spaces, aeroplanes and world record attempts. Commercial composers are just that: composers.

Music For Commerce Let’s begin by identifying two different disciplines. There’s commercial composing, which is the art of writing to picture, and there’s sync licensing, which is the licensing of pre-made music to be ‘synchronised’ with another form of media.

Commercial composition requires a more businesslike approach. You are forging direct relationships with directors and advertising agencies and providing a service for them whereby a fee is paid to cover your time and the licensing cost. The skill set needed is multi-faceted. You have to be a great communicator, learning how to get vital information from your clients. You have to be adept at running a business in order to stay relevant and financially stable. And you have to produce fantastic music under significant time constraints and expectations. People are surprised when I say I often have to produce a ‘radio-ready’ demo, from scratch, in under eight hours. Not easy.

Sync licensing and production library music are slightly different. You aren’t writing to picture. Sometimes sync licensing can be a way of exploiting old music, but often new music is written to a brief: an overview of an ‘album’ concept that an agency may want to deliver. As a basic example, you may get the brief for a ’70s psychedelic rock album with a deadline months away. You’ll

submit multiple ideas, and often enter into a dialogue with multiple revisions before tracks are accepted. Albums in this format tend to go out to TV, from which you will be primarily paid through royalty collection societies such as PRS and MCPS in the UK. This happens as and when your music gets played out in the public domain, so the income you receive will be related to the amount of usage your music gets.

Production music agencies that are affiliated with collection societies such as MCPS tend to operate standard fee structures as a consequence, but there are also online library sites out there that have no such expectations or affiliations and can charge as they choose to. They may, for example, buy out all the rights to your music for a flat fee.

Copyright, Licensing & Royalties Royalties are the payments generated from the usage of your music, with the three main types of royalty being mechanical, public performance and sheet music: • M echanical royalties are generated from the reproduction of music in digital and physical formats such as CD sales, streams and sync uses.

• Performance royalties are incurred any time a song is played in public, whether by live musicians or playback of a recording in a public space.

• Sheet music royalties come from sales of printed or downloadable sheet music.

There are two main types of copyright holder: those who own a share of the songwriting, also described as the publishing rights, and those who own shares in the master recording. The easiest way to understand this is to think of a cover song. The writers of the original song typically hold the publishing rights, while the musicians (or their label) involved in the recording of the cover hold the copyright in the master recording.

Note that practically every country has different rules about music copyright and royalties. In addition, areas not within the boundaries of a country are therefore not bound to these rules either, as I found to my detriment when producing music for an airline!

Royalties & Licences As mentioned, commercial composers are working in two areas: writing direct to picture and through the use of production

agencies, or publishers as they are often known. Publishers generally work on a 50/50 copyright basis. If you look at a track’s submission sheet in PRS For Music, the UK’s primary royalty collection society, it will state that the publisher owns 50 percent of the Performing Share and 100 percent of the Mechanical Share. They give you the 50 percent back later, because this is direct income from their clients, not fully distributed through PRS or MCPS. When you’re writing directly to picture, you often play the part of the publisher too, and it’s likely you haven’t or won’t be selling any copyright to an agent. This means you will be dealing with the administration of royalty collection and licensing yourself, so for example it’s up to you to submit the relevant information to PRS. A sync licence is simply a document that outlines exactly how two parties intend to use the music and what is expected from either side within the agreement. A licence will typically specify the parties and music involved, who owns what share of the copyright, the working title(s) (known as works), the services and terms, distribution, territories covered, validity period and any additional terms, such as exclusivity or usage within cut-downs.

My advice is to be accurate and vigilant. These are documents intended for legal use, and lines often get blurred.

Not only is there the need to future-proof these documents, but it’s important to understand that even a word change in an advert means it is considered a different ‘work’ and therefore requiring of a new licence. Thankfully, MCPS provide a fantastic template for any licences you write yourself, and following their rate cards helps immensely.

Royalty payments are complex, and determined by each country’s royalty collection societies. When it comes to licence fees, MCPS generally set the precedent and therefore the best publishers for artists will likely be those associated with the MCPS directory. It’s also a good basis for determining your own fees.

As mentioned previously, online library sites are often not subject to these higher fees, so it’s important to look into each company’s individual price structure and how you will be paid, plus the agreement you are making, and judge from there. Most publishers on the MCPS directory will not accept submissions that are on library sites, or will ask you to remove them (which can take significant time).

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www.soundonsound.com / September 2024

33

TECHNIQUE

THE WORLD OF COMMERCIAL COMPOSITION

A Foot In The Door How do you get into making music for adverts and TV? I can only speak from my own experience. I had the desire to write music in this format from the beginning, so pitched my shop front up and tried to connect with any local film-makers I knew.

Around the same time, where I live in Cornwall, there was a tide of corporates trying to get their product involved in surfing. I knew most of the surfers and, as there were no real budgets to speak of and consequently no industry heads dictating anything, I ended up writing for some big corporations. One in particular helped grow my portfolio significantly, as they liked what I was doing and linked me up with the many other sports they sponsored.

This fuelled rapid growth and development, especially when some of the film-makers I’d worked with climbed the ladder and continued to use me or my ever-growing library of music. I got to know other clients primarily through word of mouth or by forging a direct relationship through constant networking, and only rarely over email or social media. I must have sent 10,000 emails in my time, through which I’ve gained only a handful of jobs.

By contrast, getting your foot in the door with libraries can take a lot of submissions and consistent emailing.

It’s worth the effort to find one that suits you and your style of music, and which can help you grow. I’ll always be an advocate of publishers over online-only libraries, as they are based around more interpersonal dealings, with a higher value placed on you and your music.

Goods In The Window It’s important to remember that any connection you make, even informally,

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is often followed by the person secretly checking out your website or online shop front, so these have to be engaging, creative and savvy spaces.

What makes a good ‘shop front’? First, your website has to engage the end user and stop them immediately looking elsewhere. We’re creatives, let’s wow them!

Secondly, we are in the commercial world where people have very little time and need an easy process. Ask yourselves these questions: • C an you make your website more immersive, impressive or easy to use?

• Do all your links work, and can potential clients easily demo your work with a click or two?

• Does your site give the impression that you are an experienced business person (even if you aren’t)?

• Does it give the impression that you are brilliant at what you do? If not, why?

• Looking at it subjectively, are there clear gaps in your work? How can you plan to fill these?

To add to this, I would consider thinking about consistent branding and communication strategies that you can implement across all your social channels and online presence, so everything feels connected.

Spread Your Wings

of potential gigs, but they can also create a lucrative niche. If you are particularly prolific, one possibility is to work under multiple artist names.

Invest In Yourself As composers, we are in competition with many others. Quality of product is key here, and investment in the best is necessary.

That applies both to equipment and skills.

We’re often asked to beat temp music from a hugely successful artist, and we may also be spanning multiple genres and disciplines.

What you use to make, record and mix your music is a matter of individual choice, but it’s important that it be immediate, fast to use and fully recallable. Some people expect all film composers to have a surround mixing setup, but as of 2024, 99 percent of adverts and TV features are asking for stereo only, so the need for any expansion above that will be rare at this stage. What is important is that your monitoring is translatable. Part of this is having great speakers in a room that you know, understand and trust, but you also need to be able to send clips quickly to your phone for reference because that’s where 99 percent of consumers will be listening. In addition, I personally have a range of really lovely headphones, as well as the ones that come with my phone, and a Bluetooth speaker that I’ve got wired directly in for single‑button A/B comparisons. Most phones are stereo these days, so I very rarely check mono compatibility.

Lastly, you have to just make stuff sound fantastic to the end user. So much of this industry is about trust in one another, and people have a lot riding on their work.

So I want to end by placing importance on personal learning and self-development.

Learning techniques such as how to make bass come across on a phone will be hugely valuable. This is an emotion-led industry, and we’re here to take the emotion up to a level they didn’t realise was possible!

Getting there is going to require experience and graft.

Does the commercial composer need to be a jack-of-all trades, or is it best to focus on a specific genre that you’re good at? There are no rules, but I would say this question is mainly about how you advertise yourself. I think you’ll find any working musician is incredibly talented in many fields and genres, and it can only be a good thing to expand your knowledge.

As a small example, my first jobs were all dubstep-based, although I had no experience or even any intention of being associated with dubstep. The same layering techniques I learnt 10 years ago in doing that recently came in handy on an orchestral piece, where the client stated, “You’re the only one who’s managed to get the drums powerful enough for me.”

In terms of business practice, sometimes it’s a pro and sometimes it’s a con when you put yourself out there as a jack of all trades, because people love to put you in a box.

It’s vital to audition your Generally, those who work on smartphone speakers, have a ‘sound’ risk ruling since that’s how much of your themselves out from a lot audience will hear it.

Photo: Santeri Viinamäki

In terms of what we’ll call ‘enterprise income’, there are no rules, but the industry will naturally set amounts for you. For instance, it’s very rare that a publisher will pay up front for your music or any submissions. But you hope they are very good at what they do, and you’ll make money together. The same can be said if you choose to sell the rights to your music, or upload to a micro-transaction site. These are all business decisions you have to make.

Similarly, bespoke composing for commercials is a business in itself. You have to consider your experience, the budget of the client, time taken and distribution. They will often dictate their budget and it’s up to you to negotiate.

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ON TE ST

Zoom H4essential Portable Recorder

“Just press record,” they say.

Can it really be that easy?

CHRIS TIMSON

F

or the best part of 20 years, Zoom Corporation have been making themselves a force to be reckoned with in the field of portable audio recorders, and I have a couple of their devices: the H3-VR Ambisonic recorder and the F8n eight-track field recorder. Both are well implemented pieces of kit, and I believe the H3-VR remains unique. So it was with some interest that I accepted an invitation to review the Zoom H4e (‘e’ for essential) recorder. Superficially, it isn’t dissimilar to the Tascam Portacapture X6 I reviewed in SOS June 2023, so I wondered how they might differ. It turned out that they represent two very different design philosophies and I’ll go into more detail about that below — but first let’s consider the Zoom on its own merits.

Hard Times The H4e is one of three recent arrivals in Zoom’s H-for-Handy range of recorders, all of which share the same design philosophy. The H1e is a basic handheld

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stereo recorder, with two built-in mics in an X-Y arrangement. The H4e adds two XLR/TRS combo sockets for external mic/ line sources, and the ability to record four tracks simultaneously. Finally, the H6e adds another two XLR/TRS combo sockets, and can record six tracks at once.

All three appear to use the same built-in mics, so I’d expect them to sound pretty similar when recording through these in stereo. The big headline for all these recorders is the 32-bit floating-point recording format — in fact, they can record only in this format, and that’s a significant pointer to the intent of these machines, as I’ll explain in more detail below. The H6e is very similar in spec and use to the H4e apart from those two extra XLR/TRS inputs and the larger physical size necessary to accommodate them, so my conclusions will likely apply to both models. The H1e looks to be quite a different beast, though, with different controls and UI, so I won’t consider it further in this review.

The H4e is quite light in weight for its size, not least because it requires just two AA batteries, and while the case is plastic it still feels pretty sturdy so it’s quite practical as a handheld recorder.

The shape is no surprise either, with mics at one end, a colour screen on top and

controls scattered around as required.

This layout works, so why change it? The built-in mics function as an X-Y stereo pair by default but they can be configured for mono capture. The screen is quite small compared with some other recent recorders but it does suffice. Below

Zoom H4e £189 pros • Extraordinarily easy to use.

• Good built-in mics.

• 32-bit floating point recording.

• Fast boot time.

• Doubles up as an audio interface.

• Good battery life.

• Accessibility option may be seriously useful for the visually impaired.

cons • Some might prefer more user-adjustable settings.

• No PSU, SD card, batteries or physical documentation included.

summary The H4essential takes advantage of 32-bit conversion to streamline the process of recording. You can’t tweak much, but it’s quick to power up and, if you point the onboard or external mics in the right direction and hit record, the resulting recording should be good.

Zoom haven’t followed competitors in going down the touchscreen route, but there aren’t deep menus to dive either. Everything is accessed using either the thumbwheel and enter button on the side, or the dedicated buttons beneath the screen.

the screen, you have a set of transport controls and a couple of odd bods such as Mixer and Mic. Along the left-hand side is a line in socket that can also provide plug-in power for a suitable mic. Said mic will take precedence over the built-in mics if you use it. A line out socket doubles up as a headphone output that gives plenty of volume and quite a good sound for monitoring. Next to that there’s a volume control for the headphones and also for the in-built speaker that’s found on the back. As usual with portable recorders like this, to describe the sound quality of that speaker as ‘poor’ would be unusually kind, but of course that’s not it’s purpose — it’s really there for basic checking like, “Did I actually make a recording at all?”

A plus point is that it’s quite loud. Next to the volume control is the microSD card slot (the H4e accepts microSDHC and microSDXC cards with capacities up to 1TB).

At the lower end are the two XLR/TRS combo sockets that can provide 48V phantom power if required, and can thus support the full range of professional mics. These don’t replace the in-built mics, which are always available, but can record to additional tracks, giving us the full four that are advertised.

On the right-hand side is an on/off/hold slider switch, and a slot for the optional BTA-1 Bluetooth adaptor. There’s a USB‑C port that can be used to supply power and/or to support file transfer. You’ll have to provide your own PSU for external power, but most households will have such things coming out their ears, and for location work you can have the option of using an external USB power brick.

Finally there’s a twiddly wheel and an Enter button. Both are key to the user interface, of which more anon.

On the bottom panel, a quarter-inch threaded insert enables mounting on a camera tripod but, if you’d prefer a mic stand, 1/4‑ to 3/8-inch adaptors are widely available.

Getting Started The H4e recorder arrived in a neat little cardboard retail box accompanied by a few bits of paper concerning warranty and safety issues (it’s amazing how many safety issues it seems a device like this can have!), the Bluetooth adaptor (a cost option) and... nothing else: no batteries, no SD card, no PSU, no user manual and no quick start guide. I enquired to see if this was intentional and was told all was as it should be.

Apparently, Zoom wish to minimise waste and do their bit for the planet, the idea being that you download the PDF manuals from Zoom’s website. I do have some sympathy with that idea, but I didn’t notice the QR code on the box until quite a bit later, and do think that just a couple of sheets providing a getting started guide would be a Good Thing.

After I’d scrabbled around trying to find an unused microSD card (eventually

borrowing the one in my Zoom H3-VR) this gave me the perfect opportunity to try out my standard test with new kit: seeing how far I could get without a manual!

The answer, once I’d worked out how the user interface worked, was a very long way. In fact, within about an hour, without any recourse to manuals, I had pretty much full control of the recorder — the user interface takes simplicity to a whole new level.

Having said that, when I first turned the H4e on I found myself vainly poking at the screen, thinking it might be touch‑sensitive like the one on my Tascam Portacapture.

Then I started prodding buttons randomly, but not understanding the results until I got onto the little twiddly wheel (officially named the Selection Dial) and the Enter button.

When first turned on, the recorder takes you through a setup procedure for language, accessibility, data and time. I soon twigged that you use the twiddly wheel to move between fields on the screen. Once the field you want is highlighted, you press the Enter button and a list of options is displayed. You use the twiddly wheel to scroll to the desired option and press Enter again to choose it. Apart a few single-purpose buttons on the front, this is the whole and only way you interact the user interface. Once I’d got used to it, I found working with it reasonably straightforward, helped by the fact that the option lists are deliberately quite short. However, when I tried the remote Bluetooth app on an iPad I realised how much I missed the touch screen on my Portacapture.

Once past the startup procedure, I arrived at the normal starting screen, at the top of which is displayed the elapsed recording time, the total amount of available recording time and an indication of the battery life. In the middle is a section for each microphone, and all are turned off.

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Although unable to set the mic gain (you don’t need to given the 32-bit recording) you can set the output levels and mix of the different mics, and the mix can both be auditioned and recorded alongside the mic channels.

At the bottom are a series of icons that you select to set up various things.

Setting Up First, there’s File List, and selecting this accesses a list of the recordings you’ve made and lets you play or delete them.

This would be a good point at which to describe the H4e’s file system: in brief, it hasn’t got one! Well, under the bonnet it undoubtedly has, but the user has no control over things like file names or folders; it’s all decided for you. The file list is a list of recordings you’ve made, identified only by the date and time you made them. You can’t, for instance, supply a root name to be used to identify a set of recordings as you can on, say, the Portacaptures, or create a folder to group a set of recordings by theme.

A typical recording name might look like 240615_122813, which would mean it was started on 15th June 2024 at 12:28 and 13 seconds.

Another icon, Input, accesses settings such as low-cut filtering, mono mix on/off for the built-in mics (by default the mics work in stereo, mono mix on makes them behave as a mono mic), plug-in power, phantom power and 1+2 Link for the external mics (provides stereo linking, but Mid-Sides is also an option for the external mics). While there is what amounts to a way of setting input levels if you want to, it isn’t here! An Output icon accesses a menu option to use or disable the volume control — I can’t see why you’d want to disable it, myself.

The Rec icon accesses a few recording parameters, including sample rate (44.1, 48 or 96 kHz) and metadata to the iXML standard. Notably ‘bit depth’ (digital

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word length) is missing and that’s because, as I mentioned, this machine always records files at 32-bit float. A significant parameter is Rec Source, which offers two options: Pre Mixer and Post Mixer. More about this below, where I discuss the output mixer.

There’s also an SD Card icon — you can test or format the SD card here — and another called USB, for when you connect the H4e to a computer. You’re offered a choice of File Transfer or Audio Interface modes, and for the latter there are a few setup options, such as whether the H4e functions as a stereo or multi-channel interface. Apart from verifying that it did indeed show up on my Mac as a valid interface offering 32-bit floating point recording, I didn’t pursue this further.

Choosing File Transfer means it will show on your computer as a storage drive. You will see a list of folders, each with the ‘name’ of a recording made earlier, and in each folder is a set of files that actually hold the data. For instance, imagine the recording 240615_122813 we made earlier had the onboard mics set to on and each of the external inputs active as two mono mics. You’d see four files: ‘240615_122813_Tr1.WAV’ for the first of the external mics; ‘240615_122813_Tr2.

WAV’ for the second; ‘240615_122813_ TrMic.WAV’ for the two onboard mics (whether stereo or mono); and ‘240615_122813_TrLR.WAV’, the output of the mixer section of the recorder.

Finally, we come to the System icon. This is for the settings not already covered above, with notable examples being Accessibility, Bluetooth and Help. Accessibility turns on what Zoom call Guide Sound: a female American voice speaks every heading when it’s highlighted, speaks every value you select and so on. It beeps whenever messages appear too. This is audible over the speaker except if headphones are plugged in. I’ve not seen this in a recorder before and I’m genuinely impressed, as it could make high-quality recording available to the visually

impaired. Nice one, Zoom! Bluetooth relates to the optional Bluetooth adaptor, the BTA-1, of which more later. The Help option brought a wry smile...

Mixing It With The Best The transport controls on the front of the recorder do exactly what you’d expect, but what about the other four blue buttons? The Mic button turns on/ off the built-in mics for recording, 1 turns on the left-hand external mic and 2 the right-hand one. As each is selected, a red light shows above the button to indicate that the corresponding mic/array is armed for recording, and the display shows traces that correspond to the volume on each mic’s channel. It’s the sort of thing you might normally use to set your recording levels — but, of course, on this machine you don’t need to do that! The state of the Mic buttons is retained when the recorder is switched off, so if you left, say, the built-in mics armed when you switched off then when you next switch on they would still be armed and ready to record. It’s when you press the big red button in the middle that recording starts.

Pressing the Mixer button brings up a new screen that, as you might expect, allows you to mix the output of the mics into stereo audio that can be listened to using the speaker or headphones. When

For those who like touchscreens, or want remote control, there’s an iOS app (though as yet there’s no sign of an Android equivalent).

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you’re recording, this mix is also saved as a stereo file, as I mentioned in passing.

I also mentioned the Rec Source option above and, in conjunction with this screen, this can have a big effect on the files you record. When set to Pre Mixer, each of the mic files will be at a default level, while the stereo mix file uses the levels set in the mixer. When set to Post Mixer, each of the mic files will use the levels set in the mixer, so, by a slightly circuitous route, you can set the gain levels for your recordings if you wish to.

You can also buy, separately, the optional BTA-1 Bluetooth adaptor, which works for the H4e and a number of other Zoom devices (it’s the same as the one for my H3-VR, for example). When plugged in, this provides a Bluetooth connection to compatible devices that are running the free control app. Currently, these seem only to be iThings, with no support for Android devices, which was a surprise — it’s a control app after all, not an audio one — so hopefully that will change in the future. Fortunately, I have an iPad, so I downloaded the H4essential Control app from Apple’s App Store, installed the BTA-1 in the H4e, started the app and established the connection.

As you can see from the screen, the app largely mirrors the H4e’s hardware UI, though you don’t have equivalents to the twiddly wheel and the Enter button.

Instead, you just tap icons and lists in time-honoured tablet fashion. Frankly, it’s nicer than the twiddly wheel, but the main reason for using the app on an iPad or iPhone is to be able to control The test recording session, captured on both a Zoom H4essential and a Tascam Portacapture X8: the quality of results was very similar (and good in both cases!).

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the recorder when you can’t be with it.

For example, you might have it recording a band from the front while you’re back in the audience, or maybe you’re recording yourself playing guitar and don’t want to have to put down your instrument between takes. You can start and stop recording and perform similar tasks very easily. It’s not an essential add-on, but it’s a very useful option.

The Proof Of The Pudding...

The H4e offers an interesting, simplified approach to recording, then, but what’s it like in use and how good are its recordings? On the evening of the day the H4e arrived, my wife’s band were playing in a charity concert at a local church. I’d already decided to record the concert using my Tascam Portacapture X8’s built-in mics, which I regard quite highly, but it was a no-brainer to take the H4e along and set it up next to the X8.

I spent a little time setting up the X8 using its superb mobile‑phone-inspired interface. For the H4e, I left Rec Source at its default Pre Mixer setting, and did little more than point it in the right direction and ensure its built-in mics were turned on. I set the X8 to record in 32-bit float, but with the H4e didn’t have to, since it’s the only option.

Later, I loaded the recordings into Logic Pro, which can now use 32-bit float files natively. The Pre Mixer recordings of the H4e were startlingly quiet to my ears; obviously, the designers had taken advantage of the extremely low noise floor of 32‑bit float to set a very conservative recording level that would not be overloaded under almost any circumstance. The X8 file was a good bit louder, but only because of my old habits when setting the gain; I could have set its recording levels as low as the H4e’s, or used the latter’s Post Mixer setting to bring its levels up, without affecting the quality of the result. To my ears both recordings were very good and I’d be happy to use either device.

Clearly, the H4e is capable of making excellent recordings, but so too are other recorders. What makes this recorder different is Zoom’s conscious drive towards simplifying its use. Rather than opt for a nice-to-use, very well‑thought‑out UI like the Portastudio X8’s, they’ve replaced many of the parameters normally found on recorders with fixed default values.

For instance, and probably where this all started, removing the need to set recording levels by using only 32-bit float — because of that choice, I’d always leave Rec Source at its default setting of Pre Mixer; who

Audio Examples On the SOS website (https://sosm.ag/zoom-h4e) you’ll find a couple of recordings to accompany this review. Originally 48kHz/32-bit float WAV files, but converted to 48kHz/24-bit WAVs, these were captured using the onboard mics of two recorders, the Zoom H4e and my Tascam Portacapture X8, simultaneously. The originals were made at a concert on 31st May 2024, and the piece, ‘Lament For A Music Teacher’, was

composed by Anne Gregson and performed by Fiery Dragon Company. The only processing on the files was gain, to make them approximately the same volume.

File recorded on the Zoom H4e: Music Teacher on Zoom.wav File recorded on the Tascam Portacapture X8: Music Teacher on Tascam.wav

needs to be mucking about with gain all the time anyway? Then there’s the deliberately simple file system: why should you have to worry about what to call files?

The date and time of the recording are there. What else do you need?

Although I’m now more used to touchscreens, I found the H4e’s simpler user interface OK to work with, and this was greatly helped by there being fewer decisions to make than with most other recorders. Were there the number of options offered by my X8, I think I’d have struggled and it might have proved quite clunky and awkward over time.

But there aren’t, and once you’ve set up the few things you need to the first time, you probably won’t want to do much more. You will, as its makers suggest, “just press record”.

The lack of touchscreen no doubt contributes to the good battery life; the H4e requires only two AA batteries, compared with my Tascam Portacapture X8’s four, and they power it for at least as long (quite a long time if not supplying phantom power for external mics).

The SERIES SlatFusor

Finally, booting up is admirably quick (4-5 seconds).

As someone who’s been recording for quite a while, I do feel happier using a more controllable device. My X8 has, for instance, a manual mode that allows me to tweak lots of parameters, plus goodies like a built‑in compressor and an elegant UI. So I won’t be trading it in for an H4e.

But perhaps that says more about me than the devices themselves — do I really need those features? Arguably not. In fact, there will be plenty of people who actively wish to avoid that kind of messing about, and just want to guarantee they can be up and running quickly. For them, the H4e could be a revelation. Just press record. I suspect Zoom will sell these by the bucketload. They deserve to.

£ H4essential £189. H1essential £95.

H6essential £285. BTA-1 Bluetooth adaptor £38. Prices include VAT. T Sound Service MSL +44(0)20 7118 0133 E sales@soundservice-msl.co.uk W www.soundservice-msl.co.uk W https://zoomcorp.com

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www.soundonsound.com / September 2024

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FE ATURE

Hear The Sound

David Lord • XTC ‘All You Pretty Girls’ J O E M AT E R A

R

espected British record producer and engineer David Lord has manned boards for artists ranging from Peter Gabriel and Tears For Fears to the Korgis and Jean-Michel Jarre, as well as Australian legends Icehouse.

Between 1978 and 1988 he owned and operated Crescent Studios in Bath; now semi-retired, he still works with a select group of artists at his smaller home studio base, Inner State Studio, which houses a selection of the Crescent Studios equipment. Asked to nominate a favourite sound from the many he’s recorded, he chooses XTC’s ‘All You Pretty Girls’, from their 1984 album The Big Express.

Mellow Feeling “XTC had purchased a rather beaten-up Mellotron, sometimes known fondly as ‘the first sampler’. Beloved by

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prog-rockers, and of course the Beatles, it’s an electro-mechanical keyboard instrument in which, simply put, each key pulls a recorded loop of tape across a playback head. Different banks of tapes contained different sounds, for example strings, flutes, choirs and so on.

“We used this on a couple of songs on The Big Express album, which was recorded at my Crescent Studios in 1984, including the ‘choir’ which forms the intro to the song, ‘All You Pretty Girls’ — a rollicking sort of sea shanty, which Andy Partridge told me grew from him ‘dicking about’ with some quasi Hendrix-like guitar licks and a Melos echo unit. ‘A real Nuremberg rally of an echo, the sort of thing Hitler would have had under his podium!’

“Andy wanted to open the track with Mellotron choir samples, but we thought it somehow sounded too real. So, I tried a little trick — nicked from Peter Gabriel — of feeding the recorded Mellotron part into a tiny handheld transistor amp and speaker combo, nicknamed by Peter as the ‘9.95’ after its price in Radio Shack.

It basically de-hi‑fi-ed the sound, which was then miked up and re-recorded.

This got us some of the way there, but then I had the idea of putting the tiny ‘9.95’ into an empty metal waste paper bin and hanging a Shure SM57 over the top, with a cardboard tube from the

W https://open.spotify.com/ track/1qhOSwR4o1Ox9MdFCSNTw7 W www.youtube.com/watch?v=i3qweL-nxCg

middle of a kitchen roll taped to the mic. This coloured and gave the ‘choir’ a much more interesting and unusual sound which, with some added ambience and compression, along with a touch of — I think it was either an Eventide Harmonizer H910 or Roland Dimension D chorus — to ‘stereo-fy’ it, came out sounding a bit like a cartoon Albert Hall!

“Incidentally, though the budget for the whole album and studio, including me, came to £75,000, Virgin Records thought nothing of then spending £33,000 on a video for ‘All You Pretty Girls’. The album’s multitrack masters were then lost for many years, preventing any remixing, but were located in 2022 and recently remixed in immersive Dolby Atmos by Steven Wilson.”

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ON TE ST

Moog Muse Polyphonic Synthesizer

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A Moog polysynth is always a big event, but is the Muse the inspiration you’ve been waiting for?

GORDON REID

T

here was a time in the early 1970s when all synthesizers were called Moogs, no matter which company built them, just as all ballpoint pens were Biros and all vacuum cleaners were Hoovers. But despite being the world’s most recognisable synthesizer brand, Moog have hardly been prolific and, if you list their polyphonic instruments from the past 55 years, you’ll find that there have been just seven, two of which weren’t released, and another of which wasn’t a polysynth at all.

So today I have in front of me Moog’s eighth (or maybe fifth) attempt to capture the hearts and minds of the polysynth market.

I wonder whether it will succeed.

While neither as large nor as heavy as the Moog One (with which it shares some features) the Muse is still a chunky beast weighing in at 14.5kg and, notwithstanding some wobbly faders that hark back to the days of the Rogue, Liberation and Opus III, it feels solid and robust. Its 61-note keyboard offers both velocity and aftertouch sensitivity but neither polyphonic aftertouch, nor MPE, nor the more exotic performance capabilities that recently appeared elsewhere. So, if you want to control this with that while wiggling something else, you might need to look somewhere else.

Despite boasting almost 200 controls, its panel is clear and (in my view) very attractive, and its menus — many of which are accessed using the triangular buttons in the voicing panes — have just a single tier, which helps to keep things simple.

All of the housekeeping is carried out in the Programmer section toward the centre of the panel and, while small, the monochrome screen is adequate if you don’t mind scrolling up and down a bit.

The Signal Path The basic sound generation unit of a Muse sound is the Timbre, and this can be polyphonic, monophonic, or unison monophonic. A patch contains two Timbres and you can switch between these, split them or layer them. The eight voices are then distributed according to your choice from the various voicing modes.

At first glance, a Timbre appears to be based upon a dual-oscillator-per-voice architecture, but the powerful Modulation Oscillator offers an audio mode and tracks

the keyboard perfectly, so triple-oscillator patches are never far away. Oscillators 1 and 2 are based upon those used in the Voyager and, in addition to the standard facilities, they offer sync and bidirectional FM for a wide range of clangourous noises.

The Modulation Oscillator is also based on the Voyager design but generates a different set of waveforms and can be disconnected from the keyboard for constant modulation rates and drones.

You can use it as a modulator in either range and it provides eight simultaneous modulation destinations directly from the control panel. Glide is also provided.

It offers linear constant rate, linear constant time and exponential options, and you can direct it to any selection of the three oscillators as you choose.

The audio mixer lies next in the signal path. We often take such devices for granted, assuming that they sum the signals presented to them without imparting any sonic characteristics, but that’s not the case here. The Muse’s mixer is based upon the Moog CP3 module and mixes the outputs from all three oscillators, the osc 1/osc 2 ring modulator (which was inspired by the Moogerfooger MF-102) and the white‑noise generator. At low levels, it does this without clipping, but you can overload it to distort and fatten the signal. Interestingly, you can create DC offsets in the mixer to cause it to clip more asymmetrically, and this creates a further range of distortions.

Next come two 24dB/octave filters based upon the 904a filter module. The difference between them is that filter 1 has high-pass and low-pass modes, whereas filter 2 is dedicated to low-pass duties.

Both offer the expected facilities, but there are just three keyboard tracking options available from the panel, although you can obtain anything ranging from zero to more than 100 percent using the menus. You can use either or both filters as additional oscillators, which means that you can have up to five sources in a Timbre, even before you invoke the ring modulator and noise.

There are three signal routing options — serial, stereo and parallel — and you can link the filters so that Filter 2’s cutoff knob adjusts both equally, allowing you to use them as a single notch or band-pass filter.

Unfortunately, the Muse has no audio inputs, so you can’t use its filters as external audio processors.

The Muse’s VCAs are based upon the Moog 902 module, but implemented in stereo so that you can pan voices or Timbres within the stereo field. You can even link the phases of applied LFOs to the pan position, which is a novel idea, and the VCA menus offer a gain offset so that you can create drones and other effects.

Following this, the output section offers independent levels for the main audio and headphone outputs, a mute for the main outputs, and a simple 6dB/oct high-pass filter to remove deep bass that might interfere with other instruments in a mix.

Paralleling the path between the VCAs and the outputs section, the Diffusion Delay is a digital effect unit that, in addition to stereo delays and ping-pongs, can create diffuse effects that approximate reverberation. But it’s unable to create choruses and other modulation effects because — at least at first sight — it contains no LFOs to sweep the treated signals. Nonetheless, the manual mentions ‘chorused diffusion’, so I asked the chaps at Moog what was going on. They told me, “The character knob is bipolar. If you turn it clockwise from 12 o’clock you wind up with non-chorused diffusion (ie. the sound is fed through a diffusion network made up of 24 individual delay lines), and if you turn it counter-clockwise you obtain chorused

Moog Muse £2999 pros • It’s a genuine Moog polysynth and it sounds like it.

• The well-designed control panel encourages programming and experimentation.

• It offers good connectivity for both analogue and MIDI studios.

• It looks and feels ‘just right’ — attractive, solid and robust all at the same time.

cons • It offers neither poly aftertouch nor MPE. • It has very limited effects capabilities by modern standards.

• There are no audio inputs.

• Some important features are still in development.

summary I suspect that Moog’s latest polysynth will be their most successful to date.

Despite one or two limitations, it looks great, it’s pleasant to use, it has a big Moog-y sound, and it doesn’t come with an unobtainium price tag. If you’re in the market for an upmarket analogue polysynth, you have to try it.

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The Rear Panel The rear panel sports a good complement of connectors. Starting on the left you’ll find the left‑right quarter-inch output pair.

(The headphones socket is on the front of the instrument, where it should be.) Next to these lie two quarter-inch control pedal inputs, a pair of 3.5mm sockets for analogue clock in and out, and four 3.5mm sockets for CV1 and CV2 in (which are Mod Map sources) and out (which are Mod Map destinations). Digital I/O is provided by 5-pin MIDI in, thru and out, plus USB‑B and USB‑A for backing up and connecting to computers. The final hole is an IEC socket for the internal universal power supply, which is right and proper for an instrument of this size, power and cost.

diffusion where six LFOs are round-robin allocated to modulate the delay times of the 24 delay lines.” These LFOs could in principle offer all manner of opportunities for choruses, flangers and ensemble effects and, when I asked about this, they added, “We will be allowing the delay to be modulated in the modulation map soon. We will have a post-launch update to address this before long.” That sounds promising.

Other settings allow you to adjust the nature of the diffusion, select which Timbres are treated (or not) and whether the resulting effect is sent to the main outputs, the headphones, or both. If you bypass the Diffusion Delay, the audio signal remains in the analogue realm all the way from the oscillators to the outside world.

Modulation At first sight, the Muse’s digitally generated contours appear simple, with a dedicated ADSR (whose default response is modelled on the Moog 911 module) directed to the filter cutoff frequencies and another directed to the VCA gains. However, for each contour, you can choose the shape of the curve for each of the attack, decay and release stages, the amount and curve of the velocity response, and the trigger type in mono and unison modes. In addition, you can determine whether new contours are initiated from zero or from the current value when you press the next key that uses a given voice, and you can loop contours

to create additional cyclic modulators. With the attack and decay set at or near zero, the loop frequency strays into the audio range, which means that your next sci-fi soundtrack need never be far away.

The two digitally generated LFOs offer three standard waveforms, sample & hold, plus your choice of one from 11 complex shapes to insert into a fifth slot. They can be unipolar or bipolar, synchronised to clock, key-sync’ed and applied globally or per-voice. Furthermore, you can determine their ranges individually on a per-patch basis with a maximum of up to 1kHz, which means that yet more AM and FM effects are available, and you can choose different modulation depths for each destination. In addition, there’s a dedicated per-voice pitch LFO optimised for vibrato that can also be used as a ‘one-shot’ AD contour generator.

From the panel, you can assign this to the oscillators, the modulation oscillator and,

controller or both, and 50 destinations.

You can direct multiple sources to a single destination, and a single source to multiple destinations, which is always a good thing. There are six physical controllers that you can use as both sources and controllers — keyboard CV, velocity, aftertouch, the mod wheel, the expression pedal and a macro knob — but there’s no ribbon or touch pad, which is a shame.

Chords, Arpeggios & Sequences Many polysynths have a chord generator that allows it to learn a chord and then play it up and down the keyboard, but the Muse goes further, allowing you to save a different chord under each note.

Philosophically, this is the same principle as the pads on (say) a Korg OASYS when used in the same fashion, but with 61 chord triggers rather than eight, so you can record what you need for some pretty extensive pieces.

You can even use the arpeggiator to play the chords in sequence, which brings us neatly to... ...the Arpeggiator. This is monophonic (or, rather, single-key) and saved on a per-patch basis. From the panel, it appears to be a simple affair with three modes, a maximum four-octave range, a forward/backward switch, and latching.

But when you enter its menu, you’ll find that you can also create patterns of up to 64 steps. You can insert rests, and even choose whether any given step might

“It was straightforward to obtain the forceful and full-bodied sounds that we associate with the large analogue polysynths of the late ’70s and early ’80s.”

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surprisingly, to the amount of detune, which creates a pleasing ensemble effect.

All of these modulators can be used as sources within the modulation matrix.

This offers 16 slots per Timbre that you can populate with your choice from 17 sources (some of which offer both unipolar and bipolar modes), 17 modulation controllers, 14 mathematical transform functions that can affect both the source or the

randomly become a rest. If it does, you can then decide whether the Muse plays no note on that step or skips to the next.

There are controls for looping, octave shifts, pivoting (whether the first and last notes are repeated when the arpeggio changes direction), mirroring lengths of random sequences, alternating patterns, excluding repeated notes, and leapfrogging steps with the forward and backward lengths set to different values. If all of this sounds a bit complex, don’t worry... it is. Many of the parameters interact with one another, leading to unexpected results that may or may not be useful. But if you want to find the gold amongst the mud, you can record the arpeggiator’s output into the Muse’s sequencer to edit and develop it.

Ah yes, the sequencer. This starts out looking fairly conventional, allowing you to determine the length and then enter notes step-by-step or play into it while it’s running. You can also use the Value encoder to enter notes (Moog call this ‘Gesturing’) and things such as velocities and gate lengths. Overdubbing is possible, and you can extend notes across multiple steps if desired. In addition, you can record up to eight control panel values on each step so that, when you replay the sequence, each note or chord might have an individual filter setting, or use a different waveform, or experience different amounts of modulation... or whatever. It offers extensive editing capabilities so that you can fine-tune your sequences. One such facility — and this came as a surprise — is the ability to shift the timing of each note to divisions between the steps. There are 23 subdivisions between every 16th note,

and using these creates far more ‘human’ sequences. But then the really complex stuff starts. In addition to easily understood controls for direction, swing, scale, further quantisation, clock source and division options, and determining which note plays which Timbre, there are numerous parameters that use random probabilities

to add fluctuations to the sequence. These can range from a single randomly changing step, to changes throughout the sequence, to complete uncorrelated mayhem. Then there’s Coin Toss. This adds another set of lanes to the sequence, and you can then determine the probabilities that the sequence will play the notes from the main

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sequence or from the second! Having created a sequence, you can then save it to one of the 256 available locations (16 banks of 16) and, if you wish, lock the current patch to it so that the appropriate sound is loaded when you recall it. You can also create one ‘Sequence Chain’ in each of the 16 banks.

This can contain up to eight sequences for a maximum of 512 steps.

Finally, the dedicated Clock section allows you to determine the clock source — internal (with tap tempo), external analogue, or MIDI Clock over 5-pin or USB — and

The Muse is a substantial instrument, measuring 990 x 420 x 110mm and weighing 14.55kg.

whether the arpeggiator and sequencer respond to MIDI Start/Stop messages.

Interestingly, you can program a MIDI Clock offset of up to 50bpm in either direction.

There must be a reason why Moog did this, but I have no idea why you would want the Muse running at (say) 103bpm when fed a 100bpm clock! You can also choose whether a clock pulse is output each time that the arpeggiator produces a note. Given that you can program rests

in your arpeggios, this means that you can cause external equipment to mirror more sophisticated patterns than might otherwise be possible.

In Use Unreleased synths often require firmware updates during a review because they tend to evolve as launch day approaches.

So, upon receiving the Muse, I contacted Moog to ask whether the installed version was still current. To my surprise it was, although the engineers told me that there

A Brief History Of Moog’s Polyphonic Synthesizers A year before the appearance of commercially available polysynths, there was the Apollo.

Built in 1973 as part of the Moog Constellation system, this was the first instrument of its kind, but only one was ever seen, played by Keith Emerson on ELP’s Brain Salad Surgery tour. It offered just a single oscillator per voice and, crucially, its contour generators had no sustain phase, which meant that it was closer in concept to a piano than an organ or monosynth.

Nonetheless, it introduced many concepts that would later be adopted elsewhere, including individual voicing circuits for each key and preset voices that could be modified by the player.

The lessons learned from the Apollo underpinned the development of the Polymoog (later renamed the Polymoog Synthesiser), which was released in 1975. This offered a wider keyboard, velocity sensitivity, a number of modifiable presets, and one user-programmable patch. However, its oscillators used the ‘top‑octave divide down’ technology from combo organs and string synths, and its VCF section comprised just a single filter, making users’ sounds paraphonic rather than polyphonic.

Nonetheless, it was used by many top-tier bands including Genesis, Yes and Abba.

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Moog built two further prototypes called the Apollo. These were very different from Emerson’s instrument and, in 1978, the product derived from them was launched as the Polymoog Keyboard. This offered just 14 preset voices and, since it was launched the same year as the first fully programmable polysynths, it was inevitable that it wouldn’t be widely adopted. However, it included one sound that has since became part of synthesizer folklore; the Vox Humana. This resulted in it becoming far more sought-after than its meagre facilities might have suggested.

Moog discontinued the Polymoogs in 1980 and replaced them with something that was not a true polysynth at all. The Opus 3 was a multi‑keyboard with a single divide-down oscillator bank and three treatments that generated its strings, organ and paraphonic brass timbres.

A range of interesting sounds could be coaxed from it, but it was never going to be a world-beater.

Two years later, Moog finally released their first, fully programmable polysynth. The Memorymoog looked in every way like a serious, top‑of‑the‑range instrument, and it could sound glorious. Unfortunately, affordable polysynths had already started to appear and then, in 1983,

digital synths were In and analogue synths were Out. The release of the Memorymoog Plus did little to arrest its slide and, despite being adopted by the likes of Jean-Michel Jarre, Jan Hammer and Nick Rhodes, the Memorymoog was discontinued in 1985.

