THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZINE 1985 — 2023

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

“I got my spark for recording back”

Neumann KH 150
New heights in nearfield monitoring

SSL 12 & Avid MBox Studio WIN

Desktop interfaces on test

How to start a mix WOR TH €2500

...by the pros who do it every day!
www.soundonsound.com

REVIEWS: SONY / IK / ROLAND / SPITFIRE / ELECTRO-VOICE / WAVES / IZOTOPE / SONICWARE / INGRAM

TECHNIQUE: SYNC YOUR DAWLESS SETUP / PROBLEM SOLVER GADGETS JJanuary 2023 £6.99
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 LE ADER

TEAM TALK
When Stormzy agreed to be interviewed for this
month’s cover feature, he set one condition. We
would, he insisted, be interviewing him along
with his entire production crew — and they
would be equally prominent in the cover photo.
It’s refreshing to see such a big-name artist
be so determined to share the limelight, and it
made me think about how music production has
evolved. In the ’60s and ’70s, engineers were
usually on the payroll of a studio, where they
worked as a team. Their role in the production
process was invaluable, but it was universally
acknowledged to be a supporting one.
Fast-forward 50 years, and almost everyone
who makes a living in studios does so as
a freelancer. Building a brand has become as
important as knowing your way around an EQ,
and many successful producers and engineers
are public figures with carefully curated public
images. And it’s not just the production side of
things that has become more individualistic. At
the time of writing, the entire UK Top 50 singles
chart contains precisely one entry credited to
a band: OneRepublic’s ‘I Ain’t Worried’, at
number 29.
The music industry has obviously reached
one of those points on its never-ending
cycle where bands are out, while solo artists,
DJs, rappers and electronic producers are
in. Consequently, there’s less of a need for
old-school studios with their teams of highly
trained staff. But will the pendulum ever swing
Sam Inglis
Editor In Chief
back? Could it be that this phase has been
driven less by fashion and more by the fact that
teamwork just doesn’t pay well enough?
There’s no money to be made playing original
music live, and label advances are not what they “On a creative,
used to be. Streaming royalties are making label a technical and
executives rich, but that money is not filtering
down to ordinary musicians. Being in a band a human level,
means slicing the pie four or five ways, and there’s simply no
when the pie is a Mr Kipling affair an inch across,
it’s just not economic to do it. The same goes for substitute for
producers and engineers; when the majority of working with other
artists are solo acts who can record themselves
in hotel rooms, it’s hard enough to keep the people.”
wolf from the door without thinking about hiring
assistants or going into partnerships.
The democratisation of music technology
over the last 20 or 30 years has had many
positive effects, but I don’t think this atomisation
is one of them. On a creative, a technical and
a human level, there’s simply no substitute
for working with other people, whether that
means starting a band or recruiting a dedicated
production crew. I salute Stormzy for standing
up for that ideal, and I hope his success helps to
highlight the value of teamwork.  

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 124 SPIKE EDNEY

IN THIS ISSUE www.soundonsound.com

January 2023 / issue 3 / volume 38

FEATURES
WIN
26 How To Start A Mix JZ MICROPHONES
S
Just how do professional mix engineers tackle the earliest stages WORTH €2500
of their projects, and can the rest of us learn anything from
their approach?

46 Inside Track: Leandro ‘Dro’ Hidalgo

With hit albums for Dave, Wizkid and now Stormzy,
Brazilian-American mixer Leandro Hidalgo is at the top of his game.

62 Meet The Problem Solvers!

These gadgets might not be the most glamorous devices, but at
some point in your recording adventures you’ll probably find you PAGE 70
need each and every one.

74 Talkback
Billy Nomates, aka Tor Maries, tells of her love of Bowie’s Low 117 Modular
and explains why a relaxed studio is a productive studio. We talk to Future Sound System’s Finlay Shakespeare and
round up some of the best Eurorack cases.
86 How I Got That Sound
Goldfinger frontman and pop-punk producer John Feldmann on 124 Spike Edney:
his favourite vocal sound. Queen’s Keyboard Player
For Spike Edney, touring with one of the biggest bands in the
88 Sync Outside The Box world has meant keeping up with the very latest in stage
Going DAWless presents many challenges — not least keeping keyboard technology.
all your gear in sync...
144 Q&A
102 Stormzy: Making This Is
R

Your studio and recording questions answered.

E

What I Mean
O V

Britain’s most high-profile rapper has made his most ambitious 162 Why I Love... Compact Discs
album yet — with a little help from his friends. David Baer explains his abiding love of CDs.
C
 12 ARTURIA MINILAB 3

ON TEST
8 Sony C80 78 Spitfire Audio Abbey Road: 134 iZotope RX10
Capacitor Microphone Low Percussion Audio Repair Software
Sample Library
12 Arturia MiniLab 3 138 NovoNotes Ambience Enhancer
Controller Keyboard Multi-channel Ambience Plug-in
84 Waves Harmony
16 Accentize DeRoom Pro Multi-mode Harmony-generation 139 TC Electronic Bucket
Plug-in
Reverb Removal Plug-in Brigade Delay
Analogue Delay Pedal
18 Cong Burn Strokes 92 Roland Fantom-0
MIDI Sequencer Plug-in
Workstation Synthesizer 139 Warm Audio Centavo
Overdrive pedal
22 Avid MBox Studio 98 Interphase Audio Modules
USB Audio & MIDI Interface 500-series EQs 140 Warm Audio Warmdrive
Overdrive Pedal
36 Dawesome Novum 108 Massenburg Design Works 142 Sample Libraries
Software Synthesizer MDWDRC2-Native Heavyocity Vocalise 3
Dynamics Plug-in Vegas Audio Drumslot
40 Neumann KH 150 Fluffy Audio Timedrops
Active Monitors 110 IK Multimedia iLoud Sampleson Alienize
52 IK Multimedia AmpliTube Tonex Precision MTM
Amp Profiling Software Active Monitors
WORKSHOPS
58 SSL 12 116 Eventide Misha
USB Audio & MIDI Interface Eurorack Module 146 Reaper
150 Studio One
72 Ingram Engineering EQ52 120 Sonicware Liven Bass & Beats 152 Reason
500-series EQ Groovebox
154 Pro Tools
76 Rockruepel Sidechain.one 130 Electro-Voice Everse 8 156 Cubase
VCA Compressor Portable PA Speaker 158 Logic
 ON TE ST
Sony C80
Capacitor Microphone
Sony’s latest microphone makes their
C-series magic more affordable than ever.
SAM INGLIS
ony have made great studio
S microphones since the ’50s,
but in the West at least, they’ve
sometimes been better at designing them
than at marketing them. Models such as
the C48 and C38 haven’t always been
easy to discover or buy, and even the
flagship C800G valve mic has been an
elusive beast in recent years.
In 2018, the Japanese electronics
giant tried to change this, rebooting their
studio mic range with three entirely new
models. The ECM-100U and ECM-100N
are small-diaphragm ‘pencil’ mics, whilst
the C100 is a more unusual design that
features a second capsule mounted
directly above the first, in order to
capture ultrasonic frequencies. Intended
8 January 2023 / www.soundonsound.com
as a studio all-rounder, the C100 offers
switchable polar patterns, plus a pad and
a low-pass filter. I reviewed it in February
2018 (www.soundonsound.com/reviews/
sony-c100) and came away impressed.
However, although the C100 is less
costly than the U87s of this world, it’s still
quite a bit more expensive than typical
project-studio mics, and probably out of
the reach of some potential customers.
Enter the new C80: a simplified,
cardioid-only version of the C100
without the ultrasonic capsule, at a much
lower price.
Little Brother
With its businesslike, black shell, the
new mic looks very similar to the C100,
except that it’s about half an inch shorter.
Build quality appears equally good,
and the C80 comes in a robust padded
case. It comes with a fitted shockmount,
which seems reasonably heavy-duty
even though it’s made almost entirely of
plastic, but unfortunately there’s no room
for this in the case. The C80 retains the
C100’s low-cut filter and 10dB pad.
One surprising feature of the C100
was its self-noise, which is specified
at 19dBA in cardioid mode and more
in the other polar patterns. This is
an unusually high measurement for
a modern solid-state mic, but self-noise
wasn’t at all problematic in use, which
made me wonder whether the second
capsule was somehow being factored
into the noise calculation, even though
A-weighting should see ultrasonic noise
largely discounted. The C80’s specs lend
some weight to this idea, as the quoted
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 ON TE ST
SONY C80
The capsule measures around 25mm across in
total, making it somewhat smaller than that of most
‘large-diaphragm’ microphones.
self-noise is only 12.5dBA. In other
respects, its performance is very close
to that of the C100 in cardioid, with the
same 90Ω impedance and a fairly high
sensitivity of -30dB referenced to 1V/Pa,
which is equivalent to 31.6mV/Pa. The
C80 will accept up to 138dB SPL before
1% distortion is reached.
One of the main factors that affects
self-noise is diaphragm size, and this
is probably why the C80 doesn’t quite
achieve the same inky-black quietness
as, say, the Neumann TLM103 or the
Rode NT1. The transducer used here
and as the main element in the C100 is
a conventional dual-sided, externally
polarised, circular capacitor capsule
which is described as being 25mm in
diameter — almost exactly one inch.
However, this measurement seems to
refer to the capsule as a whole rather
than just the active membrane area,
and it’s clearly smaller than a typical
‘one-inch’ capsule. A better comparison
might be the medium-sized capsules
used in mics like the Neumann TLM107.
High Up
You’d expect the C80 to sound very
similar to the C100 in cardioid mode,
and although I didn’t have a C100
around for side-by-side trials, it certainly
matched my memory of that mic. No
frequency response graph is available,
but losing the ultrasonic capsule
doesn’t seem to have affected the
10 January 2023 / www.soundonsound.com
The C80 ships with a padded case and plastic shockmount.
C80’s ability to capture high-frequency
content in the audio range. I tested it
alongside various rivals including an
Austrian Audio OC16, which is itself not
shy in the highs. In every case, there
was measurably more energy above
10kHz in the signal captured by the
C80. This was confirmed at Xaudia’s
anechoic chamber, where we compared
it with older Sony mics including the
C48, C36P and C450, which likewise
has a medium-sized capsule. None of
the older models showed the same
emphasis in the top octave. (This
measurement also showed that the C80
high-pass filter is quite gentle, being
only a few dB down at 50Hz.)
Yet, just like the C100, the C80
pulls off the neat trick of not sounding
unbalanced or subjectively bright.
This, I think, is because its response
is impressively flat and pure through
the midrange, especially in the
crucial presence region. As it’s a very
clean-sounding mic, with no transformer
or other potential source of saturation,
the additional energy at the very top
is heard more as extra detail and not
so much as a change to the timbre
of vocals or acoustic instruments. On
male vocal, for example, the OC16
came across as the brighter of the two,
although its peak is an octave lower,
and the C80 actually seems to throw
more emphasis on the 1-2 kHz region.
I really liked this character on my own
voice, and if you do feel the need for
more of a push in the treble, it will
respond well to EQ. The flip side of
this is that you might need to take care
recording anything that already has a lot
of energy above 10kHz: cymbals, for
example, and some female vocals.
I sometimes wonder how often
people really use multi-pattern mics in
patterns other than cardioid. Big studios
will have dedicated omni and figure-8
models available when needed, and in
small studios, opening up the pattern
risks capturing more of an imperfect
room. When your sonic diet mainly
comprises close-miked vocals, acoustic
instruments and guitar amps, the C100’s
extra polar patterns and ability to capture
frequencies above 30kHz perhaps aren’t
very relevant. For those who only ever
record in cardioid, at standard sample
rates, the C80 sounds just as good, has
better specifications in some respects,
and is less than half the price. This is
a very good mic for the money!
summary
The cardioid-only C80 offers the same classy
sound as the much more expensive C100 in
a very affordable package.
£ £514.72 including VAT.
T Karno Sound +44 (0)20 8064 0340
E contact@karnosound.com
W www.karnosound.com
W https://pro.sony/audio
 ON TE ST
Arturia
MiniLab 3
Controller Keyboard
SIMON SHERBOURNE
rturia’s Lab range marries hardware
A keyboard controllers with the
company’s extensive soft-synth
portfolio. Their Analog Lab V plug-in deploys
the engines of the mighty V-Collection as
a unified sound module that integrates with
the keyboards. MiniLab is a portable and
significantly more affordable alternative to
the larger KeyLab master keyboards. As well
as the compact controller, MiniLab 3 comes
with a pared-down version of the sound
suite, dubbed Analog Lab Intro.
The MiniLab 3 controller maintains
roughly the same shape as the MkII, with
its deep panel giving it the same footprint
as a 15-inch laptop. I had the white model
and thought it looked like a cute tiny
Mellotron that had sprouted controls, or
the synth edition of Is it Cake? Don’t get
me wrong: I think it’s a lovely form factor.
Previous MiniLabs had 16 encoders and
eight pads with a two-octave mini-keyboard.
For this iteration Arturia’s designers have
replaced eight of the encoders with four
sliders, perhaps thinking to trade off the
12 January 2023 / www.soundonsound.com
Anyone fancy a versatile controller keyboard with
the tip of the Arturia software iceberg included free?
total number of controls for some versatility,
and tick off more boxes for people
shopping around.
Encoder count aside, the 3 has levelled
up as a controller, gaining stand-alone
capabilities. A DIN MIDI out port has been
added to the rear panel alongside the
USB-C connection, and there’s now an
onboard arpeggiator. In the last couple of
years a MIDI out has become a significant
differentiator in the small controller
market: it’s great to have the option to
work host-free and plug straight into
a hardware module.
In The (Recyclable) Box
I’m not normally one for an unboxing
paragraph, but special mention should
be made of the MiniLab’s eco credentials.
Not only is the packaging 100-percent
recyclable, the unit itself is constructed
from 50-percent recycled plastic. Thinking
outside the box (sorry), Arturia are providing
the MiniLab 3 with a five-year warranty,
promoting longer product lifecycles as part
of the bigger sustainability picture. Bon
travail Arturia!
Included in the package is a software
bundle that will get anyone new to
computer-based music up and running.
As well as Analog Lab Intro, you get two
piano plug-ins: UVI’s Model D and Native
Instruments’ The Gentleman. Then there’s
the ubiquitous but perfectly capable Ableton
Live Lite.
A smart move is the inclusion of
a subscription to Melodics, an online
interactive music learning service. There’s
some really cool apps and sites springing up
like this, which make learning an instrument
like a game. This one looks particularly
intriguing, as in addition to follow-along
keyboard lessons it covers drum pads and
production in general.
This Is My Laboratoire!
MiniLab is a flexible, general-purpose MIDI
controller, but let’s start with what sets
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 ON TE ST
ARTURIA MINIL AB 3
it apart: Analog Lab. The idea is simple:
Analog Lab is an app and plug-in that hosts
Arturia’s synth engines, offering a single
point of contact with an overarching preset
library, pre-mapped macro controls and
a live show manager.
Although I’m an avid V-Collection user,
it’s been about three years since I dipped
into Analog Lab, as day-to-day I use Native
Instruments’ Komplete Kontrol, which
Arturia also support as a platform for their
instruments. (Incidentally if you’re weighing
up both options, they have different
approaches: Analog Lab gives you access
to sounds from all premium synth engines,
but only full panel control of synths you own.
With an equivalent-level Komplete Kontrol
option, you get full access to the synths but
there are fewer of them).
The app/plug-in has had a facelift and
looks colourful and friendly. The home
screen is now dominated by icons rather
than text, presenting the content in different
ways. You can browse by Types (Bass,
Keys, etc), Instruments (the V synths) or
patch designers. Clicking on a Type pops
up a bubble with an audio example and
three preset examples. I’m not convinced of
the utility of this, but double-clicking takes
you straight to the patches in the more
familiar Explore view. This scrollable list
has a search field and tag lists. Tags stack
up in the search bar as you click them,
progressively narrowing down your search.
Intro has around 500 patches (as
opposed to 9000 in the full version of
Analog Lab) so you don’t need to make
much use of filters. This equates to about
20 patches per instrument, which doesn’t
provide a lot of room to access the range
of sounds these iconic synths can reach.
I managed to boost this to 840 by installing
most of the free sound packs in the integral
Store. There are many paid packs available,
or you could upgrade to the full version of
Analog Lab, but the most compelling way to
start unlocking the potential is to purchase
some of the actual instrument plug-ins. Any
instrument you own can be accessed with
its full user interface from inside Analog Lab.
Macro Conditions
The bottom of Analog Lab’s window
presents macro controls that match
your hardware. The first four are always
high-level sound shapers: the actual
parameters these control varies by patch,
but they are always titled Brightness,
Timbre, Time and Movement. The next four
encoders control levels or mix amounts for
insert effects and returns.
14 January 2023 / www.soundonsound.com
That leaves the sliders. For a reason
I can’t fathom, Arturia have mapped these to
Analog Lab’s master Bass, Mid, Treble and
Level in all the factory patches. Why, when
you have the most incredible, responsive
software synths at your fingertips, would
you use half your controller to adjust EQ?
You are left with just four macro controls per
patch that actually control the instrument. It’s
consistent but restrictive, considering that
the MkI and II MiniLabs made full use of their
16 encoders to control a range of relevant
controls on the underlying instruments. It is
possible to reassign controls to Analog Lab
effects and master controls, and whichever
few macros get exposed to Intro by the
instrument. But you can only do this on
a per-patch basis, not a whole synth.
There is good news too. First, the
controls are rotary encoders so there’s no
parameter jumping when you take control.
Second, preset browsing from the hardware
has been improved since I last used a Lab
controller. The MkIII has a dedicated push
encoder with indented action. With Shift
this lets you step through sound categories;
otherwise it scrolls through lists, with a push
registering a selection.
Multis
Analog Lab can host two synth patches
side by side. You can create splits and
layers, transpose them relative to one
another, and choose where standard
performance controllers like pitch wheel are
routed. The encoder and sliders point to
one side at a time, toggled in the software
or via a Shift+Pad combo. Both channels
share two insert effects, and delay and
reverb sends. The effects section is laid out
like a little stompbox rig, with a selection of
modules to choose from in the four slots.
Further customisation and performance
configuration is available in the form of
multiple Favourites lists, and a Playlist
feature designed for live shows.
Unfortunately, you’ll need the full version of
Analog Lab to configure Playlists. Teasingly,
Intro can make a Playlist but can’t add
anything to it!
In Control
The MiniLab 3 has DAW-control
functionality for leading DAWs except for
Pro Tools. Scripts are provided, but I didn’t
The rear panel features a full-size MIDI out
socket, a control pedal input and a USB port.
have to install anything for the current
versions of Live and Logic. However, the
transport buttons also played notes in both
until I manually disabled the control port.
Shift+Pad 3 on the controller steps through
its various modes: Analog Lab, DAW and
any User Sets you’ve made.
In DAW mode the pads become
transport buttons, and the main encoder
selects tracks, or scrubs the timeline with
Shift. In Live mode you can dial through
scenes and launch clips from the pads.
The faders adjust channel controls on the
selected track, and the encoders map
to devices. Although there are discrete
pages, you can access DAW transport
controls using Shift while in Analog Lab
mode. It would be cool if this was extended
to the encoders and sliders as well.
We’ve focused a lot on specialised
functionality, but the MiniLab 3 has
plenty going on to make it a great
general-purpose portable MIDI keyboard.
It’s customisable using Arturia’s familiar
MIDI Control Center app, where you can
define CC or NRPN functions for all the
physical controls and pads. You can store
five pages in addition to the Analog Lab
and DAW modes. This is a great feature
to have, although I’d have preferred if
they were accessed by a different pad,
instead of adding to the list to tap through
when switching between Analog Lab and
DAW modes.
Conclusion
The MiniLab 3 carves a niche for itself
in the sub £100 controller space: it’s
portable without feeling cramped and
with a decent spread of controls including
endless encoders. Stand-alone MIDI and
the arp greatly extend its utility, as does its
integration with Analog Lab Intro.
summary
Less control of the V instruments than the
larger Labs, but a compelling starter package
and portable controller.
£ £95 including VAT.
T Source +44 (0)20 8962 5080
W www.sourcedistribution.co.uk
W www.arturia.com
 ON TE ST
required, and it can process
mono or stereo sources at
44.1, 48, 88.2, 96 or 192 kHz
sample rates.
Overview
The default GUI is pleasingly
clean and uncluttered, with
a representation of the
reverb removal, along with
a handful of main controls:
you can set the sensitivity
of the detection, the
amount of reverb reduction
(a dynamic process whose
activity is reflected in real
time on the curve above),
and fine-tune the result with
attack and release controls,
to make the process a touch
smoother at the cost of
letting a shade more reverb
through. At the top, you can
choose to link the L and
R channel processing (or
Accentize not), hit a Room Only button
to audition what’s being
removed, bypass the plug-in,
or enter SpectralFocus mode,
DeRoom Pro
Reverb Removal Plug-in
of which more below.
For the analyser, you
have a range of options. You
can display a waveform or
spectral view, choose to hide
the visualiser (reducing the
Removing ‘baked in’ room sound is no temptation to judge settings
longer the stuff of dreams, and this clever by eye rather than ear), and
display just the output signal,
plug-in does it better than most. or both the input and output
signals in adjacent ‘lanes’.
M AT T H O U G H TO N
Nearby is an option to scale
’ve been doing a lot of podcast production recently, dealing the GUI: 80, 100 or 150 %, the
I with recordings of a few people captured remotely using
Zencastr. While I’ve been able to offer a little advice on
the participants’ recording setups, much remains beyond my
last almost filling the height
of my MacBook Pro Retina’s
screen when in SpectralFocus
control and the recordings inevitably suffer from a number mode. Speaking of which...
of problems, not least of which is the undesirable sound of In SpectralFocus mode
the domestic rooms and offices used for the recordings. you’re given three bands to
There are various options open to us now to reduce ‘room play with and have plenty
sound’ in post-production. For example, I’ve used iZotope RX of freedom in setting the
(version 10 of which is reviewed in this issue) and Accusonus crossover points: the first can
ERA 6 (still working, but sadly no longer available to buy) to be anywhere from 100Hz to
reasonably good effect, and Descript’s Studio Sound feature 2kHz, while the second can
can be an impressive ‘quick fix’ too. But, dissatisfied with the be set from 300Hz to 6kHz.
side-effects, particularly when more assertive room removal is So you have the ability to
required, I decided that I should audition some other options — really zone in on particularly
and my pick of the pack so far is Accentize DeRoom Pro. problematic aspects of
Authorised via iLok, DeRoom Pro is a real-time plug-in a room that might require
underpinned by machine-learning techniques, and it can be more aggressive treatment.
authorised to a dongle or to up to two separate machines. Each band is treated to its
Mac and Windows are supported, an AAX, VST3 or AU host is own Sensitivity, Reduction,
16 January 2023 / www.soundonsound.com
Attack and Release knobs,
and the equivalent global
controls from the default
view are retained.
It’s all very easy to use
and to see what’s going
on. Really intuitive. More
importantly, the results are
superb — it’s smoother
sounding than any other
real-time reverb removal
system I’ve tried, and while
there will always be some
side-effects when you ask
too much this sort of tool,
those encountered here
are mercifully minimal and
their onset subtle. I’ve
found I that can push the
processing much further
than the equivalent RX
or ERA processes before
encountering distracting
‘warbles’. I’ve also found
that I can sometimes use
two instances in series to
good effect where more
assertive processing is
needed. And while they
might not know precisely
what’s changed, clients
have certainly noticed the
improved results!
DeRoom Pro is intended
to treat dialogue and, it
being based on machine
learning, I wasn’t surprised
to discover that it wasn’t
hugely effective on other
sources, such as guitars,
drums or piano (in theory,
the same tech could be
‘trained’ to recognise them,
but it hasn’t been — yet).
Still, I like to experiment,
and found a cool alternative
use for it in Listen mode to
fake a drum ‘crush’ mic!
I thoroughly recommend
DeRoom Pro, then, but as
there’s a free fully functional
trial, you needn’t take my
word for it.  
summary
Stunningly good use of
machine learning to suppress
unwanted ‘room tone’ in real time.
£ £215 including VAT.
W www.accentize.com
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 ON TE ST

