THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZ INE 19855 — 2022

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MUSIC PRODUCTION TECHNIQUES / INDEPENDENT IN

IN-DEP
DEPTH
P TH PPRODUCT TESTS / ENGINEER & PRODUCER INTERVIEWS / LIVE SOUND

MAKE THE RIGHT MIX MOVES

Idles
: ‘His kes hits!’”

TWICE: crafting a K-pop hit

“It’s definitely different mixing nine singers rather than just one!” PMC6 MONITORS
W RTH £5010
WO
Which gear got your votes?
www.soundonsound.com

REVIEWS: EXPRESSIVE E / ZAOR / API

REVIEWS AP SELECT / ROYER / ROLAND / STEINBERG / JZ / KORG / EARTHWORKS
February 2022 £6.99
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 L E A DE R

THE PEOPLE’S CHOICE

T
here’s been a lot of controversy about
awards lately. Stars such as The Weeknd
have chosen to boycott the Grammys, while
the Golden Globes have been cancelled altogether.
Other gongs such as the Oscars have faced
mounting criticism, with charges of corruption, bias
and lack of diversity being fired from all directions.
The roots of these problems lie with opaque
judging criteria. When nominations and winners
are chosen by shadowy groups of industry
insiders, it’s hard to convince the world that these
decisions are always made purely on merit.
The annual SOS Awards don’t have the same
illustrious history as the Oscars. Winners don’t get
to thank their mum and dad at a lavish ceremony,
hosted by whatever edgy comedian is in vogue
that week. In fact, most of the hardware designers
and software coders I know would be horrified at
the thought of getting up on stage at all. But when
you talk to manufacturers and developers whose
work has been honoured, you’re in no doubt as to
how much importance they attach to winning.
Why? Because the SOS Awards aren’t decided
by obscure committees meeting behind closed
doors, or influenced by lobbying from smarmy PR
executives. They are voted on by the people who
actually buy the products that are nominated: you,
the readers.
Now you might argue that a reader vote isn’t
an ideal system for deciding awards either. Yes, it
has the merit of being transparent and democratic,
Sam Inglis
Editor In Chief
but at the same time, no-one is likely to have tried
every single product nominated, so how can it be
a truly informed choice?
Well, if there’s anyone out there who really
does have in-depth, real-world experience of
several hundred new products, I’d hope they
would be writing for SOS as well as voting in our “Your votes really
awards! But with over 7000 people taking part
this year, we can be confident that the winners
do matter. In fact,
really deserve their recognition. Even when they’re the only
awards are carried off by high-end products that
are aspirational rather than universal, it still shows thing that matters.”
the manufacturer is doing something right. The
fact that few of us can own an Arturia Polybrute
doesn’t mean people were wrong to vote for it;
it means that Arturia have succeeded in creating
a truly desirable instrument.
So, thank you for taking the time to log on to
the SOS website and put your crosses in boxes.
Whether you have used this equipment every day
in a professional context, or just looked at it from
afar with lustful thoughts, your votes really do
matter. In fact, they’re the only thing that matters.
And if you want to know how much they matter,
ask the manufacturers who won.

ADMI NI S TRATI ON EDI T OR IAL ADVER TISING

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ISSN 0951-6816
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expressed are those of the contributors and not necessarily

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4 February 2022 / w w w. s o u n d o n s o u n d . c o m
 Introducing Duet 3

Legendary Sound Quality, Total Portability & Hardware DSP

All Come Together in a Beautiful New Design

Sounding Amazing Never Looked So Good

In 2007, the original Apogee Duet shattered the expectations of
what a home studio interface could be. The all-new Duet 3 brings
next-generation Apogee performance and features to a beautiful,
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(&6&KDQQHO6WULSWXQHGE\%RE&OHDUPRXQWDLQ studio or on the go.

www.apogeedigital.com

TECHNOLOGY
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 40 IDLES

INTHISISSUE www.soundonsound.com

February 2022 / issue 4 / volume 37

WIN
FEATURES
26 How I Got That Sound PMC6
Producer Howard Benson explains how he crafted the drum
sound on POD’s iconic 2001 track ‘Youth Of The Nation’.
MONITORS
30 Vocal Mix Preparation
With your lead vocal performance comped and tuned, and its
timing tightened, is there anything else to do before you
start mixing?

40 Idles: Recording Crawler

For their latest album, Idles guitarist Mark Bowen adopted the WORTH £5010
role of producer for the first time.
PAGE 110

66 Talkback
Hinako Omori has lent her synth and keyboard skills to some of
122 Modular
the biggest artists around. She talks to us ahead of the release of
her solo debut. We share some tips on getting the most out of your VCAs, and
catch up with Expert Sleepers founder Andrew Ostler.

72 EQ: What Do All Those Knobs Do?

C O V E R

140 Golden Gear: The KEF KM1

The right EQ settings can make a mix — and the wrong ones can Main studio monitors these days are packed full of technology —
make a mess. Our in-depth guide will help you get more from but that wasn’t always the case, and it took an innovative British
this all-important tool. hi-fi brand to drag mains out of the dark ages.

92 Inside Track: Taesub Lee 160 Q&A

When you’re polishing a nine-piece vocal band like TWICE to
K-pop perfection, organisation is essential!
112 SOS Awards: The Results
Find out what gear got your votes!
Your studio and recording questions answered.
162 Why I Love... Arranging
SOS reader Julian Schmauch explores the power of
a good arrangement.
 54 ZAOR MIZA Z FLEX & GRIPRACK MKII

ON TEST
8 Sennheiser HD 400 Pro 68 Royer Labs dBooster2 128 Korg Collection 3
Open-backed Headphones In-line Microphone & Instrument Software Synthesizers
Preamplifier
10 Goodhertz Tupe 136 Apple Logic Pro X 10.7
Saturation Plug-in 80 Eventide SplitEQ Music Production Software
Transient-shaping Equaliser
12 Toontrack Upright EBX 144 EHX Nano Deluxe Memory Man
EZbass Expansion
84 API Select T25 Delay Pedal
Dual-channel Valve FET
14 PreSonus R80 V2 Compressor-Limiter 144 Audio Damage Other
Desert Cities
Active Nearfield Monitors 90 JZ Microphones Vintage 12 Delay Plug-in
Large-diaphragm Cardioid
18 Milab VIP-60 Microphone 145 Z-Plane Tonic
Multi-pattern Capacitor Microphone Music Theory Assistance Plug-in
98 Steinberg Wavelab Pro 11
22 Boss Eurus GS-1 Audio Editing Software 146 Sample Libraries
Guitar With SY Synth Engine FrozenPlain Slow
102 IK Multimedia AmpliTube X-Space IK Multimedia The Resonator
36 Sound Particles Digital Reverb Pedal
Space Controller 106 Roland SP-404 MkII
Spitfire Audio Abbey Road
One Instruments
Smartphone-activated Surround
Sampler UJAM Virtual Drummer BRUTE
Pan Control Plug-in

Expressive E Imagine 120 Qu-Bit Cascade

WORKSHOPS
48 Eurorack Module
Software Instrument
121 Clatters Machines
54 Zaor Miza Z Flex & Griprack MkII Garden Listener
Studio Furniture Eurorack Module
148 Digital Performer
58 Blackstar Dept. 10 152 Live
124 ErgoLab Stealth
Overdrive & Distortion Pedals Studio Chair 154 Pro Tools
62 Sonnect Sound Bullet 126 Earthworks ETHOS 156 Cubase
Portable Audio Test Device Capacitor Vocal Microphone 158 Studio One
 ON TE S T
Sennheiser HD 400 Pro
SAM INGLIS
G Sennheiser’s new studio headphones offer some serious
erman audio titans Sennheiser
operate across multiple markets,
and although they manufacture
many headphone models that are widely
used in studios, not all of these are officially
‘pro audio’ products. In particular, high-end
open-backed designs such as the HD 600,
HD 650 and HD 800 are categorised as
‘audiophile’ headphones, and marketed
mainly at hi-fi enthusiasts. So, although
the new HD 400 Pros are by no means
Sennheiser’s most expensive open-backed
headphones, and are more affordable than
all the aforementioned models, they could
nevertheless be seen as the new flagship of
the company’s pro audio range.
For open-backed headphones, in
fact, the HD 400 Pros are aggressively
priced, undercutting potential rivals such
as the Audio-Technica ATH-R70x, Austrian
Audio Hi-X65 and Shure SRH1440, and
going more or less directly up against the
Beyerdynamic DT900 Pro X. It’s fair to say
this price-consciousness is a little apparent
in both the packaging and the industrial
design of the HD 400s. They don’t come
with a carry case or even a bag, and the
frames and earcups are made almost
entirely from utilitarian black plastic. The
earcups themselves are very elongated, as
now seems to be common in headphone
design, and offer a relatively small degree
of movement in both the fore and aft
and the up and down planes. The plastic
construction makes them very light at
240g, and the cushions are soft, but they
nevertheless clamp quite firmly against
the sides of the head. Comfort is a very
8 February 2022 / w w w. s o u n d o n s o u n d . c o m
Open-backed Headphones
bang for your buck.
personal thing, but I would rank the HD
400s only so-so in this respect.
The HD 400s are fed audio through
a removable cable which locks securely into
the base of the left earcup. Both straight
and curly cables are supplied as standard,
and the eagle-eyed will note that the locking
connector is a TRRS mini-jack rather than
the cheaper and more widely used TRS.
This means that the left and right channels
have independent return paths, in theory
reducing crosstalk.
The HD 400s use conventional
moving-coil dynamic drivers, with a large
moulded grille allowing sound from the
back of each driver to escape into free air.
The impedance is quoted as 120Ω, which
is higher than most consumer headphones
but not so high as to make them hard
to drive from standard audio-interface
headphone sockets and the like. Sensitivity
is specified at a very decent 110dB SPL for
a 1V RMS, 1kHz tone, while total harmonic
distortion is 0.05 percent for a 1kHz tone at
90dB SPL.
HD Audio
One of the reasons that Sennheiser’s HD
650s are so widely used in mixing is that
they have a reputation for being ‘unhyped’:
they deliver a flat, neutral, uncoloured
sound that allows you to listen to music
in an analytical way. I didn’t have a pair of
650s on hand for comparison, but I was
able to compare the HD 400s directly
with open-backed phones from other
manufacturers, and came away impressed.
The Shure SRH1840s, for example, are
longstanding favourites of mine, and
retail at around twice the price of the HD
400s; yet the HD 400s come pretty close
to matching them in terms of neutrality,
and sound marginally cleaner on some
material. A gentle presence boost is audible
somewhere around the 5kHz mark, and the
bass is perhaps a touch leaner on the HD
400s, but there’s less to choose between
them than the price would suggest.
Meanwhile, Austrian Audio’s Hi-X65s
sounded positively ‘scooped’ in comparison,
with a recessed midrange and very forward
bass and treble.
In short, then, the HD 400 Pros are
exactly what a ‘pro audio’ product should
be. The overriding sense you get from them
is that the development and manufacturing
budget has been spend not on cosmetics or
accessories, but where it matters — on the
sound. Try these and you may not feel you
need to spend more.
summary
The HD 400s are neither the most
comfortable nor the most stylish pair of
open-backed headphones around, but they
are very affordable and sound great.
£ £219 including VAT.
T Sennheiser UK +44 (0)333 240 8185
W en-uk.sennheiser.com
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 ON TE S T

Goodhertz Tupe
NEIL ROGERS

S
ince I reviewed Goodhertz’ plug-in
range in 2018 (www.soundonsound.
com/reviews/goodhertz-plug-ins)
I’ve used their tools often, especially
Saturation Plug-in
when I want something ‘a bit different’. As the name implies, Tupe emulates both tape and tubes
I particularly like their Wow And Flutter — but it goes a bit further than most other saturators.
tape emulation, so was very happy
when I learned of their latest release, Flicking through the tape and tube the transients that I didn’t immediately
Tupe, which combines the much-loved choices gives you distinct tonal options, appreciate until I hit bypass.
sound of analogue tape machines with, and I found them refreshing compared Some may find the layout a bit ‘funky’
you guessed it, the much-loved sound with other tape emulation plug-ins I’ve but there are dark (pictured) and light
of vacuum tubes. As if that weren’t used. There are also presets that mimic (white background) modes, and you soon
enough, the core functions are joined by the sonic characteristics of some classic get used to it. I do like that there are
a generous set of creative processing records and you shouldn’t be tempted no fancy tape animations or worn-paint
options, including Goodhertz’ take on to dismiss this as a ‘throwaway’ feature graphics; you just move the sliders and
a vintage opto compressor. — it’s a fun and handy way to change the use your ears. In my book, a good plug-in
The GUI’s Tube
section allows you to
choose one of eight
different tube flavours,
and the manual offers
a detailed explanation
of each. Some are
modelled on simple
preamp or signal tube
circuits, others recreate
the sound of entire amp
stages with multiple
tubes and transformers,
and some don’t mimic
real-world signal chains
at all, but were created
in ‘the Goodhertz Lab’.
The Tape section treads
a similar path, offering
an initial choice of four tape types. You can personality of a source quickly while you should sound great straight off the bat.
decide whether the audio hits the Tubes get to know the features. It should have usable presets for quickly
or the Tape first, and have the option of On a mix bus, Tupe did a very changing the character of a sound, as
bypassing either section and/or the opto convincing job of mimicking the sonic well as extra features that allow you to dig
compressor. As well as a master Drive character of an expensive hardware deeper as you get to know it. Tupe ticks
control, there are slider controls that allow tube processor that I had in for review all those boxes: this is digital doing what
you to crank the tape or tube sections at the time, and on a drum bus it worked digital does best, whilst offering some of
past their Normal (100 percent) setting superbly for ‘pulling in the edges’ of the most authentic analogue sounds I’ve
up to a maximum 200 percent, for more the cymbals and generally creating heard to date. From creating the effect of
saturation. Bias and Noise sliders allow a unified, cohesive drum sound. I also a cranked guitar amp to subtle mastering
further character tweaking. had great success creating a deliberately duties, it does the job of pretty much
As with most Goodhertz plug-ins, ‘blown-out’ bass guitar part and some every saturator in my plug-in folder and
an additional section accesses more vintage-style vocal effects, with a little more besides. In fact, the only downside
advanced features. Emphasis controls help from those presets. was having to decide what I liked most,
determine which frequencies are most The filters can have a huge effect on given the sheer number of options!
affected by the saturation, while a more the tone and removing large chunks of
traditional EQ offers bell and tilt filters. the frequency range produced particularly summary
Lastly, a High Quality mode improves the dramatic results on a programmed A very usable tape-plus-tube emulation that’s
sound at the cost of more CPU cycles. drum part. The compressor is styled easy to use straight off the bat but also allows
on a famous ’60s optical device and if you to dig that bit deeper when you need to.
On Test you use this with care you’ll find that
I used Tupe on a number of mixes and it’s a worthwhile addition. On drums £ $129
liked what it did in various scenarios. especially it does interesting things to W https://goodhertz.co

10 February 2022 / w w w. s o u n d o n s o u n d . c o m
HEAVYWEIGHT GEAR.
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 ON TE S T
Toontrack
UprightEBX
EZbass Expansion
Toontrack add a fine and upstanding
acoustic expansion for EZbass.
JOHN WALDEN
S
ince launching EZbass, Toontrack have not held
back in terms of add-on content. The subject
of this review is, however, something a little
different and, I suspect, something many EZbass users
have been hoping for: the Upright EBX. The ‘upright’
bass is, of course, a core element of the orchestral string
section and is therefore featured in many orchestral
sample libraries. However, Toontrack have placed the
focus here on a plucked approach as suits jazz, pop and
rock contexts rather than an orchestral ‘bowed’ one.
Double Up
A high-end upright bass (apparently a Corsini Elite
made by Calin Wultur) was meticulously sampled for the
EZbass engine with an acoustic/modern (as opposed
to classical) performance role in mind. A combination
of on-board piezo and five different microphones was
used to capture the full dynamic and frequency range
of the instrument. Each of the presets offers you a blend
of some combination of these sources, whether that’s
the piezo and a mic or two different mics. And, as with
12 February 2022 / w w w. s o u n d o n s o u n d . c o m
other EZbass instruments, each preset
also includes a specific selection of further
controls to shape your sound as well as
the option to blend in a sub-bass for extra
bottom-end goodness.
That said, I didn’t find myself really
needing too much additional sub-bass; the
Upright EBX doesn’t disappoint in that regard.
Indeed, across the range of presets, you can
dial in an excellent range of tonal characters,
from warm, deep and solid, through to lighter,
brighter and more agile. The acoustic origins
of the source shine through in all cases,
though, and you get a genuine flavour of
the acoustic upright character. Whether it’s
a traditional jazz trio style sound, or a more
contemporary indie rock, bluegrass or folk
feel, Upright EBX undoubtedly has the tone.
Indeed, unless you have your own upright
bass, a classy selection of mics, and an
acoustic space to do it justice, I expect you
would be hard-pressed to match the quality
of the sound.
Of course, while the sound is one piece of
the jigsaw, there is also the playing required.
Like the other EBX libraries, Upright ships
with a goodly collection of MIDI groove
content. This is organised into midtempo,
uptempo and 12-bar categories, includes both
straight and swung grooves and options in
3/4 and 4/4. The supplied grooves make it
easy to see (hear!) just how convincing the
Upright EBX can be within a mix. Toontrack
also have steadily expanding MIDI groove
packs for EZbass and, in testing, patterns
from the Jazz, Latin Jazz,
Singer Songwriter and Acoustic
Songwriter packs also worked
well with Upright EBX. For those
that don’t play a real upright
(that’s me then), these groove
expansions are an important
element of the overall EZbass
experience, so I do hope
Toontrack keep them coming.
Conclusion
If you need the sound of an
upright bass in your productions,
the combination of EZbass
and the Upright EBX is a very
attractive option. It sounds
fabulous and, while a virtuoso
player would undoubtedly give
you those extra human details
(and perhaps cost you something
similar for a single day of session
work), EZbass Upright EBX
delivers a quality, characterful,
performance. A joy to use.
summary
The Upright EBX for EZbass
delivers the essential character
of an upright bass with
a minimum of fuss.
£ £59 including VAT.
W www.timespace.com
W www.toontrack.com
_PolyBrute
Analog evolves. Introducing the free 2.0
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 ON TE S T

PreSonus R80 V2 Active Nearfield Monitors

PreSonus have updated their ribbon-tweetered R series
with a set of tonal adjustment options.

PAUL WHITE
as a well-designed AMT is capable of its directivity so as to produce a wide
reproducing a clean and detailed high but shallow pattern. This allows a wide

P
reSonus build a wide range of end, partly because it doesn’t suffer sweet spot while reducing the effect of
studio monitors, including the from resonances to the extent that desk and wall reflections, which should
affordable Eris range, but the a conventional tweeter does. translate into better sound quality, at least
company’s two-way R-series speakers The folding of the ribbon also gives in the higher registers, when the room
are a little different from most of the it a much greater surface area than the acoustics are less than perfect.
competition as the high end is handled ‘space’ it takes up on the front baffle: We reviewed the original R80
by an Air Motion Transformer or AMT, if unfolded, the 0.01mm-thick Kapton model back in June 2017 (https://sosm.
sometimes referred to as a folded AMT membrane used in the R80 V2 ag/presonus-r-series), but this new
ribbon. This type of tweeter, designed reviewed here would measure 6.8 V2 update adds a number of extra
by German engineer Oskar Heil and square inches — roughly equivalent in features, including EQ controls on the
popularised in the private studio market area to a conventional tweeter three rear panel, specifically Low Cut, Mid
by ADAM Audio, has been adopted inches in diameter. The shape of the and High controls. The Mid and High
by a number of manufacturers now, tweeter housing, meanwhile, controls bands offer both cut and boost (±6dB),

14 February 2022 / w w w. s o u n d o n s o u n d . c o m
 12Mic
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Worldwide Distribution: Audio AG, +49 8133 9181-0, www.audioag.com • UK Distribution: Synthax UK, +44 1727 821870, www.synthax.co.uk
www.rme-audio.com
 ON TE S T
PRESONUS R80 V2

Bundled Software
As a deal-sweetener, PreSonus include a special edition
of their Studio Magic software suite with their studio
monitors. The bundle includes Studio One Prime and
a large collection of plug-ins, loops, virtual instruments,
effects and tutorials. All the included plug-ins come in
VST, AU and AAX formats.

and the Low Cut filter can be switched off or set

to either 80Hz or 100Hz. An additional Acoustic
Space switch offers 0, -3 or -6 dB of low-frequency
attenuation to compensate for placement relative to
room boundaries.
Housed in a ported two-way cabinet measuring
241 x 309 x 379 mm weighing a reassuring 9kg,
the R80 V2 is driven by a pair of updated Class-A/B
amplifiers delivering 75 Watts to the woofer and
65 Watts to the tweeter, crossing over at 2.6kHz.
The R80 V2’s eight-inch woofer features a rigid,

“Voices come over with plenty

of presence and detail, and the
stereo imaging is good too...”

custom-woven composite cone with a slightly

concave dust cap, with the aim of producing more
consistent dispersion across its frequency range. The
moulded baffle has a somewhat angular appearance
and sits at the front of a conventionally plain box with
a black vinyl finish.
There is a range of connection options,
comprising a balanced quarter-inch TRS jack,
a balanced XLR and an unbalanced RCA phono
socket, all operating at line-level. Power comes in
on the usual IEC socket, there’s a separate power The R80 V2 adds a range of EQ options, including a low-cut filter, mid and high EQ,
switch, and a front-panel LED shows when the plus a three-position room placement compensation switch.
speaker is powered up. The operating voltage is
selectable via a recessed slide switch. sound sitting comfortably in doing a mix on them. There’s more than
On paper the speakers cover the frequency my own room though, as the enough level for close-range monitoring,
range 40Hz to 22kHz at the -3dB points, while speakers sounded a little too and the only extra I’d suggest is a pair of
the maximum SPL at one metre is 107dB, and the bright in the ‘flat’ setting. decent monitor platforms, especially if
A-weighted signal-to-noise ratio is better than 100dB. Voices come over with you plan to put the speakers on your desk
plenty of presence and or on a shelf. Overall then, a worthwhile
In Use detail, and the stereo update to the original R80.
Because of the wide, shallow dispersion of these imaging is good too, placing
speakers, it is important to get the tweeters aimed centre-panned sounds solidly summary
at your head in the vertical plane. If you have them between the two speakers.
aimed too high or too low, you may lose a little of the As with many ‘affordable’ These are very detailed monitors with a wide
sweet spot and no unacceptable faults
HF. Get them right though and you’ll be rewarded by speakers, the lower midrange given their price, though you may need to
a detailed sound completed by firm lows. While the is slightly coloured due to tweak the EQ settings to tame their slightly
bottom end isn’t as tight as you might expect from small resonances, but again forward character.
a fully sealed cabinet, it doesn’t display excessive there’s nothing that worries
overhang, so dry kick drums don’t suffer excessive me too much given their price, £ €858 per pair including VAT.
smearing. I rolled the HF back just a hint to get the and I’d feel very comfortable W www.presonus.com

16 February 2022 / w w w. s o u n d o n s o u n d . c o m
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 ON TE S T

Milab
VIP-60
Multi-pattern
Capacitor
Microphone
Swedish mic manufacturers
Pearl and Milab have been
doing things differently
for 80 years — with
outstanding results.
SAM INGLIS

I
f there’s one piece of kit that I have
a slightly irrational passion for, it’s
microphones, and I hope I never
find myself in a situation where there’s
only time to rescue one of mine from
a fire. If that does happen, though, it
won’t be a U87 or a C414 I drag from the
smoking ruins. Instead, it will be my ST8,
a little-known stereo mic made by Pearl
Microphone Labs.
PML were founded in Sweden
in 1941 by Rune Rosander. A keen
experimenter, he pioneered a number
of innovations, including the use of
rectangular diaphragms in capacitor
microphones. Introduced as long ago
as 1958, Rosander’s rectangular capsule
has a number of interesting and useful
properties. The circular membrane used in
conventional capsules has a single, strong
resonance, which needs to be tightly
controlled by damping. By contrast, the
rectangular diaphragm has two, smaller
resonant modes. And whereas a circular
capsule has the same directivity in the
vertical and horizontal planes, rectangular
capsules are markedly less sensitive to
off-axis sound in the plane of their long
axis than the short axis. The rectangular
capsule would go on to be an enduring
hallmark of PML designs through the
years, including the ST8.
In 1980, a second company called Milab
was established under the leadership
of Rosander’s son, the idea being that

18 February 2022 / w w w. s o u n d o n s o u n d . c o m
Rosander and PML would focus on R&D
while Milab built a wider market for the
products. In the event, however, the two
companies ended up as competitors,
even though some models were marketed
under both brands.
A key figure in this story is Bernt
Malmqvist, who originally joined PML as
an 18-year-old machinist in 1954. When
Milab was created, he became that
company’s production manager, before
leaving to buy back the Pearl company
on Rosander’s retirement in 1987. As chief
designer at PML, Malmqvist continued to
build on Rosander’s research, introducing
many further innovations until, after 63
years in the business, he formally retired
in 2017, and the two brands were at last
reunited. The current Milab catalogue thus
features both Milab and Pearl mics, and
support is available for older models from
both companies.
VIP Endorsements
Although neither Pearl nor Milab has ever
quite achieved ‘household name’ status
in the way that Neumann and AKG have,
their products have a deservedly high
reputation, with advocates including Ray
Charles, Quincy Jones and Bruce Swedien.
The mic that did most to achieve this in
the USA was the Milab VIP-50. Launched
in 1987 and still in production today, the
VIP-50 is a multi-pattern capacitor mic that
houses Milab’s 2700 rectangular capsule
in an equally unconventional rectangular
body. It’s not sleek or elegant in the way
that other high-end mics are, and in fact
many would consider it downright ugly,
but it has the merit of being instantly
recognisable! To celebrate PML’s 80th
anniversary in 2021, Milab launched both
an Anniversary Edition of their equally
classic DC-196, and an updated version
of the VIP-50. The VIP-60, as this mic
is known, uses the same case as its
predecessor, but there are numerous
internal changes and improvements.
The VIP-60 is a phantom-powered,
large-diaphragm, true capacitor mic,
drawing a fairly modest 4mA. Rather than
the original 2700 capsule, the new mic
uses the 2900 capsule also found the
DC-196. This retains the dimensions found
in historic Milab and PML models, rather
than the more elongated shape of the
newer ELM capsules. Like its predecessor,
the VIP-60 is a transformerless mic,
and perhaps the most significant
improvement Milab have made is to
reduce self-noise. The specifications quote
14dB (A-weighted), and the individual
test result supplied with the review mic
reported 13.0dBA in cardioid, where only
one side of the capsule is active and thus
self-noise tends to be lowest. This is not
state-of-the-art, but it’s 4 or 5 dB better
than the VIP-50, and comparable to the
current Neumann U87Ai.
A number of factors contribute to
a microphone’s self-noise. One of the
most important is the surface area of the
capsule. In this respect the rectangular
diaphragm has the advantage — for
a given width, a rectangle has a greater
area than a circle — but other differences
can counteract this. For example, whereas
the familiar Neumann K67-style circular
capsule has an intrinsic high-frequency
emphasis because of its single primary
resonance, Milab’s rectangular capsules
are less sensitive in the high frequencies.
To achieve an overall flat response, the
mic’s electronics incorporate a gentle ’tilt’
across the audio spectrum, which incurs
a small noise penalty. Either way, though,
unless you’re recording earthworms
fluttering their eyelids, the VIP-60 will be
plenty quiet enough! Sensitivity is given as
14mV/Pa, which is about what you’d want
in a general-purpose studio mic.
60 Something
With its unique shape and rough grey
finish, the VIP-60’s appearance will
continue to divide opinions. Personally,
I rather like it, and it’s certainly the
antithesis of the ‘blingy’ cosmetics found
on some mics. It also feels extremely solid
and well made. The VIP-60 is shipped
in a very stout and practical, watertight
plastic flightcase, and comes with a Rycote
InVision shockmount. This is a good choice
from the point of view of performance,
but it only fits in the case if the bolts that
hold the mic are screwed all the way
in. I suspect most users won’t have the
patience to do this every time they want to
pack the mic away. As a ‘quick and dirty’
alternative, there’s also a 3/8-inch socket
in the base of the mic itself.
One of Milab’s many distinctive
design touches is the use of miniature
rotary switches to change the polar
pattern and select filter and pad settings;
unusually, these are on the front of the
mic rather than the back. Five polar
patterns are available, with subcardioid
and hypercardioid adding to the usual
omni, cardioid and figure-8. Also novel is
the choice of filter settings. Rather than
offering a steep roll-off at a low frequency
to eliminate subsonic noise, these are
designed to counteract proximity effect
when used up close, providing a gentle
slope that turns over at 200 or 500 Hz.
The pad offers -10 and -20 dB settings,
and with the -20dB setting engaged, the
mic can accept SPLs of up to 138dB before
1 percent distortion is reached.
The Pearl/Milab rectangular capsule is
often described as having a ‘ribbon-like’
sound, perhaps due to the absence of
that prominent single resonance that
characterises circular capsules. If ‘ribbon-
like warmth’ is another way of saying that
these mics have a relatively restrained
high-frequency response, there’s probably
something in it. Generalised frequency
response charts for the VIP-60 weren’t
available when we went to press, and
the individual plot for the review mic only
showed the response in cardioid. This was
almost dead flat from 40Hz to 13kHz or
so, with very gentle dips around 6kHz and
11kHz. The response does fall away at the
extreme top end, but less dramatically than
the VIP-50 does. If that mic is anything to
go by, the VIP-60’s pickup should prove
more uneven through the midrange
in figure-8 than in cardioid, and more
attenuated at the top end in omni, and that
chimes with what I heard in those patterns.
A look at the Milab catalogue will
tell you that the VIP-60 doesn’t just
compete with high-end mics from other
manufacturers. Milab themselves still
offer the DC-196, which uses the same
capsule in a much more compact housing,
and achieves better noise and headroom
Milab VIP-60
£2399
PROS
• A great-sounding studio workhorse
that works well on almost anything,
especially vocals.
• Worthwhile improvements over the
existing VIP-50.
• Useful filter settings.
• Asymmetrical polar patterns open up
some interesting possibilities.
CONS
• If you want state-of-the-art noise and
headroom specs, you’ll need to look
elsewhere.
• Packing the shockmount away is
tedious.
SUMMARY
A well-judged update to a cult
classic, delivering improved technical
performance without compromising
the sound that Milab devotees know
and love.
w w w. s o u n d o n s o u n d . c o m / February 2022 19
 ON TE S T
MIL AB VIP-60

figures, though it lacks the intermediate

polar patterns and the filters. As I wasn’t
able directly to compare the two mics,
I asked Milab about the differences
between them. They explained that
the two mics have completely different
electronics, and that the radically different
headbasket designs also make quite a big
difference to the sound. The DC-196 is the
more ‘modern’ mic in design terms, but
the VIP-50 and now the VIP-60 remain
in Milab’s catalogue because there is still
a demand, just as Neumann continue to
offer the U87 even though they make
other mics with better technical specs.
Compared with a U87, the VIP-60
sounds almost plain, without the Neumann
mic’s forward quality in the midrange, but
both mics have two things in common that
I think of as being the hallmarks of really
good large-diaphragm mics. One is that
although sources aren’t hyped, they are
nevertheless presented with a substance
and solidness that makes them easy to
work with at the mix. It’s easy to add top
end with EQ if you do want things to be
brighter, but somehow that often doesn’t
feel necessary, because they make
themselves clearly audible without it. The
other is that you can be confident in their
ability to tackle pretty much anything that
comes through the studio door. In the real
world, there’s never time to try out every
mic in the locker in pursuit of the perfect
match, so the ones that get used the most
are the ones that work on the widest range A Rycote InVision shockmount
is included in the package.
of sources. By that measure, the VIP-60 is
up there with the best.
for recording singing guitarists and is not a forensic tool like a measurement
Raising Voices the like. My informal tests showed that mic. Its dryness is that of a dry Martini or
Milab told me that the VIP-50 is particularly off-axis capture certainly varies from the a good white wine: not as immediate or
popular as a vocal mic, and so they didn’t horizontal to the vertical, but it’s perhaps as obvious as the sweetness of chocolate,
want to change anything in the VIP-60 an oversimplification to say that rejection but ultimately more satisfying.
that would undermine its strengths in this is better or worse in either plane. There The VIP-60 is priced at a level which
department. Having used it on quite a few are quite pronounced differences in the makes it a competitor to the Neumann
voices in the review period, I’d say they tonality of the spill in each case, so if U87Ai, Microtech Gefell UMT-70S and
got it right: this is never less than a good you’re after maximum isolation, it may Sanken CU-44X, among others. All of
vocal mic, and frequently an excellent well be worth trying both options in order these are tried and tested designs, and
one. And on vocals in particular, those to see what works best subjectively with you can’t really go wrong with any of
filter settings really come into their own, the source. Finally, although the VIP-60 them. The VIP-60 easily deserves its
helping to deliver a natural and balanced doesn’t have the endless headroom place in the same category. It represents
sound when singers want to work the mic of some modern mics, it does have a worthwhile and sensitive upgrade to an
up close. As is the case with many classic that -20dB pad, and I really liked it as existing classic, and will deliver a lifetime
multi-pattern mics, the tone of the VIP-60 a guitar amp mic, where it delivers a solid, of high-quality recordings.
does change somewhat depending on up-front sound without exaggerating any Now, any chance of reissuing
the pattern settings. On my own voice, harshness or mud. the ST8...? 
I preferred cardioid, and found omni the If I had to pick one word to sum up
least useful, but I can imagine it working the character of the VIP-60, it would be
£ £2399; matched pair £5059.
well for other singers. ‘dry’. By that, I don’t mean that it magically Prices include VAT.
Off-axis rejection in figure-8 mode ignores reverb. Rather, it doesn’t obviously W www.funky-junk.co.uk
is very good, making this a useful mic sugar the source, yet at the same time, W www.milabmic.com

20 February 2022 / w w w. s o u n d o n s o u n d . c o m
 ON TE S T
Boss Eurus GS-1
Guitar With SY Synth Engine
Pedal legends Boss have now built their clever
synth engine into a regular guitar — no hex
pickups required!
PAUL WHITE
A
lthough their parent company
Roland are well-documented
pioneers of guitar synths, for Boss
it’s a relatively new venture. However,
though the same Roland engineering
team is behind the GS-1 and related Boss
products, in some ways the GS-1 turns the
clock back to the Roland GR-300, which
had its own guitar plus a box of analogue
electronics. The big difference is that
the GS-1 incorporates all the necessary
electronics into the guitar, and no hex
pickup is required: like the Boss SY-300
standalone guitar synth system, the SY-1
pedal and the SY-200 pedal, the GS-1
processes the sound from a conventional
guitar pickup to produce a range of
analogue-style, polyphonic synth sounds,
albeit digitally.
Overview
The guitar itself is made in one of the
leading far-eastern guitar factories and
22 February 2022 / w w w. s o u n d o n s o u n d . c o m
has the same 25.5-inch scale length
as a Fender Stratocaster or Telecaster.
Its fingerboard is Indian rosewood
rather than one of the currently popular
substitutes while the C-shaped maple
neck has easy access truss-rod
adjustment at the body end and is
fitted with a bone nut. Its fingerboard
has a compound radius from 12 to 16
inches, which is comfortable and doesn’t
choke out on extreme note bends.
Locking Gotok tuners are fitted as
standard and the vibrato bridge is also
by Gotoh with a push-in arm secured
by a variable-pressure grub screw. The
body is alder. Unlike a Strat, the guitar
has two Eurus humbucking pickups but
wired via a five-way switch to offer neck
humbucker, neck single coil, neck and
bridge humbuckers, bridge single coils
in parallel and bridge humbucker. This
provides a wide range of tones straddling
the traditional Fender and Gibson camps,
all presided over by a single master
volume and single tone control.
The synth section is accessed by
a six-way rotary switch to bring up each
of the six sounds that can be stored
on board with just one other control to
Boss Eurus GS-1
£1933
PROS
• Easy to use with no latency or
pitch-tracking issues.
• Free, simple editing app using
Bluetooth connection.
• Compatible with optional EV-1-WL
Bluetooth foot controller.
• Plausible analogue-style sounds.
CONS
• No onboard effects.
• No dedicated synth/guitar
balance control.
• Stores only six presets.
SUMMARY
While the SY-300 pedal offers far more
sonic flexibility, this is a great all-in-one
solution for the guitarist who wants to
add synth sounds without hooking up
separate hardware.
adjust the most important parameter for the patch,
usually related to the filter. A Mode switch selects
Synth or Guitar though there are also two output jacks
so the guitar can be sent to its own amplifier and the
synth sound to a keyboard amp or PA system. It is
also possible to use a single output carrying a mix
of guitar and synth, though this isn’t as flexible as
having separate amps, and if you use a lot of amp
overdrive, the synth sounds will get muddied. There’s
no dedicated synth/guitar balance control though the
synth control knob can be assigned to this purpose if
required. If using the separate outputs, then feeding
the synth output via a volume pedal is a practical This extremely limited version
WNW]ZI_IZL̉_QVVQVO5MZK]ZaQ[Å\\ML
option. There’s a voicing option for the synth section _Q\PIZIZM]V][ML5]TTIZL+>^IT^M
for playing though a guitar amp but a full-range <PM[M^IK]]U\]JM[PMTLQVUQTQ\IZa[\WZM[
[QVKM\PMCold War eraOQ^M5MZK]Za
speaker is really the best-sounding option if you can IVM^MV[UWW\PMZ[QTSQMZ\WVMIVLVI\]ZIT
arrange it. KWUXZM[[QWV_PI\M^MZaW]¼ZMZMKWZLQVO
Power for the guitar can come from an external 9V )VM`\ZỈ[XMKQITUQK\PI\Q[I
must-haveNWZIVa[\]LQW
PSU (not included) but more normally from four AA
batteries. Plugging in a jack lead powers up the guitar.

The GS-1 features separate outputs for the regular humbucker

pickups and the synth sound, as well as a USB port for system
updates from a Mac or Windows computer.

.
sontronics sontronics /sontronicsuk
 ON TE S T
BOSS EURUS GS-1
The GS-1 can communicate with an optional
EV-1-WL wireless footpedal or a smartphone app
(below), both of which can be used to adjust key
synth parameters in real time.
Unusually, the guitar also has Bluetooth
that can be used with an optional EV-1-WL
Bluetooth foot controller or for editing
the on-board sounds using the free GS-1
app for Android or iOS. The depth of
editing is fairly limited, constrained to
the key parameters: Tone, Resonance,
Depth, Direct Level and Synth Level plus
functions for Hold and Pitch. The App
also allows for a larger choice of source
24 February 2022 / w w w. s o u n d o n s o u n d . c o m
sounds such as bell-like, string-like, pad
or organ. The EV-1-WL Bluetooth foot
controller and its switch functions can be
assigned only to these parameters.
In Use
The factory sounds cover a fair amount
of analogue-style ground, but unlike the
SY-300, the GS-1 has no onboard effects
and no patches that combine synth
sounds with processed guitar. Having
separate outputs means you can use
different external pedals on the guitar
and synth sounds, but a built-in reverb
would have been welcome, since some
of the synth sounds can sound a bit dry
without it. The quality of the sounds
more generally is impressive, though,
encompassing those classic GR-300
pseudo brass and string types of sound
as well as fat octaves, filter sweeps and
a really convincing organ, albeit it with
a permanent fast rotary effect applied. It
is also possible to conjure up some useful
sawtooth and square-wave lead sounds.
I’d have felt more comfortable with
a dedicated synth/guitar balance control
— for example, accessed via a pull-switch
on the synth control knob. It may also
have been possible to add a pull function
to the Mode switch to double the number
of stored sounds. On the plus-side,
the way the sounds are created by
processing the sound coming off the
strings means latency is imperceptible
and there are no pitch-tracking errors
so at least players don’t have to relearn
a new playing technique. The synth
follows the string pitch exactly so any
string bends are rendered faithfully.
While the organ sound is impressive,
I would have loved to have a way to
change the rotary effect speed during
performance. Using an EV-1-WL controller
extends the real-time control of the
sounds to a useful degree, and though
setting up the Bluetooth is a bit of a faff
to start with, once you’ve done it the
guitar and pedal pair up reliably.
As a guitar, the GS-1 is no slouch
either; it arrived well set up and is
comfortable to play. The pickup
combinations cover a lot of tonal ground
and the general character is bright and
lively with good note definition and
a well-balanced sustain. The guitar also
rings out cleanly when not plugged in,
which is always a good sign. It certainly
doesn’t feel like a budget instrument
so you could use it as your main
gigging guitar and certainly not feel
short-changed.
It could be argued that if you have
a guitar that you like already, you could
get more flexibility, and a bit of change,
by going for the SY-1 or SY-200 pedal
or the more flexible SY-300, the latter
having the benefit of some very sweet
effects. However, if you hate extra
cabling and just want a simple live
solution for adding the occasional synth
sound, then the GS-1 is an effective
solution. I still think that basic effects
should have been built into the synth
engine, but one more pedal on your
board will take care of that.
£ Eurus GS-1 £1933. EV-1-WL footpedal
£152. Prices include VAT.
W www.boss.info
No tricks
No copies
No gimmicks
No emulations
Only quality. . .

You can't unhear quality

 F E ATU R E
Howard Benson
POD ‘Youth Of The Nation’
J O E M AT E R A
G
rammy Award-winning American
producer Howard Benson has
helmed the desk for numerous
multi-platinum selling albums since he began
his production career back in the mid-1980s.
These include albums by Kelly Clarkson,
the All-American Rejects, Adam Lambert
and Daughtry to name but a few. Here he
discusses how he achieved the drum sound
on POD’s 2001 hit ‘Youth Of The Nation’.
“That drum sound is something I have
always been asked about. Also, ‘Youth Of
The Nation’ was a seminal song after 9/11.
26 February 2022 / w w w. s o u n d o n s o u n d . c o m
We wanted to get the deepest drum that
we could. We were recording on digital at
the time and decided that on that particular
track, we would record the drum part at
around 30 to 40 percent faster. And we
sped the tape up too. So we recorded
drummer Wuv Bernardo playing the toms in
the way they were supposed to be played
and tuned but with the music sped way up.
That way, when we played the tape back at
regular speed, the toms, in their pitch and in
the way they were played, went down a lot.
And that was the best way to do that kind of
thing back then.
“The drums were recorded at Bay 7
Studios in North Hollywood via a Neve 8058
desk. We could’ve used Pro Tools, but the
plug-ins weren’t that good for lowering the
pitch, they’d be glitchy, so instead we sped
the tape and we recorded the whole song at
what sounded to me like double-time speed.
And we still weren’t happy with it, so what
we did was we rented a bunch of timpanis
and we recorded timpani drums over the top
of the toms as well. And we pitched them
really well down. We used three timpanis
for the timpani part. When we had it all, the
drum tracks (after transferring to Pro Tools)
were sent to a guy in Vancouver, Canada
who [engineer] Randy Staub had been
using to do drum editing, to have it edited
so that the toms and timpanis were tight. It
was new to me to use outside editors so it
changed the way I worked because I started
outsourcing the editing that I was always
doing at that time.
Mission Impossible
To this day I have artists calling asking me
how to get that drum sound I got on that
Hear The Sound
W https://open.spotify.com/track/0UrWr7Jnu
1heq1o99ZwUd0?si=76cce61ef2d7433a
W https://www.youtube.com/
watch?v=EDKwCvD56kw
record, but its almost impossible to do
that. We have to rent timpanis, and do the
whole speeding everything up thing and
slow it down because the sound of the
tape machine slowing down, you can’t
really replicate that easily. There’s a lot of
compression that would be needed, the
analogue warmth it gives and everything.
“Miking was done with AKG C414s on
the drums with [Shure] KSM 32s used for
the Toms, while for the timpanis it was most
likely AKG C24s. We recorded the timpanis
separately, together with a room mic. The
drums were pretty complicated on that
record. We wanted it to sound kind of tribal.
“We added a lot of compression in
the mix to make it all fit in because it was
pretty jumpy with all the drums. [Mixer]
Chris Lord-Alge usually samples the drums
but this time around on this track, he
couldn’t really do that. He had to use the
live drums.”
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 T E C H NI Q UE

Vocal Mix Prep

Processing Options Before Mixing
With your lead vocal performance
comped and tuned, and its timing
tightened, is there anything else to
do before you start mixing?

30 February 2022 / w w w. s o u n d o n s o u n d . c o m

MIKE SENIOR
O
ver the past couple of months,
I’ve written about comping vocal
performances from multiple
takes and correcting any tuning/timing
issues they might have. And, up until
about decade ago, that was pretty much
all there was to the mix-preparation
process, as far as vocals were concerned.
In recent years, however, facilities for
manipulating individual audio regions/
clips within your DAW have continually
improved, and professional vocal
producers now routinely use the hell out
of those functions to refine lead-vocal
sonics and lyric intelligibility. In this article
I’m going to explore some of these
techniques, and how you might get the
best out of them.
A Fader For Everything
The first vital tool is the humble ‘clip gain’
setting, which lets you apply a simple
level offset to any audio region, and
it’s advisable to use this to even out
big level disparities between words/
phrases before you apply any general
mixdown compression.
But isn’t compression supposed to
sort this kind of stuff out anyway? Well,
yes, but compressors often won’t actually
do what’s most musically appropriate.
For example, if your singer accidentally
moved closer to the mic for one phrase,
then the compressor will just see
more signal level and will compress
more aggressively, potentially giving
that phrase a different sound flavour
or emotional character. By using clip
gain to turn down the overloud phrase
instead, before it hits the compressor,
the compression and
vocal sound remain more
consistent. Similarly,
if you’ve set up your
compressor with a slow
release time in search
of a fairly natural sound,
but then the singer
Here you can see two
versions of the same vocal
performance. The wide dynamic
range of the upper track will
likely make it difficult to find
a musical-sounding compression
setting at mixdown, which is why
applying clip-gain adjustments
to some regions (highlighted)
makes a lot of sense at the
editing stage.
suddenly hits one particularly loud
note, the compressor may trigger heavy
gain reduction that lingers too long,
making subsequent notes too quiet. Again,
a simple clip-gain move neatly side-steps
this problem.
You might also want to use clip gain
for more detailed rebalancing within
vocal phrases. So where you’re planning
on creating a heavily compressed vocal
sound, say, it can make sense to fade
down the breaths and sibilants with clip
gain, knowing that these will invariably
be overemphasised by most compressor
designs otherwise. Plus some elements
of a vocal recording are inherently
more difficult to bring out in a busy mix
than others, so I regularly use clip-gain
adjustments to improve the intelligibility
of low-energy consonants such as ‘h’, ‘l’,
‘m’, ‘n’, ‘ng’, and ‘v’. That said, sometimes
singers will swallow a consonant so badly
that there’s actually nothing there to
boost! In those cases, I’ll often search for
a replacement consonant from an alternate
vocal take if possible, and occasionally
a consonant from elsewhere in the timeline
will also work, although your success there
will depend heavily on the specific adjacent
vowels in each instance.
Equaliser Sharpshooting
Although clip-gain settings are already
tremendously useful on their own, the
real power of region-specific adjustment
lies in the ability to apply EQ to individual
audio snippets. For instance, if your
singer has ‘popped’ the mic, resulting in
strong low-frequency wind-blast pulses on
plosive consonants such as ‘p’ and ‘b’, the
best way to sort this out is by high-pass
filtering just the offending moments.
Admittedly, you might be able to get
Additional Resources Online
To support this article, I’ve set up a special
resources page on my website, where you’ll
find audio and video demonstrations of the
editing techniques discussed in this article,
downloadable multitracks with unedited vocal
takes (so you can practice your region-based
editing), and links to further reading.
www.cambridge-mt.com/vocal-editing
similar plosive reduction by just high-pass
filtering the whole vocal channel, but that
runs the risk of thinning out other sections
of the performance too, unlike the more
precisely targeted region-specific strategy.
Furthermore, it’s easy to apply different
high-pass filter settings to different
plosives if some of them are more
severe than others (as is often the case)
— or even to deal with other sporadic
low-frequency problems such as the
thumps of accidental mic-stand collisions
or a musician absentmindedly tapping
their foot on the stand base.
A similar tactical approach can also
elegantly address over-prominent breaths
or noise consonants (such as ‘k’, ‘ch’, ‘s’,
‘sh’, and ‘t’ sounds), because you can tailor
the EQ processing to respond to each
moment’s unique demands. For example,
many studio condenser mics have a strong
brightness peak in the 10-12 kHz zone
that doesn’t subjectively enhance vowel
sounds nearly as much as it does noise
consonants, so it’s easy to end up with
an apparently paradoxical mix scenario in
which the singer simultaneously seems to
lack ‘air’ (because the vowels need more
high end) and sound abrasive (because
the noise consonants are over-hyped).
Applying simple 12dB/octave low-pass
w w w. s o u n d o n s o u n d . c o m / February 2022 31
 T E C H NI Q UE
VOCAL MIX PREP

If you’re trying to match the tone of vocal takes

recorded on different sessions, the necessary
equalisation curve can be quite difficult to work out
by ear. Fortunately, there are plug-ins such as Logic’s
Match EQ, Melda’s MAutoEqualizer and Tokyo Dawn’s
VOS Slick EQ GE that can analyse the frequency
profiles of the different takes and calculate sensible
EQ suggestions automatically.

filtering to just the abrasive consonants

can sort this out instantly, although you
may want to tweak the filter’s cutoff
frequency on a per-case basis too — after
all, a sharp ‘t’ sound may feel a lot more
abrasive than a mellow ‘sh’. On occasion,
I may use more elaborate EQ settings
too but, to be honest, most of the time
that’s just chasing diminishing returns and
a quicker alternative fix might be to copy
a better-sounding consonant from another
vocal take.
Granted, you could use a mixdown
de-esser to handle some of these tasks
in a more set-and-forget manner, but
even if that plug-in manages the sibilance
flawlessly, it’ll still struggle to tame strong
‘k’ or ‘t’ sounds as effectively, and it may
also end up killing some of the vocal’s
delicate ‘air’ frequencies into the bargain.
Moreover, even the best software
de-essers rarely work equally well on
every instance of sibilance in a given
performance, whereas region-based EQ
can be made to measure. What I’ll often
do with a particularly thorny sibilant, in
fact, is loop it while I set up the EQ. It’s
not a pleasant listening experience, as
you can probably imagine, but it does
make it much easier to home in surgically
on the most unpalatable frequencies.