Like their first, Moog’s next polysynth was never released. The SL-8 was designed to be cheaper to manufacture so that it could compete with low-cost polysynths and, when demonstrated in 1983, it garnered significant interest. Verbal orders for hundreds of units were taken, but the Yamaha DX7 was introduced at the same time, so the financial controllers at Moog Music refused to allow it to enter production.

Following the bankruptcy of Moog Music in 1987, four decades were to pass before the reborn company dipped its toes back into the murky waters of polyphonic synthesis. When it appeared in 2018, the One was heralded as the successor to the Memorymoog and, in many ways, that was right. Large, heavy, feature-laden and capable, it split the synthesizer community into those who adored it and those who couldn’t see what the fuss was all about. Despite its high price, it remained in production until 2024, when it was replaced by the Muse.

is a new one in preparation that should be ready for the launch which, at the time of writing, is still a couple of weeks away.

Having allowed the Muse to warm up, I wasn’t surprised to find that it had arrived out of tune, so I ran the Quick Tune procedure. Things were still a little out of kilter, so I initiated the full tuning and calibration routine. (Don’t do this immediately before a gig — it takes almost an hour.) Everything was then fine and although further Quick Tunes were necessary after switching on and during long sessions, they were sufficient to keep everything in order.

I was pleased to find the pitch-bend and modulation wheels in their rightful places, and I liked the control panel, which has the right amount of variation and colour to break up what might otherwise have been a bit impenetrable. The chaps at Moog also pointed me toward a parameter in the menus called Show Param Changes, which allows you to see the saved and current values of the parameter that you’re editing, and this proved to be very helpful.

But what I didn’t like was the keybed —

or, to be more precise, its response to aftertouch, which felt jerky and abrupt.

So, whenever I programmed a destination for this, I added Slew as the Function Source in the Mod Map to smooth the response and make it more musical.

I created some typical analogue sounds using single Timbres: brasses, solo and ensemble strings, pads, pipe organs, basses, leads, and even a few Clavinets.

As I had expected, it was straightforward to obtain the forceful and full-bodied sounds that we associate with the large analogue polysynths of the late ’70s and early ’80s.

This shouldn’t surprise anybody; the Muse is, after all, a Moog based in large part on vintage Moog technology. But, even with the gains in the mixer backed right off, it’s not what I would call naturally smooth or silky, and it didn’t take kindly to being asked to do things that it would rather not.

To investigate this further, I viewed its output on a signal analyser. With no overdrive, the filters wide open and the VCAs set to moderate levels, the raw waveforms proved to be spikey no matter what the mixer levels were. What’s more,

when listening to them through high‑quality speakers and expensive headphones, the tone was audibly crunchy. Using the Diffusion Delay and — given the paucity of onboard effects — hooking the Muse up to some external effects units smoothed things out and increased the range of sounds that I could obtain. At one point, I stumbled upon a family of ethereal pads that I’ve loved since the 1970s, but the Muse is far from all-encompassing, and there are some sounds that you’ll obtain more easily elsewhere. Don’t misunderstand me... there’s nothing wrong with that. Synths have characters, and it would be daft to want the Muse to sound exactly like a Prophet, or an OB-something, or a PolyBrute, or whatever. Just make sure that it has the character that you want.

Since every Muse patch contains two Timbres, I moved on to creating some splits, and this was where I encountered my first serious concern. If you import existing Timbres into a split patch, it’s likely that you’ll want to be able to shift their octaves to play them at sensible pitches. I could find no way to do this, so I contacted Moog’s

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engineers again to ask The small triangular buttons in each section whether I was being are shortcuts to the a numpty. I wasn’t. relevant menu pages.

They told me, “This feature is not present in the current firmware but will be addressed in coming updates.

For now, it would need to be handled via the oscillator controls, or through Mod Map routings.” I didn’t bother to point out that shifting the oscillators up or down by an octave or two is very different from shifting the whole Timbre up or down by the same amount.

Next, I created some patches with Timbre A directed 100 percent to the left output and Timbre B directed 100 percent to the right, and played them as independent synths, even applying the left-hand delay to Timbre A and the right-hand delay to Timbre B. Sure, this means that you can’t use the panning features in the VCA another serious shortcoming. I spent half section, nor can you use the Diffusion an evening creating a complex setup with Delay to spread the results across the chords distributed right across the keyboard stereo field, but I found this to be a more and then looked for a way to save this. than acceptable compromise, especially I couldn’t find one, so I hoped that the map since I was able to apply separate external would be retained as a global setup. But effects to each output. when I power cycled, all of my work was I also layered the Timbres to obtain lost. This renders the chord/key capability some composite sounds. These often almost useless, so I again contacted the sounded excellent, but bear in mind that this chaps at Moog. To my great relief they told reduces the polyphony to just four voices. me that, “The chord map will be non-volatile It was now time to take a deep breath with eight available maps you can store, before diving into the complexities of but that’s not currently implemented. the arpeggiator and sequencer. To be We’re working on it.” Phew! honest, it would have taken far longer Next, I hooked the Muse up to my MIDI than a pre-release review to explore the studio to see whether it played nicely with possibilities that they offer. Given all of the other children. the Muse’s connectivity options (and, in It did. Its MIDI specification includes an particular, its use of 3.5mm connectors) extensive range of MIDI CCs that allow I suspect that Moog are encouraging us to you to control other equipment as well as see it as the centrepiece of an experimental automate the Muse itself, and there’s even electronic music studio and, if used in this a Multi mode that allows you to control the way, I think that many of their features will two Timbres in each patch using separate make a great deal of sense. However, it MIDI channels. You can also choose was while I was combining the arpeggiator different MIDI output channels for the and the Chord mode that I discovered keyboard and MIDI CCs (on the one hand)

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and the pitch-bend and modulation wheels (on the other).

That’s odd, but interesting.

Finally, I hooked it up to my MacBook Pro and powered up the synth in its Disk mode, whereupon it appeared as an external drive on the Mac and gave me access to all of its 256 patch and 256 sequence memories.

I could now move and rename them, build new banks and back up the results. Power cycling then returned the Muse to its usual operation with all of the changes made.

Final Musings The Muse will evolve further after I’ve submitted this review, but some things are already clear: it was stable and well-behaved throughout the review, it’s pleasant to use, it has that big Moog-y character that many of us love, it’s superb at generating some sounds but less so at others, and it will take time to learn how to use it to its full potential. Of course, no synth is perfect, but none of the shortcomings I’ve mentioned are dealbreakers, especially since one or two of them might be history by the time that you read this.

So, finally, we reach the issue of its price, and I think that there are two ways to look at this. On the one hand, it’s expensive.

Although £3000 is half the price of the equivalent Moog One, nobody in 2024 would blame you for wanting more voices, poly aftertouch and even MPE for this much money. On the other hand, it’s cheap. For £3000, you’re getting the genuine Moog sound for £375 per voice, which is a fraction of what it would cost if you were to try to obtain it by other methods. Would I buy one? I would have to think about it. Would I use one? Of course I would. £ £2999 including VAT. W www.moogmusic.com

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TECHNIQUE

The Tape Mindset

Steve Osborne.

An Analogue Approach To Our Digital World

Working with analogue tape wasn’t all good, but it did impose a discipline on recording sessions that was hugely beneficial — and which our DAWs have made optional.

NEIL ROGERS

A

recent chat with producer Steve Osborne (New Order, U2, Elbow, Doves, Happy Mondays and more) prompted this article. We were discussing techniques he’s carried over from the days of tracking bands to analogue tape to his current DAW-based work, and it struck me that, despite advances in technology, the basics of his approach haven’t really changed that much. It seemed to be a conscious choice too: he sees real benefit in the approaches that working with tape imposed on both engineers and artists, and he’s very clear about the importance of making and committing to creative decisions as you go.

I’ve recorded to tape many times, as an engineer and a drummer, but when I started as an engineer it was already yesterday’s technology. Steve, on the other hand, was working on big sessions when tape still ruled the roost; he has vastly more

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experience with it than I do, especially when it comes to higher-end commercial sessions. So, wanting to explore this in more detail, we had further chats and in this article — which is less of an ‘interview’ and more a co-authored feature, really, with lots of input from Steve — I’ll discuss the ways in which DAW-based engineers and artists might benefit from getting into a bit more of a ‘tape mindset’.

Ready To Roll?

You probably know something about analogue tape: its sonic qualities, the limited track counts or even that edits entailed using a razor blade. But something Steve stressed that’s often overlooked is that working with tape imposed a different mindset on a session, right from the very start. Engineers needed to ensure that the machine was properly aligned and biased.

The tapes had to be organised, ready to be put into action quickly. On every session, someone needed to operate the tape

machine — not just hitting Play, Record or Rewind, but also loading the reels and locating sections of a song quickly for ‘punch-ins’. They needed to know what was recorded on which tracks, and which tracks to arm for the next take. Filling out track sheets, keeping notes... the list goes on.

Steve reckons this was all really important in creating a sense of focus and attention. For any session to run smoothly, everyone in the production team had to be present and actively engaged most of the time. Today’s tech gives us so many ‘safety nets’ that we risk losing that sort of focus and control, which means we need a good dose of self-discipline!

Track Counts Back in the days of tape, there were significant technical limitations compared with our DAW-based setups, and one was the number of tracks available to work with.

There were mixing desks with vast numbers of inputs, but the tape machines would

have only 16 to 24 tracks, at most. In most recording sessions, therefore, you were limited to recording to 16 or 24 tracks (fewer in some smaller studios), and you might have to reserve a track for time code too.

It was possible to synchronise two machines for more tracks, but good, low-noise tape machines were expensive, it used twice as much tape, and this wasn’t really a desirable or very practical way of working.

All this had a huge influence on how studio sessions were run. It was necessary to mix some parts together and record them to the same tape track, and choosing what should be mixed (there was no ‘undo’!) influenced other decisions: what needed to be recorded when, for instance. Even with the best-laid plans, a need to add more parts would often arise, and to free up the necessary tracks you had to ‘bounce down’ multiple recorded parts to the last one or two remaining tracks. Again, such decisions were destructive, and the more times you bounced things down the more quality you’d sacrifice. So, great care had to be taken that sounds being committed to tape were in phase and generally as close to a ‘finished’ sound as possible.

In short, it was necessary to plan ahead and to commit to lots of decisions as you went along that, today, could seem tempting to put off. DAWs offer us unlimited tracks if we want them, but while that freedom can be a good thing, it creates a risk: because we don’t need to plan ahead, it can be tempting to keep our options open ‘just in case’ — in which case, our projects can start to drift along a bit aimlessly.

You don’t have to stick to 16 or 24 tracks, but planning a project as if you do can be very helpful. It ensures you’ve really thought about what a project should sound like, and that will help you to remain

Tape machines needed maintaining, calibrating and operating throughout a session, all of which can be viewed as chores. But such jobs ensured everyone on a session was constantly focused on the task in hand.

more focused throughout the project. Also, when it comes to mixdown and you ‘pull up the faders’, you’ll probably find the mix is already halfway there, with the sounds already working together and being at roughly the right level. So mixing becomes quicker and, because you’re freer to focus on the vibe and the creative side of mixing, much more fun.

Takes, Drop Ins & Edits Our DAW software gives us powerful editing tools, but basic editing was still possible with tape recordings. A competent engineer could cut tape and splice different bits together, whether to create a comp from different takes or to make arrangement changes.

A good engineer could

even chop out problem noises, duff notes or drum hits. There was a whole different level of jeopardy and time involved, even if Steve reckons wielding the razor blade wasn’t nearly as terrifying as it might sound (you don’t miss what you’ve never had).

But Steve reckons that, presumably because neither engineers nor performers thought of editing as a safety net, major mistakes were rare. Engineers like him, who cut their engineering teeth in the world of analogue recording, had to be musically engaged in a session, not just technically. They needed to know if there were any issues with that last take, be able to communicate with the artists about that, and be ready to manually ‘punch in’ (record over a specific section of a recorded part), with limited visual cues. When I talk to them about sessions in those days, they often describe a more collaborative environment that was more fun and rewarding than when you’re glued to a computer screen, poring over the endless options for ‘perfecting’ that last take.

Artists & Performances Artists and musicians were used to the same way of working. Before I was an engineer, I was a drummer, and I can clearly remember when recording to tape the focus being on getting a good, full take, before perhaps dropping in a few ‘fills’ or ‘frills’.

Editing was a pain (there was certainly no

Tape machines had 16 or 24 tracks at most, so careful session planning and record keeping was a necessity. Today, we’re free from such limitations — but that means we’re also faced with the risk of becoming lazy in our preparation and decision making.

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WOR KING W IT H A TA PE MIN DS E T

Editing tape recordings was possible, but it was a fiddly and time-consuming job — so the focus was very much on getting three or four good takes and comping those into one good performance during the recording session. No matter how powerful the comping options available in our DAWs, we can still approach sessions with a similar mindset.

Beat Detective or Elastic Audio to fall back on!), and knowing before entering the studio that I’d be expected to produce the goods focused my mind when preparing and rehearsing for a session.

Frankly, I don’t always see this level of preparation by the bands coming into my studio today. The knowledge that editing is easy and tuning and timing enhancements are readily available can sometimes seem to make artists reluctant to put in the hard yards that lead to the best results. I spent many hours, often ‘off the clock’, comping and editing drum takes to give the illusion of a tight, rehearsed drum performance.

It’s possible to achieve decent results like that, but it takes a lot of time and can sap your energy — I’ve often wondered quite seriously whether I’m actually doing a young musician any favours!

Perhaps the biggest change, though, is our ability today to record and keep endless takes, then choose the best bits to create a perfect ‘comp’. Guitarists have always wanted to spend six months perfecting one solo, of course, but it’s

probably when it comes to vocals that we’ve seen the greatest difference. Many engineers who still sometimes work on tape today will concede that they eventually return to Pro Tools when it comes time to do the ‘proper’ vocals. It’s understandable: the convenience of working with playlists and ‘on-the-fly editing’ is tough to give up once you’re used to working that way, especially for the artist. But if we’re not careful, this approach can breed a ‘kick the can down the road’ mentality, where we continually defer decisions.

I asked Steve to describe his vocal sessions when recording to tape, and he mentioned having maybe four tracks available on the tape machine for a series of takes. After a few takes, he’d get the singer into the control room and comp the best parts of those takes — one track would become the ‘comp’, and the remaining three could be reused to repeat the process, if necessary, until they got to a point where there was a finished vocal take on the reel. Another case of decisions being made as you go.

There’s no reason why we can’t work like this in our DAWs but, again, with more options available, temptation can easily lead us astray. I know some engineers actually enjoy comping a vocal from 20 takes but personally, I reckon returning to a project knowing I still have to establish what the finished vocal track sounds like is a real buzzkill — especially if it’s an EP or album, not just one track!

Also, it’s sometimes only when you return to it, that you realise you didn’t quite get a particular line or phrase in the bag, and might have to get the singer back in...

As engineers there can be another voice in our heads suggesting our lives might be easier if we record just a few more takes and work on the comping later, free from the pressure of time or the singer’s presence.

Steve made a simple but pertinent point about this: in pursuit of getting the best possible takes and performances, an engineer shouldn’t be afraid to be authoritative or even a bit ‘difficult’ in a recording session. Many musicians can quickly enter a mindset in the studio that leads their own standards to drop when they feel frustrated. It’s our job to push, cajole and encourage people to deliver the very best performances they can. How best to approach this is another article in itself, but shying away from this side of an engineer or producer’s job is perhaps the worst ‘lazy way out’ that recording digitally has enabled. It’s also a false economy, You don’t have to let the computer dictate your approach to a session! Left: For the Happy Mondays single ‘Step On’, the production team recorded live drums to tape without a click, and then sampled the bits with the best feel to create the loops that underpin the song. Right: Most DAWs include tools that allow the grid to be adjusted to match live performances, making a similar approach easy today.

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TECHNIQUE

WOR KING W IT H A TA PE MIN DS E T

(limited) ability to nudge the pitch of problem notes, but we now have tools that can automatically quantise the pitch of every note, and even iron out natural variation and vibrato within notes. This can be a great way to remove all a singer’s feel and nuance!

Unless you’re going for a specific ‘tuned’ aesthetic, it’s almost always better to take your time to appraise the performance before reaching for the pitch processing — focus on the sound and feel, and change only what needs changing (and when you make changes, ensure the result doesn’t sacrifice any of the vibe).

Best Of Both Worlds?

Knowing that they were expected to produce the goods on a tape-based session, performers typically rehearsed and arrived well prepared. Many still do... but the knowledge that performances can be ‘fixed’ can also lead to a very different frame of mind!

because recording a bunch of takes to put together later creates work for another day — a day we’re probably not getting paid for!

Perfect Imperfections Our DAWs give us access to powerful pitch‑ and time-manipulation tools and, as with edits, it can be tempting to use them to create the illusion of a ‘perfect’ performance — often with just a few mouse clicks. But there’s a risk that in ironing out every imperfection, you sacrifice something important in terms of feel, particularly when you use tools that promise to achieve this automatically.

When deciding how to approach a ‘problem’ during a recording session, Steve, like most successful producers I’ve talked with, prioritises ‘vibe’ above all else.

For example, he explained to me how the production team approached the sampled drums for the Happy Mondays’ hit single ‘Step On’. Working on analogue tape, they spent a lot of time choosing the best bits of

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longer sections of the live drum takes that had the best ‘feel’, and then created loops from them using an Akai S1000 sampler, and he suggested that this sort of approach — looking for the right energy in a live-played part — can be a better tactic for ‘enhancing’ drums than using DAW-based tools like Beat Detective to pull everything into perfect time. He’s not against embracing what our DAWs have to offer, though, and pointed out that many people don’t fully understand the tools they offer us. It’s easy to assume, for example, that because a DAW project has a grid, your recordings should be aligned to it, but most DAWs also enable you to adapt the grid to match the changing tempo of a live performance recorded without a click.

So we can choose not to be a slave to the metronome — just as we might have done years ago, but without sacrificing any of the control our DAWs give us over other aspects of a production.

The same applies to pitch-correction.

In the ’80s and ’90s, samplers gave us the

Hopefully we’ve given you some idea of how technology, whether new or old, can shape our approach to not only the technical aspects of recording, but also creative and psychological ones. But where does all this lead those of us working in the box today? Most of the technical limitations of tape recording, in terms of channel counts and our ability to edit recordings, are long gone. We don’t need to mix things together on the fly any more, and we have the ability to hit undo or save and return to multiple versions of a project. The positives are plain to see, and I benefit daily from the ability to quickly jump between different projects — I have no plans for my studio to acquire a real tape machine anytime soon!

But I want to finish this article by suggesting a few ways in which we might all benefit from retaining at least some aspects of analogue tape-style recording sessions.

It’s Not All About The Computer!

Try not to allow your DAW to become the focus of a recording session. When recording bands, my DAW is often nothing more a glorified tape machine with some clever tools that I can pull out of the bag if needed. The extent to which you can do that will vary according to genre, but I’ve worked on enough projects to understand that I can bring the most value to sessions as an engineer by engaging with the artist, rather than being a computer programmer.

If you struggle with this, you might need help to shift your mindset — even a small thing like tucking the computer screen off to the side slightly can help you keep your focus in the right place.

There’s also an element of health and self-preservation here! Steve and I discussed just how easy it is for a modern engineer, especially one who does a lot of mixing and programming, to lead

a fairly solitary, sedentary work life based around a computer screen. How one counters this is a very personal thing, but I find that interacting with musicians and other engineer types creates better music, and makes me a happier engineer.

And if I’m working on an 8-10 hour session with minimal breaks, it’s essential for my wellbeing that a decent chunk of that time is spent standing up, and not looking at a screen!

Trust In The You Of Now The above heading is one of the Oblique Strategies, Brian Eno and Peter Schmidt’s series of random choice playing cards, intended as a fun way to break creative blocks. It applies well here: as producers and engineers, we should trust ourselves to be brave in our decision-making. After all, why should our taste or judgment necessarily be any better on another day?

There are counter arguments, but in an age where it’s more important than ever to be efficient with our time, committing to decisions and making progress rather than putting them off is a win. I’m sure

that many SOS readers can relate to the creative paralysis sometimes caused by a continual lack of commitment when working on projects at home!

The Enemy Of Art “The enemy of art is the absence of limitations,” according to legendary filmmaker Orson Welles. It can be hard to create arbitrary, artificial ones where they don’t actually exist, but plenty of artists do just that, because they believe it encourages originality and creativity.

If you’re in a position literally to record a project to tape, it would immediately push you into a different way of working, but this won’t be a practical option for most of us. So I’d suggest simply making a conscious effort to step back and actively decide on the priorities of each song or production you work on. Where it’s appropriate to chase a more ‘feel’ based original sound, try leaning into the tape mindset a bit more — it could well help you achieve your goals. As Steve pointed out, although any such rules we set ourselves can be easily broken,

it’s about creating a framework. When recording vocals his way today, we might end up with six vocal takes instead of four — but at least it’s not 20!

Final Thoughts I’m not one of those engineers who laments about ‘the good old days’ — I always look to embrace the possibilities new technology opens up. But I do think technology has made it too easy to forget just what the tape mindset did for recording sessions. It’s all about conscious and timely decision-making, whether about the creative aims of a project, the performances themselves, or whether/how to use the incredible tools we now have available to us. If we can encourage ourselves to commit to these little decisions throughout a project, we’ll keep our focus on what’s best for the art, rather than what’s easiest or most convenient for those making it. We can get a better feel for when to respect the performance for what it is, and when that fancy new AI tool we just installed can make a positive contribution. We really can have the best of both worlds!

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Icon Pro Audio/ Harrison 32Ci USB Audio Interface

Boring preamps have long been a given in audio interfaces, but all that is about to change...

SAM INGLIS

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udio interfaces have had mic preamps built into them for almost as long as there have been audio interfaces. But for many years, it seemed as though manufacturers had signed a secret pact to make their preamps as boring as possible. Interface designers were slow to wake up to the fact that not every user wanted a completely transparent, neutral input stage, and that in fact many people were spending really rather a lot of money to augment their transparent, neutral input stages with third-party mic preamps that offered a bit of character.

In the last few years, we’ve seen a sprinkling of interfaces with transformer-balanced preamps that can add a little mojo to proceedings, including Steinberg’s UR-RT models, Neve’s 88M and the forthcoming Heritage Audio i73 Pro. It’s still a surprisingly sparsely populated market sector, though, so the emergence of a new contender is very pleasing.

That contender is a collaboration between a relatively young company, Icon Pro Audio, and veteran console designers Harrison. In some ways, it’s a fairly conventional USB audio interface, but in place of the usual boring mic preamps, it employs the input stage from Harrison’s 32-series consoles. So, as well as heritage preamps, it has an unusually extensive filter implementation featuring high- and low-pass filters that can be swept right through the midrange. And, as we’ll see, it also has a few other unusual qualities.

Short Back & Sides The 32Ci is a desktop device with a gently raked, squarish case measuring nearly nine inches by 10. The controls are housed on the top, the two headphone sockets are at the front, and all the other I/O is located on the slightly deeper rear panel. Build quality and industrial design are impressive, and there’s a real sense of class to the 32Ci that belies its competitive pricing.

One reason why boring preamps are the norm in audio interfaces concerns power requirements. As well as offering good specifications, modern off-the-shelf,

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chip-based designs are very efficient, which is an especially important consideration in bus-powered devices. If Icon Pro Audio had taken that route with the 32Ci, I’m sure they’d have been able to make it powered over USB, but the inclusion of the Harrison preamps necessitates an external PSU. The unit also gets noticeably warm in use, though thankfully there are no fans to kick in and ruin your best takes.

All of the 32Ci’s controls seem to use conventional pots, and there is nary an endless encoder to be seen. The master volume control is flanked by 12 LEDs which act as a pair of six-segment meters.

The two interesting preamps are fed from combi XLR/jack sockets, and can be switched into a high-impedance mode to accommodate signals from electric guitars and basses. There’s an additional pair of inputs on quarter-inch jacks; depending on the setting of a rear-panel button, these can operate as balanced line-level inputs accepting levels up to +24dBu, or as unbalanced ‘consumer’ inputs optimised for the -10dBV standard. On the output side, there are two pairs of quarter-inch jacks in addition to the aforementioned headphone outputs. These are configured in a slightly unusual arrangement, which I’ll return to in a moment.

There’s also digital I/O, comprising a single ADAT optical in and out, MIDI in and out, and two USB Type‑C sockets. One of these is used to connect the 32Ci to the host computer, while the other is labelled OTG and is intended for streaming audio from the 32Ci to a phone or tablet. If you use an Apple device, you’ll need the official Camera Connection Kit or similar rather than a generic USB cable.

Buffering The Hits Basic audio interfaces often handle cue mixing and low-latency monitoring using only hardware controls. As the I/O count

mounts up, though, this gets impractical, and so more complex interfaces have digital mixing and routing facilities, configured using a software control panel.

The 32Ci is the only interface I’ve come across that falls into both camps at once.

How so? Well, inasmuch as there is mixing and routing functionality built into the 32Ci, it’s all-analogue and controlled from hardware. However, you can also install a macOS and Windows program called IO Pro. This looks similar to the mixer utilities supplied with many other interfaces, but instead of adjusting settings on DSP in the interface, it runs entirely natively. It’s perhaps best thought of as a host-based alternative to your DAW’s own mixer window that runs in parallel with your DAW. The main Core Audio or ASIO driver presents a total of 14 physical I/O paths to and from your DAW at base sample rates (four analogue plus stereo OTG and eight ADAT), while a second ‘loopback’ driver presents four stereo virtual I/O paths, if selected. The 32Ci itself is blissfully unaware of the virtual I/O, but both sets appear in the IO Pro mixer. On macOS, therefore, the IO Pro mixer has a total of 12 mono and five stereo input channels; things get more complex on Windows, thanks to the need to integrate ASIO and WDM paths.

There’s also a stereo master output and four stereo aux busses, each of which can be returned either to the master output or direct to a physical or virtual output path. Pressing the button labelled Mon adds what is in effect a second master channel that can be routed to a different physical output. This monitor channel has an undocumented pop-up labelled Direct, which presents a 4x4 matrix with sliders, and appears to do nothing whatsoever in my system. Perhaps it has a function in the Windows version.

Icon Pro Audio/Harrison 32Ci £429 pros • Stylish design and construction.

• Good sound quality and specifications.

• Surprisingly creative and useful filtering options on the inputs.

• OTG port for integrating phones and other USB devices.

• IO Pro mixing utility provides versatile virtual devices and routing options.

• Good value.

cons • Built-in low-latency monitoring is quite basic.

• IO Pro needs further development in some areas.

• As IO Pro is host-based, you might as well use your DAW’s mixer unless you need its virtual I/O. • No control over or information available about ADAT clocking.

• Some important topics aren’t covered in the documentation.

summary The 32Ci has some frustrations, but its pedigree preamps, funky filtering options and novel blend of software and hardware cue mixing make for something refreshingly different.

Clocking arrangements concerning the 32Ci and IO Pro are opaque. On macOS, Audio MIDI Setup allows you to select a clock source, but this only seems to relate to its relationship with the loopback driver and other Core Audio devices, and ADAT does not appear as an option. My tests suggested that if the 32Ci receives a valid ADAT signal at the sample rate your DAW is using, it’ll automatically clock to that and make the ADAT inputs available to record. If no ADAT source is connected, or there’s a sample rate mismatch, the 32Ci reverts to its internal clock and mutes the ADAT inputs. However, there is no visual indication at all to tell you

Beginning at the left of the rear panel we find a USB‑C port for computer connection and a second USB‑C port labelled OTG for interfacing with a phone or tablet.

Moving on, there are ADAT I/O ports, full size MIDI I/O sockets, quarter-inch main and ‘Alt’ outputs, quarter-inch line inputs and a pair of combi inputs. A pair of headphone ports is found at the front of the unit.

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ICON PRO AUDIO/HARRISON 32CI

what clock source is currently being used, and nor is there any useful information in the manual on the topic.

Generous Host Because IO Pro runs natively, it’s able to act as a plug-in host. The input channels, auxes and master bus all have insert slots where plug-ins can be inserted in VST2, VST3 or Audio Units formats. It also has its own buffer size setting, which is independent of the buffer size set by the DAW; so you could, for example, run IO Pro at a low buffer size and restrict yourself to using plug-ins that don’t use lookahead, whilst your DAW operates at a higher buffer size and with no such restrictions.

The IO Pro installer also installs several plug-ins designed specifically for working in this environment. These include a simple reverb called iVerb, a grid-based one-shot sample player called Stingers (which I couldn’t get to work) and, most surprisingly, a full-blown modular signal processing and routing environment called Patch. This is itself a plug-in host and chainer, and allows you to set up labyrinthine audio and MIDI routing paths.

All in all, though, IO Pro is a bit of an odd duck. It allows you to do some quite complex stuff that could be very handy for particular use cases such as live streaming or podcasting. Yet, at the same time, it’s frustratingly lacking in basic features. For example, it doesn’t seem to be possible to link two analogue or ADAT channels together to make a stereo channel, and I couldn’t find any simple way to reset pan controls to their default positions. DAW playback bypasses IO Pro altogether,

as a consequence of which there is no monitor control on offer there. There’s no solo/PFL architecture, and each channel has only a single pre‑/post-fader button, which applies to all four aux sends on that channel. If you have a pointing device that supports sideways scrolling, such as the Apple Magic Mouse, you might expect this to scroll sideways across the mixer channels, but instead it does exactly the same as vertical scrolling, namely moving the nearest fader up and down.

Yes There Are Two Paths You Can Monitor By Assuming you’re not streaming live or casting pods, though, you always have the option simply to ignore IO Pro. Mac users don’t even need to install it, in fact, as the 32Ci is class compliant. The routing and monitoring facilities baked into the 32Ci hardware will be available regardless.

There are two built-in monitoring paths, one feeding the main outputs and the first pair of headphones, and the other feeding the second headphone output. Each of these paths has a potentiometer that allows you to set a balance between DAW playback and input signals from the mic preamps and line inputs. There’s no way to adjust the relative levels of the different inputs in the low-latency monitor feed, and the 32Ci doesn’t offer any options as to where they’re panned: the two preamps are always heard centrally in mono, whilst the line ins are hard-panned.

Both pairs of rear-panel analogue outputs are tied to the first of the two monitor paths. The first pair is intended for connection to speakers or other devices

IO Pro is not a ‘control panel’ but a fully native software mixer.

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that expect a ‘professional’ signal level, and can deliver up to +24dBu for a 0dBFS signal. The second pair shares the same master volume control and carries the same audio signal, but at a less hot level intended for consumer devices, and are designated ‘alternate’ outputs. Under normal circumstances, the main outputs are active and the alternate outs muted; pressing the Alternate button reverses this situation.

This is an arrangement that might suit you perfectly, if for example you want to augment a pair of conventional studio monitors with a consumer grotbox for A/B mix checking, but equally, it could easily become limiting. There’s no way to alter the fixed level offset between main and alternate outputs, nor any way to address the latter independently from your DAW.

Character References As I’ve already mentioned, one of the reasons why manufacturers often use boring off-the-shelf chip-based preamps in their interfaces is that their specifications are usually better than those of older console preamp designs. If what you’re after is a more interesting and colourful sound, it’s likely to be precisely the extra distortion and non-linearity in the older design that is responsible for its character.

Perhaps excusably, then, Icon and Harrison have listed specifications for the preamp and for the A-D conversion independently, even though there’s no way to use the one without the other. The converter specs are beyond reproach, with a dynamic range of 125dB available on the input side and a humongous 129dB on the output side. The preamp specs are less

satisfying, partly because there are no measurements given for equivalent input noise or distortion, and partly because they’re copied from the 32Classic console, with references to a non-existent fader being at unity.

The non-existence of this fader is germane to the question of character. Many standalone rackmounting or 500-series preamps offer an output attenuator, allowing the input gain to be driven harder without overloading downstream devices. Others offer insert points, where a compressor or other processor can be patched in to offer control over the level of the preamplified signal. The 32Ci has neither, so there’s no way to manipulate the amount of ‘hair’ the preamp is adding to your signal independently of the level feeding the A-D converter. And as it happens, it’s not obvious what part of the circuit might contribute desirable distortion anyway.

Transformations Inasmuch as mic preamps add character, it is usually because they are based around components such as transformers or valves that can behave in a non-linear fashion. Back in the ’70s, Harrison consoles went through at least three major design revisions. As far as I can discover, the first two iterations used a Jensen input transformer designated the JT-115K-E. Thanks to its 10:1 turns ratio, this provided 20dB of voltage gain and thus formed a key part of the preamp circuit. Then, in the third and most desirable iteration of the console, Harrison redesigned the input stage to be capable of transformerless operation: a 1:1 transformer could optionally be fitted in front of this, but wasn’t necessary and didn’t carry out any actual amplification.

Now part of the Audiotonix group, Harrison recently introduced an updated 32-series console called the 32Classic, and have since announced a 500-series reissue of the 32C preamp. As these use a Jensen JT-MB-CPCA transformer with a 1:1 turns ratio, they are presumably based on the third revision of the circuit.

At launch, Icon Pro Audio’s website stated that the 32Ci’s preamps were

frequencies. For example, I often find that DI’ed bass guitar can sound a bit too heavy below 100Hz or so; engaging Bump can trim that excess weight whilst adding more substance in the frequency range an octave up that’s audible on small speakers.

Summing Up The 32Ci is an unusual product, and not only because it bucks the trend for interfaces to have boring preamps. Even after using it for several weeks, I’m not entirely sure what to make of it.

Let’s start by getting the negatives out of the way. The way Icon have The 32Ci measures 291 x 243 x 92mm and weighs 1.83kg. augmented hardware direct monitoring with the separate IO Pro likewise “equipped with high-quality utility is distinctive, but the former is rather transformers”, but there obviously isn’t basic, and the latter feels undercooked space for them within its compact shell, and at present. The two don’t work together Icon confirmed to me that this was an error. in any way, and unless you need the So, presumably, they too have used the last loopback functions, there is little to be revision of the preamp circuit, but without gained by monitoring through IO Pro rather the optional transformer. Consequently, than your DAW’s own mixer. Elsewhere, it’s not coloured in the sense of offering hardware features that would potentially saturation or other non-linear behaviour, be useful in a studio context, such as but it’s certainly a high-quality preamp, speaker switching, are undermined by with plenty of gain, low noise and a decent inflexible or restrictive implementation. amount of headroom. The gain pot has But the positives shouldn’t be a 50dB range, which is extended further by overlooked. Transformers or no a -20dB pad button, allowing signal levels transformers, the 32Ci sounds great, up to 18dBu to be accommodated. albeit without adding any noticeable Moreover, the lack of obvious saturation warmth or substance to input signals. doesn’t mean that the 32Ci’s input The filters are more rewarding and stage has no character — just that this creative to work with than mere filters character resides elsewhere. Specifically, have any right to be, there’s ADAT I/O it is imparted by the unusual filter if you need to record more sources, and implementation. The high-pass filter can be Icon Pro Audio have really thought about swept from 25Hz all the way up to 3.15kHz, use cases other than music recording, and there’s also the much less common with the OTG socket and some highly option of a low-pass filter, which can turn advanced software routing features over anywhere between 20kHz and 160Hz. available in case you need to patch These provide plenty of tone-shaping in audio from multiple applications and options on their own, but the icing on send it over the Internet. And when the cake is a button labelled Bump. This you’re getting two channels of Harrison introduces what was apparently a common preamplification and filtering for much less modification to Harrison consoles back than a single 500-series Harrison preamp, in the day: a small resonant peak at the it’s hard to dispute that the 32Ci offers corner frequency of the high-pass filter. plenty of value for money.

This doesn’t transform it into a screaming Moog ladder filter, but it’s brilliant for those situations where you want to convey the £ £429 including VAT. impression of a rich, full low end without W www.soundservice-msl.co.uk cluttering the mix up with unwanted sub W www.iconproaudio.com

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Ken Scott & Emre Ramazanoglu: Ziggy Stardust In Atmos David Bowie’s classic album has been reinvented in three dimensions.

SAM INGLIS

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n July 6th, 1972, an unearthly figure in a blue and gold quilted onesie made music history. David Bowie’s Top Of The Pops performance of ‘Starman’ influenced countless future hitmakers, taking glam rock in a strange and thrilling new direction. The song and its parent album The Rise And Fall Of Ziggy Stardust And The Spiders From Mars established

Ken Scott at Abbey Road Studios.

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Emre Ramazanoglu has become one of Britain’s leading Atmos mix specialists.

Bowie as a superstar, and would form the cornerstones of his incredible musical legacy.

Fifty-two years on, that legacy is being celebrated with a box set of outtakes and rarities entitled Rock & Roll Star, and an entirely new Atmos mix of the Ziggy album. Both were overseen by original co-producer Ken Scott, who worked with Emre Ramazanoglu to carry out the immersive remix from the original multitracks.

Star Quality Back in 1972, Ken Scott was an ambitious house engineer at London’s Trident Studios who was looking to spread his wings. “I had done two albums with David as an engineer, with Tony Visconti producing, and David and Tony split ways.

And I was working with David, who was taking some time off after having very little success with the two TV albums.

He came in to produce a friend of his, a gentleman by the name of Freddie Beretti who finished up being a clothes designer for David.

“Freddie wasn’t a great singer to say the least, but it put David and I back together again and it was at a time when, as an engineer, I was getting very bored with what I was doing. Everything seemed the same, and I wanted more artistic say. So, during one of the breaks in the sessions, I happened to mention this to David, and he said ‘I’ve signed a new management deal. They want to put me into the studio to record an album whilst they shop a record deal. I was gonna produce it myself, but don’t know if I’m capable of doing it. Will you co-produce with me?’ An instant ‘Yes!’ and that was the start of a four-album relationship.”

Co-producing, as far as Bowie was concerned, meant joining a core group of creative people who worked together. “David was brilliant at putting together great teams. For the Ziggy period, that was Trevor Boulder, Woody Woodmansey, Mick Ronson, myself and then a pianist. That changed on three of the albums, but we all had our input, and then the final thing for me was David’s trust, because I think of the four albums

that I did with him, he came along to only two mixes. The rest was just left up to me. Whatever I chose to do in the mixes was fine, I guess, because once the album was completed, we never spoke about it again!

“Some of the songs had been routined, but not all of them. Quite often he would just teach the band the song in the studio, and then they got to work out what they were going to play themselves. I’ve heard that from various people, like Carlos Alomar and Mike Garson. They all say the same. When he put a team together, he picked them because he knew what they would bring to it. And he wanted them to do that.

He didn’t want anyone to do it the way he wanted it, if you get it — it was the way they would normally do it. He didn’t want to direct them that much.”

Tracking Ziggy Stardust was recorded at Trident, with Ken Scott engineering. A single 16-track machine proved ample for capturing everything. “We made

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Z IGGY S TA R DUS T IN ATMOS

Ziggy Stardust was originally recorded at London’s Trident Studios, where Ken Scott was a staff engineer.

Although the studio is long gone, the Trident name survives, and Trident Audio Developments manufacture consoles, outboard and plug-ins based on the original in-house designs.

decisions, then, as to what things were going to sound like!” Scott laughs.

Tracking sessions usually involved the rhythm section being laid down live, perhaps with one or two additional parts and a guide vocal. “Bass and drums would always be live, with possible punches for the bass. There might be an electric guitar or an acoustic guitar or acoustic piano along with them, then overdubs from there.”

Woody Woodmansey’s drums were mixed live either to two or three tape tracks. “I think some of it was stereo and some of it was three tracks; the kit would be stereo and the kick would be on its own track. I think more often than not the bass was DI, but at times I’m sure I used the amp as well, mixed in with it.”

For Mick Ronson’s guitar, meanwhile, Scott says he used “One Neumann U67, about that far away [he indicates a distance of a foot or so] from the cabinet, and we got his sound. With Mick it was always the same. He played his Les

Paul through a Cry Baby wah-wah pedal into his 100 Watt Marshall, and the way we’d get the sound is that he’d start the wah-wah pedal at the bottom and slowly bring it up until we said ‘Stop!’ And that was the sound we’d use for that particular section, or the whole song maybe.”

Bowie himself, meanwhile, was an engineer’s dream. “With regard to his

what we had back then, I’d need to look at pictures to remind me what we had, but probably an LA‑2 or something like that, and the mic would have been either a Neumann U67 or an AKG C12A. He was a one off!”

There were also orchestral overdubs, which at that time had to be recorded in a less than ideal way because of the preferences of the musicians. “The orchestras refused to wear headphones back then, so we had to feed the track through large speakers. If it was loud enough for them to hear it, it was picked up by the mics. I remember when I was doing the 5.1 remix, I hadn’t listened to the multitracks or anything for 30-odd years, and then the first time I brought up the orchestra faders I suddenly heard this track coming out. I thought ‘What the hell is going on? What’s this?’

And suddenly I realised ‘That’s coming through the speakers!’ I had completely forgotten about that. I got so used to them wearing headphones. But yeah, that’s something that we used to learn, the whole thing of blending things like that so that it doesn’t appear as if there’s too much spill coming in when you start to cut live. You just learned to deal with it back then. Now, you get everyone having to be in their own little booths, you can’t have anything crossing over because who knows what we might want to do with it later!”

From Surround To Immersive Ken Scott carried out the 5.1 remix of Ziggy Stardust in 2003 and, in retrospect, sees it as something of a missed opportunity. “I said to the label, ‘OK, there are two ways I can do this.

I can modernise it. Or I can just do that as it was originally, just putting things in different places.

Which would you prefer?’

And they said ‘We’d love both versions, but we can only afford the one version — which has to be the same as the original.’”

Two decades later, the call came to revisit the album for the next generation of surround formats, and specifically, Dolby Atmos. “I insisted going into this that it had to be a whole different thing. It was going to be a new album, basically. Same music, different album.

I believe that if the listener is in a different

Ken Scott: “The orchestras refused to wear headphones back then, so we had to feed the track through large speakers. If it was loud enough for them to hear it, it was picked up by the mics.”

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vocals, on the four albums I did with him, I think probably 95 percent of the vocals were one take, first take, beginning to end. I would get the level, the sound, hit record on the tape machine, and what he did that one time through is what you still hear today. He was amazing. I think I probably used a little compression; I can’t remember

FE ATURE

Z IGGY S TA R DUS T IN ATMOS

To create ambiences for the immersive remix of Ziggy Stardust, Ken Scott played stems out through a pair of large floor‑standing B&W monitors into Abbey Road Studio 2 and re‑recorded them using a variety of microphones placed around the room. In the event, it was the Core Audio OctoMic in the centre of the room that provided the best results.

environment, you should try and make it that they’re in a different environment.

“There’s also the whole thing where Apple have to have the stereo matching up exactly with the immersive audio.