immediately open up intriguing

and addictive possibilities
for polyrhythms. The ability
to mute channels or instantly
‘invert’ patterns — turning
programmed steps off and
empty steps on — adds a lot of
flexibility for experimentation.
Channels 5-8 occupy
a separate smaller area of
their own, and these really are
esoteric! They’re algorithmic
note generators driven by
interactions of events on
channels 1-4, and sometimes
the accent lane. As such they
are ‘programmable’, but in
a way that’s really not as most
of us know it (captain). More
about this, if you’re interested,
in the ‘Logic’ box.
Next up (or rather, down)
is a section labelled Shares,
which, to ruthlessly summarise,
reduces event density on the
note channels with varying
degrees of randomness, for
more or less unpredictable,
ongoing variation of

Cong Burn Strokes

MIDI Sequencer Plug-in ‘channels’ that send out MIDI notes, and
12 modulation sources alongside that
Strokes brings unconventional generate MIDI CC messages.
Channels 1-4, along with an adjacent
Eurorack-style sequencing to Accent lane, are relatively speaking the
your DAW. most conventional bit of the plug-in.
The programming lanes often look and
ROBIN BIGWOOD
behave a bit like Roland drum machines
trokes is an unusual beast: a dedicated going back decades. However, the
programmed patterns.
With these sections alone
a lot of sequencing action
is possible. But there’s
a whole other world nestling
just alongside...
Modulation
Matrix and Weights are
Strokes’ modulation
generators: between them they
output 12 streams of MIDI CC
data, which any good DAW can

S sequencer plug-in that runs inside your DAW

sequencer, generating MIDI note and other
messages to drive other instruments of your choice,
number of steps (from 1-32) is adjustable
on a lane by lane basis, and they can run
backwards and in alternating directions
whether software or hardware. It can do simple too. More importantly, there’s another
pattern-based programming, but also has facilities for mode on offer, ‘Euclid’, which is distinctly
generating patterns that you might not naturally come un-Roland-like. This programs steps
up with yourself, and can be a platform for exploring for you according to four dedicated
polyrhythms and randomness in repetition generally. parameters. Euclidean sequencing
Strokes started life as a Max For Live device but might sound esoteric, but in practice it’s
now it’s available as a VST3 and Audio Unit plug-in about dividing the chosen number of
for Windows 10/11 and Mac OS 10.11 or later. I tested triggers as evenly as possible across the
version 3.4 on the Mac, and as I write an AUv3 plug-in steps available. This can lead to entirely
has just been released which will work on the iPad in evenly distributed patterns, but also
hosts like AUM. dozens of others that are more complex
and syncopated. At the same time the
Notes combination of variable lane lengths,
Strokes’ resizeable GUI can be bewildering at first time divisions, and the other Euclidean
sight, but with only a little familiarity it proves very clear parameters (including loop length, which
and straightforward. Essentially there are eight parallel can be longer or shorter than lane length)
distribute elsewhere.
Weights is the simpler
to understand: it’s a bank
of four separate envelope
followers/generators that
are triggered by events on
the note channels. All four
channels at once, if you like.
The result is a variety of
adjustable modulation shapes,
from simple pulses to complex,
vacillating mayhem.
Matrix, meanwhile, is more
abstract: a sort of ‘chess game
meets step sequencer’ that
takes inspiration from Make
Noise’s Eurorack module,
René. Modulation values are
generated as a position

18 January 2023 / www.soundonsound.com

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 ON TE ST
CONG BURN STROKES
Strokes’ single configuration screen is for
assigning MIDI channels to note lanes, and MIDI CC
numbers to modulation outputs. This lets it work in
multi-device, multi-channel environments, and with
gear that has fixed MIDI CC assignments.
indicator moves across a 4x4 grid of value
knobs, in response to note events. This
happens on eight layers simultaneously,
according to different rules: some move
vertically, others horizontally, a couple
crab-like, and not necessarily across all
16 positions. It’s arguably bonkers... but
the beauty of the Matrix lies in the way it
forces you to think about step sequencing
in a totally different, two-dimensional way.
The remaining sections are more
utilitarian in purpose. Scope is a handy
visualiser of the modulation values being
generated by Matrix and Weights. And
finally Patterns switches between five
plug-in-wide pattern snapshots and,
independently, seven alternative MIDI
note layouts for channels 1-8. These
numbers are not randomly chosen but
correspond to the five black and seven
white keys in an octave, with which (via
MIDI) Strokes can itself be ‘played’ in
real time.
Genius
Strokes is one of those software tools that
could all too easily seem intimidating and
conceptually slippery. In fact, it’s pretty
DAW Games
Because Strokes sits outside the box of what
plug-ins normally do, DAW compatibility is
definitely a ‘thing’, and there are variations in
the user experience from one DAW to another.
There’s some specific documentation and
support on the Cong Burn website for setting
up in Ableton Live, Bitwig, Cubase, FL Studio,
Logic, and the virtual modular environment VCV
Rack. I tested in Studio One 6 without difficulty,
and other DAWs will doubtless work too, as
20 January 2023 / www.soundonsound.com
straightforward to
grasp, particularly
if you have any
prior knowledge
of typical Eurorack
sequencer workflows
or Elektron devices.
The interface is clear
and responsive,
and with almost no
additional panels or
views, no right-click
functionality, and
nothing at all related
to loading from or
saving to disk, it’s
a quasi-analogue experience. Hit Play
in your DAW and Strokes starts doing its
thing. Press Stop and it stops too. The
fact that there are only four conventional
note lanes, constrained to single
pitches, feels old-school but refreshingly
uncomplicated too.
For me, the musical potential sang
out from word go; at least with various
concoctions of electronica, EDM, glitch
and minimal-leaning genres in mind. It’s
a wildly different experience to recording
to your DAW or clicking data into note
and automation lanes: in many ways more
liberating and experimental. As I used
Strokes to drive various drum samples,
synths and sampled instruments,
I continually ended up with patterns,
ostinatos and riffs that wouldn’t have
happened with more conventional ways
of working. Crucially, many were worth
saving and developing. Polyrhythmic
interactions and randomness is a big
factor here, and as time went on I found
more and more use for the modulation
outputs too. To actually save what
Strokes does, by the way, you literally
‘record in’ its output to the other MIDI
or instrument tracks it’s driving, in real
time. Subsequently sifting through and
selecting material can in itself spark
different and productive ways of working.
long as they support the flow of virtual MIDI
from plug-ins to other tracks. On the Mac, the
AU version is developed exclusively for Logic
users, and everyone else will need to use VST3.
Ironically, given Strokes’ heritage, it’s Ableton
Live that provides the most non-standard
experience: a companion Max For Live device is
still required, and there’s an unusual (but entirely
effective) scheme for getting MIDI messages out
of the plug-in and routed somewhere else.
Logic
Channels 5-8 are out there, no question.
They produce notes — or not — depending
on interactions between other note lanes
and the accent channel, according to a
range of eight preconfigured logic functions
that are arbitrarily numbered. It’s arguable
whether they are ‘programmable’ in any
meaningful sense, but they’re certainly a
vast mine of happy rhythmic accidents.
What really helps here is the bank of
flashing indicators that show the note
activity associated with each logic mode,
for all four algorithmic channels, all the
time. So you don’t have to have a clue how
a pattern is produced, but if you spot one
that is flashing away with the sort of density
of events you might want for a hi-hat part,
say, then it’s just a case of choosing the
corresponding mode and configuring the
appropriate MIDI note for the channel.
Any downsides? Well, there’s no way
of programming velocity in note lanes, let
alone flams or ratchets. Meanwhile, I’d
gladly accept some more fully featured
ways of copying and clearing patterns.
Selecting an empty and unused pattern
for the first time will copy into it anything
you currently have programmed, which is
something, but it’s a blunt tool. Also, for
all its visual clarity, Strokes’ interface is
a bit light on information, in places. The
algorithmic channel modes and Matrix
layer behaviour are unguessable, and
only lightly documented in the manual.
Some Weights knobs are unlabelled
too. I put some of this to John Howes,
the developer of Strokes, and he
enthusiastically proposed a few really
elegant solutions. As I submitted this
review there was already a new beta
version that tackled some of the issues,
and had other nice features.
So the future of Strokes looks
bright, but even in its current form it’s
an inspiring and fascinating tool: an in-
DAW playground for modular-inspired,
pattern-based experimentation. At
its very modest asking price it’s an
absolute bargain.
summary
A dedicated pattern sequencer for your
DAW that opens up all sorts of polyrhythmic,
pseudo-random, non-linear possibilities. Huge
fun, and musically potent too.
£ Plug-in £35, iOS £19.99.
Prices include VAT.
W www.congburn.co.uk
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 ON TE ST
Avid MBox Studio USB Audio & MIDI Interface
The MBox is back and it’s much better and certainly
much bigger than it was before.
SAM INGLIS
he original MBox, launched more
T than 20 years ago, was a hugely
influential product. It made Pro Tools
an attractive option for home studios
and schools as well as professional
environments, and established a design
template that is still followed even now. It
was followed up by another two generations
of increasingly powerful USB and FireWire
MBoxes, before the ability to use third-party
interfaces in Pro Tools apparently signalled
the end of the road.
For quite a few years after that, the
native version of Pro Tools was positioned
as a software-only product, and even when
Avid introduced new HDX units such as
the MTRX, they were essentially rebadged
versions of other manufacturers’ products.
So it came as a very pleasant surprise when
Pro Tools | Carbon appeared late in 2020.
22 January 2023 / www.soundonsound.com
Not only were Avid back in the business
of making hardware interfaces: they were
innovating once again, with a new Hybrid
Engine that exploited the AAX DSP plug-in
format to effectively banish the problem of
latency during tracking. And now they’ve
followed it up by reviving the MBox.
Hot Desking
Like most earlier MBoxes, the new MBox
Studio is a desktop interface that connects
to the host computer over USB, but that’s
pretty much where the similarities end. Its
physical design is very different from that
of any other audio interface I know, and the
MBox Studio is a seriously chunky piece of
kit. Its wedge-shaped case measures about
29 x 22cm, stands more than 10cm proud of
the desktop, and is correspondingly heavy.
This must be a conscious decision on Avid’s
part, because there are rival interfaces
that offer lots of connectivity in a smaller
Avid MBox Studio
£899
PROS
• Refreshingly original design with
many unique features.
• Caters especially well to guitarists.
• Great-sounding and unusually
versatile preamps with effective
variable impedance options.
• Built-in tuner is a real boon.
• Large controls and clear, bright
metering.
• It won’t slide off your desk...
CONS
• Some teething issues with the v1.0.0
firmware and the initial version of
MBox Control.
• Low-latency performance isn’t
amazing.
• Less portable than other desktop
interfaces.
• You’d hope that a box this size would
have room for an internal PSU.
SUMMARY
It’s always good to see fresh thinking
at work in a world full of ‘me too’
designs, and the new MBox displays
innovation in abundance. Recording
guitarists in particular should take
a close look at what’s on offer!
package, Antelope’s Zen Tour Synergy Core
being one obvious example.
The blackness of the case is offset by
an attractive lightshow from the meters and
controls, and the enormous pair of rotary
controllers in the lower part of the top panel
make it look more like a DJ controller than
an audio interface. As on the Carbon, the
buttons have internal lights that change
colour to indicate different statuses, while
the rotaries are backlit in purple. It all feels
as heavy-duty as it looks, to the point
where the push action on the main rotaries
requires quite a bit of force.
Connection to the computer is made
through a Type-C USB port, and
the supplied USB cable
has a neat adaptor to fit
Type-A or C ports at the
other end. The MBox
Studio requires mains
power, and despite its
generous dimensions,
does not have an
internal PSU, which is
a shame. It does, at least,
have an on/off button, though.
The Ins & Outs
Previous MBoxes were mostly quite
small devices with only a handful of ins
and outs, but the MBox Studio positively
bristles with connections. To start with the
more conventional aspects, there are four
analogue inputs on combi sockets. The
first two of these are located on the front
panel and can accept mic, line or instrument
signals, whilst inputs 3 and 4 are round
the back and lack the high-impedance
option. The front panel also houses two
independent headphone sockets and
their associated rotary level controllers.
Connection to loudspeakers is made using
The MBox Studio’s battery of sockets gives you some idea of its size — large.
two pairs of quarter-inch jacks on the back
panel; the second pair can be configured
as alternate speaker outputs or as generic
line outputs depending on your needs. But
that’s not the half of it...
As well as optical and S/PDIF sockets
for digital interfacing, the back panel also
houses eight more quarter-inch jacks. Two
of these are for footswitches or expression
pedals, two are standard balanced line
inputs, but the other four are altogether
more interesting. They can operate as
a pair of line outputs and line inputs, but
they are labelled Line/FX, and their primary
application is to integrate external effects
devices. That includes guitar stompboxes,
and the outputs can be switched into
a special high-impedance mode for that
purpose. The second of these outputs is
also mirrored on a front-panel jack labelled
Hi-Z Out To Amp; this always operates in
high-impedance mode and is intended for
re-amping. It even has an optional ground
lift function. It’s fair to say that someone on
Avid’s design team is pretty serious about
guitar recording.
What’s more, although you might think
that one mic/line/instrument input is much
like another, Avid have supplied some fresh
thinking here too. All parameters including
preamp gain are under digital control, and
most can be adjusted either from the front
panel or the host computer. Up to 50dB
gain is available in 0.1dB steps, and there’s
also a switchable 10dB pad, making a total
gain range of 60dB. Not only is phantom
power switchable per channel, but each
mic/line input also has optional high-pass
filtering, soft limiting and polarity reversal.
And although inputs 5-8 are described as
line inputs, they too have the full 60dB gain
available, so as long as you have the right
cables, they can be used with mics that
don’t require phantom power. The headline
feature, though, is variable input impedance.
In mic mode, this can be set to 1, 5 or 10
kΩ, but once again, it really comes into its
own when recording guitars or basses.
There are no fewer than seven impedance
settings, and each has the option to engage
or disengage modelled input capacitance.
Audio specifications are very decent, with
112dB dynamic range on the mic/line inputs
and 110dB on the outputs. The outputs are
calibrated to produce a maximum level of
+19dBu, while the hottest mic signal that can
be accommodated is a sensible +14dBu.
Guitarists also benefit from the MBox’s
ability to turn itself into a tuner, while
recording engineers will appreciate its
built-in talkback mic. This is available as
a recording input in your DAW for slating
takes and so on, but can also be routed to
the tuner, which is a nice touch. Finally, as
well as supporting stereo S/PDIF and up to
eight channels of ADAT optical digital I/O,
the MBox Studio also has both Bluetooth
in and out, while the last pair of ADAT
channels can be replaced by a stereo
loopback signal if you want to record the
output from another program in your DAW.
That adds up to a total of 21 physical
inputs and 22 or 23 outputs, depending on
whether you count the mirrored re-amping
output. However, although all 21 of these
physical inputs are available for recording,
the DAW only sees four stereo pairs of
outputs, and can’t address physical outputs
directly. Routing of software playback to
physical outputs is one of many things
handled in software.
MBox Control
Like Pro Tools | Carbon, the MBox Studio
comes with a year’s subscription to
Pro Tools Studio, which would cost £249 if
www.soundonsound.com / January 2023 23
 ON TE ST
AV ID M BOX S T U DIO

purchased separately.
However, whereas
Carbon is designed
to be integrated
seamlessly into Avid’s
DAW, with low-latency
cue mixing handled
entirely inside the
Pro Tools mixer,
the MBox Studio
is fundamentally
DAW-agnostic.
Routing and
low-latency cue
mixing are all
handled from within
a separate software
utility called MBox
Control, which is so
fundamental to the
use of the MBox
The MBox Control software.
Studio that there’s a dedicated pink button momentary and press-and-hold actions.
at the top of the unit itself to call it up. In the current version, there is a total of 19 are a few niggles in the current version.
Interface control-panel mixers seem to actions that can be assigned, and most are For example, the first time you fire up your
fall into two basic groups. Some, like the potentially things it would be useful to have new MBox and play back some audio, you’ll
SSL 12 Mixer reviewed elsewhere in this under hardware control. However, the lack hear it in mono, because the Software Input
issue, present a single set of faders for the of labelling means it’s hard to remember channels don’t default to stereo. Mixer solo
main mix, with alternate cue mixes set up what’s assigned to each button until you’ve is always AFL and both mute and solo are
using aux sends. The other approach is to used it for a while. At top right, meanwhile, global, which is a surprising limitation in an
ditch the sends and have a separate fader you’ll find buttons relating to monitor otherwise hugely flexible utility. Channel
panel for each output or pair of outputs, and control, most of which are self-explanatory. linking and unlinking is done on screen
that is what Avid have done. Each physical What isn’t present on the unit itself is any by clicking tiny circles that are difficult to
output is fed by its own, colour-coded mixer, visual indication of successful clocking to an target with the mouse. Faders occasionally
as is the internal Avid FX reverb and delay external source. move in ‘fine control’ mode even when the
combo. The main, alternate, headphone and appropriate modifier key is not pressed. And
S/PDIF outs are stereo, and the Bluetooth
At The Controls return levels from the Avid FX can only be
out always duplicates the alternate output, Like the MBox Studio itself, MBox Control adjusted from within the Avid FX mixer view.
but the Line/FX and ADAT outs can be is an ambitious creation, and that has both It would make sense to add a separate FX
split to mono, conceivably giving you up good and bad consequences. To take return channel to each individual mixer.
to 15 different mixer panels not counting the latter first, its complexity makes it a bit While I’m in nitpicking mode, I also
the FX. Each of these can draw from two intimidating to get to grips with, and there encountered a few issues with the
rows of sources. The top row of Hardware
Inputs presents all 21 physical inputs,
whilst the eight mono or four stereo DAW Tuna
returns appear in the middle row labelled
Hitting the MBox Studio’s Tune button cleverly repurposes the meter LEDs to provide a clear indication
Software Inputs. of the detected pitch and how far sharp or flat things are. This is mirrored on screen within a pop-up
A Channel Focus pane to the left of the window that looks somewhat like a typical rackmount tuner. Tuning is accurate and fast, and I like the
fader banks shows a detailed overview of all fact that you don’t have to select an input — it acts on any audio signal coming into the mixer. Tuning
settings for the currently selected channel. using plug-ins can be a faff, so I really like this feature, which is every bit as immediate as using a tuner
These include four-band EQ on every input, pedal without the need to patch one in. The only down side is that using the tuner temporarily mutes
which can be switched into the record path all input signals, which isn’t always what you want.
if desired. Pressing the left encoder on the
MBox cycles the focus selection through
the eight analogue input channels, and the
main input controls on the focus channel are
accessible from dedicated buttons, while
rotating the left encoder adjusts input gain.
Four coloured buttons with no legends
in the centre of the MBox panel are
assignable within MBox Control, and each
can have a separate function assigned to

