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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
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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Mix Atmos With Ease
Introducing the Symphony Mk II 16x16 Special Edition with Control Remote
Our Highest Performing System... Ever. Used by Alan Meyerson, Eddie Kramer, Shawn Everett & more. * Remote sold separately
IN THIS ISSUE
FEATURES
32 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?
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’. 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.
62 Ziggy Stardust In Atmos
Ken Scott and Emre Ramazanoglu explain how they reinvented
David Bowie’s classic album in three dimensions.
80 How Virtual Instruments Work
We cut through the jargon to explain how software instruments achieve their incredible realism.
88 Talkback: Ramera Abraham
Producer and engineer Ramera Abraham on dissecting Beyoncé and why she’d love to sit in on an Ariana Grande session.
92 Modular
Cycle Instruments’ Josh Wilkinson talks about modules and reinventing the sequencer.
104 Inside Track: Julian Bunetta
Taking the decision to follow his instincts has elevated Julian Bunetta to the top rank of songwriters and producers.
120 Spotlight: Kick Drum Microphones
We’re spoilt for choice when it comes to dedicated kick drum mics. We check out some of the best. 144
Your studio and recording questions answered.
Dave Gale explains why he craves the simple life.
116 ORCHESTRAL TOOLS BENJAMIN WALLFISCH STRINGS
ON TEST
Wallfisch Strings
Origin Effects Cali76 FET
Cali76 Bass
Processors
The Jazz Sessions SDX Superior Drummer 3 Expansion
Audio
Channel Strip Plug-in
MIR Pro 3D
Vienna Power House Immersive Audio Reverb
Waldorf Iridium Core
Zoom H4 essential
Recorder
Zynaptic Morph3 Pro Audio Morphing Plug-in
130 Digital Performer
Logic
Pro Tools
Cubase
Studio One
McDSP APB Tape
Tape Emulation Plug-in For APB
McDSP’s analogue plug-in platform can now replicate the complex dynamic behaviour of tape.
McDSP’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.
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!
“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!”
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
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
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.
£ Free to APB owners. W www.sxpro.co.uk W www.mcdsp.com
FOCAL I SHAPE TWIN
Shape Twin is the largest and most versatile member of Focal’s Shape series. Featuring dual passive radiators, twin flax woofers and Focal’s latest ‘M’ shape aluminium/magnesium inverted dome tweeter, Shape Twin is designed to perform in a wide range of monitoring environments.
Beyond the curve!
AIXDSP DrumEQ
Equaliser & Dynamics Plug-in
Could this clever, labour-saving plug-in change the way you mix drums?
MATT HOUGHTON
AIXDSP 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
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
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
Originally 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.
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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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,
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
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.
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
Well 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
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
“I
Neil Rogers, Sound On Sound
Soundtheory Kraftur
Want to add power, energy and excitement to a source? Perhaps Kraftur can help...
PAUL WHITE
Soundtheory’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,
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
Fireface Series
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Zynaptiq Morph 3 Pro
Audio Morphing Plug-in
Ever wanted to blend the characteristics of two different sounds? Morph Pro makes that easy.
MATT HOUGHTON
Available 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
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
The Neumann U47 is legendary for a reason — as this homage makes clear.
NEIL ROGERS
Since 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
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 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
Atmos just got
Immersive Audio Interface & Monitor Controller
Meet ORIA, the world’s first audio interface and monitor controller designed specifically with immersive audio in mind. Built for creating immersive audio mixes for formats such as Dolby Atmos, ORIA lets you calibrate, control and monitor multi-channel speaker arrays from stereo up to 9.1.6 and everything in between. Perfect for music, film, TV, game and VR production.
JOHN WALDEN
While 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
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
Waldorf Iridium Core
Synthesizer
Waldorf’s all-encompassing Iridium engine is now available in its smallest form yet.
RORY DOW
As 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
(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
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).
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.
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. £ £1349 including VAT. W www.handinhand.uk.net W www.waldorfmusic.com
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.
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.
It’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
SAM INGLIS
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
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.
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
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?
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.
A single RPS power supply can power up to eight Box Line 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.
Moby’s decision to license tracks from Play for use in commercials helped the album sell better, and made him a lot of money.
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
The 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
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:
• Mechanical 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
Photo: Bill Ebbesen
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).
Be careful what you sign. Licence agreements are legal documents!
SCAN ME
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.
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,
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:
• Can 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
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 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. Generally, those who have a ‘sound’ risk ruling themselves out from a lot
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,
It’s vital to audition your work on smartphone speakers, since that’s how much of your audience will hear it.
If you thought you knew what a polysynth was capable of - think again. If you thought you knew what the interplay between human touch and analog synthesis could sound like - think again.
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Zoom H4essential
Portable Recorder
“Just press record,” they say. Can it really be that easy?
CHRIS TIMSON
For 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
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.
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.
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.
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.
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
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.
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).
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!).
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.
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
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
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).
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
The SlatFus or Modern Elegance
Acous�c Panels & Bass Traps
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.
David Lord • XTC
‘All You Pretty Girls’
JOE MATERA
Respected 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
“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
Hear The Sound
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.”
Moog Muse
Polyphonic Synthesizer
A Moog polysynth is always a big event, but is the Muse the inspiration you’ve been waiting for?
GORDON REID
There 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.
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
“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.”
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
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
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
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
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.
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,
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
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 its output on a 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
engineers again to ask whether I was being a numpty. I wasn’t. 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 section, nor can you use the Diffusion Delay to spread the results across the stereo field, but I found this to be a more than acceptable compromise, especially since I was able to apply separate external effects to each output.
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
I also layered the Timbres to obtain some composite sounds. These often sounded excellent, but bear in mind that this reduces the polyphony to just four voices.
It was now time to take a deep breath before diving into the complexities of the arpeggiator and sequencer. To be honest, it would have taken far longer than a pre-release review to explore the possibilities that they offer. Given all of the Muse’s connectivity options (and, in particular, its use of 3.5mm connectors) I suspect that Moog are encouraging us to see it as the centrepiece of an experimental electronic music studio and, if used in this way, I think that many of their features will make a great deal of sense. However, it was while I was combining the arpeggiator and the Chord mode that I discovered
another serious shortcoming. I spent half an evening creating a complex setup with chords distributed right across the keyboard and then looked for a way to save this. I couldn’t find one, so I hoped that the map would be retained as a global setup. But when I power cycled, all of my work was lost. This renders the chord/key capability almost useless, so I again contacted the chaps at Moog. To my great relief they told me that, “The chord map will be non-volatile with eight available maps you can store, but that’s not currently implemented. We’re working on it.” Phew!
Next, I hooked the Muse up to my MIDI studio to see whether it played nicely with the other children.
It did. Its MIDI specification includes an extensive range of MIDI CCs that allow you to control other equipment as well as automate the Muse itself, and there’s even a Multi mode that allows you to control the two Timbres in each patch using separate MIDI channels. You can also choose different MIDI output channels for the keyboard and MIDI CCs (on the one hand)
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.
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Enjoy a Summer of Creativity
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The MindsetTape
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.
Arecent 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
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
NEIL ROGERS
Steve Osborne.
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
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.
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.
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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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
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!
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
(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?
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
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’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