Inconsistent Vocal Timbre

When mixing vocals, it’s usually desirable
to achieve tonal consistency throughout
the performance. In other words, you’re
trying to make every single syllable
sound great, not just some of them. This
might seem like an impossible task when
faced with real-world vocal recordings,
but region-specific EQ processing is the
magic ingredient you need.
Some tonal problems arise from
recording technique. If the singer
doesn’t maintain a constant distance
from the mic, for instance, then the mic’s
proximity effect may cause dramatic and you loop the syllable in question (another At the other end of the spectrum,
unmusical fluctuations in the subjective mildly unsettling listening experience) the amount of ‘air’ in the vocal tone can
warmth of different notes. I’ll usually and examine its frequency content on often be frustratingly variable, not only
use a low-frequency EQ shelf to restore a spectrum analyser, then you’ll see the because open vowel sounds (‘ah’, ‘eh’,
balance here, with the shelving frequency note fundamentals pretty clearly, and ‘ee’) naturally have less airiness than
adapted to suit the nature of the voice setting the shelving frequency just above more closed ones (‘oo’, ‘aw’, ‘uh’) when
and the register it’s singing in. Again, if those will often be a good place to start. captured close up, but also because

32 February 2022 / w w w. s o u n d o n s o u n d . c o m
 Dealing With Hiss
No recording chain captures audio perfectly, so every vocal recording
inevitably contains some background hiss. With modern recording equipment,
this is seldom worth worrying about for music applications, but if noise begins
to feel intrusive then there are specialist tools you can use to reduce it at
the editing stage. Plug-ins such as Acon Digital DeNoise, iZotope RX Voice
De-noise and Sonnox DeNoiser are all capable of great results, although
it definitely pays to dive into the settings with whatever option you go for,
because small differences in configuration can have a significant impact
on the sense of ‘air’ and breathiness of the processed vocal tone. In my
experience, you’ll get best results if you use the software’s ‘learn’ mode to
train the noise-reduction algorithm on a section of isolated background noise,
and it’s also better not to reduce the noise levels any more than absolutely
necessary, because unwanted processing artefacts tend to accumulate the
more you push things in this respect. I’d also suggest tackling noise reduction
prior to your tuning or timing edits, because pitch/time-manipulation engines
usually operate better on cleaner (and therefore more mathematically
predictable) audio signals.
Another thing to bear in mind with background noise is that, while you may
not consciously notice it while it’s constantly in the background, it’ll become
a lot more noticeable if it suddenly cuts in or out. So if you want to maintain
the illusion that your vocal is one natural take without technical manipulation,
then don’t reflexively cut out all the ‘silent’ parts of the vocal recording
between phrases or song sections, because that’ll kill the hiss and draw
attention. If there’s unwanted shuffling around or coughing from the singer
during those gaps, then by all means patch things up with cleaner sections of
hiss from elsewhere in the take, but if that’s not possible then at least try to
mute the noise at a section boundary or on a strong musical beat, where the
sudden textural change will be less obtrusive.

high frequencies are very directional, so
small movements by the singer can end
up firing some of them far less strongly
into the mic than others. A simple band
of high-shelving EQ around 10kHz can
work wonders remedying this, though,
boosting the upper spectrum for each
‘air’-deficient region to even out the tone.
Of course, you may also encounter
a performance which has been comped
from two separate tracking sessions with
different vocal sounds, and region-based
EQ can deal with that as well. However,
such tonal differences can sometimes
be quite complex, making it pretty
challenging to arrive at a decent timbral
match by ear. This is where a specialist
‘EQ ripper’ plug-in such as Logic’s Match
EQ, Melda’s MAutoEqualizer or Tokyo
Dawn’s VOS Slick EQ GE can lend a hand,
analysing audio from both sessions and
calculating a multiband EQ curve to map
the frequency profile of one onto the
other. A word of caution here, though:
sometimes these algorithms come up with
overly extreme settings, especially at the
frequency extremes, so be prepared to
take their suggestions with a large pinch
of salt. I’ll usually take the precaution of
at least bypassing each of the suggested
 T E C H NI Q UE
VOCAL MIX PREP
Mouth Clicks & Lip Noise
The physical mechanics of a singer’s lips, mouth,
and tongue moving against each other while
performing can naturally give rise to tiny little
high-frequency snicking sounds often referred to as
‘mouth clicks’ or ‘lip noise’. These are normally so
low in level that they rarely cause a problem with
full-bodied vocal performances, but can become
a massive pain in the arse if you’re close-miking the
kind of whisper-quiet singing style that’s become
so fashionable in the wake of Billie Eilish. Making
sure your singer takes regular sips of water can help
somewhat, but you’ll not fix the problem entirely
without a good deal of careful editing besides.
Where the mouth clicks happen during gaps,
breaths, or noisy consonants, it’s usually very
straightforward to get rid of them. First, zoom
right in on the waveform so you can see the click
— it’ll usually be extremely short, roughly one or
two milliseconds in length. Then snip either side
of it, slide the audio within the snipped region to
one side (so that the click is effectively replaced
with a copy of the audio
immediately before/after
it), and then put miniscule
crossfades at the two edit
points. Sometimes you’ll get
little bursts of two or three
mouth clicks that seem to
happen all at once, but once
While simple
audio editing methods can
deal with the majority of
mouth clicks, occasionally you
may need to bring in specialist
click-removal tools such as
Acon Digital DeClick, iZotope
De-click and Sonnox DeClicker.
bands to double-check by ear that it’s
actually doing something sensible.
Battling Nature
Other tonal inconsistencies derive from the
basic mechanics of the voice itself. When
a male singer switches from chest voice into
falsetto, for example, the tone will usually
change radically, trading midrange focus
for a combination of pillowy low end and
extra breathy highs. However valid this
may be from an expressive standpoint, it
can nonetheless present a mix-translation
problem. You see, the midrange carries
better over small speakers, so it’s easy
to end up in a situation where falsetto
notes that seem to be at the right level on
a full-range listening system become less
easy to hear in the balance on mass-market
playback devices. Some judicious region EQ
delivering 3-6 dB of peaking boost at around
1-2 kHz can really help square that circle.
34 February 2022 / w w w. s o u n d o n s o u n d . c o m
you zoom in you’ll still typically see a gap of
several milliseconds between each one, so you
can deal with them in exactly the same way.
Where mouth clicks get trickier to manage is
in the middle of sung notes, often triggered by
vowel-consonant transitions involving ‘l’, ‘m’, ‘n’,
and ‘ng’. Here, my simple little slice-and-slide
method would cause audible lumps and bumps in
the note’s sustain, so my first port of call is to find
the waveform cycle the click appears in, slice at
the zero-crossing points on either side
of it, and then drag the audio within that
region to replace it with the waveform
cycle before or after the click. If you use
short equal-power crossfades, this’ll
give a totally transparent fix 90 percent
of the time, but if it doesn’t work,
another option is simply to render the
click’s tone less distracting by isolating
it into its own region and applying
a low-pass filter at around 2kHz.
Another common issue is where
some vowel sounds on certain (often
higher-register) notes generate
powerful individual harmonics in the
vocal spectrum, giving the voice an
unpleasantly whistling/piercing timbre at
unpredictable moments. The solution to
this is usually a narrow-band region-EQ
notch or two, but you need to aim
accurately in the frequency domain if
they’re going to sufficiently tame the
harshness. This is another time when it
can make sense to loop small sections of
the audio while watching a high-resolution
spectrum analyser, since the vertiginous
spectral peaks you’re looking for should
be clearly visible on that.
Editor’s Choice
If you use all of these editing tricks
sensibly, you’ll discover that the
mixdown process becomes a whole lot
Alternatively, if nothing else works, you can
try specialist click-removal software such as
Acon Digital DeClick, iZotope De-click or Sonnox
DeClicker. As always, I’d recommend applying
such plug-ins only to the specific affected
syllable, rather than as a blanket process,
because by the time they’re working assertively
enough to fix prominent mouth clicks, they’ll
also merrily remove all sorts of other desirable
transient detail too.
more straightforward, because you’ve
pre-balanced all of the vocal’s many
sonic components better against each
other. After all, it’s much easier to get
musical and predictable results from
your channel EQ and compression if
you’re not having to use it for serious
remedial work too.
I’ll offer you one final piece of advice,
though: if you find that you have to
split up the audio in order to apply clip
gain or plug-in processing to a specific
region (you will need to in some DAWs),
then remember when crossfading
your split points to use equal-gain
crossfades rather than equal-power
ones — otherwise, you’ll get little gain
‘bumps’ at the crossfade points. The
equal-gain type is usually displayed
graphically as a straight ‘x’ shape,
whereas the equal-power fades are
usually shown as curved.
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 ON TE S T

SoundParticles
PAUL WHITE

F
rom our previous reviews of Energy
Panner and Brightness Panner, you’ll

know that Sound Particles have got
multichannel panning down to a fine art, and
their new plug-in Space Controller builds on
that in a way that’s both simple in concept
and ingenious. It uses your mobile device
as a pointer, so that you can steer the sound
source in all directions just by pointing it as
you would a laser pointer.
Immersive multichannel outputs or
binaural processing also make use of the
height dimensions, and this mode of control
is much easier to manage than struggling
with two joysticks. Thus, it could clearly
be attractive to anyone involved in film
soundtrack work and game design — it
makes placing things like sound effects
very intuitive — but there’s no reason at all
not to use the system in music production,
especially now that the streaming
platforms are starting to pick up on more
immersive formats.
Overview
Supporting plug-in formats AAX, AU and
VST, and with app control via an iOS or
Android phone (or an iPod touch), there are
Space Controller
Smartphone-activated Surround Pan
Control Plug-in
Wouldn’t it be nice if there were a simple, intuitive way
to place sources in your immersive, surround or binaural
soundstage? Well, now there is!
two versions of the DAW plug-in available. Atmos Objects and to pan in more audio
The standard version connects to the phone formats: Quad, 4.0, LCR, LCRS, 5.0.2, 5.1.2,
app over Bluetooth control and supports 5.0.4, 5.1.4, 7.0.4, 7.0.6, 7.1.6, 9.0.8, 9.1.8, 11.1.8.
surround formats up to 5.1 as well as There’s also control over the LFE channel,
Binaural. The Studio version can operate via while the number of communication links
either Bluetooth or Wi-Fi (beneficial where between phones and outputs has been
greater distances are involved between the increased to 128.
phone and computer) and supports four The Space Controller smartphone app is
stereo output formats and surround formats a free download. Once you have it talking to
up to 7.1.2, including Dolby Atmos. With the the computer, there’s an area of screen to
version 1.1 update that came along as I was tap to establish which way is ‘forward’: just
writing this review, there’s now the option aim the phone between your speakers or at
to copy and paste automation to Dolby your centre speaker if you have one, press

36 February 2022 / w w w. s o u n d o n s o u n d . c o m
BiG SiX
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Advanced hybrid production

BiG SiX combines tools from iconic SSL studio consoles with the
latest in hybrid production. Whether you’re wanting to take
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TPVOE to the next level with SSL analogue processing or require
advanced hybrid workflow for multi-track recording or stem mixing,
BiG SiX offers absolute flexibility and creative agility for any

production scenario.

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EQ,
glorious-sounding mic pres with channel processing, and
advanced
monitoring and routing, all built on a SuperAnalogueTM
foundation
and brought together with full 16-channel USB integration.

BiG SiX is the essential SSL studio.

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 ON TE S T
SOUND PARTICLES SPACE CONTROLLER
Sound Particles
Space Controller
£77
PROS
• Supports most surround, binaural and
immersive formats.
• Easy and intuitive to use.
• Bluetooth and Wi-Fi options.
• Android and iOS supported.
CONS
• None.
SUMMARY
A simple idea nicely executed, Space
Controller makes smooth-sounding
immersive, surround and binaural
panning easy and intuitive.
the button and you’re all set. After that,
touching the screen activates the panning,
while releasing it leaves sound sources
at their last position. If you prefer, you can
set a latching operating — touch to pan,
touch to stop.
At the centre of the plug-in window is
a representation of the soundstage and
the speaker positions. Here you select
either Cube and Sphere Modes, and the
display changes shape accordingly. If
you’re working with Ambisonics or VR,
the Sphere Mode might feel more familiar,
while for the more traditional sound mixer
the Cube Mode may be preferable. The
control layouts in each mode are different.
When in Cube mode, there are three knobs
controlling Left/Right, Front/Back and Up/
Down. In Sphere Mode there are two
knobs: Azimuth and Elevation. Normally,
elevation is from the speaker level upwards
but you can configure the panning to go
Cube mode, pictured here, is an alternative to Sphere mode shown in the main screenshot.
38 February 2022 / w w w. s o u n d o n s o u n d . c o m
below if the project and the output format
requires it. Click on the Advanced mode
switch and you get a few more options. In
both modes there are controls for Size and
Gain, the former adjusting the width of the
source(s). It’s also possible to lock individual
position-related parameters.
In addition to the two main modes,
Space Controller offers several options
for the way in which it handles the inputs,
these ‘panning modes’ being described as
Mono, Channels, Rotation and Symmetric,
and the input and output formats may be
set independently. Mono uses a mono sum
of the inputs as the source, while Channels
allows independent panning of the various
input channels. Rotate works on the entire
soundfield — useful, for example, in a movie
when the viewpoint is rotating — while
Symmetric Mode mirrors the panning
moves, so can be used to control the
width of the output. As an alternative to the
usual plan view option, you can switch to
a rear view, so as to get a visualisation of
the panning height.
The plug-in and app can also be
The phone app, which can connect to the DAW
configured so that multiple phones can plug-in via Bluetooth or Wi-Fi.
be used to control multiple plug-ins or,
for example, to control the left and right up and running it does precisely what it’s
channels of a stereo signal independently designed to do. To use Space Controller,
using a phone in each hand. It’s also worth the plug-in is inserted into a channel strip or
noting that when there’s no paired mobile bus insert, after which you launch the app
device, Space Controller can still be used as and make sure the plug-in and app are set
a more conventional 3D panner plug-in. to the same channel. Bluetooth connection
is as easy as you’d expect and in the case of
In Use Wi-Fi you can scan an on-screen barcode,
Space Controller is, as I’ve already said, which makes the process similarly simple.
really intuitive. So much so, in fact, that Set the centre position for your phone as
there’s really not a huge amount else to described above, choose your input and
say about the user experience; once you’re output formats, select Cube or Spherical
mode and, not forgetting to touch the
phone screen to activate the panning, you
then can start waving your arms about.
Sonically, the panning sounds smooth
and the Binaural output when heard
on headphones produces a believable
immersive experience — something that
came as no surprise, since it’s also the case
for the existing Sound Particles panners;
the USP here is the intuitive method of
control. Despite the complexity going on
behind the scenes, Space Controller is
one of the simplest plug-ins to operate and
certainly makes multi-dimensional panning
more fun... even if you do run the risk of
smacking your assistant in the face as you
pan left!
£ Standard version £77.22; Studio version
£311.22. Prices include VAT.
W https://soundparticles.com
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 IN T E R V IE W
IDLES
Recording Crawler
For their latest album, Idles guitarist Mark Bowen
adopted the role of producer for the first time.
TOM DOYLE
S
ince 2009, and over four
increasingly successful albums,
Bristol five-piece Idles have built
themselves a towering reputation as an
intense, soul-baring punk/metal outfit.
With their latest record, Crawler, the band
have expanded into more atmospheric
sonic territories, without losing any of their
inherent heaviness.
Idles’ guitarist and co-producer Mark
Bowen tells SOS that the aim for the album
40 February 2022 / w w w. s o u n d o n s o u n d . c o m
was to create a “dark warmth”, inspired by
the moody soundscapes of Portishead and
John Carpenter’s film soundtracks, while
maintaining what he describes as “doom
metal techno sludge in the low mids”.
Down the years, the band have worked
with various producers, including Space
(the Prodigy) and Nick Launay (Nick Cave
& the Bad Seeds). But, for Crawler, Mark
Bowen chose to step into the role himself,
working alongside Kenny Beats, the
American hip-hop producer whose credits
include Vince Staples and slowthai.
“It was daunting,” Bowen admits. “I was
worried about producing it on my own. So,
I got Kenny involved. He’s an incredible
producer. But he was just there to give me
confidence, really. I knew if I surrounded
myself with the right people around this
project, that it was going to be successful.
“Honestly, it was the most pain-free
recording process, and mixing process, that
we’ve had, from the audio standpoint,” he
adds. “Because I think that everyone we
were dealing with was open. Everyone was
listening, and paying attention.”
Having spent years on the other side
of the studio glass — as the guitarist
with his amps in the live room, trying
to capture his often brutally noisy and
aggressive sounds through the control
room monitors — Bowen admits he’s often

Photo: Tom Ham
Idles (from left): Joe Talbot, Jon Beavis, Adam
Devonshire, Mark Bowen and Lee Kiernan.
been left frustrated. “I’ve always been,
like, a production guy, and I’ve always
been interested in effects,” he says. “And
I’ve always been interested in how you
transfer what you’re hearing. How putting
a microphone in front of something doesn’t
do it justice a lot of the time.
“It’s always really bothered me, that
battle [laughs]. Especially as a guitar
player, I stand in front of my amps, and I’m
always like, ‘This sounds incredible.’ And
then you get into the control booth and
you’re like, ‘Aw… you’ve not found what it
was that was great.’
“So, it’s been a real journey across the
lifespan of Idles. We’ve been a band for
12 years, and we’ve been in a studio a lot
of that time. A lot of stuff just never got
released because we were struggling in
that process. There’s been a big, steep
learning curve.”
Out Of The West
Idles formed in Bristol at the end of the
’00s around the core of singer Joe Talbot
and bassist Adam Devonshire, before
adding Lee Kiernan and Belfast-born
Mark Bowen as guitarists, along with
drummer Jon Beavis. Starting out with
a series of EPs, culminating in 2015’s
Meat, the band mainly worked in local
studios, including Moles in Bath, and Bink
Bonk and the now-defunct State Of Art in
Bristol (owned at the time by Portishead’s
Geoff Barrow).
“My first experience of studios was
really through Idles,” says Bowen. “A lot
of it was kind of relying on the engineers.
You’re very lucky in Bristol because you’ve
got your kind of Massive Attack heritage
and your Portishead heritage. There’s a lot
of pretty wonderful engineers floating
about in the studios.”
But it was with producer Space that
Idles formed their most successful early
studio alliance, resulting in their first two
albums, Brutalism (2017) and Joy As An
Act Of Resistance (2018). The former
was recorded at Sugar Cane Studios in
Wandsworth, while the latter was done at
Monnow Valley Studio in Wales.
“We got on board with Space, who’s
a really open-minded producer,” says
Bowen. “He was the right guy for the bill.
He helped kind of engender and cultivate
that move into the heavier sound, the
more kind of blistering, chaotic sound. He
definitely was very proactive in exposing
that and pushing that with the band.”
In 2019, for Idles’ third record, Ultra
Mono, the band decamped to La Frette,
a residential studio on the outskirts of
Paris. There, working with Nick Launay
and his partner Adam Greenspan,
the band were afforded more time to
experiment with different methods of
tracking and overdubbing.
As the title of Ultra Mono suggests,
the idea was to further define — and
even narrow — the range of Idles’ sound.
“Joe really wanted it to be mono,” Bowen
laughs. “There was some negotiating done
there because it wouldn’t have worked.
“With Ultra Mono it was the first time
that production was going to take a role
in the song craft on the album. So, we
were writing these deliberately obtuse,
deliberately simplistic songs. Everyone
was kind of playing the same part and the
drums were bombastic.
“Everything kind of had to sound
singular. So, there were a lot of discussions
around that. The album was recorded in
less than two weeks, but it took about
four months to mix. It was about trying
to find that sound within the cacophony
a lot of the time. And that’s why the
album sounds like it does, y’know. It’s
supposed to be this abrasive, obnoxious,
singular-sounding thing.”
One of the great discoveries for Idles
at La Frette — a studio previously used
by Arctic Monkeys and Nick Cave & the
Bad Seeds — was that it wasn’t a posh,
upmarket residential facility: “It’s this
beautiful, massive chateau,” says Bowen,
“and because it’s got this faded glory, it
doesn’t feel stuffy. It’s very vibey, very
moody, and all the recording space is
downstairs. A bit dingy, but an absolute
banger of a [1973-built] Neve [A646]
desk. Lots of interesting mics, lots of
interesting, weird vintage gear. It definitely
adds a vibe to the recording process that
is useful.”
The basement drum room at La Frette
was formerly a brick-walled wine cellar, so is
particularly live-sounding. To control the kit
recording, Launay suggested to drummer
Jon Beavis that he try out a trick the
producer had learned when working with
Kate Bush in the early ’80s.
“It’s where you record the cymbals
separately from the drums,” Bowen
explains. “So, Jon was doing all his parts
and hitting a pillow whenever he was hitting
the cymbals, and then swapped everything
and he was hitting pillows and then hitting
the cymbals. Luckily, he’s a robot, so it’s
really straightforward to get him to do that.
“Then we set up in the other rooms. We
had a big bass room, and for the guitars, we
normally set up a big wall of our amps, and
then kind of pick and choose what ones are
being used.”
After the La Frette sessions for Ultra
Mono, Idles — looking to add hip-hop sonic
dimensions to their traditional rock sound
— first asked Kenny Beats to get involved
with them in an additional production
role. “The low end of the bass, the subs,
needed to be cleaned up,” says Bowen. “If
you listen to grime and drill and trap, the
low end’s very, very concise. It’s in a kind
of pocket, which guitar music just doesn’t
have. And also it was to make the drums
cut through and slap.
“It’s always this thing when you work
with guitar band engineers — the battle
w w w. s o u n d o n s o u n d . c o m / February 2022 41
 IN T E R V IE W
IDLES

The control room at Real World

Studios, where the majority of Crawler
was recorded.

between guitars and drums is

always won by the guitarist. And
then with hip-hop producers, it’s
the opposite. The drums always
supersede anything else. And we
kind of want to sit somewhere in
the middle. Kenny did a lot of drum
programming, triggering, put some
808s in and did some processing
of the bass.
With this fresh perspective
on the tracks achieved, the files
were sent back to Launay and
Greenspan, who mixed Ultra Mono
at Sunset Sound in Los Angeles.
“It showed them where we wanted
to go,” Bowen says of Kenny
Beats’ additional production. “It
went to this really extreme version
that sounded crazy. And then
Nick and Adam kind of processed what “Joe was talking about how he wanted “I did a lot of, like, guitar synthesis,
Kenny was doing.” to sing more,” Bowen adds. “So, that kind or was using the guitar as my primary
of led us down this more melodic route. oscillator, let’s say. That was going into a lot
Crawl To Arms Also, I think if you’re singing melodically, of Moogerfooger pedals, and I was using
Approaching the February 2021 recording you can’t really have that obnoxious CV to have those communicate with each
of Crawler at Real World Studios near Bath, music as the background to it. The idea other. Then using the Red Panda pedals
it was clear to Mark Bowen and the other was just to approach the drums and as well, the Particle and the Raster. So,
members of Idles that they wanted to make the instruments and vocals in a very like, granular delay and heavily modulated,
an album that sounded markedly different different way.” pitch-shift and delay.”
to Ultra Mono: “Very much so. We definitely Ahead of the sessions at Real World, In the end, elements of these demos,
didn’t want to retrace the steps of Ultra Mark Bowen demoed all of the tracks in his particularly the treated guitar sounds on
Mono. I think that the obnoxious chaos of North London home studio, using Ableton the songs ‘MTT 420 RR’ and ‘Progress’,
Idles had reached its zenith. It was starting Live, a Push 2 hardware programmer and ended up on the finished album. “What you
to become annoying [laughs]. Y’know, for Universal Audio’s Apollo Twin interface, hear is actually the demo of ‘Progress’,”
members of the band as well. along with a heap of effects pedals. Bowen says. “Those heavily granular delay,
pitch-shifted, weird sounds. When we got
to the studio, we just kind of thought, ‘Why
would you do anything differently?’”

Pressing Matters
Unusually, Bowen had drummer Jon
Beavis initially record the drum parts for
the majority of the tracks at Holy Mountain
Studios in Hackney, East London, before
having them pressed onto an acetate test
pressing. At Real World, Beavis played
alongside his own turntable beats, with
Bowen hoping that the fast-degrading
acetate would produce interesting sonic
results (as heard on tracks such as ‘Meds’
and ‘MTT 420 RR’).
“I was really hoping that after 100 plays,
it would be unrecognisable,” he laughs.
“But actually after 10 plays it kind of blows
out a little bit. You get this real warmth in
the low end, you get lots of compression,
Hip-hop producer Kenny Beats was enlisted to help Mark Bowen produce the album. and then it kinda stays the same.”

42 February 2022 / w w w. s o u n d o n s o u n d . c o m
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scan.co.uk/proaudio . 01204 47 47 47
 IN T E R V IE W
IDLES
A first set of drum recordings was captured at a studio in Hackney and then pressed onto acetate,
for drummer Jon Beavis to play along to at Real World (pictured).
Recording Beavis’s drums at Real
World, Bowen was keen to use very little
kit processing, preferring to set up “tons of
mics” and get the sounds at source. Beavis
was positioned in the studio’s Wood Room,
with the microphones fed to the Big Room
with its Neve 1073 preamps routed into its
72-channel SSL 9000K desk.
“We were moving mics when we
needed to rather than processing,” says
Bowen. “We recorded all the drums to
tape as well, which was a big thing. Ran
it quite hot, but not like crazy hot, onto
the tape.
Pedals played a large role in
shaping Mark Bowen’s guitar tone.
44 February 2022 / w w w. s o u n d o n s o u n d . c o m
“We had a mantra: ‘Hiss makes hits’
[laughs]. All your favourite albums are
hissy as hell. Like, you listen to Led
Zeppelin, to the Beatles, to Motown, it’s
filled to the brim with hiss. And I think
that’s part of the charm.”
In terms of drum-miking choices,
Bowen was partly inspired by Steve
Albini’s famously unorthodox methods.
“So, an Octava [MK-12] and a [Shure
SM] 57 on the snare. We taped them
together. Coles [4038s] pointing on the
floor. We used the AEA [R88] split stereo
mic in front of the drums, and this other
really battered AEA as a room mic, up on
the balcony, with Schoeps [CMC6Us] up
there as well.
“I love a distorted room sound of heavily
compressed drums. But adding all those
processes creates a limitation in the sound.
You’ve got to do a lot of carving and so we
wanted to create that in the acoustic side of
things as we were moving from acoustic to
analogue. I think we had like 22 channels of
drums or whatever, all running into tape.”
When a more distorted, processed
sound was needed, the team used the
Thermionic Culture Vulture valve rackmount
unit. “We had this really ridiculous
setting on the Culture Vulture that was
our distortion sound,” Bowen says. “For
compression, we used the Valley People
[610] compressor and the [Empirical Labs
EL8] Distressor.”
All Your Bass
For Adam Devonshire’s bass parts, the
band worked with a fairly basic setup.
“Two amps... dirty, clean, and both dirty,”
Bowen explains. “There was a ’70s Hiwatt
100 that was absolutely stunning that they
had at Real World. So we had that, and
we intermittently changed between Dev’s
Fender Super Bassman and another Hiwatt.
We used a [Shure] Beta 52, a [Sennheiser]
MD 421 and a Beyer M80 on the bass. Then
two DIs, one clean, one dirty. We ran one
hot into preamps and kept one really clean.
“Again, it was always first principles.
So, we were in the control room and if the
bass didn’t sound right, we were running
 IN T E R V IE W
IDLES

Joe Talbot plays the

Philips Philicorda, which Bowen
says “looks like a piece of
hipster furniture...”

out to the amps, changing

the settings, changing the
position of the mics, and
then just balancing in the DI,
but trying to avoid as much
as possible carving the shit
out of everything. Everything
took a lot longer, but it
sounds a lot better.”
One standout bass
sound opens Crawler :
the low, synthy tone that
underpins ‘MTT 420 RR’.
“That bass synth sound is
actually transposed guitar,
using the Moog Freqbox,
and a POG [Polyphonic
Octave Generator],” Bowen
points out. “And then also
just pitch-shifting it in
Ableton, using the transpose… not even,
like, AlterBoy, or some swish kind of plug-in.
The terrible, terrible processing in that
Ableton transpose, where you get these
weird wobbles and glitches, I love that.
“It’s hard to get bass guitar to sound like
that. And it’s also difficult to get that plucked
guitar feel. It fits that John Carpenter/
Portishead vibe but stays within the Idles
punk rock band ethos.”
Guitars
Elsewhere, in the guitar department,
things get a bit more complicated with
Idles. Although normally a Fender Strat
or Mustang player, on Crawler, Mark
Bowen mainly used the Electrical Guitar
Company’s EGC500, with its aluminium
resonator plate. “Super hot, super resonant
because of the aluminium thing,” says
Bowen. “You can get really weird sounds.
Especially when you’re going into DI, the
charm of the guitar comes through.”
During the making of Ultra Mono, Bowen
also began using baritone guitar, having
tried out a Fender Bajo Sexto. “That got
me really hooked on baritones,” he says.
“So, Fender made me one. I don’t like Tele
bodies, so I had it as a Strat. Then I used
Hiwatt amps, mainly. I also got into using
a Sunn Model T preamp, going into the
Orange AD200. You’re stood in front of it,
going, ‘This is AC/DC... come on!’”
Idles’ other guitarist Lee Kiernan
typically uses Marshall rigs, though on
Crawler expanded out into a Fender Twin
Reverb and a ’70s WEM. “Just anything
that would be caustic and bright,” says
Bowen. “A hangover from Ultra Mono that
we did again on this album was using
straight-to-desk DI guitars.
“A big thing that Lee did on this album
was he kind of played a lot of snare
rhythms, and a lot of percussion rhythms.
And then as always with Lee, he was the
top end violence that you get on certain
songs, like ‘New Sensation’. He’s very
prominent there.”
Bowen in particular is an avid collector
of effects pedals, with his current
favourites including Adventure Audio’s
Dream Reaper (for fuzzy, modulated
feedback) and Death By Audio’s
Waveformer Destroyer distortion.
“Dream Reaper is kind of
a feedback-scape carver. So, basically you
can create a lot of harmonic overtones in
your feedback. You can really manipulate
it with the filter. I’m still obsessed with
the Death By Audio pedals, like the
Reverberation Machine. If you ever hear
a reverb on any of my guitars, that’s what’s
happening. And the Echo Dream, especially
the modulation, is still my go-to.”
The only keyboards to feature on
Crawler, in the brooding, slow-building ‘The
Beachland Ballroom’, were Mark Bowen
playing Real World’s Yamaha C3 grand
piano and Philips Philacorda organ.
“It looks like a piece of hipster furniture,”
he says of the latter. “Everyone gravitates
towards it. You’re sat in front of the desk,
and if you go and sit on the sofas at
the back, it’s right there in your eyeline.
In-between takes, everyone’s going over to
it. I just started playing it and it really kind of
had that Motown feel to it.”
Vocals
When it came to recording singer Joe
Talbot’s whisper-to-scream vocals, the
process was perhaps unsurprisingly an
intense one. Bowen credits Kenny Beats
with having the skills and patience to
capture Idles’ hyperactive frontman’s
performances on record.
“It was new territory for us because
Joe was singing. Joe is the most impatient
human being in the world in the studio.
And he’s also the most hard-on-himself
performer. So, he does a line, stops,
screams in furious anger, goes back, does
the next line, stops. And that process he
wants as quick as possible. So, while he’s
doing his screaming, he wants you to get
back to the line and do the next line.
“Kenny works with rappers and watching
him process and comp vocals as he goes
is one of the most incredible things. That
was novel for us, having the recording of
the vocal process run smoothly. Basically,
everyone left the room and left those two to
it. And every time you went back into them,
they were just arguing with each other.”
Mic-wise, for the louder vocals, Talbot
used a Shure SM7b. “Normally we’re trying
to find a mic that will take his barking,”
Bowen says. “And so on the barkier

46 February 2022 / w w w. s o u n d o n s o u n d . c o m
passages on this album, we’re using an
SM7, that he can hold in his hand, which
is a very useful thing for a singer like him.
He is very dynamic in his movements. If we
can get him physically moving around and
gesturing, you get a lot more out of Joe.
He’s very physically involved in his singing.”
For the softer or more melodic vocal
tracks, the team attempted to use a range
of vintage Neumann U67s and U87s on
Talbot. “We broke maybe, like, £50,000
worth of microphones at Real World,”
Bowen laughs. “It was so funny. They’re so
old, they’ve got dust in them, and he’s so
fucking loud. Like, he is the loudest vocalist
I’ve ever heard. So, like, 60 years’ worth of
dust is just getting bashed into this capsule.
“And again, we were using Distressor
a lot with Joe’s vocals, because he’s
a hugely dynamic vocalist. He really wants
his quiet moments, and he really wants
his loud moments and it’s very hard to
negotiate that with his mics. Those vintage
Neve preamps that they’ve got at Real
World are stunning, into the SSL, which
is kind of just accepting everything you
throw at it.”
Bowen singles out “one amazing
moment” during the recording of Joe
Talbot’s vocals on Crawler, when late at
night Kenny Beats handed the singer
a Shure 520DX ‘Green Bullet’ mic (typically
used for harmonica) to allow him to
improvise over the bare bones of the
instrumentation of ‘MTT 420 RR’.
“It was really dark, February in Real
World, y’know, bay windows everywhere.
We looped the song and he just started
doing all these weird passages and
finding what he was going to do in the
song through that. And we just kept all
those loops in there. If you listen to that
song, in the background, there’s lots of
very warbly passages and Joe kind of like
running through where his range fits in the
song. Very, very cool. Really big moment.
Everyone got very ‘in’ on the song at
that point.”
Finishing Touches
After all of the tracking was done, Kenny
Beats and Mark Bowen spent a lot of time
editing and working on the grooves within
the files. “We really worked hard at the
end of finishing recording,” says Bowen.
“We did a lot of editing where we got the
drums and the bass to sit exactly where we
wanted them to. Kenny is very much about
the ‘pocket’. Y’know, he comes from a place
of funk and hip-hop. Dev’s always way too
far in front of Jon, so he moved Dev a little
After damaging a number of vintage valve microphones,
singer Joe Talbot eventually settled on a Shure SM7b.
bit further after. And we did the same with
the guitars. I’m miles ahead of everyone else
all the time, because I’m just thinking about
everything else to do with the song.”
The mixing of Crawler was then handed
over to Craig Silvey (Portishead, Arcade
Fire, the Horrors). “My God, talk about a mix
engineer,” Bowen enthuses. “It’s an absolute
joy to watch him work. He works all in
analogue, and he’s got this Avid controller
that’s controlling all his parameters, his
barrage of outboard. He does everything
on the fly.
“We’d worked a lot on the end of session
mixes. We really wanted Craig to have a very,
very firm sense of what the songs were and
how we wanted them to sound. He listened
to the end of session mixes, and just made
an infinitely better version of those.
“His approach to balance is mind-blowing.
His mixes are so well balanced in the
positioning of where things are in terms of
the EQ, and where things are in the stereo
scape. Just the positioning of guitars and
instruments and the flow and movement.
“But again, we were doing a lot of that
stereo manipulation in the studio ourselves.
Y’know, a lot of the stereo scaping that we
were doing on ‘MTT 420 RR’ was preserved
in the mix. Craig was able to kind of clear up
a lot of that.”
Bowen believes that the big difference in
the mixing processes between Ultra Mono
and Crawler was his and the band’s growing
knowledge of recording. “Oftentimes, our
songs have been transformed quite a bit
from the studio through the mix process.
Sometimes that’s been vitally needed. A lot
of Nick Launay’s mixes really improved our
sound and really captured the chaos of the
band, by pushing the faders and moving
things around.
“Whereas with this album, there was an
accomplishment to what we had achieved
in the studio, and a sense of confidence in
just having done that. And I think for the first
time we had that translated in the mix.”
From here, the road ahead is clear for
Idles, in terms of the areas they choose
to explore creatively and sonically. Mark
Bowen, for one, has ambitions to push
rock music into new and unexplored
territories. “I think now that I’ve kind of
grown in confidence as a producer, it
means that we can write a hell of a lot
more with production in mind,” he says.
“Again, my goal is to reduce the steps from
my head to your ears. I want to make the
mode as coherent a pathway as possible.
I’m really interested in this idea of us
setting up our own studio and creating our
own space.”
A key inspiration is Radiohead’s Kid
A, even though it might not provide an
absolute clue as to the future sound of Idles.
“Kid A was that moment where DAWs
were coming in,” Bowen points out, “and
you can hear that translation in things
like ‘The National Anthem’ with Jonny
Greenwood carving stuff up in Pro Tools.
There’s scope for a band to do that again
— y’know, a snapshot of where we are now
with music technology. But using guitars as
the primary oscillators. That’s my vibe.
“I want to push the boundaries of what
we can do as a band without ending up
sounding like a prog band. Still maintaining
that rock & roll and that punk ethos, that
hip-hop ethos, but translating that across.
That’s my next goal.” 
w w w. s o u n d o n s o u n d . c o m / February 2022 47
 ON TE S T
ExpressiveE Imagine
ROBIN BIGWOOD
I
magine is the offspring of two innovative
companies, Expressive E from France
and the Canadian Applied Acoustics
Systems. The former are probably best
known for their Touché gestural control unit
for synths, and the (at the time of writing still
much anticipated) Osmose MPE synth. AAS,
meanwhile, are old hands at the physical
modelling game: their Lounge Lizard electric
piano was one of the first really successful
implementations of the technology as
a plug-in. These days the product range
includes modelled guitars and synths too,
and Eurorack hardware tie-ins.
With forebears like these Imagine
should be interesting and unusual, and
it is! It’s essentially an acoustic modeller
of imaginary instruments, based on the
sound production behaviour of skins (in the
drum sense), strings, bars and tubes. Two
sound-producing modules can be layered
in each preset and their outputs treated
by a bank of modulatable effects. A macro
system lets you tie multiple parameters
48 February 2022 / w w w. s o u n d o n s o u n d . c o m
Software Instrument
Imagine uses physical modelling to let you create any
acoustic instrument you can, er, imagine.
together for easy simultaneous modulation,
and it’s easy to set up real-time control
with a Touché or more conventional MIDI
controllers. The graphic interface is clean
and invites experimentation. It all looks
fascinating, and promising, but how does it
work out in practice? And more importantly
perhaps, what sounds can you get out of
this thing?
Giving Me Excitations
Let’s start with an overview of Imagine. The
heart of every preset is its pair of instrument
layers, shown next to each other in the
main Instrument plug-in page. These have
identical capabilities, can be individually
enabled, and have their own independent
level, pitch (+/- four octaves, in cent
steps), release time and pan positions. In
conventional synth terms you might regard
each as a complete polyphonic ‘voice’ but
there’s very little that’s conventionally synthy
about them.
Each layer/instrument uses one of
three methods of virtual excitation: either
mallet-struck, noise-actuated or sequenced.
Ten categories of material combinations
can then be explored; pairings of skins,
bars, tubes and strings. For each of
those 30 excitation combos a number
of pre-configured starting points are
provided, often with descriptive and fanciful
portmanteau names such as ‘Dustophone’,
‘Hurdy Tuby’ and ‘Harpastral’. The names
sometimes give a flavour of how they’ll
sound and behave, but there’s scope for
altering any of them quite radically: more
of that in a minute. What’s more, there can
be huge variation between instruments
within a single category, and equally lots
of overlaps between categories, so there
are no hard and fast rules. Each instrument
is, I’d guess, a product of hundreds of
under-the-hood acoustic modelling
parameters that would be too overwhelming
and complex to expose to the end user.
Regard them as very flexible, almost
waveform-like mini-presets, and you’ll be on
the mark. Also, be aware that conventional
instruments (eg. marimba, bass guitar,
violin) are conspicuous by their absence.
You might find a way to create convincing
versions of these, but that is not really the
point of this plug-in. In general too, the
character of sustaining instruments leans
much more towards ensemble than solo.
Alongside this main sound-generating
side of Imagine is a complex parameter
modulation system, visible at the bottom
of the window as four macro knobs and an
associated MSEG (multi-stage envelope
generator) for each. And also a whole
separate Effects page, with an extensive
signal flow via a vibrato/frequency shifter,
two configurable ‘expressive fx’ (which
encompass filters, distortion, chorus and
more), a tilt EQ and compressor, and finally
a delay and plate reverb. The on-board
modulation is just as valid for use here as
on the Instruments page, by the way, and
remains visible.
As you might expect, an Imagine preset
loads (instantaneously) the whole shebang:
both instrument layers, modulation and
effects. Somewhere around 200 are
provided, organised into familiar groupings
of Bass, Lead, Pad and so on. A nicely
implemented, very easy-to-use MIDI Learn
facility operates at a global level, in the
Expressive E Imagine
€139
PROS
• A genuinely unique, distinctive
sound, with little overlap into sampled
or analogue territory.
• For the most part, a beautifully clear
user interface.
• Strikes a fine balance between ease
of use and tweakability (with most
of the university-level mathematics
hidden safely away).
CONS
• Requires the latest CPUs for the best
experience and sound.
• Non-retriggering modulation
envelopes.
• No control over velocity response.
SUMMARY
An inspiring and unusual source of all
sorts of unclassifiably clangourous,
breathy, twanging and noisy sounds,
from ethereal and luminous to heavy
and percussive.
individual instantiation in your DAW, with any
hardware control assignments you make
unaffected by loading new presets. There’s
also a Main menu that gives access to
preset management functions, an init patch,
and an online PDF manual.
It’s Easy If You Try
Without wanting to pre-empt the main
findings of this review, Imagine would
have a lot to offer even as a preset-only
virtual instrument. But in fact, digging in
and tweaking presets, or starting new
sounds from scratch, is easy and inviting:
surprisingly so maybe, given the potential
complexity of physical modelling.
Looking at the instrument layers first, it’s
quickly apparent that different parameters
appear according to the type of ‘excitator’
(which I’m not totally sure is a word, but
I rather like it...). And it’s remarkable just how
few parameters there are.
Mallet instruments, which are intrinsically
decaying in volume and brightness,
offer adjustable shine (brightness), mute
(damping), position (of strike) and impact.
This last parameter continuously varies
the virtual mallet from a hard-headed stick,
through a felt then soft mallet, and finally
a metal brush.
Noise instruments, which can sustain,
get adjustable colour (brightness) and
resonance (a slightly different version
of damping, and nothing to do with
analogue synth filters). The grains control
gradually blends the sustaining character
from a continuous noise to a randomised
flow of discrete events; it’s a granular
synthesis term of course, and absolutely
takes Imagine into that sound world. An
The instrument
browser lets you
quickly explore
excitator and material
combinations. The
instruments are
all factory presets,
and fair game for
further tweaking.
LFO with nine
waveforms varies
harmonic content
in the system
(a sort of tremolo
on steroids),
and there’s
a per-voice ADSR
too. Interestingly
the release
stage of that
affects different
components of the sound than the main
layer release parameter.
Sequence instruments are a little
different. In fact the ‘sequencing’ aspect
consists of two fairly standard bpm-sync’ed
arpeggiators combined with a 16-step
playback system. You can choose shorter
cycle lengths, different time divisions, and
disable steps along the way to create
repetitive rhythmic patterns. The main point
is that, just like other normal arpeggiators,
sounds are triggered by enabled steps
during playback, and only indirectly from
MIDI input. Sound design is boiled down
to just two generic timbre parameters
which interact to simultaneously control
brightness, character, decay and more.
A randomiser knob varies velocity and strike
position from note to note. There’s also
a useful chord/restrike mode which repeats
and pulses notes you hold, and an unusual
Arp Select parameter that seamlessly
switches which arpeggiator is in use, even
mid-cycle (only one can be active at a time).
You can modulate that parameter though,
or control it via MIDI, for some complex
sequence-like effects, or simply to achieve
patterns longer than 16 steps.
So far, so intuitive, and some new users
of Imagine might feel they’ve got the
whole thing licked after about 15 minutes.
The one parameter that can cause some
head-scratching though is the Expr[ession]
knob in each instrument’s header panel. To
cut a long story a bit shorter, Expr is pretty
much essential for sequence instruments
to get much out of them: it becomes
a sort of ‘energy’ and volume parameter
here. For the other instrument types it
adds continuous noise excitation, so you
w w w. s o u n d o n s o u n d . c o m / February 2022 49
 ON TE S T
EXPRESSIVE E IMAGINE
can make mallet instruments sustain,
for example. It can also add additional
harmonic content and volume to noise
instruments, though it’s nothing that you
couldn’t already achieve by playing them
with higher velocity values. I found this
arrangement a bit unhelpful: I often don’t
want pads and other sustaining sounds
to be velocity sensitive, but in Imagine
noise-excited instruments always are
whether you like it or not. There’s no
velocity sensitivity parameter anywhere,
and I think there should be. The problem
with Expr, too, is that it’s a somewhat blunt
tool, because it hoses natural decays and
ADSR shapes to leave sounds with crude
organ-like envelope characteristics. Still, it
can do some nice things when modulated,
and has a role in real-time MIDI control,
driven from your modulation wheel, say, to
provide expressive swells. Still, I’d love to
see some further refinement here, certainly
to the velocity response behaviour, and also
a more thorough discussion in the manual.
Above Us Only Sky
I already mentioned the extensive effects
provision: here’s some detail. The Vibrato
and Freq Shifter section, the first step in the
effects chain, is unusual. It’s there to provide
basic pitch-bend response on the one
hand (normally the left, cough...), and the
Frequency Bend parameter is permanently
tied to your MIDI controller’s pitch wheel.
On the other hand, it’s the only dedicated
way to achieve vibrato. Now what, I hear
you cry, is this doing masquerading as an
effect, rather than being implemented at
the oscillator or voice level? Well, I’m not
sure that it’s not in both places at once, so
to speak, because pitch-bend response
across different instrument types varies
considerably, and is often subtle and
complex. Sometimes we hear only some
harmonics in a sound bending, while others
stay at a fixed pitch. Then there’s the fact
ALTERNATIVES
There’s a pretty huge overlap with
an existing AAS modelling plug-in,
Chromaphone 3, which seems to
be built around many of the same
concepts but offers more sound design
parameters in lieu of the ready-rolled
instruments. I’d be prepared to bet
Imagine uses a similar (if not identical)
underlying engine. You might also look at
Pianoteq 7 Standard or Pro, which can’t
offer any noise-excited timbres, but via its
morphing feature, and ideally using a few
Chromatic Percussions instrument packs,
can cook up novel percussive sounds.
50 February 2022 / w w w. s o u n d o n s o u n d . c o m
Imagine... Owning A Super-computer
Like many new, cutting-edge instruments,
Imagine is heavy on the CPU. On my 8-core i9
MacBook Pro, with a 128-sample buffer size in
Studio One, many noise-excited patches could eat
40-50 percent of the processor with just a few
held notes. And this is with the instrument layers
running in ‘regular’ rather than the more airy,
treble-extended ‘superior’ mode, which added
4 or 5 percent more. Conspicuously, eight notes
that the bend range is expressed in arbitrary
units rather than the more usual semitones,
and some instruments have strict limits on
how far they’ll bend (sometimes no more
than a major third, even at max range), or
might bend further down than up. Intriguing
stuff. Vibrato is more conventional and
predictable in its effect, but is about as
simple as you can get. There’s no option
to have a delayed onset, there are no
waveform choices, and undulations affect
all voices simultaneously and equally.
However, you can achieve more varied
vibrato effects, including many different
waveshapes and even pseudo-sequenced
patterns, by modulating the Frequency
Bend parameter, with the proviso that it’s
still a mono/global effect.
Moving on, the two ‘expressive fx’
slots can be filled with a tremolo, resonant
low-pass and high-pass filters, autopan,
distortion, a ‘noisifier’, ring modulator,
three flavours of phaser and a funky
chorus. It’s nice to see the filters, and they
sound good in a smooth 2-pole sort of
way, but again whilst you can modulate
their cutoffs to achieve key-sync’ed filter
sweeps it’s not a per-voice implementation,
and even a simple paraphonic response
can’t currently be achieved (see the
Macro-managing
box for more on
this). All other effects
are of high quality,
including a decent
stereo chorus and
genuinely useful
ring modulator,
and I particularly
liked the noisifier.
Given how hazy
many noise-excited
sounds can sound
by themselves,
Sequence
instruments are actually
built on dual step-based
arpeggiators.
of polyphony is the maximum you’re allowed to
have per plug-in instance, with six, four and two
the other options. Individual layers can also be
run monophonically. There’s a global ‘eco’ mode
that loads up its own complement of presets, and
these for the most part do indeed run with about
half the CPU use of their ‘main’ counterparts, and
still sound good. Bottom line: it’s possible to work,
but let’s all keep saving for that M2X Mac Pro...
adding this can make for positively sooty,
smoke-filled textures.
As for the Tilt EQ, it’s a sensible choice,
and one of the quickest ways to warm up
or brighten a sound with just one knob. The
adjustable ‘pivot’ freq adds to the flexibility.
The Compressor/Limiter is a simple
two-knob design and seems more capable
of dialling in a bit of thickening or pinging
initial attacks than thunderous squashing.
Finally, the stereo delay and plate
reverb are excellent. The delay is not
over-complicated, but can be made to
cover a broad range of analogue and digital
styles, with or without modulation, and offset
L/R ping-pong effects. The plate is a good
choice: it’s not naturalistic, but can sparkle
or boom in a very attractive way, and with
variable stereo width.
Imaginary Friend
In a world seemingly now overrun with
analogue synths jamming away repetitively,
I found Imagine to be a breath of fresh air.
There’s little else like it. It’s unashamedly
synthetic, yet the sounds that come out of it
often have the aliveness and organic quality
of acoustic instruments. You’ll struggle to
get it sounding like a clean, buzzy string
machine, or an orchestral string section, and
 ON TE S T
EXPRESSIVE E IMAGINE
even less a Minimoog, but instead it has
a huge timbral gamut of its own. Strange
new idiophones and twisted bells are
a given, varying in their degree of chaos and
Macro Managing
Modulation in Imagine is done via a ‘macro’
system. There are four independent macros
— essentially multipurpose modulators — each
with its own multi-segment envelope generator
which can pull off a vast range of LFO, envelope
and step sequencer-like shapes and run them in
looping or one-shot fashion. You can set them
up yourself with mouse drags and visual menus,
or choose from one of dozens of preset shapes.
Any number of parameters can be modulated
by a single macro, and you set up value ranges
and polarity visually using mouse drags around
parameter dials. There’s live feedback of
modulation as you play too — parameter value
indicators zoom around.
Fundamentally it’s a great system, powerful
and intuitive. It can even support custom value
mappings for external controllers (like Expressive
E’s own) via a ‘manual’ mode. But there are
currently some flaws. The main one, for me, is
that there’s no retrigger option, to allow one-shot
shapes to fire for every MIDI key press and not
just the first, when playing held broken chords
for example. I can’t see any reason why this
couldn’t be implemented easily, and it would
allow for useful paraphonic triggering of the
global frequency shifter, the Expressive FX filters
and more. Another is that there’s no quick way to
52 February 2022 / w w w. s o u n d o n s o u n d . c o m
inharmonicity, but the repertoire extends
easily to harp and hurdy gurdy, bowed
metallophones, and sounds reminiscent of
(though still distinct from) complex Asian,
see which macro is controlling what — perhaps
colour-coding could be an option here. And
there’s also weirdness in the way that all the
modulation shapes are defined on a 16-step
tempo-sync’ed grid, and some of the presets
(particularly the simpler LFO shapes) load with
only one or two steps stretched out across the
entire grid. Consequently they run very slowly
even at their maximum rate. It’s as if there’s
a parameter missing that governs whether steps
have an absolute or relative duration value.
Strange, though not serious, because it’s easy
to fix by redrawing the envelope on a smaller
timescale.
Just as useful really is the MIDI Learn feature.
This allows most parameters to be controlled
by external MIDI controllers (via CC, pitch-bend
and aftertouch messages) with definable value
ranges and polarity. A nice touch is being able
to save a default setup of your own, which
saves having to continually redo assignments
you use over and over again. Owners of Touché
controllers will see lots of ready-mapped presets
in the Lié software manager, by the way, which
have the four axes of touch often associated with
the four macro/MSEGs.
Imagine’s envelope generators easily cover
complex LFO territory too.
What’s a synth these days without a big dollop
of effects? A powerful and decent-sounding chain is
on offer here.
African and Indonesian instruments. Then
there are the breathy beds, noisy, twinkling
pads, and any number of otherwordly
atmospheres. It can be a tremendous
source of sequenced drum sounds too: the
perfect antidote for an overdose of 808
(and the like). It’s not given to really chaotic,
unstable sounds, I’ll hasten to add: nearly
always things remain musically oriented.
As to who might use Imagine, well it
probably wouldn’t be the first thing I’d turn
to for my next techno hit (yeah right...).
Even less for the next cover-band wedding
gig. However, for a lot of acoustic-leaning
drum and percussion programming, for
Afrobeat ostinatos, twinkly pop filigree,
brooding cinematic atmospheres and
beds, pad sounds to be used alone or
layered with synths, it’s a great source of
semi-serendipitous inspiration.
I remember decades ago salivating at
the prospect of the sonic chimeras that
I might be able to conjure from innovative
hardware modellers like Yamaha’s VL70M,
but pretty quickly learned the truth: if you
thought FM synths like the DX7 were
difficult, this was a whole new world of hurt.
So for me (at least) a synth like Imagine
is a promise that has finally come good.
It’s a genuinely creative, controllable
modeller that delivers whether you like to
merely load and tweak presets, or work up
sounds from scratch, and has a sound all
its own.
£ €139 including VAT.
W www.expressivee.com
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 ON TE S T
Zaor MizaZFlex&GriprackMkII
SAM INGLIS
I
f you think about the items that are
essential to any studio, the first things
that come to mind are probably mics,
speakers, keyboards, recording software
and hardware, and so on. But there’s
another class of things that are just as
vital, and easy to overlook when you’re
planning a new setup. The fact that they
don’t themselves produce or capture
sound doesn’t make cables, stands,
lights or acoustic treatment any the less
important — and the same is true of
studio furniture.
Back in the day, when studios were
based around consoles costing six
figures, any furniture and racking would
typically have been constructed on site
by a carpenter as part of the build. That’s
still the preferred option today, but it
does have some disadvantages. Having
54 February 2022 / w w w. s o u n d o n s o u n d . c o m
The assembled Miza Z Flex desk with two Griprack 4 MkII units perched on top. It won’t stay this tidy for long...
Studio Furniture
Zaor’s Miza range
promises smart, affordable
self-assembly furniture to
suit every studio.
furniture custom made is costly, can
be slow, and means you end up with
something that can’t easily be repurposed
when you move to another space. At
the other end of the spectrum, domestic
and office furniture is often pressed into
service in a home-studio environment.
This is usually the cheapest approach,
but kitchen tables and computer desks
are not ergonomically or acoustically
optimised for the needs of the studio
engineer, and are unlikely to have rack
slots or speaker stands built in.
Fortunately, there’s a middle way, and
it’s catered to by manufacturers such as
Zaor, who offer self-assembly furniture
targeted at home and professional
studios. Zaor will undertake custom
commissions and adaptations, but in most
cases, buyers choose from a palette of
options with different dimensions and
features. Your chosen furniture is then
shipped in flat-pack form from Zaor’s
factory in Romania, ready for you to put
together at your leisure.
All Fours
The Zaor catalogue features four main
ranges. The Classic line includes racks
and speaker stands, plus desks aimed
both at ‘in the box’ and console-based
control rooms. Most are built from
solid wood, and as the name suggests,
the design aims for a timeless, classy
appearance. Those seeking a more
modern and stylised look can turn to
the iDesign line, which offers a striking
minimalist aesthetic. Both of these are
premium ranges, which are likely to find
their way mainly into professional and
high-end project studios. For the rest
of us, Zaor also offer the Vision and
Miza lines.
The desks and racks in the Vision
range share something of the same
‘designer’ aesthetic as the iDesign
products, with a strong black-and-white
colour theme and bold use of triangle
shapes. But the widest range of Zaor
furniture is to be found under the Miza
name. There are no fewer than 15 Miza
products at the time of writing, including
desks, speaker stands and racks. The
general vibe is modern, but in quite
an understated way compared to the
iDesign or Vision models, with smooth
curves taking the place of sharp angles.
Two colour schemes are available:
‘wengé grey’ and black with cherry
wood highlights.
Getting A Grip
Strangely enough, in over two decades
working at SOS, the only purpose-built
piece of studio furniture I’ve ever owned
is a small wooden studio rack built by
our esteemed Paul White. Most of my
gear has been housed in portable racks
for location use, or perched on top of
repurposed kitchen tables, office desks
and so on. That has always worked OK,
but it brings its share of annoyances. It’s
impossible to keep cables tidy, or easily
access a MIDI keyboard and its QWERTY
counterpart at the same time; and
perfecting monitor placement is always
a challenge, whether you try to perch
them on the table or use separate stands.
Consequently, I was very willing to
try out some of the options from Zaor’s
Miza range. These include desks with
integrated racks and/or speaker stands
as well as ‘separates’. I quickly decided
that the latter would be better options,
and because I need to swap out pieces of
gear quite often, I was particularly taken
with the concept of the Miza Griprack II:
a free-standing rack case available in 2U
and 4U variants, which is designed to
sit on top of a desk. As well as offering
easy access to the rear, the Griprack also
features a special foam damping pad
on top which is purposely designed to
support monitor speakers.
I decided to pair two Gripracks with
the biggest flat-topped desk in the Miza
range, the Z Flex. This offers plenty of
surface real estate for positioning the
The Griprack features a foam pad
on top that is designed to support
monitor speakers.
Gripracks either side of a computer call ‘system bolts’: metal bolts with an
monitor, as well as offering useful features ‘eye’ into which cone-shaped screws are
like a large cable channel and a slide-out driven to fix a connection. You’ll need
keyboard tray. For disclosure, I should a crosshead screwdriver, and a rubber
point out that manufacturers usually send mallet or hammer is useful to tap in
review gear to SOS on loan, which is then the dowels. Clear printed instructions
returned after the reviewer has done his are included, and also available as
or her work. That’s not feasible in the PDFs on the Zaor site along with
case of studio furniture, so it was agreed construction videos.
in advance that the Zaor items would
remain with me after the review period.
Zaor Miza Z Flex &
Desk Jockeying Griprack 4 MkII
The review furniture arrived rather £669/£209
unexpectedly one Thursday afternoon
PROS
on the back of a small truck (it was lucky • Plenty of choice.
I was in, really). The driver helped me • Easy to assemble.
manhandle it into the building, which • More affordable than
is good, because the larger of the two custom-made furniture.
• Smart appearance.
Z Flex boxes weighs 71kg. Unpacking
• Some nice ergonomic
the boxes revealed that everything was design touches.
present and correct. Affordability is the
raison d’être of the Miza range, and Zaor CONS
• As you’d expect at this price, the
have evidently achieved this through two
materials are not as robust as solid
main goals. First, there are almost no wood or ply.
metal components, at least in the items • Although the height of the slide-out
I tried. Second, the panels are all made keyboard tray can be varied, that of
of ‘furnitureboard’ or MFC rather than the desk itself can’t.
• Monitors placed on top of Gripracks
solid wood.
can end up too far away from
If you’ve ever put together a piece of the wall.
furniture from IKEA or similar, you’ll be
familiar with the construction principles SUMMARY
Zaor have applied a cost-effective
used in the Zaor Miza range. Pre-drilled
‘flatpack’ model to create furniture
holes need to be populated either with that is optimised for the studio at
small wooden dowels or with what Zaor reasonable prices.
w w w. s o u n d o n s o u n d . c o m / February 2022 55
 ON TE S T
ZAOR MIZA Z FLEX & GRIPRACK II
It took me a little over an hour to
put together the Z Flex desk (and much
longer to clear the space in the studio!).
Most of this can be done by one person,
but you’ll definitely need a helper for
the final stages, especially putting the
top onto the frame. There’s an element
of customisation, in that you can select
one of several heights for the pull-out
keyboard tray. I got this the wrong way
around on my first attempt and ended
up with the tray on its lowest rather than
highest setting, but that’s mainly down to
my own lack of handiness.
The Gripracks are trivially simple
to assemble, and even come with the
dowels pre-inserted into the holes. They
are designed to sit at an angle to the
desk, and this can be varied by screwing
in the feet at the front of the rack to raise
or lower it. A nice touch is that you get
a large pack of rack bolts and washers
along with the components needed to
build the actual rack itself.
Flexible Friends
With everything in place, my studio
certainly looked smarter and more
professional than it had before. The basic
nature of the materials isn’t apparent in
the assembled desk, which looks and
feels very solid. Only time will tell how
long this feeling will persist, but I see no
reason why it shouldn’t last as long as the
desk isn’t subjected to ill treatment. (The
wengé grey desktop is quite vulnerable
to liquid marking, so be careful with those
coffee cups.) My experience is that this
sort of self-assembly furniture doesn’t
like being repeatedly disassembled and
then put back together again, but Hugh
Robjohns reports that his seven-year-old
Miza 88XL still looks and performs as
good as new despite having undergone
this process at least once.
Ergonomically, the Z Flex ticks a lot of
boxes. There are discreet cable ports at
both rear corners, and these open onto
a capacious channel with Velcro cable
tidies. Two further ports lead from this
to the back of the keyboard tray, which
slides in and out smoothly enough on its
metal runners. I only have a lightweight
USB keyboard but I’m confident this
would happily support a hefty 88-note
master keyboard if needed — it’s rated at
40kg. Alternatively, it could be very useful
for other gear that needs to be within
easy reach.
The Gripracks are, if anything, even
smarter than the desk, and the ease
56 February 2022 / w w w. s o u n d o n s o u n d . c o m
The open back of the Griprack
makes access to rear-panel
sockets refreshingly easy.
of access to the rear for plugging and
unplugging is a revelation. The adjustable
angle is also great from the point of view
of having optimal control over the gear
in the racks, and I had no trouble fitting
two nearfield monitor speakers on top of
each rack.
My overall impression is very positive,
then, but I would note a couple of
reservations about the design. Although
the height of the keyboard/accessory
drawer on the Z Flex can be varied, that
of the desk surface itself is fixed, and at
820mm above the ground, it’s relatively
high. That’s not a problem in itself, but
it means that the top of a 4U Griprack
placed on the desk is almost at ear height
even before you place a monitor speaker
on top of it. As a result, it’s difficult to
avoid having the tweeters firing over
your head, even with the Griprack at its
minimum tilt setting.
A related issue is that the curved
panels at the rear of the Griprack extend
some 200mm beyond the foam panel
on which your monitors sit. This, in
effect, means that even with the desk
hard against the wall of the studio, your
speakers will inevitably be quite some
distance away from it. That’s generally
not recommended from an acoustic point
of view as it brings speaker-boundary
interference response (SBIR) into play.
Having said that, though, I didn’t notice
a deterioration in the low-frequency
response in my own studio, and I’ve yet
to experiment with treatment behind
the speakers.
In terms of price, there’s no doubt
that the Zaor furniture will work out quite
a bit cheaper than having something
custom made. Wood prices have gone
through the roof over the last couple of
years, and if you were to make something
the size of the Z Flex from marine ply or
similar, you’d be looking at hundreds of
pounds for materials alone. At the same
time, though, the studio furniture market
is a niche one, and doesn’t quite enjoy
the same economies of scale as the
wider office furniture market, so you’ll
pay more for a Zaor desk than you would
for a generic office desk of similar size
and quality. Having now tried both, I feel
the studio-specific features of the Z Flex
— especially the cable management
and slide-out, heavy-duty keyboard tray
— probably justify this premium, and as
long as you’re not planning on moving
studios too often, it’s a worthwhile
investment. For me, though, it’s the
Griprack that is the real life-changer, and
I think they’ll be responsible for a large
reduction in the amount of cursing
emanating from my room!
£ Miza Z Flex £669 (grey) or £629 (black/
cherry); Griprack 4 MkII £209/£199. Prices
include VAT.
W www.zaorstudiofurniture.com
 ON TE S T