24 January 2023 / www.soundonsound.com

 SONTRONICS
S
O
N
T
R
O
N
I
C
S ® ®

hardware. Changing sample rate, switching clock source and

powering up generates noise at the outputs even if they’re
muted; I also discovered problems with low-level hiss on the
main output, and crosstalk from the main output onto the cardioid valve condenser microphone
first FX/Line out. Avid were able to confirm these issues, and
say that they relate to bugs introduced in the v1.0.0 firmware.
There is a good chance that they will be fixed by the time
you read this, and my niggles with MBox Control are likewise
mostly slated for correction soon.
The main takeaway, though, is that the MBox Studio’s
built-in mixer is an extremely deep and powerful affair with
a greater than usual level of hands-on control, which enables “Sets a n
“Sets new
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enchmark”
Ja k Ruston, Reviewer (Sound
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Sound On
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some pretty advanced recording and monitoring scenarios.
For example, you could plug a guitar into input 1 and use
the mixer to mult it to the two FX/Line loops, with different
“E
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stompbox effects chains in each. Simultaneously, you could any
any vvintage mic”
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send the dry signal out of the front-panel high-impedance
socket for re-amping. Elsewhere, the built-in delay and
reverb are surprisingly good, and there are lots of thoughtful
“An absolute
“An absolute dream!”
dream!”
PJ Harvey, A
Artist/Composer
rtist/Composer
t/ er
touches such as the balance and width controls on the stereo
channels. And if it all gets too complex, you have the option “S
Soo ssilky
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and
nd
to Group All Monitor Mixes, which means that the same
fader balance gets sent to the main, alternate, Bluetooth and
beautifully
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A rew Dudman,
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Andrew Dudma
dman, Engineer
Eng
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er (Abbey
(Abbey Road)
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both headphone outputs. In most real-world situations, this
eliminates the need to flip between different mixer panes.
Of course, you do still need to flip between MBox , DEVELO
ED

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DESIGN
Control and your DAW, and it’s worth noting that Pro Tools

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itself hasn’t changed to accommodate the MBox Studio.

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For example, it still doesn’t compensate for input latency ADE IN T

when placing recorded audio on the timeline, so if you

use the high-impedance output to re-amp a previously
recorded signal, the miked amp part won’t quite line up
with the original, and nor will audio routed through the
FX loop. This isn’t hard to correct after the fact by sliding
clips, but it does feel a bit like being back in 1999.
Talking of latency, the MBox seems to use the built-in
Core Audio USB driver on Mac OS, but its built-in mixer
adds some further delay. At 44.1kHz and with a 32-sample
buffer size, Reaper reported the round-trip latency as
7.8ms, but a loopback test suggests that the true figure
was about 2.4ms higher. This is a little disappointing by
today’s standards, but at least the cue-mixing facilities in
MBox Control mean there’s not often a need to monitor
input signals through the DAW.
Not being integrated with Pro Tools also has the obvious
advantage of making the MBox Studio equally tempting
for users of other DAWs, as is not the case with Carbon.
And I think the temptation will be a strong one, especially
once the minor annoyances described above have been
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tones that you’d struggle to get on any other interface. The
MBox Studio is an ambitious product that showcases both
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 TECHNIQUE
Just how do professional engineers tackle the earliest
stages of their mixes, and can the rest of us learn
anything from their approach?
NEIL ROGERS
H
ave you ever wondered how
professional mix engineers get
their projects underway? I invited
four of them to share some insights on
both the technical aspects and the ‘softer
skills’ involved. They each specialise
in a different genre, and you can find
out more about them in the ‘About The
Engineers’ box.
We began our conversations
with an exploration of some of the
behind-the-scenes discussions that occur
before a professional mix takes place,
26 January 2023 / www.soundonsound.com
before moving on to the mix-project
preparation and the earliest stages of
the mix itself. The engineers also shared
some helpful troubleshooting tips, for
those occasions when things aren’t
progressing quite as we’d like. I also
invited them, where possible, to think
about how their advice might apply
to keen amateur engineers, not just
aspiring professionals.
Great Expectations
My first questions were about the
interactions with clients and artists which
usually take place before a professional
mixer even thinks about starting
startiing work
w
on a project, and obviously this begins
with the question of whether it’s the right
project. Maria Elisa Ayerbe explained to
me that “my lawyer says: ‘I won’t take
a case if I know I won’t win it,’ and I apply
the same philosophy to working with
a customer... When a client reaches out
to me, I ask them to send me a rough mix
or production demo. I want to judge if I’m
the right person for the job, from a craft
and experience perspective.”
But even if the project looks a good
fit, there are practical issues that need
ironing out. Romesh Dodangoda, for
instance, stressed that: “The conversation
about timescales is one my first priorities.
Sometimes when it’s a big project, I may
have a team involved, and I need to
make sure people are available. I also
want to always be on time with the
delivery, and that is something that’s
appreciated by labels and managers.”
Rachel Moore expanded on this side of
things: “Budget and timeline of when
they want the project done are the first
things I discuss. I like to be as upfront
as possible with expectations. Once
these are agreed, I will draw
up a simple contract based
on our discussions. Good
communication throughout
the process tends to solve
most things.” Likewise, Dan
Gresham stressed this need
to have “clear Ts&Cs” and the
importance of “ensuring that
everyone knows where they
stand from the outset”.
Communication about the
business side of things is
clearly important, then, but
I was also keen to understand
how these engineers ensure
they’re on the same page
as the artist from a creative
point of view — particularly
given that so much mixing is
now done remotely. Rachael
Moore pointed out that much
depends on your degree
of involvement at other
stages of the project : “If it
is something I didn’t have
a hand in recording, I like to
have a discussion either on
the phone or email about the
vibe they are going for. If they
get specific, then I ask for
references to make sure we
are thinking the same thing
before starting any work on
the mix.” Maria Elisa Ayerbe
Rachael Moore: “I’m a fan of the
also stressed the importance
of speaking to artists and
Prince ideology, where he said
discussing references, that any one track in a song,
especially when working
with less established or more paired with the vocal, should
‘up-and-coming’ acts: “I want
to know what they aspire to.
As well as a creative steer,
I’m trying to find out if their
expectations are realistic.”
Dan Gresham, who works
predominantly with electronic artists,
pointed out that: “Many of my clients have
typically worked hard to get their demo
to where it is and are not looking for me
to deliver a totally new vision of a track.
My job is to pay close attention to what
they have done already and focus on
improving clarity, separation, and crucially,
how the track will translate on both home
setups and big sound systems.”
Prep School
With the business matters and creative
aims addressed, I asked how ‘mix-ready’
these engineers felt the recordings
sound like a record!”
needed to be for them to weave their
magic, and what issues they or their
assistants might look out for in the
raw files. Romesh Dodangoda feels
it’s important to establish if there are
problems which extend beyond his brief:
“If there are vocals that are a little messy
with the timing, or noise that shouldn’t be
there, we are normally happy to clean this
up in the mix prep. If it’s anything more
stylistic or production-related, then we
will flag this up with the client”.
Maria Elisa Ayerbe, meanwhile,
explained that “my assistant or I will
check that all the files work correctly
and that the multitrack matches their
production demo. I’ve received sessions
where the multitrack was
32 bars shorter than the
demo, and I make it clear
that it’s not my job to amend
the arrangement.”
Rachael Moore generally
prefers to oversee her own
mix preparation “because
I’m a bit of a control freak,”
and explained that she takes
a little time to assess what she
has been given to work with.
“I look and see if any editing,
tightening, or tuning and
cleaning of the vocals needs
to happen. If it’s something
I didn’t track, I will either ask
the client to get someone to
do that, or I will give it to a few
people I trust to clean.”
Dan Gresham also
confirmed that he will “check
through all the parts and make
sure that there aren’t any
obvious issues that are going
to hold things back (editing
clicks, plosives on a vocal).
If necessary I’ll top and tail
sections to cut out any hissy
gaps. I’ll also check timing on
rhythmic elements to make
sure that things are sitting
together tightly (or not — if
that’s what’s wanted!)”.
I was really interested to
know, once they’ve run the
rule over the initial files and
know they’re ready to work
with them, what any further
mix preparation stages look
like for our engineers, and
how important they feel this is
to the way they work. Romesh
Dodangoda was very clear on
the importance of a separate prep stage.
“When I’m mixing,” he explained, “I really
don’t want to think about the DAW at all
if I can help it. I want to focus more on
chasing the sound in my head. Having
an assistant to handle the prep allows
me to not get too close to individual
elements.” He was conscious, though,
that an assistant is a luxury we can’t all
www.soundonsound.com / January 2023 27
 TECHNIQUE
HOW TO S TA RT A MIX
afford: “If you’re on your own, try to do it
on a completely different day, so you can
be as fresh as possible for the mix.”
Maria Elisa Ayerbe went into a little
more detail about what this stage might
entail: “My assistant will prep things like
track names, colour coding, creating
markers, and they will also take care
of basic routing, gain
staging and panning,
and do some initial lead
vocal gain levelling or
plosive riding to help with
compression settings.
Gain staging is very
important in how I mix,
and getting levels in the
right place is done during
the prep stage, to ensure
that my bus compressor
is working from the
moment I begin the mix”.
Conscious that those
ideas of ‘gain staging’
and mixing through
master and subgroup
bus processing can too
easily be misunderstood
by rookie home-studio
engineers, I pushed the
engineers to discuss
more of what they have
in mind when they
talk about such things,
particularly for those
mixing in software alone.
“When I went from OTB
mixing to ITB,” offered
Romesh, “I found it really
Maria Elisa Ayerbe: “Master bus
hard to mix inside the
DAW until I started having
processing is important to how I mix,
a bit of colour on the mix
bus, via certain plug-ins.
and I find it works best when I focus on
It’s key for me to have the loudest part of the song first.”
some excitement to feed
off, and I’ll always have
it on from the very start. I don’t get too
bogged down about trying to meet any
kind of specific levels; as long as the
faders are playing nice and any master
processing is hitting roughly where
I would expect it to be, we’re all good!”
Dan Gresham also makes use of bus
processing, and discussed how gain
structure relates to that. “I will often
use subgroup processing, with a good
example being the drum bus — to add
a bit of ‘crunch’ and to glue the drums
together. If I do that, it usually gets added
from the start, which is one of the reasons
why one of the very first jobs I do is get
28 January 2023 / www.soundonsound.com
all the source audio to an appropriate
level.” He continued: “I know there are
technical reasons why gain staging, in
the traditional analogue sense, is less
of an issue with 24- or 32-bit audio, but
for me it’s about having the audio levels
where I expect them to be with how I like
to work.” Rachael confirmed that “gain
staging is important and is something
I address first, as you don’t want to
mix yourself into a corner, so to speak.
Nothing is more annoying than to get
the mix where you want and then see
everything is way too loud.”
I also asked the engineers for opinions
on the value of mix templates in their
DAW software, and the replies were
more nuanced than I’d anticipated. Maria
Elisa Ayerbe: “I don’t use a broader
mixing template, but I do have an effects
template that I like to import to my
sessions: delays, reverbs, chorus, parallel
compression, parallel distortion, panners,
phasers, etc. Every time I create a cool
effect for a song, I make sure I add it to
my template, so that it can be repurposed
for another song in the future.”
Rachael, on the other hand, does have
“a general template with effects, a basic
mix bus I am feeling at the moment, basic
vocal chain, drum bus chain that I like
to start with. I almost
always will pull those
things into the session
and then tailor things to
the vibe of that specific
project (ie. the tonality
of the vocal, drums, etc).
I will usually figure this
out on the first song and
then make a template
for the project itself
that I use as a starting
place for the rest of the
project. So I essentially
create a template for
each project.”
Romesh also has
a template, “but not in the
sense of just importing
things into it and hoping
for the best. I’m not sure
that’s a great way to mix,
personally. However,
there are certain
techniques and plug-ins
I’ve used over the years
that I just know I like and
will use time and time
again. I look at a template
as having a bunch of
outboard that you know
you like, all plugged
in and ready to go. It’s
like pressing the insert
buttons on a console
where you’ve already got
some outboard patched
in and ready to go.”
Dan Gresham prefers not to use
a template at all: “I start with an entirely
blank project. There are processes that
I’ll follow more often than not, but I don’t
set these in stone with a template — it
pushes me down certain alleys that
hamper what I feel is the creative part of
a mix.”
Getting In The Zone
It’s all too easy with creative endeavours
like mixing to overlook the ‘human angle’,
and when asked about this Romesh
clearly felt strongly: “I think the mindset
conversation is massively important. What
 TECHNIQUE
HOW TO S TA RT A MIX

you have to remember is we’re humans, But I also wanted to glean from them for what I need to home in on first.” And
and some days it just doesn’t work. When a sense of what the early stages of the Dan Gresham’s early moves reflect the
that happens, I will just leave it and come mix itself actually look like: while, clearly, more defined electronic elements he
back to it the day after and, suddenly, it’s the approach will vary in some respects usually works with: “I’ll listen through to
all so easy.” from one project to another, is there the raw tracks with all the faders at unity.
With this idea in mind, I asked the anything they typically do during the first, I’m listening out for how the rhythmic
engineers what they do to try and get say, 10-20 minutes of every mix? Which elements work together with the bass,
themselves into a mixing mindset. instruments or song sections might they what the instrumental elements are
Rachael Moore feels that deadlines help: focus on first, for example? doing and where a vocal might be sitting
“If I have a deadline, in all of that. I want to
it makes me focus, quickly establish what’s
not get bogged down, important in the tune
and trust my first and what has more of
instincts.” Dan Gresham a supportive role”.
concurred: “I’d also As for which
mention how helpful instruments are tackled
having a deadline can first, for Dan Gresham,
be for a project. As well working with electronic
as helping me schedule sources “more often
projects, it brings some than not... it’s about
clarity to getting up and nailing the drums and
running with a mix.” the bass. Often, initially,
Most were also just the kick drum and
conscious that they the bass. Getting those
work better at some two working together
times of day than cleanly is critical. I’m
others. Maria Elisa generally looking for
Ayerbe, for example, a solid bass that can
said: “I try and keep underpin the entire
a certain part of my day tune, and a punchy
clear for mixing — I find kick drum that pokes
I do my best creative through with enough
work in the afternoon clarity and definition.”
and evenings. I try hard Maria Elisa Ayerbe
to stay organised with
my admin and keep
Romesh Dodangoda: “If you’re spending all starts in a similar place,
but also mentioned
it away from my ‘mix
time’.” Dan Gresham, on
your time EQ’ing guitars without the vocals some other priorities:
“I’ll quickly look to get
the other hand, tends
“to work on mixes in
in place — these elements can often sit the groove locked,
(the kick and bass,
the middle of the day. in the same range frequency-wise — then before focusing on
By that point I’ve gone the main storyteller —
through my admin, you’re likely to have a hard time getting the lead vocal — the
dealt with a few audio snare, and then any
jobs and my ears and the vocals to fit in” primary accompanying
brain are warmed up.” parts, such as guitars,
Rachael is equally piano or loops, make
conscious of when she works best, and Romesh stressed that “the first period accommodations for their best friends
how to get into the zone when that’s of a mix is crucial for me. I work fast and (background vocals and harmonies), and
not possible: “Evening is definitely more like to quickly get a feel for the song then I just deal with the crowd (the rest of
inspiring for me but needs often must, and where things should live sonically. the band).”
and if I’m working in the day I like to light I want my gut to dictate things as much Romesh also starts with the groove:
Palo Santo [wooden incense sticks] and as possible, and within about an hour the “I like to plant my kick, snare and bass
maybe listen to some music in the vibe song is usually in a pretty good place. It’s into a good place and then if I’ve got
of the project, just to get my brain going the finer details, such as automation and those right, it’s a lot easier for me. I find
before I start.” finer balances, that take more of my time”. that with rock music if you get the drums
Rachael stated that “generally, I like to to stand proud, everything else falls
Where To Start work on getting a solid balance among on top so easily. If you haven’t got the
So far, we’ve talked largely about what all the instruments first without hitting drums right, you spend a lot of time
happens to get an engineer to the point any solo buttons. I want to see how the covering it up and that’s when it can all
when they can start mixing in earnest. vocal is sitting and then start to get a vibe sound a bit undefined. I rarely start at the

30 January 2023 / www.soundonsound.com

 TECHNIQUE
HOW TO S TA RT A MIX

vocal, I can totally see how dance music is often the first
that works for people but it drop. Although a track will
rarely works for me”. often build from there, it is
Rachael offered the most critical section.”
up a slightly different
perspective: “I try to
What About The
determine the core, or Vocals?
heart of the song first. If it Even if the engineer doesn’t
is something where lyrically work on a vocal first, most
the singer is telling a story put significant work into
then I will start with that them fairly early. Romesh,
and the main instrument for instance, confirmed that:
propping that up. If it’s “The vocals will be in pretty
a singer-songwriter track, quickly, and it’s important
for example, I will start with to remember that they are
the vocal and acoustic guitar almost certainly going to be
track and work everything the focus of the mix. If you’re
else around that, making spending all your time EQ’ing
sure nothing is hindering guitars without the vocals in
the core of the song, which place — these elements can
is the lyric. If it is a pop often sit in the same range
track, where the lyric isn’t as frequency-wise — then you’re
important, then I focus on the likely to have a hard time
groove of the song and the getting the vocals to fit in.
excitement factor first. I’m There are certain plug-ins that
a fan of the Prince ideology, I just know I love on vocals —
where he said that any one the UA [1176] Blue Stripe, for
track in a song, paired with example — so these might

Photo: Garry Jones Photography

the vocal, should sound like come into play very early on.”
a record!” Rachel expanded on that:
Given our earlier “The vocal is one of the first
discussion about bus things I like to get sounding
processing, it wasnt like a record. With the right
a surprise to learn that the treatment, it can get your
engineers all focus on the
biggest or most crucial Dan Gresham: “I know there are creativity moving and dictate
where to go next in the mix
section of a song first. For
Maria Elisa Ayerbe this
technical reasons why gain staging, — do we want the drums to
be a certain vibe now, based
means she likes “to focus
on the big chorus first, then
in the traditional analogue sense, on the vibe of the vocal, for
example?” Dan Gresham,
verse 1, then intro, and is less of an issue with 32-bit audio, though, again highlighted
then everything else. There that the considerations
are musical and technical but for me it’s about having the in dance music can be
reasons for this, as I like somewhat different. “I’ll
to get any bus processing audio levels where I expect them to usually have the instrumental
working well as early as
I can. Master bus processing
be with how I like to work.” mix fairly well established
before introducing the vocal,”
is important to how I mix, he told me. “Whilst [the
and I find it works best when vocal] can be important, in
I focus on the loudest part of the song If the verse has so many layers and parts, a lot of dance music it can be more of an
first.” Rachael offered similar thoughts: the chorus is possibly not going to have instrumental layer rather than the primary
“I will identify where the biggest part the impact they want. So, often, starting at focus. Sometimes that leads to reworking
of the song is and work to balance that the chorus is a good way to see how far the instrumental mix though, depending
section. I can then understand where the you can push everything.” on the writing and arrangement”.
sections before and after it need to be Dan Gresham’s advice again had
dynamically — from a musical as well as a slightly different focus, borne of the
Keeping Things Moving
a technical perspective.” And Romesh also nature of the genres in which he works Coming back to that point about being
expanded on this theme: “I think one of — dance tracks are not typically based human, even engineers at the top of their
the things that can be forgotten is to make around traditional verse/chorus song game and with a strong track record of
a chorus big, you need to think about the structures: “I’ll usually start with the most creating well-received mixes can face
arrangement and what the verse is doing. important section of the track, which in the same sorts of issues and moments