Well, that’s OK as long as you’re working totally digitally, but with regard to the old stuff that was done on tape, no tape machine ever runs at the same speed twice. So there is no way of getting the stereo to match up to the Atmos mix unless you go in and you do things like Elastic Audio to match them up, and then you’ll start to lose quality. So what I insisted was ‘OK, we’re going to do the immersive audio mix — but there’s also going to be a brand new stereo mix as well, which will be based around the immersive audio.’ Anything we changed on the immersive audio will be reflected in the stereo mix, so that they will match up perfectly.”

The New Team There was never any doubt that Ken Scott had to be the person to oversee the Atmos remix. “David insisted that

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if anyone was going to do any future work on his product, it had to be the people that did it originally. It had to be the people that knew what he was going for, that was part of that original team.”

But there was a problem.

“When it was first brought up about two years ago that we might be doing this, I thought ‘Oh, great. I’ll start learning about it.’ But I learned very quickly there is no way that a 76-year-old would learn Atmos well enough to make a classic record into a classic Atmos record.

“Luckily I was introduced to Emre at an event, and a bit later, I said to a friend of mine. ‘Can you put me in touch with that guy I spoke to at the awards ceremony?’

He said ‘You mean Emre?’ I said ‘Yeah.

Yeah. OK.’ And we met up, we were speaking and it was amazing. It was as if Emre was part of me. We work together so well, it was so easy. It was such an enjoyable 10 days and it’s turned out to be something I am exceedingly proud of.

And once again, I keep on going back to the team thing. There was the team of Emre and myself, and it worked perfectly.”

Praise indeed — especially when you consider that it’s the first time this has happened in Ken Scott’s 60-year career. “I started off as an engineer, and then decided in 1971-ish that I wanted to move into production. I have tried since 1971 on several occasions to work with another engineer, and it never seemed to work for me, because it would take my mind away from the music. I could always reach over and do something quicker than explaining it to the other engineer.

So I have always engineered my own projects — until now!”

Precision Matching Emre Ramazanoglu is a London-based musician, who has enjoyed varied success as a producer, songwriter, drummer, programmer and engineer.

Lately, he’s become something of an Atmos specialist, with credits including the immersive remixes of Eno’s classic ’70s albums, and he brings an impressive degree of rigour and technical understanding to the often confusing business of immersive mixing.

“A lot of the work I do is exactly matching the stereo remix, making a fully immersive version where the binaural fold-down has exactly the same balance as the stereo mix,” he explains. “Often artists aren’t involved, and I’m not gonna change the artist’s mix when they’re not in the room and the producer’s not involved and the mixer’s not involved. It’s certainly not my job to go and reimagine their mix, when they’ve spent a year getting that one approved. That’s my gig on those jobs, and it’s very clearly stated especially by UMG and Sony that that’s the job, because it’s very very difficult to do correctly. Matching the mastering from pre-master stems is not straightforward in immersive audio.”

Immersive audio formats can be channel-based or object-based. In the former case, a master recording contains a fixed number of mono channels, each of which is mapped directly onto a specific loudspeaker. In the latter, the channels within the master typically represent individual sources or groups of sources, and are packaged with metadata that allows their spatial position to be assigned on the fly during playback. Dolby Atmos is a hybrid format, containing both surround ‘bed’ tracks and mono or stereo objects. Engineers who mix music in Atmos have very different opinions about the relative usefulness of these elements, and Emre Ramazanoglu insists that if you want the binaural fold-down to sound right, you need to avoid using the beds altogether.

“The bed doesn’t translate to headphones very well, and wasn’t really intended to. That was not the purpose.

It’s not a really useful tool if you want any headphone translation. You need specific control of the binaural metadata per element, and precise control of objects, in order to get a deeply translating stereo companion in binaural, and also being aware of what that does with Apple Spatial. So now the only thing I use on the bed is a touch of LFE. And very little, often, because you can’t rely on LFE for an Atmos mix because several playback systems just don’t have it, or reproduce it in unexpected ways.”

Instead of using the conventional Atmos beds, Ramazanoglu creates what he calls ‘object beds’: groups of mono or stereo objects routed to specific positions within the 3D space. Individual sources or stems that don’t need to be treated as objects in their own right can then be

Conceptual Art The Rise And Fall Of Ziggy Stardust And The Spiders From Mars has sometimes been described as a concept album — a notion with which co-producer Ken Scott has no truck. “That’s a complete fallacy.

If you consider that, the main thing that brings it all together as a ‘concept’ is the track ‘Starman’, and that was never part of the original album. We had a cover of Chuck Berry’s ‘Round And Round’ in there until we gave it to the record company. They came back saying they didn’t hear a single, so we went back into the studio, spent two days recording ‘Starman’, and just put it in place of ‘Round And Round’.

So as the main focus of the concept was never in there originally, how can you possibly consider it as being a concept album? There are some songs that link together about Ziggy Stardust, of course, but as a full-on concept album, no.”

Not that Bowie himself was averse to laying false trails: “You could never tell with him. A track off Hunky Dory called ‘The Bewlay Brothers’ was the last thing we recorded for the album. He came rushing in one day and said ‘I’ve got a new song we’ve got to record!’ And he said ‘But don’t listen to the lyrics.’ I said, ‘OK, why?’ and he said ‘Because they don’t mean anything. I’ve written it specifically for the American market, because they’ll read things into anything.’ And I must have heard five, six, seven different ideas as to what that song’s about — and David would agree with every single one of them. So, yeah, you could never quite tell. David loved to float some porkies every now and again!”

fed to these object beds as required. “I’ll have multiple object beds depending on what binaural positions work for the intention of what I’m doing. So you end up with quite large object-based mixes.

But again, I don’t even use the object beds that much for anything positional, because then you’re lying inbetween two often different binaural positions, and if you have to hear it in headphones, it won’t be a useful translation. So, I’m much more definite and will be very specific about where things go.

“Another issue in Atmos that’s really constant is that you’ll get a heavily front-loaded system and then small satellites or surrounds, and all the bass management’s at the front. You have to

bear that in mind a little bit when you’re moving elements around. You technically will get a non-discernible location from your low end, but it depends how high your bass management is going up, and what speakers there are. You have to think a bit carefully about where the weighting is going and how it’s going to feel, because you can go absolutely mental and it’ll maybe sound quite nice in your room, but in headphones or other rooms it could be just a real mess.

“And this is the problem you get with Atmos a lot. People say it sounds ‘watery’, and it doesn’t. It doesn’t sound any different from stereo — unless you make it sound different by not making the mix feel coherent. What’s often happening The immersive remix was carried out at RAK Studio 4, newly equipped with a Genelec Atmos monitoring setup.

www.soundonsound.com / September 2024

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Z IGGY S TA R DUS T IN ATMOS

were a couple of places I used a third kick drum, which was really sort of low end, just to emphasise a couple of spots.”

Getting Wet

David Bowie’s 1972 Top Of The Pops performance of ‘Starman’ influenced innumerable future musicians and was eventually voted the most memorable in the show’s entire history.

is people are mixing from pre-master stems, and these are from a track that has been mixed with a limiter or more on throughout the mix process. This mix bus is then removed and not replaced, so you’re not actually hearing a mix that has ever been approved or is even finished!”

Future Proofing By the time Scott and Ramazanoglu started work on the Atmos mix at RAK Studio 4, with its state-of-the-art Genelec monitoring rig, the producer had already done a good deal of detailed preparation work. “The original Ziggy sound was very much — quote unquote — the ‘glam rock’ sound. And that was fine for the ’70s. It needs to be beefed up a bit. It needs to be modernised. So one of the things that I did was I went in and I added samples to the drums. Initially just a snare and kick, not replacing anything, just adding to the original sound, to beef it up a bit and make it more modern. As it went on, for one thing, back in the day, I hated cymbals. I now quite enjoy cymbals. So we found in doing this: ‘Hang on, there’s no cymbals. What’s going on?’ So I then had to go in and program in hi-hats and cymbals and all of that kind of thing.”

“I was slightly concerned when Ken mentioned samples and adding stuff,” admits Ramazanoglu. “I’m very very fussy

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about that kind of thing. Especially with the hi-hats, which is really not often done sample by sample — but I don’t think you’d tell they were there. It was just part of the drums. If you’d put them up, I wouldn’t have questioned it. I might not even have known.”

“It’s ridiculous!” laughs Scott. “It literally is hit by hit, making sure it’s on there. I use the Massey DRT to get it into MIDI and then trigger from there, and there are always some things that are going to be off, and you have to move it around. Obviously when you’re coming to hi-hat, especially when it’s very quiet with regard to everything else, it’s going to be off more. So it really was going in and moving around almost every beat to make sure it’s exactly the same as the original. Otherwise, then it starts to sound fake. And also, doing it via MIDI, you do get the builds and the quiet passages, all of that kind of thing, a bit better. So that makes it more human.”

Scott’s painstaking efforts also addressed another respect in which the original sound has dated somewhat. The album’s relatively tight low end made it perfect for cutting to vinyl, but arguably unnecessarily bass-light on modern playback systems. “The kick drum sample I used the most throughout had a lot more low end than the original, and there

Once the sample augmentation process was complete, Ken Scott bounced stems of important elements and took them into Abbey Road Studio 2. “I can’t remember what it is they use at Abbey Road these days, but we had a pair of their regular monitors down in the studio, and I had them set up Neumann U67s and AKG C12s and U48s all the way along the studio, and played back the stems.

“As luck would have it, there was an engineer came along that it looked as if he was going to be working on this project with me. He came along and he brought this mic called a Core Audio OctoMic, and he said ‘Would you mind if I set this up in the middle?’ I said ‘Fine.’

He set it up, we recorded it, we never listened to it when we were doing all the stems. We just listened to the Neumanns and all of those. And it wasn’t until we came to start mixing that we finally managed to pull up the OctoMic — and that was the one. It was so much better than the rows of mics that I had along Studio 2!”

“This mix was really interesting,” continues Ramazanoglu, “because when we first talked about it, my very first thought was ‘God, that’s a dry album!’ So when Ken said ‘I printed rooms’, I was like punching the air going ‘Oh my God, yes!’

Because you won’t listen to this and think it’s wet. Not even the stereo mixes. You don’t think ‘It’s a different record. It’s all huge reverbs!’ It’s nothing like that at all.

What reverb there was in the original mixes came from an EMT stereo plate, and although Scott and Ramazanoglu weren’t trying to recreate those mixes exactly, they did end up leaning on modern equivalents. “We did quite a bit of looking for reverbs,” says Ramazanoglu. “We were making sure it sounded natural in the track. It wasn’t just whip up that plug-in and it’s done. We had to search a bit for a couple.Actually we used Altiverb’s plate, more than I thought we would. They did an unbelievable job that I think is slightly unsung taking Altiverb 7 and making it immersive. And so they’ve got sort of combinations of all the plates to make a 9.1.6 plate, which is really good. They’ve done a lot of really clever stuff in there that isn’t being shouted about really much, but

it’s absolutely brilliant. And otherwise, Liquidsonics’ Lustrous Plates were used quite a bit.

“We didn’t really reference the original mixes that much until quite near the end, where we were having a listen going, ‘I’m not sure these are present enough,’ and then we had a quick listen to the original mixes, and I would sit with headphones and make a small tweak here and there just to get a bit more edge, because I think we were feeling out the room as well. It was some of the first mixing we’d done in RAK 4, which is fabulous.

But we were working out, ‘OK, what sort of reference level are we working at?

And could it just have a little hair more presence?’ and that’s the first time we really listened to the original mixes.

When Ken was saying it was nice working together, it really was, because we’d both go ‘1dB on the vocal!’ at the same time.

‘This word, take the ess down!’”

Third Dimension Freed from the requirement to match an existing mix, Emre Ramazanoglu found he needed to use much less processing than usual on the Ziggy Stardust remix. “Normally I will be recreating the mix bus or at least the mastering, so I’ll have linked side-chain processing going across. If I have

100 mono objects, it’ll be 100 mono channels of processing all side-chained and all working together and creating glue. Here, we used one compressor on one channel on one track on the entire record, which is testament to the recording I have to say. There were a few EQs — a few — and it was mainly subtractive.

“There was so little done across the whole record, but this is why I was so impressed by this, because you could move things. I did very little of what I would often do to create space.

There’s all kinds of ways of making space immersively that feel satisfying and coherent that mean I can spread effectively a mono or stereo source around a bit more, but we didn’t really do that, because of the way it’s put together. It’s slotted into a mix that was in 3D rather than in stereo, and the whole basis was thinking in 3D and thinking about... We found some interesting things with masking. Some things that we hadn’t heard before.

Even some melodic stuff I’d never heard before, because it was interacting before with other elements. So there was recreating that feeling of interaction.

“One of the things I hadn’t done before which is quite interesting is have the vocal ‘floating’. It’s actually

slightly forward and up. It made it all quite spacious feeling immediately, not having it all just pinned to the back wall.

I really don’t use the centre speaker much in Atmos, because of headphone translation issues. But this worked!

There’s a sense of belonging, every piece belongs in its new position for a reason, and it’s not just sort of plonked there.

It’s like a 3D supported mix.”

Ken Scott sums up the philosophy behind the immersive mix thus: “I wanted it to feel like you were in the club or the theatre or whatever with the band, and it’s the way they’re positioned on the stage. The vocalist will always be out front, and the band just around like that, not all in one line, because when they’re live they’re not all in one line. So it was very much that kind of thing that I was after, more of a live experience.”

“It’s quite immersive in that there’s a lot of sides and rear,” concludes Emre Ramazanoglu, “but unless you turned your head, I don’t know if you’d notice massively — except if you turned it to stereo, you’d go ‘Oh no, put that back.’ If you didn’t even notice it was Atmos, I wouldn’t be sad. As long as you felt enveloped by the music, then I’ll be delighted, personally. In the nicest possible way, it’s not about making a Dolby advert!”

This Pro Tools screen capture shows part of one of the sessions from the remix. Note the additional sample-based drum tracks, here rendered as audio, and re‑recorded stems with multiple mics.

www.soundonsound.com / September 2024

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Lauten LS-408 & LS-508 Drum Microphones

Lauten’s classy new capacitor microphones are purpose-made for toms and snare drums.

PAUL WHITE

W

e’re accustomed to using dynamic mics on the individual drums in a kit, but Lauten have taken a different approach with their LS-508 and LS-408 models, which are also dubbed simply the Tom Mic and Snare Mic. The Tom Mic (LS-508) uses a 32mm supercardioid capsule in a side-address configuration. Its FET circuitry is combined with integral analogue sound-shaping specifically designed to meet the challenges of tom miking, such as minimising cymbal bleed. As this is a capacitor microphone, 48V phantom powering is required. Lauten Audio use

Lauten LS-408 & LS-508 £385 each pros • Sturdy.

• Voiced for their specific applications.

• Solid sound.

cons • Not cheap — but then these are very well-engineered mics.

summary A well-built, great-sounding pair of drum mics specifically voiced for toms and snare drums.

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what they describe as a bleed-blocking head design, and this, combined with their proprietary dual‑bias circuitry, is said to offer up to 28dB of off-axis rejection. The capsule is shockmounted to minimise the effect of vibration if mounted on a rim clip, and the basket even includes a mildew-resistant dust filter to avoid problems caused by the decay of accumulated drum stick dust.

With a stubby, side-address profile, the mic comes with a swivel mount secured via a locking ring, and I’m pleased to see that it has a very strong locking mechanism so mic droop shouldn’t be a problem. The overall size of the mic is 127 x 60mm, or 131 x 106mm if you include the mount. Weight is 450g with the mount in place. A satin black finish is applied to the mic body, basket and mount, embellished with a red Lauten crest that denotes the front of the mic.

A pair of three-way toggle switches on the mic body engage high-pass and low-pass filtering, with two frequency options for each filter, plus a flat setting. This filtering isn’t as straightforward as it seems: the response curve has been shaped specifically for toms, and the same is true of the Snare Mic, the latter having a 1kHz dip combined with a small peak at around 3kHz. With the low-pass filter set to its ‘flat’ position, the frequency response is not actually flat but rather unmodified. The mic’s

‘default’ frequency response exhibits a lift from 1-2 kHz, dips sharply at 3kHz, then rises at around 5kHz to capture the stick attack. There’s also reduced sensitivity from 9kHz to 14kHz to minimise cymbal bleed.

Using the 12kHz low-pass filter tames bright tom strikes and further reduces cymbal bleed. The 5kHz setting attenuates cymbal bleed even more, while still preserving the general impact and tone of the tom.

With the high-pass filter switch in the flat position, low frequencies are not filtered

Unlike the LS-508 Tom Mic, the LS-408 Snare Mic is an end-fire design.

out. Since smaller toms may not produce so much low end, there are 80Hz and 140Hz high-pass options for cleaning up the lows and reducing kick‑drum spill. The frequency response of the mic (without filtering) is quoted as 20Hz‑20kHz, though with no ‘dB down’ limits specified, those figures are not particularly informative.

Dynamic range is specified as 135dB minimum and the self-noise a low 15dB A‑weighted. Expressed as a signal-to-noise ratio, this works out at 79dB. While normal drum miking doesn’t require particularly low noise figures, it is good to know that this mic is quiet enough to use on alternative forms of percussion that may not be so loud. Both the Tom and Snare Mics have a deliberately lower sensitivity than for a typical general-purpose capacitor mic, as close‑miked drums present a loud environment. The LS-508’s sensitivity is specified at 1.3mV/Pa.

LS-408 Snare Mic The LS-408 Snare Mic is also a FET capacitor microphone, with a similarly stubby form factor body and the same mounting hardware, but this time configured as an end-fire mic. The capsule is also a 32mm design, but here with a cardioid pickup pattern rather than supercardioid, and it has a different response in the mids and highs. This model has an overall size of 109 x 50mm, or 112 x 71mm with the mount. There are the same body-mounted EQ switches, with the same options as for the Tom Mic. As before, the flat low-pass option delivers an unfiltered high-frequency response to capture the stick impact, maintaining an airy high end while the low end captures the punch of the fundamental frequency.

As with the LS-508, the low-pass filtering options can be used to shape the high end or to reduce bleed from cymbals and hi-hats. Overall, the general specifications are very similar to those of the Tom Mic, but with a slightly different sensitivity of 0.5mV/Pa. Both mics are shipped in cardboard boxes and include the mount, a cleaning cloth with Lauten logo, and a padded zip-up storage bag.

Performance While they’re not the cheapest options for recording toms and snares, given that these are FET capacitor mics they are not unduly costly either, especially taking into account the reputation of their manufacturer. The mounting hardware is excellent, and the short, stubby bodies

The Tom and Snare Mics both feature the same high- and low-pass filter options, but have different ‘default’ frequency responses to suit their respective sources.

make positioning easy. They are also solidly engineered so the odd wayward drum stick shouldn’t upset them too much.

As with most drum mics, you have to know what to expect from a close-miked tom position, as some of the stick articulation invariably gets suppressed, and the proximity effect of the supercardioid capsule also beefs up the low end as you get closer. If that becomes too much, the built-in high-pass filters on both models are a very welcome addition. Once you add in your overhead mics, you get a really solid tom sound with just the right amount of stick impact.

I also tried the Tom Mic on my set of three Rototoms; positioned maybe half a metre above the toms, the drums stayed pretty well balanced and I didn’t lose the ringiness and stick impact as you often do when putting a mic very close.

With both models, you will capture more stick attack by aiming the mic across the head, starting from the usual ‘just inside the rim’ position around 50mm above the head, which means that you can get a good position using a rim clip if that is your preference. Aiming the mic across the head in this way is Lauten’s recommendation.

The Snare Mic also demonstrates what most engineers already know about positioning. If you just mic the top head, you lose some of the snare sound but capture a good solid punch, so the overheads might

bring back just enough snare rattle for the drum mix to work. If not, you can mic the bottom of the snare (remembering to invert the polarity relative to the top mic, if using both). Used beneath the snare, the Snare Mic captures all the snare snap and rattle you could wish for, and you can filter out the lows using the onboard EQ to avoid too much kick-drum spill. Used on its own, the sound from an under-snare position becomes very snares-heavy, so balancing the top and bottom mics in the context of what the overheads are capturing is the key to a well-balanced snare sound. Another advantage of adding an under-snare mic is that you can use the filters on the top mic to suppress the spill from hi-hats and cymbals without losing high-frequency articulation, as the under-snare mic puts that back in. Having two different HF roll-off positions is a big help in that respect.

Both these mics are strong performers, and though they won’t compensate for poor mic technique, they do allow for more flexibility than most commonly used snare and tom mics when in the right hands.

Their drum-specific voicings are invaluable, as are the flexible filter settings that are specifically tailored to the job in hand. £ £385 each, including VAT. T Synthax Audio +44 (0)1727 821 870 E info@synthax.co.uk W www.synthax.co.uk W www.lautenaudio.com

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SPL Phonitor 3 With its clever Phonitor Matrix, this high-end headphone amp aims to deliver a more speaker-like listening experience.

PHIL WARD

Y

ou perhaps don’t need me to point out that there’s something of a disconnect between the way most commercial music is mixed (on loudspeakers) and the way it’s increasingly consumed (via headphones).

One result of this is that mixing on headphones is becoming much more common. Whereas, a decade or two ago, headphones in studios were primarily present for tracking and overdub duties, these days they also have an important mix role. This is especially the case for Dolby Atmos music mixes (other immersive systems are available), where the most often consumed end product is probably an Apple Spatial binaural headphone downmix.

A further consequence of the expanded studio role of headphones is that headphone amplifiers have become more relevant. Previously, we all just plugged our headphones into an interface or mixing desk headphone socket, or maybe into a headphone distribution amp, and perhaps didn’t think too hard about how that affected the headphone sound.

Yet, mixing on headphones means we’ve become more sensitive to headphone sound quality. The subject of this review, SPL’s Phonitor 3, reflects the increasing significance of headphone amplification.

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But, more than that, it offers some intriguing headphone output configuration options. Christened by SPL the Phonitor Matrix, the configuration options enable some significant manipulation of the headphone listening experience, primarily with the aim of making it more speaker-like.

It’s probably worth me mentioning that while I’m looking at the standard version here, SPL also offer a slightly more expensive option, the Phonitor 3 DAC — this covers much the same ground, but it also features a built-in 32-bit, DSD-capable D-A converter.

Ins, Outs & Basics While I’ve introduced the Phonitor 3 as a headphone amplifier, and that function does generally cover its defining set of features, it’s also a monitoring preamplifier that offers three balanced stereo XLR input pairs and one balanced XLR stereo output pair, for the connection of active monitors.

Input source selection for both monitor preamp and headphone amplifier functions is achieved by a simple three‑position rotary switch on the front panel, and output volume is adjusted by a large and satisfyingly tactile machined‑aluminium volume knob.

Having just deployed the term ‘tactile’, it strikes me that it’s a suitable one for the Phonitor 3 more generally, because the traditionally constructed and black‑painted

Headphone Amplifier

all‑metal enclosure feels really substantial and no-nonsense. It looks and feels like it means business. Also apparently meaning business is the Phonitor 3’s headphone amplifier itself. It’s powered, say SPL, from ±60V rails to provide significantly greater output power with very low noise and distortion, and its very low output impedance means there’s negligible change in frequency response with variable headphone impedance.

I’ve mentioned the volume control and input selection switch on the Phonitor 3’s front panel, but there’s a lot more besides.

Firstly, dominating the right-hand-side of this panel are two retro-looking VU input level meters that, say SPL, display BBC-style meter ballistic characteristics. (At this point I will recommend Hugh Robjohns’ fascinating history and explanation of the

SPL Phonitor 3 £1899 pros • Very high‑performance headphone amplification.

• Fascinating and versatile headphone processing.

• Very high build quality.

cons • Perhaps a solution looking for a problem.

summary The Phonitor 3 is an interesting and very well‑engineered product that does extremely competently exactly what it sets out to do.

VU meter in SOS May 2024, called ‘VU Meters: Virtually Useless or Very Useful?’ you can find this on the SOS website at: https://sosm.ag/vu-meters.) Second up, beneath the VU meters and alongside the 6.3mm headphone output socket, are six slide switches that engage or disengage the Phonitor Matrix options, and provide left/right channel swap, left/right channel polarity, left/right channel solo, VU range, and headphone or XLR output. On the left side of the front panel are five rotary controls and a further slide switch that configure and adjust the Phonitor Matrix and an option called Laterality.

Enter The Matrix To begin with, let’s consider the Phonitor Matrix options. The first is Crossfeed and, with the Phonitor Matrix enabled, turning this knob feeds an increasing amount of left-channel signal into the right headphone channel, and right-channel signal into the left headphone channel. The idea is to simulate the psychoacoustics of listening on loudspeakers where, it should perhaps go without saying, both ears hear both channels to some degree. Crossfeed, though, also adds some EQ to compensate for the fact that more interaural crossfeed from a pair of speakers is likely also to be accompanied by slightly emphasised high‑frequency balance. Adding crossfeed, say SPL, corrects to some extent the ‘false’ stereo image presentation of headphone listening that results from its absence.

The second Phonitor Matrix option, Angle, is related to Crossfeed, in that it compensates for the interaural crossfeed time delay phenomenon (it takes fractionally longer for sound from each speaker to reach one ear than the other) of the speaker listening experience that’s absent on headphones. Again, the intention is to make headphone listening more speaker-like. The angle

ALTERNATIVES The professional audio world isn’t short of high-performance headphone amplifiers: the Benchmark HPA4, DACS Purity and Violectric HPA V550 for example. But if the Phonitor Matrix functions of the Phonitor 3 particularly appeal, then it’s one of a kind.

referred to in the parameter is the one that would be created by the stereo spacing of the imagined loudspeakers.

The wider the spacing of the loudspeakers, the greater the interaural crossfeed time difference between them. As with Crossfeed, the Angle adjustment also comes with EQ to reflect the variation in head-related acoustic absorption at different imagined speaker angles.

options to be routed to the rear-panel XLR outputs, with or without the volume control operational. This enables, for example, a second headphone amplifier to be fed with a signal that’s been processed by the Phonitor Matrix.

Sound Opinion But what, I’m sure you’re wondering, does all this actually sound like? Well, to get the basics out of the way first, the fundamental headphone sound quality of the Phonitor 3 is, to my ears, excellent. I have a pretty wide variety of headphones available, with significantly differing sensitivities, impedance characteristics and tonal qualities, and each one sang with an impressive and seductive sense of clarity and impact when driven by the Phonitor 3. When considering its Phonitor Matrix functions, I have to admit that to begin with I was somewhat underwhelmed, because I found their effects decidedly subtle.

But as I began to listen more and tune into their characteristics and behaviour, my opinion changed: I found the Phonitor Matrix functions increasingly interesting and useful, particularly the Angle and Centre functions. In fact, after getting used to the Phonitor Matrix processed headphone sound for a while, switching back to unprocessed headphones was a genuine disappointment.

There’s no getting away from the fact that it’s a niche product, and when it comes to my opening point about the disconnect between people mixing on loudspeakers but listening on headphones, processing headphones to sound more like speakers probably asks more questions than it answers. But the Phonitor 3 is a likeable and strong product that’s impressively constructed and technically interesting.

The headphone amp itself is good, and I could see the Phonitor 3 becoming indispensable in the right setting. It does a good job of emulating the speaker listening experience on headphones, and I can certainly see its potential in mastering studios or in mix environments that are extensively headphone rather than speaker based.

“I have a pretty wide variety of headphones available... and each one sang with an impressive and seductive sense of clarity and impact.”

The third Phonitor Matrix option, Centre, enables adjustment of the relative level of the centre signal to compensate for the over-emphasis that will potentially result from adjustment of the Crossfeed and Angle parameters.

The final front-panel knob, Laterality, controls a function that’s related to a traditional balance control. However, it simultaneously reduces the level of one channel while the other increases, and also offers extremely fine adjustment (just ±2.25dB across the Laterality control’s full range). The intention is to provide a balance control that can compensate for reduced hearing sensitivity in one ear.

One more function to mention is the DIP switch array on the rear panel. As well as offering a +12dB headphone output option, this allows the headphone Phonitor Matrix

£ Phonitor 3 £1899. Phonitor 3 DAC £2599.

Prices include VAT.

The Phonitor 3 can switch between three sources, and can deliver a clean output for speakers, or a Matrix-processed signal for other headphone amps.

T SCV Distribution +44 (0)3301 222 500 E sales@scvdistribution.co.uk W www.scvdistribution.co.uk W https://spl.audio

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Korg MPS-10 Drum, Percussion & Sampler Pad Korg’s MPS-10 is a compact percussion instrument with hidden depths.

MARK GORDON

K

org may not be the first name that springs to mind when it comes to electronic drums and percussion, but they actually have a great history of creating innovative and interesting percussion instruments, such as the classic DDD-1 from the 1980s, the Electribe R Analogue Modelling Rhythm Synthesizer and, of course, the iconic Wavedrum.

It’s not exactly a prolific output, but Korg have often been thought-provoking and challenging of the status quo, and their entry into the sampling multi-pad market, the MPS-10, is no exception.

Boasting an impressive 32GB of RAM, 2350

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assignable instruments and more than 3000 high-quality samples, the MPS-10 also features ground-breaking Continuous Controller Pads that give you real-time control over internal effects, depending on where the pads are struck. A powerful four-track looper also adds to a package that is an intriguing and slightly different proposition from the usual suspects.

If Looks Could Kill At first glance, the MPS-10 has a familiar look, with its six square pads and a further four ‘shoulder’ pads placed across the top of the unit. A rubber covering extends across the whole of the top surface of the unit, almost as if each of the individual pads have been carved out of it. The covering wraps around the edges of the unit, giving it a rugged feel and creating the impression that you could safely hit the MPS-10 anywhere without damaging

it — although, ironically, Korg have chosen to locate the control panel and screen between two of the pads. While hardly radical, this placement does set the MPS-10 apart and reduces the overall footprint of the unit but, depending on how many times you end up hitting the screen while playing, it could either be a stroke of genius or a design disaster.

In a nod to the new Roland SPD-SX Pro and the Alesis Strike MultiPad, coloured LEDs sit below each of the pads, but there is no option for setting your own colour configuration. Three preset colours are offered, depending on the sound type that’s assigned to the pad: red for a one-shot sound, blue for a repeating sound or loop, and green if no sound is assigned to the pad.

As I mentioned above, the control panel sits neatly between pads one and two, and stretches the width of the two pads above it, making it smaller in terms of real estate than its competitors. This results in a slightly smaller colour LCD display than I would have hoped for.

To the left of this LCD are Main Volume and Phones Volume controls, along with a knob to alter metronome tempo,

which incorporates push-button on/off functionality and two small LEDS either side, as visual tempo indicators. To the right is a large data-entry knob, alongside dedicated Edit, Write and Exit buttons.

Two large +/- buttons are provided to advance through the kit presets (pressing the data-entry knob also ‘engages’ it as an alternative way to quickly scroll kits), and an equally large Sound Off button offers an immediate way to silence the unit. This can come in very handy when using loops and long samples.

Dedicated buttons that take you directly to the Kit, Instrument, Sample and Utility screens are located beneath the LCD, along with a second row of buttons that control the looper functions: Looper FX, Record and Erase. The SW Mode button switches the functionality of the four large Track buttons between controlling the looper tracks and enabling the CC Pads. All the buttons are clearly backlit, making it very easy to read their labels on a dark stage. However, none of the knobs offer the same visual assistance.

At the rear of the unit are a single quarter-inch stereo headphone socket, a left and right Main output, and an additional stereo Sub output pair, both on quarter-inch mono jacks. A stereo quarter-inch line input and mono quarter-inch mic input are provided to connect external signals, which can be processed by the MPS-10’s effects engine or used as a source for the sampler and looper. The mic input features a preamp and can be configured as a balanced or unbalanced connection, but it is worth noting that you can’t use the line and mic inputs simultaneously.

Two ‘dual’ triggers are provided on stereo quarter-inch jacks, so four individual trigger inputs can be achieved using Y-splitter cables. While you could use two dual-zone pads with these inputs, there’s no option to select combination ‘Head & Rim’ triggers within the MPS-10, so you’d have to assign samples to both head and rim individually if you were using this type of trigger.

Alongside the triggers are two footswitch inputs, which can be used to control various functions of the MPS-10, such as kit advance and looper start/ stop, and an expression pedal input that can be assigned to looper volume, filter cutoff, reverb mix, pad decay and pad pitch. I had hoped that the expression pedal would allow for open and closed hi-hat control, but it appears that this is not possible.

Two USB ports are provided: one for connection to a USB flash drive for backing up data or importing samples, and a second that allows both MIDI and audio information to be sent to and from a connected computer. Four channels of audio can be sent from the MPS-10, that correspond to the physical main outputs (1+2) and sub outs (3+4).

A little surprisingly, the MPS-10 only has a MIDI out socket and no MIDI In. While many people will use USB MIDI communication, in my own hybrid drum setup I send MIDI Program Changes and MIDI Clock to my SPD-SX Pro via a good old-fashioned MIDI cable, something I couldn’t do with the MPS-10.

Play For Today The MPS-10 includes 100 preset kits and a further 100 user locations to store your own creations. The presets cover a very wide range of genres and styles, combining one-shot drum samples, loops, tuned percussion and sound effects that demonstrate the MPS-10 very well.

The Funk Kit is a great example of a regular acoustic kit, and many of the instruments in this preset utilise both round-robin samples and velocity switching to great effect, giving the kit a natural and realistic sound. As you play the 14 x 5‑inch steel snare harder, it opens up with a wonderful metallic rim shot. One particularly nice touch is that Korg have added a bass drum sample to some of the cymbal and hi-hat instruments, which is triggered at a specific velocity. You can easily play a bass drum with a crash

Korg MPS-10 £949 pros • Great sounds and presets.

• Innovative CC Pads.

• Very good and easy-to-use Looper built in.

• Can process external audio and include it in loops.

cons • No MIDI in.

• Editing and sampling a little complicated.

• Limited software editor.

• Smallish LCD and non-configurable LEDs.

summary The MPS-10 is a very powerful tool, offering exceptional creative and innovative features in a unique and compact form.

or, with careful dynamics, add a kick-drum pattern to a hi-hat part using a single pad.

As you might expect, electronic styles are well catered for with a host of hip‑hop, trap, dubstep and techno kits, many of them featuring percussion loops and musical phrases alongside 808-style kicks and EDM snares. Percussion presets span everything from traditional cajons, bongos and congas through to more exotic Middle Eastern and melodic instruments and, again, use round-robin and velocity-switching samples that enhance realism.

As you switch between kits, the MPS-10 employs SST (Smooth Sound Transition), which allows instruments, loops and effects to continue playing when you switch to a new preset. This enables seamless transitions and allows you to access different kits while loops, samples and effect tails continue to play in the background.

The only small criticism I have from a playing point of view would be the sensitivity of the pads. Even with sensitivity set at maximum, some lighter hits weren’t picked up. In general, percussion pads don’t offer the same subtlety offered by mesh pads on an electronic drum kit, so you won’t be doing any snare buzz rolls, but even so, compared to other percussion pads I’ve

Round the back we find a host of quarter-inch jack sockets for headphones, main and sub audio outputs, line and mic input, trigger inputs and footswitch and expression inputs. There’s also a MIDI out port and USB ports for an external drive and connection to a computer.

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used the MPS-10 doesn’t offer quite the same lightness of touch. In fairness, I am looking at it from a subtle drumming point of view, and in most real-world playing situations I doubt it would be a major issue.

One of the features that sets the MPS-10 apart from its contemporaries is the unique functionality of the four shoulder pads (or CC Pads, as Korg call them). In addition to working as regular triggers, Korg have enabled each of these pads, either individually or in combination, to control transposition, note delay, multi-effects and filtering, depending on where you strike the pad. This feature has been put to great use on many of the preset kits, and exploring one of them offers probably the best way to discover its potential.

Any kit that utilises the CC Pads displays a line of four rectangles (representing the CC Pads) at the top of the LCD screen. In the case of preset 10 (CC PAD:FILTER kit), each pad plays a tambourine sound when struck, but the fun begins as you work your way from left to right across the CC Pads. This action progressively imposes a low-pass filter across the whole output. Each CC Pad has 32 possible locations along its length (a total of 128 across all four pads) where it can be struck. As you hit each pad, the location of the strike is shown in the four rectangles at the top of the LCD and also

In addition to displaying the kit name and number, the four rectangles across the top of the LCD screen represent the CC Pads and offer useful visual feedback by showing where each pad is struck.

transmitted as a MIDI CC4 message. Think of it as the percussive equivalent of turning a value knob or fader on a synth.

You can set a number of different filter types (low-pass, band-pass, high-pass), along with cutoff and resonance, and also change the way in which the pads respond by selecting one of nine different preset patterns.

Preset 72, Reggae Loop, makes use of the CC Pads to control the MPS-10’s internal multi-effects, by increasing delay feedback as you move from left to right across the four pads, but you can assign any of the internal effects and control any available parameter, such as the speed and

depth of a chorus, the gain of a distortion or the reverb time of a classic plate.

Another neat trick the CC Pads have up their sleeve is the ability to apply transposition to other pads. This is most effective when using tuned percussion such as marimbas, or pitched samples such as bass notes or keyboard stabs.

Preset 85, Poly Pipe, puts this facility to good use. The preset itself features pitched samples of plastic pipes being struck (in the style of the Blue Man Group), spread across eight of the pads. The two centre CC Pads, when struck, will either raise or lower the pitch of all the pads by a tone, thus doubling the number of available pitches. A transposition value from -24 to +12 (in semitone steps) can be assigned to each of the four CC Pads and this in turn can be applied to any of the 10 main pads.

So, as an example, hitting CC Pad 1 could lower the pitch by three semitones of any instruments assigned to say, pads 3, 4 and 5, and hitting CC Pad 2 could raise the pitch by 12 semitones of instruments assigned to pads 3, 6 and 9 — and so on. It’s a great feature that really expands the flexibility of pitched instrument samples.

The final, and simplest, CC Pad function is Note — or, more accurately, Note Delay.

This allows you to apply a repeating pattern to any instrument assigned to the four CC

Loop Da Loop Many percussion pads have the ability to record loops and phrases, but the MPS-10 takes things a step further by including a four-track looper that records any sound routed to the main outputs.

What this means in practice is that not only can you record what you play on the pads, but you can also incorporate audio signals that are plugged into the line or mic inputs into your loops. When the looper is being used, the LCD is divided into two sections (upper and lower). The current kit name is shown above and the looper information is shown below.

The looper ‘tracks’ are marked by four circles on the lower half of the LCD that change colour depending on their status: red for record, yellow for overdub, or green for playback. The length of the loop is displayed in the centre. Above this are displays showing the looper volume, track length and bpm setting (which, as default, matches the bpm assigned to the currently selected kit).

The looper is intuitive and easy to use. Pressing any of the track buttons immediately puts the track into Record Standby, turning its status to red and flashing the record button. The default is for the looper to start recording as soon as you strike any pad, but you can configure it to start manually by pressing the record button or via a pedal input.

Once the track has recorded one cycle (the length of which can be configured in the looper settings, from 0.5 to 64 measures), it drops into overdub mode and turns yellow. The track continues to cycle round the predefined measure length in overdub mode, so you can add more material, until you press the track button again, at which point the display turns green and the track enters playback mode.

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During the overdub process, you can freely switch presets and add instruments from any kit to the loop. The remaining tracks work in the same way, but prior to recording you can set the length of subsequent loops as a multiple of the first — up to 32 times the length of the first loop.

For example, if you’d set the initial loop length as one measure, you could set the second track to ‘4x’ and it would record for four measures before cycling around. There’s also the useful ‘Free’ option, which makes the track continue recording until you choose to stop. This is great for adding improvisational elements when you don’t want to be constrained by a predefined length.

During playback, the four track buttons can be used to switch each part on and off, and you can freely drop back into overdub mode using the Record button, or use the Erase button to delete a track. The Looper FX button enables you to apply one of the internal effects to the tracks in real time, but it would appear that this is only possible across all tracks simultaneously.

As I mentioned above, the looper will also record audio via the mic and line inputs, and I had great fun adding synth and guitar lines to loops. You can mix and match audio with MPS-10 parts on the same track, and build up complex patterns and rhythms, switching tracks and effects on and off in real time to create some really interesting live performances. All loop data is deleted as soon as you power off the MPS-10, but if you’ve hit upon something you’d like to keep, you can save individual loop tracks into the MPS-10 itself. These can them be re‑imported back into the looper at a later date, or exported as WAV files onto a USB stick, to use in your DAW.

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ON TE ST

KO R G M P S - 10

Pads. The repeat can be set to eighth, 16th, 32nd or 64th note resolution, may be up to four bars in length and follows one of four simple rhythmic patterns.

Typically, you could use this to create dance-style snare rolls of increasing speed across the four CC Pads.

You could potentially use all four functions (Transpose, Note Delay, MFX and Filter) simultaneously. To make this easier to manage, when the SW Mode button is pressed to display CC Pad mode, the four large track buttons become on/ off switches, so you can quickly enable or disable each CC Pad function individually.

I’m really only scratching the surface of what it’s possible to achieve using the CC Pads in various combinations, and this feature adds a huge amount of fun, creativity and uniqueness to the MPS-10.

Digging In The MPS-10 offers a comprehensive range of editing options. Pressing the Edit button selects the Kit and Pad edit pages, where you can adjust overall kit settings including level, footswitch control and CC Pad assignment. You can also select which effects are assigned to the kit: Master FX, which includes everything from overdrive and chorus through to ring modulation and bpm delays; a separate reverb that has halls, plates and early reflections settings; low-pass and high-pass filters that can be applied to the overall output; and two Insert FX that can be assigned to each pad individually.

The Tempo knob doubles as a ‘page select’ button, and turning it switches from Kit Edit to Pad Edit, where you can select which instrument is assigned to each pad.

The way the CC Pads respond can be dictated by assigning one of nine preset patterns. The corners of the triangles represent the minimum parameter level, the long sides of the triangles represent the maximum level and pads represented by a grey line revert the parameters to their default value. Pattern 1 will increase the assigned parameter value as you play from left to right across the CC Pads, and Pattern 3 will decrease the parameter value as you move toward the centre of the four pads and is ideal for playing with two sticks

The 2350 onboard instruments are sorted into individual Drum, Percussion, SFX and Loop categories, to make selection easier, and they sit alongside user folders for your own creations. You can also adjust individual instrument levels, pan, pitch and which of the two Insert FX are assigned to a specific pad.

The final parameter here is Pad Mode, which has five settings: Normal, Exclusive, Series, Random and Latest. Groups of pads can be configured to play exclusively (as one pad is struck, other pads are muted) or concurrently. For example, striking pad 1 repeatedly can play the instruments

The sounds assigned to each pad are referred to as instruments. Although complex, the instrument architecture, which comprises two oscillators and up to 16 individual samples, does allow for the creation of very realistic-sounding and articulate instruments.

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assigned to pads 3, 4, 7 and 9, either in order or randomly. This is similar to the sequence option on the new SPD-SX Pro and is a great way to play complex patterns by only striking a single pad.