PAUL WHITE

T
he Dept. 10 Dual Drive
and the Dept. 10 Dual
Distortion both originated within
Blackstar Dept. 10
UK company Blackstar’s Dept. 10, which Overdrive & Distortion Pedals
is a section of the company dedicated
to exploring new and innovative design These valve-based pedals offer DSP speaker emulation,
approaches. In a nutshell, these are USB audio interfacing and comprehensive EQ.
dual-channel overdrive/distortion pedals
that combine full-voltage valve circuity voicing switch. All the other controls are treble EQ augmented by Blackstar’s ISF
with some additional analogue circuitry, common to both channels: a three-way control, which morphs the EQ voicing
plus digital speaker-cabinet emulation. cabinet emulation selector for three from a more edgy US style to smoother,
The reason that I’m reviewing both different cabinet types and a bass, mid, more reserved UK tonality.
together is that the pedals are identical
in format, and differ only in colour and
voicing. Conceptually, they appear
to build on the ideas that have been
explored in the company’s existing HT
valve pedal range.
Housed in tough steel cases with dual
footswitches, the pedals require 500mA
of 9V DC power, and a suitable PSU is
included. Internal circuitry then multiplies
this voltage to over 200V to drive the
valve, so that it functions in the same
way as it would if in a traditional valve
amplifier. There are two jack outputs
on the left of the case, one a stereo
headphone jack with speaker emulation,
and the other a mono jack with no
emulation — the latter is the one you’d
use to plug into a guitar amp. On the
rear of the case is a balanced XLR line
output, again with speaker emulation,
but a pair of send/return jacks allow an
external effect to be inserted between
the overdrive and the speaker emulator.
There’s also a USB port, to allow the
output to be recorded directly into
a computer.
Each channel has its own set of Gain
and Volume controls plus a two-way

Blackstar Dept. 10
£249
PROS
• Flexible two-channel design.
• High-voltage valve circuitry.
• Effective DSP speaker emulation.
• Free editing software.

CONS
• At this price? Nothing.

SUMMARY
While there are many all-digital, and
some solid-state, analogue amp
emulators, this one is unusual in
combining full-voltage valve circuitry
and analogue EQ with DSP speaker
and power-amp emulation.

58 February 2022 / w w w. s o u n d o n s o u n d . c o m
 On the Dual Distortion model, the while on Channel Two it selects between The stock cabinet switch settings provide
voicing switch on Channel One selects two flavours of overdrive, the second a bright-sounding cab to the left,
between CL (Clean) and CR (Crunch), option having slightly more edge. a scooped metal-style voicing in the

A free Mac/Windows app allows you to select different speaker emulations, power stages and EQs.
 ON TE S T
BL A C K S TA R D E P T. 10
middle and a warmer, fatter sound to the
right. For the Dual Drive model, the first
channel again has a choice of clean or
crunch voicing while Channel Two offers
Crunch or Overdrive.
Driving Range
The Dual Distortion’s clean setting can
be pushed to add a little ‘hair’ to the
sound, while the Crunch setting gets
you into blues and classic rock territory.
For more raunchy rock or for more
aggressive contemporary rock sounds,
the Overdrive settings of channel two fit
the bill nicely, so this pedal covers a lot
of ground. Importantly, it imparts the
right feel to the guitar, with just the right
degree of touch-responsiveness. The
pedal’s EQ section behaves in a very
amp-like way, while that ISF control really
does make a big difference when you are
searching for exactly the right tonality; it
seems to shift the character of the whole
EQ section. Overall, the experience feels
very much like playing through a proper
valve amp, even when the pedal is
plugged into a solid-state amplifier set
to clean.
Plug in the Dual Distortion to
your computer, either feeding the
speaker-emulated analogue output into
an audio interface or directly over USB,
and those three speaker emulation
settings provide even more variety.
Choose the middle setting, set the ISF
control anti-clockwise and use Channel
Two with plenty of gain and you have
instant metal, with the familiar hollow mid
and edgy top. However, using the third
cab option with the ISF control clockwise
and the gain backed off a little brings you
back to a familiar UK rock sound.
The Dual Drive pedal sounds to me
like a slightly warmer, cuddlier version.
The clean setting ranges from very clean
to a nice jangle that’s well-suited to ’60s
and ’70s pop — great for where a Kinks/
60 February 2022 / w w w. s o u n d o n s o u n d . c o m
As well as a clean mono out for your amp and a speaker-emulated headphone out, both via jacks on the
side panel, the rear panel of both pedals boasts a speaker-emulated line output, a USB output for recording
directly to your DAW and a (pre-speaker emulation) insert send and return.
Who/Beatles range of sounds is needed.
Shifting over to crunch can move you
from slightly murky blues into ZZ Top and
even Gary Moore territory. In Overdrive
mode, there’s more gain on tap, so at
lower settings there’s some overlap with
what you can do in Crunch mode, but
wind up the drive and you can cover all
the classic rock genres, ending up with
a Black Sabbath meets Deep Purple kind
of angst. Again the EQ and ISF controls
provide a lot of scope for shaping the
sounds, and the speaker emulations are
again extremely convincing. The Dual
Drive should cover pretty much all your
needs unless you’re into Scandi Death
Metal — and if you are, then maybe the
Dual Distortion is for you.
In the studio, you can either use the
straight output in conjunction with one
of your own speaker emulator plug-ins
or work with the emulated output. I don’t
know precisely what DSP techniques
Blackstar have used to recreate these
cabinet sounds — they say their
approach goes beyond IRs — but they’re
decent, and I didn’t feel the need to
switch them out for something else.
There aren’t many pedals you can plug
into a computer that just make you want
to sit and play, but these certainly do,
and I think it’s the way these recreate the
experience of playing through a valve
amplifier that makes them so comfortable
to work with.
Yet, there’s more. If you still don’t
find enough variety in these pedals as
shipped, you can download the free
Blackstar Architect Mac/Windows app
and this offers a much wider choice of
speaker cabinets, microphones and on/
off-axis mic positions, as well as extra
EQ options. There’s also power-amp
emulation that’s handled by the DSP, so
you can choose between EL84, EL34 or
6L6 output valves, as well as adjusting
resonance, drive and presence. You can
also add in a little emulated room mic
ambience with a choice of the three room
sizes in both damped and undamped
variants. Any of the three Cab Rig presets
in the pedal can be overwritten by your
edited version, and you can also back up
favourite setups on your computer.
You may not get the additional effects
that some of the all-digital platforms offer
but my guess is that most players would
want to use their own effect pedals
when playing live, and in the studio
we already have plug-ins that can add
anything from delay and reverb to the
seriously outlandish.
Drive It Home
In summary, then, whatever your guitar
style, one of these pedals should
meet your needs: each covers a lot
of tonal ground, while also providing
a gratifyingly touch-sensitive playing
experience. For live use, the ability
to set up two channels, or to bypass
them to give you your unadulterated
amp sound, means you now have three
sounds available, and the ability to
customise the sounds using software
is an unexpected bonus. The speaker
emulations are effective and mean you
can send an oven-ready feed to the PA
for gigs, and the insert means you can
place more pedals after the preamp but
before the speaker emulator — so you
can get all your effects running in the
order you want them. These are very
well thought-out, tonally versatile and
attractively priced pedals.
£ £249 each including VAT.
W https://blackstaramps.com
THIS THING A BADASS “I’ve used a lot of 2 channel interfaces but
when I pugged this in I noticed everything

SOUNDS GREAT! L I T T L E got clearer and the depth was incredible.

WAY BETTER THAN B O X . THIS PRODUCT

MY PREVIOUS -Ben Bernstein WILL CHANGE THE
INTERFACE. (TapeOp Magazine)
WAY YOU HEAR
YOUR MIXES”
THIS MAY BE THE
-Don Gilmore
(Linkin Park, Korn, Good Charlotte)

BEST SOUNDING
-Chris Lord-Alge
(Green Day, Keith Urban, Bruce

2CH INTERFACE
Springsteen, James Brown, etc.)

I’VE NOT HEARD

UNDER $2K.
A BETTER IT’S CRAZY TO ME TO
THINK THAT A $400
- TC from TX
[HEADPHONE
AMPLIFIER] IN ANY UNIT CAN SOUND
I was in the market for 2 new channels
of i/o. I have a $2,000 a/d and a $2,000

USB BUS-POWERED [MUCHBETTERTOMY

d/a on loan that I was going to be
looking into buying. After having both

INTERFACE. EARS] THAN SOME

setups side by side, listening... I am
sold! This is the clearest and BIGGEST

OF THE 2-6K UNITS

sounding interface I have ever heard...
- Sam Inglis

BUT FOR THE PRICE WE TESTED OUT!!!

(Sound on Sound Magazine)

DIFFERENCE I CAN -Noam Levinberg

THE REVOLUTION (Pluto Studios / Artist.io)
STILL AFFORD A
2X2 IS AN
MICROPHONE, 2 “OMFG HOLY $#%”
ABSOLUTE BEAST.
MONITORS,ALLTHE I have reviewed everything and you guys

CABLES NEEDED
If you’re looking for a 2 channel interface, are going to be one of THE names in pro
audio (more so then currently) very very

THIS IS THE ONE. AND A USED 1999

quickly. I plan on keeping this gorgeous 2x2
for my home mixing. Do we have a 8X8 or a

TOYOTA COROLLA.
ADDA with this clock in it in the works for
HANDS DOWN. the studio? I hope so. Just amazing guys.

- Ask Audio - PP from Virginia - AB3D

Also available: a
limited time bundle,
with the Izotope Music
Production Suite 4.1
for a special price.

VTN  
Learn more at www.blacklionaudio.com.

©2022 All Rights Reserved, Black Lion Audio. All other trademarks are property of their respective owners.
 ON TE S T
SonnectSoundBullet
HUGH ROBJOHNS
N
o matter how well we maintain
our audio equipment and how
carefully we rig complex audio
systems, sooner or later something won’t
work as expected and the Law of Sod
says it will be at the least convenient
moment! Fast and efficient fault-finding in
a complex audio installation requires solid
underpinning knowledge of the technology
being used, a calm and logical approach,
and some reliable test equipment. Of
course, it’s that last element that’s so
often absent when you really need it. To
identify a broken cable you really need
a cable tester. To identify a plugging or
signal routing issue you need a signal
generator and/or a signal level meter and
speaker/earphones. And to spot which
phantom-power button has mysteriously
managed to deselect itself when nobody
was looking, you’ll need something that
measures and indicates phantom power.
While I’m sure most of us have access
to a decent cable tester, and a signal
generator, metering, portable monitor
speaker, and even a multimeter, this isn’t
normally the kind of gear instantly to hand
while rushing on stage or into the studio
while everyone is staring at you to find and
fix a problem. I’ve seen dozens of audio
testing devices over the years, some more
useful and usable than others, but it’s
quite telling that the only one I’ve found
useful, reliable, and convenient enough
to carry with me everywhere for the last
30-plus years is a home-made Bright Eyes
— a simple phantom power detector that
can also be used to identify faulty XLR
cables, which takes the form of a Cannon
XLR plug on a keyring!
Sound Bullet
That’s why Sonnect’s Sound Bullet was
invented. David Scorteccia, a professional
live sound engineer working in the
UK, tried to find a comprehensive but
convenient audio tester he could use in
his day job and couldn’t find anything that
met his needs. So he decided to design
his own. Like most good ideas, the Sound
Bullet was born out of a practical, everyday
necessity. David needed something
compact that he could carry with him
62 February 2022 / w w w. s o u n d o n s o u n d . c o m
Portable Audio easily to test mic cables, check for the
Test Device presence of phantom power, and prove
audio lines to and from the stage. Having
Every engineer needs identified these requirements, it took
David four months to come up with his first
portable problem solvers, working prototype.
and this one could become Of course, having a great idea is
a constant companion. worth nothing if it can’t be turned into
a realisable product, but David had an
advantage there: his father owns an
electronics manufacturing company in
Perugia, Italy. As David used the prototype
in earnest on a daily basis more ideas
emerged for additional useful functionality
and, with his father’s help, the design was
refined and turned into a practical product
which is now manufactured entirely in
Italy. The PCB, metalwork and other
components are all sourced within 100
miles of the factory, which also takes care
of the assembly, testing, order fulfilment,
and invoicing.
The Sound Bullet is a very compact
yet surprisingly capable audio test
system, measuring just 130mm in length
and roughly 30mm square, housed in
a metal box and powered by an internal
rechargeable battery. It feels very durable
and solidly put together, with male and
female Neutrik XLR connectors extending
Sonnect Sound Bullet
€220
PROS
• Extremely compact, and with
a convenient carry case to ensure
it’s always available when needed.
• Carefully optimised range of
audio signal, monitoring and
testing facilities.
• Easily rechargeable battery with
a reasonable duration.
• Responsive manufacturer.
CONS
• One or two very minor weaknesses
in the design, although they may well
have been updated by the time you
read this.
• May seem a lot of money for a small
box with lights on!
SUMMARY
A very practical, well-designed audio
test device which will make any audio
technician’s life considerably easier
when it comes to system testing and
fault finding.

As the XLR connectors protrude from
the end, as well as hooking up to XLR cables
the device can plug directly into equipment’s
panel-mounted connectors.
from each end of the matt blue-grey
Nextel suede-coated chassis, and both
quarter-inch TS and 3.5mm minijack
sockets on the top. The protruding XLRs
are key to the design, as they allow the
Sound Bullet to be plugged directly
into equipment without needing cables.
Everything is controlled via a handful of
buttons, switches, and a thumbwheel, all
clearly labelled in white, and there are no
screens or menus.
For fast problem solving on stage or
in the studio the Sound Bullet covers
four core functions: it is a simple signal
generator (so it can substitute for a mic
or DI box), a basic signal analyser (with
metering and a monitor speaker), it
measures phantom power, and it tests
XLR cables. Its internal battery is charged
via a micro-USB port from any standard
USB charger (a USB-A to micro-USB cable
is supplied, but not a mains adapter),
providing over two hours of continuous
operation. In practice, as the unit is
only switched on when it’s needed (via
a recessed slide switch on the side),
one charge will easily last a full day of
PA rigging or studio sessions. When the
battery reaches 20-percent capacity the
‘Signal’ LED flashes three times every
six seconds, and when it is nearing full
depletion it flashes more urgently at 6
times every two seconds.
Each Sound Bullet ships with a fabric
storage case (these are available in
two different designs — see the Carry
Case Options box for more on this),
a double-ended jack plug adapter,
a USB charging cable, and a well-written
paper manual.
Signal Generator
Exploring the testing facilities in more
detail, the signal generator provides
either a 1kHz sine-wave tone or pink
noise, chosen by pressing the ‘Sel’
button. The ‘Signal’ LED lights solidly
when producing pink noise, but pulses
slowly for tone, and its colour indicates
the output destination (yellow for the
Carry Case Options
The carry case supplied as standard comprises
a pouch with a magnetically secured flap and
a quarter-inch socket on the side of the pouch
to store the supplied adapter plug. A carabiner
attached via elastic loops at the rear allows the
case to be clipped onto anything appropriate,
including a rigger’s belt, equipment bag, or
whatever. It’s a neat, practical and versatile
design, but when clipped to a belt it can move
around which some users find undesirable. In
response to customer feedback, an alternative
The whole kit and caboodle fits into a protective pouch, and a carabiner is supplied for easy
attachment to belts, bags and so forth.
default male XLR output, and blue for
the quarter-inch TS output — see below).
The output level of both noise and tone
is set by a miniature three-way toggle
switch at -10, -20 or -40 dBu (see the
Generator Signal Levels box). The tone
levels are within a 1dB of the stated
level, while the pink noise measured
a couple of decibels low on the review
model, but such minor technical
inaccuracies are inconsequential for the
intended applications.
On power-up, the signal generator
output is available at the male XLR as
a fully symmetrical balanced output
signal, while pressing the ‘To’ button
re-routes the output to a quarter-inch
unbalanced TS socket instead.
A double-ended jack-plug adapter is
included to convert the TS socket into
a plug, allowing the unit to be plugged
straight into a DI box, effects pedal, or
stage amplifier to check those signal
paths directly without needing an
instrument cable, which is a nice feature.
Cable Tester
The Sound Bullet’s second function is
as an XLR cable tester, and this mode
is activated by holding the ‘Sel’ button
down for two seconds, whereupon the
Signal LED flashes white once every four
seconds. With an XLR cable plugged
between the unit’s two XLR connectors
the integrity of each connection through
the cable is tested individually by
repeated presses of the ‘To’ button. A low
design has been produced, which is identical
except that the carabiner is replaced by a broad
elastic belt loop. This variant ensures that the
pouch sits flush against a user’s belt and stays
tightly in place as the wearer moves about. When
purchasing directly from Sonnect the preferred
type of case can be specified, while models
bought from retailers come as standard with the
carabiner version. Replacement cases of both
types and spare jack adapters are also available
from the Sonnect website.
w w w. s o u n d o n s o u n d . c o m / February 2022 63
 ON TE S T
SONNECT SOUND BULLET

(3.3V) DC voltage is applied to each of

the female XLR’s terminals in turn and A male-to-male TS jack adaptor is included,
a set of six LEDs on the side of the Sound and this allows the Sound Bullet to be plugged
directly into amps, stompboxes and the like.
Bullet indicate the presence of the test
voltage on each of the male and female
XLR pins so that any missing, mis-wired,
or intermittent connections become
immediately obvious.
The Sound Bullet cannot currently
detect whether a cable’s XLR shells are
linked to pin-1 or not, and while many
commercial cables have this link between
the XLR shell and pin-1 — as per the AES
recommendation — in my experience this
often results in unexpected ground-loops,
particularly when working with legacy
equipment. For that reason, I have
suggested to David that adding two
more LEDs to indicate when pin-1 is
connected to the shell at each XLR would
be a very useful, and fairly simple to
implement, enhancement.

Phantom Power Checker

A common reason for non-functioning
mics and DI boxes is that phantom power
isn’t reaching the end of their cable for
some reason, and a simple check for the
presence of phantom power can often
resolve the problem in seconds. The
Sound Bullet indicates the voltage on pins
2 and 3 of the male XLR (relative to pin-1)
with a pair of LEDs on the top of the unit.
If the phantom voltage conforms
to the IEC specification (48± 4V) each
LED illuminates solidly. Although some
equipment struggles to meet the specs,
especially when powering a lot of mics
simultaneously, most mics and DI boxes
will still function perfectly well with
phantom power below 44V. So, very
sensibly, the phantom power status LEDs
flash slowly if the voltage is between 24
and 44 V and, in the more worrying event
that phantom power exceeds 52V, the
LEDs flash ‘in an alarming way’!
Monitoring
Often all that’s required for fault-finding
is to be able to audition a connection to
confirm whether the right signal is on that
cable. The Sound Bullet accommodates
this requirement with a built-in miniature
speaker and 3.5mm minijack headphone
socket (the speaker mutes when
headphones are connected). Its volume
is controlled with a thumbwheel on the
side of the unit and it monitors signals
connected to either the balanced female
XLR or the unbalanced quarter-inch
jack socket (although the XLR seems to
dominate if signals are applied to both
at the same time). The volume control
covers an impressive range, and I was
able to monitor XLR input levels between
about -65dBu and +20dBu just on the
internal speaker (although I had to hold it
to my ear for the quiet end of the range
— earphones connected to the minijack
output make it easier).
As the internal monitoring amplifier
powering the speaker and headphone
socket is permanently wired across
the quarter-inch socket, if the signal
generator is routed to the jack socket its
noise/tone can be readily auditioned too.
It also means that the 3.5mm headphone
socket can be used as a stand-in sound
source for a presentation laptop or other
portable sound devices.
Generator Signal Levels
An audio test set that doesn’t generate the
standard alignment level of +4dBu seems a little
odd at first glance, but in the context of simple
functional checking of equipment on stage and in
the studio, +4dBu is actually a pretty hefty signal
level — your ears wouldn’t thank you for sending
a powered PA speaker pink noise at +4dBu!
The loudest output from the Sound Bullet
is -10dBu, but this is a reasonable (and safe)
level to check line inputs as it will register
However, having the quarter-inch
socket as both an output (for the
signal generator) and an input (for the
monitoring) creates a technical dilemma.
Ideally, the signal generator’s output
impedance should be low, to drive
long lines, but the monitoring input
impedance should be high to avoid
loading connected sources too heavily.
David decided a low output impedance
was the priority for the Sound Bullet’s
typical applications, and a sound source
plugged into the quarter-inch socket
‘sees’ an input impedance of 610Ω. This
is an unusually low input impedance
and while some equipment won’t mind
at all, others will respond with a slightly
reduced output level and possibly some
transient distortion — although neither
clearly the meters of most pro and semi-pro
equipment. The -20dBu option provides
a notional instrument level, and is therefore
ideal for checking DI boxes, effects pedals and
guitar amps. It’s also useful for testing powered
PA speakers without deafening anyone. And
the lowest -40dBu setting replicates a typical
dynamic mic’s output level, so is ideal for
checking mic lines without having to turn console
mic preamps down.

64 February 2022 / w w w. s o u n d o n s o u n d . c o m
are particularly significant if only testing
for the presence of a signal.
Nevertheless, if an output signal
needs to be auditioned more accurately
there is a neat workaround: selecting
the cable-test mode disconnects the
signal generator’s output circuitry from
the quarter-inch socket and then an input
source ‘sees’ only the monitor amp’s input
impedance of 12kΩ, which is high enough
to keep any device’s line output drivers
completely happy. (In case you were
wondering, the XLR input impedance is
a reasonable 5.9kΩ).
Where it’s not practical or necessary to
listen to the monitored signal, the Sound
Bullet also includes ‘signal present’ and
‘peak’ LEDs which illuminate at -25dBu
and +10dBu, respectively, to give a very
rough idea of received signal levels.
Bizarrely, though, these only monitor
the XLR input and don’t register signals
connected to the TS socket at all. Although
these are useful, it seems to me that what
is more often required is confirmation that
a signal is passing through a unit at unity
gain, and the Sound Bullet’s combination
of signal generator levels and metering
LEDs is of no use in that respect. So
I have suggested to David that perhaps
he could add an extra tri-colour LED which
would indicate whether the input level is
lower, the same, or higher, than the signal
generator output. Again, this would be
relatively simple to implement but would
add worthwhile practical functionality.
I’ve also recommended that the metering
reflects the input to the speaker/
headphones amplifier rather than just the
XLR, although that may be a more difficult
thing to engineer.
Impressions
I am genuinely impressed with the Sound
Bullet. It combines a plethora of pragmatic
facilities while being easy to use and it
covers all the common requirements for
fast, efficient fault finding and system
checking, whether on stage or in the
studio. It is also conveniently sized, and
is robustly built; it certainly feels rugged
enough to survive life on the road. That
it comes with a two-year warranty is
also reassuring, and I’ve been delighted
ALTERNATIVES
There are few broadly comparable
portable audio test units in current
production. For example, Whirlwind
offers the QBox, RDL the PT-AMG2, and
CTP both the TESTA and more elaborate
dBbox3, but none of these match the
convenience, compactness, practical
feature set, or value for money of the
Sound Bullet, in my opinion.
with the designer’s very quick and
positive responses to my queries and
suggestions, which bodes well for the
company’s future.
While €220 may initially seem
expensive for a small device like this, the
quality and functionality easily justify the
price, especially given that this is a tool
which will make the working lives of audio
professionals easier — the Sound Bullet
a very worthwhile investment for any
serious ‘soundie’.
£ €220 including Italian VAT (around £230
including shipping and UK VAT).
W https://sonnectaudio.com
w w o d.com / F r 2
 TALK B AC K
Hinako Omori
WILLIAM STOKES
J
apanese producer and artist
Hinako Omori can often be
found lending her synth and
keys expertise to the live shows of
the likes of Radiohead’s Ed O’Brien,
electro-pop sorceress Georgia and
spoken-word artist Kae Tempest. Her
own music weaves synths and samples
into enveloping ambient soundscapes
that often incorporate field recordings
made on her travels using a portable
recorder. Her first full-length solo effort
a journey… is set to be released in March
2022 via the Houndstooth label, and
incorporates recordings of the natural
area surrounding Real World Studios
in the west of England, originally taken
as part of WOMAD festival’s lockdown
series. Hinako has lately been developing
scores for animation, as well as working
on music for Belgian-American artist
Cécile B Evans’ installation at Centre
Pompidou. She also hosts a monthly
show on Worldwide FM with close
collaborator Kloxii, through which, as
Hinako explains, “We explore the topic of
identity and heritage through music that
helped us reconnect to our roots.”
At the moment I can’t stop listening to
Taeko Onuki’s album, Signifie. I’m
really embarrassed to say I didn’t know
about Taeko san’s [‘san’ is a title of
respect in Japanese] music until I came
across this record in a record shop in
Shimokitazawa a few years ago. It was
one of those moments where I picked up
the record, saw the incredible artwork
and immediately needed to listen to it!
I handed it straight to the gentleman
behind the counter who put it on for
me, and I instantly fell in love as soon
as I heard it. Such a beautiful, warm
66 February 2022 / w w w. s o u n d o n s o u n d . c o m
atmosphere and sound, with Taeko san’s
ethereal voice soaring over the lush
instrumentation… Many of the tracks
on the album are arranged by Ryuichi
Sakamoto also; they were frequent
collaborators. I later realised that Taeko
san was a huge key figure in Japan’s City
Pop scene in the 1970s.
The project I’m most proud of
I think it would be the project I worked
on last year, which has become the first
solo album project of my own. During
lockdown last year, my wonderful friend
Oli [Jacobs], the head engineer at Real
World Studios, kindly invited me to take
part in a series of sessions for WOMAD
festival called WOMAD At Home. The
premise was to invite nine artists to
create an immersive audio experience
that reaches beyond the possibilities
of traditional stereo listening. I felt
incredibly honoured to be amongst
a line-up including Sarathy Korwar, Native
Dancer, Voka Gentle, Blue Lab Beats
and Cheng Yu. At first I was planning to
do an ambient improvisation, but then
thought it could be a nice idea to piece
together the demos I’d started creating
throughout lockdown into one continuous
piece. Somehow quite magically they
all naturally seemed to fit together side
by side!
I’ve always been very excited at the
thought of working with binaural audio,
and I was lucky enough to be able to
travel down to Real World Studios a day
before my session to borrow a binaural
dummy head, drive out to the surrounding
woodlands and lakes and record field
recordings from the area for the project.
The idea with the recordings was to
create a sonic environment that could
teleport you out of your current location,
especially during this time of lockdown, to
somewhere that you can be surrounded
by nature. I was thinking about the art of
forest bathing, ‘shinrin-yoku’, and how this
can have such a positive effect on our
mental health and wellbeing.
We also reamped some of the
vocal stems using a D&B Audiotechnik
loudspeaker array system. Being able
to pan audio in a 360-degree space
was extremely fun, especially in the
stunning acoustics of the Wood Room at
Real World.
The first thing I look for in a studio
I hope this doesn’t sound too cliché,
but I think the vibe and atmosphere of
a studio is so important! Making music
can be such a vulnerable process, so
a safe space where you can fully immerse
yourself is ideal. Somewhere you can feel
completely comfortable and lose yourself
for a while. Of course, a lovely collection
of synths never goes amiss either…
The person I would consider my mentor
My ‘A’ level music technology teacher
at Reigate College, Lloyd Russell. While
we were studying Lloyd was in an
electro-pop band called Schmoof, playing
[Roland] SH-101 keytars. It was amazing
and eye-opening to hear the ins and outs
of the music industry from someone who
was directly involved in it. I’m so grateful
to Lloyd for all the encouragement and
advice, inspiring me to continue working
in music and for the help he offered us
post-college too. He even very kindly let
me borrow me his SH-101 long-term, as
he was very keen for me to further my
understanding of synthesis.
My go-to reference track or album
That’s a tricky question! I think it depends
on the type of music I’m working on.
I might delve into a playlist I’d created
of favourite tracks around a theme
— the thing I’m really interested in is
the creation of space, environments
and immersive elements in music, and
how artists and producers create their
own sonic architectures. My favourite
records on this theme at the moment
are by Pauline Anna Strom, Eliane
Radigue, Gia Margaret, Tim Hecker
and Emily A Sprague. And of course
Oneohtrix Point Never, Daniel Lopatin, for
cinematic soundscapes.
My desert island studio item
I’d love to take an original Theremin, the
RCA, by Leon Theremin. I’ve been delving
into the beautiful performances by Clara
Rockmore, which always brings tears to
my eyes. The expression, emotion and
tone of Clara’s playing speaks a thousand
words, especially her performance of
Tchaikovsky’s ‘Valse Sentimentale’. It
would be amazing to spend time on the
desert island with no distractions, really
diving into the instrument and learning
how to play it. It would take me a very
long time I’m sure, but I’m willing to
dedicate the time to it!
The piece of gear I’d bring back into
production
I was going to say the original Moog
Modular — watching videos of the
incredible Wendy Carlos and her Moog
synthesizer creating an orchestra of tones
has been such a huge inspiration for
me — although I think our dreams were
answered a few years ago when Moog
produced a limited run of the IIIc… so,
instead I’m going to go with the Ekdahl
Polygamist by Knas. It’s this amazing,
incredibly colourful semi-modular
analogue synthesizer I discovered for the
first time recently, which looks and sounds
like an out-of-this-world board game. It’s
controlled using a pressure-sensitive
ribbon controller. I believe there were only
55 units ever made.
The producer I’d most like to work with
Karin and Olof Dreijer. The Knife were
the first band that really inspired me to
delve into synthesis from listening to
their intriguing synth worlds, rhythms and
soundscapes… everything about their
creations is just magic. I became fixated
by the sound that they created in ‘Like
A Pen’, which always makes me envisage
a dripping tap, and found it so fascinating
that they’d built a beat for the whole track
around this sound. It would also be an
absolute dream to work with Susanne
Sundfør — I’ve been listening to her
record Ten Love Songs on repeat.
The advice I’d give myself 10 years ago
Enjoy every moment of the journey
and try not to worry about the smallest
details. I think if you go into something
open-hearted and open-minded, you’ll
eventually end up where you imagined.
Or, perhaps the path will lead towards
something you’d never thought of, but
that’s equally exciting. I’m forever grateful
for the wonderful and inspiring friends
and collaborators I’ve met along the way,
and the experiences I’ve been lucky
enough to have with them.
w w w. s o u n d o n s o u n d . c o m / February 2022 67
 ON TE S T
Royer Labs dBooster2
Inline Microphone & Instrument Preamplifier
Not content with having created arguably the world’s best preamp
booster, Royer’s boffins decided to make it better.
HUGH ROBJOHNS
A
merican manufacturers Royer
Labs are very well known for
their range of high-quality ribbon
microphones, but a few years ago they
also introduced the dBooster, an inline
microphone gain-boosting device that
was intended for use with ribbon and
other low-output passive microphones.
I reviewed the phantom-powered dBooster
in July 2019 (www.soundonsound.com/
reviews/royer-labs-dbooster) and found
not only that it was the most sophisticated
design of its type I had ever come across,
but also that it delivered consistently
excellent technical performance.
The dBooster is still available but,
being a single-channel unit, it was almost
inevitable that Royer would sooner or later
be persuaded to produce a dual-channel
version for the convenience of those using
stereo ribbon mics...
Overview
Well, here it is: the predictably named
dBooster2. As you’d expect, this new
version effectively provides two dBoosters
within a single extruded aluminium box,
and it uses an updated version of the
68 February 2022 / w w w. s o u n d o n s o u n d . c o m
same sophisticated circuitry that has
proven so effective and reliable in the
original single-channel model. I described
that original version in detail in my
previous review, but as a brief recap
each channel’s input section comprises
four paralleled discrete transistors on
each side of the balanced input to
form an ultra-low-noise input stage,
complete with elaborate power-supply
conditioning and a symmetrical balanced
output driver based on the OPA1692
high-performance dual op-amp. Current
feedback is employed to ensure
minimal distortion, and construction is
on a high-quality fibreglass PCB using
mainly surface-mount components, but
with high-quality through-hole Rubycon
electrolytic capacitors.
At 3.6kΩ, the input impedance presents
a relatively easy load for ribbon mics,
in order to maximise output level and
headroom, while buttons on the input end
of the box switch the overall gain boost
between nominal +12 and +20 dB settings.
Each channel draws up to 7.5mA of current
from the phantom supply, but there are
no LEDs to indicate when the box is
adequately powered. Importantly, phantom
power isn’t passed through to the input
XLR connectors, so delicate ribbon mics
are fully protected.
If you simply place two channels in
a box there’s always a risk of creating
inadvertent ground-loops — but not in
this case! Royer have neatly avoided that
possibility by building in DC isolation
for channel 2’s connector ground paths,
linking them to the unit’s chassis ground
only via capacitors. It’s an arrangement
which ensures effective RF shielding while
preventing low-frequency ground-loop
currents. In developing the dBooster2,
Royer have also tweaked the circuitry
slightly to further increase the headroom
margins and better align the actual gain
with the nominal values, and I discuss this
in more detail below.
Preamp or DI?
Now, if two channels and these minor
performance upgrades described the new
dBooster2 completely it would be a very
worthwhile product but, surprisingly, Royer
have also taken the opportunity to extend
the versatility of this elegant device much
further by incorporating instrument input (DI)
facilities on both channels, too. So, unlike the
original model, the new dBooster2 can serve
either as a dual-channel DI box in recording
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 ON TE S T
ROYER L ABS DBOOSTER2
Royer Labs dBooster2
£336
PROS
• Highly sophisticated design.
• Low- and high-gain/headroom modes.
• Ultra-low noise and distortion mic
gain booster.
• Phantom power cannot reach
ribbon microphones.
• Instrument input options, with two
gain/headroom levels.
• Very clean and accurate DI
sound quality.
CONS
• No phantom-power indicator LEDs.
• No link sockets to feed
backline amps.
• Such sophistication and build quality
is expensive.
SUMMARY
A clever development of the
original dBooster, adding a second
channel along with instrument
inputs to serve as a very high-quality
recording DI box when not needed as
a passive mic booster.
applications, or as a very high-quality stereo
mic booster — or, of course, it can provide
one channel of each at the same time.
To facilitate this new DI feature, the
standard input XLRs have been replaced
with ‘combi’ types, with a TS jack socket
at the centre of the XLR pins. Plugging
a standard instrument cable into the
combi-XLR routes an unbalanced signal
through a separate high-impedance input
stage, built around a discrete FET (Field
Effect Transistor) to provide the instrument
with an input impedance of 270kΩ. This is
lower than many active DI boxes (which are
typically around 1MΩ), but it’s similar to the
ever-popular Radial J48’s input impedance,
so it’s in very good company.
The dBooster2 doesn’t have any ‘link’
sockets which, traditionally, are used to
feed backline stage amps, because the
dBooster’s end panel simply doesn’t
have enough space to incorporate the
extra quarter-inch sockets that this would
require. But while this might frustrate in
ALTERNATIVES
While there are now various options for
passive mic boosters, I’m not aware of
any that offer two channels, boast the
sophistication and technical specs of
the dBoosters, or that can double up as
a DI box. Perhaps the closest recording
DI box equivalent is Orchid Electronics’
Dual Channel Micro DI, but that can’t
also boost passive microphones and
doesn’t have the dual gain modes.
70 February 2022 / w w w. s o u n d o n s o u n d . c o m
The XLR outputs draw phantom power from the main mic preamps in the recording chain.
some applications, link sockets are often
unnecessary in the intended studio setting,
where artist monitoring is now typically
direct via the recording system rather than
a backline amplifier.
I should also note that this is a DI which
is intended primarily for capturing signals
from electric guitar and bass pickups or
keyboards, etc, and not for piezo-equipped
acoustic instruments, which generally
sound much better when they’re fed into a
device with a much higher input impedance
of around 1MΩ or more.
Bench Tests
As instruments typically produce higher
signal levels than passive microphones,
the DI signal path offers less overall gain
than the mic path, but the gain switches
are still operative, boosting the level by
8dB. Testing the review unit on an Audio
Precision system, I found that the low-gain
mode for the instrument input attenuated
the signal by around -3.3dB, while the high
gain boosted it by +5dB.
The FET buffer introduces second and
third harmonic distortion progressively as
the signal level increases, but it remains
very low for normal signal levels, typically
below 0.005 percent, only reaching
0.2 percent just before clipping.
There’s plenty of headroom in the
circuitry and the low-gain instrument mode
clips at a reassuringly high input level
of +13dBu. In the high-gain mode it clips
with -6dBu at the input, while the residual
noise floor measured -96dBu in low gain
and -87dBu in high gain — both very good
figures for this type of device.
Looking at the frequency response, it
is ruler flat between -3dB points at 8Hz
and 22kHz for the DI input, whereas the
balanced mic input’s LF response falls
to -3dB at 6Hz. The HF response remains
flat to well over 80kHz.
Running through the bench tests for the
microphone booster, it measures pretty
much as the previous single-channel
model did, except that the low-gain
mode now provides the intended +12dB
(previously this was a little high, at +14.3dB).
The high-gain mode measured a tad over
+21dB, while the headroom and clipping
levels were both a couple of decibels
better than the original mode (-2 and -22
dBu for the low- and high-gain modes,
respectively). Total harmonic distortion
remained below 0.0005 percent at all
practical signal levels, which is excellent.
Conclusion
Fundamentally, then, Royer’s dual-channel
dBooster2 is a very clean and sonically
accurate DI box — you shouldn’t expect it to
add musical coloration — and it’s obviously
very convenient when working with passive
stereo ribbon mics in particular. The
dBooster2 is a predictable development
of the original design, but one which offers
the bonus of a slightly enhanced technical
performance. The addition of instrument
input facilities was completely unexpected
and, while it is a relatively easy upgrade
from an engineering point of view, it
adds considerably to the versatility and
cost-effectiveness of this device.
As we have come to expect, the
dBooster2 is also engineered and
constructed to very high standards indeed,
with a concomitant level of technical
performance and flawless sound quality.
I can’t recommend the dBooster2 highly
enough — gain boosters simply don’t get
any better than this!
£ £336 including VAT.
T SEA Distribution +49 590 39 38 80
E info@sea-distribution.com
W www.sea-distribution.com/en
W https://royerlabs.com
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 T E C H NI Q UE

What Do All Those Knobs Do?