32 January 2023 / www.soundonsound.com

of doubt as the rest of us! So I asked our
engineers when they find themselves
struggling, and what they do about it —
and, specifically, if they ever feel it’s better
to just start over.
Rachael suggested that “if I run
into problems, it’s usually because I’ve
become too engrossed and have started
to hyper-focus on one particular part of
a mix. I don’t tend to start again, but if I’ve
completely lost the vibe of the track I will
bypass what I was doing and return to the
core of the song again, to figure out where
I went wrong. Generally, the culprit is too
much processing!”
Dan explained earlier how important
the demo is in his line of work and how he
uses that as a reference point: “The only
time I’ll start things again is when I go back
to the demo and I feel like I’m not adding
anything to the mix. That usually leads to
a change in approach, which hopefully
[delivers] an improved result.”
Maria says that she “used to backtrack
a lot in my early career, but not anymore.
I have become good at keeping track of
my decisions and staying on top of my
moves as I advance further into the mixing
process. This ‘only’ took me about eight
years to figure out correctly! If I feel I am
getting stuck, I will just move to a different
song, go for a walk, or grab a glass of wine
and chill for a moment.”
Romesh told me that: “I usually know
within the first 30 minutes if the mix is
going to be straightforward or a tricky one
to finish. For me, I always want to be super
proud of the mix before I send it to the
client so whatever it takes to get there is
what I’ll do!” The question also prompted
a relevant anecdote: “I was mixing Bring
Me The Horizon’s Live At The Royal Albert
Hall. It was such a hard mix, as the band
were playing with a full orchestra and choir.
I mixed six songs initially, took them home
to listen and felt it wasn’t exactly what
was in my head. I had two options: I could
either edit the existing mix or just go again
from scratch. I chose the latter and I’m
so glad, because when I started again
everything fell into place and it was exactly
how I wanted it to sound.”
Words Of Wisdom
I asked our engineers if, once they’d got
the mix underway, they had any other
tips on where to take things from there.
Nearly all of them stressed the importance
of taking regular breaks, to avoid fatigue
and retain their perspective. Rachael, for
example, told me: “I force myself to take
a break every hour to an hour and a half
to reset, and I’ve noticed I have a bad
habit of wanting to mix louder when I’m
feeling something. I frequently like to zoom
out and assess what I am doing to make
sure it’s working. You learn a lot [through]
freshening your perspective frequently.”
Romesh felt the same: “It’s so important
to take breaks. When you’re constantly
listening to the same piece of music on
repeat, you can start to lose perspective.”
Dan offered some different advice: “I always
work at a consistent monitoring level and
also a consistent RMS level [at the mix
bus]. I know how things should ‘feel’ at that
level — how heavy a bassline should feel,
how punchy a kick, how snappy a snare. It
helps me to get to a sonically balanced mix
without having to refer to analysers.”
Before we wrapped things up, I asked
each engineer if they had any final words
of wisdom to offer less experienced mixers
who are keen to improve. Dan Gresham
explained that “a good mix starts with
good writing and good arrangement.
Sometimes, elements in a track just won’t
sit together well even with a good mix. If
you’re struggling to get things to work, ask
yourself whether a change in arrangement
or choice of sounds would solve the
problem.” Romesh reiterated something
he’d mentioned earlier: “If you feel a mix
is not coming together, don’t be afraid to
start again on a different day. More than
likely you’ll find it all comes into place
much easier!”
Maria suggests that you “don’t
hyper-focus on single elements or
instruments — it’s always about the
big picture. Remember you are mixing
within the context of a song and that the
bypass button is your best friend. If the
EQ you worked on in solo does not sound
good within the context of the song it’s
pointless.” Rachael also focused on EQ: “If
you’re struggling to get something to come
through in a mix, or you can’t get the vocal
to sit above the track, it is almost always
an EQ thing... Use any of the amazing
visual EQ tools we have now and figure out
where the [frequency] build-ups are and
what track is causing it. Manage the space
in your mix frequency-wise, and you’ll find
you suddenly don’t need to make things
louder or use loads of automation. Resist
making things louder and focus on what is
taking up your room frequency-wise.”
Takeaways?
So what lessons can we draw from all
of this? I’ve focused on the process of
About The Engineers
Rachael Moore is a Grammy-nominated
engineer/mixer who, as T Bone Burnett’s
engineer, has worked with the likes of
Robert Plant, Alison Krauss, Elvis Costello
and Jessica Chastain. Based in Nashville,
Rachael is also a busy freelance recording
and mix engineer.
Maria Elisa Ayerbe is a Latin
Grammy-winning and multiple
Grammy-nominated engineer, producer
and composer living in Miami. She has
been involved in the musical productions
of a host of successful artists, including
Mary J Blige, Ricky Martin, Marc Anthony,
J.Lo, Natti Natasha, Il Divo, Laura
Pausini, Juanes, New Kids On The Block,
Aterciopelados, Gian Marco, Kronos
Quartet and Adriana Lucia.
Romesh Dodangoda is
a Grammy-nominated record producer/
mixer from Wales. Specialising in rock/pop
and metal, Romesh has worked with the
likes of Bring Me The Horizon, Motörhead,
Nova Twins and Twin Atlantic. He also
runs Control Room, an online collaborative
networking space for the production and
mixing community: www.control-room.net
Dan Gresham is a British artist, producer
and engineer who is well known in the drum
& bass world under his Nu:Tone pseudonym,
and for his long association with Hospital
Records. Dan also works as a mixing and
mastering engineer, specialising in all forms
of electronic and dance music. You can find
out more at www.ntmastering.com
starting a mix because of the relatively
short window in which we can assess
what’s really going on with a piece of
music. A few things struck me in these
conversations, but what came across
most strongly was the disciplined,
professional approach; the effort put into
organisation and communication.
If they work to understand their clients’
aims and expectations, and then to get
some of the niggly technical stuff out of
the way, an engineer can focus on the
more creative, instinctive side of the job
when they mix — and that enables them
to deliver something that really works in
terms of the music, energy and all the
stuff that really matters. We might not all
have the luxury of an assistant, but any of
us can approach our mixes with that sort
of mindset! I’d like to say thanks to these
engineers, who were very generous with
their time, and I hope you’ve found some
ideas in here that will help you get your
next mix off to a great start.
www.soundonsound.com / January 2023 33
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 ON TE ST
Dawesome Novum
Software Synthesizer
Dawesome’s Novum is a granular synthesizer with
a multi-layered approach to sound design.
Dawesome Novum
$179
PROS
• Novum’s unique approach means
that you can get a lot of mileage
out of splitting and recombining
even a relatively small number of
source samples.
• Easy to use at a basic level but
there’s plenty of depth to explore.
CONS
• As with many granular synths, the
resulting sounds do tend to have an
identifiably granular character that
often sounds somewhat diffuse.
SUMMARY
For creating shifting textures and beds,
Novum does a great job. Definitely one
for the more experimentally minded.
36 January 2023 / www.soundonsound.com
PAUL WHITE
escribing Novum is no easy task as
D it is a synthesizer unlike anything
I’ve used before. It comes with
over 300 factory patches and despite the
apparent complexity of what is going on
under the hood, it is highly optimised for
CPU efficiency and has a very approachable
GUI that can also be resized. Novum
requires Mac OS 10.13+ or Windows 10+
supporting AU and VST3 plug-in formats.
Even my 2012 Mac Pro (Mojave) ran it OK,
though at some settings the CPU load was
still close to the limit, so if using multiple
instances on an older machine, it may be
best to bounce the tracks as audio once
you’re happy with them. The samples that
form the basis of the sounds come as
a separate download from the instrument
plug-in so that you don’t have to download
the entire library each time there is
a software update.
Basic Principles
For those unfamiliar with granular synthesis,
the general principle is that the original
sound, in this case a sample, is loaded into
memory, but instead of simply being played
back as it would be in a conventional
sampler, small snippets of sound can be
lifted from anywhere in the sample and
then used as the basis for creating new
sounds. By scanning different parts of the
sample and by overlapping the grains,
moving textures can be created, while
using fewer and longer grains produces
discrete repeats. Using modulation to
change the way grain samples are taken
can produce sounds that are far removed
from the original sample. Novum goes
even further.
 Samples are first broken down into six
layers using algorithms based on machine
learning. These operate somewhat like
the aspects of human perception that
allow us to mentally separate sounds
when several things are going on at once.
Simple sounds tend to be broken down into
fairly similar-sounding layers while more
complex sounds or drum loops show very
obvious differences between the parts. As
I understand it the original sample is broken
down into a very large number of frequency
bands, then the layers are created by
adjusting the levels of each frequency band
according to the separation algorithm. If all
six layers are added together, the original
sound is reconstructed. Each layer then
becomes a timbre in its own right and has
its own complex envelope created by the
extraction process, though the user can
draw in new envelopes for the individual
layers if so desired. These layers, which can
be turned on or off, or soloed individually,
are then subjected to granular processing,
so what you finally hear may bear little
relation to the sample you started with.
Further processing is possible
via synth-style filters, effects and
comprehensive modulation. There’s full
MPE support and most parameters can be
modulated either by internal modulators,
such as LFOs, or by external controllers.
You can also import your own samples or
drum loops, giving you the opportunity to
turn familiar sounds into something very
different. Samples can be dragged into
the main waveform display, after which it
takes a few seconds for your sample to
be split into layers, or different samples
can be dragged onto individual layers. For
example, you could combine six different
choir samples, one on each layer, to create
something very rich and textural. It is
recommended that imported samples are
at 44.1kHz or 48kHz. If the sound is pitched,
Novum is able to extract the pitch.
Getting To Know You
The GUI may look a little unfamiliar at first
but it’s actually pretty intuitive once you’ve
spent a short time exploring it. The included
waveform library, called Basics, appears
down the left-hand side of the screen in
the browser section, each with a graphical
depiction of its waveform. When one is
selected its six layers are also depicted
graphically. The currently loaded sound
is shown top centre with power buttons
below for turning the layers on or off, along
with depictions of the envelope for each
layer. Clicking and dragging on the display
rotates it so that you get a view of the six
layers, each of which has a different colour
assigned to it.
Patch presets are accessed via the
icon to the left of the patch name box at
the top of the browser and are arranged
in categories such as Pads, Basses, Leads,
Drones and so on, plus there are several
banks contributed by individual sound
designers. Sounds can also be searched by
name — for example, you may want to view
all the patches containing the word Strings.
A star can be added to favourite patches
so that you can also filter by favourites. On
the main wave display, white circles depict
the position and movement of the granular
‘emitters’. Modulation sources used in the
patch are shown on the right of the screen
with a plus button to add more sources
if needed.
Two Lock icons allow the user to lock in
place the timbres and/or their envelopes,
making it possible to try new presets while
retaining the timbre and envelope settings
from the previous patch, which can bring
up some interesting variations. User sounds
can be dragged directly on to the spectral
display, after which it takes a few seconds
for Novum to deconstruct them into timbral
layers. When dragging different samples on
to individual layers to create more complex
layered sounds, these can be from the
basics library or from an external sample
library or short audio file.
The fun starts when you start combining
attributes of one sound with another, which
you can do by dragging a new sound
waveform on to the line of Timbre power
The ‘Timbre Flower’ allows different sonic textures to be explored.
buttons. Now you have the timbral filters
and envelopes from the new sound applied
to the sound you originally loaded. If you
want to replace only a single layer, you
can drag and drop single layers from the
sound waveform selected in the browser
on to a specific layer in the current path.
Taking the envelopes from a drum loop,
for example, then applying them to a more
consistent sound, can produce some very
interesting rhythmic effects.
Five view sections access controls for
Timbre, Envelope, Synth, FX and Mod, with
the relevant controls being shown below.
In the Timbre view, the ‘Timbre Flower’ is
bottom centre and this offers a means of
varying the sound of the currently selected
layer with the original timbre in the centre.
Moving upwards softens the sound while
to the right the sound becomes more
noise-like. Moving down makes the sound
more like that of a conventional synth.
‘Homogenize’ is designed to help smooth
out sounds that might start to sound a little
dirty after editing or exchanging envelopes,
apparently by removing phase information.
The small orange brackets to the left of
a control allow linking so that edits apply to
all six layers rather than to individual layers.
At the very bottom of the window you can
see the incoming MIDI notes.
Grains
As the sound engine is based around
granular synthesis the settings for Grains
and Grain Emitter in the Timbre view have
a profound effect on the sound — as do
the filter and effects settings found in the
www.soundonsound.com / Janaury 2023 37
 ON TE ST
DAWESOME NOVUM
other views. There are six effects types,
with the option to run them in series or
parallel. Of particular note are the gorgeous
Shimmer and Cloud reverbs, which can
have a massive effect on the overall sound
— so much so that it can be useful to turn
them off when editing patches as they
can overwhelm some of the more subtle
changes. Some of the patches can sound
quite lumpy without the effects switched
on but add the effects and the end result is
transformed into something silky smooth.
This is evident in some of the string and
choral settings, which manage to sound
really ‘expensive’.
There are various settings for the
start position of the grains, grain loop
points, Grain Density, positional Jitter, Blur,
Grain Size and so on, all of which can
be modulated via the available internal
LFOs, envelopes, random generators,
sequencers and so on, or via external MIDI
control. There’s also the option to draw in
custom LFO shapes. Where relevant, the
internal modulators have the option of free
running or DAW sync. As the Grain Density
is reduced, the individual grains cause the
sound intensity to be modulated, as sound
is only generated when a grain is playing.
Grain Size generates more grains at lower
settings so the pulsing effect heard at lower
densities will be faster. It is also possible to
add random panning to each grain.
Speed and playback pitch are
independent, so drum loops can be
changed in pitch while maintaining their
original tempo. Via the Envelope tab, the
envelopes for the layers may be viewed
in detail and new envelopes drawn in as
required. As new envelopes are created,
they are reflected in the main waveform
display. There’s also a familiar ADSR
envelope shaper at the end of the chain but
before the FX section. Comb filtering can
be applied to introduce string-like timbres,
and controls for the grain emitters are also
found here.
In the Syn menu, there is a Dirt control to
add harmonics, as well as resonant filtering,
and a Syntify control that manipulates the
harmonic structure to make it sound more
synth-like if required. There’s a lot of heavy
maths behind this seemingly simple control
and there are two choices of character for
Syntify, so you just pick whichever sounds
best. Other synth-style controls allow for
detuning, pitch adjustment, transpose
and so on. To modulate a parameter,
click on it to select it and then adjust the
modulation amount using the circular dial
next to the desired modulation source. You
38 January 2023 / www.soundonsound.com
The Syn section features a Syntify slider, and more familiar controls including filters and distortion.
can even use one modulator to modulate
another. A similar process is used to assign
parameters to external control sources such
as the mod wheel.
Practicalities
It pays to start fairly simple, first by exploring
the presets, then by making some tweaks to
them to hear what that sounds like. Switch
off the effects to hear what the granular
engine is really doing and then experiment
with the granular settings. You can also solo
individual layers to hear what they sound
like, and sometimes a sound appears more
focused if not all the layers are used. When
you get around to combining elements from
different sounds, it is best to keep an open
mind and just experiment: sometimes you
get a wonderful surprise and other times
a grainy mush that is too dense to be useful.
I don’t have any real criticisms of Novum,
as the designer has put in a lot of hard
work to provide intuitive controls for what
is in reality a very complex instrument. It
might have been fun to add in the option
of adding random octave shifts to some
of the grains as you find in many of the
stand-alone granular effects processors
on the market, so I mentioned this to the
designer and it was hinted that something
along these lines and beyond is already in
the pipeline for a future update.
Granular synthesis does tend to
have a certain diffuse character, though
constraining the grain activity close to one
part of a sample can yield more focused
results. On the other hand, if you like
soundscapes that continually evolve, fat
bass drones or silky pads, then Novum
offers an array of sonic delights from the
gritty to the super smooth. In this respect
its FX section does a lot of the heavy lifting,
especially where you want to add a dreamy
ambience. It perhaps isn’t the instrument
to choose if you like realistic depictions of
acoustic instruments, but then it could be
argued that the world has enough of those
already. Novum is a playground for sonic
experimentations and excels at providing
backdrops for other instruments, while some
simple external controls or basic modulation
setups can add a lot of movement to the
sound. When you consider the complex
technology underpinning this instrument, it
makes you appreciate its very approachable
user interface even more, especially
the drag-and-drop operations and the
straightforward way in which modulation
assignments are made.
As to whom Novum is aimed at, sci-fi or
fantasy soundtrack composers are obvious
candidates — creating dramatic beds and
cues is very straightforward and the end
results can be truly epic. Then you have
the experimental EDM, chill or ambient
composers for whom Novum is simply the
gift that keeps on giving — just recombining
elements of the existing library could keep
you busy for months. In fact anyone who
wants a break from the mundane will
have a lot of fun with Novum, and with
a generous 90-day demo period there’s
nothing to lose.
£ $179 including VAT.
W www.tracktion.com
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 ON TE ST

Neumann
KH 150
Active Monitors
The latest model in Neumann’s
celebrated KH series sets a new
standard for nearfield monitoring.
PHIL WARD

eumann’s range of active monitors, from the

N diminutive KH 80 to the much more substantial

KH 420, have long been admired by Sound On
Sound reviewers and readers alike. But there has always
been a gap in the Neumann range between the compact
two-way KH 120 and the significantly larger three-way
KH 310. It’s a gap that I suspect has long frustrated those
aspiring Neumann monitor owners for whom the KH 120
doesn’t quite meet bandwidth and volume level needs,
and the KH 310 doesn’t fall within budget. That gap has
now been filled by the subject of this review: the KH 150.
You can’t really mistake a Neumann monitor for any
other brand, and the KH 150 follows the established
industrial design style closely. In fact, without something to
provide scale in photographs, it’s not easy at first glance
to tell the difference between the KH 150 and the smaller
KH 120. Under the skin, however the KH 150 is significantly Downstream of its not to drive tweeters to break-up
more advanced technically, and offers an even more inputs, and following frequencies above audibility, and that
ambitious level of electro-acoustic performance. 48kHz/24bit A-D means there’s little point in an internal
conversion (for the sampling rate that would enable audio
The Drivers analogue inputs), the bandwidth to extend much above the
The KH 150 is a relatively compact two-way monitor KH 150 incorporates high-frequency limit of human hearing.
of comfortable nearfield monitoring dimensions, and amplification rated at 145 The KH 150 uses a new and
it combines a nominally 165mm bass/midrange driver Watts for the bass/mid patent-pending amplifier technology that,
with an elliptical-waveguide-loaded 25mm tweeter. The driver and 100 Watts for say Neumann, “combines superior audio
enclosure design incorporates softened edges and the tweeter. The 48kHz performance with the energy efficiency of
a smoothly integrated waveguide that will help minimise internal sampling rate might Class-D”. Further research established that
the response aberrations that can be caused by diffraction sound limiting in today’s the amp topology incorporates some novel
from sharp edges. The enclosure is constructed from world, but Neumann explain techniques that take driver impedance
composite polycarbonate front and rear mouldings with that this is a choice driven into account within the feedback strategy.
a wooden carcass, and finished with a textured paint. by tweeter performance. Fundamentally, the KH 150 amplifier
Dark grey and white options are available. In the context All tweeter diaphragms technology takes advantage of the fact
of multi-channel monitoring environments, mounting reach a frequency where that the architecture of active speakers
facilities are becoming a significant issue, and the KH 150 they will enter a resonant enables the driver and amplifier system to
enclosure usefully incorporates M6 threaded inserts on its break-up mode and even be engineered and optimised holistically,
rear panel for this purpose. if, as is said to be the case rather than as disconnected entities.
As well as the two colour options, the KH 150 is also with the KH 150 tweeter,
available with two different sets of connection facilities. the break-up frequency is
DSP
The standard KH 150 offers just conventional balanced above audibility, break-up Perhaps of more immediate interest to
analogue and S/PDIF digital inputs, while the KH 150 intermodulation distortion users is what’s between the monitor’s
AES67 offers AoIP (Audio over Internet Protocol) network effects can appear in inputs and its amplifiers, and that’s
connectivity compatible with AES67-compliant protocols the audible band. So it’s a whole load of DSP power aimed at
such as Ravenna and Dante. beneficial, say Neumann, optimising both the KH 150’s inherent