The sounds assigned to each pad are referred to as instruments, and in-depth editing is available via the Inst button.

MPS-10 instruments have quite a complex structure. Each instrument can have up to two oscillators, which can play simultaneously. Each oscillator can have up to eight blocks, each of which contains a sample, adding up to, potentially, 16 samples per instrument.

Each sample can be configured in up to eight velocity layers or eight round-robin samples, so you could, for example, have four velocity layers, each with two round-robin samples, on both oscillators.

As you can probably gather from just the explanation, it’s quite an involved process, and I found some of the terminology Korg have chosen to use, such as oscillators and blocks, a little confusing, giving more of a feel of synthesizer programming than other sample pads and percussion instruments I’ve used.

Although I didn’t find the process particularly user‑friendly, the complexity on offer does ultimately allow for the creation of very realistic-sounding and articulate instruments, and once you’ve got to grips with the concepts it starts to make sense.

Sample Analysis A major feature of a ‘Drum, Percussion & Sampler Pad’ is, of course, the ability to sample, and the MPS-10 doesn’t disappoint on that score. It can sample internally generated sounds or audio via the line, mic or USB inputs, and the 32GB of onboard storage offers plenty of space for samples, which are all recorded as 16-bit/44.1kHz WAV files, with a maximum record length of 60 minutes.

Pressing the Sample button presents two options: Recorder and Sample. Recorder, as you might expect, is the option you select to record a sample, while Sample, perhaps less obviously, takes you to the waveform display and editing tools.

The Recorder page offers three options: • C hange Sample in Inst: record a sample that replaces one in an existing instrument.

• New Sample & Inst: record a brand-new sample and create a new instrument.

• Only new Sample: simply record and save a basic sample.

A Soft Option The MPS-10 Editor is a web-based application that works via Google Chrome on Mac and Windows (URL available via the Korg website).

With the MPS-10 connected to your computer via USB, the provided link launches the software and clicking the ‘Go to Kit List’ button loads all the kit data into the application and presents a list of the Preset and User kits. You can then click on each kit and open a Pad, FX or CC Pad editing window.

The Pad page offers the ability to change instrument, along with its level, pan and pitch. You can also adjust the Pad Mode to Normal, Exclusive, Series, Random or Latest (see main text for an explanation of these modes). The FX pages enable you to select Main and Insert FX, adjust parameters

If you’re creating a new sample and instrument, the next step is to set a name for the instrument, and once you’ve done this you can finally get down to recording the audio! The whole process struck me as a little over-complicated but, again, familiarity and regular use might change my opinion.

The recorder can be set to automatically start when it detects audio above a set threshold, and can also auto stop after a set number of beats. I was able to record by playing the pads or routing audio via the mic, line and USB inputs. Unusually (to me, at least), you can set a tempo and click that plays while you’re recording, which is a nice feature and useful to play against when creating loops.

Once recording is completed, the Sample screen automatically appears, displaying the audio waveform, and the four large track buttons can now be used as transport controls to stop, play, fast‑forward and rewind. Start and end points can be created, using the data-entry knob, and vertical and horizontal zoom allows you to edit with accuracy. You can also normalise and truncate the recorded sample, as well as adjusting pitch and time-stretch. Each function offers a preview option before you Execute.

and assign effects to pads, and the CC Pad page displays Transpose, Note Delay, MFX and Filter settings (again, see main text for more on these).

Visually, the software looks quite basic and offers no more functionality than the corresponding pages on the MPS-10. I had been hoping for something a little more inspiring that not only

It’s also possible to import samples via USB, although you will need to save them to the MPS-10 before you can edit them in the sample editor. Both WAV and AIF file formats are supported, at 44.1 or 48 kHz and in 16-bit resolution.

Hit The Spot?

The MPS-10 is both powerful and fun to play. The instruments and samples sound fantastic, and are put to excellent use in the preset kits, which are all very usable in

included the ability to configure kits but also route outputs and edit samples.

Unfortunately, you also can’t drag and drop samples from your desktop onto pads to create kits, nor is there the option to edit instruments, a process that would have benefited from the larger screen available on a computer.

Roland suffered a little from a similar issue with their SPD‑SX, whose editing software, whilst more visually appealing than that of the MPS-10, was also limited.

Fortunately, the software included with the new SPD-SX Pro is exceptional, and I had hoped and expected the MPS-10 editor to at least give Roland a run for their money, but I’m afraid it falls a little short on both features and design.

specific pad functions. The MPS-10 offers so much functionality that a larger and more informative LCD, along with user-definable pad LED colours, would be a huge benefit, as would a MIDI input.

The MPS-10’s Instrument editing is very powerful, and I’m not aware of other sample pads that include round robins (which enables dynamic, natural sounds), but the process didn’t feel particularly intuitive. The same could be said for the sampling options, which, again, are very powerful but felt a little over-complicated to me.

Unfortunately, I didn’t find the manual particularly helpful, as it seems to feature lots of descriptions of functions but not quite enough detail on exactly how to use them.

That said, the MPS-10 is still a great performance tool that offers a different slant on the usual ‘percussion pad’ routine.

It isn’t cheap, but it has a lot of power and potential under the hood if you’re prepared to invest the time in getting familiar with the editing and sampling processes. Oh, and I didn’t hit the screen while playing, so... stroke of genius!

“The MPS-10 is both powerful and fun to play.

The instruments and samples sound fantastic.” real-world playing situations. The CC Pads can add a real twist to a live performance and the looper is both flexible and easy to use. I particularly liked being able to use the CC Pad effects on external audio sources and incorporate live audio into loops.

On the negative side, I found the MPS-10’s LCD a little small compared to those of its competitors, and the information it offered felt limited. Both the Alesis Strike MultiPad and the Roland SPD-SX Pro offer an overview of the pad layout and sound assignment, which I find very useful and which the MPS-10 doesn’t have. I also found it odd that the LEDs beneath each pad could only display three colours preset to

£ £949 including VAT. W www.korg.co.uk

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Simulation Theory

How Virtual Instruments Work We cut through the jargon to explain how software instruments achieve their incredible realism.

CHRIS KORFF

R

eading some recent SOS reviews of flagship orchestral sample libraries from the likes of Spitfire, EastWest and VSL, I was struck by just how much jargon is involved in describing their features. Anyone new to the idea of creating a realistic orchestral composition in their

computer, or whose most recent encounter with ‘fake’ musical instruments involved comparatively simple hardware ROMplers and workstation keyboards, would struggle to understand what they’d be getting from a piece of software that could cost hundreds of pounds and consume an entire hard drive on its own. So, in this article, I’ll try and explain how these things work,

Virtual drum instrument BFD3 features up to 80 velocity layers per kit piece.

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so that the next time you see a review of Gigantic Trombones Vol 9, Enchanted Violas or Sensual Hollywood Banjos, you’ll know what the heck we’re talking about.

The Basics Most virtual instruments are built using samples — that is, audio recordings of the instrument you’re ‘virtualising’. (There are some exceptions, most notably those that use ‘physical modelling’; for more info see ‘Getting Physical’ box). So let’s start with a simple hypothetical virtual instrument: a solo violin. Because we want this to be the best virtual violin it can possibly be, we’ll hire a world-class session player to bring his Stradivarius into our studio, and play the clearest, crispest, most mellifluous Middle C known to humanity for us to record. Now, we take that sample and load it into a software sampler like Native Instruments’ Kontakt.

Hit Middle C on your keyboard, and you’ll hear that lovely note again. Congratulations on your first virtual instrument!

You might eventually consider playing notes other than Middle C, but don’t worry: all software samplers have the ability to pitch-shift audio, so that our violin sample can be played back at any other pitch.

This works well enough for notes a few semitones either side of the recorded note, but stray too high or too low and it’ll start to sound quite unnatural. So let’s call our session violinist again, and ask him to play another C, this time an octave higher. If we map the first C to Middle C on our keyboard, plus all the notes half an octave either side, and then do the same an octave up for the higher C, we’ll have two octaves of pretty convincing violin to play with.

Twenty years or so ago, when hard-disk space was expensive and limited, that’s exactly how it was done. But with storage space so cheap and fast nowadays, we can afford to be uncompromising, so why not go the whole hog and ask our session player to play every single note in the violin’s range, for maximum realism?

Dynamics This is exactly what most modern sample libraries do, and it allows us to play some relatively convincing solo violin lines from our MIDI keyboard, across the entire instrument’s range. But what if you want some notes to be played quietly and others loudly? You have two options. The first is to apply your MIDI keyboard’s velocity data to the synth-style controls in your software sampler, to modulate the volume at which the samples play back. You can even add a low-pass filter too, setting it so that lower-velocity notes sound duller than higher-velocity ones, mimicking the softer timbre you get when playing a real violin more quietly.

Again, this is very much how dynamics were handled many years ago, and it can work well enough, but since we’ve all got multi-terabyte drives and our sample library is still only a few megabytes, why not get our session player in again and ask him to play every single note once again, but at three different volumes? Then we’ll tell our sampler to play the quiet version of each note for MIDI velocities below, say, 50; the medium-volume samples from 50 up to 100; and then the loud samples for anything above that. These are what’s known as velocity layers and, again, most modern sample libraries use this technique to enhance realism. If we were selling our virtual violin, we might boast about it having ‘three velocity layers’, but you can use more — indeed, virtual drum instruments in particular tend to use a great deal more (BFD Drums, a division of InMusic, use up to 80 velocity layers per drum kit element). You can start to see why some of them ship on their own hard drives!

EastWest’s Hollywood Strings 2. On the right you can see faders for the different microphone options, and on the left is the ‘round robin reset’ button, which ensures consistent playback.

Getting Physical If recording 60,000 samples and spending weeks programming a sampler to play them just right in order to get your computer to sound like a violin sounds like a slightly brute-force approach... Well, you’d be right. And there is another, arguably more elegant way to go about it.

Physical modelling is the name for a type of synthesis that’s aimed at recreating the sound of acoustic instruments. Consider the violin: when a string is played, it vibrates, generating a pitched sound. But the vibrations of the string also travel through the bridge and into the soundboard, which resonates at a number of frequencies related to its shape and dimensions.

The other strings might vibrate in sympathy too, with more string vibrations travelling through the neck of the violin, exciting the back and sides of the body, interacting with the other vibrations travelling through the bridge, cancelling here and reinforcing there... And as that complex pattern of sounds radiates from the violin and into the air for us to hear, radiation from one part of the instrument will cancel or reinforce with sound emanating from another, and so on. Building up a mathematical model of everything that contributes to the sound of an acoustic instrument is an immensely complicated business, but many physical modelling instruments manage it remarkably well. Modartt’s Pianoteq, for example, offers realistic pianos, while Applied Acoustic Systems’ String Studio VS-3 uses the physics of stringed instruments as the basis for its sounds.

So what advantages does physical modelling have over a sample-based approach? Well, the obvious one is that a physically modelled instrument doesn’t require hundreds of

gigabytes of audio samples. Indeed, most physical-modelling-based instruments weigh in at perhaps a couple of hundred megabytes, compared with the many hundreds of gigabytes in a modern, multi-articulation sample library.

The down side, though, is that the maths required by physical modelling is much more demanding of CPUs than the comparitively simple job of playing lots of audio files at the right time.

The difference between physically modelled and sampled virtual instruments is somewhat analogous to algorithmic versus convolution reverb. Just as a convolution processor can give you the exact qualities of a specific acoustic, so a sample library made using a 1685 Stradivarius recorded in Abbey Road Studio 2 with a Neumann M49 will sound exactly like... A 1685 Stradivarius recorded in Abbey Road Studio 2 with a Neumann M49. But that realism comes at the cost of flexibility. In the same way that an algorithmic reverb will let you adjust things like room size, diffusion, damping and so on, physically modelled instruments usually let you adjust the parameters of the mathematical model itself — which allows for changes as subtle as a slightly duller or more strident violin, perhaps some more bow noise, or adjustable sympathetic resonance of the unplayed strings... or as radical as making your violin sound like it’s 16 feet long and made out of concrete. That kind of adjustability is largely absent from sampled instruments.

Some developers have sought to combine the two methods, for a ‘best of both’ approach — notably Audio Modeling, whose SWAM series woodwind instruments employ both sampling and physical modelling techniques.

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HOW VIRTUAL INSTRUMENTS WORK

NI’s S Series controllers have a Light Guide feature, which can be used to show you at a glance where your keyswitches are.

Cinesamples’ CineBrass instrument uses a custom GUI over Native Instruments’ Kontakt sampler. Shown here is the mixer page.

To get the most from our humble three velocity layers, though, we can even combine them with the MIDI-controlled filtering scheme I mentioned earlier. And, to avoid an obvious ‘switch’ in samples between notes played at a velocity of 49 and those played at 50, we can crossfade between them, for a smoother transition.

In The Round Our virtual instrument is now quite usable, but try playing repeated notes at the same velocity and your ears will tell you that you’re just hearing the same sample over and over again, rather than a real violin.

If this were a virtual drum instrument we’d complain about the ‘machine gun’ effect (it’s particularly audible on drum hits, but our virtual violin isn’t immune to the problem). One way around this would be to put our programming hat on again and tell the sampler to apply a tiny amount of

Signature Sounds In much the same way that Gibson have the Slash collection of Les Pauls, and Fender offer the Jimi Hendrix Signature Strat, many software instrument makers enlist the help of big-name artists and composers when building their instruments. For example, Spitfire’s Hans Zimmer Strings promises the bombastic sound of the legendary Hollywood composer; while UJAM were founded by the man himself, and their Symphonic Elements Drums was recorded using Zimmer’s own extensive drum collection. Other sample libraries employ renowned musicians in their sample capture (Native Instruments have just released a second Alicia Keys collaboration, and harpist Justyna Hebda features in Steinberg’s new Navia Harp), while others sell themselves on the pedigree of their engineers, for example EastWest’s Quantum Leap series, which was recorded by Grammy‑winning engineer Professor Keith O’Johnson.

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random pitch‑shift to successive samples, to mimic the imperfections that our human violinist might exhibit if we told him to play E7 eight times in a row. Better still, though, would be to drag our patient violinist back into the studio, and get him to play every sample — each note on the violin, at each of our three velocity layers — all over again. Then we tell our sampler to alternate between samples from the first session and the new ones we’ve just recorded, and now the machine-gun effect is eliminated.

This is known in the trade as a round robin. Our virtual violin has one round robin per note, because our session player got bored and left before we could ask him to record more, but a fully featured orchestral instrument could easily have half a dozen — and some of EastWest’s virtual string instruments have as many as 16.

Now our virtual violin — which started with just a single sample — already has close to 300 (six distinct samples for each of the 50-odd semitones in its range).

One potential, albeit usually pretty minor, issue with round robins is that when you hit Play at the start of your song, you won’t know for sure which of the round robin samples will play back. In practice this is rarely an issue (because sample library makers are pretty meticulous types and will have been consistent in recording one mezzopiano Middle C after another), but if you simply must hear exactly the same thing every time you play your composition, some orchestral instruments feature a ‘round robin reset’ button, to allow you to do just that.

Take A Bow So far, we’ve only asked our violinist to play straightforward notes at a variety of pitches and volumes. But the violin is capable of much more than that. You can pluck it (or play ‘pizzicato’), you can play harmonics, you can hit the strings with the back of the bow (‘col legno’), you can use the bow over the fingerboard (‘sul tasto’) or near the bridge (‘sul ponticello’) for different timbres.

The Sound Variations window in PreSonus Studio One. Sound Variations are PreSonus’ way of making keyswitches a bit more intuitive to work with.

Front Ends Most of this article has been about recording samples, but what about playing them back? In the hardware days, a sampler would do both jobs, but in software, it’s more common for ‘samplers’ to only concern themselves with the playback side of things. Probably the best known one is Native Instruments’ Kontakt, and there’s a huge number of companies making third-party content for it. Other major platforms include IK Multimedia’s SampleTank 4, Steinberg’s HALion and UVI’s Workstation 3 — all powerful sampling engines that have attracted legions of third-party developers.

Some companies, such as Spitfire, VSL and Orchestral Tools, have built their own sample playback engines (the Spitfire Player, Synchron Player and Sine Player, respectively). These perform much the same function, but allow the devs to use their own proprietary technology, and to customise the user experience more than using an off-the-peg sampler would allow.

These are all powerful pieces of software with all the features that both developers and users need to wring maximum realism from a sample set, so the choice of whether to buy one instrument or another will often depend on which ‘ecosystem’ you’ve already bought into.

The same is true for other instruments: a drummer might play with brushes or mallets instead of sticks; bassists can slap and pop (whether they should is another matter), woodwind players can use flutter tongue... In musical terms, as well as in sample library parlance, these techniques are known as articulations, and unlike for pitch and volume, there’s no realistic way to ‘fake’ these using MIDI messages to tweak the sample playback. If we want these playing techniques to be available in our virtual violin, we’ll have to record them.

By now our beleaguered session violinist is quite used to our phone calls, so in he pops to record his pizzicatos, harmonics and fancy bowing tricks, with multiple velocity layers and round robins. Our sample library has grown commensurately, and now has a different patch for each of these articulations, selectable from a drop-down menu in our sample player of choice. But wouldn’t it be nice if we could switch playing styles instantly, in the same way a real violinist might switch between bowing and plucking?

This is where keyswitches come in. Many multi-articulation sample libraries employ the very lowest or highest MIDI notes (those outside the range of the instrument) as switches, to tell the sampler which set of articulations to play. So, if you play C1 (the bottom C on an 88-note keyboard), you won’t hear an unnaturally low violin note; rather, it tells the sampler to use your ‘default’ bowed sample set. Play D1, and it’ll switch to pizzicato, and so on. Using keyswitches, you can play or program a realistic violin part with multiple playing techniques, just by drawing in or hitting the occasional extra MIDI note.

Clearly, this is a bit of a fudge: if you don’t have an 88-note keyboard, you’ll have to mash your ‘octave down’ button repeatedly to get to the right articulation keyswitch, before jumping back up again to the playable range. And even if you do have a full-size keyboard, remembering which key activates which playing style is far from intuitive, given that the keyswitches are unlikely to be in the same key as the music you’re playing. Native Instruments’ S Series controllers offer a helping hand here, by illuminating the relevant keyswitches in blue so you can find them easily, and some DAWs have

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83

FE ATURE

HOW VIRTUAL INSTRUMENTS WORK

A photo of one of the recording sessions for Spitfire Audio’s Abbey Road One series of orchestral instruments.

attempted to simplify things by including a dedicated keyswitch automation section (Expression Maps in Cubase, Sound Variations in Studio One), so that you can easily see and change your articulations, with useful names rather than simple note numbers, when editing your MIDI performances. Even so, some composers find it easier to just run multiple instances of an instrument, with one set to legato, one to pizzicato, one to sforzando, and so on. That

does require a lot of extra CPU and RAM resources, though, so if your computer can’t handle that many instances, you’re stuck with keyswitches.

Mic Options Our virtual violin is now feeling pretty complete: we can play and program a realistic-sounding performance of our violin, using a number of different articulations via the keyswitches. But what if the recordings don’t fit our production? Perhaps the violin was recorded in mono when we’d prefer stereo; or it was captured too close-up and lacks reverb; or the room we recorded it in didn’t have the best acoustics?

We could, of course, try to simulate a better acoustic using a reverb in our DAW. Or, we could build a reverb into our sampled instrument — and that’s something that many sample libraries do, often using convolution reverb for maximum realism.

Modartt’s acclaimed Pianoteq achieves highly realistic piano sounds without using a single sample.

But if we want true flexibility, the thing to do is to record the violin using multiple microphones in multiple positions, so that we can change the sound of the recording even after we’ve played or programmed our part.

Many large orchestral sample libraries offer multiple mic options — close mics, an ORTF array, a Decca Tree, distant omni mics, maybe a Mid-Sides setup — and these can usually be blended together in a mixer section to offer the kind of ambience your production needs.

Speaking of ambience, many companies go to the additional trouble of using world-renowned concert and recording venues to capture their samples: for example, Orchestral Tools recorded their Berlin Symphonic Strings library at the world-famous Teldex Studio; Spitfire Audio are regular customers of Abbey Road (their Abbey Road series of instruments covers the complete orchestral line-up); and Toontrack’s Legacy Of Rock expansion for Superior Drummer was captured at the legendary AIR Lyndhurst. And some virtual-instrument makers have gone even further, by buying their own studios: EastWest own the Bill Putnam-designed Western Recorders in LA (and naturally renamed it EastWest Studios), while Vienna Symphonic Library did much the same with scoring studio Synchronhalle (now Synchron Studios).

Wrapping Up And so, what started innocently enough as a single recording of Middle C has ballooned into a multi-gigabyte behemoth.

We’ve got samples for each of the violin’s

Further Trickery Extensive sampling will get you most of the way to a realistic-sounding virtual instrument, but there are some clever tricks to be played with sample manipulation and MIDI data too.

Your MIDI keyboard’s pitch and mod wheels can be put to good use, allowing for real-time playing of pitch glides and vibrato. The former is straightforward pitch‑shifting of the sample, while the latter is usually a simple LFO modulation of the sample playback speed, with the mod wheel controlling either the modulation amount, or speed, or both. While some sampled instruments will record vibrato notes as an additional articulation, the mod wheel/LFO approach gives you continuous adjustability, and allows for things like tempo-sync’ing to your DAW project.

Another neat trick is the use of ‘release samples’. These are additional samples that capture what happens when a musician stops

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playing a note — for example, a guitarist might intuitively rest their finger on the strings to stop them ringing, resulting in a quiet but audible ‘thud’.

Spectrasonics’ bass instrument Trillian includes such release noises, triggered by the Note Off MIDI messages that are generated when you release a key, and most virtual pianos do the same thing for the ‘hammer return’ sound when a key is released.

MIDI aftertouch can also be pressed into service, perhaps for controlling tremolo depth or speed, as an alternative means of controlling vibrato, or for adjusting playback volume so that you can play crescendos in real time.

Legato — a playing style whereby a run of notes is played ‘smoothly’, ie. without any gaps between them — is another area that’s ripe for playback tweakery. If we play our sampled violin in legato style on a MIDI keyboard, we’ll hear a bow attack for each note, which is not how a violinist would

play legato. So what we can do is program our software sampler so that, if one note is played before the preceding note ends, it cuts off the beginning of the second note and crossfades into it from the first, giving us a smooth transition and approximating the sound of a violinist playing two notes with one smooth bow movement.

It should be noted that some of the more fully featured software instruments now go the whole hog and record actual legato note transitions, so that when you jump from a C to a G, say, you’ll hear the whole C sample, then a brief sample of a violin jumping from C to G, and then the sustain portion of the regular G sample. Vienna Symphonic Library pioneered this ‘true legato’ approach back in 2002, and it now appears in a number of other orchestral instruments.

Similar trickery can also be used to achieve realistic trills, portamentos and glissandos.

Toontrack’s EZkeys 2 includes a virtual Bandmate, to help you generate suitable keyboard parts for your projects.

50 playable semitones; three velocity layers for each of those samples; a round-robin alternative for each of those again; half a dozen or so different articulations (each with its own velocity layers and round robins); and 10 different microphone signals for each note, articulation, velocity layer and round robin... Rather like the ancient ‘grains of rice on a chessboard’ problem, our simple virtual violin now comprises around 18,000 audio files! It’s also got an awful lot of clever stuff going on under the hood, such as MIDI-modulated filtering

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Grooves & Players A meticulously multisampled virtual instrument is all well and good if you have the keyboard skills to play it well, but for those who don’t, many software instruments now include a ‘performance engine’ to help you get the best out of them. For example, Native Instruments’ Action Strings 2 includes a number of preset phrases that you can trigger with just a single key strike, and even features an editor so that you can write your own.

Virtual drum instruments likewise often ship with a collection of ‘grooves’: MIDI performances played by real drummers on electronic kits, which can be used to trigger the underlying samples. Spitfire Audio’s The Grange, for example, features real drum performances from the likes of Chad Smith, Roger Taylor and Andy Gangadeen, so you can incorporate those drummers’ performances in your productions while still having the flexibility to, say, swap out drum kit parts or tweak the MIDI notes to fit.

Toontrack have gone even further, with the Bandmate feature found in their EZdrummer 3 and EZkeys 2 instruments. Drag and drop an audio or MIDI file from your DAW into the Bandmate window of any of those instruments, and they’ll automatically generate a suitable accompaniment.

Even better, you can share the auto-generated drum part with with EZkeys 2, which will then use the drum groove as the basis for its own playing.

Apple’s Logic now also includes some ‘virtual session musicians’, which you can read about in this month’s Logic workshop.

and pitch-shifting, keyswitches mapped to the different articulations...

And perhaps a nicely designed graphical user interface to hide it all behind.

So, next time you play a virtual violin, or program a sampled sousaphone, perhaps spare a thought for the recording engineers who had to capture all those files, the assistants

who had to name them all, the software engineers who had to program the samplers — and, of course, the musicians who played all those notes!

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85

COMPE TITION

Win! Electro-Voice Everse 12 Worth £1050

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September 2024 / www.soundonsound.com

BLACK

It’s coming, and you will hear it coming...

TALKBACK

Ramera Abraham WILLIAM STOKES

A

n engineering residency at Spotify’s Noteable Studio, under the mentorship of Berklee College of Music, helped Filipino-Canadian producer and engineer Ramera Abraham realise her passion and talent for vocal production. “What it really gave me was the opportunity to be alone, with an artist in the room with me. Just myself and another creator in a room, which was brilliant,” she reflects. Abraham has since worked with artists including Nick Cave and Warren Ellis, Elton John, Little Mix and many more.

At the moment I can’t stop listening to I’ve recently discovered that Beyoncé has released a cappella versions of her tracks from Renaissance, and I’ve basically been using those as study materials. I’m trying to listen to what effects she’s using, what kind of stylistic arrangements she’s going for, which lyrics she’s emphasising in those tracks... You can also hear where they have maybe gained up some of the plosives, which I think is interesting.

I’m listening to a lot of Elyanna: she mostly sings in Arabic. She’s like, the Arab Beyoncé. I’m really trying to listen to the inflections in her voice. The way she sings is so different to how we would stylise an English pop vocal, I think. So, I’m just trying to expand my horizons a bit.

Also artists like Victoria Monét and Tyler Lewis; with them it’s all about the vocal arrangements. With Tyler, for example.

I like picking out what her influences are, because I think that she uses a lot of what I would reach for as well. Obviously, she’s South African. So musically she’s going for a lot of amapiano, but I can hear lots of R&B in the way that she sings, in her intonation and everything. I think vocal production is 100 percent about the performance. It’s the performance from these artists that is going to pull you in

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to the song itself. The production could be crazy. But if the vocal performance falls flat, then I don’t think you have a hit, personally!

The first thing I look for in a studio Their vocal chain and their valve mic selection. That tells me a lot about the kind of work that studio does, as soon as I see the mic list. Analogue warmth on a vocal is a must for me, especially with the volume of pop and R&B records that I’m working on. So, if you give me a [Telefunken ELA M] 251, a [Neve] 1081 and a [Tube-Tech] CL‑1B, I’m a happy girl. I just love that expensive, gooey, full-bodied sound. That being said, you know, some of the greatest performances have been captured on, like, a [Shure SM] 57. And that’s cool. If I don’t have access to those things, it’s up to me as the engineer to warm up the vocal plug-ins, do my best with like what I have: with EQ, and in some cases now I’ll just bring the vocal home and run it through my own CL‑1B! Usually with live stuff, especially where there’s a lot of instruments involved, I’ll reach for a [Shure] SM7B. Because I think that will do the best job, period. But in some cases maybe the artist has said ‘I don’t like it! It’s too heavy, can I just have a [Shure SM] 58?’ And then I have to really remember that the most important thing is that the artist is comfortable. And if they’re uncomfortable, that’s first and foremost going to affect the performance. So if the artist is comfortable, I will absolutely get the job done with just a 58. And it’s surprised me more than once! The best mic is the mic that you have. You just have to be able to be flexible.

The person I would consider my mentor There are three, really. They’ve all played sort of different roles, but I’ll try not to make this too long. There’s my Dad, who has taught me a lot about being self-employed, because he is as well.

I grew up in a single-income household, but was blessed to have him around all the time. So I watched him work really hard for everything that he has. And I think that instilled something in me to also work really hard, to do my best.

The second one would be Kamille.

She motivates me to keep going. Even though there have been so many days where I’ve felt like quitting, she’s just always been there for me. She’s been my main cheerleader this whole time.

She knew me when I was a runner, sharpening pencils, and she’s just supported my journey ever since then.

And the third mentor, I would say, is Jerry Barnes from Chic. He is their bass player and he let me engineer a session with him and their drummer Ralph Rolle, one month after I had graduated. And I was honestly just bricking it! I was so nervous, but he walked into the studio just cracking jokes and wanting to have a laugh. Just saying, ‘Oh, yeah, let’s experiment with this. Let’s try that.’ And it just changed my whole perspective on how to approach sessions. And I just remember that day thinking, ‘Oh, my gosh, they’re going to be very picky. I have to get the sound just right’. But it was really the opposite of that. It was just about capturing the right vibe in the room. And that taught me the most important lesson.

And we’ve been in touch ever since.

My go-to reference track or album Anything written and produced by the ‘big three’ who, to me, are MNEK, Tre Jean-Marie and Kamille. Oh, also, anything produced by D’Mile. MNEK, Tre Jean-Marie and Kamille have worked on Little Mix, FLO, Mabel, Anne-Marie...

They’re all about that punchy, sassy vocal, which doesn’t hold back on performance; it’s a lush, thick arrangement that just goes, throughout the track. It’s pop hit after pop hit. When I was an assistant, that’s what I kept hearing, day in and day

The studio session I wish I’d witnessed Thriller! I want to see Quincy Jones and Michael Jackson in action. How did that concept come up? Thriller is just so iconic.

It tells a completely fresh and fictional story, and now that Linn Drum is used on every single pop record. And I believe that the Thriller vocals were recorded on an SM7. It’s just so iconic. All those records.

We hear a lot of those influences in pop records today, and that’s why everyone wants to know: what did they do on the original recordings?

I’d also love to sit on a vocal tracking session with Ariana Grande or Rosalía.

They both produce and record their own vocals, so I just want to be a fly on the wall and watch them both in their element and see what makes them comfortable.

The producer I’d most like to work with I’m gonna have to say Timbaland or D’Mile. Either would be a dream.

I think I would just learn so much.

out, when we were working on the Little Mix albums. And all of that just really stuck.

With D’Mile, obviously, he’s worked with Victoria Monét, Silk Sonic, Mary J Blige, Janet Jackson, to name a few! Anything he’s touched makes me pull a ‘stank’ face and want to dance. That is my own goal, that’s what I want to make other people do when they listen to something I’ve worked on. I love music that you can just get lost in and groove to. If I had to choose one track, I’d say ‘Leave The Door Open’

by Silk Sonic — I mean, who is not going to list that as their reference track!? Even though that obviously pulls from a lot of Motown stuff. But it’s that fat, tapey drum sound, the call-and-response sections, the top of your lungs singalong parts.

That gets me. Those are the things that I really try to achieve.

My top tip for a successful session Leave your ego at the door. Remember that everything you do is to serve the record and the artist’s vision, and that’s

it. Especially with with vocal production.

Being mindful of the language I’m using, the tone you employ when you’re correcting somebody or even when you’re just proposing an idea.

That’s so important. It all kind of just boils down to... don’t be rude to other people in your session! I have been in many sessions where I’ve been spoken down to by the producer, or the producer has spoken down to the assistant — in front of the artist. And all it does is create a divide. And as soon as that happens, you’ve lost the session.

Learning how to manage people in a session isn’t something that can easily be taught, I would say. But reading the room is important. Just remember that you have to preserve this environment where the artists and the collaborators feel safe. They need to feel safe in order to create their best work. And the way you act and treat people in a session is the thing that’s going to decide whether or not you get to come back.

The part of music creation I enjoy the most Speaking to the artist about what inspired them to create that record. I would say that’s an element that I love, because it informs so much of what happens afterwards. I love being almost... petty as well, with production. Say, there’s a song about a break-up. And let’s say, this person broke up with someone over text, I would 100 percent want to put, like, a texting ringtone or whatever inside the song. I’d want to try to mimic something to be a nod to the situation that inspired the creation of that song. That’s what I interpret my job to be: to take that information and just make sense of it, and then funnel it into the record on the day.

The advice I’d give myself of 10 years ago Believe in myself and trust my gut. I always had gut feelings about the direction that my life would take, and how good I was at something. And for some reason, I always just pushed that voice down, never listened to it. I think I’ve always felt like an underdog. But I’ve always had this sort of sixth sense. And when I’ve leaned into her, and I listen to her, I’m usually right about things. Or it takes me to another opportunity. And that’s amazing! So I think If I had learned to do that sooner, I would have a lot more self-confidence.

And I would have handled a lot of different situations better.

www.soundonsound.com / September 2024

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ON TE ST

Cycle Instruments Tetrachords Eurorack Module

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ustralian company Cycle Instruments have burst onto the scene with a very interesting and highly ambitious debut, the Tetrachords sequencer. A substantial module at 36HP, the Tetrachords promises big: polyphonic sequencing over both CV and MIDI, four sequencer tracks that can work separately or together and bounteous options when it comes to customising scales and chords.

To best understand the Tetrachords’ philosophy, it’s worth taking a closer look at its name. The word ‘Tetrachord’ literally translates to ‘four notes’, and constitutes, for all intents and purposes, half a diatonic scale. Traditionally they’re bound by a perfect fourth — so the first tetrachord of a C major scale, for instance, is C, D, E and F, and the second is G, A, B and C. Of course, when it comes to modular, traditions are there to be broken.

The Tetrachords presents its workflow as two rows of four buttons. Depending on the selected mode —

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Arp or Chord — these represent notes in a sequence or notes in a chord. Alongside each row of four is an encoder for selecting one of six scales: ostensibly, with both rows set to the same scale and order, and all steps engaged, the Tetrachords will play through that scale from start to finish. I say ‘ostensibly’ because even at this rudimentary stage the Tetrachords starts to open its behaviour up to wildly variable customisation and performance.

Neighbouring Spread and Amount controls for each row can create octave leaps (of multiple octaves, if desired) and dictate the order in which these occur, labelled with little dots and lines and a little white LED meter respectively.

A Reverse button switches the direction of travel; since there’s one of these for each tetrachord row, it’s possible to reverse the order of one but not the other. Ordinarily the forward motion of the transport is left to right on the top row (the first half) then left to right on the bottom (the second half). But a Path encoder on the right of the panel allows pretty much any route between these rows to occur: forwards then backwards, zig-zagging between rows, hopscotching back to the first note of the sequence after each step, quasi-random (with a re-randomisation button) and more. If that’s not enough a Roland

SH-101-inspired step‑sequencer mode allows user-programmable custom sequences of up to 32 notes as well.

The last piece of the Tetrachords’ fundamental architecture is perhaps the most rewarding to play with, at least in the first instance. The Root encoder, with its handy ring of LEDs to show which notes are in play, can be used to change key or to change the starting note of a sequence within the confines of a scale. This might not sound particularly creative, but whistling through iterations of a sequence across one scale is essentially akin to creating chord progressions and is a fantastic compositional tool. It’s the closest thing I’ve come across to the tone‑row-based workflow of the Eventide Misha sequencer, and if you have read my review of that module you’ll know what an accolade that is.

Now: all of the above applies when crafting one pattern. Above each of the eight main sequencer buttons is a little white ‘pill’ LED that represents the selected pattern, so program another pattern, or seven, and then use a pair of arrow buttons, or patch in some triggers, to cycle between them. In this sense, the Tetrachords offers a rather wonderful thing in an incredibly streamlined package: sequencing sequences.

Fans of the Teenage Engineering OP-1’s Finger Sequencer will be familiar with the creative delights contained in sequencing sequences — let’s dub it meta‑sequencing — and those delights are here in spades.

Each pattern can have its own slew settings, play as a chord or a linear sequence, have a different order and the rest. I found particular reward in setting the lot to Chord mode and working through progressions in orders that I would never otherwise have thought of.

This leads us to the topic of polyphony: the Tetrachords can work over MIDI or CV, and interface between them, usefully. It has four tracks, all of which can work together or independently to various degrees. When using CV alone, these can be multitrack — that is, can output different sequences to different destinations — or they can be used to stack up to four oscillators polyphonically.

Whew! Fair to say the Tetrachords is a rather esoteric module on first impression, and in many ways its slightly congested faceplate doesn’t offer the most intuitive layout. This is a complex module, and it feels like not much has been done to mitigate that complexity or give much of a foothold. That said, spend a little time getting to know it and things get much more fluid. And after all, you might say it’s part of the territory with something so incredibly powerful. The manual, if you’re interested, is 167 pages long.

Glasgow Synth Guild Oct Tone Eurorack Module

O

n a passing glance, you’d be forgiven for thinking this is a module from Glasgow-based Instruo Modular. However, Oct Tone is the first module from a new endeavour.

The Glasgow Synth Guild emerged from the remarkable growth of Instruo over the last couple of years, and while the cross-modulations run deep, they are their own thing. The Guild is a collective of Instruo builders and specialist designers who will be producing modules for other companies, as well as themselves.

The Oct Tone is the reimagining of the Octone, originally designed by Instruo’s Jason Lim for Qu-Bit Electronix back in 2016, and it’s an eight-step sequencer.

The interface is circular. It starts with a Rate knob sitting proudly in the centre. Surrounding that are eight knobs setting the pitch or CV level for each step. Shooting off from the eight knobs are eight light-infused pulse-activation buttons, and finally, eight gate output sockets complete the cascade. These four elements control the hypnotic dance of lights in our circle of sequencing.

Each knob sets the voltage for the step in the range of 0-5 V. This can be unquantised or you can engage one of the six scale modes to ensure some musical behaviour.

The modes are switchable from a button at the bottom left and reflected in some illuminated hieroglyphs that form an arc around the left side of the Rate knob.

Interestingly, the Tetrachords has no onboard clock, which might come as a surprise. Instead it expects triggers, over MIDI or CV, which on reflection I think is a very good move. The message from Cycle Instruments is clear: this is a tonal, melodic module, and it’s ready to get to work with your other, rhythmic modules. In that capacity it truly excels, and Cycle Instruments have managed, somehow, to stretch the already well-explored boundaries of sequencing in Eurorack.

If this is their debut, I can’t wait to see what’s next. William Stokes £ W

£615 www.cycleinstruments.com

The pulse buttons enable the sending of a pulse to the pulse output for the corresponding step. It doesn’t affect the sounding of the step, just whether a pulse is sent at the same time. This enables you to set up some rhythmic triggering of an envelope to a VCA or filter. The fact that they exist in a circle somehow gives it a different feel than a straight-line sequencer with gate switches. With an enabled pulse you also get a gate from the step’s individual output. This is something I found I had to work hard at in order to make it useful. So, once per cycle I get to trigger an event of some kind, or up to eight different events in a series as the steps go around. It does give you an alternative trigger to the main pulse output, and you just have to think creatively about ways to use it.

You can run a sequence either clockwise, randomly or with a ping-pong action. A separate knob controls the number of steps in the sequence, and it always starts with the step in the 10 o’clock position. A Retrig CV input and button lets you restart from step 1, and a clock input lets you sync to an external clock source, rendering the Rate knob superfluous. It would have been nice if the Rate knob could have offered a clock division or multiply function when clocked.

There are three modes of sequencing. Classic Mode is what we expect an eight-step sequencer to be, Tune Mode halts the cycle and lets you play the steps from the buttons like it’s a keyboard, and Pulse Mode is the only time things can get a bit ham-fisted. In Pulse Mode, you can set up note repeats for each step. To do this, you have to first

of all be in Tune Mode; then you select a step, hold the Retrig button and turn the Steps button to illuminate the circle of pulse buttons up to as many as you want the step to repeat (got that?). You can then double-tap a step to enable it for pulsed repeats or no-pulse repeats, which means the note is held for the same length without retriggering the pulse output.

When you double‑tap a button, the Scale button responds by flashing very quickly for enable, and just a fraction longer for disable. Once you’ve added the repeats, you switch back to Pulse Mode, and the sequence runs with the repeats in play. It’s just a tiny bit too difficult to get right every time. The double‑tap is tricky to do, and you end up flicking back and forth, which can kill an otherwise suavely spontaneous sequencing experience. However, it is a useful feature, and it’s really nice to be able to swap between Classic and Pulse Modes on the fly for throwing in some natty pulse repeat variations.

If you have some spare room, you can cosy up another Oct Tone to give you 16-steps. The steps run from one to another in a sort of misshapen circular motion. The combination also brings some additional functionality in the shape of transposition and the saving of sequences.

I enjoy a good eight-step sequencer.

There’s something about having eight notes in a cycle that hooks into the rhythms of our souls — well, mine anyway. The best ones tend to be easy, obvious, front-panel-led, playable and performable. The Oct Tone ticks all those boxes and has a visual flair that makes for a very happy sequencing experience. Despite the compact size and closeness of the knobs and buttons, I’ve found it a breeze to play with during a performance. It’s the most fun and fluid eight-step sequencer I’ve encountered to date, and the possibility of 16 steps definitely tempts me. Robin Vincent £ W

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www.soundonsound.com / September 2024

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Modular Profile: Josh Wilkinson

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usband-and-wife duos, siblings, parent-child teams; we’ve seen Eurorack companies grow out of all sorts of relationships, but it’s not often we hear of a guitar teacher and student teaming up to design modules. Brand new Australian developer Cycle Instruments is one such company, started by Josh Wilkinson and his teacher Nick Lavers.

Here Josh shares his own favourite modules, the story of Cycle Instruments and that of their impressive debut module: the Tetrachords.

On his entry into modular I got into modular after visiting Five G [Music Technology] in Tokyo and hearing the [Mutable Instruments] Clouds module for the first time. I still remember thinking, ‘If I can create this many sounds just by turning the knobs on this one module... what can this entire wall do!?’

I bought Clouds and a DIY PSU from Schneidersladen immediately when I got home. It was pretty expensive and I figured out that the DIY module kits were a lot cheaper so ended up just buying a bunch of those Hexinverter drum modules at the start, which forced me to learn how to solder really well.

On his go-to modules (aside from his own!)

I’m as basic as it gets. I still only have a 104HP system after all these years, based around the Doepfer A111-4 Quad Precision Oscillator. I like that oscillator because you can stack the voices and do the really thick analogue sounds that you can’t really do with a single oscillator.

Mutable Clouds is still my go-to effect module, I know that thing like the back of my hand. I also love my trusty Hexinverter Mutant Bassdrum that I built way back from a kit. The kick is now pretty sketchy, its pots are all touchy and scratchy, but there’s just something about it on a big PA that’s absolutely in another league.

I’ve tried to get rid of it a few times but missed it instantly.

On the story of Cycle Instruments Cycle Instruments started out as a progression of conversations between myself and Nick Lavers. Nick was my guitar teacher at school and we kept jamming after I left school, and we both got into modular around the same time.