The right EQ settings can make a mix
— and the wrong ones can make
a mess. Our in-depth guide will
help you get more from this
all-important tool.
 SAM INGLIS
Mathematician Charles Fourier
proved that any complex repeating
waveform (in this case, a C7 chord)

R
ecord sound into a DAW, and it’s
represented visually as a waveform. is the sum of multiple sine waves at
different frequencies and amplitudes.
Zoom in, and you’ll notice repetition.
Peaks and troughs of similar size and shape channels of our mixing consoles,
follow each other at regular intervals. Their in dedicated outboard units and,
spacing reflects the fundamental pitch or above all, in software plug-ins.
frequency of the note. Their height is known They are applied to individual
as the amplitude of the waveform, and the mics and instruments, to group
shape of each peak and trough reflects the busses, to the master bus and
timbre of the source. even to the monitor path to correct
The simplest waveform is a sine wave, the failings of our loudspeakers
which looks like a series of half-ellipses. One and headphones. Equalisation is
of the basic facts that makes digital audio a transformative rather than an
processing possible is that all other repeating additive process, and as such, an
waveforms can be described as, or reduced EQ is almost always used as an
to, combinations of sine waves at different insert, not on an auxiliary send.
frequencies and amplitudes. In a pitched
sound, the sine wave with the lowest
Bands Of Gold
frequency is known as the fundamental; the Another way of representing
others are mathematically related to this, and sound visually is a spectrogram:
are known as the harmonics or overtones. a two-dimensional graph with
A recording of a typical piece of music amplitude on the vertical axis and
will contain multiple fundamental notes and frequency along the horizontal.
harmonics, all playing at once. The bass Unlike a waveform display,
instruments might sound fundamentals a conventional spectrogram doesn’t have We can display the action of the
as low as 30 or 40 cycles per second a time axis. It shows a single balance of equaliser as a line running horizontally
(Hertz), whilst the upper harmonics of frequencies, averaged over a time period. across the spectrogram. With the EQ doing
the high-pitched instruments will be well However, if we keep this time period short, nothing, this line will be flat, indicating
above 10,000Hz (10 kilohertz). Inasmuch as we can continually update the spectrogram that it’s not altering the amplitude of any
it consists of sustained, pitched material, so that it follows the evolving content of the frequencies within the overall sound.
what we hear as a single piece of music can audio. This sort of animated spectrogram is Where the line is not flat, it raises the
always be broken down into lots and lots of now the standard form of visual feedback amplitude of frequencies in areas where
different sine waves at different amplitudes in many software equaliser plug-ins, and is it’s above the flat, and reduces them where
and frequencies. That goes for full mixes just extremely helpful for understanding the way it’s below. On some software equalisers,
as much as it does for individual instruments. in which EQ acts on the sound. the line itself is a control. The response of
When a recording fails to be true to the
source, one common reason is that the
balance between all of these different sine
waves has been altered. A good example
of this is the proximity effect. When placed
close to the instrument they are recording,
directional mics exaggerate the amplitude
of the lowest sine waves. This can manifest
itself as a muddy or boomy sound. There
are many other ways in which the balance of
frequencies within a recording can become
skewed, either because of circumstances
or the limitations of equipment. The idea
of correcting — or ‘equalising’ — this
imbalance through electronic means has
been around almost as long as electrical
recording itself. The term has remained
with us, usually abbreviated to EQ, but the
process itself has become a routine part of
recording and mixing.
Nowadays, we don’t only use EQ to
Photo: Hannes Bieger

match the sound of a recording to that of the FabFilter’s Pro-Q 3 is arguably the definitive ‘modern’ EQ plug-in, with its animated spectrogram display
original source, but for many other reasons. and click-and-drag interface. In this image the spectrogram clearly shows a fundamental pitch at 250Hz
Equalisers are everywhere: in the input and harmonics at 500, 750 and 1000 Hz.

w w w. s o u n d o n s o u n d . c o m / February 2022 73
 T E C H NI Q UE
EQ : W H AT DO A LL T HOS E K NOBS DO ?
Phase & Transients
I started this article with the great insight of the
French mathematician Charles Fourier: that any
continuous repeating waveform can be broken
down into a number of sine waves at different
frequencies and amplitudes. Equalisation is
a process that changes the relative amplitudes
of these sine waves — but that’s not all it does.
Most EQ designs, including all analogue EQs,
also alter the phase relationship between them.
If, for instance, we boost at 10kHz, we aren’t just
making sine waves in that region louder compared
with those at, say, 1kHz. We’re also delaying
them slightly.
In many circumstances this is a non-problem.
Our ears are not very sensitive to phase shift, and
EQ is not the only thing in the signal chain that can
cause it. In fact, directional microphones distort
the phase of the sound they’re capturing, so we
often aren’t starting with a faithful representation
of the original phase relationship anyway. However,
there are occasions where it can be an issue. The
amount of phase shift generated is related to the
steepness of the EQ slope, so we are most likely
to hear it when we apply steep filters to the entire
mix. This isn’t something you’d do for its own
the equaliser can be set simply by clicking
and dragging to change its shape.
Try this, though, and you’ll notice that
there are constraints on the types of
shape that can be created. The line cannot
contain sharp corners, or vertical sections.
The shapes you can create are invariably
curved and, unless they go right off the
sides or bottom of the graph, they’re usually
symmetrical. When you click and drag in this
style of EQ to change the shape of the curve,
you are adjusting what in a more traditional
EQ would be called a band. Modern
software EQs can have an arbitrarily large
number of bands, but in an analogue device,
each band requires its own dedicated circuit.
Analogue equalisers thus have a fixed
number of EQ bands; and often, each band
is of a fixed type. The three main types of
EQ band used in mixing are filters, shelving
bands and bell or parametric bands.
Friendly Filters
The simplest of these is the filter. Create
a high- or low-pass filter band on an EQ
with the sort of graphical display described
above, and you’ll see that one end of the EQ
line now dives towards, or off, the bottom
of the graph. A high-pass filter (also known
as a low-cut filter) progressively reduces
the amplitude of frequencies below a fixed
point. A low-pass (high-cut) filter does the
same with frequencies above a fixed point.
The fixed point in each case is known as the
cutoff or turnover frequency.
74 February 2022 / w w w. s o u n d o n s o u n d . c o m
sake, but it’s necessary for certain applications.
For example, most studio monitors use at least
two separate drivers to handle different frequency
ranges, and this means using steep filters to
ensure that only the low frequencies reach the
woofer, while the tweeter receives only the high
frequencies. This is one reason why sealed-box,
single-driver speakers such as the Auratone are so
valuable: although they can’t reproduce the entire
frequency spectrum, they also don’t introduce
crossover-related anomalies in the midrange.
In mixing, phase shift is often apparent with
multiband dynamics plug-ins. These use steep
filters to divide the frequency spectrum up into two
or more bands, which can then be compressed
independently. Simply inserting one of these
plug-ins on the master bus can cause the sound
to change noticeably even before you do any
compression, as a result of phase shift caused by
the filters. Phase shift caused by EQ can also be
apparent when the same source is captured on
multiple microphones: your bass drum might no
longer cohere quite so well with the overheads
once you start using EQ, even if the EQ improves
the sound of the close mic in isolation.
The two key parameters that can be
modified in a filter are the cutoff frequency
and the slope. Referring back again to an
EQ with a spectrogram display, the latter
is a very literally named parameter! The
higher the slope setting, the more abruptly
frequencies beyond the cutoff are reduced.
The value of the slope is usually expressed
in decibels per octave (dB/oct) and for
reasons I won’t go into here, the options
The effect of the slope setting is best understood visually! In this screen from Eiosis’s AirEQ, the
low-pass (high-cut) filter at the right has been set to a gentle 6dB/octave, whereas the high-pass (low-cut)
filter is operating at 12dB/octave.
It’s also worth remembering that Fourier’s
insight strictly applies only to continuous,
unchanging waveforms. In practice, any sound
that is sustained enough to have a timbre
or tonality at all can probably be considered
a reasonable approximation to a continuous
waveform, but many real-world sounds also
contain transient elements. This goes for almost
all drum and percussion sounds, as well as the
initial note onset in many plucked or hammered
instruments. These, by their definition, are
momentary rather than continuous, and don’t
last long enough to have a tonal character as
such. However, phase shift caused by EQ can
undermine their impact and rob them of punch.
In a digital system such as a DAW plug-in,
it’s possible to implement what is called a linear
phase EQ. As the name suggests, this makes
possible equalisation without phase shift.
However, there’s a down side, and again, it’s
one that primarily affects transients. Instead of
‘smearing’ caused by phase shift, linear phase
EQ causes a different phenomenon known as
pre-ringing, producing artefacts that are heard
before the transient itself.
available are usually multiples of 6dB/oct.
The steepest slope available in analogue
designs is usually 24dB/oct, though steeper
filters can be constructed.
Slopes For Improvement
Most mixers and EQs have high-pass filters
on every channel. Quite often, these are
entirely preset affairs, with the only control
available being an on/off switch. It’s cheaper
for a manufacturer to build fixed settings
into the design than to make them variable.
However, this doesn’t mean all fixed filters
are the same, so it’s worth consulting the
manual for your mixer to find out what
exactly your particular filters do.
Despite its simple nature, the high-pass
filter is arguably the only truly indispensable
type of equaliser, and certainly the most
important for corrective equalisation. Broadly
speaking, it can be used for two functions.
One is to remove unwanted low-frequency
noise whilst leaving the wanted signal
intact. The other is to correct the balance
of a recording where the low end is
exaggerated. A typical example of the first
situation might be to deal with a recording
of an acoustic guitar where the player is
tapping his or her foot. The fundamental
frequency of the lowest note played on
the acoustic guitar might be 80Hz or so,
whereas most of the sound picked up from
the foot-taps will be at lower frequencies.
In theory, by setting a high-pass filter with
a cutoff frequency at 80Hz and a steep (18
or 24 dB/octave) slope, we can reduce the
unwanted sound without affecting the guitar
sound at all.
By contrast, if the guitar has been
recorded using a directional mic placed too
close to the body, we may have a different
corrective problem on our hands. Thanks to
the proximity effect, that 80Hz fundamental
and those of other low notes will be
exaggerated, making the overall sound
boomy and muddy. In this case, we want to
configure our high-pass filter with a much
more gentle slope and have it turn over at
a higher frequency — a 6dB/octave slope
from 200Hz might not be a bad starting point.
Low-pass filters are less commonly used
in mixing, but they do have their place. For
example, most guitar amp cabinets don’t
produce a great deal of sound above 6kHz
or so. So if a cabinet is miked up next to
a drum kit, most of what is captured above
6kHz will be spill from the cymbals and
snare; if this sounds bad, it can be removed
using a low-pass filter without affecting the
guitar sound. There are also percussion
instruments such as tambourines and
shakers that generally belong in supporting
roles in a mix, but are possessed of
ferocious amounts of high-frequency jangle.
A low-pass filter can help them to be audible
in the mix without dominating it.
On The Shelf
Most equalisers have just a single
high-pass filter per channel. And likewise,
most have a single low and high band of
Like the stock EQs bundled with most DAWs, the Pro Tools EQ3 plug-in is simple but very capable. Here
you can see the characteristic shapes which give shelving bands their name — in this case, the low shelf is
cutting whilst the high shelf is being boosted.
shelving equalisation per channel. Often
this is provided in addition to the filters,
sometimes as a switchable alternative.
Create a shelving band on an EQ with
a spectrogram and you’ll immediately see
where it gets its name: it offsets the EQ
line by a fixed amount above or below its
turnover frequency. Usually, you’ll be able
to choose this frequency, and you’ll also be
able to specify a level of gain in decibels.
This sets the extent to which the shelf is
offset from the flat part of the curve, with
0dB being off or inactive.
Sometimes, you’ll have the option to
determine how abruptly the EQ line moves
from its flat position to the shelving level. This
setting is sometimes called the ‘slope’ but,
borrowing a term from the next type of EQ
we’ll encounter, can also be known as the
bandwidth or Q. Where this is isn’t adjustable,
it’s often set at a gentle 6dB/octave.
Shelving bands have both corrective
and creative uses. A low-frequency shelving
cut can be another good way to tackle
excessive boominess caused by proximity
effect, whilst a low shelving boost can do
the opposite, beefing up the low end of
an anaemic recording. And although you
wouldn’t often use two low-frequency
shelves on the same source, there are
situations where combining a low shelf
and a high-pass filter can be effective. For
example, a recording of a bass instrument
might have excessive sub-bass, yet be
inaudible on small speakers because it
lacks weight in the low midrange. We
could address this by applying both a low
shelving boost from, say, 300Hz down, and
a high-pass filter turning over at 60 or 80
Hz. This would reduce the amplitude of
the low-frequency fundamentals and raise
that of the harmonics, changing the sound
of the instrument and helping to create
a mix that ‘translates’ across every type of
listening system.
Shelving equalisation is particularly
good for subtly altering the overall tone
of a source, or even of an entire mix. With
a gentle bandwidth setting, the shelving
bands on a good EQ will sound extremely
transparent. The most common use
for a shelving boost is to subtly add air
and presence at the top end. In fact, it’s
probably fair to say that the bright sound
of modern pop and rock mixes is hard to
achieve without using shelving boosts over
many of the individual sources, or the mix
as a whole.
It’s worth experimenting with a very wide
range of turnover frequencies when you
do this. You might find that all you want is
the hint of sheen added by a boost from
10 or 15 kHz upwards — but often it can be
beneficial to use a shelf that extends right
down into the midrange, perhaps even as
far as 1.5 or 1 kHz. One of the most common
faults in mixes by inexperienced engineers
is a failure to fill out this midrange area
enough. This part of the frequency
spectrum really is the ‘engine room’ of any
mix. It’s the area where key signals such
as vocals mostly lie, and it’s the one part of
w w w. s o u n d o n s o u n d . c o m / February 2022 75
 T E C H NI Q UE
EQ : W H AT DO A LL T HOS E K NOBS DO ?
The broad and narrow path: in this screen from Apogee’s ModEQ 6, there are two active bands. The red
band is applying a high-Q/low-bandwidth cut at 200Hz, while the blue band implements a broader (low-Q/
high-bandwidth) boost around 2kHz. Note that the shape of each band is symmetrical according to the
logarithmic frequency scale along the bottom of the graph.
the mix that is guaranteed to be heard on
any playback system, no matter how bad. It
is therefore vital to make the best possible
use of it, and even if you don’t retain it in
the final mix, experimenting with a broad
shelving boost on the master bus can be
a great way to find out which sources can
be pushed harder, and how far.
The Bell Curve
The third common type of EQ band
is the bell or parametric band. Unlike
filters or shelves, parametric bands aren’t
intrinsically ‘low’ or ‘high’, and the frequency
setting doesn’t mark the outer limit of the
band’s action but its centre. Assuming the
design of the EQ allows it, they can be
positioned anywhere within the frequency
spectrum. And, as is also obvious from any
EQ graph, the name ‘bell’ refers not to the
sound but to the shape of the curve that is
created. The action of a parametric band is
greatest at the centre position determined
by the frequency control. Either side of
this, it curves back to the flat, forming
a symmetrical hump or dip.
The bandwidth or Q control
encountered with many shelving EQs really
comes into its own with parametric bands.
As the name suggests, this governs the
width of the hump or dip. Where Q (short
for ‘quality factor’) is given a numeric value,
it can seem a bit counter-intuitive, because
76 February 2022 / w w w. s o u n d o n s o u n d . c o m
a low Q value indicates a wide bandwidth,
and a high value a narrow one.
Bandwidth is expressed in octaves, and
this is often the most helpful way to think
about it in a musical context. It’s important
to remember that the relationship between
frequency and musical pitch is logarithmic
rather than linear. An interval of an octave
represents a doubling of frequency. An EQ
band that spans 100Hz to 200Hz thus has
a bandwidth of one octave; but so does
a band that spans 400Hz to 800Hz, or 3kHz
to 6kHz. The same goes for the relationship
between the centre frequency of the band
and those of its outer limits. EQ bands are
symmetrical on the same logarithmic scale,
so for example, a two-octave parametric
Automating EQ
Among the many advantages that modern
digital mixing environments have over the
old-school recording studio are the availability
of unlimited tracks, and the fact that almost
everything can be automated. Both of these are
relevant to the practical use of EQ, especially
in a corrective context. Very often, problems
that require correction using EQ aren’t entirely
consistent throughout a recording. For example,
an over-excited singer may move around too
much in front of the mic: when they get too close,
plosive pops and proximity boost are apparent,
but when they go off-axis to the side, the sound
will become darker at the top end. The sort of
API designer Saul Walker
is said to have invented the
‘proportional Q’ equaliser, and
it is used in modules such as
the current API 550B.
boost centre ed at 100Hz
will have its lower
l and
upper limits at 50Hz and
200Hz respe ectively.
Incidentally, not all
equalisers thhat offer bell
bands offer ccontrol over
the bandwidth. Typically,
these semi-p parametric
designs have what’s
called a ‘proportional
Q’ response. What
this means iss that the
bandwidth narrows as
the extent off cut or boost
increases. Th his might
sound like a limitation,
but in practicce it is usually
sympathetic to our needs,
and makes the gain control more useful
over a wider range.
Cuts Both Ways
The best way to understand how the
frequency, gain and bandwidth controls
interact is to experiment on a modern
software plug-in with a helpful graphical
interface. It will quickly become apparent
that even a single fully parametric EQ band
is a hugely powerful and versatile tool, and
that an EQ with multiple bands is capable
of drastically reshaping any source. So how
can we turn all this power to our advantage
when mixing?
Once again, parametric EQ has
applications that could be described as
corrective or creative. One almost universal
piece of guidance to bear in mind is that
high Q (narrow bandwidth) settings fall
into the former category, and that they are
EQ cuts that are needed to deal with the former
problems when they occur will actually be
detrimental to the sound elsewhere in the track.
There are two main ways of dealing with this.
One is to automate the gain (and perhaps other
parameters) on some EQ bands, so that they act
only when needed. The other is to split different
sections of the part to different tracks within your
DAW, and use different plug-in EQ settings on
each. Which you use is a matter of personal taste
as much as anything; personally, I tend to the latter
approach unless there’s a need to do something
very complicated such as have the frequency of an
EQ band track notes within the part.
 Flying Solo
Some mixing rules are honoured more in the breach than in the
observance, and none more so than the oft-quoted dictum that
you should never adjust an EQ while listening to something in
solo. I don’t know a single engineer who would not cheerfully
admit to doing this all the time! Even so, it expresses a valid point,
which is that the aim of EQ’ing things within a mix is not to make
each individual source sound as good as possible. Rather, the
goal is to make the mix as a whole sound as good as possible —
and reaching that goal often means equalising individual sources
differently from how you would if there was nothing else playing
at the same time. In particular, a satisfying overall mix balance is
sometimes achieved only by cutting quite a bit of low midrange
from individual sources, which consequently sound rather thin
when you hear them in solo.

much more useful for cutting than boosting. Some of the most
common sources of unwanted noise in recordings actually
manifest themselves as sine waves, or at least as fairly simple
continuous waveforms with relatively few harmonics. These
can often be effectively eliminated by using narrow-band
parametric EQ cuts. Examples include feedback in live
recordings, 50 or 60 Hz mains hum, the high-pitched whine of
old cathode-ray TVs and monitors, and so on.
Problems associated with the instruments themselves
sometimes show up in similar fashion. It’s quite common for
a poorly tuned or damped snare drum to have a noticeable
and unpleasant ‘ring’. Watch the spectrogram and you’ll see
narrow peaks in two or three very specific frequency areas.
With a parametric EQ on the snare track or the drum bus,
we can apply narrow, sharp cuts centred on the offending
frequencies and reduce the ringing without otherwise
changing the sound of the drum. A badly set up bass guitar 6 -Vo i c e P o l y s y n t h
or a cheap amplifier can give rise to similar problems,
whereby one note seems to jump out compared with the
others. Recordings of the human voice, too, can exaggerate
unwanted honks and resonances in the midrange, and
sympathetic narrow-band EQ cuts can often minimise these
without hugely affecting the overall vocal tone. THE
Where a problem really is confined to a single frequency,
it may even be possible to use what’s called a notch EQ. This POLYPHONIC
UPRISING
can be thought of as an approximation to a parametric band
that has a bandwidth of zero and an infinite negative gain,
and is useful in dealing with problems like mains hum. A 50Hz
mains hum will typically manifest itself as sine waves at
50Hz, 100Hz, 150Hz and 200Hz; notching out each of these
separately will have minimal impact on the overall sound,
whereas using a high-pass filter or a single parametric band to
address all of these frequencies will lose all the wanted bass
£459
inc VAT
content in the signal.

Cutting Remarks
With musical signals, a narrow bandwidth means the EQ
affects some notes more than others. For example, if
you tightly focus an EQ cut at 220Hz, you’ll reduce the
fundamental frequency of the A below middle C. The C itself
will be less affected, whilst the A an octave down will retain
its fundamental frequency unchanged, but its first harmonic
will be reduced in level. This can be useful if, for example,
an instrument has a pronounced resonance on one or two sales@nova-distribution.co.uk
tel: 0203 150 4446

w w w. s o u n d o n s o u n d . c o m / February 2022 77
 T E C H NI Q UE
EQ : W H AT DO A LL T HOS E K NOBS DO ?

notes, but also brings obvious potential

for things to go wrong. If you often find
yourself making narrowish cuts at the
same frequency, on multiple instruments,
it might be worth checking whether the
fault lies in your monitoring rather than the
recordings themselves!
Broader parametric EQ cuts have
numerous uses. For example, guitar amps
can often sound a bit shrill or harsh in the
upper midrange, and shaving a few dB
off somewhere in the 2-4 kHz area can
ameliorate this. And although ribbon mics
can make fine drum overheads and room
mics, in my experience they are frequently
a bit too present in the 1.5kHz region and
benefit from a slight cut thereabouts. Close
mics on bass drums and toms quite often
sound ‘cardboard-y’ on their own, and you
can sometimes help this by scooping out
the lower midrange with a cut somewhere
around the 500-700 Hz area. Acoustic
guitars recorded with pickups rather than
mics tend to sound very unnatural, and here
too the midrange can be overpowering.
A handy but not infallible trick for
identifying the best frequency region to
cut is to temporarily set the EQ band to
a fairly hefty boost instead. You can then
sweep the frequency control up and down
to find the point where the problem you’re
hearing gets worse. Be warned, though,
that absolutely any frequency region
tends to sound bad when you boost it
indiscriminately, and some self-discipline is
required to avoid creating more and more
bands to deal with things that aren’t actually
a problem.
Behind The Mask
The distinction between ‘corrective’ and
‘creative’ equalisation is a very fuzzy one,
not that that’s something to worry about.
Making cuts in the lower midrange across
multiple sources is often helpful when it
comes to getting them all to work together
at the mix; you’re not so much improving
the sound of individual instruments as
crafting a compromise arrangement that
benefits them all. Likewise, cutting some
upper midrange from an electric guitar
sound might actually make it less impressive
in isolation, but from the point of view of
making the mix work, perhaps that is what
is necessary to help it sit behind the vocal
rather than fighting for the same space.
The idea that elements within the mix
should not overlap too much in terms of
occupying the same part of the frequency
spectrum has been the focus of much mix
advice over the last few years, and there
Before & After
One of the most common questions asked by
novice mix engineers concerns the order in
which processing should be applied. Should you
place your EQ plug-in before your compressor in
the signal chain? Or after?
The answer is: It depends.
The usual reason why this dilemma arises is
that people haven’t paused to consider two other
questions: why does this source actually require
EQ or compression at all? And what am I hoping
to achieve by using it? Having clear answers here
will often suggest the most appropriate order, if
indeed both are really needed.
For example, let’s suppose we decide that
a vocal needs some dynamic control from
a fast-acting compressor, as well as a 100Hz
high-pass filter to cut out plosives and other
Acustica Audio’s Lime is one of many ‘channel strip’ plug-ins that allow you to change the order of
the processing elements. In this setting, the red display reads ‘F-E-C’, indicating that the filters are first
in the chain, the EQ second and the compressor last.
are even automated AI-based plug-ins
that will analyse individual instruments
in your mix and tell you where overlaps
are occurring. When these overlaps are
problematic, the result is a phenomenon
known as ‘masking’, whereby elements
obscure one another. If this happens, it
can be helpful to apply EQ cuts to different
areas in these sources to reduce the
degree of overlap. In the example above,
we might cut 2kHz from the electric guitar
and 400Hz from the vocal to help them fit
together better.
However, it’s important not to be led into
the unconscious inference that frequency
overlaps are always bad. Eliminating them
can help us achieve clarity at the mix, but
clarity itself is only a means to an end. The
pursuit of clarity for its own sake often
low-frequency noise, and a 1.5kHz EQ boost to
help the upper midrange cut through the mix. If
we place this EQ before the compressor, it will
also shape the side-chain signal that triggers
gain reduction. Compared with placing it after
the compressor, we’re thus more likely to get
gain reduction triggered by peaks in the 1.5kHz
range, whereas plosive pops are less likely to
trigger compression.
Since the aim of the compression is to
achieve a subjectively more consistent vocal
level, my instinct in this case would be to put
the EQ before the compressor. The point of
the high-pass filter is to remove the plosive
pops altogether: if we allow them to trigger
compression first, we will in effect hear the
compressor acting on sounds that aren’t there!
leads to sterile, thin-sounding mixes. It
also closes off creative possibilities, such
as layering multiple instruments with the
specific goal of having them be perceived
as a single sound. Phil Spector’s ‘wall of
sound’ was anything but clear, but it was
mighty effective!
Booster Shots
Novice mix engineers are sometimes
advised never to use EQ to boost, only
to cut. Personally, I think this advice is
outdated. It was perhaps a good rule of
thumb when people were mixing on cheap
analogue consoles that lacked internal
headroom, because EQ boosts on those
mixers often sounded strained or introduced
distortion. But a modern DAW has basically
unlimited headroom, and plug-in EQ can

78 February 2022 / w w w. s o u n d o n s o u n d . c o m
 be as clean as you like even when applying
large EQ boosts.
However, one reason for being careful
about EQ boosts is still as valid as it ever
was. By definition, an EQ boost makes the
source louder, and often brighter too — and
that in and of itself tends to make things
sound more impressive. So when you do
use EQ to boost, it’s a good idea to use
either the channel fader or the output gain
control on the EQ plug-in to try to match
the apparent level to how it was before.
That way, you can make an informed
judgement as to whether the EQ really is
doing something useful, or whether you
should just have pushed the fader up in the
first place.
We’ve already looked at the possible
benefits of boosting with a high shelving
EQ across the master bus or many of the
sources feeding it. One or two parametric
bands can often be put to good use in the
same role, the key being to keep the Q
value and gain both relatively low. In fact, I’ll
often combine a shelving and a bell boost
on the master bus. For example, a shelf
could be used to lift everything from 2kHz
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up by a couple of dB, whilst a parametric
band adds an extra focus on the 5kHz or
10kHz region. But in this context above all
it’s vital to be cautious about perceiving
louder and brighter as better. It’s also an
extremely good idea to A/B your mix with
known reference material at the same level.
On individual sources, one way to
think of EQ boosts is as a more focused
alternative to pushing up the fader
when you want something to be more
audible. Let’s say, for example, that your
vocal is being overshadowed by guitars,
keyboards or other instruments. Pushing
up the fader makes every frequency in
the vocal louder; this might achieve what
you want, but in a busy mix, you might
achieve just as much cut-through with
a gentle boost at 1.5 or 2 kHz, without
unnecessarily emphasising aspects of
the vocal that are already prominent
enough. Likewise, if you want a ‘subby’
kick drum sound to be audible on small
speakers, simply raising the level can be
problematic, because you’ll be bringing
up the low end to the point at which it’s
overwhelming on large speakers. Better
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to use EQ to boost 100 or 150 Hz. The
same applies to other bass instruments.
For example, a gentle boost somewhere
in the 400 to 800 Hz region can help
a double bass to sound more articulate
and present in the mix, without also
bringing up the woofy low end that will
throw things out of balance.
Keep It Simple
Whether you are boosting or cutting, you’ll
sometimes end up with EQ settings that
look wrong, or which are the exact opposite
of the settings you’ve used previously on
the same instrument, or which violate some
Internet ‘rule’. No matter. If it sounds right,
it is right. As with so many other aspects
of mixing in the digital age, though, the
easiest way to mess up with an equaliser
is to overcomplicate things. You should
never be afraid to use EQ if it helps you
to achieve the results you want — but it’s
seldom a good idea to use EQ when you
don’t know what results you want! EQ is an
invaluable tool, but you may well find that
the better you get at mixing, the less EQ you
actually do...
Image shows The Ultimate Collection Lyra 2
 ON TE S T
M AT T H O U G H TO N
W
e’re spoiled for choice when
it comes to EQ plug-ins. Even
those that come bundled with
our DAW software work well and have
extensive feature sets, or offer a more
than adequate feature set. But Eventide’s
new SplitEQ is no normal equaliser.
Supporting the usual plug-in formats
for Mac and Windows and authorised
via iLok (no physical dongle required),
it seems to be underpinned by the
transient-separation technology Eventide
first employed in their Physion plug-in
(see Paul White’s review in SOS June
2017, written before its name change:
www.soundonsound.com/reviews/
eventide-fission). And while the idea
of separately EQ’ing the transient and
sustain portions of a sound isn’t new in
itself (I can’t be the only one to have used
transient shapers and EQs in parallel) it’s
the first time I’ve encountered an insert
plug-in that takes care of all this task in
one place. That convenience makes it far
more than a problem-solver...
Signalling Intentions
Conceptually, SplitEQ starts with
a Structural Split process: it divides
the incoming signal into two separate
80 February 2022 / w w w. s o u n d o n s o u n d . c o m
EventideSplitEQ
Transient-shaping Equaliser
An EQ that gives you separate control over the transient
and sustain portions could be more useful than you think!
processing paths, one of which operates
on the Transients and the other on the
sustain or, to use Eventide’s term, the
Tonal part of the signal. Each path flows
through an EQ stage, with six configurable
bands plus dedicated high- and low-pass
filters. Each band also has an L-R pan or
M-S stereo width control (though both
can’t be used simultaneously for the
same band), before the two paths are
recombined to form the output signal.
Useful global controls including
separate but linkable ±30dB level controls
for the Transient and Tonal signals,
a ±200-percent EQ scaling facility (with
separate linkable controls), a pan/width
stage for each path, a global bypass and
another bypass for the EQ only; engage
the latter and SplitEQ can behave more
like a conventional transient shaper.
The split process sounds transparent
to my ears, in that when no EQ or stereo
manipulation is applied and the global
Transient and tonal faders share the same
setting, the signal at the output sounds
the same as at the input. This splitting
process is performed automatically but
while you don’t need to tweak anything
for it to do its thing there are some useful
controls for refining the split.
First, at the bottom of the GUI, you
can choose different transient detection
presets, each fine-tuned to suit a different
source (Full Mix, various drum/percussion
options, Bass, Piano/Synth, Guitar, vocal
and General). Helpfully, you can lock (or
unlock) this parameter once selected:
choose a different global preset and this
selection will remain unchanged if locked.
A Transient Separation parameter
determines the point after a transient
is detected at which the source is split,
and Transient Decay acts like a release
control: longer times lengthen Transients
and the attack/swell phase of the Tonal
path. Finally, a Smoothing control can
help you solve problems: lower values
give greater differentiation, but if set too
low, sound abrupt; higher values sound
smoother but make the paths increasingly
similar. Thus far, I’ve felt the need to tweak
this only rarely, but it’s good to know the
facility is there when needed.
Eventide recommend starting by
choosing an appropriate source preset
before moving on to these ‘refinement’
controls to target the specific source in
question, and I’d agree that this is the best
approach. To aid your decision making
when selecting and refining these presets
you can solo either the Tonal or the
Transient signal, and this solo auditioning
process is essential if you’re to get the
best results from this plug-in.
Band Controls
Once you’ve determined how the signal
is best split, you can turn your attention to
manipulating each path. As
I mentioned in passing, there “By boosting only in the Tonal path of
are eight bands in total: low-
and high-pass filters along a vocal part you can change the overall
with six more configurable
bands, each of which can
brightness without bringing up much in the
be set to act as one of six way of sibilance or lip smacking.”
types (Low/High Shelf, Peak,
Notch, Tilt Shelf, or Bandpass).
For each of the eight bands you can
choose independently whether the EQ is
applied only to the Transient signal, only
to the Tonal element, or to both paths
simultaneously. Obviously, operating on
both from the outset means SplitEQ acts
as a regular EQ, but note that if you’ve
previously adjusted the Transient and/
or Tonal paths, you can boost/cut both
together, so that the difference between
them is preserved. As with the global
signal, you can solo the Transient or Tonal
part in each band, to aid you in making
these adjustments.
You have the option of making your
tweaks by click-dragging on, or typing
in the parameter fields at the bottom
of the GUI or, my general preference,
click-dragging nodes on the GUI’s EQ
curve and using key modifiers to tweak
certain parameters for the selected band.
The way in which you select the nodes is
pretty slick: hover over a band’s default
grey/white node, and green and blue
wings/tabs appear above and below it;
click-drag the grey part to manipulate
the whole signal for that band, the green
wing to change only the Transients, or
the blue one for the Tonal path. (This
colour-coding is reinforced throughout
the GUI, making it all very easy to digest).
Once you’ve adjusted the Transient or
Tonal component, you can click anywhere
between them in the highlighted band to
act on both simultaneously. Modifier keys
allow you to adjust the Q (Shift-drag), reset
a node (Alt/Option-click), or make finer
adjustments (Ctrl-drag).
There are some small limitations but
nothing to really put me off, and some
could be addressed easily in a future
version. For instance, while you can set
the Q and gain for the main six bands’
Transient and Tonal paths independently
(I’ve often found that I want to boost one
and cut the other, each with a different
Q), those paths always share the same
frequency. The high- and low-pass filters
can operate on different frequencies for
each path, and extending this approach
to the six configurable bands would make
SplitEQ considerably more versatile.
A minor gripe was that I couldn’t find
a modifier key to lock the frequency
while dragging up/down; you can of
course use the lower section’s controls
to achieve the same thing. And while you
can use the Alt/Option-click command
to snap a node back to its zero position,
I’ve sometimes wished for a similar way to
snap the Tonal node to the Transient one,
or vice-versa (irrespective of whether that
position is zero). But, as I say, these are
small details, and on the whole it works
very smoothly.
Split Siding Away
As well as the EQ, each band offers you
a means of adjusting the stereo image. At
the bottom-left of the GUI you can switch
the lower section to display these controls
instead of the EQ ones. (They don’t take
up a huge amount of space, and I wonder
whether it could be useful to have
the option to display both control sets
simultaneously?) Essentially, you have the
option of panning each path left or right,
or of changing the Mid-Sides balance of
either path, or both of course. You can
also pan/widen the two paths globally (all
bands at once).
This might seem a curious facility
to include on an EQ, but it’s potentially
very useful here, simply because of the
transient-based split. For instance, by
panning the Transient and Tonal elements
of a mono acoustic guitar recording, I was
able to create a lovely, natural-sounding
sense of stereo width — it’s nice to
be able to do this using the ambience
already baked into the recording rather
than relying on artificial reverb — in a way
that didn’t seem to cause problems
for mono playback.
Split: The Difference?
So far, I’ve written mostly about SplitEQ’s
functionality, but not a huge amount about
what you might actually want to use it
for, and there’s a vast range of potential
applications. While we often think of
transients as being the most fleeting of
moments, and maybe comprising mostly
noise, SplitEQ allows you to define fairly
precisely what constitutes the ‘transient’,
and depending on the split
settings it can be surprising
just how much difference
boosting or cutting at any
given frequency in one path
or the other can make.
Perhaps the best way
I can convey this in words is
to describe some examples, and I’ll start
with vocals. By boosting only in the Tonal
path of a lead vocal part you can change
the overall brightness without bringing
up much in the way of sibilance or lip
smacking. Alternatively, you can apply
that boost like a regular EQ, and then pull
down any unwanted transient details that
boost brings up without compromising
that overall lift. I can’t think of any other
EQ that can do that; I’ve used multiband
transient shapers and dynamic EQs to
achieve similar ends but the former is
Eventide SplitEQ
$199
PROS
• Very intuitive and sounds good.
• Transient/Tonal boundary can be
defined to suit the source.
• So many potential applications.
CONS
• None of note.
SUMMARY
SplitEQ offers the user separate
control over the transient and sustain
aspects of a signal, and the potential
uses, both corrective and creative, are
probably greater than you think.
w w w. s o u n d o n s o u n d . c o m / February 2022 81
 ON TE S T
EVENTIDE SPLITEQ
Analyser
The GUI includes a frequency analyser with several options. It can display the
whole signal (summed), the Transient or Tonal signals, or both of those, and
can display the pre- or post-EQ signal (though not both). The resolution can
a blunter tool and the latter requires more
fine-tuning. Alternatively, if you have
layer upon layer of background vocals, I
find you can sometimes get an unhelpful
build-up of consonants and mouth noises
that undermine the sense of timing. Pulling
down distracting frequencies only in the
Transient path can be a wonderfully subtle
way to address that.
Drums are an obvious candidate for
any transient-based process, of course,
and SplitEQ can be really effective here.
For example, my ears often seem to be
offended by too much of the 200Hz
region of an acoustic kit’s kick drum, and
I almost always end up with a fairly narrow
cut somewhere around that area. But it’s
rarely a perfect solution. The ability to
pull down only the sustain of the kick at
that frequency — or to pull it down more
than I do the transient — makes it easier
to achieve what I want with less collateral
damage. Higher up the spectrum, a boost
to only the Transient path can be great if
you need to bring out more of the beater
click without raising unwanted content
spilling onto the mic from other drums.
Drums, toms and snare can be
manipulated in similar fashion. In particular,
if you judge the split settings nicely, it’s an
incredibly useful tool for pulling annoying
resonances out of tom-toms without
robbing the initial hits of their impact. But
SplitEQ also has plenty of potential for use
on mixed drum signals too, whether they
be drum loops or the drum bus in a mix.
82 February 2022 / w w w. s o u n d o n s o u n d . c o m
be adjusted (a trade-off between the frequency of updates and the amount of
detail) and the decay rate can be adjusted. A Freeze function sets this decay
time to infinity, so you can easily home in on any problem frequencies you hear.
For example, I often use tape emulations
on the drum bus but this can often have
an undesirable softening effect on the
cymbals. I really appreciate the control that
SplitEQ gives me when EQ’ing ‘into’ the
tape emulation, to manage that side-effect.
Perhaps less obviously, SplitEQ can
also be a really helpful tool when it comes
to EQ’ing one source to create space for
another, and not least in those regions
where instruments often tend to clash with
vocal parts. Take that 2.5-ish kHz area in
electric guitars, for example, which can
help them cut through a mix (particularly
when heard in mono, which often leaves
impressive, aggressive wide-panned guitars
sounding rather duller than in stereo) but
can sometimes also mask details in a vocal
that are important to the lyrical intelligibility.
If you get the separation/overlap between
the Transient and Tonal paths right, you’ll
find that you can often either boost the
Tonal path of the guitar in that area, making
it stand a bit prouder in the mix without
suffocating important vocal consonants, or,
if the guitar is present enough already, you
can pull down just enough of the Transient
that you can hear the lyrics more clearly
without unduly compromising the guitar.
While I’m on the subject of guitars, SplitEQ
can be great for ‘de-quacking’ piezo
pickups or controlling unwanted ‘boom’ in
acoustic guitar recordings.
Finally, while I don’t consider myself
a ‘mastering engineer’ as such, I have
tried a spot of DIY mastering with Split
EQ and am convinced it could have
applications in this role too, particularly
for those projects which need a little more
by way of corrective work, whether that
be taming boomy bass instruments, or
lifting up the high end without highlighting
unwanted transient details or making
things sound too ‘brittle’.
Verdict
I could discuss more examples but these
ones should provide enough of a picture.
The smoothness and precision of the
transient detection process is impressive
and, while the source presets are well
judged, for the best results it’s important
to use the path Solo buttons and fine-tune
the split by ear. Do that and, for corrective
work, it’s often possible to get away with
much less radical moves than when using
a conventional EQ. Meanwhile, when it
comes to gentler tonal shaping there’s
oodles of potential, but I’ve particularly
appreciated how smooth a high-end boost
it can deliver, and the per-band, per-path
pan and width controls are icing on the
cake. It won’t replace my preferred EQ
(which supports Mid-Sides and dynamic
bands, among many other features) but
it will join it, and I’m certain that I’ll be
reaching for it on pretty much every mix for
the foreseeable future.
£ Full price $199. (Discounts were available
when going to press).
W www.eventideaudio.com
ANDREW DUDMAN
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 ON TE S T
API Select T25
Dual-channel Valve FET Compressor-Limiter
API have been making excellent solid-state outboard
and consoles for decades, but their latest venture sees
them using valves for the very first time.
HUGH ROBJOHNS
M
ost of our readers will know
that API’s famous console and
outboard gear designs employ
the revered 2520 discrete solid-state
op-amps and related designs, and that
these play a huge role in the distinctive
sound character associated with that
marque. The company’s main range
continues to draw on those design
traditions but they’ve also been keen
to allow their engineers more freedom
to experiment with other technologies.
Thus, last year, they announced a new
API Select brand, which hosts some
refreshed designs from the company’s old
84 February 2022 / w w w. s o u n d o n s o u n d . c o m
JDK-badged sub-brand along with some
entirely new creations, notably the T12
mic preamp and T25 compressor-limiter.
Both of these are hybrid devices which
combine conventional solid-state
op-amps with valve circuitry, and it’s the
latter that I’m reviewing here.
Construction & Circuitry
A block diagram included in the user
manual reveals how these different
technologies are employed in the
T25. The audio signal path starts with
an electronically balanced solid-state
input stage (presenting a 20kΩ
input impedance). This drives a FET
gain-reduction element which is driven by
an RMS-sensing side-chain connected in
a feedback arrangement.
A Class-A valve amplifier employing
a 12AT7WC (ECC81 equivalent) provides
the make-up gain, while a beefier 12BH7
drives a custom API output transformer.
Both valves are Electro-Harmonix
dual-triodes plugged into good-quality
sockets, and I spotted a number of
standard NE5532s and TLO82 op-amps
among the solid-state circuitry, with the
former appearing to serve the audio
paths and the latter probably in the
metering and side-chain sections.
A single, selected, NTE457 N-channel
J-FET provides the gain-reduction
element for each channel, each one
being mounted on small plug-in circuit
board. Most of the circuitry is carried
on a single PCB covering the whole of
the chassis base, and surface-mount
components are widely employed.
The user controls and LED metering
are built onto secondary circuit boards
mounted behind the front panel, and the
whole construction is tidy, robust and
well thought-out.
Operation
The T25 is a dual-channel compressor
that’s encased in a substantial 2U
rackmounting chassis with a smart
blue front panel. The controls for the
two channels are laid out identically,
with plenty of space and clear function
labelling, although as there are no
intermediate scale markings on any of
the controls you have to count and note
detent clicks to recall settings.
Perforated metal grilles allow each
channel’s 12AT7 valve to be seen
(the 12BH7s are towards the back of
the circuit board and harder to view),
and LEDs mounted in the centre of
their valve-holders provide a cosy (if
misleading) golden glow.
Moving left to right, the first vertical
pair of rotary switches control the attack
and release time-constants, both with
11 settings. The attack time spans an
extremely fast 0.02ms (20 microseconds)
down to a still very quick 0.8ms, while
the release time ranges from 50ms
to 1.2 second; there’s no auto-release
mode. The ratio control is a four-way
rotary switch marked 2:1, 4:1, 8:1 and 20:1,
although this device is generally rather
more assertive than those labels might
suggest (as I’ll explain in more detail
below) and I noticed that the threshold
changed over roughly a 6dB range with
different ratio values (lower for the 2:1
setting and higher for 20:1).
The lower part of the panel carries
input and output level controls, both
having 31 detented actions and covering
the range from fully off up to +20dB. Unity
gain is with both knobs set around 10
o’clock. The output level knob doubles
up as the gain make-up control, while
the input serves as the threshold,
both functions being identified with
secondary labels. As the T25 has a fixed
internal compression threshold level
(slight changes introduced by the ratio
control excepted), to introduce more
compression you turn up the In/Thresh
control to push more of the input signal
over the threshold. I found this input
control to be surprisingly sensitive,
API Select T25
£1999
PROS
• A valve compressor from API is
inherently intriguing!
• Harmonically rich signal path.
• Hybrid circuitry.
• Separate GR and output meters.
• Dual-format I/O.
• Thrust side-chain filter included.
CONS
• Compression ratios all tend towards
limiting rather than the stated values.
• No scale markings on any controls.
• Detented input and output controls
make stereo gain matching tricky.
SUMMARY
API’s new Select product range takes
the brand into new directions, starting
with the T25 hybrid compressor.
and rarely had it above 11 o’clock when
working with nominal line levels.
A vertical 10-segment bar graph
indicates applied gain reduction
over a 24dB range, while a round,
vintage-style, backlit VU meter displays
the output level. An associated peak-level
LED illuminates at +18dBu (6dB below
w w w. s o u n d o n s o u n d . c o m / February 2022 85
 ON TE S T
A PI S E LECT T 25

On the rear panel, the inputs and outputs are presented on parallel-wired XLRs and TRS jacks,
and each channel’s output can thus feed two separate destinations simultaneously.

clipping), and although the meter’s 0VU When working with stereo signals the such designs, both channels’ controls
mark is aligned at the factory to +4dBu, Link button uses the strongest side-chain must be set exactly the same to ensure
an internal jumper allows that reference control signal from either channel, sharing equal responsiveness on both sides.
to be changed to +14dBu for those who it with both so that the stereo image A bypass button connects the input
prefer to work with hot signal levels. doesn’t pull to one side or the other when sockets directly to the corresponding
Using an Audio Precision test system, gain-reduction is active. As with most output sockets via a relay, which means
I measured a low THD of 0.05 percent at
an output level of -10dBu, but that rose
progressively to 0.5 percent at +4dBu,
1.2 percent at +10dBu, and 3 percent
by +14dBu — so if you’re looking for
old-school character, there’s plenty of it
available here. Regarding the metering
arrangements, I really like that the output
level and amount of gain reduction are
indicated at the same time; so many
compressors force the user to select one
or the other on a shared meter!
Five buttons under the meter activate
sealed relays to introduce two different
side-chain filters, soft and hard bypass
modes, and link together the side-chains
of both channels for stereo applications. This plot shows the transfer curve with different ratio settings. The top red line is with the compression
The unit’s on/off button is over on the switched out, revealing some valve/transformer saturation creeping in at higher levels. The four almost
right-hand side of the chassis, and I horizontal lines are from the four ratio settings, clearly demonstrating how the selected ratios all quickly
trend towards limiting as the signal rises more than about 6dB over the threshold. The way the threshold also
second API’s recommendation to allow
changes with the ratio can also be seen. The lowest curve is the 2:1 setting, which trends to more than 8:1.
15 minutes for everything to warm up and
stabilise before serious use.
Most compressors offer a high-pass
filter to desensitise the side-chain at low
frequencies, making them less likely to
‘pump’ on kick drums and bass guitars,
but API’s acclaimed (and patented) Thrust
design is rather more ingenious than
that — you can find out more about that
in the separate box. A second side-chain
filter option labelled D-S is a de-esser
mode, and the Thrust and D-S filters can
be used together, if desired. The de-esser
employs a narrow 6kHz boost in the
side-chain to increase the compressor’s
sensitivity to sibilant frequencies, thereby
ducking them a little more assertively This plot is an expanded version of the previous chart, revealing more clearly an initial soft-knee
to reduce sibilance. character which acts over a few decibels before flattening into a high-ratio slope.