40 January 2023 / www.soundonsound.com

electro-acoustic performance and its
integration with listening-room acoustics.
One example of the former is a bass/
mid driver limiting function that kicks in to
stop diaphragm displacement exceeding
±12mm. This means, say Neumann,
that overall bass/mid driver distortion
is reduced because the usual need to
engineer non-linear mechanical limiting
into the driver suspension is avoided. The
DSP handles fine adjustment of individual
driver responses to ensure no more than
±0.8dB divergence between any two
monitors, and it also facilitates the KH
150’s fourth-order crossover filter slopes
and wide-band phase correction. The
latter ensures, say Neumann, that the KH
150’s phase response remains within ±45
degrees between 105Hz and 16kHz.
So the KH 150 is very much a monitor
of the DSP age and while, as I’ll describe
further down, there’s much also to write
about in its traditional electro-acoustics, it
wouldn’t be half the monitor it is without
those audio digits getting processed.
When it comes to user-accessible
elements of the KH 150’s DSP, there are
effectively two modes of operation. The
first is local control, in which the KH 150
operates as a conventional active monitor
with a variety of rear-panel switches
adjusting input selection, ground lift, local
or network control, input sensitivity and
gain, and three EQ bands. The EQ offers
0dB to -4dB of low-frequency adjustment,
0dB to -6dB of low midrange adjustment
(centred around above 200Hz),
and +1dB to -2dB of high-frequency
adjustment. However, the KH 150 is also
fully integrated with Neumann’s MA
1 monitor alignment technology, and
the second mode of operation sees
MA 1 deployed to enable both TCP/IP
network-based monitor configuration and
measurement-driven room optimisation.
The MA 1 package comprises
an individually calibrated
measurement microphone and
a Mac/Windows-compatible analysis and
control application. The measurement
microphone, as might be expected from
Neumann, exudes a sense of engineered
precision, and the app looks suitably
professional and inspires confidence. The
app is also intuitive in use and demands
very little in terms of time and effort. What
it does demand, however, is that, while
the optimisation process is underway, the
monitors are connected to the same TCP/IP
network as the computer to be used. In
most cases this will be the DAW computer,
In local control mode, the KH 150 operates as a conventional active monitor, with rear-panel switches
offering control of the input, ground lift, input sensitivity, gain, and three-band EQ.
but it doesn’t need to be, because once
the optimisation process is complete,
the optimisation data is uploaded to the
monitors so they can be disconnected from
the network. If you subsequently need to
run the monitors in un-optimised mode,
a flick of the KH 150 rear panel Network/
Local switch will do the trick. Having
mentioned a few paragraphs back that
the KH 150s bracket mounting facilities
suit the monitor for use in multi-channel
systems, that’s not yet the case for MA 1,
which at present is stereo-only. I imagine
a multi-channel version is on the way.
As with conceptually similar products
such as Sonarworks, ARC, Trinnov, Dirac
and Genelec GLM, the calibration process
relies of a series of monitor measurements,
seven sine-wave sweeps in this case,
captured using the measurement mic
in a variety of positions, focused on the
primarily listening position. MA 1 derives
from the measurements the characteristic
acoustic signature of the environment,
along with how the monitors interact
with it, and from that analysis constructs
equalisation data that aims to optimise the
in-room listening position performance.
Interestingly, MA 1 doesn’t simply aim for
a ‘flat’ response, but creates a target curve
that its algorithms propose will best suit the
specific monitors and room.
I’ve included a couple of MA 1
screenshots that illustrate the process.
Screen 1 shows an MA 1 composite
measured response of the pair of KH 150s
in my room, and Screen 2 shows the result
of the optimisation. You’ll see at the bottom
of the screens that MA 1 also provides the
facility for EQ modifications through 10 (two
shelf and eight parametric) user-accessible
filter stages, so if you want to tweak the
optimised MA 1 response and upload that to
the monitors you’re perfectly able to do so.
Now, you might expect that, in
optimising the performance of a stereo
pair of monitors, MA 1 would display the
pre- and post-optimisation data for the
two monitors individually, but it doesn’t: it
displays a summed response curve. MA 1
works like this because although it analyses
and generates optimisation data for each
monitor individually, it needs to take into
account that if the optimisation data for
each monitor is significantly different, the
resulting stereo image quality will suffer.
This is because it is direct sound reaching
the listener’s ears that is primarily the
source of stereo image information, and
the fundamental modus operandi of MA
Neumann KH 150
€3300
PROS
• Remarkable bandwidth and volume
level from a compact monitor.
• Utterly neutral and uncoloured tonal
balance.
• Brilliant stereo imaging and mix
clarity.
CONS
• MA 1 is stereo-only at present.
SUMMARY
The KH 150 and MA 1 is a fabulous
example of what can be achieved with
the best of advanced electro-acoustics
and DSP. It comes with the sound of
a standard being raised.
www.soundonsound.com / January 2023 41
 ON TE ST
N E U M A N N K H 15 0

Screen 1: The combined in-room

response of a pair of KH 150 monitors,
as measured by the MA 1 software.

1 (and the other conceptually

similar systems I’ve mentioned)
is to equalise the direct sound
with the optimisation data
in order that, when room
effects are ‘printed’ on it,
the resulting composite is
tonally optimal. But if each of
a stereo pair of monitors were
to need significantly different
equalisation because, for
example, they were positioned
in acoustically different
environments, the stereo
imaging could be compromised.
It’s for this reason also that
the user EQ offered by MA
1 can only be applied to the
summed system response. If
users were able to equalise
the two monitors of a stereo
pair individually, stereo imaging
would again be at risk.

Crossing Over
I mentioned the monitor’s
fourth-order crossover filter
slopes a few paragraphs up.
The crossover frequency is at
1700Hz, and that’s not far short
of an octave lower than is more
traditionally found in similarly
configured two-way monitors.
The low crossover frequency is
made possible by the sensitivity
gains that result from the tweeter
waveguide enabling its usable
bandwidth to be extended downward.
That in turn also enables better directivity
matching between the two drivers around
the crossover frequency than is typically
found when crossover frequencies are
higher and tweeters are used ‘naked’.
A similar technique can be found on, for
example, the Amphion range of monitors.
The KH 150 tweeter is a 25mm dome
unit that’s closely related to the drivers
found in other Neumann monitors. It
benefits from Neumann’s proprietary
Mathematically Modelled Waveguide and
Extreme Linear Force Factor technologies,
and incorporates what Neumann describe
as an “alloy fabric dome”, in an apparently
deliberate move to be unspecific about
the actual material. The KH 150 bass/
mid driver is a newly developed unit that
Neumann say was specifically engineered
to offer a significant improvement in
distortion, low-frequency bandwidth
and maximum level. In particular, the
driver’s motor system is designed to offer
extreme linearity and low distortion over
extended diaphragm displacement. By
way of illustration, a typical off-the shelf
165mm bass/mid driver will offer between
±6mm and ±8mm of linear diaphragm
travel. The KH 150 driver, say Neumann,
offers ±12mm. This was achieved through
intense FEA modelling of the magnet
system and diaphragm suspension
components, and the use of a variable
voice-coil winding profile technique. The
latter is particularly unusual. Forward of
the voice coil and magnet system, the
KH 150 bass/mid driver employs a paper
diaphragm and dust cap, paired with
a rubber roll surround. The relatively low
Screen 2: The MA 1-corrected response.
crossover frequency will help with bass/
mid diaphragm design and optimisation,
because it makes the diaphragm break-up
characteristics and surround termination
characteristics a little less critical. Even so,
Neumann say that the particular demands
placed on the surround by the driver’s
very high linear displacement make its
optimisation extremely complex.
Finally on the bass/mid driver, its
custom-designed die-cast aluminium
chassis forms a structural element of the
enclosure front panel — recovering the
structural panel weakness that results from
the big hole the driver demands.
Pass The Port
Beneath and either side of the KH 150
bass/mid driver are a couple of reflex port

42 January 2023 / www.soundonsound.com

 exits. They are triangular in form and flared
to help retain laminar airflow to as high
a volume level as possible. The KH 150
reflex tuning frequency is at a relatively
low 42Hz, and the length of port required
to achieve that means the ports tubes
internally bend upward inside the cabinet.
Neumann development engineers say that
much time, effort and simulation work was
carried out on the ports to ensure both
laminar airflow and minimal organ-pipe
resonance effects.
Considering the reflex loading and Diagram 1: The KH 150’s measured on-axis frequency response (red trace), and second- and
bass/mid driver together, the resulting third-harmonic distortion levels (green and blue, respectively).
low-frequency bandwidth specification
is -3dB at 39Hz, with a maximum One related area where the KH significance of low-frequency group
low-frequency (50Hz to 100Hz) level for 150 and KH 310 fundamentally differ, delay and many, including me, believe
3% total harmonic distortion of 107.5dB. however, is in their low-frequency it to be potentially audible. However,
These are impressive numbers for time-domain performance. The KH 310, the situation is far less black and white
a relatively small nearfield monitor, and due to its closed-box loading technique, than raw specification numbers might
to add some context, the equivalent demonstrates low-frequency group suggest, and all sorts of other factors play
specifications for the much larger delay (think of group delay as frequency- a role in the subjective low-frequency
Neumann KH 310, with its 210mm bass dependent latency) of just 12ms at 40Hz, character of monitors, the choices monitor
driver, are 34Hz and 106.8dB respectively. whereas the KH 150’s group delay designers make, and the needs of monitor
The KH 310 offers 5Hz extra bandwidth, reaches 27ms at a similar frequency. As manufacturers to make products that
but is actually 0.7dB shy of the KH 150 in I’ve written on quite a few occasions in work across different professional sectors.
terms of maximum level. the past, there’s much debate on the There’s also a lot more to subjective

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 ON TE ST
N E U M A N N K H 15 0
Diagram 2: The KH 150’s frequency response measured on-axis (red trace), and 30 degrees above and
below (blue and green, respectively).
Diagram 3: The axial frequency response (red),
overlaid with the MA 1’s equalised response, as
measured in the free field.
low-frequency performance than group
delay: for example, low distortion and
high volume levels at low frequencies,
especially when driver and enclosure size
is limited, are generally easier to achieve
with reflex loading. And while group delay
defines how promptly a monitor is able
to ‘accelerate’, it says nothing about what
happens when it needs to stop (that’s all to
do with port and driver Q).
Port dynamics are also of vital
significance in defining how well
a reflex-loaded monitor is able to play
bass. Poorly designed ports that become
non-linear and compress at relatively low
volume levels, or demonstrate organ-pipe
resonances that result in the radiation
of significant midrange energy, can give
ported monitors a worse name than
perhaps they deserve. Considering the
low-frequency bandwidth and maximum
level requirements defined by the very
broad range of potential roles it needs to
satisfy, I’m sure Neumann had little option
but to choose reflex loading with the KH
150s. The issue then, as with any other
aspect of monitor design, was to find the
combined set of engineering parameters
that result in optimal performance. And
speaking of performance, as is traditional,
I took a KH 150 along to my big measuring
44 January 2023 / www.soundonsound.com
space (a church, now you’ve asked) and
fired up FuzzMeasure to investigate a few
elements of objective performance, and
this is what I found...
Taking Measurements
Diagram 1 illustrates the KH 150 frequency
response and harmonic distortion
(second and third harmonics). Sound
pressure level was 90dB (at 1m) and the
data is accurate down to around 100Hz.
The measuring mic was 1.5m away and
positioned on an axis midway between
the bass/mid driver and tweeter. The
resulting KH 150 frequency response is
probably the flattest I’ve ever measured
on a commercially available speaker
and, while I’m not one to get too excited
about a single response curve (monitors
are much more than a single axial
frequency response curve), I will admit
Diagram 4: The KH 150’s in-room response before and after MA 1 optimisation (blue and green traces).
to a mild double take and a quiet “wow!”
(I was in a church, remember) when the
curve appeared on the laptop screen.
On the particular axis I measured, the
frequency response is flat to within ±1dB
from 100Hz to 20kHz. The distortion
performance revealed in Diagram 1 is
similarly notable. The second harmonic
is typically at -50 to -55 dB, and the third
harmonic is typically another 5 to 10 dB
lower. For context, -50dB equates to 0.3%
and -60dB equates to 0.1% distortion, so
these are seriously impressive figures.
Diagram 2 illustrates the KH 150’s
vertical dispersion at 30° off-axis upwards
and downwards. As expected there’s
a crossover interference dip where the
path lengths from the two drivers go
out of phase, but it’s over quickly and is
remarkably symmetrical up and down. This
suggests extremely well-managed driver
integration through the crossover region.
The off-axis curves also reveal a gentle
reduction in tweeter level without any
sudden discontinuities.
Diagram 3 illustrates the same axial
response of Diagram 1 overlaid with
the result of MA 1 optimisation in my
room. This shows how MA 1 changes the
frequency response of the KH 150 once it
has the room optimisation data uploaded.
Finally, Diagram 4 illustrates in-room
measurements of the pair of KH 150s with
and without room optimisation engaged.
Along with the illustrated measurements,
I made some others, to check horizontal
dispersion and to look for port organ-pipe
resonances, for example. The KH 150
threw up nothing of concern.
Listening Tests
So a little tyre-kicking analysis of
the KH 150 suggests an extremely
well-engineered monitor that packs a lot
of electro-acoustic and DSP expertise
into a conveniently dimensioned and, in
high-performance monitor terms, relatively
affordable package. But what does the
 ALTERNATIVES
I’ve given the KH 150 and MA 1 a very positive review, but of

THE NEW
course there are other very fine monitors in its price range.
The PS1 A17M, the Genelec 8350A, the Focal ST6 Solo 6
and the Dynaudio Core 7 are four such examples.

RS660
KH 150 sound like? As a confirmed admirer of Neumann’s
existing monitors I had high hopes for the KH 150 and MA
1, and I wasn’t disappointed. I’ve left myself not very many
words to describe my subjective thoughts, but in some
respects that’s fine, because there’s not actually much
to say other than that the KH 150 and MA 1 is as near to
flawless as I’ve heard compact nearfield monitoring get.
I began with the KH 150 working in local mode without
the benefit of MA 1 room optimisation, and the rear-panel
low-frequency EQ switch at -2dB, to suit my room. From the
start, the flatness of the KH 150’s frequency response, its
well-managed dispersion, and its very low distortion, wide
bandwidth and serious volume level capabilities combine to
create a powerfully analytic monitoring tool. The KH 150 is
deeply insightful through the entire audio band with stereo
imaging that’s rock solid, as much mix clarity as you could
ever want to hear, and a tonal character that’s authentic,
trustworthy and apparently unchanging whatever the volume
level. KH 150 bass is unfeasibly extended considering the
monitor’s relatively compact dimensions, yet it sounds
dynamic and dependable with great subjective pitch and
dynamic accuracy. I would never have guessed its 27ms

REDEFINED
group delay.
Adding the MA 1 to the KH 150 mix, for me, wasn’t
a massive game-changer, not least because the game is
already so good, but it did sharpen the stereo imaging still
further, help reveal yet more mix detail and significantly ELEGANT
COMPRESSION
reduce the tonal balance and coloration of room effects. The
benefits of MA 1 will always be somewhat context-dependent,
so in an acoustically more challenging installation it will

CIRCUIT.
likely enable the qualities of the KH 150 to shine though
when otherwise they might struggle to. Having said that,
considering the relatively modest cost of adding the MA 1
kit to a pair of KH 150s, you’d be crazy not to employ it, and
certainly, once I’d configured MA 1, I didn’t go back to using
the KH 150s without its optimisation engaged.
But along with performing the nearfield role all but
faultlessly, there’s more to the KH 150 and MA 1, because
on the right material I also found listening to be hugely “Wade’s done it again, right out of the
rewarding. If truth be known, I pretty quickly stopped box, the RS660 sounds like everything
listening analytically to try and get a handle on such things
I look for in a vintage tube compressor,
as coloration and tonal balance, because they were so
clearly on point, and started listening just for the simple fun without having to worry if it’s going to
of discovering new insights into material I thought I knew breakdown on a session.“
intimately. For example, there’s a bass part in Joe Jackson’s
‘You Can’t Get What You Want’ from the Body And Soul
— DAVE COBB
album that I simply hadn’t noticed before, and I’ve been
listening to that track since the mid-1980s. Monitors that
enable that kind of
revelation are few
£ KH 150 €3300, KH 150 AES67 €3790.
and far between, Find a dealer @ nova-distribution.co.uk
Prices are per pair, including VAT.
T Sennheiser UK +44 (0)1628 402200 so the KH 150
W www.sennheiser.com joins a very select
W www.neumann.com bunch.  

www.so onsound.com / January 2023 45

sales@nova-distribution.co.uk
TEL: 0203 150 4446
 INSIDE TRACK

Leandro ‘Dro’ Hidalgo

With hit albums for Dave,
Wizkid and now Stormzy,
Brazilian-American mixer
Leandro Hidalgo is at the
top of his game.
PAUL TINGEN

“I
always say: ‘If you want to find out
how to be an amazing assistant,
be an engineer for a day, and have
an assistant.’ You will know everything you
want that assistant to do. When you go back
to being an assistant, you will be handing
the engineer things before he even thinks
of them.
“Similarly, if you want to learn how to be
a great mixer, be a producer or an artist for
a day, and you will know what they want from
a mixer. Most mixers view their job purely from
a mixer standpoint. But I’m also looking at it
from an artist/producer standpoint, because
I have played all roles. I record, I write demos,
I produce, and I mix. I have done it all, A to Z.”
Leandro ‘Dro’ Hidalgo’s multi-disciplinary
approach to his work is in evidence on his
Instagram account, where he still bills himself
as a “musician/band”. But his credits as
a drummer, engineer, writer and producer
notwithstanding, it is his work as a mixer
that has in recent years attracted the most
attention. In 2020, Hidalgo mixed almost all
of Made In Lagos, the breakthrough album in
the west of Nigerian singer Wizkid, who has
already been a household name in Africa for
many years.
Next up, Hidalgo got to mix the whole of
Dave’s most recent album, We’re All Alone
In This Together (2021). And at the end of
2022 Wizkid’s More Love Less Ego was Leandro ‘Dro’ Hidalgo.