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Coming from a ‘guitar jams’ background I think both of us were getting frustrated about how we had all this harmonic power in our cases, but it was very difficult to control and jam together using the sequencers we had at the time. I think Nick really brought it all together when he mocked up the first faceplate design using the tetrachord theory to build scales.

That’s when it all clicked and we got really excited. Then we just had to figure out how to build modules... which took about seven years.

On the Tetrachords I see the Tetrachords module as an instrument. It requires you to learn how to play it, have a dialogue with it, and it inspires me to think about music theory in very interesting ways. Western harmony is very interesting and beautiful but it is often pretty inaccessible unless you go down a traditional learning path.

For me once the module prototypes got further along I began just to sit with it for hours and move the melodies and chords against each other in ways that I liked, and began to really understand what different intervals and chords sound like.

I think the coolest thing about the module is that you can just follow your internal

taste when it comes to harmony and then learn — almost by osmosis — what you’re doing in a traditional music theory sense. I like to think the essence of Nick’s 20 years of teaching guitar somehow got uploaded into the module along with the code he wrote.

On the culture of modular I like modular culture. I think it has aspects that some people describe negatively, like the hoarding or the expensiveness.

But I think fetishising, or collecting for the sake of collecting, shows how passionate people are about the module designs and the platform. Which I think is cool.

I also like the almost cult-like relationships you can find yourself in.

I’ve cold messaged people while on tour and been invited into people’s studios for jams all over the world, simply based on the fact that we both are into modular. You don’t really get that if you play piano or guitar.

From a purely music perspective I am super inspired by people like Colin Benders, JakoJako and Barker who are playing out and using their systems as instruments, not simply as experimental sound boxes in the studio.

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ON TE ST

Sonic Charge Synplant 2 Software Synthesizer

Synplant 2 takes a more horticultural approach to synthesis than you may be used to...

JOHN WALDEN

T

here can’t be many products with a 15-year update cycle, but that’s how long fans of the original Synplant software synth have had to wait for v2. I have to admit, it was not a product I was familiar with, but the new release has caused a bit of a buzz amongst both users of the original and those with an interest in synth-based sound design. So, what exactly is Synplant and, 15 years after the original, what’s new in v2?

Biological Science Under the hood, Synplant is essentially a twin‑oscillator synth with FM and a flexible system for envelopes and modulation, a rather nice filter section, plus reverb, saturation, EQ and panning effects.

The core elements of this engine have been brought up to date in v2 in order to improve the overall sound.

However, as with the original release, Synplant 2’s UI and sound‑design controls are somewhat unconventional. With concepts such as planting seeds, growing branches, and DNA editing, designing sounds within Synplant might seem to

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have more in common with a biological research lab than a synthesizer. The original’s Bulb and DNA Editor pages have been refined but, by deliberately sidestepping a more conventional graphical representation of the knobs, sliders and buttons found on a typical hardware synth, the UI was created to make experimental sound design accessible to any user.

Sonic Charge have not undermined that ethos in any way, but a more conventional graphical representation of the synth’s controls within the DNA Editor is now available if you want it. And, with the addition of the very intriguing Genopatch system, there is now a whole new (AI‑fuelled) method of starting the sound‑design process. So, if you have got your wellies on, and a suitable spade to dig with, let’s plant some (sonic) seeds...

Sound Gardening Designing a new sound within Synplant requires you to plant a ‘seed’ at the centre of the Bulb display (for example, by simply holding Cmd/Ctrl and clicking on the seed icon). This triggers a randomisation process to provide some initial synth settings. By default, new branches then appear, growing outwards from that seed, each arranged upon one of the 12 notes laid out around the circumference of the surrounding circle. If you click and drag on one of these branches, it will ‘grow’ and, the further it grows from the centre, the more the sound changes relative to the original seed. You can also drag back towards the centre if you wish to reverse the process. The changing sound will audition as you drag, but you can also audition a branch by triggering the associated note from a MIDI keyboard.

If you drag a second branch, its sound will ‘evolve’ in a different way, and you

can repeat this process to see which branch produces the most promising sonic evolution. Find something you like, and the main menu then lets you create a patch from the currently selected branch.

There are various other neat options available within the Bulb window. For example, by default, you can grow all the branches together by moving the mod wheel. If you like the way a particular branch has evolved, you can take that sound and ‘replant’ it as the main seed and then start evolving away from that. Equally, holding Alt and clicking the central seed will clone the sound of the selected branch to every other branch, essentially giving you the same playable sound mapped across the full MIDI note range.

The Bulb page now also includes four different bulb ‘modes’ to control how the branches respond to MIDI notes (this includes an impressive new option for layering sounds), as well as the main sliders for tuning, overall effect level, volume and release (which interacts with the amplitude envelope controls). When growing new sounds, the Atonality control influences whether you get more conventional playable sounds with an obvious pitch (at lower settings) or more abstract/ experimental sounds (at higher settings).

Once you have grown a suitable sonic starting point, switching to the DNA Editor page lets you dig a little deeper.

Sonic Charge Synplant 2 €149 pros • Fascinating sound‑design workflow.

• Very capable synth in its own right.

cons • Perhaps not an obvious choice for bread and butter synth sounds.

summary It was worth the 15-year wait. The new Genopatch sound cloning feature adds to the cool and quirky sound‑design options but Synplant 2 is also an impressive sounding synth.

Alongside the DNA strand, the DNA Editor page now includes the option for conventional rotary controls for all the synth’s parameters.

This can be done via the rather beautiful — if somewhat abstract — DNA strand graphic, where each of the ‘genes’ (horizontal lines within the strand) acts as a synth parameter control. These are organised into three sections from top to bottom; envelope and LFO, oscillators, and filter and effects. However, new for Synplant 2, if you select one of these three sections, a full set of more conventional rotary controls for that section are then shown on the right side of the display.

A very useful text description is shown at the base of the window as you hover over a specific control.

Whether or not you fully understand what’s going on within the nuts and bolts of the synth engine, the sound‑design workflow is fascinating and intuitive. The results of this sound evolution process can be beautiful or bonkers, from playable sounds to weird sound effects. It’s both fun and inspiring to use.

a combination of synth engine settings to match the sound of the audio sample.

Once you trigger the process, the engine grows four different strands, each of which offers multiple versions of this sound matching process. Each possible match is represented by a small coloured dot within the display and clicking on any of these will audition the specific version of the synthesized sound. The toolbar’s Save button lets you save the currently selected sound match as a user patch, and you can also select the Plant Seed option from a drop-down menu; your matched sound then becomes a new seed for further evolution.

This is a fascinating process to watch unfold. Yes, it is unrealistic to load a two-second sample from one note (for example, from a guitar) and expect Synplant 2 to then generate a fully playable, totally realistic patch that both spans the full range of MIDI keys and offers the dynamic response of the original instrument. That

However, it’s probably the Genopatch sound cloning that most obviously grabs the attention. This is a really interesting concept. Surely other developers will soon bring their own take on this kind of process given just how quickly AI is finding its way into music technology software?

An obvious next step might be to allow the user to import two or more samples at different pitches or dynamics levels and then let the AI create appropriate synth engine settings that interpolate between these multiple starting points.

Hopefully, we won’t have to wait a further 15 years for Sonic Charge to offer that in Synplant. With AI moving at such a pace, my money would be on something closer to 15 months... Fingers crossed.

Synplant 2 is a great‑sounding synth that’s also a very accessible and downright fun to use sound‑design tool for folks that don’t usually spend their time doing sound design. OK, so the pricing might deter the more causal user, but the free three-week trial is well worth giving a spin.

Is it perfect? No. Is it cool, quirky, interesting, inspiring, and capable of some great sounds? Yes. When it comes to a ‘grow your own’ approach to creating synth sounds, Synplant 2 can supply endless sonic seeds to get you started. Very interesting stuff.

“Synplant 2 is a great‑sounding synth that’s also a very accessible and downright fun to use sound‑design tool.”

Genetic (Sound) Cloning Refinements and additions elsewhere aside, perhaps the headline new feature of Synplant 2 is the Genopatch process.

This allows the user to import an audio sample of their choice and, based upon up to two seconds of that sample, Synplant 2 applies its in-built AI capabilities (and not an inconsiderable amount of your computer’s CPU grunt during this phase of the process) to attempt to find

said, in practice, while the results are very source‑sample dependent, they can be both very good and very usable. That’s particularly the case if your starting point is a short sound such as a pluck or short bass tone, or perhaps a percussive sound such as a kick, snare, hi-hat or clap; it’s very impressive stuff. You can, of course, then do additional tweaking within the DNA Editor to finesse the ‘genetically cloned’ starting point that Genopatch has provided.

Synth Evolution Space precludes a discussion of all the refinements and additions to be found, but v2 certainly moves the Synplant concept forwards with its improved sound engine, the conventional controls option within the DNA Editor and the new ‘sound cloning’ process offered by the Genopatch.

The first two are impressive in their own right and would be enough for many long‑standing users considering upgrading.

For potential new users, Synplant 2 might not be an obvious choice as a ‘first serious virtual synth purchase’, but if you already own a few ‘standard’ synth emulations, this is a very cool curveball. Biological concepts within the UI aside, the sound can certainly get ‘organic’.

Via the new Genopatch feature, Synplant 2 will attempt to match the sound within an imported audio file as a starting point for your sound designing. £ €149 including VAT. W www.soniccharge.com

www.soundonsound.com / September 2024

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Vienna Symphonic Library MIR Pro 3D & Vienna Power House

Screen 1: A 3D view of the author’s test project.

Immersive Audio Reverb Suite In a collaboration with tech pioneers GPU Audio, an add-on to this multi-impulse-response immersive-audio reverb promises to take the strain off your computer’s CPU! TREVOR MICHAEL

T

his latest update to VSL’s MIR Pro reverb brings the company’s lauded Multi Impulse Response convolution technology to the immersive audio environment, but the developers didn’t stop there. Also available, as a cost option, is Vienna Power House, an add-on that promises to farm up

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to 50% of the processing out to your computer’s graphics processor, courtesy of technology from a company called GPU Audio. Given the heavy demands that immersive audio projects can make of a computer’s CPU, that really is an interesting prospect, and in this review, I’ll cover both MIR Pro 3D itself, and this novel CPU-saving feature.

MIR Pro 3D Overview Unlike in most reverbs, MIR Pro 3D’s processing takes place in a standalone room-modelling app that runs alongside your DAW, but audio is routed to and from the app by instances of a plug-in that are inserted on your DAW tracks in the usual way. You have a choice of two versions. The full version is called Vienna MIR Pro 3D, but considerably cheaper is Vienna MIR Pro 3D (24), which offers all the same features but limits the number of instruments/audio signals that can be placed on the virtual sound stage to 24.

Licensing is through iLok, and you’ll need a macOS (10.14 and up) or Windows (latest 64-bit version of 10 or higher) DAW that hosts AU, VST, VST3 or AAX plug-ins, as well as a graphics card supporting OpenGL 4.0 with at least 512MB of VRAM (more is recommended). VSL offer various system spec recommendations on their website, which change according to the scale of project you plan to undertake, but I tested the software on a 2022 M2 MacBook Pro with 16GB RAM, running macOS 14.5.

Upon opening the app, you’re prompted to select a ‘venue’ from one of the Room Packs that are installed. There are currently seven Room Packs available and you get one (of your choice) free with your purchase, but you can pay to add more and if you purchase more than one the most expensive of your selection comes free. Each Room Pack includes different rooms that the VSL team have captured meticulously and in incredible

detail, such that you can place a source in various positions in a room, and even define the direction in which it’s pointing; there’s more on that side of things below.

Pack 1 is the famous Vienna Konzerthaus, and this includes 11,000 impulse responses captured in its four halls and foyer. Pack 2, Studios and Stage, has 24,000 impulse responses from Berlin’s Teldex recording hall, the Austrian Broadcasting Corporation building in Vienna, and two rooms from Austrian recording studio Weiler. Pack 3, Mystic Places, contains 4200 impulse responses captured in various Gothic churches.

Pack 4, The Sage Gateshead, has 14,000 IRs from two large concert spaces in the North East of England. Pack 5 has 5000 IRs from the Pernegg Monastery in Austria.

Pack 6 features 6000 IRs from VSL’s own 540 square‑metre Synchron Stage in Vienna, where many film scores by composers including Hans Zimmer, Rupert Gregson-Williams and Laura Karpman, to name just a few, have been recorded.

Finally, Pack 7’s responses were captured in two prestigious venues in Salzburg, the Grosses Festspielhaus and Karl Bohm Hall.

Venue selection is organised by venue type (concert halls, foyers and special places, mystic spaces or studios and sound stages), reverb time and location, so it’s pretty easy for you to find what you’re looking for even if you have several packs at your disposal. Multiple venues can be opened simultaneously and it’s easy to move instruments from venue to venue too, making it possible to audition different spaces quickly for creative reasons. Once you’re happy with your choice of venue, you insert an instance of the MIR Pro 3D

Screen 3: The default plan view shows the room layout, but you can also view the room in 3D (as in Screen 1), and in either view you can place both instruments and mics very precisely.

plug-in on any track in your DAW that you want to send to the virtual 3D space, and that signal will show up in the app, in the venue you’ve selected. You can insert the plug-in on a mono or stereo track, and open it out in the app into anything from stereo to 9.1.6, including first‑, second‑ and third-order Ambisonics.

The Plug-in In the MIR Pro 3D plug-in, simple controls let you to get things where you want them quickly. A meter display shows the level being sent to each output channel. The bypass, mute and solo options here are self-explanatory, but there’s also a hide function that makes an instrument (or group of instruments) invisible in the MIR Pro 3D app venue display — useful when you’ve got 20 or 30 different instruments on a stage and the GUI is starting to become cluttered. There’s a dry/wet ratio (mix) control, as well as an overall output level control and a polarity inversion switch for the wet signal.

Bottom left of the plug-in pane is the room tone generator button, which plays background ambience for the selected venue. Simply throw up a spare track in your DAW, insert this and, hey presto, you get that little background hubbub ambience going on. This may strike you either as a gimmick or unnecessary attention to detail, but it’s actually very useful — often, when compiling live albums that play continuously from track Screen 2: Audio is routed to/from the main MIR Pro 3D app from a DAW plug-in, inserted in the usual way, and in this you have control over a number of basic functions, as well as instrument grouping.

to track, I’ve had to search for some ambience to play between some of the quieter moments because going down to ‘digital zero’ between tracks sounded wrong, and surprisingly disruptive. This feature, which VSL describe as a “natural dither”, overcomes that issue neatly.

In the centre of the plug-in pane, at the bottom, you can assign each instrument to a group. When instruments are grouped together, changing one parameter adjusts that parameter for all instruments in the group. It’s not an all-or-nothing option, though, as you have control over which of the following parameters are linked: volume, rotation, width, dry/wet, position, bypass, hide, mute and solo. So, for a standard string section, you might group your violins together, and the same

VSL MIR Pro 3D & Vienna Power House pros • Great sound.

• Good range of spaces to choose from.

• So much control over instrument and mic placement.

• Good integration with your DAW. • MIRacle is a nice bonus.

• Optional GPU power could be very useful.

cons • GPU power is a cost option.

summary A stunningly good reverb that takes us well beyond the realms of conventional convolution. The integration of GPU Audio’s technology to move convolution processing to your computer’s GPU could be a real boon for immersive audio professionals.

www.soundonsound.com / September 2024

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VSL MIR PRO 3D & VIENNA POWER HOUSE

Screen 4: A zoomed-in view of the MIR Control icon. As well as moving the icon to position the source in the room, you can grab and drag the white squares to change various parameters.

for your viola, your celli, and so on, and could then quickly place all instruments in a group on the stage, solo or mute them, or apply the same amount of effect.

App Basics All these controls and more are also available in the main app. At the top of its GUI are the usual main menu and command bar, with preset management, a help button and a useful reset switch to return the selected venue to its default state. Below this lies the instrument channel list, a list of all the sources being sent to the app from your DAW, and, to help you keep control of your session, you’re given basic hide, bypass, mute, solo, volume and group assignment controls.

When multiple venues are open, they appear as tabs along the top of the page and in the instrument list. From that list, the instruments can be dragged and dropped into the various different venues. Screen 3 shows the Vienna Konzerthaus loaded in MIR Pro 3D’s default view, looking down on the venue from above, while Screen 1, the main image at the top of the article, shows a 3D view of the venue, with some instruments on stage and the chosen mic cluster in the foreground.

You can see and tweak the position of the mics in the venue, and of the instruments (audio signals) on the stage.

The latter are represented by icons that can be moved around the stage and, in some cases (depending on the selected venue), to the back of the room — or even to the balcony! If you hold the Command (Mac) or Control (Windows) key you can zoom in and out of the venue and, when zoomed in, not only can you place instruments in the room with greater precision, but you can also grab an icon’s white squares and drag to adjust key parameters such as the dry/wet ratio, volume control and stereo width. Each icon also lights up to indicate when the instrument is playing (ie. when audio is

present), and there’s a directivity control that allows you to position the instrument so it faces towards the mics in the venue, or to rotate it anywhere in a 360-degree range.

Essentially, any audio source can be positioned almost anywhere in the room, facing in any direction with regard to the microphones that are capturing the sound in/of the room.

You can also move the virtual microphone cluster around the room, of course. Moving the mics nearer to the stage brings everything closer: sonically drier and more intimate. Moving the mics

a look at the specifics for each venue because it’s sometimes also possible to place things in more obscure places (like balconies or the back of the room, as mentioned). Other view options are to display a virtual grid (useful for lining things up), show/hide instrument titles, and focus mode, which, when selected, means the venue view will move around when you select/move an instrument.

Advanced Settings Hopefully you can see that MIR Pro 3D can be simple, but it can also be as complicated as you want it to be. Those who want to make more detailed tweaks will want to check out the right-hand side of the GUI, where there are three main sections: Instrument, Output and Venue. The Instrument section displays information for whatever instrument is selected, either in the main venue view or on the instrument list on the left.

At its most basic, you get a lot of the same controls you’ll find in the plug-in and on the MIR icon; there are a lot of ways to adjust the same parameters. But there’s also a very handy ‘find in venue button’ — great for when one of the second violinists gets lost at the bar in the foyer!

There are three drop-down menus in the Instrument pane, and the first, called Assignment, is interesting. Naturally, it allows you to select which of the open venues you want the instrument to be

“The strings in the studio really came to life.

I was immediately ‘in the venue’.”

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towards the back of the room results in more reverb. From a user point of view, it’s simple, effective and sounds very realistic.

Along the bottom of the venue view display are four further view options. The first highlights the room’s ‘hotspots’ — areas in which you can place instruments in the venue — with a yellow/orange glow; placing an instrument outside them mutes the audio signal. Hotspots always include the stage, obviously, but it’s worth having

Screen 5: While most work can be done in the main GUI, the Instrument pane gives you more than enough scope to place, assign and otherwise tailor individual sources.

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VSL MIR PRO 3D & VIENNA POWER HOUSE

placed in, but you can assign a profile for each individual instrument, and this is what sets MIR Pro 3D apart from any other IR‑based plug-in I’ve used.

My explanation could get a little complicated, but the shorthand version is that VSL didn’t just capture impulse responses for stage positions and mic configurations; they also considered the different instruments being played and how their sound is projected into the room, taking into account factors including instruments’ natural volume, frequency response and stereo dispersion/width.

The algorithms are based on VSL’s Vienna instruments but there are general profiles available for anything else, and a ‘character’ EQ option provides another way to tailor each instrument’s tone.

The final settings in the Instrument pane are Group, which I’ve discussed already (this is just another place where you can assign an instrument to a group) and Role. Whereas Venue presets cater for the listener perspective, Role presets handle the instrument perspective.

Clicking on Role opens another window containing factory presets for all VSL’s library of instruments and venues for each room pack, and these can be tweaked and saved as user presets for future use.

Another drop-down menu is dedicated to the dry signal. By default, MIR Pro 3D operates on the assumption that a spot mic in the room that’s set up for a particular instrument will follow that instrument: change the direction in which the instrument is facing, and the spot mic will move accordingly. There are various options for turning off or fine-tuning that, so that rotating the instrument away from the spot mics results

in the room very precisely on X, Y and Z axes. You’ll almost certainly find it easier to grab the MIR icon in the main venue view and position things that way, but I’m sure some will appreciate the greater precision here.

Following the Instrument section comes the Output pane. Here you can EQ the wet signal and select the microphone configurations. There are so many mic configurations available: anything from a mono mic in omni, to multi‑channel 9.1.6 organised into six drop-down down menus, 3D to stereo downmix, classic stereo, 3D surround downmix, classic surround, 3D immersive, and raw Ambisonics.

Each venue provides up to four microphone position presets. They operate on an X/Y/Z axis, just like the instruments, with X being the horizontal side-to-side position, Y horizontal front-to-back, and Z the height. Again, while you can place them precisely on these axes, the easiest way to position the mics is usually to drag them in Venue view. Two microphone configurations can be used and placed in different parts of the room. So, for example, you could have a mic at the front of the room and a different configuration further back, and you can specify a volume offset on the secondary mic configuration to blend the two together.

The third and final section on the right is the Venue pane, with three options (Classical, Scoring and Pop) for dry and wet signal handling. Classical mode is the default linear setting, whereas Scoring and Pop modes tend to sound a little drier, and there’s a Global

Screen 6: the Output pane, which offers control over the wet signal and the microphone configurations.

in different changes to the sound and volume of the instrument.

The Instrument section’s final drop-down menu is the Mixing + Spatialisation pane, with options for positioning and rotating each instrument

Vienna Power House: Using Your GPU My MacBook Pro’s CPU handled my test projects perfectly well with MIR Pro 3D running on its CPU alone. But if I were using multiple venues, lots of virtual instruments being processed and fine details being tweaked in terms of how each instrument placement reacts to the room, then two lots of microphone clusters capturing the room, thousands of IR’s being processed and taken into account every time an instrument or group of instruments gets moved around... There’s a lot of number‑crunching going on below the surface, and you’d be correct in thinking this is a relatively power hungry app. Add in what’s going on in your DAW, with all its other plug-ins consuming resources, and it’s not hard to see that you could soon reach the limits of what your CPU can handle.

This is where the clever new (cost option) Vienna Power House comes into play. Vienna Power House is described as a GPU Power Extension. From a GUI perspective, very little changes: you get one more button in the Output section that you can toggle on/off (Screen 7). But, courtesy of GPU Audio’s clever technology, switching it on moves all the convolution processing (a significant

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proportion of what MIR Pro 3D does) to your computer’s GPU. VSL tell us that: “an orchestra consisting of 80 stereo sources, mixed in 7.0.4... will create a maximum of 1,760 positional impulse responses and consequently the same number of convolutions (80x2x11), calculated in real time preferably at a low latency. By lowering the CPU load by up to 50% or more, depending on your system, Vienna Power House provides you with significantly increased raw computation power at the click of the GPU Power button.”

It worked flawlessly for me, but you must have a computer with the right specs. Mac users will need an M1, M2 or M3 (Pro/Max/Ultra) Mac, with a minimum 16GB of RAM (32 is recommended), running macOS Monterey (12.6.8) or later. Mac users of Cubase or Nuendo will also need ASIO Guard to be on. On Windows you’ll require certain video cards with a minimum of 8GB VRAM to take advantage of this option (Nvidia GTX 1000, RTX 2000, RTX 3000, RTX 4000 series, 2016/17 or later; or AMD RX 5000, RX 6000 series, 2019 or later). An Nvidia RTX 3060 or AMD RX 6700XT with 12GB of VRAM are recommended.

, mashers, , thumpers rs e ll e y , rs e p ders, stom kers and boom-chic mers, shred , m rs e ru v st ri , -d rs e e esign , pil plink in. Grace D rs, noodlers , pluckers, e e rs ff m e ri o k , c ic rs e p e w p u p yo ere oners, cho es. You’ll d that’s wh rinners, cro s of all strip n etime. An ia m c o si s u m in grassers, g g ic otta plu s for acoust g on. ers, y’all g le preamp ’re standin b u a o li y re e , n song-sling g o in e t sound ybe even th ously grea where, ma ry e v builds fam e t u o st ab onstage ju find them

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www.gracedesign.com

ON TE ST

VSL MIR PRO 3D & VIENNA POWER HOUSE

was tracked three times for a bigger sound, and some samples were used to augment the arrangement. Each instrument in the quartet was spot‑miked, and each of the three recording passes also had a stereo room capture.

I started experimenting by sending the spot mics and sampled strings to MIR Pro 3D’s Vienna Konzerthaus Grosser Saal venue (Screen 1). Initially, I was working on headphones and monitoring through the Dolby Atmos Renderer in Binaural mode, and the results were very satisfying. The strings sounded huge: instant, cinematic film-score vibes from just four people tracked a few times in a reasonable room!

I could move them around the stage, placing things exactly where I wanted, and could easily set a good blend of wet and dry signal. So far so good.

I decided to bring in the piano and place the lead vocal at the front of the stage, which worked really well for this arrangement. In fact, if I’d been mixing this new stripped‑back arrangement of the song for real, I’m sure I could have happily made all my changes in MIR Pro 3D and finished the mix without having to return to my DAW. I then took off my headphones and worked in my Dolby Atmos 7.1.4 studio.

Screen 7: The Venue pane allows you to tailor the venue to suit different material and, if you’ve invested in the Power House add-on, is where you switch GPU processing on.

Dry/Wet Offset slider below. The helpful Reverb Time Scaling shortens the venue’s decay, and it’s great if you find an almost perfect space for a particular project but find the length of the room makes things sound just a little too ambient for some tracks. Finally in this Output section is another way to add/control the aforementioned room tone.

It sounded incredible! The strings in the studio really came to life. I was immediately ‘in the venue’. The room captures are so good if you’re mixing with stereo stems in Atmos, which I do a lot — you could probably get great results using MIR Pro 3D to upmix selected stems, just spreading them around the room and shortening the reverb times to taste. Even some quirkier positioning could be useful too. Moving the piano to the back of the concert hall for fun, for example, resulted in the piano sound becoming duller, obviously behind you and a little further away. (I guess just like moving the piano to the back of the hall!)

Placing the close drum mics in another venue, the Weiler Studio live room, required a little more ‘fiddling’ to arrive at the desired results, but nothing too drastic; it was simple enough to dry off the kick drum to keep it punchy, while adding a little more snare and toms to the room.

Moving the mic cluster toward the drums in this room really pulled them in, though, to retain their energy, clarity and punch.

Opening another venue, ORF Studio 2, I put my electric guitars in there, and then threw my backing vocals and percussion back into the Vienna Konzerthaus Grosser Saal, with some group vocals on the big choruses at the end placed right at the back of the room.

By now, I had 45 channels of mono or stereo sources coming out of Pro Tools

In Use MIR itself might not be new, but I must say that having this sort of technology available for immersive audio is a game‑changer for those of us who mix in this format. VSL say that, with the number of IRs captured for each room, MIR knows how each instrument type reacts to each given venue, and certainly having a virtual space like this doesn’t feel the same as simply sending instruments to a multi‑channel reverb. Just as importantly, though, VSL have chosen to recreate some truly amazing venues, and it’s super easy to use.

For the sake of this review, as well as playing some orchestral instruments through MIR, I revisited a song I mixed a few years ago, a standard band setup, with drums, bass, guitars and keys with a male vocal. We’d also recorded a string quartet in Studio 1 at Echo Zoo Studios in Eastbourne — a reasonably sized live room, about 6x12 metres with four-metre high ceilings. The quartet

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Screen 8: Using MIR Pro 3D to place non-orchestral sources in different spaces.

It’s A MIRacle The MIRacle plug-in — a nice bonus!

There’s a really nice bonus that comes with MIR Pro 3D: a very useful multi-channel reverb plug-in called MIRacle, that’s independent of the app and can be used in a DAW as you’d use any regular reverb.

You can place it on a mono or stereo channel and open it out to anything from stereo to 9.1.6 channel widths. It has controls for all the usual

into MIR Pro 3D and being returned to Pro tools in 9.1.6 channel widths, feeding the Dolby Renderer. All running on my CPU — but there will be more demanding projects out there, and for those the GPU Audio option might well be worthwhile. You can read more about that in the separate box.

One last important observation is that when folding things down to stereo in the Dolby Atmos Renderer, quite often when using conventional multi‑channel reverbs things can get a little cloudy and muddy— here there was still clarity and detail in the mix.

Verdict When mixing in Dolby Atmos, placing the instruments in MIR Pro 3D in this way makes a lot of sense. You position your mics in the venue and place your instruments around them. It won’t be right for every project, as, obviously, you’re adding ambience to everything, but if starting a mix from scratch rather than trying to match a stereo one, MIR Pro 3D offers a good, enjoyable way to work.

parameters, including pre-delay, length, modulation level, modulation frequency and stereo width, as well as separate pre-delay, length, volume and damping controls for the height and rear channels. You can also mute any of the channel outputs. It sounds great, has a good selection of factory presets, and is very usable.

You don’t have to delve too deep, either. It’s super easy to get going and achieve excellent results, even if geeky engineers (guilty as charged!) could spend hours selecting and positioning mics and instruments before getting any ‘proper’ work done! When you do feel the need to take things further, VSL’s website has plenty of tutorial content to help you out.

I do lots of mixing work in Dolby Atmos, and already know that I’ll be using this app a lot. Being able to place instruments within such a natural-sounding 3D space, with this degree of control over both the instruments and the space is very effective, and as I said the sense of clarity when folding back down to 2.0 was welcome. The room packs available so far constitute a good selection of very usable spaces, but I’m sure there will be more to come in the future.

If you work with classical music, film score production or orchestral music, MIR Pro 3D is something you’ll probably want to check out — but if working with those genres in immersive audio, you owe

it to yourself to check it out! For those operating more in the pop/rock fields and stereo, it could still be useful, although the benefits of the attention to detail in the venue captures, especially within dense mixes, might be lost on you.

None of this is inexpensive, of course, especially if you want those extra Room Packs or the Vienna Power House add on. But the 24-channel version brings the prices within reach of mere mortals, and there are good discounts for upgrades from previous MIR versions. Thankfully, there’s a free 30-day demo of all the software, including Vienna Power House so you can try everything before you buy — and you really do get what you pay for.

This is top-notch stuff.

£ MIR Pro 3D €745, MIR Pro 3D (24) €355,

Vienna Power House €165.

Prices include VAT. T Best Service +49 (0) 89 4522 8920 E uk@bestservice.de W www.bestservice.com W www.vsl.co.at

www.soundonsound.com / September 2024

103

INSIDE TRACK

Taking the decision to follow his instincts has elevated Julian Bunetta to the top rank of contemporary songwriters and producers.

PAUL TINGEN

Julian Bunetta

“F

or a lot of my career, I tended to follow the rules of what you should and shouldn’t do. You set up your busses this way, you find one reverb space, you never have too much on the master bus, and so on. But at a certain point, I threw that rulebook out. Now I’m just splatter‑painting on the canvas. I’m not comparing myself to Picasso, but if I want to, I put the nose on the forehead, because it’s fun. I’m constantly trying to experiment and trying new things. I like to mess stuff up and see what happens and stumble on happy accidents. The only way you can do that is by trying something you haven’t tried before.”

A few years ago, Julian Bunetta also decided throw out the rule book with his career. This led directly to him having writing, engineering, multiple instrument, mixing and production credits on one of last year’s biggest hits, Teddy Swims’ ‘Lose Control’, and now on Sabrina Carpenter’s huge hit ‘Espresso’.

Stop Chasing The moment that divided his career into a before and an after phase came in 2019. “I’d had a lot of success working with One Direction from 2012 to 2015, and then global hits with Niall Horan and Rudimental. In 2018, I moved from Calabasas near Los Angeles to Nashville, and had three or four number ones with Thomas Rhett and country. Then, in 2019, when my daughter was born, I had to figure out again how to work. I used to be in the studio until 3am, and all of a sudden it was clear I couldn’t do that if I wanted to spend time with my family.

And then, obviously, the pandemic hit.

“I didn’t do much Zoom writing, it just isn’t for me. Instead I went back and learned my favourite old songs on the piano, by Babyface, Diane Warren, and so on. I dug into synthesis and how to actually work these plug-ins that I have.

I’m not a gearhead, and I normally just turn some knobs, and either I get a cool sound or I hit the next patch. I wanted

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to go back to the fundamentals of what I was doing, and why. I wanted to reframe how I worked, and also who I worked with.

“I didn’t want to chase things as much any more. In the songwriting community in LA, you’re working with a different artist every day or every two days, because the more people you work with, the more of your songs are cut. But that never worked for me. When someone’s really, really famous, you don’t get as much time in the studio with them. I decided I wanted to do less speed dating, less writing with different people every day. I had done that since I was 19.

“Instead, there was just a slowing down. I wanted to work with friends that I trusted and loved.

The thing that I love the most is sitting in a studio with them and hashing out lyrics and hashing out a production and musing over what we should do on the next song.

All my success came from working with people I spent a lot of time with, where I really understood who they are, and got their vision.

I wanted to work more with new artists, because I found that some of my most enjoyable times were when creating music with artists who were game to do anything, who want to try new stuff, who are hungry.”

A Family Affair As the sons of well-known drummer and producer Peter Bunetta, known for his work with the Temptations, Donna Summer, and many other legends, Bunetta junior and his brother Damon grew up among the stars. The brothers and their father later founded a publishing business called Big Family (now under the umbrella of Hipgnosis Songs), which continues to provide the framework for Julian’s career.

“My mission statement during the pandemic became: ‘Who do I really want to work with?’ I said to Damon, who is my manager, that I wanted to work with people who are interested in working with me, because of the way I like to work. So we set out on a journey to find new people to work with, and help them with their creative process. We found a few artists that I really enjoy working with, and I decided to dedicate all my abilities and focus on them, and not

I later played on guitar in my studio in Los Angeles, and we ended up chopping it to half the length.

“Teddy co-wrote the song, but by the time it came to recording his vocal part, he had lost his voice. So someone else sang the demo. The demo was bare bones, and when I came home, I started producing it up. When Teddy came to my studio in Los Angeles to lay down his vocals, he did a lot of takes to get it right, because he had to adapt to the new production, and make the song his own. He had to unlearn what the other singer had done and get that out of his head. He’d sing it again, and a nuance would happen and I would be like, ‘Oh, that was cool, that was yours. Let’s do it again and build on that.’ We tried out many different things, and the magic started to happen.”

On Song

Julian Bunetta has been playing drums since he was a child and frequently plays on records he produces.

spread myself thin any more.

“Teddy Swims was one of the up-and-coming artists we found.

I started working with Teddy in 2020. ‘Lose Control’ was written at a songwriting camp in Palm Springs.

I love songwriting camps. There are no interruptions, no meetings, no lunch appointments, no family obligations.

We do a lot of these camps, and we go somewhere remote with a selective crew of people. We just write and we write, and don’t worry too much.”

‘Lose Control’ bears writing credits for Swims, Joshua Coleman (aka Ammo), Marco Rodriguez-Diaz Jr (Infamous), John Sudduth (Mikki Ekko) and Julian Bunetta, with the latter and Ammo co-producing. “That song was based on a rough demo by Ammo. Infamous and he developed it into the three-time musical bed. The demo was just an MP3 that I dragged into Logic and recorded over. Later, when I got home, I got the stems from him, and then I added more things to the arrangement. The guitar solo was a mouth solo by Ammo that

Bunetta’s credits on ‘Lose Control’ also include drums, bass, guitar, keyboards and vocal production.

While some of the instrumental parts are programmed, and Bunetta is comfortable with the modern in-the-box, click-in-the-notes approach, he likes to play actual instruments. “Because my dad was a drummer and producer, I went to the studio after school, and grew up there.

I’d do my homework and witnessed session after session, with camaraderie and laughter. It was an environment that I gravitated towards and loved. It was my secret place that no‑one else knew. It was a sacred space. There was always a drum set in the house, and I simply knew how I’d play drums when I was old enough. I never learned. My dad fostered all my musical inclinations, and I played in jazz bands and orchestras and Summer bands and so on.

“When I was in high school, in ’97-’98, I was doing demos for songwriters in LA that were my dad’s peer group, people like Tom Snow, Cynthia Weil and Andy Goldmark, who had had hits in the ’70s and ’80s, and were looking for someone to update their production. They were recording just piano/vocal to cassette tapes. That obviously wasn’t going to cut it, so I would produce these songs in Logic, and initially I’d hire a background singer or a guitar player, but then I’d play things myself. If you’re given $350

www.soundonsound.com / September 2024

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INSIDE TRACK J U LI A N BU N E T TA

Bunetta’s studio in Calabasas is called the Perch.

to do a production, you don’t want to give away $200 to an instrumentalist.

This is where I started also playing bass, guitars and keyboards.

“When I was 19, I enrolled at the Berklee College of Music, as a jazz drummer. After a year, Warner Chappell offered me a publishing deal. I left school, and spent the next 10 years or 11 years cutting my teeth, learning the craft of writing songs. Before that, I had been more into producing. I loved hip-hop, and wanted to make beats like Dr Dre and Timbaland. But I then did the grind of songwriting with tons of different people. I really fell in love with writing lyrics and I started singing, so I could write melodies. The things that kept me up at night were not the guitar sound or the reverb or the mix, it was the lyric in the chorus or the melody.

So my focus really shifted to the song.

At the same time I was still engineering, mixing and producing.

“I was doing well, but the songs that I was involved in writing did not become

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singles, and when they did, they did not become hits. My brother and I had met this young kid named John Ryan, and in 2010 he joined our company. The chemistry John and I have unlocked everything. He was obviously the missing component. Around this time I was asked to work as a producer for The X Factor, and this led to an A&R asking me to write for One Direction. He paired me with Jamie Scott from the UK, and I brought in John Ryan. John and I ended up being involved in the writing of three songs on the Take Me Home album [2012].

“When the next One Direction album came around, the label flew us out to England to write for Midnight Memories [2013]. Magic stars aligned, and we wrote all the singles and the title track of the album, including ‘Story Of My Life’ and ‘You And I’. It was one of those magic moments. We also did 10 songs on the Four album [2014] and nine on the Made In The AM album [2015]. After that I did songs with Harry Styles, Niall Horan, Rudimental, Thomas

Rhett, and others, and I moved to Nashville and got married.”

Home From Home In addition to his studio in Calabasas, the Perch, he also set up a studio in Nashville, called the Nest. The latter is a huge space, with a kitchen at one end, a sitting area and various keyboards and guitars in the middle, and the studio gear and a drum kit at the other end. It’s set up for live playing, says Bunetta. “When I worked in Nashville in 2019 and 2020, I learned a lot from working with Dan Huff and other country producers and musicians. Moving here reintroduced me to five musicians sitting in a room and playing together, and watching a song go to unexpected places that I never would have gone to if I was by myself in my studio. After doing a lot of the country sessions, I wanted to incorporate that more into pop. I am a musician, I come from that, and I wanted to find a world where you fuse that with programming. So you

don’t just have a 100-bar loop, but there and the Kala acoustic-electric U-Bass, are ebbs and flows and dynamics. which is one of these little rubber-string “My studio is above a garage. It’s guitars. You can make it sound like an perfect, because I have everything in one. upright bass, or like a Motown bass.

The guy who’s been mastering all my stuff They’re amazing. I record them DI. for the last 10 years, Nathan Dantzler at “I have several keyboards here, the Hit Lab, helped me design the room including an M-Audio MIDI controller and get the treatments. I have the PMC in front of me, and the Roland Juno MB2S XBD monitors here, the super and Jupiter. In LA I have the Sequential big ones with the cabinet, and they’re Oberheim OB6. But I’m not really great. They’re too big for the Perch, a synth guy. When I’m in the middle and because John has PMC 8-2s in his studio in LA, I’m thinking of having the 8-2s fitted in the Perch, and here as well, so that my two studios and John’s studio have the same monitors.

“I started with Logic and Pro Tools in ’98, but the problem was that the latter was of writing, I just need speed. I don’t terrible with MIDI at the time. I didn’t have want to stop the flow to try to get microphones and stuff in my bedroom a synth sound. And until I find a synth when I was 17, so I had to do everything that I can operate super quickly, with samples. The EXS24 sampler I just stick to my go‑to VSTs, like Xfer changed the game. I went fully Logic, and Serum, Spectrasonics Keyscape and I’m still very happy with it. I use it with Omnisphere, NI Kontakt, and so on. the UAD Apollo 8XP interface, which we I’ve used them all. Sometimes I try new have in all our studios, for consistency. ones, to get me out of a habit. When We apply the same philosophy with the I’ve used something for too long, I stop mics. I have a Lewitt LCT 1040 tube/ being creative with it.

FET condenser microphone here, and the LCT 940 in LA. When I go to writing camps I bring Aston Spirits, because they sound great. I bring two in case something happens to one, and sometimes we have two rooms.

The mics go straight into the Apollo.

I don’t go through anything else because, again, I need to make sure that if we start something in a writing camp, or in Nashville, and then end up in LA to finish it, it sounds the same.

So the easiest thing is just to go directly Much of the equipment from into the Apollo 8XP. the Perch is replicated at Bunetta’s second studio, the Nest, in Nashville.

“I have Hofner and Bacchus bass guitars,

“Plug-ins that I like to use include the Valhallas, which are obviously a standard. I also love the UAD EMT 140 and the new Soundtoys SuperPlate. The FabFilter stuff is so good, and I like the iZotope Trash, as it pushes things forward. I also use the Logic plug-ins a lot. I’m reckless with plug-ins. I just pile plug-ins on and twist and mess them up and move the order of them, to see what it does. I don’t set up my sessions with dedicated sends to a big room and a small room and a short delay and a long delay. It’s more fun when I just throw something on.

“Being reckless yields unpredictable results, and sometimes it’s great and sometimes it sounds like a mess.

It’s the same with my master bus. It’s different every time. Sometimes it’s just a FabFilter Pro-L 2. I use iZotope Ozone 10 a lot. The Imager is amazing, and the multiband compression is great.

Sometimes I go crazy. There are some records where it’s OTT, with Ozone and then SketchCassette II and then another Ozone and then a Pro-L 2. It depends.

“Nine times out of 10, I know I need to pass a mix to Serban Ghenea or Josh Gudwin, because they’re going to squeeze something out of it that I’m not going to. But there are some very rare cases when I know that a mix is just right, and there’s no need to go further.

You’re going to end up asking the mixer, ‘Can you make it sound like my rough mix?’ I hate that, and they probably hate it, too. What’s the point? In any case, Jeff is an amazing mixer. And at 41, I trust in my ears and my own judgment.”

Enough, clearly, to help create major hits with the rule book gathering dust in the bin.

Julian Bunetta: “I’m reckless with plug-ins. I just pile plug-ins on and twist and mess them up and move the order of them, to see what it does.”

www.soundonsound.com / September 2024

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ON TE ST

Grace Design Rex & Roxi Microphone Preamps

These classy preamps combine exemplary technical performance with some unusual stompbox‑friendly features.