86 February 2022 / w w w. s o u n d o n s o u n d . c o m
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 ON TE S T
A PI S E LECT T 25
the unit can be powered down to
preserve valve life while maintaining
a signal path and avoiding the need
for re-patching. However, there’s also
a ‘soft bypass’ mode which effectively
turns the T25 into a characterful line
amp: if the In button is not pressed the
side-chain control signal is disabled, so
there’s no gain-reduction but the signal
still passes through the entire circuit path,
and the valve and transformer sonics can
be used as a signal conditioner. Flipping
the In button also makes it practical to
compare the sound with and without
compression, of course.
Audio connections for all inputs and
outputs are presented on rear-panel
XLRs with TRS quarter-inch sockets
wired in parallel, so you can choose
which you prefer to use and, although
only one input connector format can
be used at a time, both output formats
can be used simultaneously to feed the
same signal to two destinations. Mains
power is provided to the hefty internal
linear PSU via a grounded IEC inlet
complete with a fuse-holder and voltage
selector (115/230 Volts AC), and the unit
consumes 45 Watts.
In Use
As well as trying the T25 on a range of
material, I performed standard bench
tests, which revealed that its audio
Full Thrust
Unlike the conventional high-pass side-chain
filter offered by many compressors, API’s
acclaimed Thrust filter is essentially a gentle
3dB/octave spectral tilt, which reduces low
frequencies and boosts high ones. It can be
thought of as having the inverse frequency
response to pink-noise: with typical music
each octave band typically has half the
amount of energy as the octave below, and
in a conventional RMS-sensing side-chain
the lower frequencies tend to determine the
amount of gain-reduction.
The idea of the Thrust filter is to give
each octave band the same amount of
energy as every other octave band, and
so the side-chain derives the necessary
gain-reduction more equitably across
the whole spectrum. As a result the
compressor’s sensitivity to low frequencies
is reduced, with less ‘pumping’ driven by the
bass line, while the more even-handedness
across the whole spectrum gives a greater
sense of overall dynamic ‘punch’ and,
simultaneously, the output sounds less
heavily compressed than it otherwise would.
It’s a clever, pleasingly effective solution.
88 February 2022 / w w w. s o u n d o n s o u n d . c o m
A plot showing how the Total Harmonic distortion varies with input level. Note how the
distortion rises with increasing level.
bandwidth (with no gain reduction)
has -3dB points at 6Hz and around
90kHz, which is impressive. Relative to
+4dBu at the output, the signal-to-noise
ratio measured 80dB, and crosstalk
between channels at 10kHz was
a healthy -101dB.
Driving the input quite hard with
the output backed off, intentionally to
‘push’ the thermionics and transformers,
I achieved some really nice, musically
pleasing saturation, even without
engaging the compressor. As I mentioned
earlier, there is more than enough
‘warm analogue character’ available
here to satisfy the strongest digi-phobe,
and I found myself running all sorts of
stereo mixes through it just to fatten and
warm them up. Unfortunately, though,
I found precise matching of channel
gains for stereo tracks a little tricky,
because the detented input and output
level controls impose relatively large
adjustment increments.
Most FET compressors are inherently
fast-acting and colourful, but the API
Select T25 goes further than most in both
respects. However, I found it very hard
to dial in modest amounts of more subtle
compression, and when I explored the
dynamic behaviour on the test bench
I discovered that once the signal exceeds
the threshold by about 6dB, all four ratios
very quickly become extremely high.
Even the 2:1 setting soon tends towards
8:1, and the higher three settings soon
reach ratios of at least 15:1. So the ratio
control has almost no effect on the nature
of compression outside of a fairly small
sweet spot: when you have a signal
that’s already fairly controlled in terms
of dynamic range, you can experiment
with the input level to achieve subtly
different compression effects in that
region, but push it further and the T25
generally behaves more as a limiter than
as a compressor,
With that in mind, I wouldn’t really
recommend the T25 for use as a subtle
bus compressor to glue things together.
Yet, it certainly is powerful and effective
on heavily dynamic instruments such
as basses, and on percussive sources
like drums (both the individual close
mics and stereo kit overheads); anything
that requires both assertive control
and character. The attack range allows
precise control of the transients, while
the release is fast enough to make
the result as dynamically bouncy and
rhythmic as you like. The Thrust and
D-S modes are also useful and effective
additions, and I found myself using the
Thrust mode a lot.
Verdict
The new API Select range is interesting
in that it allows the company space to
explore new directions of technology
and sound texture, and while there
may be a few foibles in the way the
T25 works that mean it’s probably
not the best choice as anyone’s first
compressor (easier access to more subtle
compression settings would address
that) it could be argued that they add to
its character rather than detract from its
abilities — it is a powerful tool that can
certainly be used to great musical effect
in the right circumstances.
£ £1999 including VAT.
T KMR Audio +44 (0)20 8445 2446
E sales@kmraudio.com
W www.kmraudio.com
W www.apiaudio.com
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 ON TE S T

Can a solid-state mic

really give you the
sound of an AKG C12?
JZ Microphones
NEIL ROGERS Vintage 12
L
atvia-based JZ Microphones
have been producing their
quirky-looking mics for well
Large-diaphragm Cardioid Microphone
over a decade now but, in the UK at
least, it’s only in the last few years its inspiration. For one thing, it’s
that more of a buzz has appeared not a valve mic; and it uses a
around their products. This is mostly centre-terminated capsule rather
thanks to their Vintage series than the edge-terminated AKG-style
mics, which a number of respected design. This capsule also uses JZ’s
engineers claim offer sound quality proprietary Golden Drop sputtering,
beyond what you’d expect from whereby gold is deposited in
their price tag. Their Vintage 67 small circles. Compared with
is the model that I’ve heard being traditionally made capsules, this
talked about most, but they also design lowers the mass of the
make a mic called the Vintage 47. diaphragm, and is said to improve
As their names suggest, both are transient response. Solid-state
inspired by classic Neumann mics. electronics have practical benefits
For review here, though, we too — they tend to have lower
have the Vintage 12, which takes noise, are more rugged, and don’t
the AKG C12 as its sonic inspiration. require a separate power supply
The V12 isn’t a new mic as such, — but can they really produce the
but the success of their V67 and magic that a well-designed valve
V47 models has led JZ to reissue it microphone can?
as part of the Vintage line to get it
the attention they feel it deserves.
First Impressions
Like those other models, the V12 I respect companies that do their
is not a straightforward copy of own thing from a cosmetic point
of view, and I haven’t reviewed
a microphone that looks so
JZ Microphones different from anything else in my
Vintage 12 microphone cabinet as the V12.
€1299 As with other mics in their range,
PROS the body features a physical hole,
• A lot a microphone in and what struck me most about the
a small footprint. design was just how little ‘body’
• Bright but smooth frequency there is to the V12. It really is all
response sounds superb on the
about the capsule and its basket.
right source.
• Good SPL handling. The mic ships with a small screw
• Well-balanced low-frequency attached to the top, which holds
and midrange response. the capsule in place in its internal
CONS shockmount during transit. Once
• A little fussy on some sources this has been removed, the mic
such as sibilant vocals or is designed to be used without
acoustic guitar. an external shockmount, and it
SUMMARY comes with its own ball-and-socket
The Vintage 12 by JZ attachment that allows a small
Microphones is an degree of rotation on top of a mic
original-looking and stand. I liked the look of the V12
impressive-sounding when it was in action in my live
microphone that offers many
room, and despite the minimalist
of the benefits of a high-quality
tube mic, with the conveniences design and unusual looks, it still
of a solid-state design. manages to have enough physical

90 February 2022 / w w w. s o u n d o n s o u n d . c o m

presence that an experienced studio
singer will not feel short-changed.
In Use
Female vocals seemed like a great place
to start when I first put the V12 to use, and
it was obvious straight away that this was
a mic with a generous high-end boost.
A quick look at the included frequency
response confirmed this, with a 2dB dip
at around 2kHz giving way to a broad
bell-shaped lift that begins
around 3kHz and peaks “You get a subtle sense of valve-like
around 8-9 kHz. This makes
the V12 feel quite ‘airy’ and
character to your recordings but with
‘hi-fi’, but it also manages to a more focused transient response.”
sound smooth, restrained
and, well, a bit like a good valve mic,
which is nice! Female vocals were a great
fit, which wasn’t a huge surprise, but I was
also impressed with how it sounded on
a male voice, where it exhibited a solid
low end that could be enhanced, if
needed, by working the proximity effect.
Listen Online
You can hear for yourself how the Vintage
12 performed on all the sources discussed in
the main text, at the URL below.
W https://sosm.ag/jz-12-media
The Vintage 12 isn’t only useful on vocals! Its
airy top end also makes it a good choice for sources
such as fingerstyle guitar.
On acoustic guitar, I found the
presence boost made the V12 a bit fussy
in terms of placement, although once I’d
found a position that had a nice amount
of low end, it had the effect of balancing
out the more ‘zingy’ top end that the mic
was exaggerating. As a drum overhead,
that 8-9 kHz boost was a little too edgy
on the cymbals in my room, but I was
very impressed with how it captured the
tone of the drum shells. On the snare,
in particular, it produced an impressive
representation of my Ludwig 401, with the
perfect balance of body and crack — so
much so that I recorded some extra snare
hits with a view to adding some new drum
samples to my collection.
As a last test, I tried the V12 outside
a kick drum to get a feel for the mic’s
SPL handling, and it worked very well
indeed. Placed around 2 inches from
the lower half of my 22-inch kick drum’s
resonant head, it produced a solid, clean
low thump, seemingly without breaking
a sweat.
Final Thoughts
This was my first encounter with a JZ mic,
and on any source that didn’t already
have any inherently unpleasant high mids,
I was very impressed. It delivers a subtle
sense of valve-like character, but with
a more focused transient response than
you’d get from most actual
valve mics. It was the sense
of solidity across the low end
and midrange that perhaps
impressed me the most, and
it also has that airy quality
that engineers most associate with a
good C12. On the right source — female
vocals in particular — it can excel.
Although this is not a budget microphone
by any means, it’s impressive that JZ
have produced a capsule that does
all the heavy lifting in capturing studio
recordings with both clarity and character.
The V12 is well worth checking out if you
get a chance.
£ €1299 including VAT.
T JZ Microphones +371 2947 0301
E info@jzmic.com
W www.jzmic.com
w w w. s o u n d o n s o u n d . c o m / February 2022 91
 IN S IDE T R AC K

Entertainment, SM Entertainment and YG

Taesub Lee K-pop is a crucial part of this wave, and its
place on the international music scene has
When you’re polishing grown almost exponentially. In 2020 alone, it
a nine-piece vocal band experienced 44.8 percent growth in the US.
This multi-billion-dollar industry is driven
like TWICE to K-pop by a whole swathe of boy and girl bands.
perfection, organisation First amongst them are megastars BTS,
is essential! followed by TWICE, BlackPink, TOMORROW
X TOGETHER, Stray Kids, ENHYPEN,
PAUL TINGEN NCT127, ATEEZ, SEVENTEEN and ITZY.
(In line with the brash, extrovert, colourful
Entertainment were all founded in the
’90s, while HYBE Corporation was started
in 2005. Each of these multi-million-dollar
companies was started by a famous record
producer, respectively Jin-young Park, Lee
Soo-man, Yang Hyun-suk and Bang Si-hyuk.
Apart from HYBE, they take their names
from the initials of their founders. Park, Bang
and Yang also are singers and songwriters.
We covered K-pop for the first time in
August 2018, when London mixer James
F Reynolds appeared in the Inside Track
series, talking about his mix of the BTS
hit ‘Fake Love’, the lead single of their US
number one album Love Yourself: Tear.
More than three years later, it’s time to take

T
he Korean Wave has been image that K-pop projects, many acts like to a look at the production and mix process of
sweeping the planet since the stylise their names entirely in capital letters.) K-pop in Korea itself.
early ’90s, and now challenges At the heart of the enormous proliferation Behind-the-scenes interviews about
even Hollywood for the crown of leader of K-pop acts is a Korean music industry K-pop are rare, but JYP A&R Jane Kim and
of the worldwide entertainment industry. that is dominated by four companies. JYP JYP head engineer and mixer Taesub Lee

92 February 2022 / w w w. s o u n d o n s o u n d . c o m

were thrilled to talk about their work on
TWICE’s third album, Formula Of Love:
O+T=<3, which was a number one on the
Billboard World Albums chart, and number
three in the main Billboard 200. (The
O refers to the TWICE fanbase, who are
collectively called Once, and so Once +
TWICE = heart shape, or Love).
TWICE consists of nine young women,
and the band were formed in 2015 by JYP
Entertainment. The group had their first
Korean number one in 2016, with ‘Cheer
Up’, and have enjoyed a string of Korean
and Japanese number one singles since.
The group’s debut album, Twicetagram
(2017), also went to number one in Korea, as
did their first Japanese album, BDZ (2018),
in Japan. TWICE broke through in the US
with their Taste Of Love EP, released in
June 2021 and reaching US number six,
and Formula Of Love: O+T=<3, released
last November.
Taesub Lee at JYP Studio’s Blue Room, where he mixed ‘Scientist’.
Both Jane Kim and Taesub Lee have album. I talked with the girls about what they
worked with TWICE from the group’s wanted musically, and they suggested 3:3:3
beginning. Kim, who is Korean-American, unit songs, with just three girls singing in
was born in New Jersey, and studied each, so they can have more variety in live
Cinematography, Film/Video Production concerts. We tried new things, like having
and Linguistics in New York. Following this, several English tracks to appeal to all global
she did a Master of Arts degree in A&R fans, and also doing different genres.”
and Music Business at Berklee College Several tracks on Formula Of Love were
of Music in Valencia, Spain. Kim started made with the involvement of Western
to work for JYP in 2009, initially as artist producers, including the Stereotypes, who
manager for the Wonder Girls, who went have worked with Bruno Mars and Justin
on to become the first Korean Billboard Hot Bieber; Tommy Brown, who worked with
100 artists, and pioneers of K-pop crossing Ariana Grande and Justin Bieber; and
over to the West. Since 2014 she has been British producer Shift K3Y, who worked
an A&R manager. She has worked with with Tinie Tempah.
groups like 2PM, GOT7, Stray Kids and Producers and writers who TWICE
ITZY, and is now in charge of TWICE and have worked with in the past include Zara
several other artists. Larsson, MNEK, Kim Petra and Charli XCX.
As K-pop has become more successful in
Love Me Two Times
Kim’s command of the English language
meant that she often translated for Lee
during a 90-minute Zoom conversation,
conducted from Lee’s studio in JYP’s
headquarters. Kim started by elaborating
on her role with TWICE in general, and the
making of Formula Of Love in particular.
“I may suggest a direction to the team,
and the group, but in the end we make
all decisions together,” explains Kim. “So
it always ends up being a big team effort.
[JYP founder] JY usually is very involved
with the singles, and sometimes writes
and produces them. He trusts the team to
handle the album tracks. ‘Scientist’
“A&R’ing Formula Of Love was pretty Written by Anne-Marie, Tommy
Brown, Melanie Fontana, Steven
challenging and fun. I was the only A&R for
Franks, Michel Schulz & 72
the album and worked on all 17-18 tracks, Produced by Tommy Brown, Steven
including the remix. I tried to communicate Franks & Michel Schulz
as much as possible when preparing the
w w w. s o u n d o n s o u n d . c o m / February 2022 93
 IN S IDE TR A CK
TA E S U B LE E
the west, an increasing number of Western
producers are getting involved in its making,
leading to crossovers between Korean and
Western artists.
The credited writers on ‘Scientist’ are
Anne-Marie, Melanie Fontana, Michel
‘Lindgren’ Schulz, Tommy Brown, Steven
Franks and 72, while production was by
Brown, Franks and Lindgren. Se-hee Eom is
credited as engineer. Eun-jee Sim, a Korean
lyricist, was vocal director and digital editor,
Taesub Lee was the mixer, and the song
was mastered by Nam-woo Kwon.
Winning Formula
Once the vocal parts have been divided up
between the nine singers, TWICE usually
JYP A&R Jane Kim.
94 February 2022 / w w w. s o u n d o n s o u n d . c o m
go to JYP Studios to record. This part of
the process is supervised by Taesub Lee.
“Taesub directs the engineers who work
with him,” explains Kim. “There are six
engineers, and when vocals are recorded,
Taesub will be around to give direction for
what equipment to use and how to use
it. Myself and the vocal director also are
around to guide the girls and help select
the best performances.”
Lee details that there are six studios at
JYP, consisting of “two mix rooms called
Blue and Red, and four recording rooms,
called Whitney Houston, James Brown,
Stevie Wonder and Prince. All rooms have
outboard like the Teletronix LA-2A, Purple
Audio MC-77, Neve 1073 DPA, and the
rooms share preamps like the vintage
Neve 1073 and 1084.
“The Blue, Red and Whitney Houston
rooms have Amphion Two18 monitors
with Amphion’s FlexBase25 LF extension
system, the Stevie Wonder room has
Barefoot MicroMain 26s and the Prince
and James Brown rooms have Genelec
8351A monitors. The Telefunken ELA M251
is the only microphone we use to record
TWICE. We tried mics like the Telefunken
U47, Sony C800G and Manley Reference
Gold, but the 251 worked best for all nine
girls. We run that through a Neve 1073 and
the LA-2A.”
Most of the TWICE recordings were
done in James Brown. After the recording
sessions, the vocal takes are produced
in-house, or sent to an external editor
to be comped and tuned. In the case of
‘Scientist’ the vocal producer and editor
were the same person, Eun-jee Sim, but it
may be two different people. “We like to
keep things in-house,” says Kim, “so we
have a consistent sound for recording,
mixing and mastering. JYP Entertainment
has many artists releasing albums, so the
A&Rs communicate with the producers
and with Taesub to find the right mixers for
each song. We use people from all over the
world. In the end we simply use the best
person we can find who is available. With
regards to the mixer, I asked the producers,
and in this case, they asked for Taesub.”
Seoul Man
Born in Seoul, Taesub Lee started his
musical career playing guitar in bands.
He moved to Moscow with his parents
when he was 13, and attended Moscow
State University there. Ten years later
he was back in Seoul, where he first did
military service, and then went to Seoul
Jazz Academy, the most prestigious music
academy in South Korea. He studied there
as an engineer because “there are so
many great guitarists in the world, it’d be
impossible for me to do that for a living”.
After completing Seoul Jazz Academy,
Lee worked at two different studios before
joining JYP as chief engineer in 2013.
There are seven mix engineers
credited on Formula Of Love, including

two American star mixers, Tony Maserati
and Kevin ‘KD’ Davies, while Serban
Ghenea’s right-hand man John Hanes
has an “immersive mixing” credit on ‘The
Feels’. Lee mixed three tracks. For him
to be asked to mix the lead single of the
album in this stellar company seems quite
an achievement, but Kim stresses that it’s
quite normal, and in part because Lee is
known for a “specific unique sound” which
fits the bass-heavy nature of the song, in
part due to Tommy Brown’s hip-hop roots.
“I like rock, and I am very inspired
by that,” says Lee. “So I like things to
sound loud, and noisy, but also to be
well-organised and in the right place. The
tracks need to be very balanced. When
dealing with a hip-hop-influenced track,
I think of the directness of rock. I always
try to not bring out the voice too much,
but emphasise the kick and the snare, to
make it sound more like rock music, but in
a hip-hop context.”
Out Of The Blue
The Blue room is Lee’s mix room,
where he conducts his mixes using the
aforementioned Amphion monitoring
system, and an impressive amount of
outboard including a Manley Massive
Passive EQ, Burl B2 and Crane Song
HEDD Quantum converters, Tegeler
Audio RaumZeitMaschine, Bricasti M7
and Yamaha SPX990 reverbs, Chandler
EMI TG12413 and Neve 33609J/D
compressor-limiters, an SSL Fusion
analogue colouring device, Focusrite’s Red
3 dual compressor-limiter, a Thermionic
Culture Vulture Mastering Plus compressor,
a BetterMaker Darth limiter, a McDSP
APB16 analogue plug-in box, and Empirical
Labs Distressors and a Fatso.
“I love the Amphion monitors,” Lee
explains, “They sound very clear and very
flat. With some speakers you get distracted
when the room conditions change, but
these remain very stable. I always use
some analogue outboard in my mixes.
I started using analogue gear on my mix
bus seven years ago. Plug-ins are nice, but
you can’t really add enough coloration to
textures with plug-ins, so I use my outboard
for that. The Culture Vulture is important
for adding different colours, as well as the
SSL Fusion.
“These outboard units are adding just
little bit of compression, but that gives
me a huge amount of satisfaction. The
Raumzeit is a reverb that I can control
in the box, so it’s very easy to use, and
sounds fantastic. It sounds totally different
from other reverbs. Very spacious but not
like a hall or stadium. It’s really different.
I don’t use the Massive Passive very much
any more, because the plug-in sounds as
good and is easier to recall.
“I tend to mix several songs at the
same time every day, switching between
them as I go. I receive all vocals tuned
and comped. In house we use Melodyne
Lee’s mix room houses an
enviable collection of outboard,
much of which found its way onto
his mix of ‘Scientist’.
for tuning. Tuning is the
most important part of vocal
producing these days. When
we do the tuning here at JYP,
the producer, Jane, the vocal
director and the girls and
myself all listen, and decide
what needs to be done.
Because of the pandemic we
communicate via text!”
Session View
The Pro Tools session for
‘Scientist’ is a whopping 160
tracks large, and is one of the
best-organised that has ever
featured in the Inside Track
series. Lee makes extensive
use of folder tracks, and
groups and colour-codes
everything in such a way
that it is obvious what one is looking at
with one glance. At the top of the session
are five sound effects (swooshes, risers,
and so on), grouped in and sent to an
‘FXFX’ folder track. The same is the case
with the pre-chorus bass and the main
bass tracks.
The drums are organised in subfolder
tracks, with six kick and snare tracks
going to a ‘KS’ routing folder, eight hi-hat
and cymbals tracks going to an ‘HHHH’
routing folder, a fills folder track, and
three percussion tracks going to a ‘RHY’
routing folder. All folder tracks go to
the main drums folder track, which is
called ‘DRDR’.
Below this is an instruments section,
followed by the Lead Vocals. These consist
of 18 audio tracks going to an ‘LDV’ routing
folder track. The session continues with
three VCA tracks for volume control, and
then a whopping 70 backing vocal tracks,
grouped in three sections, with backing
vocals by TWICE, Sophie Pae and Melanie
Fontana. Finally, at the bottom are 10 aux
effects tracks, two parallel compression
tracks and the master section.
“Organisation is very important!”
stresses Lee. “I have various templates.
I have template tracks for vocals, for
background vocals, and several reverb
and delay aux template tracks. I also have
specific presets for TWICE, like for fast
songs, slow songs and different genres,
for example with a focus on sung vocals,
w w w. s o u n d o n s o u n d . c o m / February 2022 95
 IN S IDE TR A CK
TA E S U B LE E
With 160 tracks to navigate, a well-organised
session is essential!
or a focus on rap vocals. Because I’ve
been working with them for so long,
I know what works. But I change these
presets every two or three years, and will
adjust the EQ and compression depending
on the song.”
Plugging In
There are very few plug-ins in the
‘Scientist’ session on the drums and
instruments, but quite a few on the lead
vocals, and the backing vocals by TWICE
and Sophie Pae. The template origins can
be seen in the fact that the signal chains
on each of the audio tracks for each group
are almost identical. On the lead vocals it
consists of the Waves CLA 76 compressor
and FabFilter Pro-Q 3 EQ on the inserts,
and four sends to the aux effects tracks:
‘Doubler’ with the SoundToys MicroShift
plug-in, UAD Lexicon 480 reverb, TC
Electronic 2290 delay, and an eighth-note
delay from the Waves H-Delay.
“I was fortunate that I did not have
to do much on the production in this
session,” says Lee. “I did a few small
things, like adjust the balance of the kick
and the bass a little, and some EQ here
and there to make space for the vocals,
but that was all. By contrast, there are
other sessions in which the instruments
need extensive work!
“In this session my main concern was
that I was dealing with 100 vocal tracks,
96 February 2022 / w w w. s o u n d o n s o u n d . c o m
which is a lot. With regards to the lead
vocals, it’s definitely different mixing nine
different singers rather than just one
lead vocal. I need to make sure that the
individual voice and character of each
voice shines through, yet also that there’s
a flow from the beginning to the end, and
you don’t lose the attention of the listener.
A lot of that is about riding the volume of
the vocals.
“With regards to the treatments,
I started the mix from Nayeon’s hook
part, and then adjusted the other vocals
from there. I have the Waves CLA-76 with
a slow attack, set to 3, and a fast release,
set to 5. My standard setting is 4 on the
attack and 6 on the release, but I had
to adjust it a bit here. The Pro-Q 3 is on
a fairly standard setting, with a high shelf
adding lots of high end. There’s an
additional Waves CLA Mix Hub plug-in on
a couple of the tracks, which is for a radio
voice effect.
“All lead vocals go to a folder track,
which has the Sonnox Oxford Inflator, to
make the lead vocals sound fatter. Below
are the TWICE backing vocals, on which
I again have the CLA-76 and Pro-Q 3,
on different settings, as well as the SSL
Channel, the FabFilter Pro-DS de-esser,
and the SoundToys Devil-Loc. The latter
is to get these vocals to sound more
crunchy, so they have a different sound
from the lead vocals. These backing
vocals also have just one send, to the
UAD 480, so they are in a different space.
Sophie’s vocals have a similar chain,
with the addition of the Waves S1 Stereo
Enhancer, to spread them wider and make
them sound fuller.
“At the bottom of the session, all tracks
apart from the drums and bass go to the
‘RearBusL’ and ‘RearBusR’ tracks [153-154],
for parallel compression with the UAD
1176AE plug-in. It’s something I adapted
from Andrew Scheps. These two tracks
then go the mix bus fader, with which
I control the volume.
“All my tracks go to track 156, named
‘AD’, because it originally was a channel
going to my Lavry 122-96 MkIII AD
converter. I didn’t use that in this song, and
instead it has my outboard Neve 33609
and SSL Fusion as inserts. That track goes
to track 157, on which I have an insert for
my outboard Culture Vulture. It also has
the Waves InPhase LT phase-correction
plug-in, to correct for latency coming from
the outboard.
“Finally, everything goes to the ‘All’
bus, in which I have the Waves PuigTech
EQP-1A and UAD Curve Bender Mastering
EQ plug-in, and then I print the mix. When
I send out the mix for comments, I increase
the volume with the iZotope Ozone 9,
but I take that off when I send the mix
for mastering to Nam-woo Kwon at 821
Sounds Mastering.”
From where the track went on to make
a splash around the world. As the Korean
Wave continues, we’re sure to hear more
of Lee, Kim and TWICE… 
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 ON TE S T
SteinbergWaveLab Pro 11
Audio Editing
Software
The WaveLab department
at Steinberg have been
very busy indeed...
JOHN WALDEN
W
hile WaveLab is well-established
as leading audio editing software,
audio formats and delivery
platforms are constantly evolving, as are
the demands placed by users on their
software tools. Thankfully for WaveLab’s
loyal following, it would appear that
Steinberg are not resting on their laurels,
on paper at least; WaveLab Pro 11 looks
like it delivers a bumper bundle of new and
improved features.
This follows some impressive advances
in the two most recent iterations of
SpectraLayers Pro and, if your audio
production work happens to span the broad
catalogue of Steinberg applications, I think
98 February 2022 / w w w. s o u n d o n s o u n d . c o m
it is a good thing to see both WaveLab and
SpectraLayers getting their share of the
development attention. So, with a long list of
‘new for 11’ with this WaveLab Pro upgrade,
let’s get started.
Ride The Wave
Over the 25 or so years of WaveLab
existence, a diverse user base have joined
the ride. This includes musicians, audio
editors in traditional radio, TV and film
broadcasting, mastering engineers, audio
analysis and audio forensics. However, the
needs of those users are always changing
while, at the same time, the potential user
base has expanded.
For example, for those working in film,
the ability to work with an ever-expanding
set of multichannel audio formats is
important. For mastering engineers (or
musicians doing their own mastering),
audio analysis and options for meeting
the different loudness standards set by
a diverse suite of streaming platforms,
have become critical over recent
years. Independent podcast producers
represent a massive new potential user
base and need a software environment
that suits their workflow and provides
export formats targeted to their different
broadcast platforms.
Existing WaveLab users can be
reassured that Steinberg have not
reinvented the wheel with Pro 11 and its core
features and major workflows (covered in
many previous SOS reviews) remain intact.
The ‘new and improved’ in v11 contains an
interesting mix of things that will appeal
long-standing and potentially new users
alike; I’ll focus on these new features here.
Improve Your Vision
WaveLab has always provided a range of
different audio metering/analysis tools but
Steinberg added a dedicated plug-in for
just this job in Cubase Pro 11 — SuperVision
— and that plug-in is now also available
to WaveLab Pro users. While SuperVision
undoubtedly duplicates some metering/
analysis options that were already part
of the WaveLab feature set, it does offer
a number of attractive workflow advantages.
For example, the layout is highly
customisable, with up to nine individual
metering/analysis modules (from the 27
module types available) being displayed
within a single instance. In addition, if you
have a multi-screen host system, placing
SuperVision on its own dedicated display
means you always have your analysis tools
available. And, as you can use multiple
instances of SuperVision, you can easily
meter before/after a key audio processor
(or processor chain) for feedback on the
changes produced.
The modules themselves provide a very
broad range of options from simple level
metering, various phase displays, a range of
different spectrum displays and a number of
different loudness options. All the modules
have their own Settings panel for fine-tuning
their behaviour. While there is plenty of
overlap between the modules available in
Cubase and those now found in WaveLab,
there are also some differences. For
example, WaveLab’s SuperVision includes
a ‘Netflix’ Loudness module and, most
welcome, a VU meter module. Hopefully,
the latter will reach the Cubase SuperVision
module list shortly. Anyway, SuperVision is
something that every WaveLab user could
potentially benefit from so it’s an addition
that will have a very wide appeal.
Plug The Gap
There are eight further ‘new to WaveLab’
plug-ins. Of these, five are also arriving
from Cubase; Imager (multiband stereo
image adjustment), Frequency 2 (multiband
dynamic EQ), Squasher (multiband dynamics
Steinberg WaveLab Pro 11
£430
PROS
• A huge number of additions and
improvements across the board.
• Group Tracks and Track Lanes add
considerable flexibility.
• Some impressive new plug-in
additions.
• Lots of new functionality for working
with multichannel projects.
• Upgrade pricing represents very
good value.
CONS
• Something of a learning curve for the
new user.
• A substantial financial commitment
for the first-time buyer.
SUMMARY
WaveLab Pro 11 is a very impressive
update that will particularly appeal to
those working with multichannel audio
or mastering projects. Pro 11 is a deep,
powerful and flexible environment for
almost any audio editing task.
New reverb reduction plug-in
DeReverb provides a flexible
means of reducing ambience in an
existing recording.
with both downwards and
upwards compression),
Quadrafuzz v2 (multiband
saturation/distortion) and
MixConvert V6. I covered
the first three of these in the
Cubase Pro 11 review, while
Quadrafuzz v2 has been
available in Cubase for some
time. Those involved with
mastering will find them all to be useful
additions, although WaveLab’s Master
Rig — which is excellent — already offers
modules that provide much of the same
processing capabilities. MixConvert V6
adds a further option for those needing
to monitor a surround audio project on
a stereo playback system. WaveLab already
has a number of options to achieve this
in the Playback Processing panel of the
Master Section, but MixConvert provides
greater flexibility.
Of the three totally new plug-ins, one
is also aimed at multichannel audio; the
impressive-looking AmbiDecoder. While
I didn’t have the chance to fully test
it, it will convert a first, second or third
order Ambisonics mix to an HRTF-based
binaural sound for headphones or
a speaker configuration.
The final two newbies are DeReverb
and LIN Pro Dithering. The former provides
a four-band reverb reduction system that,
in use, is very good indeed. It works very
well when used for cleaning up field-based
voice recordings and was impressive when
reducing existing room ambience or reverb
on stereo drum recordings. It includes
a Learn mode (which tries to optimise the
initial settings) and the ability to monitor just
the signal being removed. I can imagine
Cubase users hoping this is a plug-in that
eventually goes in their direction. LIN Pro
Dithering brings a more advanced approach
to dithering and has been developed in
collaboration with MAAT Audio (a company
with some impressive clients for their audio
processing expertise). It offers different
dither type and noise-shaping algorithms as
well as an Auto Blanking option that, when
activated, gates the dither noise during
sections of silence.
Two other plug-in-related changes are
also worth mentioning. First, the External
FX plug-in (which can assist when you
need to patch external hardware into
your WaveLab signal chain) has been
improved with new input/output meters
to help with gain-staging. There is also
an option to embed an image of your
external hardware into the window and this
could, for example, be used to record the
specific hardware control settings for later
recall. Second, all WaveLab’s plug-ins now
have a plug-in history system and an A/B
settings system. Flipping back through your
setting changes for an individual plug-in, or
toggling between two setting states, is now
much easier.
Audio In, Audio Out
If you work on audio for video, WaveLab
Pro 11’s new capability for replacing audio
in video without re-rendering the video will
be very welcome. Not only does it mean the
workflow is quicker (no video re-rendering
is required) but also avoids potential
problems caused by re-coding the video;
the replacement audio is simply embedded
alongside the original video data.
Once you have imported video into
a WaveLab Audio Montage, you can edit the
original audio using all of WaveLab’s usual
tools, or simply delete/mute it and build
a completely new audio soundtrack within
the Montage. Having tried this on a short
video clip by simply swapping the original
audio (a drum instrumental) for a new audio
clip (a different drum instrumental with
some edits added to fit the video footage),
it worked an absolute treat. Rendering
seemed to take just the usual time required
for the audio itself. For editors asked to
make last-minute changes to the audio
of a video project, this process will be an
absolute boon.
Push The Envelope
This release also adds automation
envelopes for any parameter within a VST
3 plug-in that is applied at clip level.
Right-clicking on any parameter in such
a plug-in produces a pop-up menu allowing
you to add an automation envelope for that
w w w. s o u n d o n s o u n d . c o m / February 2022 99
 ON TE S T
S T E IN BE R G WAV E L A B P R O 11
The ability to replace audio within
a video file without re-rendering the
actual video is a very welcome addition
and works a treat.
parameter. This worked perfectly
with all the Steinberg VST 3
plug-ins I tried and for some
third-party plug-ins.
To support this new
functionality, a dedicated
Automation Envelope panel has
been added within the Inspector.
This allows you to toggle the
envelope visibility between any
of the clip’s active envelopes
(one envelope is visible at any
time) and you can also bypass
an envelope. Equally, for those third-party
plug-ins that didn’t want to produce the
pop-up menu mentioned above, you can
add a target parameter within this panel.
Envelope editing is very straightforward
in terms of adding, deleting or moving
nodes and the toolset includes some useful
options of editing multiple nodes and
envelope smoothing.
If your workflow involves the Audio
Montage for assembling clips for an
album/CD project, these new clip-based
automation options are going to make
it much easier to finesse the details of
each clip to produce a coherent sound for
the entire project. I suspect this may be
a feature that Steinberg will continue to
develop but, even in this first iteration, it is
very nicely implemented and simple to use.
We Have You Surrounded
WaveLab Pro 11 delivers some significant
improvements for working with surround/
Ambisonics multichannel audio formats
in both the Audio Editor and the Audio
Montage. In the Audio Editor, multichannel
interleaved files up to a 22.2 format can be
Cast Your Net
In response to the rapid rise of the podcast,
Steinberg had recently introduced WaveLab Cast,
a streamlined (and budget-friendly) version of
WaveLab with a feature set aimed specifically at
podcast producers or video creators/YouTubers
(WaveLab Cast includes video support) who want
to ensure their audio is as good as it can be.
WaveLab Pro 11 now brings the podcast-friendly
elements found in Cast to the Pro version.
There are two obvious headlines on this
front. First, Pro now includes the direct upload
of completed podcast projects to a number of
the popular podcast hosts (Buzzsprout, Castro,
PodBean, SoundCloud and Spreaker). Second,
and also useful for tasks outside podcasting, the
100 February 2022 / w w w. s o u n d o n s o u n d . c o m
opened, edited and saved. And, given the
flexible output routing options that are also
provided (including the MixConvert and
AmbiDecoder plug-ins already mentioned),
you can still work with these multichannel
formats even if you only have a stereo
monitoring system available.
Within the Audio Montage, multichannel
files are automatically organised into mono
or stereo ‘clusters’ (for example, in a 5.1
surround format, the L+R and Lr+Rr are
each stereo clusters while the LFE and C
channels are each mono clusters). Each
cluster has its own mute/solo controls
and various editing tasks can easily be
performed on individual clusters.
Whether in the Audio Editor or Audio
Montage, the selection tools for editing
tasks are very flexible, allowing you to
select ranges on individual channels,
individual clusters, or multiple clusters.
These selection options also operate
within the Audio Editor’s Spectrogram view
for spectral editing tasks. Equally, once
a selection range has been made, it can be
moved or resized. These are impressive
enhancements and will undoubtedly
Track Inspector now has two additional sections;
Clean and Enhance. As their names suggest,
these provide easy access to tools for cleaning
(DeHummer, DeNoiser, DeEsser) and enhancing
(Voice Exciter, Reverb, EQ, Maximiser) your voice
recordings. While tools with similar functions
are available elsewhere within WaveLab Pro’s
feature set, having these dedicated panels is
a great workflow feature. And while you might
need to look beyond these somewhat simplified
control sets into WaveLab’s other plug-in
offerings if you are dealing with very problematic
recordings, for a quick polish of a voice that was
reasonably well recorded in the first place, they
really are very effective.
improve the workflow for anyone handling
multichannel surround formats for film audio
or specialist music projects.
On The Busses
WaveLab Pro 11 introduces two new features
that essentially bring audio bussing to the
Audio Montage; Track Groups and Track
Lanes. In the first of these, Tracks can be
assigned to a Group and there is a new
Tracks Group dialogue to help manage the
process of creating Groups and editing
which Tracks belong to which Group or
remain outside of a Group. A Track can
only belong to a single Group at any one
time but it is easy to move Tracks between
Groups. Groups can be colour-coded
as a visual aid in project management,
but the key feature is the ability to adjust
gain, mute/solo a specific Group, and to
apply an effects chain at the Group level.
A new Group panel is now available in the
Inspector to toggle between Groups and
to build whatever Group-level processing
chain you might require.
Beneath the Group level, stereo, mono
or reference tracks can now be divided
into up to eight Track Lanes. Lanes can
be added and removed and Clips placed
on Lanes, moved and/or copied between
Lanes and Lanes have individual mute/solo
buttons. You can also convert Lanes into
Tracks and Tracks into Lanes, with various
options for taking any applied effects
plug-ins with the conversion. Clips can be
overlapped between Lanes (effectively
layering sounds as you might for sound
design) and you can also swap Clips
between Lanes to perform track comping
tasks. Effects applied at the Track level are
applied to all Clips on all Lanes within the
Track. However, plug-ins can also be added
to individual Clips within the Lanes system,
in effect providing a further level of ‘bussing’.
 Automation envelopes
can now be created for
any VST 3 plug-in inserted
on a Clip and envelope
management is provided
by the new Automation/
Envelope panel within
the Inspector.

Clearly, these new

Track Group and
Track Lane systems
open up some fairly
complex audio routing
options and I’m sure
I’ve only scratched the
surface of what might
be possible during
the review period.
At either level — Groups or Lanes — the
obvious application is in the preparation of
audio stems (for example, separate stems
for dialogue, music and sound effects in
film/TV audio production). However, you
might also use it for preparing/processing
drum, bass, guitar and vocal stems if
a client requires that level of control for
a subsequent mixing task. Equally, you can
imagine mastering tasks where these new
features would be very useful.

Best Of The Rest

While the improvements outlined so far
perhaps represent my personal highlights of
the Pro 11 release, there is a considerable list
of further additional enhancements. Indeed,
there are too many to include them all
here, but a few deserve at least a mention.
For example, the Audio Montage now
has a streamlined audio ducking feature,
allowing you to ‘duck’ a target track based
upon the audio signal of other tracks. In
addition, under the hood, processing within
the Audio Montage is now optimised for
multicore support producing both smoother
operation and faster rendering.
Other improvements include more
Insert slots within the Master Section, a new
Start-Up Assistant dialogue with more
options, the option to add comments within
the Marker Creation dialogue, more options
when importing audio into WaveLab, and
a range of new options/settings when
rendering from the Audio Montage (some of
these related to the new Group Tracks and
Track Lane’s features).
Wave Hello?
Throughout the review period, WaveLab
Pro 11 ran flawlessly on my host system.
As with SpectraLayers Pro over the last
couple of years, Steinberg appear to have
The combination of the new Track Groups (shown here) and Track Lane systems create some considerable
audio routing flexibility for WaveLab projects.
also given WaveLab some serious love.
Speaking as a regular WaveLab user, it’s
great to see that level of commitment to
the product.
So, should you buy it? Well, for new
users coming to Pro for the first time,
there is a hefty price tag and undoubtedly
something of a learning curve. However,
for those working in a commercial context,
where time is money, it’s difficult not to
conclude that WaveLab Pro 11 is a hugely
powerful, and very flexible, platform
for editing audio; Pro fully deserves its
‘pro’ label.
For existing users, even those already
on WaveLab Pro 10, given the plethora of
new and genuinely useful improvements in
the feature set, the upgrade would seem to
represent excellent value for money. This
will, of course, depend upon exactly which
of the new features might benefit your
individual workflow but, as someone who
spends most of his WaveLab time within
the Audio Montage, I think the upgrade falls
into the ‘no-brainer’ category. I can imagine
those regularly working with multichannel
formats will have similar feelings.
Whichever way you look at it, Steinberg
have packed a lot of new stuff into
WaveLab Pro 11. This is an impressive
update with a bumper load of new features
and improvements. Is WaveLab Pro 11 for
you? Well, you can find out for yourself as
Steinberg have a free 30-day trial version
available for download from their website.
I’ve always felt WaveLab was the best
audio editor available — that’s why I’m
a longstanding user — but, for my money
at least, in Pro 11, Steinberg have made the
best even better.
£ £430. Upgrade pricing available.
Pricing includes VAT.
W www.steinberg.net

w w w. s o u n d o n s o u n d . c o m / February 2022 101

 ON TE S T
PAUL WHITE
W
hile some companies strive to
emulate hardware in software,
Italy-based IK Multimedia are
doing just the opposite with their X-Gear
range, which expands on the technology
underpinning their AmpliTube amp and
effects emulation software and puts it into
physical, DSP-powered pedals designed
for live performance. Currently the range
comprises the X-Drive, the X-Vibe, the
X-Time and now, reviewed here, the
X-Space, which as the name suggests is
designed to create reverb-based effects.
IK tell us that the DSP platform has an
open architecture so as to allow for future
updates, and I suspect that all the pedals
in this range share the same DSP platform
but run different software and firmware.
Overview
Built into a durable, blue, anodised
aluminium case and offering true stereo
operation, the X-Space hosts 16 new
reverb effects, as well as an amp/cab
simulator section with a choice of five (or
no) impulse responses, which means you
102 February 2022 / w w w. s o u n d o n s o u n d . c o m
IK Multimedia
AmpliTube X-Space
Digital Reverb Pedal
With both a hardware pedal and a plug-in version included,
this classy-sounding reverb offers more options than most.
can DI the pedal into a PA system if that’s
the way you prefer to work. The IR settings
are global and include a Bass option as
well as four guitar amp/speaker IRs.
The pedal’s audio credentials are
impressive. It boasts 24-bit, 192kHz
converters, a 5Hz to 24kHz frequency
response and a 112dB dynamic range,
while the dry signal path remains
analogue, which will please the purists.
There are options to configure the pedal
for true (hard-wired) or soft (buffered)
bypass in the setup menu, and you can
set the pedal to allow reverb tails to fade
naturally when bypassing or changing
patches. Speaking of which, the pedal
comes with 50 factory presets already
included, but it can store up to 300.
There are stereo ins and outs on
quarter-inch jacks but any permutation of
mono or stereo inputs and outputs can be
used, as required. Power comes from an
included 9V adaptor and an expression
pedal/switch input jack opens up the
possibility of real-time control for any pedal
parameter. There’s also a USB port, and
this can be used not only when running
the librarian software on a computer (see
box) or for firmware updates, but also as
a stereo audio recording interface for your
DAW. There’s also a comprehensive MIDI
implementation, with full-size five-pin MIDI
 IK Multimedia
AmpliTube X-Space
£299
PROS
• Wide range of reverb effects, both
conventional and creative.
• Easy to use.
• Excellent audio quality.
• Works as a USB audio interface.
• Amp/speaker IRs onboard.
• AmpliTube-hosted plug-in
version included.

CONS
• Nope!

SUMMARY
This hugely flexible reverb pedal is up
there with the best of them and having
a software version included for use in
AmpliTube is a genuine bonus. The X-Space can be used with mono or stereo sources and destinations.

in and out connectors, including provision
for control over IK’s AmpliTube 5. MIDI
could also be useful if you have other
X-series pedals in the chain, as you could
use the first one as a master and then
get them all to change to the appropriate
presets when you change the first.
The X-Space’s top panel is laid out in
a very clear way, so as to make setting
up your own effects largely intuitive. The
reverb algorithm types are printed below
the large LED display, which also displays
corresponding algorithm numbers along
the bottom, and to adjust the sound
there are six knobs below the display,
five of which access the most important
parameters: Time, Pre-delay (of up to
1000ms), Color (reverb tonality), Mod
and Mix. The Parameter knob on the left
adjusts deeper patch parameters such
as shimmer pitch, reverb size and so on,
 ON TE S T
I K M U LT I M E D I A A M P L I T U B E X - S P A C E

and is also used to access Global setup

parameters. (The precise parameters
available depend on which reverb
model is currently running.) Two further
turn-and-press encoders, Preset and
Model, are used for accessing and saving
presets and for selecting the effect type
or model. Presets can be saved to any
location and can be named.
Used in pairs, the footswitches may
be used to switch up or down between
preset banks. Alternatively, they can be
used individually to select one of the
three presets in each bank. Above each
switch, an indicator LED lights green if
a preset is active or amber if the preset
has been edited, and all blink amber after
changing banks. The footswitches are
also used to activate bypass or can be
held down to access the X-Mode, which
brings in an extended sustain effect for
the reverb tail. If new notes are played
after activating X-Mode, their reverb is
added to the sustain. If no lights show the
pedal is bypassed.
Librarian Software & AmpliTube Plug-in
As well as the pedal you get a software version Save As, and then name it and select where you
of X-Space. It’s hosted by IK’s popular AmpliTube want it stored. If you then switch to the X-Gear
software, which can run either as a standalone Librarian, set to use the same folder, you’ll find
app or as a DAW plug-in. It offers exactly the your saved preset easily and can transfer it over
same effects and capabilities as the hardware to the pedal. Not only is this a convenient way
and you don’t need to own the full version of to manage your presets, but it means those who
AmpliTube to use it; X-Space can be hosted like to use plug-ins while writing in the studio
in the free AmpliTube CS. Once the pedal has can export precisely the same effects as they
been registered, you can also download the used in their production to the pedal, for use in
free X-Gear Librarian software, and this makes their live shows.
it possible to move presets
between any X-Gear pedals
and AmpliTube. To save
a new preset for the pedal
within AmpliTube, you click
on the small triangular
arrow on the right-hand
side of the display window
of the pedal GUI, select
Separate librarian
software allows transfer of
presets between AmpliTube
and the pedal.