46 January 2023 / www.soundonsound.com

released, entirely mixed by Hidalgo, along
with Stormzy’s third studio album, This Is
What I Mean. Hidalgo mixed seven of the
album’s 12 songs, including the lead single,
‘Hide & Seek’ and the epic eight-minute
long album opener ‘Fire + Water’.
In The Rough
Perhaps because of his experience as
a producer, Hidalgo feels no need to
stamp his own mark on his mixes. “I love it
when people say, ‘Make it sound like the
rough,’ because it is a challenge. Many
people I know hate that statement, but
I think it is awesome. To make a song
sound better, and still keep the magic of
where it was, is an art. It is harder to do as
a mixer than to change a bunch of stuff,
and make it sound different.
“My job as a mixer is to keep the magic
alive. Again, this comes from me recording
and producing. I have been in the room
when this magic was created. There’s this
energy bubble, which is like a forcefield,
and you record it into this little digital
thing called a computer. It is alive. You
can almost feel the energy that was in the
room. And then you give it to a mixer, and
sometimes it dies completely. The energy
is done. You can kill that magic easily.
“People complain that ‘make it sound
like the rough’ is about demo-itis, and
that may be the case. But not always.
Producers know how to make the music
feel, and that is what it is about. All these
people that geek out over plug-ins and
all this other stuff, it is cool. I used to do
the same. But I am so much more focused
now on how the song makes me feel than
on doing this little notch where some
frequencies are clashing. Who cares.
Sometimes that stuff is what makes it
sound good!”
Love From Brazil
Hidalgo appears to have a strong
connection with UK-based artists, because
in addition to Dave and Stormzy, he’s
also worked with Skepta and KSI. The
common thread here is the influence
of African music, to which Hidalgo has
taken a particular liking. He is one of the
go-to mixers in the US when it comes
to Afrobeat, something which is directly
related to his Brazilian background.
“I was born in Brazil, and my family
moved to Miami when I was five,” explains
Hidalgo. “Music and dancing are central
to Brazilian culture, so I was around
that a lot, and at middle school I started
playing drums. I also got interested in
music production — I started reading SOS
in great detail! — and went to Full Sail
in 2011-12. After that I interned at Alexus
Records, with Rico Love. He has a group of
producers called Division One, and they’d
make beats on their computers every day,
with me being the floating person in the
room. I learned a lot from them and Rico.
I’m sure they had no idea that I spent a lot
of time taking notes!
“I moved to LA in 2014, and worked
at the Record Plant, where Rose
Mann-Cherney gave me a chance to work
as an intern, and this led to me meeting
top mixer Dave Pensado. I worked with
him for nearly two years [Pensado kicked
off the Inside Track series in January 2007:
www.soundonsound.com/techniques/sec
rets-mix-engineers-david-pensado]. After
that I worked for most of one and a half
years with producer Brody Brown. He was
one of the main writers on Bruno Mars’
24K Magic album [2016]. I learned a lot
about mixing and music production from
both Dave and Brody.”
Four In One
Hidalgo first met Wizkid in 2015, when the
singer was in LA working on his major-label
debut Sounds From The Other Side. It
woke Hidalgo up to the parallels between
Afrobeat and his Brazilian background.
“After meeting Wiz, I learned a lot about
my own culture that I did not even know.
How much African influence there is in our
culture, and how much of our culture stems
from specifically the Yoruba tribe, the main
tribe in Nigeria. All of it made sense to me.
“However, when I listened to early
Afrobeat, the music felt amazing, but
sounded really bad. They were recording
on terrible equipment, and doing things
they were not supposed to be doing. At
the same time, when you do something
really wrong it can become right. For
example, Wiz’s vocals tended to be
super over-compressed. When I started
working with him, if I did not compress his
vocals that hard, he would be like: ‘Dude,
something is off.’ So I had to learn how to
give him what he was looking for, while at
the same time improving the sonics. I read
that mixer Seth Firkins wanted to take
Future’s music to a place where it could
sonically compete with the mainstream.
I felt the same with Wiz. My idea was
to take this amazing music and make it
compete with the likes of Dua Lipa and
Beyoncé and so on.”
Made In Lagos was nominated
for a Grammy, and made it to many
‘Hide + Seek’
Written by Ayanna Christie Brown,
Ras Kassa Alexander, Andrew Jason
Brown, Michael Ebenezer Kwadjo
Omari Owuo Jr, Owen Cutts, Richard
Olowaranti Mbu Isong &
Temilade Openiyi
Produced by P2J, PRGRSHN &
Owen Cutts
lists of best albums of 2020. And, as
usually happens, one good thing led to
another. “We were mixing Wiz’s album in
Amsterdam, and when P2J came along as
executive producer, it changed the album
completely. P2J then introduced me to
Dave, because he worked on his album
as well. I did one song for Dave, and they
liked it and kept sending me more stuff to
mix. It ended up with me mixing the entire
album. P2J also did a song for Stormzy,
and with me already having mixed Dave’s
album, it made it easier for Stormzy’s team
to trust me.
“The first song I mixed for Stormzy was
given to me as a test, and if I passed it,
it’d unlock me mixing many other songs
as well. The song was ‘Fire + Water’, and
at eight minutes long, it was definitely
a challenge! For a mixer, a song of that
length is complicated. It’s essentially four
songs rolled into one, and how do you mix
a track that has screaming lead guitar in
one section, and almost turns into Afrobeat
at the end, and yet doesn’t? How do you
get that to sonically make sense? How do
you do the transitions, and how do you
send all that through one mix bus? In fact,
I ended up using two mix busses.”
Play Your Cards Right
At the time of writing, Stormzy’s new
album was still under wraps, but it’s clear
that the rapper has changed musical
direction. The mood is introspective,
Stormzy is singing more than rapping, the
arrangements are spacious and delicate
www.soundonsound.com / January 2023 47
 INSIDE TRACK
LE A N DRO ‘ DRO ’ HIDA LGO • S TOR M Z Y

and made to sound like a real band, and

the production is delicate and gorgeous.
According to Hidalgo, the entire album
takes a similar direction.
“Everything I have worked on for the
album was very musical, very progressive.
One of my favourite old songs of Stormzy
is ‘Wiley Flow’, and it’s very dramatic with
intense rapping, but this new album is so
different. It is all about beautiful melodies,
beautiful sonics, tons of space, a lot of
musicianship and so on. I was listening
to the tracks as they came in, and I was
amazed. And the instructions I got started
with, ‘Our goal is to make this album as
musical and sonically pleasing as possible.
We don’t care about hard or loud.’ The
other thing they said was, ‘We really like
where we are, it just needs to be mixed.’
In a way it is saying: here is this house of
cards that we built, go make it better. But
if you build a house of cards and you take
one card out of position, the entire house
is coming down. So you have to be very
careful of what you do.
“Working on this album was very
different from most of the albums that
I work on in several respects. I normally
have direct communication with the artist
and the producers, but with this album
I only dealt with his label people and two
engineers, Joel Peters and Calum Landau.
There was a big group of producers that
had contributed, so I think it was easier to
funnel everything down to the engineers.

Generally I prefer to speak with the artist,
especially when you are receiving notes
that are not in technical terms. When
you get notes like: ‘Hey can you make
it sweeter?’ I like to talk with the artist,
because something that he or she will tell
me outside of that sentence is going to
provide some kind of contextual value that
will lead me into the right direction. But
luckily Cal sent me very detailed notes that
made it easy for me to get to the end goal,
which is to make the artist happy and make
the song sound as best as it can be.”
Getting It Together
“My perspective as a mixer is that I begin
where the tracking engineer signed off. My
starting point is the rough mix. Initially I was
sent stems, but when I loaded them into
a Pro Tools session and compared that to
the demo, and A/B’ed the two, I thought,
‘No way’. They were not the same. Things
were not there. So I asked for the Pro
Tools sessions, and after that it was pretty
smooth sailing, because I could carry on
from where they had left off.
Stormzy (standing) with engineers Calum Landau (left) and Joel Peters (far right), executive producer
Tendai Senyange (centre), #Merky Records A&R Jermaine Agyako and producer PRGRSHN (front).
“However, the sessions had many
plug-ins, more than I would normally
use. This whole album was one of those
were I had to do a lot of work with what
was there, which I don’t necessarily
do normally. It’s why this album was
a challenge, and also really rewarding.
Also, in some cases I could see from
the track names that they had printed
effects, and then added plug-ins on top
of that. Clearly, they had committed quite
a number of things because it’s how they
liked them, so my job was to enhance
this, not to go all the way back and then
recreate a whole different thing.
“The scenario I was in actually took
longer, because I have to experiment a lot
more, and make sure I don’t destabilise
the house of cards that they built. I might
take a couple of their plug-ins off to hear
how drastic they work, and put them back
on, or try a couple of things myself that
Photo: Zebie
I don’t usually do. In all it would take me
maybe eight to 10 hours per session to
get the whole thing together, though
I don’t work on one session all the time.
My brain starts getting fogged up, and
I don’t really know where my reference
point is any more. So I jump between
sessions a lot. I might work on five
different songs in one day.
“I also jump around the session itself
when I mix. I am definitely not systematic
at all. I am doing whatever I hear needs
doing. Sometimes the drums need
the most work, sometimes the music,
sometimes the vocals need. Usually, the
vocals need the most work. Producers at
this level know how to make the drums
smack and music sound good, so I don’t
have to spend as much time on that.
Generally speaking the vocal is what
ties the whole thing together, so I focus
on that, and amplify the music and the

48 January 2023 / www.soundonsound.com

 The Importance Of References
“What often happens is that people are on
a wave, have worked for the whole day, and
probably have not heard an outside song for
hours, so they’re forgetting to reference,”
says Leandro Hidalgo. And, once again,
this experience as a producer helps inform
his mixing process. “I do that all the time.
During mixing I have Magic AB on my mix
bus, and I go back and forth between songs.
I play DJ. I close my eyes and put on songs
that if I was a DJ, I’d play next to the song
that I’m currently working on. When my song
comes on, the energy should be better, and
it needs to feel better. It needs to have a lift
and be noticeably different.
“For this song my references were Dua
Lipa’s ‘Levitating’, Drake’s ‘Controlla’, and
J Balvin, Bad Bunny and Mr Eazi’s ‘Como Un
Bebé’. The references help me recentre my
mind when I’m deep in the mixes. I know all Leandro Hidalgo inherited lengthy plug-in chains on many tracks, and added more of his own, but often
three songs very well and have heard them set things so that individual plug-ins were doing relatively little. The Oeksound Soothe plug-in on Stormzy’s
around the world in many different listening vocal is a good example: the Mix is set to just 10 percent.
environments. I also often use DJ Khaled’s
‘Wild Thoughts’, Burna Boy’s ‘Anybody’ and and one aux effect track; the tracks of the Neoverb, but they were happy with it, so
my mix of Wizkid’s new song ‘Wow’.
other vocalists Ayanna, Oxlade, Teni and I was not going to change it. Other vocal
Owen Cutts; the drums, with 23 audio aux effect tracks had the Valhalla Vintage
drums a little bit. They don’t tend to need tracks and three aux tracks; and the music, Verb and Valhalla Room, ModDelay 3, and
such heavy lifting. But again, every song with 16 audio tracks and two aux tracks, I added the Voxcom aux, with the Waves
is different.” which include guitar by Jack Shepherd and CLA-76 compressor and FabFilter Pro-L 1
keyboards by PRGRSHN. The song was limiter for compression.
‘Hide & Seek’ produced by the latter, along with Owen “My colour coding is male vocals in
Hidalgo’s mix session for ‘Hide & Seek’ Cutts, P2J and Calum Landau. bright blue, female vocals in purple or
contains about 90 tracks. From top to Hidalgo: “This is fairly typical for the pink, aux group tracks yellow, bass in
bottom in the Edit window, these are way I organise my sessions. I like to have green, drums in red, and instruments
a master section of three tracks, fed by my master tracks with the mix bus at the in beige. This sometimes gets messed
All Vox and All Beats group tracks; six aux top, followed by aux vocal effects, in green. up when I’m sent new vocals while I’m
vocal effects tracks; Stormzy’s vocal tracks, All but one of these aux tracks came with mixing, and when this happens, I’ll give
consisting of audio tracks, group aux tracks the session. I’d never used the iZotope them a strange colour, like lime green

SYNTHESIZER
+ GROOVEBOX

= ALL YOU NEED

16 voices
4 Timbre + FX section each
Algorithmic oscillator
Wavetable oscillator
Noise sample oscillator
Drum instruments
Arpeggiator & step sequencer
Clip launcher
 INSIDE TRACK
LE A N DRO ‘ DRO ’ HIDA LGO • S TOR M Z Y

or something, so I can recognise them

easily. Vocal samples tend to live at the
bottom of the session, amongst the music
elements, because they tend to be part of
the instrumental.
“As I mentioned earlier, I don’t mix in
a particular order, like starting with the
drums. Because of the house of cards
situation, I had to comb through everything
in this session, and make sure everything
had its place. I added some of my own
plug-ins to the many plug-ins that were
already there, but in many cases I worked
with the existing plug-ins, and my work
consisted of getting the best out of them.
This involved a lot of backing things off,
because of the brief to make it sound
natural and beautiful.”

Vocals
Stormzy’s vocals are sent to two
self-explanatory aux group tracks, Stormzy
Rap LV and Stormzy Sing LV, and there’s an
additional SIDE aux effects track with the
Waves Doubler 2 and S1 Imager plug-ins.
“The Purple Audio Alliance MC77 and
ProCompressor came with the session,
and I adjusted the settings. I’d rather use
the UAD 1176, but these worked pretty
good as well. There’s an Oeksound Soothe
2, and if you look at the settings, the main
thing is easy to miss, which is that the mix
is set to 10 percent. It’s so important to
turn that mix knob down on that plug-in,
because it can just as easily ruin the
vocal as improve it. People overhype this
plug-in a little bit, but it’s very powerful,
if used properly. People think, ‘Just put it
on’ and that’s it, and have the mix knob
at 100 percent. But when you do that, it’s
reshaping the vocal and making it sound
like someone else. But the goal is to make
it sound like the vocalist. There’s also
a FabFilter Pro-Q 3, with some small dips,
which are part of the repair work I did.
“There’s an Ayanna Vocal New bus,
with a ProCompressor and the Valhalla
VintageVerb, and another similar track,
Ayana LV1, with tons of plug-ins, including
the Waves X-Noise, iZotope Ozone 9
Exciter, and many more. Again, when it
comes with that many plug-ins, all I usually
did was back off most of what these
plug-ins were doing. There’s another
Ayanna vocal track that has some plug-ins
I added, the UAD LA-2A, FabFilter Pro-Q 3,
Waves C6, Valhalla VintageVerb and
Soundtoys PanMan. Many of them don’t
change the tonal characteristics too much.
“All vocal group busses are sent to
the All Vocal bus, to which I added the
Brainworx’s bx_subsynth v2 was used to fill out the low end on the bass.
Avid Lo-Fi, set to be barely audible, and
the Slate Virtual Mix Rack, only using the
Virtual Channel, to make it more analogue
sounding, but again, barely touching.
I have met mixers who did less, with
fewer plug-ins, and they made their stuff
sound so good. The common theme is
that these mixers get the most out of the
few plug-ins that they were using, versus
adding 10 plug-ins and trying to figure out
what to do with them.”
Music
“The plug-ins on the bass were also
in part already there. I inserted the
Brainworx bx_subsynth. I felt that the bass
was lacking a little bit in the sub region,
so I added some low-end harmonics.
I turned down the subharmonics knob
from 50 to 40 and then set the 24-36
Hz knob to 45 percent, and the 36-56
Hz knob to 65 percent. Those numbers
seem a little high, but the thing is that
this plug-in is based on the input level,
which was low with the bass. Also, if you
look on the left, the global mix is down
to 51 percent. I also added the Waves
RCompressor, which I side-chained to
the kick.
“All drums and percussion go to the
All Drums bus, to which I added the SSL
Native Drumstrip. The Drive in the Low
Frequency Enhancer section is set to
high, but the mix is set to very dry. It’s
a similar situation with the High Frequency
Enhancer, where there is some Drive
added, but the Amount is really low.
Overall, the plug-in gave the drums some
nice sheen on the high end, and a tiny
little bit of poke for the kick drum around
150Hz, but it is not like a gigantic jump.
“There’s a huge difference between
plug-ins for production and plug-ins for
mixing. The Avid EQ3 7-band on the
Fender Rhodes, for instance, is set pretty
drastic. That is a production thing. In a mix
you would never add 5.9dB of a shelf from
1.29kHz all the way up. So the production
team did that. I added a side-chain using
the Waves RCompressor, keyed to the kick,
because this Rhodes has a lot of lower
mids, which clashes with the upper part of
the kick drum.
“All the instruments are sent to the All
Music bus, to which I added the Brainworx
bx_saturator v2, which is a Mid-Sides
saturation plug-in. It gives a little bit of extra
width, to whatever is going on in the music,

50 January 2023 / www.soundonsound.com

primarily in the saturation. Again, only by
a tiny bit. On the Pro-Q 3 I am dipping
mids, like 0.89Hz at 280Hz and 1dB at just
above 6kHz. I’m clearing a tiny little bit of
space for vocals to sit on top of the music.”
Mix Bus
“The mix bus is entirely mine. I add it fairly
early when I am mixing, and it then evolves
with the mix. When I put the plug-ins on the
mix bus, they are usually zeroed, and I then
start to adjust them as necessary. I bring
my mixes up to competitive level in terms
of loudness with the mix bus, and I check
that with the iZotope Insight on the master
track. I don’t expect a mastering engineer
to make my stuff louder. On the contrary,
I ask them to keep it pretty similar.
“My mix bus chain starts with the
UAD Neve 1073, which bumps up the
volume, because I like to have a healthy
signal going into my mastering bus. The
Soundtheory Gulfoss is barely touching it,
I have it on 4 percent. The API 2500 allows
me to increase the level further without
crunching anything. The Pultec EQP-1A
adds a dB of volume, and my final step
Studio Dro
Leandro Hidalgo mixes at his private studio in
Los Angeles. “I’m in the process of remodelling
my studio, with acoustic panels being replaced
and the floor and the roof being redone. With
regards to gear, I’m all in the box, no outboard,
though I have a [Sequential] Prophet X synth.
I run a Hackintosh with Pro Tools, and have
a [Apogee] Symphony I/O Mk2, the black one.
I also have an old silver [Universal Audio] Apollo
Twin, and a UAD-2 Satellite, which I think is a 4,
and my Twin is a 2, so I have six cards of DSP
for UAD.
is the iZotope Ozone 9 Maximizer. I also
use the dynamic EQ in Ozone, to soften
some frequencies that might be hitting the
Maximizer too aggressively. It just helps me
get more juice out of the Maximizer without
things sounding squashed.
“Once I’m done, the mastering engineer
can add value by having a perspective
of the overall album, and to keep the
cohesiveness of the songs, EQ-wise.
Sometimes they know how to give it
a nice little warm hug. I have a couple of
“Nine months ago I bought PMC 8-2 monitors.
I used to have the Focal Twin 6 Be monitors.
It’s what I mixed Made In Lagos on. I did the
Wizkid and all the Stormzy stuff on the PMCs.
I never enjoyed listening to PMCs until their 8-2
and 6-2 monitors came out. They’re the new
cool speakers to have now, but I have heard
more exciting speakers, with more character. So
they’re not the most fun speakers to listen to, but
for me they are the best tool for the job. I like the
way they make me work. The way they translate
outside the room is perfect.”
guys I really trust, and every time I send
them something, it’s a bit better. It’s like
a safety net. Also, file delivery to labels is
very difficult, because of all the different
platforms. You have to master for iTunes
and Spotify and so on, and you have to
encode it a certain way, and you have to
be able to add IRC codes into all these
files. Mastering engineers are doing a lot
of that stuff. I’m sure their assistants are
doing all that for them, and we’re thankful
for that!”  
 ON TE ST
IK Multimedia
AmpliTube Tonex
Amp Profiling Software
Tonex lets you capture and share the sound of your guitar
amps, pedals and miking setups.
PAUL WHITE
he concept of modelling specific
T amplifier and mic setups by
analysing test signals passed
through them is not a new one: products
such as Kemper’s hardware Profiling Amp
range and Neural DSP’s software Quad
Cortex, for example, have done that for
a while. But IK Multimedia have long
been active in the amp-modelling game,
and they’ve now developed their own
software-based profiling approach, which
integrates with their existing AmpliTube
5 environment. Tonex is available for
both Mac (including M1) and Windows
machines, though Mac users will need
52 January 2023 / www.soundonsound.com
a machine that runs Big Sur or later to
host it (folk like me still hanging onto
their decade-old ‘cheesegrater’ Mac Pros
won’t be able to run it). For this review,
my tests were done on an M1 Mac Mini
running Big Sur.
Overview
Tonex can run stand-alone or as a plug-in,
and within AmpliTube 5 it’s available
alongside the software’s existing
content, which means you can add
AmpliTube’s modelled effects to your own
Tone Models. It’s is a two-part system,
comprising the Modeler, which is used
to capture the Tone Models, and the
Player, from where the Tone Models can
be used, tweaked and also uploaded to
ToneNET (www.tone.net) for the benefit of
other users. There is also a Tonex app for
iOS that can make use of Tone Models on
mobile devices. Connection to ToneNET
is made from within the program, or it can
be accessed via a web browser.
Tonex comes in a number of forms
including a free CS version that includes
20 ready-made Tone Models and 20
user Tone Models, but doesn’t include
the Modeler. Tonex SE comes bundled
with an AXE I/O interface, along with 200
models, or you can opt for Tonex Capture
(essentially Tonex SE plus the Tonex
Capture hardware), or the all-you-can-eat
Tonex Max, which includes 1000 Tone
Models. The supplied Tone Models
include many of the same amplifiers
that you might find in AmpliTube, the
difference being that these have all been
captured from real amps using the Tonex
process as opposed to being the more
traditional models hosted by AmpliTube.
The approach to capturing and
reproducing your sound is underpinned
by machine learning, and Tonex aims to
capture the sound of any amp or cabinet,
either miked or direct from the amplifier
output, the latter option enabling the user
to team their amp with different AmpliTube
 IK Multimedia
AmpliTube Tonex
PROS
• Excellent modelling accuracy.
• Easy to use.
• Integrates with AmpliTube.
• Affordable.

CONS
• Highest-quality capture takes a long
time.