NEIL ROGERS

G

race Design are a long-established pro audio company known for their high-end preamps and monitor controllers, and their products are typically targeted very much at the professional studio. The products we have for review here reveal a more playful side, combining a miniature studio channel strip with features and functionality more commonly seen on a guitarist’s pedalboard.

Grace describe the Rex and Roxi as the ‘ultimate solutions’ for performers who like to control their own mic output on stage while adding effects using guitar-style pedals. Such users can benefit as much as studio engineers from a high-end preamp and a quality EQ to shape their sound.

In further evidence that these products may have been conceived on ‘dress down Friday’ in the Grace office, practical examples given in the product literature include horn players, harmonica shredders or one-mic bluegrass bands who like to play with their own effects on stage. I think it’s fair to say that these are rather niche applications, though, so I was intrigued

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to explore what these unusual products could offer in a more general-purpose studio context.

Roxi Music The Rex is the little brother of the Roxi, and has a slimmed-down set of features and a correspondingly lighter price tag.

I’ll talk you through the larger Roxi first before explaining what the Rex has and doesn’t have in comparison. The first thing that stuck me when getting my hands on the full-featured Roxi was just how solid it felt — with its metal casing and ‘roll bars’ it has very much been designed to withstand the demands of the most chaotic of stages.

At its heart, this product is a professionally specified mic preamp, and features a full-fat implementation of one of Grace’s highly respected circuits. It allows for XLR mic or instrument jack input, with 48V available on the mic input courtesy of a small push switch on the back of the unit. There’s 60dB of gain available for mics and 40dB when used with a jack input, with an additional 10dB available from a variable Boost circuit that can be controlled by one of the stompbox-style footswitches.

Output-wise, we have a balanced XLR output as well as a jack output. These can operate simultaneously, and the jack output has an independent level control labelled Amp. We also have an output for a tuner, as well as jack inputs and outputs for a dedicated effects loop, which can be used in parallel thanks to the onboard dry/wet control. Alongside this creative functionality, the EQ section on the Roxi is impressive for such a small device, featuring low and high shelving filters alongside a sweepable midrange that can cut or boost from 175Hz up to 4kHz, by up to 10dB. These are the main controls covered but there’s also an additional DIP switch section on the side of the unit, which offers options for engaging a high-pass filter (at either 75Hz or 150Hz) as well as switching the output of the device between line and mic levels.

If that’s not enough, there’s also the option of switching the second footswitch to act as either a boost control or for inserting the effects loop into your chain. Power is supplied either from a standard IEC mains socket or a 9V power supply of the kind more commonly seen on pedalboards.

Rex Appeal The Rex is much smaller than its counterpart but still packs in most of the key features, just with fewer options. The unit can only be powered by a 9V DC power supply, but features the same preamp with the same 10dB boost feature. We also have the effects loop input and output connection, but without the parallel blend control and the footswitch-controlled insert option.

It’s the same deal with the additional amp output, in that you still have the option, just without the separate level control. EQ‑wise there’s a fixed 75Hz high-pass option as

well as the high and low shelving filters, but there’s no mid band as found on the bigger model. It’s not all ‘less’ with the smaller option though: many users, like myself, will prefer having the controls for 48V phantom power, ground lifting and the high‑pass filter available on the main body of the device, rather than having to reach around the back or engage a DIP switch.

In The Studio I’m predominately a studio engineer, so I was keen to see what these clever boxes could offer me when recording musicians and singers in a studio setting. However, as I had both models for review, I thought I would lend the Rex to a friend who performs live with acoustic instruments using pedal effects on stage, and then collect their thoughts here.

Starting with my own experiences, the thing that struck me first was the quality of the preamp. I’ve used Grace preamps before but I was still taken with how ‘clean’ and vivid my old Neumann U87 sounded on a vocal session. It was a similar story with my Shure SM7 dynamic mic — which needs a lot of gain — in that it seemed to bring the very best out of what can be a fussy mic, without needing a Cloudlifter or other boosting device. The EQ section has more than enough options for tracking sessions, and the low‑ and high‑pass filters worked superbly for adding a touch of weight and fullness or ‘opening up’ the top end. Being able to then also target specific ranges with the mid band seemed like a real bonus.

Putting the extra features to use, I began by connecting my RE‑201 Space Echo to the effects loop, and it was great being able to blend just the desired amount of echo effect using the parallel wet/dry control. I could also choose to send the output of the effects loop to my patchbay, so any effects used could be printed on a separate track, and also sent to a singer’s headphones mix, for example.

I found myself continually surprised by how useful these boxes were in my studio. On one session, in particular,

I had both out for tracking a jazz outfit in a live studio session. The double bass player had a fixed pickup in place on his instrument, and using the jack input on the Roxi, I was able to connect his output effortlessly into my setup knowing I had a good preamp taking care of things at source. I also added a touch of EQ, and the performer took advantage of the amp output to feed a tuner. It all worked superbly. In the same session, the trumpet player was keen to hear effects in his cans to get the right feel, so I used the Rex as the preamp for his mic whilst using the effects loop to add some spring reverb to the headphone mix. Again, it all worked perfectly and added both quality and convenience to a busy session.

On The Stage My friend Rachel from the band Fuzzy Lights was very happy to try out the Rex. She plays violin in the band, using guitar pedals through a less-than-ideal setup where various pedals are daisy-chained into a guitar amp to get sufficient gain. With its wet/dry blend control, the Roxi would really have been perfect for her live setup, but she was still able to plug her violin’s pickup into the Rex’s XLR input and then have her guitar delay pedal connected to the effects loop, with blending taken care of on the delay itself. She was then able to send the signal on to her on-stage guitar amp using the Amp output, for the distorted amp sound she likes when playing live.

Suitably enthused, Rachel popped into the studio to try out the Roxi, and the sonic improvement compared to her current setup was not subtle; the preamp seemed to bring out much more body from her violin, and the ability to smooth out some harshness with the EQ worked a treat. To quote Rachel herself: “It’s like someone has designed the perfect product for my live setup!”

Final Thoughts For live musicians who want to have more control over their sonic output on

Grace Design Rex & Roxi £609/£921 pros • Great-sounding preamps packaged for stage settings.

• Simple, good-sounding EQ. • Well-thought-out routing options.

• Built like a tank.

• Also great as a flexible studio tool.

cons • Quality comes at a price.

• XLR-only input on the Rex will limit its use for some.

summary Grace Design have taken a bold move by releasing unique pedalboard‑style channel strips that pack in a huge amount of flexibility for live performers, home recordists and small studio owners. These are well worth checking out.

stage, both the Rex and Roxi will have a great deal to offer. It almost feels as if someone in the Grace team who plays live with a mic and pedals has coaxed the rest of the company into designing a product for their live setup! So much of performing live is about reliability and trust in your gear, and these tools from Grace remove the need for ad hoc setups that can be prone to issues with power supplies and impedance matching. In this regard, these are brilliantly designed products, and it’s obvious that a great deal of care has been taken to think of what might be needed in different settings. If this sounds like you and you have the budget you should seriously consider getting one of these options: the Roxi would be great, but if you can only stretch to the Rex, the only thing you should perhaps think about is how you would manage only having an XLR input.

For a studio owner like myself, I was surprised at just how useful these boxes were: using one as an extra little channel strip on one project but then employing the same device to solve an issue on a live tracking session was just great.

One of these would be brilliant if you run a small mobile recording setup or are a self-recording musician who is looking to take a jump in quality. £ Rex £609.19, Roxi £921.97.

Prices include VAT.

Aside from their form factor, what sets these mic preamps apart is the presence of a stompbox-friendly effects loop.

T ASAP Europe +44 (0)208 672 6618 E sales@asapeurope.com W www.asapeurope.com W www.gracedesign.com

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Dirtywave M8 Model:02 Portable Hardware Tracker

This powerhouse portable tracker might be your new best M8.

RORY DOW

P

ortable tracking is a booming, albeit niche, industry. Dirtywave, the Los Angeles-based company behind the small but impressively powerful M8 tracker, have been quietly building a loyal following of M8 fans for about two and a half years.

During that time, the M8 received many software updates, and despite having only one developer, it is currently on operating system version 3.3. And with a new Model:02 revision, it’s better than ever.

Timothy Lamb, the man behind Dirtywave, is a well-known figure in the tracking and chip-tune music scenes.

Before moving into hardware production, he wrote a software tracker called Little Sound DJ, which ran on the Game Boy platform.

It proved highly popular and serves as the basis for the M8 today. So, despite Dirtywave being a new name in town, the M8 software is mature and well-developed.

The new Model:02 version improves on the first edition with a new aluminium enclosure, larger 3.5-inch display (from the original 2.8), improved battery life (up to 12

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hours), a built-in microphone, and USB‑C instead of micro-USB. In all other aspects, the Model:01 and Model:02 are identical.

Yes M8 The M8 is surprisingly sleek and compact, measuring 96 x 133 x 20mm. It fits nicely in the palm and is operated like a smartphone — with two thumbs. The 3.5-inch IPS touch display is small but incredibly sharp. Data input happens through the eight mechanical key switches and, occasionally, the touch screen. I love the mechanical keys. They give the unit a classy feel. Stereo audio inputs and outputs, plus MIDI in and out, are on 3.5mm TRS jacks and sensibly placed at the top of the unit. At a push, you can use the onboard speakers instead of the line output (which doubles as a headphone output), but they are tiny, so keep your expectations low. On the sides are a recessed on/off button and a slot for a microSD card.

The battery provides about 12 hours of use. It charges through a USB-C port, which also functions as a stereo‑in/

out 16-bit/44.1kHz USB audio and MIDI class-compliant interface. Samples and songs are stored on the SDHC microSD card. The M8 can stream samples directly from the SD card, there’s no limit on sample size, and loading and saving songs is almost instantaneous.

In many respects, the M8 is a classic tracker. Eight tracks of sample sequencing are arranged in rows of hexadecimal notes and ‘FX’ (more on these later). But the M8 has many features that take it beyond what you might find in a traditional tracker.

Virtual synthesizers cover a range of synthesis types, including classic console chip emulation, FM, physical modelling and virtual analogue, and there are effects (reverb, delay and chorus), modulation, tables (a sort of pattern within a pattern), stem rendering, groove templates and scale quantising.

You Wot, M8?

Trackers approach sequencing a bit differently from piano‑roll-based DAWs and sequencers. Tracks are organised into rows, and notes flow from top to bottom.

Each of the M8’s eight tracks can play one of 255 Chains. A Chain is a playlist of up to 16 smaller patterns called Phrases. Each Phrase can contain up to 16 steps. So, at the most, you’re working in 16‑bar chunks. Each

step within a Phrase can play a different note on any instrument and apply up to three FX, which are instructions to transform the note somehow.

The eight tracks are monophonic. This standard limitation of trackers was originally there to keep CPU usage in check, and it has its downsides. You can’t sequence chords or polyphonic phrases without using multiple tracks, although the M8 does have some workarounds, which we’ll come to. However, the monophonic tracks have some creative benefits too, making them ideal for sample slicing, bass lines and leads. The ability to address a different sound on each step of a Phrase is also one of those features that, once you click with the tracker workflow, is sorely missed when moving back to more traditional sequencers. These inherent trackerisms helped shape music genres like chip-tune, jungle, and IDM back in the 1990s.

School M8 Using the M8 requires a certain amount of muscle memory. Almost everything is done with the eight mechanical keys on the front panel. There are up/down/left/right keys for navigation, a Shift key, an Option key, an Edit key and a Play button. With just these, one can navigate pages, make selections, edit values, copy, paste, etc. Some time spent learning shortcuts is necessary to become proficient, but once the muscle memory kicks in, it’s a surprisingly fast workflow.

The navigation of the M8’s pages is very logical. When you start a new song or load an existing one, you’ll begin at the Song page. If you press Shift and Up, you’ll move to other Song-related pages. Pressing Shift and Right will drill down into more detail, first editing the currently selected Chain, then with another press, the Phrase chosen within that Chain. A further right move will edit the Instrument assigned to whatever step you had selected in the Phrase.

No matter which page you’re on, if you navigate up, you’ll get to pages relevant to the section you’re in, and if you navigate down, you’ll get to the Mixer and Effects pages. You can always see a small map showing you where you are in the system, although after a few days of regular use, you stop needing it. It’s an intuitive and fast way to move around the current project.

An M8 Song is composed of up to 256 rows. In each row, you can select one of 256 Chains to play for each of the eight tracks. To reiterate, a Chain is a playlist of up to 16 different Phrases, and a Phrase, in turn, is a pattern of up to 16 steps.

Instruments are the core of the M8, and you can have up to 128 Instruments defined per project. If you’ve used a tracker before, you’ll know that samples are the bread and butter of tracker music, so of course, the M8 has a powerful sampler on board. The sampler reads and writes directly to the SD card, so there is no limit on sample length.

Each sampler Instrument can load or record a single sample from either the line inputs, USB inputs, or the built-in microphone.

From there you can transpose, edit the start and end points, loop with different loop types, and assign up to 128 slices.

Some offline edit operations are available, like normalising, cropping, fading in or out, downsampling and auto-slicing.

Every Instrument also has access to a multi-mode resonant filter, sample rate reduction, and a selection of waveshapers like soft-clipping, pre- and post-filter hard-clipping, swap, fold and sine functions. There are certainly plenty of options for pummelling your instruments into submission. Lastly, you can set the send amount to the reverb, delay and chorus effects.

Unlike many trackers, the M8 doesn’t stop at sample playback.

There are four synths to choose from: Wavsynth, Macrosynth, FM Synth and Hypersynth. Wavsynth is an 8-bit wavetable synth with options to multiply, skew, warp and mirror the waveforms.

It’s perfect for chip-tune sounds.

Dirtywave M8 Model:02 £613 pros • Classic tracker mixed with modern features in a handheld package.

• Lightweight yet robust build using quality materials.

• Muscle-memory workflow that is fast and mostly intuitive.

• Surprisingly deep synthesis and sequencing.

• Surprisingly good sound quality.

cons • Tumbleweed...

summary The M8 is an impressive little portable tracker that feels great to use, has a fast workflow, and sounds far better than it has any right to. It’s one of those products built by one person, and his passion and expertise show.

Macrosynth offers 44 different synthesis algorithms based on the Creative Commons‑licensed Braids instruments created by Mutable Instruments. Synthesis methods include classic analogue, ring modulation, unison, hard-sync, vowel synthesis, FM, physical modelling, drum synthesis and more. Each model only offers a couple of editable parameters, but they cover a lot of sonic territory. Plus, you still get the filter, waveshaping, and modulation options common to every instrument.

FM Synth is a classic 4-operator FM synth with 12 algorithms, feedback per operator and different waveforms for each operator.

It nails those early Yamaha FM-type sounds and is probably my favourite M8 synth.

Hypersynth aims to help with one of M8’s limitations: the lack of polyphony I mentioned before. Hypersynth has six sawtooth oscillators, which can be arranged as 16 user-definable chords. When sequencing Hypersynth, you can select different chord numbers and each chord is transposed by the note number.

Back, or rather top, panel connections include 3.5mm sockets for MIDI and audio I/O, and a USB‑C port.

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So Hypersynth allows chord progression using just a single M8 track. There isn’t much synthesis possible here, although you can detune the oscillators, spread them over the stereo field, and tilt the volume balance of the oscillators. Still, it’s a neat solution to the problem of polyphony.

Regardless of which Instrument you have loaded, you have four modulation slots into which you can load different assignable modulation sources. Choose from four different envelope types, including one that can be triggered by another track, allowing for some nice side-chaining effects.

There’s also an LFO and a Tracking option for pitch-tracking modulation. Neighbouring modulation slots can even modulate each other. Four modulation slots might not seem like a lot, but it’s enough to get a sound moving and sounding organic, and I like that they are the same no matter which Instrument you have loaded.

There are a couple of final MIDI instrument options: a simple ‘MIDI Out’, which allows you to sequence from either the USB MIDI or 3.5mm MIDI port. You can assign channel, bank and Program Changes and up to 10 MIDI CC numbers.

The second MIDI option is ‘External Instrument’, where the instrument’s audio output is connected to the M8’s audio input. This allows the M8 to filter, waveshape, and apply effects to the external instrument’s audio. Sadly, external instruments cannot be rendered using the M8’s render functions, but you can record the audio using the sampler.

In all other aspects, rendering in the M8 is impressive. You can choose which tracks to include or exclude in the render and turn the effects on or off. You can choose to render stems or a stereo mix.

Stem rendering is perfect for moving your song to a DAW for its final mix.

AutoM8 The magic of tracking is in its sequencing power, and the M8 doesn’t disappoint. Every step in a Phrase can trigger a note from any of the 128 Instruments and apply up to three FX. FX are an instruction to automate The microSD card slot is found on the side of the unit.

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ALTERNATIVES Portable tracking is still a niche arena, but there are options. Firstly, the M8 is available as a ‘headless precompiled firmware’ that will run on a Teensy 4.1 development board. So, those with DIY skills can build their own M8 for a modest investment.

The other option is a Polyend Tracker Mini, which I reviewed in October 2023.

Although quite similar to the M8, it feels quite different in use. It has a much bigger 5‑inch screen, a better USB audio interface (more channels), and the recent firmware update has narrowed the gap in terms of synthesis, but the M8 is arguably more portable due to its smaller size.

something in the sequencing or synthesis engine. The M8 has over 90 different FX codes that can be applied, and they do a wide variety of things, from automating instrument parameters to sophisticated sequencing tricks.

Let me give you an example. You might use a Phrase to trigger slices in a drum loop. The three FX can be used to transform that slice in real time. You might use one to reverse the playback of some slices. You might use the second to place a conditional trigger so that the slice is only triggered every third loop of the pattern, and on the final column, you might increase the delay send for that step.

FX can be more complex than simple per-step automation. For example, the REP (Repeat) command will repeat the last command in the column for every subsequent step without copying and pasting the instruction on every step. The hexadecimal number applies an offset for each subsequent repeat, allowing you to automate filter sweeps, volume fades and pitch rolls with just a single FX command.

Randomisation is another area where FX shine.

You can randomise pitch, Instrument or FX for any given step.

Combining this with the SED (Random Seed) command can create predictable randomness, which

is a great way to come up with alternative patterns, melodies and automation that you might never have programmed yourself.

Other FX allow you to jump around a Phrase, automate scale quantising, change mixer settings and synthesis parameters, program pitch‑bends or portamento, apply vibrato, and much more.

There are so many creative FX commands you’re unlikely to run out of inspiration. But the M8 has yet more tricks up its sleeve.

Tables are another form of instrument-based automation. A Table is another 16-step page similar to a Phrase except that it is instrument-specific and runs every time a note from that instrument is played. Think of it as a pattern within a note.

It has a pitch offset, velocity, and three FX columns for each step. The Table runs faster than the main pattern (at a division of your choosing), allowing you to do ‘inter-step’ automation. A simple use of Tables might be to create arpeggio-style note trills or have some vibrato applied after a specific time.

You might also use them to automate synth or mixer parameters like effects send or panning. The critical distinction is that these automations are within every note. And if one Table isn’t enough, you can even run a second one simultaneously.

Overall, I was mightily impressed with the synthesis and automation possibilities.

There is a synergistic feeling throughout the M8 that all decisions about what features to include or exclude are well thought-out and, most importantly, true to tracking as an art form.

BandM8 The M8 isn’t just for linear sequencing.

There are some handy features for those who like to play live and improvise. In Song mode, you can quickly jump to different Song sections by cueing up any row to play next when the current row has finished. You can also create song sections by leaving blank rows. Whenever the sequencer reaches an empty row, it will loop back to the beginning of that section. This way, you can create ‘islands’ of sequences that loop until you hop to a new island. If you need even more freedom, you can engage ‘Live mode’ to trigger individual Phrases on different song rows, much like Ableton Live’s Session view.

Anyone playing live sets will also appreciate the ability to load a new Song and cue it without any stop in audio. This is possible because samples do not have to be loaded into RAM, so Song loading is effectively instantaneous. Also, if you’re

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DIRTY WAV E M 8 MODE L : 02

low on inputs and mixing options in your live setup, the M8’s inputs (USB or line) can be mixed with the internal mixer. There’s even a dual mono mode where you can use the left and right signals as two mono sources.

These inputs have their own effect sends and volume controls and can be automated using FX commands.

What about integrating the M8 into a more extensive setup? The M8 does an excellent job of MIDI integration. As well as sending and receiving MIDI Clock sync and transport information, you can map up to 128 incoming MIDI control changes to just about any destination you can think of.

This will please anyone playing live with the M8 as you can take a MIDI controller with you and have your controller pre-mapped, song-by-song, to any parameters you wish to automate.

Another potential input for automation is the touchscreen, which can be assigned to multiple destinations and used as an X/Y pad. These mappings

OS 4.0 Such is the fast‑paced development of the M8’s operating system that, just as we were preparing this issue for print, Dirtywave announced a major update. It’s currently in public beta, available from Dirtywave’s official Discord channel (discord.gg/WEavjFNYHh). It might even be fully released by the time you read this.

The 4.0 update will include support for controllers (just the Novation Launchpad Pro MkIII for now, more in the future); EQ for instruments, global effects, and master; a new instrument pool screen for easier instrument management; two new distortion types; 60 new wavetables added to the Wavsynth Instrument; two new zero delay feedback filter types; global tuning to diverge from A=440 standard pitch; and a generous number of new FX commands.

I only had the briefest chance to play with the new features but I found the EQ extremely useful, and the new wavetables turn Wavsynth from a basic chip-tune synthesizer into a full‑blown wavetable synth. The new FX commands unlock some powerful sound‑design options too. A great update, free to all M8 owners, and compatible with both the original M8 and Model:02.

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Spot the difference...

The Model:02 (left) and the original M8.

are all listed on the MIDI Mappings page, where you can edit the mapping’s range, source and destination.

The M8 can even behave like a sound module, with each track having a separate MIDI channel. Suppose you assign multiple tracks to the same MIDI channel and activate Poly mode. In that case, the tracks will respond as a single polyphonic instrument, with as many voices as tracks assigned to that channel. So you can use the M8 as an 8-voice synthesizer, with the added coolness of each voice having a different sound.

Don’t UnderestiM8 Looking at the M8, you could easily dismiss it as a handheld toy. How much music can you write with eight buttons and a teeny screen? Well, if you go to Dirtywave’s official Discord server, you’ll find an active community growing daily, and the amount of music posted here is inspiring. What you’ll hear is not just Game Boy-inspired chip-tune (although there is plenty of that) but well-polished pop, electro, techno, IDM, ambient, jungle, drum & bass, synthwave, glitch, house, hardstyle, and modular-encompassing jams (did I mention the M8 can be mounted in a Eurorack case too?).

My own experience during the review period backs this up. I cannot think of another device I own or have reviewed in the last few years that I’ve been more productive with. The M8, once you click with its workflow, is incredibly fast to work with. I’ve finished around six songs in the two months I’ve used it and have started about a dozen more. When I say ‘finished’, I mean they’re full songs with different sections, fills, complex automation,

well mixed, and not just 16-bar loop ideas. I will export these as stems and load them into the DAW for mixing, but consider them finished in all other aspects. The M8 cannot compete with a DAW for the final mix, but it’s an absolute blast for songwriting.

In the 1990s, trackers were popular on the Commodore Amiga home computer. It was a fun way to write music. Many people progressed to the Atari ST, maybe with the addition of a sampler, a couple of synths, a mixer and some rackmount effects. The M8 reminds me of these times. It’s like having the Atari ST, an Akai S950 sampler (for sampling), a Yamaha DX100 (for FM sounds), a Moog prodigy (for analogue monosynth sounds), a Korg Z1 (for crazy physical modelling), a mixer, and some respectable effects all in a palm-sized unit.

And that makes me happy.

If it’s not already obvious, I have fallen for the M8. It is clearly a passion project for Timothy Lamb, and the amount of thought he has put into its physical design, software and workflow is evident. The sequencer might be based on 30-year-old designs, but there are plenty of innovative new features you won’t find in another tracker.

And how does it sound? So often, with cheaper music devices, the general sound character can be disappointing, and no amount of clever features can compensate for that. I’m happy to say that is not true with the M8. It has depth and space, and I am constantly surprised by how well-produced M8 songs can sound. The synthesizers sound genuinely good, and so do the onboard effects.

If you’re looking for a hardware tracking device, you must check out the Dirtywave M8. In most senses, it’s a classic tracker.

The hexadecimal spreadsheet sequencing will be instantly familiar, but the M8 adds so much: synthesis, performance options, FX, Tables, modulation, stem rendering, disk streaming and more. It’s fast to use, and it sounds great. In short, brilliant. £ £613 including VAT. W www.dirtywave.com

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Orchestral Tools Benjamin Wallfisch Strings

Sample Libraries

The Berlin samplemeisters link up with a Hollywood film composer to create two major string collections.

DAV E ST E WA RT

S

peaking in a video in November 2023, film composer Benjamin Wallfisch cheerfully observed: “I spend way too much of my time sitting here playing string orchestra samples. [Join the club, Ben.] We’re very lucky to have amazing libraries now of gorgeously recorded string orchestras from around the world, but I’ve always been slightly underwhelmed at the final result. Of all the sections of the orchestra, the strings have the biggest gap between what we’re able to do using samples and the finished final performance if we’re lucky enough to record with a live orchestra. And for years I’ve been thinking, why is that?”

Why indeed? Within weeks the question was addressed when Orchestral Tools announced the release of two standalone string collections, Benjamin Wallfisch

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Strings (cinematic ensembles and divisi) and Stringscapes (orchestral string effects and textures). Created in close collaboration with the Hollywood composer and billed as OT’s largest sampling project to date, the project aims to give users (in Wallfisch’s words), “something that gets as close as possible to having a scoring session on your keyboard”.

The libraries feature identical line-ups of 14 first violins, 12 second violins (different players), 10 violas, 10 cellos and six double basses recorded on the main scoring stage at Berlin’s Teldex Studios. They are not small: Benjamin Wallfisch Strings (BWS for short) weighs in at a hard-drive challenging 501GB and Stringscapes eats up 129GB. Happily, you don’t have to download all of their numerous mic positions in one go; you can also hedge your bets by buying individual BWS and Stringscapes instruments for as little as €65 and €42 respectively.

Benjamin Wallfisch Strings and Stringscapes run exclusively on the latest version of Orchestral Tools’ Sine player, which works standalone and as a VST, VST3, AU and AAX plug-in. You’ll need least 8GB of RAM, and as with all large orchestral libraries, the makers strongly recommend SSD storage for best performance results.

Breaking Convention It transpires that one of Mr Wallfisch’s complaints about today’s string libraries is the routine use of sample loops. While looped long notes can be sustained indefinitely, the composer feels that interrupting the players’ expressive performance mid-flow with a relatively short looped section has a de-humanising effect.

Hence, BWS and Stringscapes are among the very few string libraries to offer loop-free sustains; instead, the players hold their long notes for 23-25 seconds, which should be enough for most musical situations.

BWS’s other deviations from the norm include true divisi sections, separate patches for upbows and downbows and five dynamic layers for all straight bowed artics and legatos. Another distinctive feature is a huge Atmos-ready microphone setup comprising Decca Tree L‑R, Decca Tree centre, close, wide, A‑B (an alternative wide position), outrigger, surround, immersive front and immersive mid mic positions. Each of these has a raw and processed version, the latter featuring Ben Wallfisch’s dedicated EQ and processing effects. Designed to give you “the Hollywood sound right out of the box”, these positions include an LFE mix for the basses.

Obviously having 18 versions of every sample hugely inflates the size of these libraries, but you can easily get by with just two or three positions: a mix of the default Decca Tree and close mics sound great, and if you want more room ambience you can add the distant outrigger or surround position. Personally I liked the subtle brightening effect of the processed versions, so they would be my starting point with these strings. If your project involves a large number of mic positions, you can use the Sine player’s mic merge feature to bounce them down into a single stereo mix.

BWS Overview Orchestral Tools’ long history of recording string orchestras at Teldex Studios has reached a zenith in BFS, one of the most sonically impressive string orchestras I’ve heard in recent years. The cello long notes are a thing of beauty: their hushed, breathy quiet deliveries, rich and vibrant mp range and majestic loud performances show off the library’s dynamics to great effect, and their generous C1-G5 span means you can use them as a sketching tool. Having used the cello sustains to create a chordal passage, I doubled the chords’ root notes with basses in the lower octave and was knocked out by their tremendous low‑end presence — instant orchestral grandeur and gravitas.

Another standout feature is the beautiful transparency of the high strings’ upper register. It’s unusual to find violins sampled beyond a top note of C7 (three octaves above Middle C) — these players go up to E7, maintaining pristine tone and perfect tuning all the way. Consequently, chords and melody lines sound fabulous across the violins’ entire register, with no weak spots.

BWS lacks the vibrato control of other OT string libraries: rather than providing an on-screen slider to switch between vibrato

The Sine player’s Benjamin Wallfisch Strings GUI, showing three of the available mic positions.

styles, it follows the example of Pēteris Vasks Strings by linking vibrato intensity to dynamics. The quieter layers use a light, restrained vibrato which becomes more pronounced in the mp dynamic and reaches full force in the loud deliveries, enabling you to program dramatic volume swells accompanied by emotive vibrato surges. This effect works particularly well with the violas, who step up with some impassioned loud sustains.

So, do upbow/downbow variants and unlooped sustains make a difference? I’d say definitely yes. To my ears, keyswitching between the two bowing styles at strategic moments sounds more natural and realistic than sticking with one or the other, and I felt that the somewhat gentler upbows are a better starting point for quiet romantic music. The loop-free samples are a revelation: each note in a chord develops naturally at its own pace without a loop butting in, resulting in a notably more organic sustained timbre.

BWS Tutti & Divisi Each of BWS’s five string sections was recorded with its full complement of players (referred to as the ‘tutti’ sections), then twice more with each section split into two half-size groups called Divisi A and Divisi B. Rather than dividing ‘at the desk’ in traditional style, the violins and cellos were split into front and back groups while the violas and basses divided left and right — this maintained a distinct physical separation between the divisi groups (good for panoramic mixes) while

enabling players to stay attuned to their desk partner’s performances (good for musical cohesion and intonation).

The advantages of divisi writing are well documented: you can boost orchestral realism by distributing the notes of a chord across separate divisi sections (thus adding voices without increasing the virtual player count), split cello parts

Orchestral Tools Benjamin Wallfisch Strings €958 pros • Both libraries sound fantastic and benefit from five dynamic layers, long unlooped sustains and admirable performance precision.

• BWS features half-size divisi sections and separate upbows and downbows.

• Soundscapes’ generous menu of played string effects and textures contains some startlingly original material.

cons • BWS is expensive.

summary This collaboration with an award-winning UK composer represents the acme of Orchestral Tools’ string sampling achievements to date. Performed by 52 players and recorded via an Atmos mic setup, the project comprises two collections: Benjamin Wallfisch Strings is a classic strings multisample library suitable for traditional, contemporary film and pop work, while Soundscapes’ imaginative effects and textures span an emotional range from beauty to horror. Taken together, the two are a knockout combination.

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The Stringscapes GUI.

great on rapid-fire lines and arpeggios.

Surprisingly, these artics are only played by the tutti ensembles — the divisi sections perform only sustains and legatos.

Stringscapes Tutti FX

between melodies and bass lines, and achieve a more intimate chamber sound simply by using the excellent half-size sections. While this is not the first library to offer real divisi, it’s good to see it cropping up in a contemporary release.

BWS Articulations Legato patches are supplied for all the tutti and divisi sections’ downbow and upbow sustains. Performed at five dynamics, the legato transitions maintain Orchestral Tools’ high musical standards and sound superbly expressive when you introduce dynamic movement with the mod wheel.

If you prefer, you can assign dynamics to key velocity — either way, it’s essential to overlap notes slightly to trigger the smooth inter-note transitions.

The remaining BWS articulations feature the standard trilogy of marcato, staccato and spiccato short notes along with pizzicato, tremolo (four dynamics each) and three-dynamic tone and semitone trills. The marcatos last just over a second and boast a scintillating loud layer; the staccatos are sufficiently short and tight to use for rhythmic ostinatos, while the ultra-short, briskly brushed spiccato bowing sounds

BWS’s companion library Stringscapes is an expansive set of string effects and textures inspired by the composer’s scores. The collection divides into live played effects written and orchestrated

by Wallfisch and his team, and cinematic hybrid sounds created by Orchestral Tools’ sound designers using the raw material from these sessions.

The Tutti FX are musical events performed by the full string orchestra.

Prosaic patch names belie their complex nature: the subtly menacing ‘Atmosphere’ combines gently accelerating minor third trills with a mournful lower semitone motif, while the unison of ‘Spreads’ disintegrates as low and high strings gracefully slide off in opposite directions in a series of tritone steps, only to see them slowly return to unison after hitting their outer pitch limits. In the same vein, major and minor chord glissandi slither up three octaves to a high triad before descending to their starting position. Another highlight here is ‘Trills’, a mini-composition which gradually builds trills of various pitches into a shimmering chordal latticework.

For pads, I can recommend the majestic ‘Chords’, and ‘Longs’, which weaves multi-octave unisons into lovely shifting soundscapes. I was also impressed

Music In The Blood Born in London in 1979 to violinist mother Elizabeth and cellist Raphael Wallfisch, Benjamin Wallfisch’s paternal grandparents are concert pianist Peter Wallfisch and cellist Anita Lasker-Wallfisch, a remarkable woman who co-founded the English Chamber Orchestra after surviving the horrors of the Auschwitz concentration camp. Benjamin’s great-grandfather was Albert Coates, composer and conductor of the LSO from 1919 to 1922.

An infant prodigy who could play piano and compose by the time he was three, Wallfisch junior grew up in a house full of music. He recalls: “I think my first memories with music were hearing Bach played every morning by my dad, my mum playing the violin and my grandpa playing the piano... music was always synonymous with family, and I guess to this day it just kind of feels like home.”

Film music also made an impression: “Growing up in London in the ’80s and ’90s I was always very interested in movies... those incredible scores of John Williams, Jerry Goldsmith and all the rest of that extraordinary period in film music. When I saw ET I was immediately taken by something I didn’t know how to describe, but it just felt so powerful. As a kid I was also fascinated with the process of film-making, and with my passion for music it became a case of ‘I’m gonna find a way to bring those two things together.’”

The plan worked out. By 2005 Wallfisch had scored his first feature film, the crime drama Dear Wendy. The next 10 years saw a steady climb up the Hollywood ladder with scores for Hidden Figures (in collaboration

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with Pharrell Williams and Hans Zimmer), It, Shazam!, Hellboy, The Invisible Man, Mortal Kombat, The Flash and Blade Runner 2049 (another Zimmer collaboration). To date he has over 75 film credits, has received Golden Globe, BAFTA, Grammy (twice) and Emmy nominations and was nominated for World Soundtrack Awards’ Film Composer of the Year in 2019 and 2020.

Following in his 19th Century ancestor’s footsteps, Wallfisch also continued a parallel career as orchestral conductor and composer, performing live in over 100 concerts worldwide. He has received over 50 concert, ballet and theatre music commissions and collaborated with luminaries such as Adele, Herbie Hancock, Lang Lang, Pharrell and Yuja Wang. Hard working, you might say.

If this sounds like the CV of some snobby classical composer, think again. Refreshingly unpretentious, our man also took a strong interest in the electronic music and rave scene of 1990s London, waxes lyrical about Vangelis’ Yamaha CS-80 sounds in the 1982 Blade Runner score and is well versed in analogue and soft synths, samples and sound design — all of which makes him an ideal collaborator for Orchestral Tools, and an inspirational role model for aspiring media composers.

Now based in LA, Ben Wallfisch works in his own composition facility The Scoring Lab, a state‑of‑the‑art Atmos mixing studio and writing room. Though his great-grandfather might have found the technology bewildering, I’m sure Albert Coates would be proud of his descendant’s musical achievements.

The Sine player’s Mixer view. You can save screen space by using the Mic Remote button to collapse the individual mic position channels into a single master channel.

by the Tutti Horror FX’s nerve-jarring atonal material, which comprises some terrific risers, suspense textures and the alarming ‘Stabs’, a sforzando accent followed by a hurtling glissando reminiscent of falling down a lift shaft.

Stringscapes High & Low Strings Effects The spirit of Bernard Hermann’s iconic Psycho score lives on in Stringscapes’ high and low string effects. The high violins’ creepy ‘Aleatorica’ evokes the tormented workings of a fractured psyche, with the basses’ dark, grumbling version warning of something nasty lurking in the cellar. ‘Chatter Pads’ runs the gamut from high-pitched teeth-on-edge screechiness to psychotic low strings improvs, while ‘Spreads’ re-emerges in useful high and low formats. All great stuff, but in my book the standout here is the versatile ‘Glissandi FX’ category, which features some stunning atonal material.

Hermann fans will appreciate the high strings’ slashing staccato accents, which the low strings bring up to date with a tremendous array of visceral atonal stabs. These effects are augmented by straight col legno and Bartok pizzicato performances, which really should have been in the main library — the latter artic is played with mind-boggling force, and will easily hold its own in a rock mix.

User tip: all the above mentioned effects have a ‘Rev’ reverse playback option which is well worth exploring. Many of them start very quietly, fade in slowly and are very long, so keep holding down the key!

‘The Transformation’ is a remarkably unpleasant category-defying racket spanning horror and comedy. All in all, a healthy collection of noises to disturb your listeners’ equilibrium.

Stringscapes Sound Design

Conclusion

Though less startlingly original than the live effects, Orchestral Tools’ hybrid processed sounds contain some bangers. I’ll pick out some favourites... I liked the Hellish descending cacophony of ‘Aurora’ and the sinister supernatural ring of ‘Bloody Spring’ (which sounds like ring-modulated alien church bells). ‘Distorted Strings’ and ‘Space Port Atmos’ are similarly unsettling, respectively evoking the buzzing of a giant wasp and the fearsome hubbub of a crowded deep future Tokyo Metro station. To soothe the troubled soul, there’s ‘Infinity Strings’ (serenely symphonic crescendo-diminuendo major and minor chords) and ‘Ascending’, a blissed-out ultra-slow build to a high-pitched timbre suggestive of distant choral voices.

Sci-fi and horror are covered by the ominous ‘Lift to Hell’ spaceship lift-off noise and ‘Manic Riser’, which one can imagine to be the cries of a swarm of malevolent flying creatures. I also enjoyed the unhinged electronic pitched pulse of ‘Needles’ and chuckled at ‘Infrared Resonances’, which sounds like a clumsy replica of a Caribbean steel drum built by a race of dim-witted giants. Meanwhile,

This project arose from a mutual admiration society: Orchestral Tools CEO Hendrik Schwarzer is a fan of Wallfisch’s soundtrack work (in particular his powerful score for the 2017 film of Stephen King’s It), and the compliment was returned when the composer singled out Orchestral Tools as a potential collaborator.

Both say they’re delighted with the end results, and a spokesman confirms that the composer is now using the strings to write with — no doubt the results will soon be booming from an immersive cinema sound system near you.

Since Schwarzer and co already have several strings collections in their locker, the question “What makes this one special?” arises. Hopefully, my reviews of OT’s most directly comparable products Berlin Strings and Berlin Symphonic Strings (SOS August 2014 and April 2021 respectively) may help you decide. My personal view is that its combination of excellent performances, brilliant tone quality, loop-free long notes, switchable upbows and downbows, divisi sections and imaginative musical effects mark out this collaboration as a star attraction in Orchestral Tools’ sample catalogue. £ Benjamin Wallfisch Strings: Full library

The libraries’ nine mic positions are presented in raw and processed versions, the latter featuring Ben Wallfisch’s dedicated EQ and processing effects.

€958.80, single instruments from €78.

Stringscapes by Benjamin Wallfisch: Full library €298.80, single instruments from €50.40. Prices include VAT. W www.orchestraltools.com

www.soundonsound.com / September 2024

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SPOTLIGHT

Kick Drum Microphones

We’re spoilt for choice when it comes to dedicated kick drum mics.

We check out some of the best.

LUKE WOOD

W

ith a decent selection of general-purpose mics to hand you’ll be well equipped for just about any recording task, but there are some situations where a dedicated model can be a safer bet. For example, you’ll often see all sorts of mics used outside of a kick drum, but when it comes to placing something inside the shell, extreme SPLs and strong blasts of air mean that a little more care and attention is required. So, a dedicated kick-drum mic can be a worthwhile addition to your collection, and thankfully, there are plenty around! An added benefit of such designs is that their low-end response and overall voicing often means that they can deliver great results on sources such as bass amps and cabinets or large brass instruments.

AKG D112 MkII

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AKG D112 MkII

The latest version of AKG’s D112 retains all of the features that made the original so popular, but is far easier to position thanks to a new integrated flexible mount.

A large diaphragm delivers a powerful low-end response, and a 4kHz presence boost helps to provide a punchy sound that’s capable of cutting through a busy mix with minimal EQ. An integrated hum-compensation coil keeps noise to a minimum, and it’s capable of withstanding SPLs over 160dB without distorting. £ £138 including VAT. W uk.akg.com/outlet/D112MkII.html

Audix D6 & F6

Housed in a one-piece machined aluminium body, the Audix D6 boasts a solid and compact design that’s both hard-wearing and manoeuvrable.

It’s equipped with the company’s VLM diaphragm, which promises to deliver a natural, accurate sound and features a cardioid polar pattern Audix D6 to maximise isolation and avoid feedback. The Audix range also includes the F6, a lower-cost alternative that employs a hypercardioid capsule and has a slightly lower maximum

SPL figure of 140dB (compared to 144dB for the D6). £ D6 £199, F6 £119.

Prices include VAT. W www. soundonsound. com/reviews/ audix-dp7 W audixusa.com/ products/d6/ W audixusa.com/ products/f6/

Audix F6

Avantone Mondo

Avantone say that the latest version of their Mondo kick-drum mic boasts a focused voicing and fast response that will punch through even the busiest mix with little to no EQ, and it’ll stand up to high volumes thanks to an SPL handling of 145dB. An all-metal construction helps it to stand up to life on the road as well as in the studio, and a pair of machined inserts on the body help to keep it held firmly in place using the included shockmount. £ £125 including VAT.