Effects & Impressions

In the Model section, there are 16 reverb
types. These include the familiar Room,
Hall, Plate, Chamber and Spring types,
of course, but you also get a Church, an
early reflections Ambience patch, Gated
reverb, Reverse reverb, and a host of
special types that include modulation
and/or filtering. Then there are two types
of Shimmer reverb, plus Bloom, Magnetic,
Etherial, Extreme and Swell.
Shimmer typically pitch-shifts the reverb
tail up by an octave, but here you get two
shifted voices that can each go up or down
by up to an octave, in quarter semitone
steps. The gated and pseudo-reverse
reverb effects are also well-established,
though some of the other options may be
less familiar. Bloom, for example, allows
the reverb tail to swell after the initial A software version of X-Space, which runs in IK’s AmpliTube software, is included in the price.
note, while Magnetic incorporates some
very ‘floaty’ modulation. Swell is similar guitar — but if all you need a reverb pedal mystery and texture to whatever sound
to Bloom, though appears to act more for is convincing springs and plates, this one you feed in.
like an extreme ducking reverb, as the is probably on the wrong side of overkill! Although we tend to think of pedals
reverb tail only builds up when the original If, instead, you’re one of those people who as being the province of guitar players,
sound decays. Etherial is a modulated likes to get much more creative with your this one can sound wonderful on just
plate reverb, and Extreme is another plate effects, it really comes into its own. It offers about any sound source, from piano to
reverb but this time with a pulsing phaser some lovely variations that will attract not vocals. Then there’s the X-Mode sustaining
effect modulating the tail. The quality of only those who are into creating ambient effect, which is a welcome addition for live
the effects is top-notch and while they may music, but also EDM and soundtrack performance. Getting a virtual version to
sound much more ‘wide screen’ in stereo, composers. The shimmer reverbs, which run inside AmpliTube is just more icing on
they all sound great in mono too, so work I use a lot, are as good as any I’ve heard an already tasty cake.
fine if patching the pedal into a guitar amp. elsewhere, while some of the modulated
The X-Space can give you a good spring reverb options range from huge-sounding £ £299 including VAT.
or plate reverb emulation to use with your and dramatic to super-relaxing, adding W www.ikmultimedia.com

104 February 2022 / w w w. s o u n d o n s o u n d . c o m

The final chapter in an illustrious saga:
Metropolis Ark 5 provides the modern
composer with a myriad of elements for big
melodies, dramatic accents, tensions and
swells, and more. And the entire series is now
available as an exclusive bundle: Harness the
raw power of the super-charged orchestra
with the Metropolis Ark series.

www.orchestraltools.com
ON TE S T
Roland SP-404 MkII
Sampler
Roland’s popular sampler
gets a major update.
SIMON SHERBOURNE
T
he Boss/Roland SP series of
pad-based samplers has been
around for over 20 years, maintaining
a loyal fanbase even among those who’ve
dallied with MPC, Maschine or Push. The
SPs carved a niche for themselves as live
performance tools and DJ sidecars, with
their simplicity and transition-friendly effects.
The 303 and 404 have also been used
as compositional tools by many hip-hop
beatmakers like the great J Dilla.
More recently the SP-404 has been
popular in the mushrooming lo-fi beats
scene, typified by mellow, melancholy
chopped jazz samples and loose hip-hop
beats, influenced by SP users like Dilla and
Madvillain. Much of this music is home-grown
and shared on YouTube, and the 404 is
affordable, portable, and self contained.
Perhaps this is why Roland thought this was
the perfect time to revitalise the 404 with
a significantly enhanced MkII model.
Levelling Up
The basic form factor and workflow of the
SPs has stayed remarkably consistent over
the years, so veteran users should have no
trouble finding their way around this one. But
several modern conveniences have been
added, along with a host of feature additions
and improvements. The most visible
Roland SP-404 MkII
£439
PROS
• Chopping.
• Pattern and Chain resampling.
• Audio over USB.
• ‘Snapback’ retrospective sampling
buffer.
• Graphical display improves usability.
CONS
• Still not the most straightforward
thing to use.
• On-board mic gone.
SUMMARY
The SP-404 has become a versatile
secret weapon for beat makers, live
MCs and laptop performers.
hardware change is the move from 12 to 16
pads. This brings the 404 into line with most
comparable devices, and provides more
sample/pattern storage slots.
Also immediately obvious is the new
display. The previous models had a chunky
but basic three-character screen. The MkII
gets a proper graphical interface, which
makes the machine much more accessible,
with niceties like waveform editing,
multi-effects grid display, and a file browser.
Missing from the front panel is the mic
grille seen on all earlier 404s, as the built-in
mic is gone. There’s still a dynamic mic input,
which is now switchable to guitar DI mode.
Removing the mic seems counter to the
embracing of DIY lo-fi production, but this is
perhaps made up for by the fact that you can
sample directly from class-compliant USB
devices like a tablet or laptop (but sadly not
my Lightning-connected iPhone).
The USB in question is of the
bang-up-to-date type-C variety, and provides
both connectivity and power. Yes, the SP-404
does still have a battery bay, but I’d consider
this primarily a live backup, as six AAs will
only get you around three hours of use. USB
is a much better option. The only issue I had
was that it requires a direct USB-C to USB-C
cable, and won’t work with a regular USB-A
charger outlet. A good old fashioned DC
power adaptor is also included.
While we’re talking connectivity, the MkII
now has proper grown-up audio connectors
for its line ins and outs, instead of phono.
404 On The Floor
The 404 has been used by some adventurous
DJs for live sample triggering and performance.
The MkII has an actual DJ mode, which allows
you to play back and sync longer tracks, with
the unit split into two decks. Samples are loaded
(from the current project) into the two playback
channels, and controlled from eight pads on
either side of the grid. You can sync the bpms
in either direction, change the tempo of either
channel, or nudge the speed to make fine sync
adjustments in classic turntable style. The three
control knobs give you independent level of the
two channels (rather than a single crossfader),
and a cue which either channel can feed.
I was somewhat sceptical about how many
people are going to find this mode useful, but
I can see some scenarios for it. It would be a fairly
safe way to manage a live set, especially if you’re
a solo MC for example (the mic input can be used
You may or may not be pleased to learn that
MIDI is on mini-jack instead of DIN. A clear
upgrade, though, is the inclusion of a MIDI
out — the MkII can now sequence external
devices where the earlier models could not.
More Of Everything
Before we get too hung up on what’s
new, let’s recap what the SP-404 does.
First and foremost, it’s a sampler aimed at
performance and sample-based beatmaking.
In that respect it has more in common with
instruments like the Elektron Digitakt and
Novation Circuit Rhythm than the MPC or
Maschine+. Its other specialism is effects. It’s
set up for fast access and control of multiple
effects for live mangling of the internal
samples or live inputs. The MkII now has four
effects busses; more on that in a bit.
An SP project holds one sample per pad,
across 10 banks, so the extra pads increase
the number of samples per project to 160
from 120. But that’s not all, because the MkII
can now hold 10 projects in memory instead
of one, and it only takes a couple of seconds
to load a new project. Projects and samples
can also be backed up, imported and
swapped out to and from an SD card, plus
there’s an excellent software manager.
You can record patterns on the SP-404,
which are again stored one per pad across
your banks. A single pattern at a time can be
triggered for looped playback, over which
you can continue to trigger samples live.
Patterns can be chained to create a song
in DJ mode). Sleaford Mods could certainly run
a whole show like this! A nice bonus is that your
playback runs through the effects so you can spice
things up a bit. From what I could tell, though,
both channels route to Bus 1. It might be useful to
be able to switch this routing on the fly or at least
have some kill filters on the two channels.
A really useful option for live use is the
ability to hook the 404 up to your laptop and get
audio I/O via USB. The 404 presents as a 2-in,
4-out interface, which I enabled in Ableton Live.
I then discovered that tracks routed to outputs
3/4 are fed cleanly to the 404’s stereo output,
while channels 1 and 2 appear at the Ext Input
stage, meaning they can be muted/enabled
on the front panel, and go through the effects!
This is all combined with the audio that’s being
generated internally, making for some versatile
configuration possibilities.
w w w. s o u n d o n s o u n d . c o m / February 2022 107
 ON TE S T
ROL AND SP-404 MKII

structure. Pattern sequencing is still fairly

basic when compared to many samplers and Drag The Drop
drum machines, but has got considerably
more useful on the MkII; again we’ll come The SP-404 MkII is self sufficient, with the ability parameters. As well as preparing a palette
back to that. to directly browse and import samples from SD of sounds to go, this would be great for
card, and back up projects to the same, but it still programming a live show. But perhaps the most
Sampling benefits from a really useful software companion. useful thing is being able to import samples
This can manage the device when it’s tethered back in from the unit. This opens up a fast
The SP-404 is more focused on sampling by USB, controlling it directly and facilitating file hybrid workflow where you can capture and
than most comparable devices I can think transfer and backup. Many of the settings can be resample your ideas to pads on the 404, then
of. It’s quick to sample source sounds, remotely configured from here. pull them in for arrangement and production on
and the workflow encourages resampling: You can load up the banks by dragging and the desktop. It would be fantastic if the editor
dropping samples, and you can even edit the could become a plug-in, with direct drop of
taking chunks that you’re working on and
samples and adjust most of their playback samples into your DAW.
committing them to another pad. You simply
arm the record mode, choose a destination
pad then start sampling, either manually or
triggered from an input threshold.
A dedicated button engages input
monitoring, and you can add an effect to
the input independent of the main effects
busses. This is particularly handy for guitars,
which benefit from several amp sim effects.
Recording is with the input effect, and when
sampling you can choose whether to just
sample the inputs, or to re-render a mix of
the input signal and current playback.
Once you’ve sampled something you
can edit and process using the new nifty
waveform display. Unfortunately there’s no
way to preset the length of sampling or sync
to the tempo, it’s always a manual process
except when bouncing a pattern. This would
really speed things up, saving the need to
always top and tail samples or re-renders.
Sampling is a dedicated start/stop
process rather than something you do on
the fly. You couldn’t, for example, use the
404 like a looper. However, there is a new
feature which I found myself using all the
time: Snapback. The 404 has an always-on
record buffer that constantly samples the
main output. At any time you can choose to
grab the last 25 seconds of audio generated
by the device and drop it as a sample on to
a pad.
The other major new feature is chopping.
A dedicated process for slicing a sample up
The SP-404 MkII’s desktop app provides remote control and file management via USB.
is present on many workstations, and will
be a huge boon for most SP users. Instead
of having to duplicate a sample to multiple to selecting and launching patterns. Pattern patterns by assembling them on a 16-part
pads and cut each down individually, you length must be pre-set, along with record grid. Chains can loop, and a project can
can now split up a sample that’s in one pad quantisation strength. It’s great that you store 16 different chains, which you can cue
using manual marker placement, transient can record free of quantisation, but it would up and launch to manage a longer song
detection, or an even division by number, be also be nice if you could tighten things structure or live set. If you like, a chain or
then drop these zones onto a set of other up after the fact. Mistakes can be changed pattern can be resampled cleanly to audio
pads of your choosing. by real-time erasing of pads, and you can onto a pad. This is something that was tricky
overdub on to existing patterns. The Rec to do on previous versions and opens up
Patterns, Chains & Songs button toggles between Rehearse and new workflow possibilities.
Pattern recording on the SP-404 is always Record modes. A notable limitation is that In fact the SP-404 MkII presents several
via real-time capture, there’s no step although you can play samples chromatically, ways to juggle your content into a finished
sequencing. A dedicated pattern mode you can’t record a melody as a pattern. track or set. The traditional way was to live
switches the pads from triggering samples In Pattern Chain mode you can sequence resample sections to pads, then trigger

108 February 2022 / w w w. s o u n d o n s o u n d . c o m

those pads to create your finished song, either out to an
external recorder or resampled internally once again. You
can also launch a sequence of patterns to a similar end,
or combine patterns with live sample triggering. You could
create a Chain that utilises bounced song sections, triggered
by simple one-note patterns. Whatever method you choose,
it encourages a live mixdown style which can incorporate the
excellent performance effects...

Effects
Effects are literally front and centre on the SP-404,
with six buttons around the screen engaging multiple
performance-friendly sound manglers. These can be
customised, but the default set is a filter+drive, resonator,
delay, isolator, looper/stutter, and a multi-effects grid with
quick access to another 16 effects. There are four effects
stages or busses, with some (limited) control over which
sounds are affected. Bus 3 and 4 are always in series across
the whole mix. Bus 1 and 2 are fed by the sample pads in one
of two configurations: either two separate busses, or Bus 1
routing to Bus 2. Each sample pad routes to either Bus 1 or 2.
The Bus 1 and 2 effects are controlled from the those main
panel buttons, which you can set as toggles or momentary
switches. Engaging an effect brings up parameter controls
mapped to the three Ctrl knobs. These so-called ‘Direct FX’
assigned to each of the main buttons can be changed in the
Utility screen. The Bus 3 and 4 master effects have a different,
and slightly awkward assignment method. They are always
recalled as a pair, stored as a ‘favourite’ snapshot. This would
make some sense if it was for real-time recall, but selection is
via a menu in the utilities.
One design choice on the SP-404 that can be limiting is
that the effects configuration is a persistent property that’s not
saved or recalled with projects. Any changes you make will
affect all banks and projects. You have to think of the effects
section almost like a separate hardware module that you’re
running your tracks through.

Final Thoughts
Although the MkII SP-404 has adopted some conventions
like the 4x4 pad grid and beat-slicing tools, it’s really not
trying to be like everyone else. It follows its own path that
runs somewhere between a workstation like the MPC and
a performance sampler or drum machine like a Digitakt.
And then it has another potential role as a live backing track
machine and effects unit. In the studio, its strength, and
sometimes limitation, is how it steers you toward a workflow
where you move quickly from idea capture to performance
and commitment via resampling. It actually reminds me in
HOME STUDIOS
AND SOUNDPROOFING
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 C OM PE TI T ION

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 In addition to their outstanding
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a mobile device. With 20mm of excursion, the bespoke and rock-solid imaging. Both the bass
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Chosen by the readers of Sound On Sound
This year’s SOS Awards once again place the emphasis on
innovation and contemporary design, with the nominations being
limited to products introduced in the 12 months prior to the
voting period. These results celebrate the ‘best in class’ products
launched or reviewed in the period from September 2020 through
to the end of August 2021.

You’ll see all the winners you selected here as well as the very
worthy close contenders in each category who have been awarded
our Highly Commended accolade.

112 February 2022 / w w w. s o u n d o n s o u n d . c o m

WINNER WINNER
BEST AUDIO INTERFACE BEST PLUG-IN
AUDIENT ID14 MKII FABFILTER TIMELESS 3
www.audient.com www.fabfilter.com

WINNER WINNER
BEST DAW BEST MUSIC SOFTWARE
APPLE LOGIC PRO X STEINBERG WAVELAB PRO 11
www.apple.com www.steinberg.de

WINNER
BEST EFFECTS & PROCESSING HARDWARE
RUPERT NEVE DESIGNS 5254
www.rupertneve.com
WINNER
BEST SOFTWARE INSTRUMENT
EASTWEST HOLLYWOOD
ORCHESTRA OPUS EDITION
www.soundsonline.com

w w w. s o u n d o n s o u n d . c o m / February 2022 113

 WINNER WINNER
BEST GUITAR TECHNOLOGY PRODUCT BEST DRUM MACHINE/SAMPLER/SEQUENCER
NEURAL DSP QUAD CORTEX NATIVE INSTRUMENTS MASCHINE+
www.neuraldsp.com www.native-instruments.com

WINNER WINNER
BEST KEYBOARD/SYNTH BEST MICROPHONE
ARTURIA POLYBRUTE WARM AUDIO WA-67
www.arturia.com www.warmaudio.com

WINNER WINNER
BEST PERFORMANCE CONTROLLER BEST MIXER/MIXING CONTROLLER
ARTURIA KEYSTEP 37 SSL UF8
www.arturia.com www.solidstatelogic.com

114 February 2022 / w w w. s o u n d o n s o u n d . c o m

WINNER
BEST MONITOR
KALI AUDIO IN-5
www.kaliaudio.com

WINNER
BEST MICROPHONE PREAMP
RUPERT NEVE DESIGNS
RMP-D8
www.rupertneve.com

WINNER
BEST HARDWARE RECORDER
ZOOM F6
www.zoom.co.jp
 WINNER
BEST HEADPHONES
SENNHEISER HD 560S
www.sennheiser.com

WINNER
BEST LIVE SOUND PRODUCT
JBL PRX ONE
www.jblpro.com

116 February 2022 / w w w. s o u n d o n s o u n d . c o m

 HIGHLY
COMMENDED
SOS AWARDS 2022

AUDIO INTERFACE DAW

UNIVERSAL AUDIO APOLLO SOLO STEINBERG CUBASE PRO 11
www.uaudio.com www.steinberg.de

EFFECTS & PROCESSING HARDWARE MUSIC SOFTWARE PLUG-IN

WARM AUDIO BUS-COMP STEINBERG SPECTRALAYERS UNIVERSAL AUDIO API VISION
www.warmaudio.com PRO 8 CHANNEL STRIP COLLECTION
www.steinberg.de www.uaudio.com

SOFTWARE INSTRUMENT GUITAR TECHNOLOGY PRODUCT KEYBOARD/SYNTH

ARTURIA PIGMENTS 3 UNIVERSAL AUDIO UAFX PEDALS SEQUENTIAL PROPHET-5 & 10
www.arturia.com www.uaudio.com www.sequential.com

PERFORMANCE CONTROLLER DRUM MACHINE/SAMPLER/SEQUENCER MICROPHONE

AKAI MPK MINI MKIII ROLAND TR-6S SHURE MV7
www.akaipro.com www.roland.com www.shure.com

LIVE SOUND PRODUCT MONITOR MICROPHONE PREAMP

SENNHEISER MD 435 GENELEC 8361A RME 12MIC
www.sennheiser.com www.genelec.com www.rme-audio.de

HARDWARE RECORDER MIXER/MIXING CONTROLLER HEADPHONES

IZOTOPE SPIRE STUDIO 2ND RUPERT NEVE DESIGNS FOCAL CLEAR MG
GENERATION 5057 ORBIT PROFESSIONAL
www.izotope.com www.rupertneve.com www.focal.com

118 February 2022 / w w w. s o u n d o n s o u n d . c o m

MD 435

Bigger and brighter.

Top of the range.
The MD 435 delivers the sound of Sennheiser’s famous MD 9235 capsule in a
dynamic wired microphone. Stage-proof, the cardioid MD 435 large-diaphragm
mic brings accentuated presence, sparkle and beautiful detail to every voice.
Assert yourself effortlessly in the mix with natural, transparent sound and a
very pleasant proximity effect.

www.sennheiser.com/md-435
 ON TE S T

Qu-Bit Cascade
Eurorack Module

Q
u-Bit have been on a progressive Eurorack
journey for the last five years or so. Hark back
to the much lauded Chord oscillator module, in
its MkI form, and the sense of scale was much larger,
providing generous spacing across the panel, complete
with sizeable pots. One might wonder if their many
fans pleaded with them to shrink the form-factor, to
a size which was more forgiving for Eurorack users
with limited rack capacity. Let’s face it, most of us are
working in a skiff-based world, and like prime London
real-estate, space is at a premium. Their most recent
modules have exhibited size-shrinkage, while the Qu-Bit Cascade
operational side of their latter modules appears to have 10HP
increased exponentially, or at least that’s how it feels. +12V 131mA
You could easily apply this to Cascade, which is to CV control) which -12V 40mA accompanying LED’s
their latest in a line of more bijou modules concealing make light work of colour, which will
a mighty amount of functionality. Described by Qu-Bit prescribing your offer all the clues you
as “a ratcheting envelope generator, VCA and sound preferred ratchet. It’s blissfully need regarding selected function.
source”, it packs plenty of functional content into easy to use, while placing The first two operational modes
a 10HP panel. the Repeats and Curves pots on the agenda are an Envelope
Despite its form, Cascade does not feel terribly under random CV control yields Follower and Compressor,
cramped. Qu-Bit module panels are always incredibly incredibly immediate chaos! equipped with a virtual side-chain.
clear, with three outputs blocked together at the bottom It possibly won’t come as In Envelope Follower mode,
of the module. Elsewhere, there are six pots, with CV a huge shock to learn that the the Attack and Decay pots stay
control of each. The only exception to this is the offset envelope element of Cascade is relatively true to form, while the
pot, which relates to the VCA level, for the purposes of linked to the VCA, but it is possible curve and repeat pots move to
hearing your signal without CV attenuation. to access each element separately. control response sensitivity and
Let’s begin with the first element on parade, which This is incredibly useful, as it allows threshold detection. There is also
relates to ratcheting. Providing Cascade with a gate the addressing of the VCA for one a dedicated Invert button, which
trigger will immediately get you off of the blocks, signal, while the same ratcheting will flip the envelope in place.
while the associated Repeats pot will create multiple envelope output can be applied to Thankfully, these controls are
ratchets, evenly distributed between each gate pulse. another element in the modular. largely mirrored in Compressor
The note value of the ratchet extends from x1 to x16, Using Cascade in Ratchet mode mode, which keeps things relatively
taking in every duplet or triplet note value along the is a simplistic business, and one tidy on the operational front.
way. Part of this ratchet experience relies on the that requires little explanation Moving along the modes,
settings provided by the envelope element; as an for the more seasoned Eurorack Cascade offers four different and
aside, Cascade can also operate as an AD, ASR or user. Where Cascade goes deeper simultaneous sound sources,
ADSR envelope, but within the realms of ratcheting, is via the Mode button, where which include an 808-style kick
the former applies, but with the added capacity to further functionality is available. and hi-hat. If you think about it, this
shape the envelope curve via a dedicated pot, from This is the point where you might makes perfect sense, as ratcheting
logarithmic, through to linear and exponential. There have to keep the manual to hand. is the lifeblood of the trap hi-hat,
are also dedicated Attack and Decay pots (also open The Mode button toggles the and nothing says trap more than

120 February 2022 / w w w. s o u n d o n s o u n d . c o m

sounds modelled after Roland’s much loved rhythm
composer classic. Alongside these ubiquitous timbral
elements, Cascade can also deliver a very pleasing
burst of white noise to the outside world, with the
addition of a sine, presumably sharing association with
the 808 kick.
What I find particularly appealing about Cascade,
although I could easily apply this to all Qu-Bit modules,
is its comparative ease of use. I’m a bit of a ratcheting
fan, and was delighted that I could hook Cascade into
my signal chain and get ratcheting without having
to reach for the manual. Having CV control across
the board is expected these days, but the provision
of a VCA which can be employed or bypassed at
will creates flexibility for exploring the ratchet and
envelope capacity elsewhere in your system.
As for the shrinkage, this will be welcome to any
skiff-bound user who will be fighting for row space, but
despite it’s 10HP footprint, it does not feel cramped
in use. Offering a VCA, envelope and ratchet, without
compromise in such a small panel space, is a perfect
demonstration of neat and clever design. Add in the
additional sound sources, compression and envelope
follower, and you’ve got a useful multi-function module,
which does not feel bloated or cumbersome in
operation. Dave Gale
£ £220
W www.qubitelectronix.com
Clatters Machines Garden Listener
Eurorack Module
source, what does it actually do?
The short answer is something
quite random: a gate here, a note
change there.
Sometimes
in clusters, Clatters Machines
sometimes Garden Listener
sparse. Upon 8HP
a cursory look, +12V 24mA
-12V 20mA
if one was
to be cynical
the word Krell would come to
mind (the ubiquitous and easy
self-generating patches inspired
by the electronic soundtrack of
the 1956 movie Forbidden Planet).
Such a response, however, would
say more about my ability and
creativity as a synthesist than it
would about the module itself. Just
as Mylar Melodies illustrates in his
exploration of sample & hold, such
fundamental functions have limitless
applications, and applying those
functions well is the test of any
synthesist; something I found out as
soon as I began to push the Garden
Listener’s functionality a little bit.
For one thing, it was thrilling to hear
the probes responding differently
to different things, attaching them
to the leaves of a plant (succulents

T
hose familiar with the YouTube channel of
Eurorack expert Mylar Melodies may remember
a particular video in which he deftly expounds
why sample & hold circuits can be so much more than
what they’re used for a majority of the time. Sample
& hold represents a function that is interesting and
exceedingly useful in theory, but tends to boil down to
a relatively rudimentary application in practice, which is
essentially a random voltage generator. Upon unboxing
the Garden Listener, the debut module from nascent
Italian developers Clatters Machines, my first thought
was that despite piquing my interest, such a module
may well find itself consigned to the same category.
The Garden Listener is a remarkable module which
uses twin probes to pick up impulses from the leaves
of plants, skin or any other bioelectrical surface, and
convert them reliably into MIDI and CV messages.
Three voltage outputs emit triggers, gates and variable
voltages, while the module’s 3.5mm MIDI output allows
for integration with just about anything that accepts
MIDI. An onboard quantiser gives the option to convert
incoming voltages into any one of five scales, and it’s
all packaged in a nifty 8HP.
The probes themselves ship with detachable
adhesive pads for attaching to your favourite (or least
favourite) houseplant, and I must say it’s immensely
gratifying to stand back and see an actual living plant
patched into one’s system like a module. It’s a definite
statement piece for any studio. But patched to a sound
worked particularly well), pressing
them to my temples (which brought
out more intense clusters of notes
and triggers) and even pressing
them to my heart, at which point
the module began shooting gates,
triggers and notes all over the
place. Perhaps it’s my imagination,
but something about the Garden
Listener’s flourishes of triggers,
gates and CV felt, dare I say
it, organic.
So, to the challenge of
application. Patching the Garden
Listener’s trigger and CV outputs
to a sound source’s gate and
volts-per-octave inputs achieved
quick and easy results, but I soon
realised there’s a world of fun to
be had by, for instance, patching
the gates to trigger and reset
a quick sequence, patching the
CV to a VCA via an inverter for an
almost tape-like instability effect
or even just using it to generate
a drone and modulate it according
to one of the Garden Listener’s
quantised scales. Clatters made the
right decision to incorporate three
separate outputs instead of two,
which makes all the difference.
I also realised that by interfacing
biological impulses with synthesis,
Clatters have achieved something
that very few developers have
managed to do: create a new
fundamental function. Furthermore,
it’s not as though Clatters have
decided to err on the safe side
by releasing an oscillator or
a sequencer for their first module
— to go this off-kilter straight out of
the gate is mightily impressive and
highly creative.
We’ll be sure to keep a close
eye on Clatters Machines. If the
Garden Listener is anything to
go by, expect exciting things.
William Stokes
£ €275
W www.clattersmachines.com

w w w. s o u n d o n s o u n d . c o m / February 2022 121

 ON TE S T

Get More From Your VCAs

‘You can never have too many VCAs’, goes the
ubiquitous saying among modular communities.
The humble voltage controlled amplifier is
something of a microcosm of how to approach
modular synthesis, in that the more laterally you
approach it the more interesting your results
will be. A quick recap: in simple terms a VCA is
an amplifier that varies its gain depending on
a control voltage. Generally it has one signal
input, one output and one CV input in between
the two to control its behaviour.
In practice I find it more helpful to think of
a VCA as a kind of variable gate that lets voltage
through according to the value of an incoming
voltage, as opposed to amplifying it; a circuit
to ‘put in the way’ of almost any signal path in
a modular system. Patch an oscillator into your
VCA’s input and an LFO into its CV input, and voila,
you’ll hear a pulsing oscillator which gets louder
when the wave is high and quieter when the wave
is low. Speed up that LFO and the pulses will get
faster. Keep speeding it up to audio rate and you’ll
start to create new tones entirely. Now things get
more interesting. Since VCAs often come in pairs,
we could put VCA 2 in the path of our audio-rate
LFO controlling VCA 1; feeding a second LFO into
the CV input of VCA 2 to control the amount that
LFO 1 is controlling VCA 1. The result is a dynamic,
multi-dimensional sound made with just two
LFOs and a single oscillator. Wondrous! Far more
than just about volume, VCAs can constitute the
creative heart of a patch. Chaining more of them
into the mix will exponentially increase their scope,
leading us back to the aforementioned maxim
that you can never have too many. If in doubt, put
a VCA in the way! William Stokes

Modular Profile: Andrew Ostler

A
ndrew Ostler is the man behind one of the UK’s
most eminent Eurorack developers, Expert
Sleepers. As an artist he makes sweeping
experimental electronic music with healthy doses of
acoustic woodwind, both by himself and with guitarist
Michael Bearpark as Darkroom.

On Expert Sleepers’ new all-analogue modules

It was a palate cleanser, really! I started them about
a year ago. I was needing to start something new.
The Expert Sleepers thing has always been, more or
less, to do what I personally would enjoy using or find
useful. I’m one of these people who does believe that
analogue brings something you can’t get from digital,
whether that’s to do with human interface or how it
behaves when you abuse it slightly. I’m very pleased
with how it turned out.

On his go-to modules

My usage of modular has always been very basic, in
a way. It comes back to whatever VCOs and VCFs I like
the sound of, and some effects to mess things up! I’ve
got a Metasonix valve VCA which I love using. I do
come back to the Doepfer modules still. A Doepfer
VCO is just a VCO, and a Doepfer filter is just a filter, simply that there are a lot So it became a microcontroller with
and it’s not trying to be 20 other things at the same of different people, with A-D/D-A conversion. That became the
time. It’s just that one thing and it sounds good. The different wants and needs. Disting, and then I had the idea for jack
[Flight Of Harmony] Plague Bearer and The Sound I certainly wouldn’t have sockets that lit up. The Disting was almost
Of Shadows have been in my rack for a long time. gotten far at all if it hadn’t an excuse to make a module with light-up
I have some bucket brigade delays, some [Expert been for interacting with jack sockets, really. It was almost that
Sleepers] Distings, a Birdkids Bateleur... I’m keen on those people on forums. simple. I’d made a little thing that inputs
the Xaoc modules, actually — their Moskwa eight-step and outputs audio and CV, and given
sequencer I used for a long time. On one function vs many those constraints, what could that be?
The Disting — the whole It’s partly me enjoying the challenge,
On the culture of modular multi-function thing— was but it also comes back to the modular
I think it’s more multifarious than it might seem. You a series of accidents, I guess! community — listening to customers who
get the analogue purists, you get the people who buy I think I was trying to make are on the forums who might say, ‘This
one of my interfacing things and all the expanders so an analogue VCO, and the is great, but if it could just do this one
they’ve got 64 channels of control coming from their tricky thing to get right with thing it would be super great.’ So I’ll think,
DAW, you’ve got the people who want to use hardly any VCOs is the tracking. I was ‘OK, well that’s two lines of code.’ It costs
patch cables in a system which is basically digital... And thinking, ‘Could I possibly do me nothing to add that and makes the
then you get people who are just tinkering because this in software?’ because my module more valuable to everybody. So
they want to have a go. My view as a manufacturer is background is in software. why not? William Stokes

122 February 2022 / w w w. s o u n d o n s o u n d . c o m

 ON TE S T
HUGH ROBJOHNS
R
ecording and mixing often means
remaining seated for extended
periods, so being supported
in a good chair is essential, both for
short-term comfort and longer-term back
care. Ergonomic chair design is a complex
matter, but obvious key factors include
the way the, er, buttocks are supported,
(not a phrase that crops up in many of my
reviews!), how well the backrest supports
the lower spine, and whether the seat
base maintains comfortable and at safe
angles for the legs, pelvis and spine.
Many manufacturers offer ‘ergonomic’
chairs and one of the best known and
most popular in our particular industry is
undoubtedly Herman Miller’s Aeron chair.
The Stealth Pro
Executive model,
with short piston for
normal seating.
124 February 2022 / w w w. s o u n d o n s o u n d . c o m
ErgoLab Stealth
Studio Chair
Unlike many of its competitors, the ErgoLab Stealth series
is designed specifically for use in the studio.
It seems almost ubiquitous in studios and
production suites wherever I go — as
well as in my own studio! However, there
are valid alternatives. Based in Colorado,
USA, ErgoLab have been manufacturing
ergonomic seating for around 35 years,
and their patented Stealth Pro chair
design is a convincing alternative to
Herman Miller’s ever-popular models,
intended specifically for studio
applications and offering more for less!
Sit Happens
The Stealth Pro ships in a single
large (and heavy) box, measuring
roughly two feet on each side.
The chair requires simple
self-assembly without tools:
the pedestal base, wheels,
column piston, seat and backrest just
slot together, while the adjustable
arms are attached (if required) with
large hand-screws underneath the
seat base.
Two sets of wheels are provided:
a quintet of standard office-style castors
featuring dual 60mm hard plastic
wheels, and five chunky 100mm rubber
flightcase-style wheels — the latter
being better suited to more industrial
applications. I’m impressed that both
types are included as standard, allowing
the user to fit the most appropriate
type for their own floor surface and
application. The selected wheels are
pressed into the pedestal base sockets
and lock into place.
Also included in the box are two
column pistons, with the short unit
providing seat nominal heights of 495 to
635 mm, intended for sitting at standard
office-style desks and studio consoles.
The longer piston covers heights of 571
to 762 mm, which is ideal for working
at raised mixing consoles in live-sound
applications, or at work benches.
ErgoLab Stealth
From £759
PROS
• Customisable through two supplied
column piston sizes, two wheel sets,
and a foot ring.
• Two seat and two backrest
sizes available.
• Ergonomically tilting seat base
and pivoting backrest provide
lasting comfort.
• Mesh seat and back ensure good
support and ventilation.
• Design doesn’t impede playing
instruments while seated.
• Attractive pricing.
CONS
• It is heavier than typical alternatives.
SUMMARY
A very robust and versatile studio chair
which also goes higher than most and
allows guitar playing without getting
in the way!
 With the long piston fitted, an included
foot ring can also be slipping over the
column piston before the seat is attached,
which is very useful for elevated seat
positions. A simple rotating clutch
mechanism allows the ring height to be
adjusted as required.
By far the heaviest element of the
chair is the seat base itself, which has
a substantial steel frame and a flexible
plastic mesh for the seat and backrest to
provide good ventilation/breathability and
naturally contoured support.
The seat base is available in two
sizes, and there are also two styles of
backrest: a Standard (short) version, and
a high-backed Executive version (pictured
above). Both element sizes must be
specified at time of order. For this review
I was supplied with the Stealth Pro model,
which comes as standard with a large
seat base and Standard (short) backrest.
However, I was also subsequently
supplied with the alternative Executive
(high) backrest, which is apparently
intended for “executives who recline for
much of the day”!
ErgoLab suggest that users over
1.82m tall and/or weighing more than
90kg should opt for the larger seat, while
smaller and lighter folk are recommended
the smaller version, but as an experiment
I asked Mrs R (who is unquestionably
under both size thresholds) to try the
large seat and she reported it perfectly
comfortable and supportive.
Assembling the chair takes only a few
minutes and is obvious, although clear
instructions are provided, of course. The
chair’s controls are simple and familiar,
too. Height is adjusted with a lever on
the right-hand side in the standard way,
the tilting action of the chair’s seat base
can be locked with a second lever, and
a large hand screw at the side fixes the
height of the backrest. As a complete
chair, the Stealth Pro with a large seat and
ErgoLab Pricing
• ErgoLab Stealth (standard seat, standard
backrest) £795
• ErgoLab Stealth Pro (large seat, standard
backrest) £859
• ErgoLab Stealth Executive (standard seat,
high backrest) £795
• ErgoLab Stealth Pro Executive (large seat,
high backrest): £1039
All versions ship with both short and long
pistons, regular and heavy-duty castors, and
a foot ring. Prices include VAT.
standard backrest weighs 30kg, while the
Executive model with the high backrest is
2kg heavier.
By default, the backrest is
spring-loaded to ‘float’, supporting and
‘massaging’ the user’s back as they lean
forward and back. However, another lever
on the left-hand side locks the backrest
in position, if preferred. A button under
each armrest allows its height to be
altered, and the padded rests can be
rotated in or out by 30 degrees from the
default straight-ahead position. The width
between arm rests can be adjusted with
the securing hand screws underneath the
chair base.
Take A Seat
ErgoLab say that the Stealth Pro is made
“with superior components which will last
much longer than typical office chairs”,
and I don’t doubt the robustness of this
design — it is considerably sturdier than
most conventional office chairs, and
heavier than the Aeron it stood in for
in my studio. Longevity is obviously not
something I can assess over the couple
of weeks that the chair was in my studio, The non-Executive Stealth Pro version ships
but it looks and feels capable of enduring with a lower backrest. Here, the longer piston and
a long life. foot ring have been deployed.
The manufacturer’s ergonomic claims
for the chair mostly revolve around the while also making the chair less
way the seat base (and backrest) move dominating in the room. I also found that
to minimise pressure points, which can the armrests can be lowered sufficiently
lead to discomfort and fatigue. Even more to not get in the way when playing guitars
importantly, the tilting seat base maintains and keyboards, too, and then brought
good lumbar support as leaning forward back up at the touch of a button when
pivots the entire body instead of forcing working at the desk.
it to bend at the lower back, apparently The sprung backrest means that you
reducing pressure on the lumbar spine have to kind of push your way back
by as much as eight times compared to into the seat, but once there — and
a fixed chair! The mesh seat and backrest with the backrest set at exactly the
stretch to provide natural and individual right height for the user’s back — the
support, which is claimed to be better Stealth chair is extremely comfortable
ergonomically than simple fixed foam and throughout long working days. If you’re
cloth upholsteries. in the market for a rugged, hard-wearing
After some experimentation I settled and versatile studio chair, the ErgoLab
on the standard office wheels (I have Stealth Pro/Executive has much in its
a laminated wooden floor) with the long favour. The extensive list of included
piston. Later I also replaced the standard accessories and the fact that the chair
backrest with the higher version. The goes considerably higher than most
standard backrest was very comfortable alternatives are big pluses, and the tilting
when working at the computer (and when seat base and pivoting backrest ensure
playing my bass guitar or keyboards), outstanding comfort over long periods.
but I found the high backrest better for Highly recommended.  
occasional moments spent in ‘executive
recline’. Unusually, the Stealth Pro’s
£ See pricing box.
backrests are much narrower (especially
T Universal Distribution +44 (0)845 555 1123
at the top of the high version) than most E info@universal-distribution.co.uk
alternatives, and that means they don’t W www.universal-distribution.co.uk/stealth
get in the way when playing instruments, W www.stealthchair.com
w w w. s o u n d o n s o u n d . c o m / February 2022 125
 ON TE S T

Earthworks
ETHOS
Capacitor Vocal Mic
Earthworks continue their mission to bring
small-diaphragm capacitor mics to new markets.

SAM INGLIS

O
ver the last 10 years or so,
capacitor mics have gained
widespread acceptance for
live use. Their extended high-frequency
response and clean off-axis pickup
has made models like the Neumann
KMS series, DPA 2028, AKG C636 and
Shure KSM9 familiar sights on stages
worldwide. Typically, these stage mics
use small-diaphragm capsules derived
from familiar studio mic designs — yet
it’s still rare to see small-diaphragm mics
used for vocals in other contexts. Why?
Anyone who’s tried it will know that
small-diaphragm mics are perfectly
capable of capturing the human voice.
With some voices, in fact, they even have
advantages over their large-diaphragm
brethren. Our tendency to put up a U87
or even an SM7B ahead of a KM84
when a singer walks into the live room is
probably driven as much by habit as by
genuine sonic preference — or perhaps love, but a typical pencil mic just doesn’t
by the unconscious expectation that look the part for vocal recording.
a large and imposing mic will deliver
a large and imposing vocal sound.
Small Is Beautiful
Capacitor stage mics have been accepted Earthworks have long championed the
in part because they outwardly resemble small-diaphragm capacitor capsule for
the moving-coil models we know and all applications. They see its accurate

126 February 2022 / w w w. s o u n d o n s o u n d . c o m

transient response, phase linearity, pure
polar pattern and smooth off-axis sound
as being universal benefits that confer
advantages in any recording or live
context. And, having refined their core
technology, they are now introducing
it into new form factors in the hope of
overcoming our resistance to using
small-diaphragm mics for vocals.
This process has already yielded the
stellar SV33 studio vocal mic, as well as
the SR40V and SR314 stage mics and the
ICON, a speech-oriented mic available
in XLR and USB variants. The latest
product of this development pipeline is
the ETHOS. Like the ICON, it’s described
by Earthworks as a “broadcast mic”, and
its industrial design clearly references
one of the titans of that world, the Shure
SM7B. As on that mic, a foam plosive filter
forms part of an overall cylindrical shape,
and can be removed to leave the capsule
housing looking slightly naked.
All the Earthworks mics I’ve tried have
been beautifully made, and the ETHOS
is no exception. The shell is in similar
proportions to that of the SM7B, but is
a little smaller, and lacks that mic’s yoke.
It’s manufactured from gleaming stainless
steel (though if the gleaming bothers
you, a matte black version is available
as a cost option) and the rigid grille that
surrounds the sides of the capsule is
a work of art. The front of the capsule
is protected not only by that black foam
filter, but by a circle of very fine metal
mesh that is completely untensioned
and moves freely back and forth. The
rear end of the mic, meanwhile, houses
the XLR connector; there are no pads or
filters. The smart cardboard shipping box
houses the mic snugly in a foam cutaway,
but there’s no carry case.
Before moving on, I also want to single
out for special mention the supplied
Earthworks ETHOS
£699
PROS
• Stunning appearance and
high-quality construction.
• Clean, balanced sound, with the
ability to take very high SPLs.
• Brilliant standmount.
CONS
• A carry case might have been useful.
SUMMARY
Equally suited to podcasting, radio and
recording studio use, the ETHOS is
a fine general-purpose vocal mic.
A custom foam
windshield is supplied as part
of the ETHOS package.
standmount. A third-party component
called the M2-R, made by Triad-Orbit, this
emerges halfway along the mic body,
and uses a ball joint to combine a wide
range of movement in all three axes with
instant, secure clamping. I’ve only ever
encountered a similar mount on one other
mic, an obscure Pearl model from the
’70s, and it’s a genius idea that deserves
to be much more widely used.
Flat Earthworks
Earthworks make several different
sizes of small-diaphragm electret
capsule, and the ETHOS uses their
14mm design, configured here to deliver
a supercardioid polar pattern. It can
operate on phantom power from 24 to
48 Volts and, like other Earthworks mics,
draws the maximum permitted 10mA from
the supply. Sensitivity is a comfortable
20mV/Pa, self-noise is specified at 16dBA,
and the ETHOS is said to be able to
cope with sound pressure levels up to
145dB, though the level of distortion this
represents isn’t stated. The balanced,
transformerless output has a nominal
impedance of 65Ω.
One person’s “accurate transient
capture” is another person’s “extended
high-frequency response”, so it’s
no surprise that Earthworks claim
a frequency response of 20Hz to 30kHz
for the ETHOS. As this mic is intended
for close-up use, it’s voiced to give a flat
response with the foam filter in place,
and within the distance where proximity
effect is apparent, not at 1m. The graph
shows less than 1dB deviation from flat
below 3kHz, with a gentle dip centred at
around 4.5kHz and slight
double peaks at 7 and 10
kHz. Even at 30kHz, the
response is only 2 or 3
dB down.
The ETHOS might
have the ability to
capture frequencies
other mics can’t, then,
but it isn’t an obviously
bright-sounding or shrill
mic. In fact, the adjective
that immediately springs
to mind is ‘balanced’.
Without the presence
peak that many
large-diaphragm mics
have, it comes across
as quite restrained by
comparison; and in fact,
if you were to move from
something like an SM7B to the ETHOS,
I think you’d notice the difference as
one of refinement and openness at the
top end, rather than a night and day
contrast in character. Removing the
foam filter gives the sound some extra
air, and unless you work the mic very
close, popping doesn’t seem to be a big
problem, so I quite often used it that way.
And although the standmount doesn’t
offer any shock protection, whatever
internal shockmounting Earthworks have
devised is highly effective.
I liked the ETHOS a lot on both
speech and singing, and was particularly
impressed by its ability to capture the
female voice, which can often present
a challenge. Its pattern is tight enough
to offer a very useful degree of off-axis
rejection, yet not so tight that vocalists
need to take special care to address it
from precisely the right angle. I also tried
the ETHOS on numerous instrumental
sources, and found it a very useful
studio workhorse. It will happily take
the level from snare drums and guitar
amps, and it’s equally content on things
like acoustic guitar. In fact, you could
use it pretty much anywhere you’d
normally use a directional pencil mic,
with the caveat that its voicing for close
use means you’ll get less bottom end.
With its striking appearance, superb
construction and versatile, smooth
sound, the ETHOS is much more than just
a “broadcast mic”!
£ £699 including VAT
W www.studiocare.com
W www.earthworksaudio.com
w w w. s o u n d o n s o u n d . c o m / February 2022 127
 ON TE S T
GORDON REID
S
ince the release of the Polysix,
MS20 and Wavestation soft synths
in the first incarnation of the Korg
Legacy Collection, the incorporation of
the Polysix, MS20 and CX3 in the OASYS,
and subsequent ‘soft’ recreations of the
Mono/Poly, the M1 and the Triton, Korg
have been steadily mining their rich
history of analogue synthesizers and
digital sound engines. Now the company
have turned their attention to three more
of their keyboards — the MiniKorg 700S
analogue monosynth, the Prophecy
digital monosynth and the Triton Extreme
workstation — releasing soft synths
based upon each of these within the Korg
Collection 3.
Some people might find these
strangely chosen. After all, the Korg 770
and 800DV are more revered than the
700S, the Z1 would have covered more
ground than the Prophecy, and the Trinity
probably has more of a cult following
than the Triton Extreme. But others will
find them well chosen. Wherever you
stand on this debate, there’s no doubting
128 February 2022 / w w w. s o u n d o n s o u n d . c o m
Korg Collection 3
Software Synthesizers
Korg add more instruments to their growing stable of
software classics.
that the response to their announcement
was almost overwhelmingly positive so
I downloaded and installed all three new
synths. It was time to indulge in a bit of
gratuitous comparisonising.
The MiniKorg 700S
The 700S soft synth comprises three
layers: a recreation of the 700S itself with
additional performance controls and an
arpeggiator replacing the original Repeat
function; an eight-slot modulation matrix
offering 80 destinations fed from 32
sources that can draw upon three new
LFOs and three additional ADSR contour
generators; and an effects board with six
virtual stompboxes.
To begin testing it, I selected the soft
synth’s monophonic Init patch, deleted all
of the V.Patch sources and destinations,
and then switched off all of the effects
units. This, in theory, turned it into an
unenhanced 700S playing a sawtooth
wave. I then set up my MkII 700S in the
same way. The sounds that I obtained
from the hardware and soft synth were
very similar and I could have used either
with the same musical results. So far, so
good. I then stepped through the other
waveforms and differences emerged. The
square and sawtooth waves were fine,
but the chorus waves were different, with
the modulations in both I and II being
deeper and faster on the soft synth.
These differences were so significant
that I wanted to look into them further,
 The same patch as
shown in the first screen,
but now represented in the
more accessible 2D format.
so I liberated my
other MkII 700S
from storage (yeah,
I know... shoot me)
and set up the
three instruments
alongside one
another. The two
hardware synths
were almost identical
with one another
but the soft synth
stood apart.
I then moved
on to the Travelers
(the 700S’s HP and
LP filters). On the
original, the controls
for these ensure that
the cutoff frequency
of the LPF always
remains equal to or higher than that of
the HPF so that a conventional band-pass
response is obtained. On the soft synth,
they can cross over. I know that some
owners chopped bits off the Traveler
knobs to make this possible on the
original, so I can understand why Korg
have done this. Comparing the three
synths demonstrated that the underlying
characters of the soft synth’s filters are
similar to those of the originals, although
its low-pass filter closes further and
the preset amount of filter resonance
is greater. However, you can correct
these differences in the V.Patch page if
you wish.
The early Korgs have possibly the
strangest envelope controls ever seen,
yet they are capable of creating AS,
ADS and AR contours. Experimenting
with these revealed differences in the
soft synth’s contour shapes and times,
and I found that its ADS shape is only
accessible within a narrow band at the
top of the Perc/Singing range, which
makes it harder to set up than the
original. Furthermore, both of my original
synths can generate a pleasing transient
‘bump’ at the start of the note, whereas
the soft synth does not. Again, you can
overcome this using V.Patch parameters.
Next, I moved on to the Effects (second
oscillator) panel. This offers conventional
dual-oscillator sounds, three flavours of
ring modulation and two types of noise,
and in many respects it’s where the real
power of the 700S lies. Here, I found
further differences. For example, the
depth of Modulator 3 is quite different.
In addition, the pitched noise contains
a tonal element that sounds like oscillator
bleed, and this makes it impossible to
obtain some of the exquisite breathy
sounds of which the 700S is capable.
By this point is was clear that, if my
hardware synths are responding and
sounding as originally intended, the
emulation is not a perfect replica of the
700S. Given the accuracy with which
Korg have recreated the MS20, Polysix
and Mono/Poly, I find this surprising.
So let’s forget about emulating the
original and ask what the soft synth is
like as an instrument in its own right.
In truth, it’s huge fun, and it generates
high-quality sounds with little or no
aliasing. If you keep things simple, it
has a lovely, clear ’70s vibe but, if you
start to invoke polyphony, unison, lots
of modulation matrix routings (including
those that overcome limitations such as
the on/off status of the filter modulation
and resonance amounts) and plaster
the results with effects, it becomes an
unusual but powerful polysynth. After
a while, I was even obtaining sounds
reminiscent of the CS80 from it, and that’s
not a trivial accolade.
In the 1970s, one mastered the 700S
by understanding and overcoming its
w w w. s o u n d o n s o u n d . c o m / February 2022
Korg Collection 3
$399
PROS
• Korg have made the 700S, Prophecy
and Tritons accessible to a whole new
generation of music makers.
• The 700S can take you in
new directions whether used
monophonically or polyphonically.
• The Prophecy adds physical modelling
to the Collection and takes tentative
steps toward being a ‘soft’ Z1.
• The Tritons are superb workhorses
with a huge range of factory sounds to
get you started.
CONS
• If my Minikorg 700Ss are to be
trusted, the soft synth doesn’t appear
to be a wholly accurate emulation.
• The Prophecy models need further
work to make them authentic
recreations of the original.
• The MIDI implementations of all four
soft synths leave something to be
desired.
• There’s no MPE compatibility,
which would particularly benefit
the Prophecy.
SUMMARY
Adding the 700S adds breadth to the
range of vintage sounds available in
the Collection, while the Prophecy
adds physical modelling to the mix. In
addition to these, the Triton workhorses
complement the existing M1 and
Wavestation beautifully. There are some
issues to be addressed, but the more
I used these soft synths, the more I liked
them and the better they sounded.
129
 ON TE S T
KORG COLLECTION 3
limitations and finding the sounds hidden
beneath, and the same is true today.
Sure, Korg have removed many of those
limitations but I still found that, the more
I came to know it, the better the soft synth
sounded. If you fancy something that
steps outside of the norm defined by all
of the Minimoog, Prophet and Jupiter soft
synths out there, this could be a good
place to start.
The Prophecy
When I reviewed the Korg Prophecy
in 1995 I heaped fulsome praise upon
it. In retrospect, I may have been
a little too enthusiastic, but not by
much; it was and remains a wonderful
performance-oriented monosynth
that was adopted by some important
artists, although in my view it deserved
even greater acclaim than it received.
Consequently, I was very excited to test
the soft synth especially since, with its
polyphonic capabilities, it’s the closest
thing yet to a ‘soft’ Z1 (or, rather, a soft
monotimbral subset of the Z1).
130 February 2022 / w w w. s o u n d o n s o u n d . c o m
Since the soft synth is, by definition,
a re-hosted version of an existing digital
synthesizer, I didn’t bother setting it up
next to my Prophecy when I first tried
it. After all, while not wanting to be
dismissive of the huge amount of work
involved in porting them to modern
hardware, running the Prophecy models
on a Mac or PC should provide the same
results as running them on the original
platform. Right? Well... no. I didn’t need
the original in the same studio to realise
that there was something wrong.
I had started by playing my favourite
Prophecy brass sounds. These were
a revelation when I first heard them, with
an authentic sounding ‘rip’ — the division
of a pitch-bend into discrete modes —
that lifted them far above conventional
synthesised and sampled brass sounds.
However, the soft synth didn’t emulate
this, and rapid bends also caused digital
clicks as the model struggled to keep
pace with the controller. Disturbed by
this, I moved on to some of the reed
model sounds such as saxophones. When
The Prophecy soft synth is much clearer and
simpler to use than the original hardware.
played on the original Prophecy, certain
bent pitches contain just the breath noise,
as they should. On the soft synth, I again
obtained a smooth pitch-bend as one
would on any conventional synthesizer.
And yet further differences were revealed
when using sounds based upon the
other models.
Not wanting to blame Korg for
anything just yet, I took advantage
of the soft synth’s compatibility with
the original Prophecy’s SYX dumps to
transfer the complete setup from my
original Prophecy into it. For obvious
reasons, this would ensure that I wasn’t
hearing differences in programming.
Unfortunately, I wasn’t. The discrepancies
remained, so I could only conclude that
the models themselves were at fault.
Had I submitted this review a few
weeks ago, you would now be reading
a handful of (now deleted) paragraphs
about how disappointed I was but, before
I did so, Korg released the soft synth’s
 ON TE S T
KORG COLLECTION 3

first update. This corrected the pitch-bend
in the brass model and, although the
results I obtained were still not identical
with the original’s, they were a huge
improvement. But why is there any
difference? I suspect that it’s because
the Synth Kit code on which the original
Prophecy is based has been obsolete for
so long that it can’t be ported directly on
to modern computers. If that’s the case,
a significant rewrite would have been
necessary and small differences are both
inevitable and forgiveable. It remains to
be seen whether Korg will update the
discrepancies in the other models.
These issues notwithstanding, the
soft synth can often sound very much
like the original, so perhaps I’m again
becoming too engrossed in a slavish
comparison between the two. Maybe
I should just be asking whether it’s
a great musical instrument. In many
ways it is, and there are aspects of the
soft synth that are streets ahead of the
original. It took me a long time to get the
best from the Prophecy in 1995 because
its programming system — accessible
through the letterbox of a two-line
display — was tricky to navigate. Today,
everything is laid out clearly in three main
pages: the synth itself; the effects page,
which also provides access to various
voice settings; and the arpeggiator
page, which includes the performance
control editors. This makes the Prophecy
accessible in
a way that it has
never been before.
However, one thing
that will remain hard
to emulate is the
sheer playability
of the original,
with its log (which
includes a ribbon
controller) and dual
pitch-bend and
modulation wheels
in close proximity.
Perhaps this
would be a perfect
opportunity for
MPE to take over
performance duties,
but this hasn’t been
implemented. I hope that it eventually
is, because it would be a great way to
control the Prophecy’s various models.
But even today, if you add the bonuses
of polyphony and unison and start to
explore what physical modelling can
do when running in multiple polyphonic
instances on a powerful DAW, the results
can be remarkable.
Triton Classic & Triton Extreme
There were numerous models of Triton
workstation, all based around versions
of Korg’s HI sample+synthesis engine
plus a sampler, a 16-track sequencer,
dual polyphonic arpeggiators, and the
option to install a 6-voice MOSS (Z1
sound engine) board. The series started
with the original Triton (retrospectively
renamed the Triton Classic) and — not
including the cut-down ‘LE’ and ‘TR’
versions and various sideways trips
to the Karma and other spin-offs — it
evolved through the Triton Rack and
Triton Studio before ending up at the
Triton Extreme which, amongst other
things, boasted five times the ROM, more
sample RAM, improvements in storage
and connectivity, and a valve that could
be placed in either the effects or output
signal paths.
When you purchase the Triton in
Collection 3 you receive two separate
soft synths — the Triton Classic already
available in Collection 2 and the new
Triton Extreme, the latter of which offers
three DAC output modes to emulate the
Classic, Studio and Extreme models.
Of course, neither is a full recreation of
a Triton, not least because there’s no
sampler, no sequencer and no option
to run a MOSS engine in either of them.
Less significant differences include (yay!)
a maximum polyphony of 256 voices in
both versions, and (boo!) a reduction of
the maximum multitimbrality to just eight
Parts. However, you can always overcome
the second of these by launching two
instances of the soft synth, so I can’t see
that this should ever be an issue.
So how do they sound? Having sold
my Triton Pro when I was unable to stop
myself from buying an OASYS, I was
nonetheless able to test the Classic
soft synth against my Triton Rack. I was
impressed. In fact, I was very impressed.
Stepping through the factory sounds
demonstrated that Korg have recreated
the HI engine accurately, and I was
unable to find or create anything that
would have failed to drop into an existing
track. Given that the soft synth also
includes most of the sounds from all eight
of Korg’s PCM Expansion boards (the
original could host only two), the sound
creation potential is enormous.
Moving on to the Extreme soft synth,
I was pleased to find that this sounds
slightly different from the Classic;
there were small differences (as there

You can select from

a huge range of sounds
contained within all eight
EXB expansion libraries in
the Triton Classic.