SUMMARY
What goes on behind the scenes
involves some complex and
time-consuming machine-learning
processes, but IK Multimedia have
made Tonex very user friendly and it is
capable of excellent results.

cabinet emulations, of course. For the

capture process, you can run Tonex with
a simple two-in, two-out audio interface,
though the routing may be easier if your
interface has four or more outputs; I used my
PreSonus 192 interface for the review tests.
IK’s AXE I/O would be a good option too,
and has the benefit of a dedicated Amp Out,
while the company’s optional Tonex Capture
box makes the task even easier: it connects
to the output of your usual audio interface
to act as a re-amping box, so as to feed the
test signals into the guitar amp at the right
level and impedance, and its return stage is
designed to be inserted between the amp
and its speaker cabinet so as to capture
the amplifier output (but not the speaker)
at a suitable level for recording, without
changing the way the amp and speaker
interact. (Note that Tonex Capture isn’t
a load box, so a speaker or dummy load still
needs to be connected in this scenario). The software walks you through setting things up, including focusing the machine-learning process on
If you prefer to capture the whole the sort of setup you’re trying to capture, as well as providing a helpful routing diagram.
amp/speaker chain, using your favourite
cabinet mic and miking position, you can the situation is more nuanced than the but choose Modeler and you see
do that too. The process can also capture manual suggests: “Although we say that a step-by-step path to creating your own
the sound of some types of effects pedal, Tonex can’t model compressors, that’s Tone Model. Along the way, you have
though not all. For example, fuzz, distortion, because it can’t model every compressor. to tell the system whether you plan to
overdrive and EQ can be captured as Tone Tonex can model compression within use a guitar or a bass, what type of amp/
Models, either individually or as part of a limited time frame, so power supply pedal setup you are using, what audio
a longer signal chain including the amp, sag and fast compressors are usually interface you are using and what inputs
but time-variant effects such as echo, well captured.” And apparently it’s not and outputs are being used for the cab
reverb, granular weirdness, expansion and just the amp’s characteristics that they’ve mic, your guitar and the feed to the guitar
modulation can’t be replicated. (Once in managed to capture either: “This can be amplifier. You can use one or two mics for
the software domain, of course, there are heard in some Tone Models that were miking your cabinet.
plenty of other plug-ins you can turn to for captured through a fast 1176 compressor The feed to the amplifier should ideally
that sort of thing.) and is especially evident for some of the be sent through a re-amp box of some
The manual states that the process fuzzy Tone Models that seem to ‘pump’ type, though it’s worth mentioning that
isn’t suited to modelling compressors, with strong input signals.” I managed to achieve decent results
but since there’s inherently compression simply by using a passive volume pedal
in the sound of guitar amps (for example
The Capture Process to trim the interface’s line-level output
power-supply sag) I asked the developers Hitting the Player tab puts you in familiar down to guitar level. Tonex holds your
how Tonex deals with that, and it seems AmpliTube-style amp browser territory, hand by providing concise instructions as

www.soundonsound.com / January 2023 53

 ON TE ST
I K M U LT I M E D I A A M P L I T U B E T O N E X

The capture process can take

a while on the highest-quality
settings, but the ‘lighter’
processes can get you in the
right ballpark quickly before you
commit to a ‘full’ capture.

you go through the various

steps, including prompts
on adjusting levels using
on-screen meters. Once
you reach the point where
you can actually start
to model your amplifier,
initiating the process sends
a stream of test signals and
noise bursts at different
levels to your amplifier,
followed by a rather long
sequence of guitar parts
played in different styles.
Once this has been done,
Tonex fires up its neural
network and sets about
analysing the results so
that it can build your virtual amplifier. At
this stage, you have the option of picking
three accuracy levels, the most accurate
taking the longest processing time.
Processing on my system took several
minutes at the default middle setting, and
over half an hour if I used the maximum
accuracy option. IK say that the CPU and
graphics processor affect the time (so
some computers can complete the job
faster), and just as I was concluding the
review IK informed me that an update
(which should be available by the time
you read this) will reduce the time taken
by about a third.
After the processing has completed,
you can select a ‘skin’ for your amplifier,
which includes a gain control and basic
three-band amp EQ. With the EQ set
midway, you hear the sound of your
amp as modelled by Tonex, and there’s
a page where you can directly compare
the miked amp sound against the Tonex
model, to assist you in fine-tuning the
result. Before finishing you get an info
sheet, in which you fill in details about
the amplifier, speaker and miking
arrangement. The same process is
followed when cloning pedals (though, of
course, you don’t need to mic anything
up unless you want to model a pedal and
amp combination). When complete, your
amp or pedal takes its place alongside
those already in the library. You can also
add comprehensive post-amp EQ, gating,
compression (switchable pre or post the
Tone Model) and reverb from the menu
bar at the top of the screen. My only
criticism here is that, from an aesthetic
point of view, I’d have preferred to be
able to add reverb and tremolo controls
to the amp skins themselves, given that
many classic amps have spring reverb
and tremolo on board.
Though the final amp model responds
to playing intensity in the same way as
your actual amplifier, it makes good sense
— as it does with other amp-profiling
systems — to create more than one
model for the same amp. For example,
you might want to use it set clean for
some songs but perhaps just breaking up
(or massively cranked) for others. It’s also
important to turn off any reverb on the
amp before you attempt to model it, and
to take care to position the microphone
just as you would when making a serious
studio recording.
There are already several forums and
YouTube channels discussing Tonex. Most
opinions are very positive, but some users
inevitably obsess over tiny differences
that they perceive between a direct
recording of the amp and the Tonex
model. The bottom line is that the better
the gear and the more care you take
when capturing your amplifier, the more
accurate the end result will be. Provided
that you take care in matching levels and
use a good re-amping box, the differences
really are very small — they’re certainly
no worse than a very slight change in
mic position between one recording and
the next. If your re-amping box doesn’t
have a flat response, or if you use
something else, such as a volume pedal
as mentioned earlier, you may introduce
some small sonic differences that will
impact on the accuracy of the modelling.
It’s worth mentioning that when
‘sampling’ any gear like this, there’s
a risk of feedback loops, so you should
make the effort to understand how your
interface’s routing software works! Indeed,
I experienced a few head-scratching
moments when first setting up my
PreSonus 192 interface: no matter how
I set its mixer panel controls, I experienced
feedback as soon as I initiated a Capture;
the mic signal was somehow being routed
back to the guitar amp that I was miking
up. The solution in this case was to use
the interface’s software control panel to
disable its mixer, so that it could be used
as a ‘dumb’ interface.
Down The Rabbit Hole
The first amp I modelled was my Roland
Blues Cube, itself a modelling amp but
one with a very valve-like sound and
feel. My Tone Model not only sounded
pretty close but, importantly, also gave
me a similar playing ‘feel’. There’s
a bit of a learning curve in figuring
out the best amp settings to use and
in matching the re-amping gain, and
I don’t think I matched the re-amp level
accurately enough in my first attempt;
the resulting model seemed to have
a touch more overdrive than the amp
settings I used. But IK have made the

54 January 2023 / www.soundonsound.com

 ON TE ST
I K M U LT I M E D I A T O N E X
IK offer a number of hardware devices
that make good parters for Tonex, including
the Tonex Capture (pictured), which comes
bundled with Tonex SE.
process very user friendly and, after
only a couple of attempts to hone the
technique, I found the process pretty
much painless. I repeated the capture,
being more careful to match the re-amp
level, and was rewarded with an excellent
recreation of my amp, which I’d set up to
have just the slightest hint of break-up.
I also made a very successful capture
of the amp in combination with a Tube
Screamer-style overdrive. The only very
small inconvenience in all of this was
having to wait for the program to create
the Tone Models.
My next attempt was with a Bugera
5W valve amp, which I’d rehoused in
a larger open-backed cabinet with a 1x12
Fane speaker. It’s a fairly clean-sounding
amp based on an old Fender circuit, and
tends to work well when used with some
pedals. Again, I felt that the captured
model replicated the original miked amp
sound and playing feel very closely. Many
of the Tone models that come with Tonex
favour very driven amp sounds, so I was
pleased to discover that the modelling
process can be equally effective when
using clean amps, or those with just a hint
of break-up.
Before I moved onto the next amp, IK
sent me their Tonex Capture box to try
(by the way, if you buy this, you get Tonex
SE bundled for no extra cost). I set this up
in re-amping mode and this time around
the amp was a tiny Vox MV50 head, set
to a bluesey overdrive, feeding a 1x12
closed-back cabinet. Again, I set the
re-amping level by ear to match that of
the guitar plugged directly into the amp.
I did notice a tonal change this time, when
comparing the signal path through my
interface and Tonex Capture box against
the guitar plugged in directly: there
seemed to be a little more weight and low
end in the interface/Tonex Capture signal
path, but the difference was still fairly
subtle — small enough that it could be
due to any number of factors, including
the characteristics of the instrument input
56 January 2023 / www.soundonsound.com
Pricing
There are four different versions of Tonex, each with slightly different features. Full details of each
are available on IK Multimedia’s website, but it’s worth providing an overview here. Tonex CS is
a free player, with 20 models included and the ability to download 20 more of your choice — it’s
a good demonstration of the results that can be achieved, but it can’t be used to capture the sound
of your own amp. Tonex SE (€179.99) is the most affordable version to include the capture function,
and comes with 200 models. Tonex Standard (€299.99) and Tonex Max (€479.99) both support the
capture process, but each one offers progressively more features.
If you don’t already have a suitable interface or re-amp box, then it migh prove more cost-effective
to look at what comes bundled with IK’s hardware devices. If you buy the Tonex Capture hardware
box, a licence for Tonex SE is included in the price (€479.99). IK’s audio interfaces might also be worth
consideration: their stereo AXE I/O, which has built-in re-amp and DI facilities (as well as mic/line inputs
and line outs) can be bought with Tonex Max and AmpliTube 5 Max for €539.99, while their AXE I/O
Solo with AmpliTube 5 Standard and Tonex Standard comes in at €419.99. IK Multimedia also offer
a loyalty discount scheme for existing owners of other software. All prices include VAT.
on my interface and the effect of any use the amp with a drive pedal, the Tone
connecting cables. Model is essentially a composite picture
Completing my capture on the quicker of the amp-plus-pedal characteristics.
Default mode produced a very plausible The EQ on the amp skin may not behave
rendering of the little amp and speaker in the same way as the EQ on your amp
combo, and once the process was — at first, I thought it might have helped
complete, I found I could pretty easily if the designers had offered a choice of
use the three-band EQ on the amp skin UK or US tone stacks — but if you have
to massage the tonal balance. I’d say several favourite sounds that you can
that this is a good tactic when you’re coax out of your amplifier, with or without
experimenting — it saves time while you’re drive pedals, you can capture them all as
searching for a sound, and if you like separate Tone Models, so perhaps the
the result you can then run the capture exact character of the EQ isn’t such a big
again in highest quality mode. On a more issue; it will probably only be used to
practical note, I reckon Default mode fine-tune the result.
should get close enough to satisfy most Tonex has deservedly attracted
players anyway — despite small tonal a lot of interest, and there’s been
differences, it still produces good, playable much discussion and speculation as to
sounds. whether or not IK Multimedia might look
to produce a hardware box capable of
Verdict running the Tone Models — so that the
Once you have made a few Tone Models sounds can be used live, without having
and got the hang of the process, the to rely on a computer or lug around your
accuracy of the end results that you can heavy tube amp. At the time of this review,
achieve is very impressive. As I touched nothing had been announced but IK do
upon earlier, I don’t think that Tonex can already make hardware, and I reckon
capture the full range of performance of this would make a great deal of sense.
an amplifier in one go. As with the Kemper In the meantime, Tonex makes getting
hardware, what you get is a ‘tweakable a great recorded guitar sound based on
snapshot’, which matches the sound your own preferred sound much more
you set up, but allows a certain amount affordable than the hardware options, and
of leeway courtesy of the gain and tone a lot easier to achieve than trying to tweak
controls. So it’s definitely down to the user
to find the sweet spot for both the amp £ See pricing box.
controls and the mic type/position. If you W www.ikmultimedia.com
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 ON TE ST

SSL 12
SAM INGLIS

hen SSL doubled up their SiX analogue mixer

W last year, they chose not to call it the TWeLVe.

It turns out that number was reserved for
a different product, which builds instead on the 2 and
USB Audio & MIDI Interface
2+ introduced a couple of years earlier. Like those
devices, the new 12 is a bus-powered USB audio
The SSL 12 is a cleverly conceived interface
interface, but as its double-digit designation would
suggest, it’s a lot bigger and more capable. with a very attractive price tag.
The 12 has a similar desktop form factor to the
smaller interfaces, with all the controls on the top, applications, and the available for the alternate speakers. It’s
though naturally it is somewhat larger. Connection built-in mono talkback possible to create custom cue mixes
to the host computer is made using a single Type-C mic, which shows up as for each headphone output separately,
USB port, and the 12 requires the full 900mA power a stereo input. and even to repurpose the headphone
delivery described in the USB 3.0 specification; in other On the output side, outs as additional line outputs. The line
respects, it’s fully compatible with USB2. There’s no things are a little simpler, outputs are DC-coupled, so can be used
option to connect a separate power supply or a second with two stereo pairs to send control voltages to an analogue
USB power source, as is the case with some interfaces. of quarter-inch jacks synthesizer. Oh, and the 12 also inherits
The name, in theory, refers to the total number of augmented by two the 5-pin DIN MIDI In and Out from the 2+.
available inputs. The first four of these are analogue front-panel headphone
mic/line channels on combi jack/XLR sockets. All four sockets. This is outwardly
From 2 To 12
have individually switched phantom power, high-pass the same total output Low-latency monitoring on the 2 and 2+
filters, five-segment LED meters and SSL’s 4K+ ‘vintage’ complement as on the 2+, is controlled using a knob that allows
mode, and the first two also have a high-impedance but the implementation the user to adjust the monitor balance
option for directly recording guitars and basses, with is a lot more flexible. between playback and input signals, and
separate quarter-inch jack inputs. This number is made Should you wish to a button to determine whether the inputs
up to 12 by the provision of a single optical input, use the second line are heard panned left/right or centrally.
which, depending on sample rate, accepts up to eight output pair to connect That’s fine for a stereo interface, but the
channels of digital audio using the ADAT Lightpipe an alternate monitoring 12’s greater complexity demands a more
protocol. And it doesn’t stop there, because despite system, speaker switching flexible system. More flexible, in fact, than
the name, the SSL 12 actually presents 16 inputs is catered for, with could be easily implemented in hardware
to your DAW. The additional sources are a stereo a dedicated front-panel alone, and so the 12 has its own software
loopback input that can capture audio from other button and ±12dB trim control panel.

58 January 2023 / www.soundonsound.com

 At the back we find a USB-C port, ADAT in, full-size MIDI I/O, four quarter-inch audio outputs and four combi
inputs. Two instrument inputs and two headphone outputs are found on the front panel.
The 12 isn’t just software-controlled,
however. It’s “360-enabled”, meaning
that control is enacted from the same
SSL 360 application used to interact with
the UF8 and UC1 control surfaces, and
with SSL plug-ins in their Plug-in Mixer.
The 12 gets its own pane titled SSL 12
Mixer, which looks quite a lot like the
top panel of the BiG SiX. The interface
presents a total of eight outputs to your
DAW, and these appear as inputs in the
Mixer pane alongside
the four analogue
inputs, eight ADAT “As you’d expect, it sounds very good, with the
inputs and the
talkback input; each
low-noise preamps being a particular highlight.”
channel type can be
shown or hidden as required. The input
channel phantom power, high-pass filter
and line mode switches are duplicated
in software, and cue mixing is enabled
by three send controls on each channel.
These are routed to headphones A and
B and to outputs 3+4, though this bank is
greyed out if you’re using those outputs
for alternate speakers. A nice touch is
that there’s an optional pan control for
each send, so you could, for example,
pan a click track hard left for a vocalist
who wants to take one headphone
off, but keep the click central for
another musician.
In fact, there are lots of nice touches
to be found throughout the SSL 12 Mixer
Point 1 Power
There are several different aspects to the USB
specification, and the two most important for
audio interfaces are data bandwidth and power
delivery. The data bandwidth of USB 2 was
already more than enough for most multitrack
audio applications, so the bandwidth increases
in USB 3 make no practical difference. The SSL
12 is fully compatible with USB 2 as far as data
transfer goes.
The USB 3.0 spec also raised the maximum
current that devices are permitted to draw.
page. On the monitor control side, the
hardware Cut and Alt buttons are joined
by configurable Dim, a Mono button
and one that inverts the polarity of the
left channel; pressing both of these
together lets you monitor only the Sides
or Difference signal. The physical buttons
can also be reassigned in software,
so if for instance you have no need of
talkback, you can repurpose the Talk
button as mono or dim. The bus channels
that feed headphones A, headphones
B and outputs 3+4 can be set to pick up
their sends pre- or post-fader, and any
or all of them can pick up the AFL bus
when something is soloed. A Settings
pop-up allows each headphone
output to be optimised for standard,
high-sensitivity or high-impedance
headphones, or converted to a balanced
or unbalanced line output. And finally,
if you’re monitoring everything through
your DAW and don’t need the Mixer
functionality, you can simply select I/O
Mode, which routes each source directly
to an appropriate output or DAW input.
My only minor niggles are, firstly, that
there’s no indication on the 12 itself to
This limit was raised again in USB 3.1, when the
move to Type-C connectors was made. The SSL
12 is described as a USB 3.0 device because it
requires the full 900mA specified by USB 3.0;
SSL say that there are a few circumstances
where it could draw more than this, in which
case you’d need to connect it to a USB 3.1
host port for full performance. That might
be the case if, for example, you were using
four phantom-powered mics plus two pairs of
high-impedance headphones.
show whether ADAT clocking has been
successful, and secondly that talkback
only operates in latching mode.
Hotting Up
Compared with the 2 and 2+, some of
the 12’s audio specifications seem to
have been adjusted, again mostly in
ways that seem appropriate to its role
as the centrepiece of a more serious
home studio. A 0dBFS signal sent to the
line outputs, for example, will produce
a +24dBu analogue level rather than the
+12.5dBu of the 2
and 2+. This means
that if you connect
active monitors
directly to the line
outs, you may find
you never use the main level control
above the lowest part of its range; if
so, it might be helpful to permanently
engage Dim in the Mixer page. This sort
of non-ideal gain structure also means
that if you leave the 12 switched on while
powering your computer up or down,
you’ll hear noticeable splats and thumps,
SSL 12
£399
PROS
• Good sound quality and
specifications.
• Superb cue mixer utility integrated
within SSL’s 360 software.
• Excellent attention to detail in the
design, with lots of ‘pro’ features
especially on the monitor control
side.
• Headphone amps cater for all types
of headphone and can even be
repurposed as line outs.
• Very good value for money.
CONS
• Very hot preamp alignment may be
an issue for recording drums and
other loud sources.
SUMMARY
A beautifully thought-out bus-powered
desktop interface with lots of neat little
features that lift it above the herd.
www.soundonsound.com / January 2023 59
 ON TE ST
S S L 12
The mixer software will feel familiar to
anyone who’s used an analogue console.
so if you can’t turn down the input gain
on the monitors themselves, it’s probably
a good idea to get into the habit of using
the rear-panel on/off button.
In mic preamp mode, the inputs
behave pretty much as they did on the
2+, which means they offer a 62dB
gain range and an impressively low
A-weighted EIN figure of -130.5dB. As
on the smaller interfaces, however,
they’re still aligned very hot, with the
maximum input level in mic mode being
+6.5dBu. Consequently, there’s tons of
clean gain available for speech and so
on, but the flip side is that if you want
to record drums, you’ll either need to
use very insensitive mics or get yourself
some inline pads. However, the 12 can
accommodate levels of up to +24dBu
in line mode, and the fact that line ins
and outs are both aligned to this level
should make it easier to patch in external
hardware at mixdown if you want to.
Like the 2 and 2+, the 12 uses generic
audio drivers. On Mac OS it works with
the built-in Core Audio driver, whilst
Windows users need to install the usual
Thesycon ASIO driver. My tests took
place on a Mac, and latency performance
was typical for devices of this type.
Reaper reported a round-trip latency of
7.8ms with a 32-sample buffer at 44.1kHz,
and a loopback test confirmed this within
60 January 2023 / www.soundonsound.com
a few samples. This would suggest that
the built-in mixer adds a millisecond
or so to the usual driver latency, and
SSL confirmed that the mixer latency is
0.98ms at 44.1kHz. To my mind that’s
a price worth paying for such an elegant
mixing utility, but if you do a lot of work
with soft synths or you need to monitor
input audio through the DAW, it might
be worth bearing in mind that working at
96kHz not only brings down latency due
to buffering but also reduces the mixer
latency to 0.17ms.
12 Steps
There was a time when bus-powered
devices were very much the poor relations
of the audio interface family. I expected
Thunderbolt to change that, but in fact
it’s been USB 3 that has given them
a much needed shot in the arm. And that’s
nothing to do with data bandwidth, which
was already more than enough under
USB 2, or latency, which is unchanged.
It’s purely down to the more generous
power delivery specification of USB 3 (see
box). Where once it was a real struggle
even to power a 48V microphone from
a bus-powered unit, we’re now seeing
supercharged desktop interfaces that can
handle multiple mics, run built-in DSP and
operate at full ‘professional’ studio levels.
The SSL 12 is an excellent example. It’s
compact, highly portable, and doesn’t
require an external PSU, yet is capable of
tracking a full band when paired with an
ADAT-equipped preamp.
As you’d expect, it sounds very
good, with the low-noise preamps being
a particular highlight. Dynamic range
figures of 111dB on the inputs and 120dB
on the outputs mean its conversion is
never going to be the factor limiting the
quality of your recordings and mixes. But
what really impressed me is the way SSL
have sneaked in so many ‘professional’
features and concepts from the world
of large-format consoles. The built-in
mixer is extremely well thought-out and
remarkably versatile, yet everything’s
presented in a way that will be instantly
familiar to anyone who’s spent time on
an analogue desk. Features such as
speaker switching with level trim, left
channel polarity reversal, talkback and
pre/post-fade switching for cue sends
are rare enough on large rackmount
interfaces, and almost unheard of on
a desktop device like this. There is
clearly potential here to develop larger
and more powerful systems on the same
framework, and I look forward to seeing
where SSL go next after the 12. The 16?
The 32? The 4000? Hang on...  
£ £399 including VAT.
T Sound Technology +44 (0)1462 480000
E info@soundtech.co.uk
W www.soundtech.co.uk
W www.solidstatelogic.com
 61-KEY