W avantonepro.

com/en/ products/mondo

Avantone Mondo

Earthworks DM6

beyerdynamic TG D70

beyerdynamic TG D70

Built around a refined version of their renowned M88 capsule, beyerdynamic say that the TG D70 is a must-have for every drummer. A hypercardioid polar pattern provides excellent isolation and feedback resistance, while an elastic suspension system helps to prevent mechanical vibrations from reaching the capsule. Along with its clear, natural sound, the mic boasts a compact and hard-wearing design that makes it easy to position and ensures it’ll cope with serious gigging and session work. £ £198 including VAT. W beyerdynamic.de/tg-d70.html

DPA 4055

DPA mics are renowned for their clear and natural sound, and the 4055 is no exception. It offers a full-range frequency response with a 6dB boost at 10kHz, allowing users to craft the exact sound they want through a combination of positioning and EQ, and maintains a <1% THD figure at an SPL of 156dB. Its internal shockmount design helps to prevent vibrations reaching the capsule, and an asymmetrical body design makes it easy to position either in front of or inside a kick drum. £ £672 including VAT. W www.soundonsound.com/reviews/ dpa-4055

The latest addition to Earthworks’ drum range, the DM6, is a capacitor mic designed specifically for kick drum duties. It boasts a supercardioid polar pattern that remains consistent across the mic’s frequency range, and promises to deliver a great sound straight out of the box while still responding well to EQ where necessary. It comes fitted with a Triad‑Orbit M2‑R Ball Mount that makes it easy to adjust and fix its position, and is capable of standing up to SPLs of 150dB. £ £399 including VAT. W www.soundonsound. com/reviews/ earthworks-dm6 W earthworksaudio.com/ drum-microphones/ dm6-best-kick-drumbass-amp-microphone

Earthworks DM6

E‑V ND68

The E‑V ND68 is a supercardioid dynamic mic that promises excellent isolation from nearby instruments, with a voicing that aims to deliver a deep low end while extending high enough to capture a kick drum’s attack. Sturdy housing and a tight-mesh grille help it to stand up to hard use, and it’ll be more than comfortable being put to work up close thanks to an SPL handling of 144dB. £ £238.68 including VAT.

W products.

electrovoice. com/na/en/ nd68

W www.dpamicrophones.com/

instrument/4055-kick-drum-microphone

Heil Sound PR 48

Designed specifically for kick drums and low-frequency instruments, the Heil PR 48’s frequency response is said to enhance a drum’s low end while Heil Sound PR 48 preserving the midrange and high-frequency content that can be lost with other mics. An integrated adjustable mount means it can be fitted directly onto a stand, while its compact design makes it easy to position. £ £314.99 including VAT. W heilsound.com/product/pr48

Lewitt Audio DTP640 REX

Lewitt’s DTP640 REX is essentially two microphones in one, combining a dynamic capsule that focuses on the drum’s attack and a capacitor element that takes care of the low end.

Three selectable voicing options provide some onboard tonal shaping, and independent outputs for each capsule allow users to mix their Lewitt Audio respective levels to DTP640 REX taste. Everything is housed in a solid die-cast body with a steel grille, and the mic will withstand SPLs of 150dB. £ £309 including VAT. W www.soundonsound.com/reviews/ lewitt-audio-dtp640-rex

DPA 4055 E‑V ND68

W www.lewitt-audio.com/microphones/ dtp-percussion/dtp-640-rex

www.soundonsound.com / September 2024

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SPOTLIGHT

KICK DRUM MICROPHONES

Milab BDM-01

As you’d expect from Milab, the BDM-01 boasts a no‑compromise design, and is built around their own hand‑crafted 21mm capacitor capsule. Its frequency response has been shaped to deliver a perfect kick-drum sound that combines low-end power with a detailed attack and high‑frequency reproduction. They’ve also optimised the dynamic range, sensitivity and internal Milab BDM-01 pop filter, which help the mic to handle SPLs up to 155dB. £ £898.80 including VAT. W www.soundonsound.com/reviews/ milab-bdm01

W www.milabmic.com/bdm-01

sE Electronics V Kick

sE’s V Kick is based on the same capsule technology used in their popular V7 and V7 X, and boasts a compact, solid housing that features a patented capsule suspension method. Four tonal options are on offer courtesy of a pair of Classic and Modern switches — the first alters the mic’s overall voicing and the second adjusts its high-frequency response — allowing it to cover everything from well-rounded sounds to more aggressive modern rock and metal tones. £ £177 including VAT. W seelectronics.com/products/v-kick

sE Electronics V Kick

Sennheiser E602-II

Developed alongside top producers, engineers and drummers, the Sontronics DM-1B is described as a must-have tool for anyone looking to capture a perfect drum sound on stage or in the studio.

It’s an end-fire, large-diaphragm capacitor design that promises to excel at capturing the attack, resonance and energy of a kick drum, and with its -15dB pad engaged it’s capable of handling levels of Sontronics DM-1B up to 155dB. £ £369 including VAT. W www.soundonsound.com/reviews/

Shure Beta 52A

Telefunken M82

products/microphones/e-602-ii

Shure Beta 52A

September 2024 / www.soundonsound.com

sontronics-dm1b-dm1s-dm1t

W sontronics.com/products/

dm-1b-condenser-microphone

Telefunken M82

With a large-diaphragm dynamic capsule, Telefunken’s M82 promises to deliver impressive low-frequency definition, but with a response that extends high enough to maintain clarity at the high end. A pair of EQ switches provide a midrange cut and high-frequency boost that allow users to adjust the voicing to suit their source, and a maximum SPL of 146dB means that it can be used up close with even the most enthusiastic drummers.

Shure’s offering comes in the form of the Beta 52A, a supercardioid dynamic with a frequency response that has been tailored specifically for kick drums and bass instruments. A pneumatic shockmount system minimises the transmission of vibration and mechanical noise, while a hard-wearing body design and steel-mesh grille help to provide the long-term reliability Shure are renowned for.

£ £379 including VAT. W www.soundonsound.com/reviews/

£ £205 including VAT. W www.shure.com/en-GB/products/

W www.telefunken-elektroakustik.com/

microphones/beta_52a

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Sontronics DM-1B

With a solid and lightweight design, the Sennheiser E602 is equally at home on stage or in the studio.

Although it’s a dynamic design, the mic uses a lightweight voice coil that helps to deliver a fast transient response, while generous low-frequency extension helps to capture the all-important low-end weight of a kick drum.

Sennheiser E602-II It also offers good isolation from nearby instruments thanks to a cardioid pattern that remains consistent throughout its frequency range. £ £154.80 including VAT. W www.sennheiser.com/en-gb/catalog/

telefunken-m82

product/m82-kick

PUSH IT. V7 Push To Talk

ON TE ST

Origin Effects Cali76 FET & Cali76 Bass Compressor Pedals Origin Effects’ original Cali76 compressor, launched in 2013, offered something significantly different from the compressor pedals guitar players had previously been accustomed to, such as the highly coloured-sounding and squashy Ross, DynaComp and Orange Squeezer. There’s nothing inherently wrong with ‘coloured and squashy’ in a guitar pedal, of course, unless you happen to want something cleaner-sounding and less obvious in its dynamic effects. The all-analogue Cali76 offered ‘studio quality’ compression with a full control set in a guitar pedal for the first time. Although ‘modelled on’ a 1960s UREI 1176 FET-based, rackmount compressor, it could actually run on a 9V battery, even if an 18V mains adaptor was preferred, so obviously it didn’t contain the same circuitry as a real 1176. It undoubtedly did, however, offer much of the same exceptionally musical character and quality of sound that we associate with premium studio compressors, including a very low noise floor and wide bandwidth.

It had gain-reduction metering via an LED strip, with customisation options for adding a parallel dry path and/or a transformer-coupled Class-A, discrete output amplifier.

Popular and very successful as the original, large-format Cali76 was, it was relatively expensive and demanded a lot of pedalboard space. That led Origin to develop some more compact and affordable versions, the Cali76 Compact and Cali76 Compact Deluxe. In the

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former, you had just a High/Low switch for the ratio, and a single control used to set both attack and release times (fast attack with slow release or vice versa), whilst the Compact Deluxe kept separate Ratio, Attack and Release knobs as well as offering a Dry level mix for parallel compression. Gain reduction could still be monitored to a limited extent via a colour change in the on/off LED. Now we have two more Cali76 versions that are even more compact than the previous Compacts. Designated Cali76 FET Compressor and Cali76 Bass Compressor, these use the new smaller housing employed by Origin’s effects pedal line (64 x 124 x 58mm WDH), but there’s

“The new pedals have a ton of headroom, permitting operation at line level... you’d have no problem using either of these as a vocal recording compressor on a mixer insert point.” more to them than the smaller box. The least visible but perhaps most significant change is internal operation at 24V, whilst still using a standard 9V DC input (200mA).

This means the new pedals have a ton of headroom, permitting operation at line level, and given how clean and quiet it is, you’d have no problem using either of these as a vocal recording compressor on a mixer insert point. They will also be better able to handle the extra hot output of active pickups or from another pedal ahead of it in the signal chain, a high-level effects loop, or indeed the direct output from any other line-level source.

The input gain has plenty of range for working with a direct connection from a passive instrument, as well, but I’ve always preferred a ‘character’ compressor like a DynaComp or Squeezer in front of an amp or modeller, with something smooth like an 1176 placed

at the end of the signal chain, where it can offer a degree of levelling without significantly changing the tonality.

Both the new pedals have much of the same line‑up of controls: Input, Output, Dry (adds unprocessed signal to the output) and Ratio. But whilst the FET model has separate Attack and Release controls, the Bass model has the combined Attack/ Release combo of the earlier Compact model, plus a high-pass filter. Turning up the latter to remove some of the bass from the side-chain (ie. control signal) prevents the higher energy inherent in the low bass notes from driving the whole signal into gain reduction. Although there are some slightly tweaked levels and time constants in the Bass model to help minimise the distortion, the two pedals’ circuitry is substantially the same, differing only in their control access, so you could use the Bass model in all the applications where you might have used the standard FET model. The only ‘compromise’ is not having separate Attack and Release controls.

The ‘combined’ ATT/REL control on the Bass model makes the assumption that if you’re using a fast attack setting, you’ll want a slower release to avoid obvious side-effects, and if you’ve used a slow attack to allow the leading edge of transients to get through before the compressor hits, you’ll want a fast recovery to get the level back up quickly.

But, obviously, that means you can’t have both a fast attack and release at the same time, which is a perfectly valid setting for some sources, albeit one that you have to be careful with on bass signals — one of the major selling points of the original 1176 when it was launched was its ability to offer much faster dynamic response times than previous designs, but that included a release time short enough for the compressor to act within a single cycle of a low frequency, which can cause audible distortion.

Anyone familiar with an actual 1176 will know that they have switched ratio settings, offering 4:1, 8:1, 12:1 and 20:1. The Ratio pot on these pedals is continuous, without any reference markings, but you’ll probably be using your ears and the gain-reduction metering anyway. There’s no actual Threshold control on an 1176, you just use the input knob to get the amount of gain reduction you want and then set the Output to get the level back to unity or above. In a real 1176, the nominally ‘fixed’ threshold is slightly modified by the ratio selected — the higher the ratio, the

higher the threshold — and I’m not sure that’s particularly evident in these, but that actually makes them easier to dial in. I did find there wasn’t a lot of range to play with at the lower end of the input gain pot with line-level sources, but nothing you can’t deal with by slightly attenuating the source.

What do they sound like? Well, they certainly won’t disappoint: if anything, I think they sound a little more like the original, big Cali76 than the various Compact models do. Very clean and smooth. Perhaps not as ‘fat’, but only because they’re so transparent and fast-sounding, with none of the transient dulling that guitar players are used to with pedal compressors. Of

course, this does all rather depend on the source and the settings used, as well.

The new enclosure depth will line up better with other pedals, but although these might be smaller they still exhibit Origin’s seriously high-quality, heavy-duty build characteristics, so be careful how you mount one of these to a board. It’ll do some damage if it comes loose! With the extra headroom, these really are an 1176 in compact pedal form, offering smooth, musical compression, excellent for guitar or bass and indeed a host of other applications, too. Dave Lockwood

Make Believe Studios MixHead

and used originals change hands for eye-watering prices.

Make Believe Studios’ MixHead aims to capture the spirit of the original hardware. All the common plug-in formats are supported including AAX, and authorisation requires an iLok account.

Unlike the original, sample rates of up to 192kHz are supported and all the benefits of plug-ins, such as offline bouncing and the ability to use multiple instances, add to the flexibility.

As is so often the case with plug-ins that model vintage gear, some additions have been made, such as the 0.1dB adjustment increments for the HF Adjust control (range ±6dB), and a 3.75ips lo-fi mode (modelled after a 1950s Webcor tape machine) that offers an alternative to the original 30ips and 15ips settings. There are separate controls for Input level and Drive, and these behave interactively. Two further controls adjust the EQ and the Output level, so operation is very straightforward.

Buttons set the tape speed and bypass (the red power switch is currently unused).

Snapshots of settings A and B can be saved and there’s a Blend button (MIDI mappable) that can interpolate between them. A faux LCD display shows the

Tape Emulation Plug-in Based on the vintage SPL Machine Head hardware digital rack unit that I reviewed all the way back in October 1997, the MBS MixHead plug-in aims to capture the subtle sound-massaging capabilities of the original, which was an early attempt to replicate the effects of recording to analogue tape. It approximated the saturation of analogue tape — not the wow and flutter or tape hiss — but there’s a lot more to the accurate modelling of analogue tape than simply soft clipping or saturation. For example, real analogue tape machines exhibit a low-frequency ‘wrinkle’ known as head bump, which varies with tape speed. The SPL Machine Head did not seem to emulate this and similarly, the 40Hz to 70Hz bass reduction experienced when running at 30ips was not reproduced.

But while it provided a less accurate tape emulation than the more forensic modern approaches, the Machine Head was very much its own thing — and was prized for its very pleasing sound, not its modelling accuracy. It gained many friends in the pro studio community,

£ £309 each including VAT. W www.origineffects.com

The two pedals run off a standard external 9V power supply, though internally the processing runs on 24V.

current setting of all MixHead parameters, while the four knobs function as rotary encoders for adjusting parameters. Tooltip mouse-over pop-ups are available by clicking the Help icon, but with so few control you may never need them. There are also loads of presets.

Used on a whole mix the effect is quite subtle unless you add a lot of drive, but set appropriately it adds a welcome polish to a mix and helps glue the various elements together. Lifting the EQ helps put an airy sheen on the high end too. Use it on a drum bus and you can coax a lot out of it, from a gentle thickening to a very obvious pushed tape saturation. It can also be very effective on bass guitar and can work magic on vocals. It might seem an odd world where first we model the imperfections of tape and then we model the shortcomings of a vintage device that itself attempted to model the imperfections of tape — but imperfectly. But as always, it’s the result that counts — and like the Machine Head rack unit, the MixHead plug-in has an endearingly musical quality to it. Ultimately, that’s all that really matters!

Paul White £ $179 W www.makebelievestudio.com

www.soundonsound.com / September 2024

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ON TE ST

IK Multimedia Tonex One Amp Emulation Pedal IK’s Tonex software works using a bespoke analysis procedure to capture the behaviour of real guitar or bass amplifiers as a Tone Model, rather than using the more familiar modelling approach. This was followed by the much anticipated Tonex pedal (reviewed in SOS June 2023: https://sosm. ag/ikmultimedia-tonex-pedal) which could run these Tone Models, making them accessible for live use without the need for a computer. But if you tend to use just one or two amp models live, it was arguably overkill — and the new and very affordable Tonex One reviewed here might appeal.

It takes up minimal pedalboard space and could prove a very practical backup in case of an amp failure. A 120mA, 9V PSU is required (none is supplied). A copy of the AmpliTube 5 SE software is included, along with a USB‑C cable, a nice touch being a rubber bung that protects the pedal’s USB port from dirt (and beer?).

While the Tonex One has a smaller preset capacity than its larger sibling, it can still hold 20 Tone Models, and there’s a huge choice available via IK’s ToneNET and Tonex SE. Alternatively, you can capture your own amps or load your own speaker IRs (or IK’s VIR cabinets) using the included macOS/Windows Tonex SE software.

However, there are so many meticulously crafted Tone Models available that, unless you have very specific requirements, you may never need to capture your own. Tonex One works with guitar amps, PA systems or flat‑response amps. The cab emulation can be switched off globally if going through an amp with a guitar speaker/cab emulator.

Plugged directly into the effects loop return of a guitar amp with the speaker emulation turned off is a practical way to work if you don’t want to go directly into the PA. As shipped, the preset slots are pre-loaded and include 13 bonus Tone Models taken from Tonex Max. Any two of the presets can then be loaded into

Universal Audio LA-6176 Channel Strip Plug-in Universal Audio’s vast plug-in range majors on licensed emulations of hardware products. Many of these represent partnerships with other manufacturers, but UA maintain their own manufacturing workshop where they continue to produce high-end analogue equipment, and their

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the A and B slots, which are toggled via the footswitch, and you can load a third preset into the Stomp mode slot. Using three very small knobs and one large knob plus an illuminated Alt button, the current Tone Model can be adjusted using amp-like controls covering Gain/Volume and three-band EQ, plus control over a noise gate, compression and stereo reverb. Any changes to the control knobs for the current preset are automatically saved. By using various combinations of button presses and knob turns, you can browse the 20 onboard presets for loading into the A/B and Stomp slots and also change global settings. The coloured knobs also act as a readout for the built-in tuner, which steers you to the nearest semitone with good accuracy.

The input is mono, though the output can be mono or stereo/dual‑mono if you use a TRS cable. The footswitch can be used to either bypass the pedal (Stomp mode) or alternate between two presets.

All three of the small, transparent knobs have coloured LED backlighting with a choice of custom colours (or off) that can be assigned to your presets for visual identification during live performances.

A safe mode can be used to protect against accidental parameter changes during use. The Tonex One also functions as a class-compliant USB-C audio interface, in which case its TRS/stereo output can feed headphones or active studio monitors.

I have to say that the Tonex One doesn’t disappoint when it comes to sound quality or core features. Library functions, preset

latest channel strip plug-in mashes up two of their own designs.

The hardware 6176 combines the 610 valve mic preamp and the 1176 FET compressor, while the LA-610 pairs the same preamp with the LA-2A optical compressor. The LA-6176 plug-in emulates both, though not at the same time!

handling and editing are all covered by the bundled software, and Tonex SE can also work as a standalone app or plug-in, so please refer to our original Tonex review (SOS January 2023: www.soundonsound. com/reviews/ik-multimedia-amplitube-tonex) for more details about that side of things.

Yes, the Tonex One format demands some compromises — there are fewer physical controls, fewer preset slots and no bank mode for switching beyond the two A and B presets you’ve loaded without first going into browse mode. But if you really need to use three presets while performing, you could always leave browsing mode on, then use the footswitch to cycle around the presets you loaded into the three active slots. Because of the pedal format, the three illuminated knobs really are very small, and I’d also recommend you keep a print‑out of the relevant manual pages to hand for making global changes or changing the loaded presets, as these detail the various permutations of knob/button/footswitch operations required.

As for the effects, I often add a little of the compressor to clean sounds, just to give them a bit more body and sustain, while the noise gate can be useful if you need to clean up really high‑gain tones. The factory presets (which you can adjust for EQ, drive and volume) cover everything from cleans, via bluesy break-up to full-on rock and beyond, all with good touch sensitivity, so if you don’t like using software, you don’t need to. A hint of the reverb can also take away the dryness of the DI’ed sound, which can be especially useful when playing in a space with little or no natural ambience.

In summary then, the Tonex One is a very practical and ultra-compact alternative to a two-channel amp, it sounds just as good as its big brother, it provides a natural playing response and there are Tone Models to suit just about any taste. At less than the cost of many overdrive/fuzz pedals it is something of a bargain. Paul White £ £179.99.

W www.ikmultimedia.com

LA-6176 is available both for the UAD-2 platform and as a native plug-in. Installation and authorisation are now handled by the convenient UA Connect utility but, at the

time of writing, the UAD collection can only be updated en bloc, so if you have an Apollo or a Satelliite, you’re still faced with a 10GB+ download in order to get any new plug-in. Hopefully that will change soon.

The UAD version of LA-6176 is Unison-enabled, so if you load it within Console or LUNA, it can take advantage of the preamp emulation features of Apollo hardware. And this is a plug-in where that definitely makes sense, as the 610 preamp has a strong sonic character. It’s easily driven into mild or obvious saturation, with a distinctively thick, rich tone that’s perfect for warming up practically any source.

Gentle high‑ and low‑shelving EQ bands allow this tone to be shaped further, and there’s also a 75Hz high-pass filter and polarity inversion switch.

This colourful preamp is teamed up with a choice of two equally characterful compressors, neither of which should need any introduction to SOS readers. The 1176 is a fixed-threshold FET compressor that’s capable of extremely fast attack and release times, and the implementation here is pleasingly full, allowing you to virtually press all the ratio buttons at once.

UA have also added a wet/dry mix control that’s not present on the hardware. The LA-2A, meanwhile, is an optical compressor with a valve make-up gain stage, which is famous for its smoothness and subtle coloration. There’s no wet/dry control in this case, but I can’t think that it would really ever be needed with an LA-2A anyway.

The EQ and compressor stages can be independently bypassed.

In use, the LA-6176 plug-in sounds just like you’d expect it to, and I mean that very much as a compliment! These are all classic circuits that we know and love, and the plug-in emulations behave exactly as you’d hope. There are those who like to use both an 1176 and an LA-2A in series on vocals, and you’d need to load two instances of the plug-in to do that; you’ll also need to look elsewhere in the UA collection if you want to get anal about different iterations of the compressor designs, or tweak things under the hood. But that’s not really the point of the LA-6176 plug-in. The aim is to provide an input channel that’s dead simple to use and sounds great right off the bat, and that’s exactly what this does. Sam Inglis £ $299 W www.uaudio.com

Drawmer 1972 Dual-channel Microphone Preamplifier I’ve reviewed a couple of Drawmer products recently: the 1977 channel strip (SOS June 2024: https://sosm.ag/drawmer-1977) and 1971 EQ (SOS July 2024: https://sosm.ag/ drawmer-1971), and been really impressed by both the originality of the designs and the excellent value for money. Reviewed here is the 1972 dual-channel mic, line and instrument preamplifier, which is hand‑built in the UK and puts two identical channels of what appear to be the same preamp design as in the 1977 into a 2U 19-inch rackmount case — though there are some important extras compared with the preamp on the 1977.

As with that device, a front‑panel rotary switch selects the source, and there’s 0 to +66 dB of gain available for mics and ‑24 to +42 dB for line or instrument sources.

There’s an instrument input TS jack on the front panel, so you can plug a guitar or bass straight in, while the balanced mic input has its own XLR on the rear, and there’s a separate combi XLR line input on the rear, meaning you can have the line and mic inputs available separately on your patchbay and there’s no risk of applying phantom power to outboard gear. There’s a single XLR balanced output per channel, and an IEC power inlet for the built‑in PSU. As you’d expect, you can select between line or instrument modes or mic modes using a switch on the front panel. But as on the 1977, there are more options on this switch than on most preamps, because there are three positions for different mic preamp impedances (2.4kΩ, 600Ω and

200Ω), and yet another position for +48V phantom power, with a suitable impedance for capacitor and other active mics. There’s also a polarity invert button, which is particularly handy on a dual-channel device (eg. if recording a guitar with both DI and mic, or both top and bottom snare mics).

There are variable low- and high-cut filters (15-350 Hz and 1.5-20 kHz, respectively). Obviously the low-cut one is useful for cutting rumbles and compensating for proximity effect bass boost, but having both filters here permits more creative uses than most mic pres allow, from rolling off the top end to push a sound back in the mix, to full‑on telephone-style bracket filters. In addition, there are two Shape buttons, each of which engages a tilt EQ pivoting around 500Hz, one putting more emphasis on the bass, the other shifting the focus towards the top end. But when you engage both, you effectively get a dip in the mids, which can also be a useful way to tame those harsh ‘honky’ sounds to which our ears are so sensitive.

Obviously, this is way more control than most mic pres give you, and makes the 1972 a great option if you like to make tonal decisions on the way in — but it doesn’t end there! A final processing feature, and arguably the most interesting of all, is Lift, which can be toggled on/off. We’re told only that this is “essentially a low‑level dynamic gain module which has no effect on signals above 0dB,” so presumably a form of upward expander, but whatever

lies behind this button, it works well: my tests with vocals and fingerpicked acoustic guitar confirmed that it’s particularly good at bringing up the details in very dynamic parts. Nothing that couldn’t be achieved with plug-ins but, again, if you like to shape a sound at source, whether to commit as you go or for live/broadcast use, I could see this coming in very handy.

Coming to the output stage, there’s a level trim control ranging from -10 to +12 dB, and this means it is possible to apply up to 78dB of gain per channel in total for mic signals. It also means you can drive the input stage harder if desired, of course, but while this preamp does sound classy to my ears, it’s probably not one I’d particularly be looking to for special saturation effects.

Finally, a ‑10dB pad switch applies to the signal going to the VU meter rather than the output signal, making the meter much more useful when working with hot signals.

In summary, then, the 1972 is a good-sounding preamp, with separate inputs for mic, line and instrument sources, and some really versatile impedance, EQ and dynamics facilities built in. You could view it as a preamp with extras, or perhaps as a slimmed-down channel strip. Either way, it’s great to see Drawmer again offering something different from the crowd, and if you’re in the market for a new preamp with some onboard tone-shaping facilities, it’s well worth checking out.

Matt Houghton £ £954 including VAT. W www.drawmer.com

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ON TE ST

Sampleson

Drumbada

Plug-in Instrument

HHHHH Conjure an image of Cinematic Drums and you’ll probably be thinking about a battery of percussionists. In the case of Drumbada that means up to 15 Bombos Legüeros, which is a rare form of Andean bass drum that has apparently remained unsampled.

At least until now.

Traditionally, the Bombos Legüeros is played with a stick and can reputedly be heard, in its native setting, from a distance of up to six miles away! It has an exceedingly deep and reverberant tone, and it can be muted and struck in different locations around the drum to provide other timbral colours, such as the familiar rim-clicking sound — useful for those upper rhythmic constructs.

Upon loading, the interface initialises to a slightly unnerving picture of some native drummers, albeit represented by computer imagery. From here, you can access a number of presets and settings from the upper menus, or tweak the patch settings from the row of parameters at the bottom of the window. You can also take full advantage of the macro settings, which can be assigned so that playing behaviour varies as you drag your mouse across the aforementioned graphic.

The Edit page is where the sequencer settings reside. One of Drumbada’s unique features relates to its ability to create polyrhythmic patterns. With five rows of sequencer layers available, each can be engineered to perform cycled loops from one to 16 steps, while the tempo is handled globally or synchronised to your DAW. There is a speed pot, located on the right of every sequencer layer, which allows the note value of each sequencer step to be altered. I might be inclined to label this as ‘note value’, although smaller note values will obviously result in a faster pace, just not a change in tempo, as suggested.

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The resulting polyrhythmic performance can be further exaggerated by the use of the Crew button, which creates the effect of having five players performing the sequenced layer. Increase the Spread amount, and those five players become even more authentic, with a slight relaxing in the tightness of performance.

The Crew function also works in tandem with the Accent pot, which becomes an indispensable parameter for heightening the polyrhythmic effect and tension.

Drumbada delivers a slightly different sound, which does feel reassuringly familiar. As it’s only based around the Bombos Legüeros drum, it might be worth considering this as part of a larger cinematic drum contingent, although as a standalone plug-in it more than holds its own, sounding thunderous and realistic.

The available editability at the note level is appreciated, and goes beyond many other similar plug-ins. A large number of presets are included, with the vast majority appearing to offer a triplet feel, which isn’t always the preference for cinematic writing.

There is also an onboard reverb, which only provides a rather cavernous setting, without any facility to edit the reverb tail.

My tendency is to switch this off and head to a third‑party plug-in, which would allow greater flexibility. Dave Gale $59 www.sampleson.com

Impact Soundworks

Rosette Mandolin Kontakt Instrument

HHHH With Rosette Mandolin, Impact Soundworks have brought a touch of American folk music to their Rosette engine. Compatible with the free Kontakt Player, the sound of the library is very pleasing. While it doesn’t offer multiple mic options, it is recorded in a neutral context and could easily fit into different mixes. The articulation range is solid, the pitched and unpitched releases and choke options are welcome and the legatos are well done.

The harmonics are also very pleasing.

I would have liked to have seen more tremolo options (particularly an unmeasured tremolo), though the tremolo provided sounds good and can be sync’ed to tempo

(it is recorded at about 150bpm, so you just have to be careful how you use this as at extremes the time stretching is evident).

The articulations page allows for quick customisation; I initially found the sound of the release samples a little too loud, but could easily nudge this back a little.

Each pair of strings has been recorded from the open position all the way up the neck, with upstrokes and downstrokes, and these samples can be triggered in a variety of ways.

Being a polyphonic string instrument, the library takes a little more learning than, for example, a piano library, but Rosette Mandolin offers considerable playability and customisation options.

The Performance Style presets do a good job of ‘mandolin-ifying’ a keyboard player’s input. Selecting between the different neck positions using the Performance Style presets (or for more precision, the Hand Position knob) not only intelligently selects the way of playing particular notes (ie. which string), but the display helps to guide the user’s decisions, and the more I used the library the more I felt I was learning how to write idiomatically for the instrument. The workstation mode is brilliant for cheating something that sounds like a mandolin, but doesn’t conform to the instrument’s restrictions.

This is likely the best place to start for keyboard players.

The multitracking functionality creates a successful illusion of multiple mandolins playing together.

The strumming functionality works well. The library uses high range keys (unused by the samples) to trigger full and partial up and downstrokes. Varying the Strum Speed, Strum Volume and Strum Distance controls allow for more realistic programming (I found these parameters easier to control once mapped to a CC controller, which the library allows for).

There are a couple of odd quirks; the partial strums are silent at a low strum distance level, and releasing the sustain pedal creates an odd noise

artefact. Impact Soundworks are looking at these, so hopefully these issues can be fixed in a future update.

Rosette Mandolin provides a good sound that could have many applications, with versatile input options that cater to writing realistic mandolin lines, impossible 5+ note chords and everything in between.

Thomas Field $129 www.impactsoundworks.com

Emergence Audio

Infinite Upright

Kontakt Instrument

HHHHH If atmospheric, textural, ‘music meets sound design’ piano sounds are of interest, Emergence Audio’s Infinite Upright is absolutely the right sample library for you. The source material, recorded on a vintage 1980s Baldwin Upright piano, was processed, according to the literature, “through a combination of Eurorack modular systems, advanced digital processing, our Non-Static Sampling process, and powered by our Infinite Motion Engine 2.0.”

Emergence Audio aren’t exaggerating when they put ‘infinite’ in the title. With all the tools on offer for shaping the sound, the possibilities for composing with the Kontakt instrument truly are endless. The library features over 200 presets; 50 impulse responses (from real‑world rooms to legendary Lexicon reverbs); over 100 customisable parameters including ADSR envelopes, LFO modulation, low-/high-pass filters, pitch adjustment and polarity flip; and effects like delay, tape saturation, chorus, phaser and rotator, which simulates the sound of a rotating speaker cabinet.

Where to start? For a pure piano-based sound, load an Infinite Keys preset. To explore moody, evolving performances, try the Drones or Pads. If you’re feeling experimental, choose from the Experimental selection or load a Multi which, as the name suggests, is a combination of multiple patches. Finally, if you’re feeling lucky (that is, too tired to audition presets), use the Randomizer function to create a random combination of layers to work with. While the library does take up a fair amount of space (9.7GB installed), it is remarkably light on CPU load. Multiple instances load without trouble and play back smoothly — it just feels like a well-built instrument.

The interesting thing about using Infinite Upright is that even though you play the

instrument as a solo musician, it feels oddly like a collaboration at times. As the layers evolve, the sounds they generate provide feedback (of the creative kind, not the ear-piercing kind), which helps shape the composition. Just holding down a chord and listening to the modulations generates ideas for melodies, and the multi-layered textures often conjure up moods that then inspire chord selection. Especially if you’re the kind of composer that leads with feel and instinct, Infinite Upright is a delight to play.

Virtual piano instruments are in no short supply, so Infinite Upright stands out because it brings a generous set of tools to the table with its evocative textures and evolving samples. The core piano sounds are pristinely captured, with a warm tone and wonderful musicality. Combine that with the power of synthesis and you’re equipped with the raw material to build a richly layered piece of music. A bonus — which is not exclusive to Infinite Upright, but beautifully highlighted by it — is that when you pay attention to the many ways in which you can shape tone, you realise exactly how many elements go into making every single sound what it is. Sonal D’Silva $149 www.emergenceaudio.com

Soniccouture

Pan Drums II

Kontakt Instrument

HHHHH It’s a sound you’ve likely heard in TV and film soundtracks but may not be able to readily identify: a light, melodic, melancholy metallic chime reminiscent of a Caribbean steel drum but lacking that instrument’s strident attack and carnival atmosphere.

This unique sonority is the Hang, a metal flying-saucer-shaped tuned percussion instrument played with the hands.

It was my pleasure to review the original Hang Drum library in June 2008, since when Soniccouture changed the product name to Pan Drums and added extra ‘handpans’ (the generic name for Hang-style instruments). The new Pan Drums II adds further content, bringing the instrument count up to seven including the original pair featured in Hang Drum. All of these sampled instruments can be played chromatically, but if you want to recreate

their authentic pitches, simply play the notes marked in purple on the Kontakt keyboard.

The library (7GB installed) requires Kontakt 6 or Kontakt 6 Player version 6.8 or later.

I immediately fell in love with the Halo Genesis (aka Halo in B) handpan made by the US company Pantheon Steel. Larger and lower-pitched than the Hang, it has a purer fundamental tone and a gentle attack which works beautifully for quiet ostinatos, in which role it could be mistaken for a harp. Its Halo in C partner sounds slightly more percussive.

These handpans were played by their inventor Kyle Cox, who brought two of his favourite Halos for Soniccouture to record.

Lovingly hand-crafted in Bristol, UK, the Meridian handpan has a peaceful, soothing tone reminiscent of harp harmonics, while its ‘edge’ performances introduce a nice pitter-pattering attack. Played in its original tuning (a D minor scale without the minor sixth) the instrument takes on a dreamy, almost hypnotic quality. By way of contrast, the US-built Rockcreek steel tongue drum is played with mallets, producing a clear, bright sustaining chime with Caribbean overtones.

Pan Drums II’s final entry is the impressive Rav Vast, a large, Hang-shaped steel tongue drum with a deep pitch, precisely tuned overtones and a solemn, lingering sustain.

The instruments were deep-sampled with up to 60 velocity layers using a variety of striking techniques: fingertips, thumbs, knuckles, edge and corner hits, slaps and mallets (styles vary from drum to drum). The range of acoustic textures is wonderful, but things get even better when you dig into Soniccouture’s excellent sound‑design presets: these utilise the library’s built-in effects to create haunting, ethereal leads and pads, alien electronic textures, Radiophonic tape delays, delicate high bells and a great upright bass emulation.

Also lurking in the sound design folder are inspirational rhythmic pulses and evolving arpeggios created by the Jammer generative tool in conjunction with mutated handpan sounds. You’ll have hours of fun with these funky gamelan grooves, Afro-flavoured jams, brushed drum syncopations and manic industrial rhythms.

Happy handpanning! Dave Stewart £119 www.soniccouture.com Audio examples of this month’s libraries are available at www.soundonsound.com.

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Digital Performer

DP’s enhanced instrument tracks offer some powerful new options.

MIKE LEVINE

I

n DP 11.1, MOTU introduced updated instrument tracks. For the first time, DP users can instantiate a virtual instrument, record it and edit it on a single track. Previous versions of DP required an instrument track and a separate, linked MIDI track for any virtual instrument.

The new instrument tracks have plenty of useful features and options. The more you understand them, the more creative control you’ll have over your MIDI production in DP.

Getting Started You can create an instrument track by going to the Project menu and selecting Add Track / Instrument Track. You’ll then be presented with a list showing the developer names for any AU, VST or MAS instruments installed on your computer.

Choose the developer and the desired instrument from the sub-menu, then release the mouse to create the track. If you hold down the Option/Alt key while accessing the menu, the word Multiple appears before each instrument in the menu.

When you then choose an instrument, a small dialogue appears allowing you to create two or more instances of that instrument in one operation.

As you can see in Screen 1, there are several choices in the Project menu for creating instrument tracks. Selecting Instruments with Options opens the Add Instrument Track dialogue, which allows you to create more than one instrument track. It also gives you the option to add a multitimbral instrument; that is, one that receives on multiple MIDI channels simultaneously (Screen 2).

One of the fields in the Add Instrument Track dialogue allows you to specify the number of instrument tracks to create. It enables you to assign

Screen 1: The instrument track options.

them all to one instrument or leave them temporarily unassigned.

The dialogue also lets you specify how many MIDI tracks to add to an instrument track. This option is handy if the instrument track you’re creating is for a multitimbral virtual instrument — for example the free UVI Workstation, which can support up to four parts on separate MIDI channels (DP owners can download a free soundbank for it). You’d choose Instruments with Options, assign the track to UVI Workstation and specify four MIDI tracks, with or without a Folder. The MIDI tracks will appear above the instrument track in the Sequence Editor and are organised by MIDI channel number.

If you have a session with many instruments, DP’s Folders can help you stay organised in the Sequence Editor or Tracks window. An open Folder shows all of its tracks. A closed folder hides everything but the Folder itself. If you choose the Folder option, the instrument track will be at the

bottom, and its associated MIDI tracks will appear above it, as in Screen 3.

Another option under Project / Add Track is Unassigned. It creates a single unassigned instrument track. In that same menu, you can also choose Multiple Unassigned Instruments, which opens up a modified Add Instrument Tracks dialogue (Screen 4), the sole purpose of which is to specify how many unassigned instrument tracks to create.

Another way to create instrument tracks is via Project / Create Tracks, which allows you to conveniently add any combination of multiple audio, MIDI, aux, or virtual instrument tracks, with consecutive channels and output assignments.

Making Changes To assign a virtual instrument to an unassigned instrument track, go to the Mixing Board and instantiate the instrument in the Track’s instrument insert slot (second

Screen 3: MIDI tracks assigned to UVI Workstation in a Folder.

Screen 2: The Add Instrument Track dialogue.

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Screen 4: You can add multiple unassigned instrument tracks with a single click.

from the top and shaded green). Or, if you have a MIDI track that you want to assign to an instrument track, do so using the MIDI Output menu in the track settings panel, as shown in Screen 5.

If you have several existing MIDI tracks and want to combine them into a single instrument track, select the instrument track and the various MIDI tracks, and hit Shift+Spacebar (or go to Setup / Run Command). In the Search field of the Run Command dialogue that pops up, type Merge by Channel (Screen 6). Select the command and hit Enter to invoke it. DP will merge the MIDI tracks into the instrument track. If they were assigned to different MIDI channels before the merge, they’d keep those assignments because DP supports multi-channel MIDI tracks. If you want the various MIDI tracks to play on the same channel, ensure the output assignments of the tracks are the same before invoking the Merge by Channel command.

If you decide to split them out of the instrument track later, use the Split by Channel command. That will work on any multi-channel MIDI track.

For older projects with legacy instrument tracks, DP offers an option to convert them to the new format the first time you open the project in version 11.1 or higher. You’ll need to do this if you want them to have the latest functionality. If you initially choose not to convert them, you can do so later, at any time, by opening the Run Command window and using the Convert Legacy Instrument command (Screen 7).

Drilling Down DP’s new instrument tracks are hybrids, because they have MIDI inputs and audio

Screen 6: The Merge by Channel command can combine multiple MIDI tracks into one instrument track.

Screen 7: Using this command, you can convert instrument tracks from older sessions to the new format.

outputs. The topmost insert slot on an instrument track is for insert presets.

The second insert from the top (shaded green) is for the instrument plug-in itself.

Directly below that is a section of insert slots for audio plug-ins, followed by a second group of inserts for MIDI plug-ins.

These insert slots are shown in Screen 8.

Instrument tracks can be soloed or muted, just like audio tracks, with one key difference. If you’ve instantiated a multitimbral instrument and want to solo one of its nested MIDI tracks, you need to select solo on both the instrument track and on the specific MIDI track of the instrument (Screen 9).

The solo behaviour changes if you turn on DP’s Partial Solo feature.

Partial Solo lets you solo one track in a multitimbral instrument and still hear the rest of its MIDI tracks, but lowered by a user-adjustable percentage of the MIDI velocity (Screen 10). To turn on Partial Solo, Option/Alt‑click the Solo button in the Control Panel. This will open the Settings window with a MIDI Solo Setup field in the middle. You’ll see a pop‑up menu that says Off. Open it, choose a percentage and click Done.

audio volume and audio pan, and you can edit this audio automation data in the Sequence Editor). But if you’re using a multitimbral instrument with multiple MIDI tracks, audio automation will be applied to all of the parts at once — to apply it to just a single part in your multitimbral instrument, use MIDI CC7 for volume and

CCs & Host Automation Screen 5: The MIDI Output field lets you change channels or VI assignments.

For a single instrument track with MIDI data in it, the track’s fader and pan knob can be used to write audio automation, just like on any audio track (these will be referred to as

Screen 8: You can insert MIDI effects and audio plug-ins Screen 9: To Solo the Clav track in on instrument this multitimbral UVI Workstation tracks. instrument track, both must be soloed.

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TECHNIQUE

INSTRUMENT TRACKS

Digital Performer parts one at a time — along with the instrument track they belong to — and run separate track-freezing passes for each. That’s because DP uses the same destination track to freeze each part of a multitimbral instrument track. Although it retains all the freezes in the Soundbites window, it doesn’t keep them all on the destination track.

An easier workflow is to freeze the part, use the Add Similar Tracks command to create a new audio track below it, and copy the newly frozen track into its own audio track.

Bouncing VIs Screen 10: The field for turning on and setting the percentage for Partial Solo is highlighted.

CC10 for pan. Be aware of this when editing automation data in the edit windows.

Resource Management When building a large DP template, instrument tracks can be temporarily disabled by clicking on the enable button in the Tracks Window ‘ENA’ (enable) column.

Disabled tracks don’t use up any CPU resources, and they load very quickly when opening a project. You can freely enable and disable them as needed.

Virtual instruments and audio effects can be processor-intensive. Fortunately, like audio tracks, you can freeze instrument tracks in DP. To do so, select the instrument track and its contents, open the Audio menu and choose Freeze Selected Tracks. DP will render the track onto a new audio track with the word “freeze” after the track name.

However, there’s a critical difference between freezing an audio track and an instrument track. With the former, DP automatically disables the original track after it freezes it. With an instrument track, you must manually turn it off by assigning its output to None. The reason is that you might want to do more than one freeze with a multitimbral instrument.

Because DP freezes a track at its output, a frozen track will include all volume, panning, automation and effects that are on it. On a multitimbral instrument track, all the parts from the various MIDI channels will be rendered together on one frozen track, as in Screen 11.