132 February 2022 / w w w. s o u n d o n s o u n d . c o m

 ON TE S T
KORG COLLECTION 3

The Triton Extreme

with its Easy page
displayed for quick and
simple tweaking.

should be) even

with its DAC
emulator set to
Classic mode. So
which is the better
version? In my
view, neither is —
they’re just a little
bit different. If
you want to take
advantage of the
valve emulation,
you’ll obviously turn
to the Extreme, but
I would suggest that
you simply choose
whichever version
you prefer for any
given application
or sound.
Just for fun,
I created a short track using nothing
other than one instance of the Classic
and one of the Extreme, and the results
were fab. Indeed, the 16 Parts available to
me were more flexible than the 16 Parts
on a hardware Triton because I had now
twice as many arpeggiators, twice as
many effects and much more flexibility
about how I allocated the effects to the
Parts. If I have to find a criticism (and I do)
I was disappointed to find that — like the
Prophecy — neither Triton has a MIDI
Learn function. You can allocate MIDI CCs
to allow external controllers to control
their soft equivalents, but there are
potential improvements to be made here;
attaching MIDI CCs directly to parameters
would be a step forward. Nevertheless,
for ‘in the box’ track creation, these two
soft synths could easily become the
bread-and-butter workhorses for many
users. Having been an OASYS and
Kronos user for so long, I had forgotten
how damn good the Tritons can sound,
and these soft synths were a timely
reminder of how musical they can be.
All in all, I have to admit that I enjoyed
using the Triton soft synths more
than I ever enjoyed using the original
workstations, and I soon found myself
whizzing around them as if I had been
programming them for years. Despite the
touch-sensitive screens of the originals,
I found the soft synths’ GUIs to be more
intuitive than the Menu/Select/Open
architecture of the hardware Tritons, and
the new browser and search functions
make it far easier to find the right
Program within the thousands of sounds
on offer, and therefore both quicker and
easier to build Combis.
But despite all of the above, perhaps
the most significant improvements
between the Triton soft synths and their
inspirations are practical ones. Firstly, you
can edit their Programs within a Combi.
Not only that but, when you save a Combi
containing an edited Program, the original
Program is unaffected, so you don’t end
up wrecking other layered or multitimbral
sounds that use it. It’s hard to overstate
what a huge benefit this can be to serious
programmers. Secondly, and perhaps
catering for players at the other end of
the scale, there’s an Easy Edit page that
provides a small but important subset of
Program parameters for those who want
to tweak sounds without diving into the
full HI programming engine.
Conclusions
Covering four soft synths in one review
means that, without taking up 20 pages
of the magazine, it hasn’t been possible
to cover each in any depth, but let’s be
clear about the conclusion... all of them
produce excellent sounds and each
could add something extra to your music
creation. Having said that, you may feel
that I’ve been a little hard on two them,
but I disagree. If something is sold as
a recreation of an original, it should meet
that promise. In my view, the Tritons
achieve this with honours, so let’s hope
that further updates are on the way to
correct the remaining differences in the
700S and Prophecy. If they do, I will have
little but praise for all four soft synths.
Sure, their MIDI capabilities need to be
enhanced to meet modern expectations
— I’m getting a little tired of MIDI CCs
that can only affect a single parameter at
a time — and I think that Korg should add
MPE as a matter of urgency, particularly
on the Prophecy, but there’s much to
enjoy here.
So the final question concerns value
for money. At $149 (approximately £110
at the time of writing) for the 700S, $149
(also £110) for the Prophecy and $249
(£185) for the pair of Tritons, they’re not
low-cost plug-ins. But for just $399 (£300)
you can obtain the complete Collection
comprising the 700S, Prophecy, both
Tritons, plus the MS20, Polysix, Mono/
Poly, ARP Odyssey, M1, Wavestation and
MDE-X multi-effects processor, which
equates to just $40 (£30) per instrument.
That’s excellent value and, despite the
existing caveats with the 700S and
Prophecy, the Collection gets a solid
thumbs up from me.
£ $399 including VAT.
W www.korg.com

134 February 2022 / w w w. s o u n d o n s o u n d . c o m

STEREO ANALOG FILTERS · FULL SYNTHESIS ENGINE · 150 GB SAMPLES · 50 GB USER SAMPLES

SAMPLES +
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HYBRID SYNTHESIZER

w w w . s e q u e n t i a l . c o m
 ON TE S T
PAUL WHITE & SAM INGLIS
M
odesty isn’t always a quality
we associate with Apple. But in
recent years, new versions of
Logic have been quietly released as free
incremental upgrades, when in fact they
introduce major new features. This was
never more true than with the new Logic
10.7. As ever, the full list of enhancements
is vast, and includes innumerable detailed
tweaks and bug-fixes, but the main focus
is clear. Logic 10.7 is about immersive
audio, and specifically Dolby Atmos.
This, obviously, is part of a wider
strategic plan on Apple’s part. Atmos
has been pushed front and centre in
the Apple Music ecosystem, and launch
publicity for the recent M1 Max and M1
Pro processors has likewise highlighted
music applications to an unprecedented
extent. And it has the potential to be
a game-changer for everyone, not least
because 10.7 is once again a free update
for existing users.
Goes From 11
Logic 10.7 is, of course, still Mac-only, and
requires Mac OS 11 ‘Big Sur’ or Mac OS 12
‘Monterey’. It’s still compatible with older
Intel machines, as long as they can run
these operating systems, but includes
136 February 2022 / w w w. s o u n d o n s o u n d . c o m
AppleLogic10.7
Music Production Software
Could a small step for Logic’s version number represent
a giant leap for music production?
many under-the-hood optimisations for
the new generation of M1 chips. A subtle
visual refresh brings the GUI more in line
with the look and feel of current Mac
OS versions, but won’t cause alarm or
confusion among existing users.
Taking advantage of its new spatial
audio features is remarkably simple,
but it’s a good idea to do a little bit
of housekeeping first. Assuming you
also want to retain a stereo version of
your project, it’s probably best to work
either in a duplicate copy, or in a Project
Alternative, rather than rely on a stereo
downmix from the Atmos version. Once
you’re ready to go, you can either visit the
Project Settings and choose Dolby Atmos
from the Spatial Audio pop-up, or select
the new Dolby Atmos item in the Mix
menu — and that’s all you need to do!
The sheer effortlessness of this is
pretty remarkable when you think about
what’s going on under the hood. What’s
apparent to the user is that a new Atmos
plug-in is inserted on the master bus, and
surround panners are activated on every
channel. What you don’t see is that Logic
is taking care of routing, sample rate
management and buffer size adjustment
in the background. Atmos itself demands
a sample rate of 48 or 96 kHz, with buffer
sizes of 512 or 1024 samples respectively.
For best results, it’s probably desirable
to apply offline sample-rate conversion
to audio that’s at 44.1kHz before you get
stuck in, but if you don’t, there’s no need
to worry: Logic will convert for you on the
fly during playback.
If at this point you hit the space bar to
find out what your newly minted Atmos
project sounds like, you might be in for
a surprise. That’s because the Atmos
plug-in doesn’t only handle routing and
panning: it is also a monitor controller, and
defaults to assuming you have a surround
playback system. Reconfiguring this is
done through a simple pop-up menu,
which includes both 2.0 (so that you can
hear how your mix will sound folded down
to stereo) and binaural as well as 5.1, 7.1,
5.1.4 and 7.1.4. This obviously doesn’t cover
all possibilities, but should be fine for most
setups. For mixing on headphones, you’d
choose binaural, but
note that this setting
uses Dolby’s binaural
“You could view a Logic licence as a way of getting
encoder rather than Atmos functionality with a DAW thrown in.”
the one through
which Apple Music audiences will be
hearing your music. More on this later.
Making Beds
A Dolby Atmos mix can contain up to 128
channels, divided between two types of
audio stream: beds and objects. Beds are
conventional channel-based surround
streams containing up to 10 individual
channels (organised as 7.1.2), while objects
are mono or stereo streams that are
packaged with metadata. This metadata
tells the playback device how each object
should be positioned and moved around
whatever speakers are available. (For
more on Atmos and immersive audio in
general, see last month’s cover feature.)
The Atmost standard itself permits
multiple beds, but Logic’s implementation
only supports a single bed. This is unlikely
to be an issue for music mixing: you’d
typically need multiple beds only if you
were working in post-production and
having to combine music, dialogue and
sound effects.
If you click the Output pop-up on any
channel, you’ll see an additional option
called 3D Object Panner. Selecting this
causes the channel to be converted to
an object. Its output is no longer folded
into the Atmos bed, but sent discretely to
the Atmos plug-in along with the panning
data from the Object Panner. And the
beauty of this is that no matter how many
times you change your mind about which
channels should be objects and which
should be part of the bed, Logic handles
all of the routing completely invisibly in
the background.
In some applications, the same
surround panner is used for both bed and
object tracks. This has obvious benefits in
terms of consistency, but it typically makes
the panner itself relatively complex. Apple
have taken a different approach, whereby
channels that are routed to the bed
feature an updated version of the existing
Surround Panner, while bed tracks get the
entirely new Object Panner.
To take advantage of the height
capabilities in Atmos, the Surround
Panner can be operated in Planar or
Spherical modes. In Planar mode, it acts
as a familiar 2D panner: if you want to add
height to your source, you need to adjust
the Elevation parameter separately. In
Spherical mode, by contrast, dragging the
pan position in towards the centre of the
circle increases elevation automatically.
So, if you were to pan dead centre in
Spherical mode, the source would appear
to be above your head, whereas in Planar
mode it would be inside your head, as
it were.
By contrast, the Object Panner
represents the 3D soundfield as
a half-cube rather than a hemisphere, and
provides separate X-Y and X-Z panning
grids. Again, the left-right, front-back
and Elevation parameters can also be
adjusted directly, as can the Size and
Spread of the object. It’s all about as
intuitive and user-friendly as surround
panning ever gets. One non-obvious
difference between bed and object tracks
is that Independent Panning for sends
is available only for the former, but this
makes sense, because otherwise the
send would have to be an object in its
own right.
Plugging The Gaps
The Atmos plug-in itself is a streamlined
version of the Dolby Atmos Renderer used
in or with other DAWs. You can watch
your sources flying around in a ‘head in
a box’ display, with which I think we are
all going to become very familiar, and an
optional text display lists the objects in
your Project, plus all the channels used
in the surround bed. Each of these, apart
from the LFE channel, can be assigned
a Binaural Render flag indicating that it
should be treated as Near, Mid or Far
when monitoring on headphones. The
effects of these settings are often fairly
subtle but they are certainly audible — in
Logic. Unfortunately, these flags are
currently ignored by the binaural renderer
used in Apple Music, which means that
the mix you audition in Logic might not
sound the same when it’s rendered and
distributed. There’s presently no way
to audition in real time through Apple’s
binaural encoder.
One thing that’s crucial to understand
is that there is no ‘master bus’ as such in
an Atmos mix, and hence no scope for
using master bus processing as you might
in a stereo environment. Any plug-ins
inserted before the Atmos plug-in on the
master bus affect only the bed channels,
whilst plug-ins
placed after it don’t
affect the sound at all
— though these slots
might still be useful
for metering plug-ins, for example.
Talking of plug-ins, 13 Logic favourites
have been now updated for 7.1.4
surround use. The most important of
these is probably the Space Designer
convolution reverb, which now has an
impressive library of Ambisonic surround
IRs. The surround version of the Tremolo
plug-in is interesting, as the multichannel
modulation can create movement within
and around the surround space. Also
joining the ranks of the surround-ready
are the Limiter, Match EQ, Chorus,
Flanger, Phaser, Micro Phaser, Mod
Delay, Level Meter and Tuner. There’s
also a multichannel Gain plug-in, and the
all-important Loudness Meter; Atmos is
a loudness-normalised standard so it’s
vital your mixes don’t exceed -18LUFS.
Other plug-ins such as Chromaverb
can operate in multi-mono mode, in the
same way that plug-ins can be set up
in dual-mono mode on stereo tracks
Apple Logic 10.7
£175
PROS
• Even though Logic 10.7 includes a full
suite of Atmos tools, it’s still very
affordable, and a free upgrade to
existing users.
• Streamlined implementation of Atmos
is very user-friendly and easy to get
to grips with.
• Major improvements to the
Step Sequencer.
• Implementation of surround within
Live Loops encourages new
approaches to music creation.
CONS
• There’s no way to monitor through
Apple’s own binaural encoder in
real time.
SUMMARY
Though it’s presented as a point
upgrade, Logic 10.7 and its integrated
Atmos tools have the potential to
change the way music is produced.
w w w. s o u n d o n s o u n d . c o m / February 2022 137
 ON TE S T
A P P L E LO GI C 10 .7
Live Loops can contain Atmos pan automation,
making it possible to carry out loop-based
composition in surround from the off.
to process the left and right channels
independently. Rather than have every
speaker treated to its own reverb setting,
the user can set groups for the speakers
— for example, the left, centre and right
speakers would normally use the same
reverb setting, whereas you may want
something darker and with more pre-delay
in the rear channels.
Steps Will Be Taken
In earlier versions of Logic, the main way
to edit MIDI data was using the Piano Roll
view or in the Event List. Today there are
alternative ways to handle both audio
and MIDI data, including the seemingly
old-school Step Sequencer introduced
in v10.5. Initially this worked pretty much
like a classic drum grid sequencer where
clicking in the squares activated whatever
sound (usually a drum voice) was assigned
to that row.
In Logic Pro 10.7, the Step Sequencer
has evolved quite considerably, extending
its usefulness beyond creating drum
patterns. It now includes a Live Record
mode, making it possible to record parts
directly into the step sequencer from
MIDI drum pads or a keyboard. A new
Live Record button in the top right-hand
section of the window activates this
mode, after which you just need to put the
sequence into Play and start recording.
Notes can also be entered in real time
by using the keyboard cursor keys to
navigate to the correct row and then
using the apostrophe key to enter a note.
There are record options that encompass
Transpose, Quantize, Velocity and Note
Length plus the addition of key, transpose
and scale quantise features similar to
138 February 2022 / w w w. s o u n d o n s o u n d . c o m
those already found in the Piano Roll
quantise editor. There are also some
randomise options for when you run out
of ideas.
Automation relating to plug-ins can
now be recorded directly into the step
sequencer in Live Record mode. Initially
the automation appears in sequencer-like
steps to match the sequencer grid, but
the Automation Latch mode can be
switched to Slide mode to smooth out the
transitions if preferred. Rows can now be
renamed and you can save Sequencer
setups as custom pattern templates, which
is useful when using third-party drum kits
or kits you’ve created yourself.
Another significant addition to the Step
Sequencer is a pitched mode. Originally,
changing the pitch of a drum note would
simply change the MIDI note and therefore
select a different drum voice, but now it
can be used to retune individual hits. In
this mode the expanded grid rectangles
operate as stepped pitch faders working
in semitones. The resulting part can also
be used with the new scale quantise and
transpose options, making it easy to set
up melodies using tuned percussion. You
can also sequence melodic parts, such
as bass lines, and this time around both
Mono and Legato modes have been
added so that you can avoid overlapping
notes. Notes may also be extended to
occupy two or more slots. When mono
mode is on and you enter a new note on
a step that already contains a note, the old
note will be deleted so your new note is
the only note on that step.
Originally you could convert pattern
regions to MIDI regions, but now you can
also go the other way and convert a MIDI
region into a sequencer pattern region
to work on. This allows you to move
between the Piano Roll editor and Step
Sequencer pattern view as appropriate.
Even unquantised MIDI parts can be
converted to sequencer patterns — they’ll
appear quantised to the sequencer grid
but will include timing offsets so that their
original timing is not changed. (Note that
if converting a MIDI region to a sequencer
pattern, the region must not be longer
than 64 pattern steps, otherwise it will be
greyed out. If you need to work on longer
sections, simply divide long regions into
manageable region lengths.)
In The Round
Hand in hand with the improvements
to the Step Sequencer comes a huge
amount of new sample and loop content.
Select New from Template when you
create a project and you’ll find several
templates from producers such as Mark
Ronson and Tom Misch, each of which
comes with a set of curated Apple Loops,
custom instrument and drum machine
patches in both modern and vintage
styles. Although I haven’t counted them
myself, there are apparently 2800 new
loops, 120 new instrument patches and
50 new drum kits including plenty of
808 material in there to keep the EDM
fraternity happy. Open up one of the
Producer Packs in the Loop view and
you’ll see the current Producer Pack loops
arranged in a song-like fashion — and
they can all be used free of copyright in
your own compositions. To help keep
things organised, the Loop browser filter
now includes the Producer Pack names
as well as Logic’s usual loop categories.
You’ll also find that when choosing a new
instrument from the Instrument browser,
there’s a similar filter for homing in on the
sounds from a particular Producer Pack.
Where this gets really interesting,
though, is in the intersection between Live
Loops and the new Atmos features. Load up the Spatial
Audio Demo Grid template, fire off a few cells at random,
and you’ll hear sounds moving around you in surround.
THE MIDI
That’s because surround pan automation can be applied
to regions as well as tracks — and is retained when those
regions are converted to Live Loops. This opens up the
SPECIALISTS
possibility of composing loop-based projects in Atmos from
the off. And that, I think, is the key to understanding Apple’s
approach to spatial audio in Logic 10.7.
It’s an immense challenge to achieve widespread
uptake of a new format for music listening. Apple are
probably the only company in the world who might be able
to do so, and this is partly because they have a stake in
every stage of the process. They see Atmos as an incentive
for people to buy AirPods and subscribe to Apple Music;
but they recognise that this will only work if there is a good
range of really great Atmos content available within Apple
Music. And, logically enough, they figure the best way
to achieve this is to make the process of creating Atmos
music as open and democratic as possible.
Logic 10.7 is the linchpin of this strategy, and does
PRO SOLO MK3
several things that could be seen as game-changing. For MIDI CONTROL FOR YOUR
one thing, it makes available the tools needed to produce VINTAGE & MODULAR SYNTHS
Atmos music at an unprecedented low cost. Previously, if
A must for vintage and modular synth fans, the Kenton Pro Solo
you wanted to do an Atmos mix, you needed a high-end Mk3 enables your MIDI keyboard or sequencer to control CV
workstation such as Nuendo, or a separate suite of synths with V/oct, Hz/V & 1.2V/oct scaling systems.
software tools from Dolby; Pro Tools users, for example,
would have to upgrade to Pro Tools Ultimate and pay for
the Dolby Atmos Production Suite on top. The Suite in itself NEW MIDI FOR YOUR USB
is slightly more expensive than a full Logic Pro 10.7 licence, ADDED KEYBOARD OR CONTROLLER
so unless you need advanced features such as multiple ROUTING
MODES
beds, you could view a Logic licence as a way of getting MIIDI USB HOST mk3
Atmos functionality with a DAW thrown in! And, of course, if Gott USB but need MIDI? You
you’re already a Logic owner, you get all of this for free. nee
ed a Kenton MIDI USB Host to
The implementation of Atmos in Logic is also unique provide MIDI In & Out sockets for
and, arguably, revolutionary in its own right. Stripping away keyboards and controllers which
havve only a USB B type socket.
some of the more specialised or post-specific functionality
has allowed Apple to simplify everything. If you’ve been put
off experimenting with surround because it just all seems THRU-5, MERGE-4
12 & 25 &8
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so complicated and difficult, this is the DAW for you. Your
stereo projects are converted to Atmos at a single click and
within minutes you’ll be flying objects around the room,
losing the listener in surround reverbs and generally having
a ball. Creating an Atmos mix that actually works well and
doesn’t just sound like a stereo mix with some artificial
widening is still a stiff challenge, but that’s the nature of
MIDI Thru Boxes MIDI Merge Boxes
the beast. Apple have done a sterling job of removing the Plug in your sequencer or master It takes a lot more than a splitter
obstacles that prevent people from mixing in surround. keyboard and get 5, 12 or 25 cable to merge MIDI signals
identical MIDI copies with no properly. Kenton Merge boxes
But what’s really new about the Atmos implementation audible latency for your sound combine 4 or 8 MIDI sources
in 10.7 is that it’s not just about mixing. Until now, Atmos has modules and other MIDI devices. into a single MIDI data stream,
been a post-production rather than a production format. with all MIDI data including clock,
Music has been created in stereo and only Atmos-ified as time code and SysEx just as they
should be.
an afterthought, or as the last step in the process. As the
new Live Loops show, Apple clearly want to encourage us
to think in immersive audio terms during the music creation Keep all your synths, sequencers and
process as well. There are still a few obstacles standing in controllers talking with Kenton MIDI products
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it will be fascinating to see, and Find out more at kenton.co.uk CELEBRATING

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hear, how music producers take
this idea and run with it. 35 YEARS

w w w. s o u n d o n s o u n d . c o m / February 2022 139

 GOLDE N GE A R
The KEF KM1
Main studio monitors these days are packed full
of technology — but that wasn’t always the case,
and it took an innovative British hi-fi brand to drag
mains out of the dark ages.
140 February 2022 / w w w. s o u n d o n s o u n d . c o m
PHIL WARD
I
f you visit almost any high-end recording studio,
particularly in the UK and Europe, there’s a pretty
good chance you’ll see a pair of either freestanding
or soffit-mounted main monitors comprising multiple
bass drivers arranged around a direct-radiating (as
opposed to horn-loaded) midrange and tweeter
module — quite often with multiple midrange drivers.
I’m thinking of monitors like the Genelec 1235A,
Dynaudio M3VE, ADAM S6X or PMC QB1A, for
example. The role of all of these monitors, and others
like them, is to generate very high sound pressure
levels over an extremely wide bandwidth, without
significant distortion, compression or coloration, and
to do so reliably for long periods without
rest. They need to reach PA volume levels
yet combine that with the ultimate in
electro-acoustic precision. It’s a genuinely
tough job, and it demands as much
electro-acoustic know-how as it does
manufacturing budget. So you’ll probably
imagine that one of the aforementioned
monitor manufacturers would lay claim
to have developed the first such studio
monitor, but you’d be wrong, because that
honour belongs to British hi-fi speaker
company KEF with their KM1 of 1982.
KEF, starting in the 1960s, had a close
relationship with the R&D department at
the BBC, where all the in-house monitor
designs were created. A number of BBC

Photo: Nick Jennings

designs used KEF drivers, the best known
being the B110 bass/midrange driver
and T27 dome tweeter employed in the
compact LS3/5A monitor, but KEF also
made complete systems aimed at studio
applications. The 5/1AC (which, ironically,
didn’t use KEF drivers) was used widely
in the UK’s commercial radio sector, as
was the later P60, a largely forgotten
but technically interesting semi-active
nearfield monitor.
In the late 1970s and early 1980s,
the BBC found itself with a monitoring
problem in the studios used to record
live sessions for its rock and pop stations,
Radio 1 and Radio 2 respectively. The
main monitors in those studios, primarily
Maida Vale and Golders Green in London,
were BBC LS5/8s. And although the
LS5/8 was designed to satisfy “monitoring
requirements from pop to serious music”
(as described in a BBC R&D technical
report available at www.bbc.co.uk/rd/
publications/rdreport_1979_22), it wasn’t
in practice able to manage the volume
levels required and it kept failing on the
job. The option to equip the studios with
Tannoy- or JBL-style monitors, the latter
comprising large-diameter, lightweight-cone
bass/midrange drivers and horn-loaded
compression tweeters, was of course
available. But those styles of monitor,
effectively repurposed PA speakers, simply
wouldn’t be able to meet the BBC’s needs
for tonal accuracy, minimal coloration and
low distortion. So something new was
required and KEF were asked to make
a proposal. The KM1 was what they came
up with, and it was like nothing that had
gone before.
Active Learning
KEF had at the time a success on their
hands in the shape of the top-of-the-range
The KEF KM1 monitoring setup at the BBC’s Hippodrome Studio, 1987.
hi-fi speaker, the Reference Series Model 105. The 105 amp channels were rated at
was a three-way passive system incorporating a B300, 400 Watts into 8Ω, so the
300mm-diameter bass driver; a B110, 110mm-diameter total power available was, in
midrange driver; and a T52, 38mm dome tweeter. KEF’s theory, 1.6kW per KM1. It’s said
electro-acoustic calculations suggested that a system that both the amplifier power
comprising four of the B300 bass drivers, two modified supplies weren’t quite up to
B110 midrange drivers and one very heavily modified the job of delivering 1.6kW,
T52 tweeter ought to be able to reach the required and that the system’s current
BBC spec of at least 120dB (60Hz to 20kHz) at 2m from limiting protection would often
a pair monitors. But KEF also realised that any practical cry “enough!” a little early, but
speaker designed along those lines and able to reach even so, the KM1 seemed well
the required volume levels would need to be active, so furnished with power. And
they set about working with another iconic UK audio it needed to be. Its drivers
manufacturer, Quad, to help develop the necessary were derived from hi-fi units,
amplifier and electronic crossover modules. fundamentally designed for
The amplification built into each KM1, in a slide-out, flat frequency response and
drawer-style sub-assembly, comprised eight low distortion rather than
single-channel modules derived from the Quad 405 electro-acoustic efficiency,
power amplifier. The amp channels were distributed so even in the loudspeaker
across the drivers such that a bridged pair of realm where anything over
channels was dedicated to each pair of bass drivers, five percent efficiency is going
another bridged pair of channels was dedicated some, their efficiency was
to the midrange drivers, and a final bridged pair of poor. The upshot was that
channels was dedicated to the tweeter. The bridged over 95 percent of each KM1’s
KEF At 60
Along with the KM1’s ground-breaking technical changed: KEF are still one of the most audacious,
achievement, one of the motivations for innovative and electro-acoustic engineering-led
celebrating it in the ‘Golden Gear’ series is to of all speaker companies. A recent example of
mark KEF’s 60th anniversary. Compared to KEF innovation is the development, by Vice President
in the early 1980s when the KM1 was launched, of Technology Jack Oclee-Brown and his team,
the company are a very different organisation of a ‘metamaterial’ wide-band acoustic absorber.
today. Despite their HQ still being at the same The metamaterial absorber is fitted behind, and
Maidstone premises in the South East of the UK, almost entirely absorbs the rear radiation of, the
KEF are no longer a privately owned UK company tweeter on the recently launched KEF LS50 Meta.
and, along with Maidstone, their products are A fascinating AES technical paper that describes
manufactured at multiple sites across the world. this ‘metamaterial’ can be found at https://www.
In an important respect, however, nothing has aes.org/tmpFiles/elib/20211221/20758.pdf.

w w w. s o u n d o n s o u n d . c o m / February 2022 141

 GOLDE N GE A R
GOLDEN GEAR: THE KEF KM1

Photo: Pete Thomas

At a time when studio mains almost exclusively used

horn-loaded tweeters, KEF’s KM1 employed a heavily modified T52
direct-radiating dome.

1.6kW rated power could potentially be dissipated

as heat. And then if you factor in that the Class-A/B
power amplifiers themselves would be only around 50
percent efficient, the KM1 begins to look like a room
heater first and a high-performance monitor second.
A KM1 working hard could quite feasibly be dissipating
something like 2kW.
So a major challenge with the KM1 was to manage
heat such that it was drawn away from the driver voice
coils and didn’t result in thermal compression or driver
failure. To that end, both the midrange drivers and
tweeter incorporated Ferrofluid-filled voice-coil gaps,
both firsts for KEF, and the KM1 cabinet architecture
was designed to create a central chimney, drawing air
from underneath and routing it over 64mm-diameter
cylindrical aluminium heatsinks attached directly to
the back of the midrange driver magnets and then

Photo: Pete Thomas

to a rear heatsink panel. The warmed air would
vent through the central top panel of the cabinet.
The tweeter magnet system also was able to sink
significant heat energy — it was twice the size of that The KM1’s amplification, developed in collaboration with Quad, was built into
used on the 300mm bass drivers. a sub-assembly that slid out of the bottom of the speaker like a drawer. The central part of the
But it wasn’t only thermal issues that KEF faced cabinet functioned like a chimney, drawing heat away from the amp and speaker heatsinks.
in the context of hi-fi drivers working at sustained
high levels. For example, the standard Bextrene format. It’s interesting that even active EQ circuit used widely throughout
thermoplastic diaphragm of the B110 midrange driver though all KM1s delivered to audio electronics) was employed as
was found to suffer from fatigue splits at the neck the BBC were reflex loaded, a consultant to help refine the circuits
when subject to the kind of sustained accelerative the KEF sales brochure of the and to develop a new input stage for the
shocks that 120dB imply. The solution was to swap time illustrates and describes power amplifiers. In addition to its active
to a polypropylene diaphragm (also lighter, so it a closed-box system. It’s likely filter circuits, the KM1’s electronics also
improved efficiency), however this meant not only that customers other than the incorporated a novel method of protecting
a near complete redesign of the driver, but also BBC would have received the drivers from thermal overload. The
the development of new adhesive techniques to KM1s in closed-box format. system initially relied on a continuous
ensure that the diaphragm would remain attached low-voltage DC signal routed through the
to its surround and voice-coil. The low-frequency
Electronics driver voice-coils. Measurement of the
power-handling performance also came under the Along with integrated power resulting DC current could then be used to
spotlight because, although the KM1 had originally amplifiers, the KM1 also calculate the real-time voice-coil resistance
been specified by BBC Radio as a reflex-loaded necessarily incorporated and hence the voice-coil temperature
design, the first monitors delivered were of closed-box active filtering. The filters (temperature and resistance are linked
format (KEF technical director Laurie Fincham was well were designed in-house at via the voice-coil material’s temperature
known to be a reflex-loading sceptic). Unfortunately, KEF, however the renowned coefficient of resistance: 0.39 percent
the closed-box KM1s were found wanting by the BBC analogue electronics specialist per degree C for copper). When the
studio managers in terms of low-frequency power Peter Baxandall (original temperature approached preset limits, the
handling and had to be converted to reflex-loaded inventor of the style of KM1 overload indicator would illuminate

142 February 2022 / w w w. s o u n d o n s o u n d . c o m

 Photo: Pete Thomas
Each KM1 amp module housed eight
single-channel Quad 405 amplifier boards.

followed by automatic driver attenuation

if necessary. The driver protection system
worked, however it had the significant
problem that if any driver was replaced,
the system would require recalibration to
account for the production-line variations
in voice-coil resistance. Later KM1s relied
on a protection system based on thermal
modelling of the drivers that enabled
accurate voice-coil temperature predictions
to be made based on input signal history.

The KM1 Legacy

With its ambition, technology, engineering
and sheer bulk (a single monitor weighed
140kg), the KEF KM1 effectively redefined
very high-performance studio monitoring.
But was it a success? The answer to that
isn’t simple. The KM1 achieved its basic
performance aims and served in the Maida
Vale studios for around six years before
being superseded by the first PMC monitor,
the BB5 (PMC were co-founded by Pete
Thomas, previously a BBC studio manager
with extensive experience of the KM1).
However, the KM1 also had a reputation
for unreliability, and for that reason was
reportedly not well liked by BBC studio
managers. A common subjective reaction to
the KM1 was said to have been that, despite
what the studio SPL meters were saying,
it didn’t really go loud enough, and that
perhaps led to unreliability. Even at 120dB,
the KM1 was said to sound uncompressed
and undistorted. It seems it was loud, but to
some ears it didn’t sound it.
Perhaps the fundamental shortcoming
was that the use of inefficient hi-fi drivers
to reach very high sound pressure levels
over a very wide bandwidth required
heaps of amplifier power — most of which
would wind up as heat. Viewed from
our perspective 40 years later, the KM1
has a whiff of white elephant about it. It
was perhaps something of a technical
cul de sac at KEF in that its development
didn’t appear to contribute much towards
the core business of somewhat more
affordable hi-fi speakers.
Ultimately, 72 pairs of KM1s were
made, and sold for a reported £20,000
each (around £75,000 nowadays), so
Photo: Pete Thomas
in commercial terms it represented
a significant slice of turnover — and the
KM1 undoubtedly prepared the ground
for a whole new style of monitor, ground
that subsequently grew businesses for
the likes of PMC, Genelec and Dynaudio.
Sadly, it appears that the whereabouts or
operational state is known for very few of
the 72 pairs of KM1s, which is a shame,
particularly because I never had the
opportunity to experience the KM1 back in
its heyday, and I would really quite like to
see and hear a pair.
Many thanks to Pete Thomas of PMC for his
help with this feature.

w w w. s o u n d o n s o u n d . c o m / February 2022 143

 ON TE S T

EHX Nano Deluxe Memory Man sets the number of repeats

Delay Pedal
The Electro-Harmonix Memory Man is
one of the longest-running and most
popular analogue delay pedals around
and there have been several incarnations
over the years, some of them quite bulky.
This latest Nano Deluxe version not only
manages to cram everything into a compact
pedal format but also heralds some
worthwhile improvements.
You get the familiar analogue delay
plus modulation (in the form of chorus and
vibrato) but the control layout is a little
different from that of earlier models, in that
the modulation section now has separate
Rate and Depth knobs. The maximum delay
time has also been increased to 550ms,
plus there’s an internal switch to turn delay
tails on or off when bypassed; when set
to off the pedal has true bypass switching.
A Status LED shows when the delay is
active. A standard EHX 9.6V power supply
is included and we’re informed that the
pedal requires 150mA should you wish to
run it from your own pedalboard supply.
There are mono ins and outs on
quarter-inch jacks but there are no control
pedal inputs. The delay time control goes
from 30 to 550 ms. The Feedback knob
and can deliver runaway
feedback at maximum
settings. Input gain is set
via the Level knob, so when
it’s engaged the pedal can
add a useful level boost or
attenuation if required; an
overload LED that warns
when you are pushing this
too hard. The wet/dry mix
is set using a Blend control,
while the Depth and Rate knobs adjust the
modulation, which varies in character from
vibrato to chorus depending on the setting
of the Blend control.
Tonally, this pedal produces the warm,
but by no means dull, characteristic of
a well-designed charge-coupled analogue
delay line: the repeats tend to sit nicely
behind the dry sound, adding support
without sounding too in-your-face. As
the delays repeat, they dissolve into the
background in a very tape-like way too. The
maximum delay time may be a little less
than for some digital pedals but that’s the
nature of the beast.
Longer analogue delay lines tend to
generate more noise anyway and while
there’s a little audible noise here it dies
away as notes decay, suggesting that
some form of compander noise
reduction may be being used.
It could be argued that a little
noise is an essential part of
the analogue delay sound but,
either way, I don’t think the noise
here is unduly obtrusive and, if
anything, it’s less noisy than the
original full-size units.
At shorter settings you can
conjure up a rockabilly-style
slap-back delay and the
modulation sounds rich and musical
as long as it’s not overdone, whether
used at shorter delay times to produce
chorus/vibrato effects or at longer times to
add subtle movement to the delays. With
a little careful modulation tweaking you can
also get very close to a warm and slightly
worn tape-delay sound. The smaller knobs
are a bit more fiddly to get your fingers
around but that’s a small price to pay for
having such a compact format.
Overall then, this pedal captures pretty
much everything that was attractive about
the Memory Man Deluxe. It increases
the delay time and adds more control
over the modulation, and all in a more
pedalboard-friendly package. Paul White
£ £179 including VAT.
W www.ehx.com

Audio Damage Other Desert Cities Delay Plug-in

I’ve been a fan of Audio Damage’s effects plug-ins for many years;
they’re generally quirky and expressive. Other Desert Cities, their
latest offering, is a curiously named take on delay processing, with
both new and some decidedly retro aspects. Compared with some
vendors who seem to take pride in photo-realistic interfaces, Audio
Damage’s visual clarity is refreshing and free of distraction: the
interface is minimal, with subdued colours, pencil-line waveforms
and simple arcs for the control knobs, although there are some
simple animated graphics depicting the audio processing.
Riffing off the Other Desert Cities name (which I’ll abbreviate
to ODC from here), the six distinct delay algorithms are named
after cities in Arizona: Desert Shores (stereo delay), Mecca The Mecca delay algorithm repeatedly buffers the delayed
(reverse delay), Cactus (dual ‘tape loop’), Thermal (multitap component of the signal and plays it backwards, rather in the manner
delay), Mirage (multi-head ‘drum’ delay) and Sky Valley (granular of a live looper.
processing). Regardless of the choice of algorithm, there’s a global Thermal is a stereo multitap delay with up to 16 variable-level
diffusion processor, and controls for regeneration, panning and taps. You choose the number of taps and they’re scaled into the
output level. All algorithms are built around a left-right pair of delay period, allowing for interesting polyrhythms.
audio buffers. The remaining algorithms take us well out of the realm of the
Examining the algorithms in turn, Desert Shores is the closest conventional. Cactus runs a pair of tape loop-like delays, enabling
we get to a conventional delay, with left and right delay times modulatable variable-speed, bidirectional playback. You can
(milliseconds or tempo-based), high- and low-pass filters, left/right randomly sample regions of the delay buffers. At this point we’re
time spread, crossfeed and saturation. With a bit of tweaking of starting to leave the safe territory of predictable echoes and
filters and saturation it’s pretty easy to get passable dub delays or tunings, but Cactus does at least allow the playback rates to be
edgy flangers. Delay times can be swept smoothly; crank them right quantised to semitones, fifths or octaves. (The controls are still
down, ramp up the regeneration and the delays will even behave continuous for automation, but the speed difference is snapped
like audio oscillators. to the nearest quantised interval.) A Respeed switch determines

144 February 2022 / w w w. s o u n d o n s o u n d . c o m

whether the ‘regenerated’ audio routed
back into the effect for feedback is tapped
from the buffered source signal or the
pitch-shifted output.
Mirage is similarly mechanical in concept
and has the same basic controls as Cactus.
A ‘drum’ of playback heads sweeps the
incoming signal viewed as a moving tape;
the number of heads can be altered, as can
their response ‘window’. I found Mirage to
be a smoother, more organic and adaptable
process than Cactus. The multiple play
heads can mimic a multitap delay after
a fashion, but with Respeed turned on,
it’s possible to get lovely Eventide-like
pitch glissandos.
Finally, Sky Valley is the obligatory
granular delay. I’m not a massive granular
fan in general, but a long grain size
and octave-quantised pitches allow
for some rather beautiful harmonic
landscapes to emerge out of simple
monophonic input tones.
My rather gritty granular experiments led
me to explore the global diffusion processor
— great for smoothing the effected signal
through what sounds like a soft, gated
reverb. An internal LFO modulates the
diffusion filters to create facets of chorusing
or tremolo. If the main delay effect is clean
diffusion may not be needed, but it’s great
for adding thickness and blurring artefacts.
ODC has fully modulatable pan controls for
the delays and regeneration signal.
Practically everything in ODC can be
automated smoothly (even things like
the number of playheads, which is a bit
mind-bending), and it comes with a small
but neatly designed modulation system,
sporting two LFOs as well as an envelope
follower that can accept a side-chain
input. Click any control in the device and
then apply modulation to taste; yellow
arcs dynamically show the amount of
modulation in real time. Like other recent
Audio Damage products, the LFO waveform
is continuously variable by shape and
skew (there’s no waveform selector) and,
of course, the LFOs can be tempo-locked
for rhythmic effect. The envelope follower
has attack and decay controls to smooth
its output. Apply envelope following to
the ‘drum’ playback speed in the Mirage
algorithm, and you’re firmly in classic
finger-on-capstan tape-effect territory.
A small selection of factory presets and
a set of curated ‘designer’ presets cover
all bases from dub and reverse delays
through to processes that sound like they
involve multiple pitch-shifters and yards of
open-reel tape. As such, Other Desert Cities
takes on the audio delay world with gusto
and imagination, and manages to balance
sophisticated and other-worldly effects
with a clean and easy-to-use interface.
I thoroughly recommend it as part of your
sound design toolbox. Nick Rothwell
£ $79
W https://audiodamage.com

Z-Plane Tonic Music Theory Assistance Plug-in
Z-Plane’s Tonic is a plug-in designed to work out
the key of a piece of music and to suggest musical
scales and chords that might work with it. It includes
a very basic synthesizer and virtual keyboard that
can be used to replay the suggested notes or to
trigger one-finger triad chords that fit with the scale.
Tonic supports all the common plug-in formats for
Mac and Windows, including Pro Tools’ AAX.
Tonic is used as an insert effect on a track, bus or master and
it can either do its thing in real time or you can set it to work on an
audio file offline. For working out the key of a specific part, such as
a sample loop or a guitar part, then it is best to load Tonic as an insert
on that track. To figure out the key of an entire tune, then Tonic is
best used as an insert effect on the master bus.
On launching Tonic, Realtime Mode is selected by default so
all you need to do is play the audio and the analysis starts. Tonic
shows three possible keys, with the shaded areas around the centre
circular display showing the probability for each one. The key shown
at the top is the most likely match. Sometimes the key of a piece of
music isn’t obvious, perhaps because some notes are not used, or
maybe because the odd ‘accidental’ falls outside the conventional
scale — hence the alternative key suggestions. If there are any key
changes, it’s best to break the song down into sections for analysis.
There are few user controls, but you can decide whether keys
should be shown as sharps or flats and there’s a Reset button to
clear previous data. Furthermore, if there’s a section of the tune that
you know uses non-key notes or that involves a key change, you
can press the Pause button and anything ‘heard’ by the plug-in will
be ignored until you click back into Play mode. Similarly, if you just
want to base the analysis on a specific section of music or perhaps
a sample loop, you only need to play that section. If there are no ‘trick’
sections in the music, then it’s best to play the whole track. In use
the displayed keys may change several times before settling down
as Tonic accumulates more information from the audio so it is always
best to play the longest extract possible unless there are
key changes. The root pitch of A is also displayed, which
is a help where the audio in question isn’t at concert pitch.
A Keyboard View can be opened at the bottom of the
GUI, where the suggested scale notes (always seven)
are highlighted on the virtual keyboard, and you can click
on these to play the notes using the internal synth just
to make sure they fit. A Fold Keys button displays only
the keys containing valid pitches. Selecting the Chords
button places seven compatible chords on single keys
so you can play around with these to see if they match the song just
by pressing the keys that correspond to the chords and if you use the
Fold Keys button, then non-chord notes are hidden. If the suggested
chords don’t fit, then you can switch to one of the other two suggested
keys to see if one of them works better.
Doing an off-line analysis can yield faster results, but you still
use the plug-in — so you must first place an instance of Tonic on
any DAW track. The File button at the top of the GUI can be used
to switch to File Mode, at which point the Play button will change
to a File Load button. The desired audio file is then located via
a dialogue box, and the results will be displayed as per the real-time
mode. Again, beware of key changes that might fool this analysis.
In practice Tonic worked very well for picking out either major or
minor keys and the associated scale notes, and for suggesting main
supporting chords too. It doesn’t seem to recognise exotic scale
types but you can often get clues by also looking at the notes in the
two ‘runner-up’ scales. Being able to see any tuning offset is also
very useful. As with its nearest counterpart, Antares Auto-Key, Tonic
takes a fair chunk of CPU overhead when active but that shouldn’t
be an issue as once the key is known you would normally deactivate
it. For those who can’t always figure out basic keys by ear Tonic is
undoubtedly useful, especially when working with samples, loops
or handling remixes, and it has the advantages of being inexpensive
and very easy to use. Paul White
£ €39
W https://products.zplane.de

w w w. s o u n d o n s o u n d . c o m / February 2022 145

 ON TE S T
FrozenPlain
Slow
Mirage
++++
FrozenPlain Slow is an instrument offering
a selection of mainly gentle sounds that
would slot well into ambient or New Age
compositions, though there are a few
slightly more edgy sounds in there too. As
with previous FrozenPlain instruments I’ve
reviewed, the compact sound library runs
on the Mirage player engine under VST or
AU hosts — AAX is not listed. Most of the
sound sources are pitched but there are
also unpitched textural examples including
the sound of water and various processed
noise samples. Any of the sounds can be
further shaped using the Mirage engine’s
envelope, looping and filter functions as well
as its effects rack and convolution section.
If you already have other Mirage
libraries, these can be accessed using the
library browser pane, though you still can’t
layer sound combinations from different
libraries. Each patch can comprise up to
three sounds, each of which has its own
set of controls for envelope, level, pitch and
filtering. The basic sound source elements
are divided into Textures and Tonals. In
the master section you’ll find the standard
Mirage effects rack offering Distortion, Bit
Crush, Compressor, Filter, Stereo Widen,
Chorus, Reverb, Delay and Convolution, all
with fairy simple stompbox-style controls.
Any control can be mapped to a physical
MIDI controller via a Learn function and
DAW parameter automation is supported.
The factory presets are arranged
into categories: Basses, FX, Keys, Noise
Textures, Pads and Pulsers. These
designations give you a good idea of what
to expect. I found some of the Keys presets
to be a bit aggressive for the genre but
they can easily be tamed using filters and
more abstract convolution options such as
the Shimmer, Formant or Space Digeridoo.
The convolution section really does make
a huge difference to the overall sound and
146 February 2022 / w w w. s o u n d o n s o u n d . c o m
should be considered as one of the more
important sound-shaping tools as it goes far
beyond standard reverb.
Most of the gentle sounds reside in the
Pads section, where you’ll find plenty of
piano-meets-organ-meets-‘host of angels’
patches alongside organ-esque pads and
more textural offerings.
Noise Textures are often effective when
layered with other synth sounds to add
a little Wavestation-style complexity, and
while the Bass section offers little that’s
new, there are some very usable sounds
to be had. I also rather liked the Pulsers
section, which injects a rhythmic element
into the sounds.
Given the low cost of this instrument
alongside its very straightforward interface,
I found it very appealing with plenty of
potential outside its New Age leanings.
Paul White
$59
www.frozenplain.com
IK Multimedia
The Resonator
SampleTank Instrumen
nt
++++
Whether it’s for blues, bluegrass
egrass or country
country,
the ability to play slide, pedal steel or
resonator can be considered something
of a specialist skill within the broad church
of ‘guitar playing’. Fortunately, if you
have a need for the distinctive tones of
a resonator guitar, but not the personal skills
to play the real thing, IK Multimedia’s The
Resonator, produced in collaboration with
developers Indiginus, might be just what
you need to plug the skill gap.
Available at a modest price as an
instrument for IKM’s Windows/macOS
SampleTank 4 virtual instrument (including
the free-to-download SampleTank 4 Custom
Shop version), The Resonator provides
a deeply sampled take on a Hound Dog
Dobro (the metal cone is held in a wooden
body) played with a metal slide. The 1.6GB
library includes a number of different
instrument presets, MIDI performance
patterns and, of course, allows access to
SampleTank’s usual array of sound editing
and effects options.
Sonically, The Resonator gives you an
instantly authentic sound that would fit
right into a blues, folk, country or bluegrass
context. Thankfully (at least, thankfully for
our attempts to fake the performance of
a resonator when we can’t actually play
one), it is also very playable. Each of the
presets offers a different combination
of performance articulations. These use
a combination of conventional keyswitches
but, usefully, also some velocity-based ones.
For example, the Resonator Full Mt preset
provides muted notes, sustained notes and
notes with slides at low, medium and high
velocities respectively, while keyswitches
allow the slides to toggle between up and
down. Yes, like all multi-articulation virtual
instruments, it requires a little practice to
pull off a convincing performance, but it is
much less of a stretch than learning to play
the real thing.
Other presets provide automatic
harrmony notes and there are two presets
that use SampleTank’s very neat ‘strumming’
sysstem, allowing you to pull off basic
stru
ummed parts with ease. There is also
a preset that has a suitable guitar amp
sim
m included for some blues rock attitude.
Finally, the instrument includes a range of
MID preset patterns that can be triggered
MIDI
from within SampleTank covering country
picking, bluegrass and blues. These offer
an impressive demonstration of how good
The Resonator can sound when you have
mastered the key and velocity switching.
Dragged and dropped into your DAW, the
patterns also provide a helpful tutorial on
playing techniques.
Aside from some noise added by
slightly over-zealous use of SampleTank’s
compressor on some of the presets (which
you can, of course, turn off) and the wish for
even more of the very useful MIDI preset
patterns, The Resonator is a pretty cool
package. If you are creating an album of
blues, bluegrass and country, you are going
to want to find yourself a good session
player. However, for more occasional use,
with lots of character and presented in
a very playable format, The Resonator may
be all that’s required. John Walden
£59.99
www.ikmultimedia.com

Spitfire Audio
Abbey Road
One: Grand
Brass,
Wondrous
Flutes & Vibrant Reeds
Plug-in Instrument
++++
Spitfire Audio’s large Abbey Road One:
Orchestral Foundations project features
a 91-piece orchestra recorded in a studio
so staggeringly famous it now has its own
shop and charges for guided tours. After
releasing this 60GB extravaganza in 2020
Spitfire began issuing supplementary
‘film scoring selections’ recorded at the
same time as the main collection. These
affordable mini-libraries provide users with
the immediately recognisable and colourful
instrumental combinations frequently heard
in classic movie scores.
Taking their place alongside Legendary
Low Strings and Sparkling Woodwinds
(see SOS June 2021) are Grand Brass,
Wondrous Flutes and Vibrant Reeds (rock
those adjectives!). All include true legato
performances and feature instruments
played in octaves — by contrast, the
main library has no legatos and is
unison throughout. The selections offer
a near-identical set of articulations and
were recorded from 10 mic positions, with
two additional stereo mixes created by
Simon Rhodes.
Grand Brass combines the library’s
excellent four French horns with a solo
tuba adding weight in the lower octave.
The four main artics are sustain, legato
(great for slow-paced regal themes) short,
emphatic, tightly played staccatos and
tenutos lasting around half a second. In
addition there are crescendo swells of
three different lengths. There’s more — exit
the tuba and enter 10 cello players, who
join the horns in a unison sound which is
the epitome of noble romanticism (think
Gustav Holst’s Jupiter and John Williams’
superhero themes). This majestic ensemble
excels at grandiose chordal passages and
soaring legato melody lines — it plays only
two artics, but both are superb.
Orchestral textures don’t come much
more classic than Wondrous Flutes,
a textbook pairing of two flutes with piccolo
doubling an octave higher. Super-clean
performances, pristine sound and exquisite
tuning give the piping high-register
deliveries a superior edge, with incisive
staccatos and bright-sounding marcatos
sounding delightful. As reported by
innumerable happy users, the mordents
are a treat — played in a choice of four
intervals, these little dancing three-note
figures are great fun and a tremendous
resource for gestural cartoon-ish phrases.
Vibrant Reeds repeats the trick,
this time underpinning two oboes
with two lower-octave clarinets. While
this evokes the folksy atmosphere of
the opening Shrovetide Fair scene of
Stravinsky’s Petrushka, the section’s
exotic, pure-sounding octaves will lend
themselves to any setting. So-called
‘vibrant’ patches see a cor anglais
joining the clarinets in the lower octave,
introducing a slight chorusing and making
the sound thicker and more complex. The
reedy quintet then smashes it with a stellar
set of mordents which can be used to
create amazing, swirling arpeggio-like
effects. Brilliant.
Clocking in at under 10GB apiece, these
pre-orchestrated, easy to use mini-libraries
provide users with instant orchestral colour
for less than 50 quid a pop, a welcome
price tag in these dodgy economic times.
Dave Stewart
£49 each
www.spitfireaudio.com
UJAM
Virtual Drummer BRUTE
Virtual Drummer Instrument
++++
SOS have reviewed a number of UJAM’s
virtual drummer (VD) instruments over the
last couple of years. Each
title provides a combination
of style-specific drum
sounds and MIDI-based
grooves, housed in a UI and
engine that is specifically
designed to encourage an
efficient workflow and avoid
parameter paralysis. The
latest addition to UJAM’s VD
instruments series is BRUTE
and, as the title suggests, the
sounds and playing styles
are aimed at the more raucous end of the
musical scale; think rock, garage, punk
and grunge.
The 3GB of sample content provides
five distinct drum kits while you also get
six global mix presets. And while BRUTE
is easy to use, you do still get options
for tweaking the drum sounds, mixing
and matching individual drums between
the kits, balancing the amount of room/
overhead ambience, adjusting the decay,
tuning and reverb on each kit item and
dialling in compression and saturation
amounts. This gives you pretty broard
scope in taking the raw drum sound from
natural to overblown as suits your needs.
The other key element of the UJAM
VD series is the included performance
options. With BRUTE that provides 31
different playing Style presets, each
containing 23 individual patterns. For each
preset, these are mapped across the MIDI
note range C3-B4 for easy triggering/
sequencing, while individual drum sounds
are mapped across C1-B2 for manual
playing. You can also drag-and-drop
patterns to a MIDI track in your DAW/
sequencer if you want to edit the patterns
or create variations. If your musical muse
has hit, and you need to get a full drum
performance created in double-quick
time, all UJAM’s VD instruments (BRUTE
included) make it incredibly easy to so.
Given the genre targets, it’s perhaps
not surprising that BRUTE’s sounds and
performances have a certain amount of
attitude; it can be big, aggressive and
full-on. Indeed, once I’d added a couple of
rowdy guitars and a gritty bass, there were
times I found myself easing BRUTE back
a little to make some space in the mix,
particularly in some of the busier patterns
featuring lots of splashy cymbals. That
said, in the target styles, I’d rather have the
surplus of attitude and dial it back than not
enough attitude and a wimpy drum track.
While you don’t necessarily get the
level of control offered by the likes of
EZdrummer or Superior
Drummer, if you want to
build a great-sounding drum
part with a minimum of
fuss, UJAM’s approach will
undoubtedly appeal. If you
are looking for something
more mainstream rock, VD
SOLID might be a better
starting point, but BRUTE
is ideal for loud, proud and
likely to scare the neighbours.
As with all UJAM’s products,
you can, of course, download a free trial
version via their website to audition for
yourself. John Walden
£93
www.ujam.com
Audio examples of this month’s libraries are
available at www.soundonsound.com.
w w w. s o u n d o n s o u n d . c o m / February 2022 147
 T E C H NI Q UE
Digital Performer
DP11 brings even more functionality to the handy Chunks feature.
MIKE LEVINE
You can also arrange your Chunks in projects or load Chunks from
the form of a Song. Open a Song window, one file to another using the

D
P’s Chunks feature is unique and powerful. and drag individual sequence Chunks into Load command.
It allows you to nest multiple sequences and it. They will appear as icons. You can then Another powerful feature
virtual instrument combinations (V-Racks) within freely rearrange their order. allows you to make a selection
a single Digital Performer file. You can use Chunks In addition, you can designate the from inside a sequence and
to create new arrangements or alternate versions of start and end times for Sequence and turn it into a new one. Make
a song, or organise multiple cues for a film or video Song Chunks. The end time determines a time-range selection and
score, among other applications. With the release of when one Chunk stops playing and then go up to the Control
DP11, MOTU added additional capabilities for managing the next one starts within a Song. The Panel Sequence menu (the
Chunks to make its workflow even more efficient. Chunks window, which you can open drop-down menu next to the
Before getting into the new features, here’s in a sidebar, is where you create, edit Tempo Slider and Meter Box),
a quick overview of the Chunks feature for those and organise Chunks. It’s even possible and choose Copy Selection to
unfamiliar with it. A DP project can have any number to copy and paste Chunks between DP New Sequence.
of independent sequences in it. Each can be a whole
song, a song section, a film cue or whatever you
want it to be, with its own tracks, its own mixer and its
own effects. Each sequence shows up as a separate
sequence Chunk in the Chunks window.
Chunks can share resources through V-Racks,
which are containers that hold any number of virtual
instruments and effects. One application for V-Racks is
for film-scoring situations, where you’re using the same
instrument palette for numerous cues.

In DP11, MOTU has further streamlined the Chunks workflow. You can access a V-Rack from all the sequences in the Chunks window.

148 February 2022 / w w w. s o u n d o n s o u n d . c o m

 Song allows you to create multiple arrangements using sequence Chunks.
DP also includes Chunk Control Buttons.
Cue Chunks lets you trigger individual
Chunks from their play buttons or via
external MIDI commands. When one Chunk
ends, the next one becomes active but
doesn’t play until you click the play button.
However, if you choose the Chain Chunk
option, when one Chunk finishes the next
one will start playing.
Shiny & New
All the functionality described so far was
in place before DP11 dropped. The new
version adds several modifications to the
Chunks feature set, more specifically in the
form of additional ways to organise or view
your Chunks.
One of
the most
noteworthy
is called
Control Panel
Sequence Menu.
Chunks Folders. These are folders you performance, Playlists differ from Folders
create in the Chunks window in which you in some significant ways. A Folder contains
can drop Sequence Chunks, V-Racks and actual sequences with all of their data, but
even Songs. If you’ve got a large project Playlists only hold aliases for sequences.
with many sequences, having folders to The beauty of Playlists is that because they
organise them is super helpful. contain only aliases, they don’t make your
To create a Folder, go to the Chunks project file significantly bigger, and you can
window mini-menu and click New Chunks create a lot of them. You can’t put V-Racks
Folder. Now, you’ll see a folder icon appear
in the Chunks Window, and you can drag
other Chunks into it. Click on the icon of the
Chunk you want to move and drag it onto
the folder and drop it in. Double-click the
folder to go into it (display its contents in the
Chunks list); use the back button to go up
one level. Like with individual Chunks, you
can name and colour-code Chunks Folders.
But that’s only part of it. MOTU added
another type of Chunk container in DP11:
Playlists. Designed specifically for live
The Chunk
Control buttons
give you various Folders
F ld allow
ll you to better
b organise
i
playback options. multi-sequence projects.
w w w. s o u n d o n s o u n d . c o m / February 2022 149
 T E C H NI Q UE
USING CHUNKS
in them, but a sequence Chunk in a playlist
can access V-Racks in the Chunks list or
Chunks Folder.
If you’re using DP for live performance,
you can create a Playlist with the song
order for your set so that each one is readily
available to trigger without having to look
through any folders. You can also add the
same Chunk to multiple playlists, or even
duplicate its alias within the same playlist.
Split & Polish
Yet another new Chunks feature in DP11
is the Chunks Split view. If you look
underneath the mini-menu in the Chunks
window, you’ll see an icon that looks like
a list with a divider in the middle of it. If you
click on it, it opens the Split View, which
adds a second Chunk pane under the first
one, giving you another view of the list that
scrolls independently. This allows you to
view and browse two separate sections
of the Chunks list at the same time. You
can also drag and drop items from one
pane to the other. For example, you could
double-click a folder or playlist in the bottom
pane and the fill it with sequences dragged
in from the top pane.
The Panes in the Split View don’t
necessarily have to display the same ‘level’
of the Chunks list. For example, the top
pane could display the top level of the
Chunks list, while the bottom pane could be
You can now scale the list fonts larger to better read dense sidebar windows.
150 February 2022 / w w w. s o u n d o n s o u n d . c o m
Digital Performer
Chunks with Split View activated to display top
and bottom panes.
showing the contents of a playlist or folder.
This gives you a great deal of flexibility and
control when managing Chunks.
Zoom It Up
In DP10, MOTU added the Scale feature,
which lets you resize and view the entire
GUI at different zoom percentages. DP11
brought with it another helpful customisation
option: the ability to scale the font size in
any list window, independently from the rest
of the GUI, including the text inside any of
DP’s sidebar windows. It’s straightforward to
use. Go to Digital Performer / Preferences /
Text Size and then adjust the slider for List
Font Size. Separate sliders are provided
for lists, the Project Notes window and the
Lyrics window.
If you work with DP for many hours
a day, this feature can be a real eye-saver
with today’s very high-resolution displays,
allowing you to enlarge DP’s list text to
a comfortable viewing size. If you’re using
DP in a live performance setting, you can
scale the type size so that you can see the
Chunks Lists better from a distance.
Tall & Short Faders
Yet another option for customising the view
in DP is the new Dynamically Resizable
Faders feature. If you change the vertical
height of the Mixing Board window, you’ll
notice that the throw of the faders gets
longer as you make the window taller
and shorter as you reduce the window’s
vertical size.
One of the benefits is that as the faders
get taller, their resolution increases. If you’re
writing automation using faders, you’ll be
able to move in finer increments. If you like
to write automation by moving the faders,
you’ll find it easier to be precise when the
mixing board is taller.
Over the last few major updates, features
like Themes, Scaling, font size adjustments,
and dynamically resizable faders have
made DP’s GUI easier to customise than
ever before.
The New Home For Music Software Downloads
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 T E C H NI Q UE
Put your own stamp on
Live 11’s new content.
LEN SASSO
I
n the December 2021 Live column
we covered the new sound Libraries
included in Live 11. This month we’ll have
a look at how to make their sounds your
own. With the thousands of instruments and
samples that come with Live, why bother
tweaking them? One reason is that sorting
through all the possibilities is tedious
at best and may not get you the sound
you want. Another is that modifying and
combining sounds helps you get a clearer
focus on the sound you want and how to
create it. It is also more fun.
Droning On
Despite the name, Live’s Drone Lab Pack
offers a wide variety of sounds, and it
includes the samples used in its many
instruments. This example uses two
samples from its One Shots/Miscellaneous
folder: ‘Avalon Vox Bed A#’ and ‘Nautilus
Synth Riff’.
Create an empty MIDI track and drag
the Avalon sample to the drop area in the
track’s Detail view. This embeds the sample
1. Drone Lab’s Avalon and Nautilus samples are layered using
Simplers in separate chains of an Instrument Rack.
152 February 2022 / w w w. s o u n d o n s o u n d . c o m
2. The MIDI Effect Rack (top) transposes incoming notes up 16 steps with the probability shown on the
correspondingly labelled knob. The Mood Reel chain in the Drum Rack (bottom) transposes them back down to
play the Mood Reel Rack.
in a Simpler instrument whose mode will
depend on the default setting in your User
Library (Dropping Samples/On Device
View). You’ll want Simpler to be in Classic
mode for this. When you play a C on your
MIDI keyboard you’ll hear the Avalon drone
at A#, as the sample name indicates, so
on the right side of the Controls panel, set
Simpler’s Transpose to +2 to re-pitch it to
C. To layer in the Nautilus sample, enclose
Simpler in an Instrument Rack (right-click
Simpler’s title bar and select ‘Group’), reveal
Live
the Instrument Rack’s Chain List and drag
the Nautilus sample to the Rack’s Drop area
to create a second Chain with a Simpler
holding the Nautilus sample. This Simpler
should also be in Classic mode, but no
transpose is necessary. Next turn Loop on
for each Simpler so that holding a note will
cause both samples to loop continuously.
In many cases you’ll also want to turn Snap
on and adjust the Start, Loop and Length
settings to suit. For Avalon, a Loop setting
of 86 percent with Snap turned on works
well. Nautilus is OK as it is. Now play a note
other than middle C and you’ll hear that
each Simpler’s playback speed changes
with the pitch — that’s most noticeable with
the Nautilus sample. To keep the speed
constant while following the incoming
note’s pitch, select Complex mode for
each Simpler and turn on the Warp button
(bottom right).
Several modulation and parameter
settings make Nautilus sit more nicely in
the mix. Map the MIDI Modwheel to the
Nautilus Chain’s volume to bring it in and
out while playing. On Simpler’s Control
page, turn on the low-pass filter and set
its cutoff to around 1.5kHz. Set Simpler’s
LFO Pan amount to 50 percent and set the
LFO’s beats-mode rate to 1.5.
Drum Swaps
You can add some variety to Drum
Racks and the MIDI clips that play them
by swapping kit pieces on the fly. The
simplest way to swap kit pieces between
two standard, 16-pad Drum Racks (pads
C1 through D#2) is to load the Drum Racks
in different chains of an Instrument Rack.
Insert a MIDI Pitch effect before one of the
Drum Racks and set its Pitch to -16. Then
insert a MIDI Random effect before the

3. One-bar loops of the drum clip on the Pad Swapper track are captured in Take Lanes on
the MIDI Capture track, which acts as the Vocoder’s modulator. The Vocoder’s carrier comes from
the Drone Lab Synth track.
whole Instrument Rack, and set its Choices
to 1 and its Scale to 16. The Chance knob of
the MIDI Random effect sets the probability
that the incoming note will be transposed
up 16 semitones. That takes it out of the
range of the Drum Rack in the chain without
the MIDI Pitch effect, whereas the Pitch
Effect in the other chain brings it within the
range of its Drum Rack.
Screen 2 shows a more flexible
approach in which individual drum pads
have their own probability of being
swapped with their mate in the alternative
Drum Rack. Each incoming pitch is
processed by its own MIDI Random effect
held in a separate device chain in a MIDI
Effect Rack. The MIDI Effect Rack’s controls
section is expanded to 16 knobs — one
for each MIDI Random effect’s Chance
control. To prevent each incoming note
from being replicated 16 times, each chain’s
key zone needs to be set to the single pitch
it is meant to process. The Macro knobs
then control the probability of swapping
individual kit pieces — the knob value
shows the chance of playing the alternative
kit piece. In screen 2, rather than using
a standard Drum Rack for the alternative
sounds, I’ve assembled a Drum Rack
comprising sounds in the Mood Reel Pack’s
Samples/Percussion folder. Samples with
less noticeable pitch work best, but you
can get interesting and usable substitutions
from a wide variety of sources.
Some drum swaps are better than
others, so you’ll often want to capture, edit
and rearrange them. You can capture the
swapped MIDI output in Session view clips
or as individual Takes in Arrangement view.
I prefer to do it in Arrangement
view, and the top of screen
3 shows one way. The first
step is to move the Instrument
Rack holding the two Drum
Racks intended to play the
clips to its own track and to
set that track’s input to come
from the Post FX output of the
original track, which still holds
the pad-swapping MIDI Effect
Rack. Copy the source MIDI
clip to Arrangement view and
set Arrangement view’s loop to
enclose the clip. Create a Take
Lane on the capture track and
copy the source clip there.
Now start Loop recording
in Arrangement view and
each pass of the loop will be
recorded in a new Take Lane.
Talk Shop
A vocoder can do a lot more
than make your synth sing (for
details, see the September 2011
Live column). In screen 3, I’ve
used Live’s Vocoder to combine
the Drone Lab synth in screen
1 with the Mood Reel drums
in screen 2. Here the drum
groove becomes the modulator
(the voice) in the Vocoder and
the drone synth becomes the
carrier (the synth). The first step
is to insert the Vocoder after
the Mood Reel Drum Rack on
the MIDI Capture track. Then
use the Vocoder’s Dry/Wet
knob to balance the drum and Vocoder outputs. Next
insert the Drone Lab synth on a new MIDI track and set
the Vocoder’s carrier to come from the Post FX output
of the Drone Lab Synth track. Selecting Post FX rather
than Post Mixer lets you use the Drone Lab Synth’s track
volume to mix in the dry synth. The remaining Vocoder
controls provide a broad range of options to colour
the Vocoded synth sound. Pay special attention to the
number of bands (Bands setting) and the individual
band volumes, which are controlled by dragging the thin
orange bars above the bands. The Formant knob is also
critical — it shifts all the band frequencies up or down.
Voice Box
The most straightforward way to manipulate samples
or the instruments that play them is with audio effects.
The example in screen 4 starts with the Digital Voice
Machine instrument from the Voice Box Pack. That
features four quite different presets that all work well in
this example. An Audio Effects Rack with three chains
is inserted after the instrument. The first chain is empty,
the second holds Live’s Frequency Shifter effect (‘Freq
Shift Flanger’ preset) and the third holds the Live/AAS
Corpus effect (‘Drum Rezo Gate’ preset). The Rack’s
Chain zones are set to crossfade through the three
chains, and the chain layout is designed to work well
with either a MIDI controller such as the Modwheel or
with clip automation or modulation.
The Rack has been set up with a single knob to
control the Chain Selector and that knob is mapped to
the Modwheel. Clip modulation applies to the current
setting of the knob, whereas Clip automation is absolute.
Setting the knob manually or with the Modwheel
disables any automation in the playing clip — the
transport’s Re-Enable Automation button then lights up,
and clicking it re-enables automation. Triggering a clip
containing automation also re-enables it.
4: The output of the Digital Voice Machine instrument is
processed by a three-chain Audio Effects Rack (top) whose single
Macro knob controls the Chain Selector.
w w w. s o u n d o n s o u n d . c o m / February 2022 153
 T E C H NI Q UE
Pro Tools
JULIAN RODGERS Speed up your mix workflow with these
handy keyboard shortcuts.
I
n a recent conversation about control surfaces,
I mentioned that my all-time favourite surface was
the Digidesign D-Control. I’ve used EuCon surfaces
and ICON-series controllers in the past, but in spite of
its sometimes impractical size the D-Control had an
advantage none of the others had, and that was an
integral keyboard.
When talking about control surfaces and DAWs the
conversation often draws a distinction between using
a control surface or using the keyboard and mouse. The
lumping together of the keyboard and mouse as the
default controllers for software is understandable but
they are two distinct devices, and when thinking about Many control surfaces include
my preference for Pro Tools over the other DAWs I use, all the usual modifier keys as
a lot of this preference is rooted in this distinction. buttons, which opens up a world of
Logic Pro, for all its impressive capabilities, has mouse-click shortcuts.
often frustrated me because many power users choose
custom keyboard configurations with their own choice of plug-in presets and track a limiter there as the level jump caused by
keyboard shortcut. Studio One is enormously accessible presets are all displayed, and bypassing a limiter will make meaningful
partly because it is so heavily mouse-driven, but Pro icons shown next to the name comparisons difficult.
Tools has a fixed set of keyboard shortcuts which are let you know whether you are
as powerful and useful as they are precisely because of looking at a plug-in, a plug-in
Paradigm Shift
their fixed nature. They constitute a common language, preset or a track preset. Back to the Shift shortcuts introduced in Pro
a standard in studios the world over. Another reason not to abandon Tools 11. These add useful functionality in
Leaving editing to one side, here I’ll suggest why your keyboard when you mix! that they allow bypassing by type. Shift+A
it is that the commonly asserted aim of replacing the bypasses all the plug-ins on selected tracks,
keyboard and mouse with a control surface might to
Command & Conquer and this functionality has been introduced
some extent be missing the point. Even control surface One feature which first made in the most recent EuCon update for Avid
designs themselves acknowledge this, by including its appearance in Pro Tools S1 users. The previously unused X button
hardware buttons that do nothing more than duplicate 11 and is just so useful it’s (Surface Shift + Page Right) now bypasses
the function of a keyboard modifier. And the D-Control’s hard to remember life without all inserts on a selected track when
big advantage was it didn’t draw a distinction between it is the ‘Shift+something’ displaying inserts on a per-slot basis. When
the keyboard and the surface, the keyboard was part of family of shortcuts, which an insert’s parameters are spilled across the
the surface. allow for powerful global eight encoders, the X button bypasses the
One of the occasions I most value the keyboard, bypass toggling. Before these focused plug-in.
and particularly the modifier keys, when mixing is when were introduced we could Other similar shortcuts include Shift+C
it brings functionality I associate with using hardware of course bypass a plug-in for dynamics, Shift+E for EQ, Shift+V for
keyboard controllers into the keyboard and mouse by Command-clicking on it Reverb, Shift+W for modulation and Shift+D
paradigm. ‘Clutching’, for example, is the name of the (Control-clicking on a PC). for delay. You can mute sends using Shift+Q
incredibly useful feature found on fader control surfaces Option-clicking (Alt) allowed and mute sends A-E and F-J using Shift+4
whereby a Mix Group can be temporarily suspended bypassing of all the plug-ins in and Shift+5. These are nice to have, but
by moving a fader whilst touching another fader. With a particular insert slot across the really useful ones are Shift+2 (on the
a Mix Group active, the faders would usually be ganged every track, and Control (Start) alphanumeric keyboard) to bypass inserts
together, but by touching more than one touch-sensitive allows bypass to be toggled A-E and Shift+3 to bypass inserts F-J.
fader cap, the ganged behaviour is suspended. You on that insert and all the These can be used to set up quick A/B
can replicate this from the keyboard by holding Control inserts which follow. Adding comparisons between alternate processing
on a Mac or Start on a PC while moving a fader with Option to Control brings the chains — and by selecting tracks and adding
the mouse. fantastically useful option to shift, this can be done across sections of or
While it’s always been possible on control surfaces to bypass all plug-ins in a session. even entire mixes.
select and instantiate plug-ins, the experience on older You can of course add Shift The way to do this A/B comparison is to
controllers was laborious, involving navigating menus to this combination, and set up your first processing chain in slots
using rotary encoders and viewing the menu a single that gives you the option to A-E and your alternative processing on F-J.
item at a time. While it was doable, it really wasn’t bypass plug-ins on all selected (If you’re using more than five plug-ins on
preferable to using the mouse. Since Pro Tools 2018.7 tracks. An excellent use of this a single track this won’t work, but If I were
it has been possible to search for plug-ins and busses, combination is if you want to using more than five plug-ins on a single
and this system is so simple and effective that it’s hard to bypass all your processing track I’d probably be asking myself whether
imagine how we got by without it. To display the search apart from on your main mix they are all necessary!) Once set up and
field you need to click a slot and hit Tab, but you can bus, which is particularly level-matched, if you start with inserts
actually begin a search by just starting to type. Plug-ins, helpful if you are using A-E engaged and F-J bypassed you can

154 February 2022 / w w w. s o u n d o n s o u n d . c o m

alternate between your two alternative plug-in chains
by hitting Shift+2+3 together at the same time. Add
appropriate Option and Shift modifiers to do this across
as much of your session as required.

Under Control
Intrinsic to any good mix is automation, and using the
keyboard you can jump directly to any automation
parameter’s playlist by holding Command+Control
(Control+Start on a PC) and clicking on the relevant
parameter’s control. To go to pan automation, hold the
modifiers and click the pan control; it’s the same for
send levels or mute buttons. Volume can be accessed in
the same way but actually there is already a single-key
Command Focus shortcut available: just hit minus on the
alphanumeric keypad to toggle between waveform and
volume. To get back to waveform view use the same
modifiers and click the track name.
To display automation for plug-in parameters, you
first need to enable them. By default Pro Tools only
offers automation playlists for volume, pan, send Click a plug-in slot, and you can search for all available plug-ins just by typing.
levels and mutes. You can choose to enable just the
parameters you require (the list can get overwhelmingly
long if you enable all available parameters in
a complex reverb or synth), but many prefer to hold
Command+Option+Control (Control+Alt+Start on a PC)
and tap the automation enable button in the plug-in
window’s header to enable everything for convenience.
If you navigate your automation playlists using the
Command+Control technique combined with mouse
clicks on the relevant control, you can fly around your
automation without ever having to see the huge list
of automation playists. Even better is combining this
with a EuCon controller. Using the Command and
Control modifiers built into the control surface, the
touch-sensitive encoders and faders can be used
to navigate and display automation playlists as you
write automation from the surface — an extremely
effective system.
On the subject of writing automation, something
which can really help getting tricky automation passes
bang-on is to use half-speed playback. Often first
discovered by accident, holding Shift and hitting play
will trigger this slowed-down playback mode, giving
you more time to get your moves just right. And it
sounds funny, so win-win! The other secret weapon
for nailing tricky automation is Back and Play. There
is no convenient keystroke for this so I’d recommend
accessing it as a Soft Key from Page 2 of the Shift keys
in the Avid Control app.
If you are a current user of a worksurface you’ll
probably have already discovered that it isn’t an either/or
choice between keyboard and mouse and worksurface,
but the current implementation of EuCon, the free Avid
Control app, and the accessibly priced S1 and Avid Dock
mean that it is perfectly possible to reach the mixing
nirvana of not looking at the screen for so long that your
screensaver kicks in without you noticing. Just be aware
that while you might not be using your mouse, you have
brought the most important four keys of your keyboard Command+Option+Control-clicking on a plug-in’s automation enable button will enable
with you over to your control surface!  automation for all of that plug-in’s parameters, and create a new automation lane for each.

w w w. s o u n d o n s o u n d . c o m / February 2022 155

 T E C H NI Q UE
The Input Transformer can do all sorts of useful things to incoming MIDI signals.
JOHN WALDEN
M
ost Cubase users will know of
the Logical Editor, which is an
amazingly powerful tool for
transforming MIDI parts you’ve already
recorded, but the Pro edition of Cubase
also includes the Input Transformer. This is
a more streamlined tool that operates in real
time on incoming MIDI data and it’s capable
of performing some very useful tricks. In this
article, I’ll walk you through some simple
examples that demonstrate the possibilities.
DIY Switch Builder
Many sample libraries have the ability
to keyswitch between different sounds
(for example, string section performance
articulations). But for those that don’t, you
can turn to the Input Transformer. The
main screenshot shows an example which
switches between different sounds in
the first three channels of an instance of
HALion Sonic SE (HSSE). By default, these
sounds respond to MIDI channels 1, 2 and
3, respectively. I’ve constrained the MIDI
note range to C1-G8 for each sound, so as
to free up the MIDI notes below C1 for use
as DIY keyswitches. Also note the instance
of the MIDI Monitor plug-in that I’ve placed
in the track’s MIDI Insert panel; this lets you
see, as you experiment, exactly what MIDI
data is reaching HSSE.
156 February 2022 / w w w. s o u n d o n s o u n d . c o m
To access the Input Transformer click
on the ‘squiggly arrow’ button in the
top-most Inspector panel, and you’ll see
three options: Off, Global and Local. Global
means the Input Transformer actions you
create will apply to the incoming data on
all the project’s MIDI and Instrument tracks.
With Local, selected here, only the current
track’s MIDI input will be affected. The Input
Transformer window looks similar to the
Logical Editor: in an upper Filter Conditions
panel you specify which MIDI events you
want to trigger an action; and, beneath,
an Action List panel specifies what actions
will be performed when those MIDI events
are detected. The four Module tabs, each
activated by a small ‘power’ button, allow
you to configure up to four independent
Input Transformer setups, all of which
operate on the same input signal.
Assuming your external MIDI keyboard
is set to transmit on MIDI channel 1,
the instrument in the HSSE’s first slot
will be the default sound. In the main
screenshot, though, you can see how
I’ve set up Module 1 so that when you
press a keyswitch the incoming notes are
assigned to MIDI channel 2 — so HSSE’s
second instrument plays instead. In the
upper Filter Conditions panel, the Input
Transformer is instructed to look for the
MIDI note D0 (the Last Event is Equal
to note 26/D0) and to understand that
Cubase
Cubase Pro’s Input Transformer; somewhat
intimidating but capable of some very useful tricks.
the filter condition is still being satisfied
while the ‘Note is playing’ (ie. until you
release the note). The lower Action List
panel is configured so that when the Filter
Conditions are met (ie. you’re holding
down note D0) the incoming data’s MIDI
channel is changed to 2. Thus, any MIDI
notes that arrive while D0 is being held
down are assigned to MIDI channel 2. So in
this example your keyboard’s D0 becomes
a non-latching keyswitch that accesses the
Synth Pizzicatos (HSSE slots), but switches
back to the default Ensemble Strings patch
in the first slot upon release.
The Module 2 settings are identical,
except that MIDI note E0 is the keyswitch,
and it reassigns notes to channel 3. You
could set up Modules 3 and 4 similarly so
this approach offers you the ability to switch
between up to five different instruments or
articulations, including the default one.
Hard Choices
Some virtual instruments also offer
velocity-based sound switching, whereby
different MIDI note velocities trigger
different sounds, and your DIY keyswitch
can achieve the same thing for instruments
that lack this feature — you just need to
make some small tweaks to the Input
Transformer’s Filter Conditions section, and

Using MIDI velocity to control the MIDI channel provides an alternative approach to DIY sound switching.
the second screen shows what’s required.
As before, the default sound is on channel
1. The second screenshot shows what’s
required and, again, a single condition is
specified within the Filter Conditions panel.
This time, it looks to see if the incoming MIDI
note velocity is larger than 64 (you can set
this to whatever velocity value you prefer).
If this condition is met, then the Action
List commands are executed. So, when a
MIDI note arrives with a velocity greater
than 64, the MIDI channel is switched to
channel 2 before the data is passed on to
the virtual instrument.
As seen in the image, Module 2 is
also active. The entries in this module are
identical to Module 1, but the threshold
velocity is set to 100 and the MIDI channel
set to channel 3. The combined result of
both modules is a three-way, velocity-based,
switching between MIDI channels 1 (velocity
0-64), 2 (velocity 65-100) and 3 (velocity
101-127). It’s very simple, but it’s effective.
Generic Controller
Of course, the Input Transformer can
be used with more than just MIDI notes,
velocities, or channel numbers — it can
also be used to convert any MIDI data into
anything else. Another simple example that
demonstrates this pretty well is identifying
a specific MIDI CC number and changing
it to a different CC number or, if you prefer,
into a different sort of continuous MIDI
data entirely.
One such transformation might be
to turn the Mod Wheel data (MIDI CC1)
into Aftertouch data (in this case, strictly
speaking, it will be Channel Pressure; the
same Aftertouch value will be sent to all
notes). This could be useful if you’re trying
to get the most from a virtual instrument
sound that responds to Aftertouch, but don’t
have a keyboard that offers Aftertouch, or if
its Aftertouch capability is less than smooth
(as, in my experience, is sometimes the case
with compact MIDI keyboards of the sort
you might use when travelling). Provided
that the keyboard offers some sort of CC
data control — a Mod Wheel is the most
likely offering, even on small keyboards
— then the Input Transformer can come to
the rescue.
The final screenshot shows the Input
Transformer settings that are required to
achieve this. Hopefully you can see that
the Filter Conditions panel now contains
two conditions, along with an ‘And’ entry
in the final ‘bool’ column. The conditions
are set so that incoming MIDI is identified
for transformation if it is, first, a controller
message and, second, its CC number is
1 (by default, the Mod Wheel transmits as
MIDI CC1). Only if both of these conditions
The Input Transformer can also be used to
manipulate controller-style MIDI data in various ways.
are met will the Action List transform the
CC1 data into Aftertouch data before
then transmitting the event to the virtual
instrument. Obviously, with some slight
tweaks to the Action List panel, you could
target a different parameter (for example,
another CC number) with your Mod
Wheel CC1 data.
Input Transformer
For Dummies?
A couple of points are worth making
to bring this gentle introduction to the
Input Transformer to a close. First, once
you’ve designed an Input Transformer
configuration, it’s worth saving it as a preset
(to add to those already supplied by
Steinberg), so you can recall it later.
Second, try not to be put off by the
rather opaque nature of some of the Input
Transformer’s settings. In particular, the
meaning of Value 1, Value 2 and Value 3,
seen in the column pop-up menus, can
leave you scratching your head. These
refer to different parameters depending on
whether you’re targeting MIDI notes, CC
numbers or something else. For instance, in
our final example, in the second line of the
Filter Conditions panel, I had to select Value
1 in the first column’s pop-up menu. There
is no ‘MIDI Controller Number’ entry in that
menu, but that’s what Value 1 represents
when the ‘Type is’ is set to Controller in the
first line.
This sort of thing can be difficult for
some people to get their head around,
and is perhaps the thing that new Input
Transformer or Logical Editor users find
most confusing. While the Cubase Pro
Operational Manual PDF has some helpful
information, I keep hoping that, one day,
a Logical Editor guru at Steinberg will fully
document the various options! This would
undoubtedly enable more users to benefit
from the vast potential of the Logical Editor
and Input Transformer.
w w w. s o u n d o n s o u n d . c o m / February 2022 157
 T E C H NI Q UE
Studio One
We show you how to configure your inputs, outputs and audio routing.
input and output latency of your audio system. We’ll come onto up correctly for easy selection
ROBIN VINCENT
latency considerations in a moment. You can usually also launch in your song. And if this

W
hen it comes to your audio interface’s own control panel from here, although not changes from song to song
routing audio all audio interfaces seem to support this button. then as the configuration
through your gear, It would make sense for Studio One to give you access to is also saved it changes
Studio One can act as a hub the input/output configuration of your audio interface at this along with it.
and routing matrix. You can point, but it doesn’t. You can’t get into those settings until you’ve To start configuring
control exactly where your created a project. While that may seem odd, what it means is your inputs and outputs hit
audio is going and define that the audio setup, which defines which inputs and outputs the Add (Mono) and Add
where it comes from. You are active and how they are connected, are all saved with your (Stereo) buttons. These
can create effects loops, project rather than dictated globally. That’s probably worth the will speculatively name
monitoring destinations and extra clicks it takes to get to those settings. themselves and become
internal routings to flow attached to the next available
tracks and inputs, outputs and
Song Setup input or output. It’s common
busses all over the place. In Once you’re in your song you should click on the Song menu practice to create a stereo in/
this workshop we’re going item and select Song Setup... This is where you’ll find the Audio out for each odd/even pair
to get into the audio settings I/O Setup page. Here you’ll see an Input and an Output tab of physical connections, and
and routings to see if we can which will display all of the physical hardware inputs and outputs a mono for each individually.
discover ways of handling to your system, and the virtual ones too if your interface supports To rename them simply
your audio that perhaps you software loopback. Studio One will pre-configure a few for you, double-click the text, and
haven’t considered before. usually generating a couple of mono inputs and a stereo input then click in the matrix at
based on the first two inputs on your device. the intersection of what you
Setting Up An The configurator is set up like a matrix, where you have a row want that input or output to
Audio Interface of physical or virtual inputs/outputs along the top and how you be connected to. Alongside
Before any routing can be want to address those within Studio One down the side. So, the audio channels you’re
managed, Studio One needs when you go to select an input for recording on a track you don’t creating are little signal
to know what audio interface see a list of available inputs as defined by your interface, you monitors so you can easily
or audio system you are see the list of inputs you’ve created and named here that are see that you’ve got the right
using. When Studio One assigned to the actual inputs on your interface. While this may thing going into the right
boots to the home screen, initially seem long winded, it lets you create maps of everything place with the right name.
you’ll find these settings you have connected to your audio interface and have it labelled You can colour them to
accessible in the middle at make them easier to spot
the bottom. Click on it to or for creating visual groups
bring up the audio device of inputs, and you can add
settings window. more than one at a time using
In Mac OS you can set the Add... button. As inputs
a playback and recording and outputs get allocated
device, whereas in Windows to sources and destinations
you simply choose a single within your song they get
audio device from the a dot in the Used column
drop-down list. The list to give you a useful overall
should contain all currently picture of what parts of your
installed audio devices, audio interface are getting
which in Windows will some action. 
include the regular Windows The matrix can be as
sound engine for using your powerful as you want it to be.
on-board sound facilities, as You could set up eight inputs
well as any ASIO-driven audio labelled ‘Input 1’, ‘Input 2’, and
interfaces. If you are using so on connected to the first
a PreSonus audio interface, eight inputs on your interface
then you’ll be treated to — job done. But you can also
a neat little photo of it. have them mirror your studio
Choose the one you want to layout with names such as
use for this session. ‘Vocal Mic’, ‘Guitar 1’, ‘Prophet
Once chosen you can 5’, ‘Modular 1-8’ and so on.
specify the ‘Device Block You could create coloured
Size’, also known as the buffer The Audio I/O setup window is where you create input and output tracks, and sections for synths or drums,
size, which will affect the assign them to your audio interface’s physical I/O. label up effects loops, or set

158 February 2022 / w w w. s o u n d o n s o u n d . c o m

up surround-sound speaker
configurations. There may
be inputs or outputs that you
don’t want to use, in which
case don’t create an option
for them. Taking a thoughtful
approach is going to make
selecting inputs for tracks and
effects routing far easier once
you’re in the creative zone.
Inputs For Tracks
Once you’re adding and
recording tracks in your song
then all the preparation you
did in the Song Setup starts
to pay off. For an audio track
you’ll find all the inputs that
you specified listed, along
with any colour you defined.
And one rather nice bit of
usefulness here is that the
input is present with the name
you gave it, but alongside
you’ll also find they’ve
included the audio interface
destination, meaning that you
don’t have to go back into
the Audio I/O setup to remind
yourself of what the physical
connections are (although
that setup window is only
a click away as they’ve stuck
a shortcut to it on the bottom
of the list).
As you create more tracks
you might notice that they
start appearing on the Inputs
list under the Tracks pop-out
Once you’re in your song, the
inputs you defined in the Audio I/O
Setup become available as track
input options.
a special bag of magic to
deal with this latency, which
is invoked by the use of
a strange Elfin rune that looks
a bit like the letter Z...
In the Audio Setup
window, you’ll notice a tab
called Processing. Here is
where we deal with a clever
system that maximises the
mixing performance while
keeping latency low for
monitoring and instruments.
Studio One calls it Dropout
Protection. What it does
is it splits the business of
handling audio playback and
monitoring/virtual instruments
into separate tasks. So, you
can keep the audio of large
projects moving smoothly
while keeping the latency of
input and instrument tracks
low enough for real-time use.
This process assumes that
The Dropout Protection option at the top of the Audio Setup window defines any plug-ins you’re using
an additional ‘safety’ buffer for avoiding dropouts. The ‘Enable low-latency don’t add more than 3ms of
monitoring for instruments’ option lets you selectively bypass this buffer so that
latency, otherwise the magic
you can play virtual instruments with minimal latency.
won’t work.
list. This is because Studio One lets you route the output of There are different scales
one track into the input of another, which brings up all sorts of of Dropout Protection, and
interesting possibilities.  you’ll find that it only really
comes into play once you’ve
Console Outputs boosted the protection to
All the tracks feed their audio into the console for mixing, and Medium, High or Maximum.
it’s here that we can specify the output destination. How this This adds a larger buffer
is configured will depend on your setup. If you are routing to the playback of audio.
through a hardware desk for mixing then you may have a bank Meanwhile, if you tick ‘Enable
of individual outputs. But more commonly, Studio One is the final low-latency monitoring for
destination of our audio and so all you really need are monitoring instruments’, the magic
outputs plus any virtual inter-software routing. However, if you Z will appear and you’ll
have a number of outputs available then you could be using be able to monitor at the
them as sends to external processing gear. You can label these lowest possible latency set
outputs up as such, and then label a couple of inputs for the by your audio interface’s
stereo returns back into Studio One. The more complex your buffer size. The magic Z also
setup the more Studio One can help you with this. appears on the main output
Any additional outputs you create will appear over by the fader in the console, and
Main output in the console and also in the output list for all on any tracks where you’ve
tracks. You can actually change the output assignments here enabled monitoring.
in the console rather than going back to the Audio I/O page if These low-latency
you wish.   monitoring options enable
you to make full use of
Low-latency Monitoring & Dropout Protection external processing by routing
One important aspect of your audio interface is how it holds audio out of your computer,
control over the latency of the audio passing through. This is through a processor and
usually controlled by the buffer size set in your interface’s control back in, either through audio
panel, but Studio One, like many other DAWs, tends to add in routing or using the Pipeline
some safety buffers of its own to ensure the smoothest possible XT plug-in that I talked about
playback when mixing. These safety buffers can add latency to in my very first Studio One
the audio, which can become noticeable when monitoring live workshop about three years
inputs or playing virtual instruments. Thankfully Studio One has ago. How time flies! 
w w w. s o u n d o n s o u n d . c o m / February 2022 159
Q How should I align meters
when working with both
hardware and software?
I need to align some analogue gear to
work with digital audio embedded in
a video broadcast. My main mixer has
a maximum output of +26dBu, but its
output goes to a delay/EQ unit with
a maximum output level of +21dBu, and
then an audio distribution system with
a maximum output level of +22dBu. At
the end of the chain, once the signal
has been digitised, I’m metering on
a digital PPM meter.
So, I know that I have to play a 1kHz
tone in my mixer with the master set to
read 0VU (+4dBu) on its meters, and that
I can then adjust the delay and distribution
system input levels so that their meters
show their nominal reference levels too.
I also know the maximum output level of
each item of equipment and I’m assuming
that this should equate to 0dBFS (digital
clipping)… but they are all different! So
which device should be my ‘anchor’, so to
speak, and what dBFS should my digital
PPM read at the end of the chain with that
1kHz, 0VU tone from the console? Thanks!
SOS Forum Post
Technical Editor, Hugh Robjohns
You’re over-thinking this! It’s really quite
simple: you just need to think of the
analogue signal chain as a whole. The
device that clips earliest is the one that
defines the maximum capability of the
whole chain, and that’s your delay/EQ
unit at +21dBu. So if you want to make the
digital system clip at the same level as the
analogue chain, +21dBu is the maximum
permissible analogue peak level. Your
console’s reference level (0VU) is +4dBu,
which is 17dB below the maximum allowed
peak level, so your 1kHz tone at +4dBu
should be aligned to read -17dBFS on the
digital meter.
However, while this will work perfectly
well, it is a ‘non-standard’ analogue-digital
alignment. The European Broadcast Union
(EBU) specification calls for the analogue
reference level to align with -18dBFS
— giving fractionally more headroom than
your configuration offers. The reason the
EBU selected this apparently obscure
value stems from two factors. First, as
broadcast audio inherently has a carefully
controlled dynamic range it requires only
160 February 2022 / w w w. s o u n d o n s o u n d . c o m
a modest headroom margin and, second,
the analogue PPM meter traditionally
used to assess signal levels intentionally
under-reads on transients, so the headroom
allowance must accommodate that.
European broadcasters traditionally work
with an analogue reference level of 0dBu,
and programme peaks must not exceed
+8dBu (in the UK) or +9dBu (in Europe),
as measured on an analogue ‘quasi-PPM’
(peak programme meter) — a critical fact
that is often forgotten! That (for historical
reasons) the q-PPM under-reads on
transients is not a problem in the analogue
world, as the analogue clipping level would
typically be well in excess of +20dBu. But
for the digital conversion the EBU wanted
to keep the digital signal as high as they
practically could in the digital domain, while
still retaining a sensible headroom margin
to accommodate any high transients. It
transpired that an extra 9dB of headroom
above the +9dBu programme peak level
proved sufficient, so the EBU standard
alignment became 0dBu = -18dBFS.
If you so desired, you could achieve
that in your system simply by deciding
to operate your console with the slightly
lower reference level of 0dBu instead
of +4dBu. Obviously, this would also
slightly increase the analogue system’s
overall headroom and slightly decrease
the signal-noise ratio, but the restricted
dynamic range of broadcast audio means
both are irrelevant variations, in practice.
You would have either to ignore the
console’s VU meters or to recalibrate them
so that 0VU = 0dBu. However, if you use
a 0dBu reference tone and align the digital
converters to read -18dBFS, everything
will work just like any other professional
broadcast installation.
Going back to that quasi-PPM business,
though, it’s important to bear in mind that
when broadcasters specify digital peaks
to -9dBFS, that relates only to steady-state
tones, not to programme audio — despite
what some quality-control departments
sometimes try to claim! That -9dBFS figure
comes from some simplistic maths without
context. With the -18dBFS alignment, the
analogue practice of keeping programme
peaks below +9dBu (+8dBu in the UK)
would appear to translate to digital peaks of
9dB above -18, which is -9dBFS. However,
the ‘programme peaks to +9dBu’ mantra
related to the signal being viewed on
a traditional PPM and, as we know, PPMs
deliberately under-read transients. The EBU
digital alignment of -18dBFS specifically
builds in headroom to cope with transient
peak signals, and a digital system can
obviously accommodate signals all the way
up to 0dBFS without issue. The current
loudness normalisation requirements in
broadcasting state that true peaks should
be no higher than -1dBTP, and that small
headroom margin is simply to protect
against potential measurement error in
true-peak metering systems.
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 ARRANGING

Photo: Timor Kodal

know the workflow of Ableton tools to support beginner and but in combination ultimately
JULIAN SCHMAUCH
Live or Logic Pro, I try to get intermediate music producers. create too much chaos and

I
’ve been creating, teaching
and writing about electronic
music production for
almost a decade, and I’m
still astounded by how little
arranging plays a role in music
education for most beginners.
Sure, there is the ‘arrangeur’
within the world of orchestras
and film scoring, but in the
world of electronic music
production there is tutorial
after tutorial for the best EQ
settings and most effective
drum bus compression, but few
go into the core of what music
production, in my opinion, is
actually all about: arranging.
Whenever I approach
the production of a song,
whenever I have gotten
a class’s attention after
a couple of days of getting to
them to focus on arranging.
What sounds play what
melody or rhythm, at what part
of the song, to support the
overall emotional message?
Do kick and bass sounds
or guitar and synth sounds
complement each other? Are
there percussion elements
that support or disturb the
vibe in terms of timbre and
rhythm? If there is a vocal,
does everything else step
out of the way enough? Does
the performance of the vocal
match the emotion? How can
pre-recorded elements such
as loops or programmed MIDI
instruments be manipulated
to ‘perform’ alongside like the
recorded instruments?
In contemporary DAWs,
there is a growing number of
MIDI tools help with lack of
knowledge in music theory.
Tuning algorithms help
correct spontaneous vocal
performances. Workflows such
as comping help creating the
perfect performance. Yet there
are next to no tools for helping
with the arrangement.
For example, voice
leading is one of the most
fundamental elements of
arranging instruments. But
DAW tracks do not ‘talk’ to
each other in such a way that
the software could suggest
how to create harmonies that
support the existing elements.
The same goes for rhythm.
The number of times I have
advised beginner students
that individual rhythms of their
tracks may seem exciting,
instability within their track, is
staggeringly high. Also, I tell
students early on that if your
arrangements are too dense,
and have too many competing
elements, it cannot be fixed
by a plug-in. How, with all
the tools available to us, are
there no tools in contemporary
DAWs that help with the
arrangement?
It could simply be that
the intricacies of a song’s
arrangement, of the
relationships between
frequency, rhythm, harmony
and structure, are a
fundamentally human element
of any music production that
it is impossible for software to
emulate. All the more reason
to do it ourselves, and to learn
it as early as possible.

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