If you’d prefer to freeze the individual parts from within the multitimbral instrument onto separate tracks, you’ll have to solo the

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In addition to freezing tracks, DP11 allows you to bounce VIs, either in real time or offline (faster than real time). Select the MIDI data (or region) in one or more instrument tracks and choose File / Bounce to Disk. Choose Outputs and/or Tracks depending on the audio routing for the instrument tracks. Many virtual instruments offer multiple audio outputs once they are instantiated and enabled. These channels will be presented as available instrument bundles (and show up in the input column for audio tracks). Use aux tracks in DP to monitor or bounce the outputs.

Input-monitored stereo audio tracks could also be used to print additional outputs from virtual instruments in real time.

Additional Tips & Shortcuts There are a variety of ways to open and view the UI for the instrument instantiated on an instrument track: • Double-click the VI name in the Tracks window, Sequence Editor, or Edit window.

• D ouble-click the VI name in the Track Inspector.

• In the Sequence Editor, choose Open Instrument from the Track Settings menu.

You can also assign a shortcut or invoke it from the Run Command window (Shift+Sacebar).

• In the Mixing Board, double-click the track’s instrument insert.

Track Assignments (Option/Alt+A) is a useful command for assigning selected audio and MIDI outputs at the same time. Insert Presets can be created and saved to recall an instrument track along with any additional audio and MIDI plug-ins. Simply create an unassigned instrument track and then choose the saved Insert Preset (from the track’s topmost insert menu in the mixer, as shown in Screen 8).

Instrument tracks can be moved into a V-Rack for shared use across multiple sequences (chunks) in the DP project file. To do so, select one or more instrument tracks, right-click on any selected track’s name, and choose Move Selected Tracks to V-Rack.

When doing so, DP will leave an assigned MIDI track in the Tracks window after moving the instrument track to the V-Rack.

The MIDI data in instrument tracks can be viewed in the QuickScribe Editor, even if the data is packed into clips.

Choose Setup / Enable Live Performance Mode to run all instrument tracks and audio effects in real time.

This eliminates any audio interruptions when opening and closing effects and instruments while working. Keep in mind, however, that this may place very high demands on your computer’s CPU, depending on how many effects and instruments you have instantiated and how processor-intensive they are.

Screen 11: Unlike an audio track, DP doesn’t automatically disable an instrument track when you freeze it, so remember to do so after the freeze.

PODCASTS RECORDING

PEOPLE &

& MIXING

MUSIC INDUSTRY

How To Remove Studio Hum

Daniel Miller - Mute

Paul White and Hugh Robjohns discuss the best practices for avoiding and eliminating electrical noise interference in your home studio.

Daniel Miller, founder of Mute Records, talks about his entry into the music industry, the origins of the label, and of his enduring love of modular synthesis.

Eliminating Electrical Noise

The MixBus Interview

ELECTRONIC

ELECTRONIC

MUSIC

MUSIC Oberheim

Jason Singh

Synth designer Tom Oberheim tells Sam Inglis about his early years as an engineer and his accidental route into creating electronic musical instruments.

Jason Singh is a sound artist, producer and performer. His creative output is an exploration of the natural world, voice and a wide range of music technologies, including the MIDI Sprout interface.

50 Years Of Synth Design

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Logic Logic

TECHNIQUE

Logic’s virtual drummer now has a full band to play with! We find out what it can do...

PAUL WHITE

L

ogic 11 introduced a few major new features, including Stem Splitter (Apple Silicon only) and an enhanced Session Player section. Most Logic users will be familiar with the Drummer Session Player track, which offers a range of styles, sounds and user input options enabling it to create drum parts that fit most kinds of contemporary music, from hip-hop to ballads. In Logic 11, Drummer has been improved with new sounds (the updated SoCal kit sounds great) and seemingly better constructed patterns, but the really big news is that it has been joined by Bass and Keyboard Session Players, all of which share the same style of user interface.

Session Drummer Drummer now has a different look, with all the available acoustic kits now listed in the Browser — and all those cheesy names for the different drummers have gone.

If you go to create a new Drummer track as before, you’ll now see that the Session Player menu shows the new Keyboard and Bass options.

Once you’ve created a Drummer track, you’ll see a default drum pattern created as a region, and the Session Player panel opens automatically. If you close it, you can open it again by double-clicking in the region. As before, Drummer regions (and now also Keyboard or Bass regions) can be copied and stretched, and new ones can be added using the plus symbol at the end of the region. Click on the style shown

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above the circular graphic of a drum kit (the default is Pop Rock) and you’ll see a menu listing types of drum kit at the top, which can be Acoustic, Electronic or Percussion.

Below this is a list of styles: for acoustic kits, these are Rock, Songwriter, Alternative or R&B, with a few variations listed below each heading. Once you’ve picked your style, the Preset window in the upper left of the Session Player window provides a choice of rhythmic variations on that style. The trick is to pick the style that comes closest to what you need, then use the controls in the Session Player window to reshape it to your track — for example, by making the part less busy or simplifying the fills. The controls for Complexity and Intensity are now simple sliders, and there are icons to switch on or off the various kit elements and to choose separately from rhythmic variations for the kick/snare and the rest of the kit. Fill amount and Complexity are adjusted with knobs as before, as is the degree of swing added to the beat.

Click on Details and the main panel is replaced by controls accessing deeper options such as the type of snare hit, the strength of ghost notes, the method by which the different tom hits are chosen, whether or not you want to use added percussion (shakers, claps, tambourine), set double‑ or half‑time tempo, set playing dynamics, Humanize, and adjust Feel, the latter enabling you to pull or push the beat.

A third tab named Manual lets you input your own rhythms for the kick and snare using a drum‑machine‑style grid. That’s a lot of scope for getting the right groove

The Drummer Session Player has had a major overhaul, including a streamlined Session Player window and a number of new drum sounds.

for your song, but if you still feel that tweaks are necessary, you can convert the track to MIDI by right‑clicking on the Drummer region and selecting the Convert to MIDI Region option. This feature is also available for the Bass Player and Keyboard Player tracks. I often convert my drum parts to MIDI so I can fine-tune the drum fills or remove unnecessary hits.

If you change to an electronic kit or to percussion, the main view is similar except that it breaks the kit into three sections rather than two, and the Details section has separate complexity sliders for the various kit elements. A further slider adds phrase variations. Drum Machine Designer is where you’ll find all the electronic kit sounds. The Manual page works here as before, but now you get three grids for setting your own rhythms for the kick, snare and claps.

Home On The Arrange Before I move on to the Bass and Keyboard Session Players, it is worth mentioning Logic’s Arrangement track, which you’ll find under the Global Tracks section. This allows the user to mark out sections of a song, and by default creates Intro, Verse, Chorus, Bridge and End sections, though you can also create your own custom‑named sections. These markers are not just there for navigation: when you create Session Player tracks, the parts adapt according to the section of the song mapped out in the Arrangement track, so the pattern will be

If you want your Session Players to follow what’s going on in your song, you’ll want to use the Arrangement and Chord tracks (shown here at the top), to tell them what section of the song you’re in and what notes to play.

slightly different for the verse and chorus, for example. This can be turned off if you don’t want Logic making those decisions for you, but the automatic choices often provide a good starting point from which to work.

The Chord Track The newly added Bass and Keyboard Session Players both benefit from new and improved additions to the instrument library (specifically Session Piano and Session Bass), and both work by following the chords in your project. There are a few standard chord progressions that can be loaded into the Chord tracks (which are again part of the Global Tracks), but in the main, you’ll probably want to enter your own chords directly at the cursor position using the Add Chord (plus symbol) button or by right‑clicking. These actions bring up a chord selector window where you can enter the root note and the type of chord. Chords in the Chord track can be handled much like conventional regions as regards moving, cutting, length changes, copying and pasting, and you can group a number of chords together so that they can be copied as a single block.

Chords from a Chord track can also be embedded in regions by Control‑clicking on the Session Player region header and then

selecting the Chords sub‑menu, which includes the options to paste chords to or from the global track. Where Follow Region Chords is selected, those regions will play from their embedded chords rather than from the master Chord track, and when the region is copied, its chords go with it.

A chord can also be entered from your MIDI keyboard by clicking the MIDI Input box in the Chord dialogue box, but this still only allows one chord to be entered at a time. You can’t input a performance and then have the chords extracted automatically — which brings us to the biggest user gripe concerning the new

Session features, which s that there’s no automatic way to extract chords from a live input or MIDI track that you’ve already played in. Given the number of comments on this subject, I suspect Apple will add an automatic extraction feature at some point, but in the meantime there is a workaround, albeit a rather clunky one...

If you go to the track inspector and set Internal MIDI In to the track you’d like to follow, the name of the chord will show up in the Bar and Beats display as the track plays. You can either make a note of these and then enter them manually, or select MIDI Input in the Chord selector window and then press Play — but you need to press Stop again before the chord changes. The last chord that plays is the one captured, so you could be in for a lot of starting and stopping when working this way, but it does get the job done. If you you need to grab a very short chord, you can set a loop to keep going over it so the song doesn’t get to the next chord before you can press Stop.

It is worth noting that Apple Loops accompanied by a small music symbol contain chord data that gets loaded into the Chord track automatically when the loop is dragged into the song. Furthermore, if you create a region using Keyboard Player (or Bass Player) with your own chord sequence embedded, you can drag it into the Apple Loops library for reuse at some

The new Keyboard Session Player in action.

www.soundonsound.com / September 2024

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Logic Logic

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S E S S ION P L AY E R S

future time. When you come to reuse it, you can change the Keyboard Player settings so all that’s really important is the chord sequence, not what the Keyboard or Bass Session Player is doing.

Session Player can be set to follow the Chord track The Session or chords embedded in the Player controls region (from the Chord menu are largely at the top left of the Session consistent across all three Window), and it is possible instruments. to edit chords that have been copied into the region from within the Session Player window. voicing and select the amount of movement Double‑click on the chord shown alongside in the performance. The Fill Amount, Fill the region pattern and the Chord Edit Complexity and Swing controls are the window opens up, enabling you to select same as for Drummer, the first two dealing a new chord. If the song plays past the point with how the player fills in between chords. where chords have been created, either for Details deals with grace notes, Feel, the Chord track or in a region, the Session Dynamics and Humanize, again much Player instrument will keep playing around as with Drummer. The tempo can also the last chord indefinitely. be halved, doubled, or set to Auto in any of the Session Player windows.

Consistency

Looking at the new Session Players reveals that they have much in common with Drummer when it comes to how the GUI is set out. The Keyboard Session Player offers a choice of five basic playing styles, Freely being the one that sounds most like a good accompanist and the others providing various ways of dealing with chords (Broken, Block, Arpeggiated or Pad).

The default instrument is the new Studio Piano (other than for Pad, where you get a patch from our old friend RetroSyn), but you can switch to any instrument of your choice if you prefer. There’s a lovely new vintage upright piano amongst the four piano types found in Studio Piano, along with three new grand piano options.

Each style has a number of preset variations, and for the Keyboard Session Player, you can switch off the left- or righthand parts, adjust the complexity and intensity, and designate a rhythm track for the part to be guided by — this would usually be the Drummer track, if used. This can be switched on in the panel where you usually select alternative playing patterns for both the Keyboard and Bass, near the top centre of the main Session Player view. The default is Follow Chord Rhythm, but you can always click the Manual tab for any of the Session Players if you need to create your own patterns using the drum‑machine‑type grids.

With the Keyboard Session Player, you can select the type of right‑hand chord

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Bass In The Place Bass Session Player adopts a very similar paradigm, with basic style types comprising Rock, Songwriter, Alternative or R&B, with a number of sub-styles and preset variations available. There’s a wide choice of bass sounds, with the new Studio Bass instrument offering a choice of basses along with finger, pick or slap playing styles. Studio Bass’s Detail section lets you control string noise, hiss and hum, string release noise and whether or not to use open strings. These options are somewhat reminiscent of IK Multimedia’s excellent MODO Bass, though I couldn’t find a fretless option in Logic’s Studio Bass repertoire. Articulations are normally chosen automatically by the Bass Session Player, but there’s a huge list to choose from (harmonics, damped notes, slides and so on) if you want to control this yourself.

How well do all these parts fit together?

Usually they work well, but if you pick a drum rhythm that has some unusual timing variations in it, you might find that that Bass, Keyboard and maybe other MIDI tracks don’t lock in as tightly as they should. The fix for this is Logic’s Groove Track feature.

Make sure that Groove Track is checked where you configure track headers, then tick on the track number of the part that you want the other to follow. A star appears to the right in the newly created Groove view, with tick boxes for the remaining MIDI tracks. Tick the boxes and those parts will

lock in to the drum track. The Session Player instruments also have the option to follow the rhythm of another MIDI track, which can often improve the feel of the end result.

Impressions The Drummer Session Player sounds have definitely been improved, with better kits and the newly introduced Studio Pianos and Bass, while the new‑look Session Player window is consistent across the Drummer, Bass and Keyboard. The way the Bass Session Player incorporates articulations such as slides and damped notes when using the new Studio Bass instrument is very convincing. While you won’t get virtuoso performances or killer riffs out of the Session Players, you can expect some very capable backing parts, which is what they were designed for, and with just a little adjustment, you can make them fit most styles. The only areas that seem underserved are the slower chillout and perhaps world music genres, which would make useful additions to the electronic drum and percussion repertoires in future, though you can still conjure up your own chilled beats by using the half‑tempo function and/or some creative control tweaking.

I still like to convert the final versions to MIDI just so that I can fix any minor issues that don’t agree with my vision of how the part should go, as just occasionally some odd note choices or timings sneak in that I might want to fix, but in the main the results are musically solid. I’m sure that users’ comments regarding the need to be able to extract chords from an existing MIDI part or live input have already been noted

by Apple so I’ll be very surprised if we don’t see an eventual update that makes that possible. And remember that Logic Pro 11 is a free update, so for the time being, let’s make the best use of the features we’ve been given rather than complaining about those we haven’t!

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Pro Tools

TECHNIQUE

The new Import Session Data window. The interface has been streamlined, the window is resizable, and the filtering options take full advantage of the newly updated PTX file format.

customised set of data on which to work.

Updates to the Pro Tools Session file format have made aHere are some of the highlights.

One of the biggest improvements the Import Session Data window even more powerful.

JULIAN RODGERS

O

ne source of frustration among Pro Tools users is the fact that you can only have one Pro Tools Session open at a time. It’s standard practice when working in almost all other applications to have multiple projects, documents or whatever open at once, and to be able to copy and paste, or drag and drop content from one to another. This isn’t possible in Pro Tools, but that isn’t to say that you can’t transfer content from one Session to another, and the place you go to access the contents of a Pro Tools Session other than the one you’re in is the Import Session Data window.

It’s fair to say this window has had too much squeezed into too little space for quite some time. It’s been ripe for an overhaul, and in Pro Tools 2024.6 that is exactly what it received. The changes are more than superficial; there has been a serious amount of work done.

In this month’s column we’ll have a look at some of the things that have changed.

if you reuse plug‑in chains or even entire tracks or groups of tracks complete with media, you can do so by saving them as track presets or in template sessions.

However, the Import Session Data window offers the ability to access and reuse almost anything from another session, with some handy time‑saving features such as Match Tracks, which imports to tracks of the same name between sessions.

In use, however, the window could feel cramped, like wallpapering the hall through the proverbial letterbox, and the Import Session Data window was an intimidating place for newbies. What the overhaul has set out to achieve is to update the UI, make it more usable, and add some under‑the‑bonnet work on the Pro Tools Session file format to allow comprehensive filtering, so that the more streamlined interface can display

over the old, cramped window is that it is resizable, meaning you can now expand it horizontally to display longer track names and vertically to accommodate more tracks. The old central ‘letterbox’ window for displaying tracks, and inevitable scrolling of the list, has been moved over to the right, with the tab for markers also moved to the same location. The same options are presented, but the information has been reorganised. The Source Properties pane, which always felt very oversized in such a crowded window, now has the option of being collapsed, containing as it does no editable parameters. The Track Data to Import button, which previously opened an over-long drop‑down multi‑select menu, now opens a floating window, which features five quick preset buttons as found elsewhere in Pro Tools. These are particularly useful for long menus like this.

Data Crunching Many pros rely on this powerful window, which can be used to import anything from plug‑in settings, audio clips and automation data to markers, memory locations and window configurations. There are alternative ways of sharing some of these attributes between sessions; if for example you regularly use the same plug‑in settings, they can be saved as plug‑in presets. Or

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The Show Only options. Note that the tracks in the list now also have icons showing what type they are.

You can now choose to automatically display automation lanes with automation — handy for sprawling Foley sessions!

But it’s in the right‑hand pane, which replaces the old track and ruler display area, where things have really changed.

Having more space courtesy of the new layout and the resizable window is of course welcome, while the slightly anachronistic use of text to describe the track type in the tracks list as been replaced by the track type icons, which should be familiar from the Mix and Edit windows. Track colours have also been introduced to the list.

Into The Fold There was previously a potential issue with the import behaviour of Folder tracks: the member tracks weren’t included. This new window displays the Folder track structure in an easily readable branching format, and provides the option to import all member tracks of a folder, even with the Folder track folded shut. A circle icon next to the Folder track can be used as a button while simultaneously displaying the selection status of the enclosed tracks. In much the same way as the dots next to Mix and Edit Groups in the Mix and Edit windows, sidebars display the selection status of member tracks. A solid circle indicates all

What Else Is New? Pro Tools 2024.6 was a big update, and the headline billing went to the six new ARA 2 integrations. Adding to the Melodyne integration we already had, these had been anticipated for some time; for music production, the major new additions are Sound Radix’s Auto Align 2 phase alignment tool, and Synchro Arts’ VocalAlign and RePitch, which offer doubling, take alignment and tuning for vocals. If you need even more forensic control, Synchro Arts’ Revoice Pro is also now integrated, as is iZotope’s RX Spectral Edit. There is a lot here, and it rightly received the lion’s share of the attention, but for busy professionals, the updates to the Import Session Data window might be most important.

that are not present in the target session will also be greyed out.

When suitably filtered, the relevant tracks can be assigned a destination in the currently open session, either manually or using Match Tracks, which assigns by matching the track names.

Tracks can be filtered by track name and further filtering operations can then be performed on filtered results so you can, for example, filter by track name ‘Guitar’ and then filter the results by track type to omit the folder tracks, leaving only the audio tracks that are active.

member tracks are selected, and a dot with a circle indicates some but not all tracks are selected.

The Folder track button follows the same convention, but if you click on it, it toggles the selection Down The Lane status of all the member tracks.

The changes to the layout and display As you dig into the detail there is more in the window are welcome, but the real to learn about the Import Session Data power in this overhaul comes from work window. If you work collaboratively, and Avid have done in the Session file format especially on complex or large sessions, itself. The import experience using the you’ll find more benefits to the new features, new window with legacy session files but to close I’ll instead point out some of the saved in older versions of Pro Tools is ways this refresh of the Session format and much improved, but some new features summary data helps elsewhere in Pro Tools. are only available in sessions saved using In the Track Show and Hide menus a version of Pro Tools from 2024.6 onwards. (accessed from the disclosure triangle at The way the Import Session Data window the top of the Tracks List in the sidebars works is that it accesses special summary of the Edit and Mix windows) are two data stored in the PTX file. This is how new entries, each giving the option to the window is able to display information show or hide tracks with clips, or tracks about the session without opening it. PTX with clips within the timeline selection. files already contained information about In a similar way to displaying additional things like track type, track width and track automation lanes underneath the main name, and by adding additional information, playlist (as opposed to displaying a single important new operations have been made automation playlist on the main playlist possible — for example, filtering by track lane), it is now possible to automatically colour, hidden state, frozen state, and by display automation lanes with automation, the presence of clips on the track. or automation lanes with automation This new summary data makes it within the selection. The last of these is possible to identify inactive tracks, a crucial particularly welcome as it can sometimes piece of information when importing only be time-consuming to track down stray the content relevant to a project. Likewise, automation in old or inherited sessions. tracks without any clips on them may be This feature will really speed that up. unnecessary. Once you can identify these The methods used to deploy the properties, you can filter them out of your search and filtering in Pro Tools can be tracks list prior to import. leveraged in multiple places throughout the If you open a legacy-format Pro Tools application, and the improved summary data Session in the Import Session Data window being of use in more places than just the you’ll see a warning that some track list Import Session Data window is an excellent filtering options are unavailable. In the example of this. filtering list, you’ll find that the top six options for filtering (hidden, frozen and inactive tracks, and tracks without clips, automation and markers) are all unavailable. This is due to the relevant summary data not being included — saving the Session Sessions from earlier Pro Tools versions won’t show all using 2024.6 resolves this. Track types the new filtering options until they’re saved on v2024.6.

www.soundonsound.com / September 2024

139

Pro Tools Cubase

TECHNIQUE

Lower a mix’s integrated loudness with Squasher.

JOHN WALDEN

T

he ‘LUFS’ measurements that are used by streaming services to normalise loudness on playback, and which are now catered for by so many meters and limiters, aren’t just weighted (with an EQ curve that accounts for the sensitivity of human hearing to different frequencies) but they’re also ‘gated’: any portion of a signal where the momentary loudness falls below a defined level (-10LU below the integrated loudness) won’t count towards the final integrated loudness.

When undertaking DIY mastering duties, the presence of this ‘floating gate’ raises two significant questions. First, for any given track, are there significant portions of time where the momentary loudness falls below the gate’s threshold?

Second, if the answer is yes, might there an opportunity to raise some of those quieter sections just above the level of the gate, both to add energy or clarity, or to lower the integrated loudness — for example, to reduce the amount a track might be turned down — without negatively affecting the louder sections?

When I recently (SOS July 2024) reviewed iZotope’s RX 11, I was particularly impressed by its Loudness Optimize feature, which was conceived to address precisely those questions. As a Cubase user, that got me wondering... could I achieve something similar using only the tools that come bundled with Cubase?

Analyse This For Cubase Pro users (v11 and above), the Loudness Curve module of the SuperVision analyser plug-in can help with the first question. Inserted at a suitable position in your project, this provides a real-time visual display of your audio’s loudness.

The Loudness Curve module settings (first screenshot) allow you to customise the display. You can, therefore, set the time resolution of the display to suit your needs (I used a 45-second window here) and choose which loudness parameters are shown (in this example, just the blue momentary and green integrated loudness curves are displayed).

As shown in the left-hand panel of the second set of screens, these curves update in real time as the track is played. After

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Among other things, the Loudness Curve module’s settings can be customised to adjust the Duration (length of time displayed on the horizontal axis) and to show a target Reference Integrated loudness level.

an initial playback, the integrated curve will stabilise and give a reasonable sense of the overall loudness. You can then dip back into the Settings panel and set the Reference Integrated value to 10LU below the observed integrated loudness value.

In this example, the integrated loudness is about -18LUFS, so I set the Reference Integrated value (it appears as a horizontal pale blue line) to -28LUFS. This isn’t what this parameter is really intended for (I’ll come back to that later) but it does provide you with a simple visual guide to spot sections of the track where the dynamic range may be great enough for the momentary loudness to fall more than 10LU below the integrated loudness curve.

Incidentally, for any Artist and Elements users, who won’t have SuperVision, APU Software’s Loudness Meter offers a free-to-download alternative to SuperVision’s Loudness Curve module can provide very useful visual feedback on the loudness and dynamics of your audio.

SuperVision. It provides some pretty sophisticated features that include a real-time momentary loudness curve similar to that found in SuperVision’s Loudness Curve module.

Going Up This visual feedback is a useful check to know if the gate might influence how streaming loudness normalisation is applied to a track, but it’s still your decision to do something about it or not. If you do, the focus needs to be on raising the level of the quieter sections of the track while leaving the louder parts unaffected, and there are a number of approaches that might be considered. One, for example, would be parallel compression. But the Squasher plug-in, available in Cubase Pro, Artist and Elements, can apply upward compression, and this is super-easy to experiment with.

The final screenshot shows some Squasher settings that might serve as a good starting point. For simplicity, the

plug-in has been set to operate in single-band mode.

Within that band, the Down knob has been set to 0% and the downward compression Threshold (the small downward arrowhead) has been set to 0dB. These settings should ensure that no conventional (downward) compression is applied, which would otherwise affect the louder sections; these should pass through the plug-in with their dynamics intact.

In contrast, the Up knob, which controls the ratio of any upwards compression, has been set to 100%. With this set, you can then gradually raise the Up Threshold value (the small upward facing arrowhead) until you see the upward compression kick in, as indicated in real time by the pale bar at the base of the Threshold graphic.

At this stage, Squasher’s main Input and Output controls have been left at zero (no overall gain adjustment applied; I’ll come back to the Output control in a moment). In addition, the Drive control (in the Parameters panel) has been left at zero so no saturation is added, while Gate is set at its lowest available value (-60dB), effectively bypassing Squasher’s internal gate. In this case, the Attack and Release are both on fast settings but, as with conventional downward compression, you can experiment with adjusting these based on the tempo of the track, letting your ears judge what produces the most transparent results.

Spot The Difference From this starting point, you can adjust the two main Mix controls, the Up Ratio and the Up Threshold values to vary just how much (or how little) upward compression is applied. Again, your ears are the key here, both in terms of just how much louder the quieter sections of your track become but also in terms of how transparently any loudness change can be achieved.

However, SuperVision can also be a guide: place a second instance of the plug-in immediately after Squasher and, if you use the same display settings as in the pre-Squasher instance, you should easily be able to see the difference in the loudness curves pre and post upward compression.

The right-hand side of the first screen shows an example using exactly this configuration. While the momentary loudness curves are the same in the louder parts of the audio, Squasher’s upward compression has bought up the loudness during the quieter sections.

While I’ve perhaps overcooked the processing here for illustrative purposes (and could lower some combination of the Mix, Up Threshold and Up Ratio controls for a more subtle effect), Squasher’s potential to manipulate the dynamics of your audio, increasing the loudness of quieter sections while leaving louder sections untouched, is easy to see.

Target Practice?

I mentioned above that I’d return to the Loudness Module’s Reference Integrated setting and Squasher’s Output control. Almost all streaming platforms have now published the integrated loudness reference levels used by their normalisation algorithms. They’re not all the same, but they are similar and, as an example,

Squasher is capable of upward compression, which allows you to manipulate the loudness dynamics of your track in some interesting ways.

Spotify uses -14LUFS-I as its default — on playback, songs will be turned either up (as much as peak levels allow; there’s no limiting applied in the default mode), or down to ensure they’re played back at -14LUFS-I. This is an automatic process, so it’s not critical that the integrated loudness of your mastered track is -14LUFS — the whole point is that loudness normalisation plays it back at that level for you. But perhaps you’re working for a client who has specified delivery of files at a certain LUFS-I. Whatever the reason, if you feel the urge to nudge your track towards a target level, you can. First, use the Reference Integrated setting for its intended purpose, and set it to the required target value (-14LUFS for Spotify, for example). Then you can simply use the visual reference this provides and adjust Squasher’s Output knob up or down until your track’s integrated loudness curve sits on or around the Reference Integrated target line. For the example used here, it would

require a setting of about +4dB. Providing this doesn’t also raise any peaks close to clipping, on playback your final mastered track might give Spotify’s loudness normalisation algorithm very little to do.

And Finally...

There are a few caveats to keep in mind.

First, it’s not all about the numbers: in raising the loudness of quieter sections of your audio, you’ll be changing the track’s overall dynamics. It’s always worth making loudness‑matched comparisons, so that you can judge whether the processed version sounds better or worse than the original!

Second, you’ll probably also be raising the level of low-level noise in the source material. This may not be an issue if the original recordings were of decent quality, but do make the effort to listen out for unwanted side-effects. If noise becomes an issue, you might be able to counter that by de-noising your sound sources, whether with a spectral noise removal tool or strategic use of noise gates or expanders.

Finally, not all mixes and mastered tracks can benefit from this sort of loudness dynamics adjustment. If your initial SuperVision visuals don’t show the momentary loudness dropping 10LU below the integrated loudness curve for significant portions of time, further loudness manipulation might well be unnecessary or unfruitful. If it’s not broke, don’t fix it!

www.soundonsound.com / September 2024

141

Studio One

TECHNIQUE

We talk you through the basics of side-chaining in Studio One. ROBIN VINCENT

L

ast month, we took a look at the various compression plug-ins that ship with Studio One. One feature that I glossed over was their ability to use an external key input. This allows the ‘side-chain’ signal feeding the detector section of the compressor to be different from the audio signal the compressor is processing. The example you’ll probably find most familiar is using a kick drum to pull down the level of a big synth sound so that the sound of the kick comes powerfully through the mix. It can subtly emphasise the kick in the mix, or it can be pushed harder, making the track throb and pulse.

So, this month, we’re going to look at how we do that in the context of the Studio One plug-ins. We’re going to use the example of the kick drum to see if we can rescue it from being lost in the mush.

You can of course, use this to highlight or emphasise any instrument you want, or use it purely for the pumping effect it can bring to your music, but let’s keep things simple.

Setting Up The Side-chain There may be lots of tracks that are covering up the action of your kick drum, but we’re going to focus on a single track of synths and a kick drum so that we can get a handle on what’s going on. So, open a nice fresh song and, delving into the Studio One Loop Library, head over to Pads / Loops, grab the file called ‘154_ Disco_Loop_Synth Pad 1_Cmaj.audioloop’ and drop it onto a new track.

Next, poke around in Loops / Drums / Kick and you should find a sample called ‘Kick 90908.wav’, which is exactly the sort of thing we need. Drag that into another track. Now, this is just a single hit, so we’re going to need to populate the track with copies that run for the length of the pad.

Select the single clip and press D (for Duplicate) as many times as it takes to fill the track out with kicks.

From the Effects Browser, drag the PreSonus Compressor plug-in onto the synth pad track. Above the compressor controls, next to the preset menu, is the Sidechain button. Click to enable it and then click the arrow to the right. This brings up a list of possible side-chain

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sources. In a big project, you’d have a long list of every track’s output, but here we just have the kick drum track.

We have a choice between Send and Output, which largely depends on how we want to use the side-chain source. The Send option works like an effects send, routing the signal to the compressor while the kick track continues to operate as normal.

This can be pre‑ or post-fader, so that the effect can be carried through regardless of whether the kick fader is turned up or not. With the Output option, the destination of the kick track becomes the compressor, so you’ll no longer hear the kick at all.

This can be useful when you’re triggering compression from a sound source that you don’t actually want to hear, such as a click track. It also doesn’t have to be a single source; you can take sounds from anywhere and use them together.

Pump Up The Jam With our routing established, we are ready to start tugging on that side-chain. Set Studio One playing back around the synth pad loop, and we’ll play with some settings in the compressor.

Pull down on the threshold and you’ll see the orange compression indicator appear on each thud of the kick. Pull it down to about -30dB, and you’ll experience the pumping effect that’s very common in EDM, with the kick drum thumping clearly through the synth. But there’s more we can do here.

You can make the effect more extreme by increasing the ratio for some serious

Here, a Compressor plug-in on the synth pad track is being triggered by audio sent from a kick drum track.

throbbing. For a more subtle effect, push the threshold back up to around -20dB, and reduce the attack and release times.

The slot for the kick is now a lot more surgical and less like a throb.

You may want to be able to hear the effect without the kick. If you mute the kick drum track, you’ll also remove it from the compressor side-chain, so that’s no good. Instead, set the side-chain source to pre-fader. Now you can turn the kick down in the mixer, but it’ll still hit the compressor.

The level of the kick also has an impact on the side-chain. If you set it back to post-fader and vary the volume, you’ll see that as you turn the kick fader up, the amount of compression increases. That’s because the threshold is set against the level of the incoming kick, not that of the synth pad. Again, setting the side-chain to pre-fader will keep the compression consistent regardless of how you mix the level of the kick.

Side-chain Filtering The Studio One Compressor has a filter that can be used to weed out unwanted frequencies from the side-chain signal. Why would you do that? Well, we don’t always have a nice clean kick track to use as our sound source. You could, of course, create one using a software instrument, or isolate a kick drum sample and route it directly to the compressor so it’s doing the work and not being heard. You could even extract the

groove from the drums you do want to hear in order to get the side-chained kick drum exactly where you want it — but that sounds like a load of work to me! Instead, we’re going to use the side-chain filter to remove everything other than the kick so that our compression works correctly.

Let’s try this out with the ‘fh_drm120_ chugger_stp.audioloop’ file from the same Kick library. Drag it into another track, mute the original kick track, and select the send for this track as the side-chain source. This loop contains a bunch of tambourine and sticks mixed in with the kick, so when you run it, you have a lot of compression being triggered that’s in the spaces between the kicks. To clean this up, enable the Sidechain Filter and pull the high cut down until it’s removed all the higher‑frequency percussion, just leaving the kick.

Even when you’re not using an external key input as the side-chain source, you can still use the filter to modify the signal feeding the side-chain. On a vocal, for example, you could filter out everything below 5kHz or so such that compression happens only when a high-frequency transient happens.

This turns the compressor into an effective de-esser — a useful alternative application of the same process.

In The Mix If you are using this effect in a proper scenario with lots of different tracks, you can very easily add the same side-chain effect to some or all of them. In the console, simply

Our ducking delay setup. A Gate plug-in has been inserted after Analog Delay on an FX Bus, with a vocal routed to its external side-chain input.

Now, when the vocalist is singing, the delays are gated out, but when the singing stops the gate opens again.

drag the compressor onto other tracks that you want to manipulate. Studio One carries all the side-chain routing across, and it’s all there set up for you.

We can see this in action by dragging in the ‘154_Disco_loop_Synth bass 3_Cmaj. audioloop’ from the Loops / Bass / Loop folder into another new track. Copy the compressor across, and it will instantly work. If you check the sends for the kick drum, you’ll see individual sends are set up for each compressor, letting you fine-tune how much is being sent to each one.

Beyond The Kick It doesn’t have to be all about getting the kick drum to poke through. You can use the exact same process to jazz up the rhythm of a piano or guitar with a hi-hat track.

And it doesn’t have to be a compressor; you can use the Gate plug-in to give a very clean and complete drop-off, or you could use the Multiband Dynamics compressor to focus on one particular frequency range of the track.

Side-chaining is great for vocals.

If there are a lot of instruments with similar frequency content, then the vocals can get a bit The side-chain filter in Compressor lets you specify a narrow band of frequencies to trigger the compression, and it can be used on an external side-chain trigger, or on the internal trigger signal, for example to set up de-essing.

lost. Using the vocal as the side‑chain source will pull the guilty tracks down whenever the vocalist is singing. You could also apply this to spoken word over a soundtrack or demo’ing a product.

This is usually known as ducking.

Ducking Delay Another cool trick is to side-chain a delay.

When you add a delay effect to a vocal, the delay can start to drown out the original vocal by repeating on top of the performance. You can sort this out by dropping in a Gate plug-in, setting it to ‘Ducker’ mode and side-chaining the vocals back into it. Let’s do that.

From Loops / Vocal / Electronic, drag in ‘Dubstep Vox 05 150bpm C.audioloop’ and bring up your tempo. If you put a copy at bar 1 and bar 5, that leaves plenty of room for the delay. Drag the Analog Delay plug-in into the Console to create an FX Bus. Add a send from the vocal to Analog Delay. Set the Analog Delay time to 1/2, and you’ll hear the first reflection coming back rudely over the end of the vocal. Drop a Gate plug-in onto the FX Bus after Analog Delay. Turn Ducker mode on and Enable the side-chain on the gate, selecting the dubstep vocal as the source. Now, whenever the vocals are playing, the gate closes — the echoes ‘duck’ out of the way, so we don’t hear them until the vocal stops and the gate opens again.

You can fiddle around with the attack and release settings to tidy it up a bit, but it’s brilliantly effective.

Side-chaining is one of those ‘secret sauce’ things to use while mixing that can give your tracks a sense of space and dynamics. But we all need to throb a kick drum through a pad from time to time, and that’s a great place to start.

www.soundonsound.com / September 2024

143

Q

Can I send mic signals over Cat5 cable for live sound?

I see four-channel XLR breakout boxes connected by Ethernet cable, such as Thomann’s Cat Snakes, becoming increasingly popular. As a decidedly low-tech option for live sound, four sets would route 16 channels of audio over four runs of cheap Cat5 cable, with the huge advantage that if somebody chews up one of the cables while rigging, or whatever, I could just whip out a reel of Cat5, cut to length, quickly crimp connectors on it, and be good to go.

It’s a much cheaper option than a Dante stagebox — but does this really work, at least for fairly short runs, at mic level?

SOS Forum post Hugh Robjohns, Technical Editor Does it work at mic level? Yes — but there are very important caveats. First, though, although Ethernet cable is used, it’s not carrying digital data, so it’s not comparable with Dante or other Ethernet-based interfaces. The passive Cat5 snake systems, which are available from several manufacturers, simply pass four balanced analogue signals over four twisted pairs of wires; the Ethernet cable is just a convenient and cost-effective product.

I mentioned caveats. As a bare minimum, this system needs screened cable with screened connectors — so definitely not the cheapest Cat5 UTP (Unscreened Twisted Pair) cable. The screening prevents interference getting into the cable, and is vital for allowing phantom power to work.

No screen means no phantom power, as some disappointed users have discovered the hard way! In practical terms, Cat6 FTP or STP (Foil Screened or Shielded Twisted Pair) cable is specified, usually with robust EtherCon connectors (an RJ45 plug within an XLR-like shell).

Ethernet cable is designed to pass symmetrically balanced digital signals of identical strength, and in that scenario there’s negligible crosstalk between the four channels (and negligible interference).

Hence the success of UTP cable generally but, with faster connection speeds, an overall screen helps to reduce interference (outwards, as well as inwards) so it’s more commonly employed today, hence the wide availability of inexpensive STP cable.

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September 2024 / www.soundonsound.com

Analogue multicore cables used in live sound, though, must convey signals at vastly different levels, and not always symmetrically balanced ones. This increases the risk of crosstalk dramatically, so in mitigation analogue multicores typically employ individual shields around each pair of signal wires, rather than an overall screen. Usefully, Cat6 Ethernet cables are also available with individually screened pairs. These are usually denoted as S/FTP, F/FTP or sometimes SSTP (Shielded Foiled/ Screened Twisted pair).

I didn’t have any Cat6 STP and SSTP cables to hand, but did have two-channel individually screened multicore to simulate SSTP cable, and star-quad cable (which can be wired as two unscreened pairs with an overall screen) to simulate STP cable. So I tested the crosstalk issues and generated a plot using my Audio Precision test system.

The green line (bottom) is the inter-channel crosstalk over 25m of two-channel multicore cable (Van Damme Studio Grade Blue Series) with individually screened pairs. The test signal was symmetrically balanced at +4dBu, sweeping from 20Hz to 20kHz. The marginally higher blue line is the same symmetrically balanced signal running down 25m of Canford Audio’s star-quad cable, with two unscreened pairs tightly twisted together within an overall screen. This demonstrates that with symmetrically balanced signals at similar levels, crosstalk is negligible regardless of whether the pairs are individually screened or not. The worst-case crosstalk is -90dB at 20kHz, while at mid-frequencies it’s around -120dB. The gentle rise in crosstalk level with frequency is normal, and due to capacitance between the wires.

Symmetrically balanced signals largely self-cancel their magnetic and electrostatic

fields, thus hugely reducing induction into adjacent wires, minimising crosstalk.

In audio applications, symmetrically balanced signals aren’t guaranteed, and sometimes we work with unbalanced signals. ‘Impedance-balanced’ outputs are common on budget mixers and other equipment, including many transformerless capacitor mics from respected manufacturers. The impedance-balanced format conveys the signal over one wire in the pair, with the other grounded, much like an unbalanced signal — as there’s no self-cancellation of the magnetic/ electrostatic field, there’s more crosstalk.

The khaki line across the middle of the chart shows the crosstalk arising from an unbalanced source feeding a balanced input over the individually screened multicore cable. The crosstalk is about 40dB worse through the midrange than for the symmetrically balanced test, but, at around -70dB, it’s still practically inaudible. The red line (top) is the same unbalanced source feeding the overall-screened star-quad cable and, as predicted, crosstalk is 20dB worse than using individually screened pairs.

At -50dB at 2kHz it’s audible too.

Although I’ve not tested Cat6 STP/SSTP cables, the constructions of the multicore and star-quad cables are similar, and it’s safe to assume that equivalent performance differences would be obtained when conveying impedance-balanced or unbalanced signals. So while STP (overall shielded) cables are fine where symmetrically balanced signals can be guaranteed, individually screened (SSTP) cable has very significant advantages if there is any possibility of conveying impedance-balanced or unbalanced signals.

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of digital control and recall that modern engineers

MINIMAL SYNTH SETUPS DAV E G A L E

D

o you ever yearn for a simpler existence?

A move to the country, or maybe a television that offers fewer channels? Putting the latter point to one side, there is a reason why so many creative types opt to shut themselves away in out-of-town studios or workspaces. It has everything to do with the way that individual creativity works, and removing distraction.

I had a moment of revelation a few years ago while composing the music for a BBC TV documentary series. This was a time when samples, plug-ins and software instruments were changing our working practices, in the extreme.

The director suggested that they wanted the score to sound more like the output of a teenager in a bedroom,

146

with a guitar, a bunch of pedals and 9V batteries.

This was the polar opposite of the clean signal ethos being promoted at that time.

Not being a guitarist (I never understood why anyone would play an instrument that hurts your fingers!) I had to resort to playing around with a synth, pedals and those ubiquitous 9V batteries, but the general ethos remained.

Hums and crackles infiltrated my production, and while I would have previously done my utmost to eliminate these annoying noises, they seemed to go down well in the edit suite.

This learning process — which, to be fair, was generated by a non-musical TV director who yearned to play the guitar as a teenager — sparked a wider change to my working practice, which I still try to use as often as possible.

September 2024 / www.soundonsound.com

A good synth-friend of mine (we’ve all got them) often recalls how he was far more creative when all he had was a four-track Portastudio and a Roland SH-101. There’s a lot of truth in that! Working with minimal equipment often forces you to be more creative, so I decided that it’s never too late to try. Before you ask, no, I didn’t buy a four-track Portastudio (despite them being back in fashion, thanks to artists such as Trent Reznor) but I did decide to produce a library album which was exclusively based around a Moog Voyager. It was nothing short of a revelation; challenge after challenge presented itself, but nothing seemed to stand between this fine, singular synthesizer and creativity.

I purposely left the Voyager in an unprogrammed state, forcing me to create sounds as I needed them. Everything was

created from scratch, from the drums and hats, through to the multiple tracking of pad-like sounds, building chords from a monophonic source. It was some of the most fun I had had in years, and resulted in an album which you may well have heard on various sci-fi and tech-based TV shows, as is the library album way!

If there were a moral to this synth tale, which is apologetically garnered with midlife reminiscence, it would be that honing your craft with less may create better musical outcomes.

I continue to adopt a similar philosophy, by moving hardware around my studio on a project-by-project basis, in the synth equivalent of ‘I’m the King of the Castle.’ It really does help to maintain creativity and production motivation. I’ll sign off now, as I think my 9V batteries need replacing!

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