73-KEY

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Redefining performance workstations, the Nautilus packs the power of
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 FE ATURE
These gadgets might not be the most glamorous
devices, but at some point in your recording adventures
you’ll probably find you need each and every one.
HUGH ROBJOHNS
W
hether you’re running
a live-sound system, working
in a professional studio, or
recording yourself in the spare bedroom,
sooner or later you’ll come across
a situation where you either need to track
down a fault or connect two ‘incompatible’
devices — and you’ll need a ‘problem
solver’ to do it. This article takes you
through the most common types of audio
problem solvers, explaining what they do,
how they work and when and how you
should use them — and sometimes how
you might creatively misuse them, too!
DI Boxes
Probably the most frequently used
problem solver is the direct injection or
62 January 2023 / www.soundonsound.com
‘DI’ box, which is designed to accept an
unbalanced, instrument-level signal and
provide a balanced ‘mic-level’ output. The
main advantages are that the mic-level
signal can travel happily down long cables
unaffected, and the XLR cable from the
DI box’s output can plug into any stage
box without scaring anyone! This allows
a guitar (or almost any instrument or
electronic signal) to be connected safely to
a distant mic preamp, such as on a mixing
console or audio interface.
Naturally, there are variations on
the theme, and prices range from £20
to several hundred, the variance being
mostly due to features, sound quality,
and robustness. It’s perhaps also worth
pointing out that ‘sound quality’ can refer
both to the technical performance and
to the sonic characteristics, and some
of the most expensive are as much amp
simulator or saturation processor as they
are DI! Choose well, and a DI box should
last, literally, a lifetime — and in that
context, the higher-priced units might offer
rather better value than you initially think.
DI boxes are available in single, dual-
(stereo) and multi-channel models, and in
either ‘active’ or ‘passive’ forms. Passive
versions are often less expensive than
active types, but while they can do the
job they may be technically and audibly
inferior in some situations, particularly
when used with electric guitars, as I’ll
explain shortly. But passive DIs can be
more versatile than active ones, and I’ll
come back to that later.
A passive DI box contains little more
than a specially designed transformer
with a fairly high ‘turns ratio’ of 10:1 or
12:1, which means the output is a 10th or
12th the size of the input. That means it
reduces the signal voltage from instrument
level (nominally -20dBu) to mic level
(around -40dBu). The way the transformer

At the heart of
every passive DI box lies
a transformer. As well as catering
for the balancing, levels and
impedances, these provide galvanic
isolation which protects the gear on either
side in the event of faults.
is wired allows it to convert the unbalanced
input to a balanced output, too. The design
of the transformer allows it to present
a moderately high input impedance of
around 150kΩ, and that is fine for anything
with an electronic output like electronic
keyboards, DJ rigs, drums machines, and
so on. However, it is often too low to work
properly with electric guitars and basses,
which typically need a load of at least
250kΩ and ideally much higher. For these
kinds of instruments, an active DI box is
a better choice because it can provide
a much higher input impedance.
A very important benefit of the
transformer is that it provides electrical
(galvanic) isolation between the source
and destination equipment. This
means that there is no direct electrical
connection between the input and output
connections — the signal is passed from
the transformer’s primary winding to the
secondary winding only as a varying
magnetic field. This isolation means
that passive DI boxes inherently protect
the input source from phantom power,
which may well be present on the mic
cable connected to the output of the DI
box — nobody one wants their shiny new
ke
keyboard or DJ console fried by
48V phantom power the moment
it’s plugged in on stage!
The same isolation is also a safety
feature It protects the source or
feature.
destination in the event of a serious
electrical fault in the other, and can be —
indeed it has been, literally — a lifesaver.
That’s why I always advocate using DI
boxes at live gigs, especially so when
working with ‘unknown’ equipment, to
electrically isolate the performers’ gear
from the live sound equipment.
If seeking a DI box for keyboards or
other electronic sound sources, a good
quality passive DI is a good choice. The
Palmer PAN 01 and ART Passive Direct
Active DI boxes can provide higher impedances, which allows electric guitars and basses to
perform as their designers intended. They can be run off battery or phantom power, the latter usually
being preferable where available.
are g good single-channel options, while for
stereo applications the Mackie MDB-2P,
stere
Klark Teknik DI20P and Radial ProD2 are
solid choices. There are many more on
offer from other manufacturers, of course.
The
Th alternative form of DI box, as
I men
mentioned earlier, is the active DI box,
so-called because it contains active
so-ca
elect
electronics which provide a much higher
input impedance than is possible with
a pas
passive transformer. Most active DI
boxes present an input impedance of
boxe
aroun
around 1MΩ (1000kΩ), which allows
electric guitars and basses to perform and
elect
soun
sound as their designers intended — their
pickups are not unduly ‘stressed’. Active
picku
elect
electronics need power, of course, and
that ccomes either from an internal battery
or fro
from phantom power over the mic
cable. The latter is generally considered
cable
the m most secure and reliable arrangement
as it ccan’t run down mid-gig!
M
Most active DI boxes also contain an
output transformer, so that they offer all the
outpu
same safety benefits as discussed for the
passive DI, but some omit the transformer
passi
for co
cost and quality reasons — while they’re
perfectly capable of converting the signal
perfe
level and performing the balancing, and
can b be a good cost-effective option in
your own studio, do check before buying
if you’re concerned about electrical safety.
Popular active DI boxes include the BSS
AR-133, Klark Teknik DN100 V2, Warm
Audio WA-DI Active, and the Radial J48,
amongst an ocean of others.
As well as performing their basic
functions, the vast majority of DI
boxes of both types incorporate some
user-configurable facilities, such as
selectable attenuators (pads) to allow them
to accept different input levels (commonly
instrument, line, and/or speaker levels).
www.soundonsound.com / January 2023 63
 FE ATURE
M E E T T H E P R OBL E M S OLV E R S

Almost all DI boxes also have a switch

that separates the input ground from the
output ground, to avoid ground loops
— typically this is called a ‘ground-lift’
switch. Most also have a ‘link’ or ‘thru’
socket (same thing, different name), which
is wired directly in parallel with the input
socket so that the source signal can, if
required, be passed both to a backline
amplifier (via this output) and the mic
preamp (via the XLR).
More elaborate models might also
feature an extra, electronically-buffered
unbalanced output to feed a tuner or
similar, or might be able to mix stereo
inputs to create a single, mono output
signal. Some are designed specifically
to accept inputs directly from a laptop
computer or even a vinyl record pickup
cartridge, while others might include
guitar amp/cab emulation circuitry or other
musically-beneficial tone shaping facilities,
such as in the A Designs REDDI and Tech
21 SansAmp Para Driver.
Although the 1MΩ input impedance
presented by most active Di boxes might
sound a high number, it’s still not really Similar to DIs, line isolation boxes offer galvanic
high enough to work properly with the isolation but without the drop in level. The ART DTI
piezo pickups found on some acoustic is a popular two-channel model, partly because it change the signal level the transformer
guitars and other acoustic instruments. offers a choice of output connector types. has a ratio of 1:1. Instead, it’s there to do
These devices typically prefer to see two things. First, it provides electrical
10MΩ or more and, consequently, there bodge isn’t an ideal solution for various (galvanic) isolation between the source
are special active DI boxes intended technical reasons, but it can be a useful and destination, this time principally to
specifically to cater for piezo pickups — get-out-of-trouble fix in an emergency. The prevent ground-loop noise (hence the
examples are the Radial PZ-DI, Caveman proper tool for the job is a ‘re-amper’, of common alternative name of a ‘hum killer’),
AP1 and Fishman Aura Spectrum DI. course, which I’ll cover below. but also to act as an electrical safety
I mentioned earlier that some problem barrier between source and destination
solvers can be ‘misused’ in a creative
Line Isolation Boxes equipment. Second, it acts as a format
way, and one option is to use a passive DI A line isolation box is very similar in converter, providing a balanced output
box backwards! Obviously, the electronic concept and implementation to a passive from an unbalanced input, or vice versa,
circuitry within an active DI box means DI, but it is intended for use with line- or depending on the connection options.
the signal can only flow from input to instrument-level sources and it provides Most line isolation boxes are
output, but the transformer in a passive the same level at the output as the two-channel units and, like passive DI
box doesn’t really care which way the input, and since the intention is not to boxes, prices vary considerably — the
signal goes. That means that you can more expensive models generally
provide it with a balanced mic-level signal have better-quality transformers, which
at its output (via a suitable adaptor cable) introduce less distortion (especially at
and generate an instrument-level low frequencies) and can handle higher
unbalanced output. Okay, so this signal levels. As already suggested,
typical applications for a line isolation
As well converting signal box include breaking ground loops
levels and impedances,
between computer interfaces
active and passive DIs
can be designed to add and active monitor
musically appealing speakers, or connecting
character. The A Designs keyboards or DJ rigs to
REDDI, for example, uses a PA mixer’s line-level
custom transformers and inputs (instead of
valves to deliver an almost
using a DI box to feed
amp-like tone, while the
passive Acme Motown mic inputs). Popular
DI WB-3 also employs models include the
a custom transformer. ART CleanBox-2 and

64 January 2023 / www.soundonsound.com

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 FE ATURE
M E E T T H E P R OBL E M S OLV E R S

Diagnosing Problems
One problem solver I find particularly useful is
a ‘phantom-checker’, used to test for the presence
of a phantom power voltage on each wire in an XLR
cable; I’ve carried one on my keyring for decades!
Several mic manufacturers used to make these in
various forms, but when writing this article the only
commercial unit I was able to find which remains
currently available is Canford Audio’s. The most technical
models only illuminate if the phantom voltage is within
the prescribed voltage limits and/or capable of providing
adequate current, but simpler versions are still useful and
easily within the bounds of simple DIY if you can solder; there
are several designs on the Internet. Most are built into a standard
XLR plug body, making them easy to carry around for fast, efficient
determination of whether phantom power is reaching a mic or DI
box (and for confirming broken cables). You’ll also find other useful
test gadgets built into XLR connectors, including phantom-powered
pink-noise and tone generators.
Of course, a more comprehensive way of testing cables is with a dedicated
cable tester, which will help you identify what the fault actually is! There
are many such devices and they cater for many different types of
connector. I’ve yet to see one that caters for everything in one box but
there are several which offer most of those in use in the audio world
today, and some also include useful additional features, including
phantom-power checking facilities, tone/noise generators and
monitor speakers, so that connections can be tested audibly.
Some, including the Hosa CBT-500,, also a allow you to test not
only analogue audio connections and MIDI DINs but also USB
and Cat cables. I find the Sonnect Audio Sound Bullet to
be a particularly useful and all-in-one compact portable
audio tester. Some cable testers come with two separate
parts, so as to facilitate easy testing of continuity over
long cable runs: dbx and Studiospares both offer
suchh a device, for example. Be sure to find a model
that has the connections you need to test, though.
For example, there are now few (the
Behringer CT-100 being one)
ch can test TT bantam
which
cabl without requiring an
cables
p cable.
adaptor

DTI, and the latter is particularly versatile
because it has multiple connections in
different formats. Behringer’s MicroHD
400 is another affordable option.
Mic Splitters
It’s often necessary to share the output
of a microphone between two or more
separate destinations. A live-sound
application, for example, might be to split
the stage mics between the front-of-house
(FOH) and monitor consoles, or between
the FOH and a separate recording setup.
In a studio, it might be to share the mic
Mic splitters such as the Radial JS-2 use special transformers
to route the same signal to multiple destinations without a change
in level — and line splitters such as the Whirlwind Split 6 do the
same thing for line-level signals.
between different preamps for comparison
purposes. The usual technology for this
application is, once again, entirely passive,
but it uses a special transformer, which
typically will have a single primary winding
(as usual) but two or three secondary
windings, configured in 1:1:1(:1) ratio,
meaning that all the outputs are at the
same level as the input.
Because phantom power is a DC
voltage it can’t pass through the
transformer, so all mic splitters have
a ‘direct output’ socket which is wired
directly across the microphone input
socket (in exactly the same way as the ‘link’
or ‘thru’ socket of a DI box). This direct
output is always connected to the ‘main’
destination, which is the one that provides
phantom power to the mic if needed.
The transformer’s primary winding is also
wired to the microphone input socket,
so independent (electrically isolated)
outputs can be provided through the
secondary windings.
Each secondary winding normally has its
own Faraday shield within the transformer
connected to the corresponding output
XLR’s pin 1, and usually there are separate
ground-lift switches for each output —
these tie the separate output shields to the
microphone/direct output ground, if needed
(or ‘lift’ them to avoid ground loops).
Again, the more expensive mic splitters
have better-quality transformers which
introduce less distortion and can handle
louder signals. Some also include an input
pad to avoid overdriving the transformer
when the microphone is placed in front
of very loud sources. Typical examples
include models from ART, EMO, Palmer,
and Radial.
As with DI boxes, there are active
versions of mic splitters too, but these
are typically only available in large
multi-channel formats and most act like
simple mic preamps, providing phantom
power (if needed) to the mic, and
something approaching line level at the
transformer-isolated balanced outputs.
Re-ampers
Re-amping is a popular music production
technique, whereby a cleanly recorded

66 January 2023 / www.soundonsound.com

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 FE ATURE
M E E T T H E P R OBL E M S OLV E R S
(for example through a DI box but not an
amp) instrument track is sent back out to
an amplifier/speaker in the studio, and
then the acoustic sound from that amp
is re-recorded. Alternatively, it may be
desirable to route the cleanly recorded
instrument through a chain of stomp
effects pedals and re-record the output of
the effects chain. The technical problem
to be solved here is that the outputs of
most mixing consoles and audio interfaces
are balanced, line-level signals, whereas
the guitar amp or stomp pedal expects
unbalanced, instrument-level signals.
The best solution is a re-amper or
re-amp box and, again, there are many
variations on the theme. The simplest
form employs a step-down transformer
Other Handy Gadgets
Arguably a must-have problem solving accessory
for every engineer is the humble in-line XLR pad.
When a moderately sensitive mic is placed in
front of a very loud source it can easily generate
line-level output signals, and these can overload
many mic preamps. A lot of preamps have a ‘pad’
button to accommodate such situations but where
this option is absent (or if, on an audio interface,
it’s inconvenient to access using software) an
XLR in-line attenuator can save the day. There
are many fixed attenuation options from 10 to 50
dB, but 20dB is often a good compromise value
if you only want to buy one type. A few (including
the Hosa ARR-448, the Audio-Technica AT8202
and Shure’s A15AS) offer two or three switchable
settings, typically ranging between 10 and 40dB.
It’s worth noting that, with an input
impedance of between 600Ω and 1000Ω and
an output impedance of 100Ω to 200Ω, almost
all commercial in-line XLR attenuators are
optimised for microphone applications — and
all pass phantom power too. Although they’re
physically capable of handling line-level
signals, their relatively low input impedance
may prove too much of a challenge for some
electronically-balanced line-level outputs to drive.
A potential result of that is transient distortion,
so I wouldn’t recommend their use for dropping
68 January 2023 / www.soundonsound.com
which performs both the signal level and
connection format changes. Most also
have facilities to adjust the output signal
level and to isolate the input and output
grounds, to avoid ground-loop problems
(ground lift). A proper re-amper is much
easier and more convenient to use than
the ‘backwards’ passive DI box bodge that
I mentioned above.
But some re-ampers are pretty
elaborate and can be expensive. Often,
they have DI circuitry in the same case,
so that the device can be used both for
recording and re-amping — and can both
send to and receive a signal from guitar
pedals, without involving an amp. But more
conventional examples include the Orchid
Amp Interface or the Radial Reamp.
line-level signals down to mic level; a DI box is
a safer bet for that application. That said, there
are a few boxes around designed to offer variable
attenuation of line-level signals, and these can
be useful if, for example, you’re deliberately
driving a mic preamp which lacks its own output
level control into saturation, and wish to avoid
overloading the next device in the signal chain.
Sometimes, of course, the problem is the
opposite — a mic’s signal is too low, rather than too
loud — and the solution is an in-line mic booster,
or ‘booster preamp’. These have become very
popular accessories in recent years and a lot of
manufacturers now offer variations on this theme,
with some models being very compact and simple,
and others larger and technically far more elaborate
(and costly). All are phantom-powered, but most
of them don’t pass phantom on to the microphone
because they’re usually designed for use with
low-output passive mics like ribbons and insensitive
moving-coil types. Cloud Microphones’ Cloudlifter
CL1 was one of the first in-line boosters to reach
the market, but other popular models include
Imperative Audio’s FP1, Triton Audio’s FetHead,
sE Electronics’ Dynamite and TNT models, and
Royer’s dBooster and dBooster2, to name a few.
Where there’s a need to boost the signal from
a capacitor mic, Triton’s FetHead Phantom and
An alternative re-amping solution,
which can be very convenient for stereo
signal paths, is to (mis)use a ‘consumer
equipment interface’ (more on these
below). This provides the appropriate level
and connection format changes, but does
so through active circuitry rather than
transformers — this means there’s a risk of
ground-loop issues in some circumstances.
Nevertheless, if you have such a device
available and lack a dedicated re-amper,
this approach can prove convenient when
wanting to process recorded signals
through stomp boxes.
Consumer Equipment
Interfaces
A very common requirement in all
pro-audio installations is to be able
to connect consumer equipment into
a professional system. Going back a few
years, the most frequent example was
to hook up a cassette machine to record
the output of a studio console and play
back demo tapes, and later the cassette
gave way to the CD burner. Today, we’re
more likely to be connecting laptops, MP3
players, smartphones or video machines,
or perhaps plumbing a secondary
consumer monitoring system (hi-fi amps or
active speakers with unbalanced inputs)
into a monitor controller or audio interface.
FetHead Broadcast models both pass phantom
power on to the microphone.
Other worthy mentions for the problem-solving
toolbox are polarity inverters and mutes. The
polarity inverter is typically another in-line XLR
connector, wired internally with the hot and cold
signals swapped over (so the input pin 2 goes to
the output pin 3, and vice versa). This arrangement
flips the polarity (wrongly referred to as ‘phase’ in
some quarters) of a balanced signal, and that’s very
handy if the console or DAW interface lacks this
facility. For example, when top- and bottom-miking
a snare drum it is essential to invert the polarity of
the bottom mic as the lower snare head moves in
the opposite direction to the batter head. Without
the appropriate polarity inversion mixing the
outputs from the two mics will result in a very thin
drum sound! Polarity inverters can also be handy
when combining mics located in different locations
— such as near and distant mics on a guitar amp, or
for decoding Mid-Sides stereo mic arrays.
Signal mutes can be in the form of an in-line
XLR connector with a rotary collar switch, or
a stompbox. Either way, the idea is simply to make
it easy for an artist to mute their own microphone
(or other signal source) when it’s not being
used — this can be very useful in ‘self-managed’
stage shows.

Consumer equipment typically has
unbalanced connections operating
at a nominal level of -10dBV (which is
roughly -8dBu), whereas professional
equipment is balanced at a nominal
+4dBu. This near-12dB level difference
and connection format incompatibility
is routinely addressed with some
active electronics in a dedicated
‘consumer equipment interface’, with
most providing two channels in each
direction: unbalanced
consumer signal at -10dBV
to balanced line level at
+4dBu, and vice versa.
Most consumer equipment
interfaces provide fixed
±12dB gain/attenuation,
though some allow the gain/
attenuation to be adjusted
to optimise signal levels
to and from the connected
equipment.
As I hinted above, these
boxes are also handy
solutions when re-amping
signals through stompbox
effects, since the latter
generally operate with
unbalanced connections
at levels that are not too
dissimilar to consumer
equipment. Note, though,
that these interfaces tend
not to employ transformers,
which means there is no Re-amp boxes are used to convert a mixer/audio interface’s balanced line-level
galvanic isolation between output to an unbalanced signal with a suitable level and impedance for guitar amps
the source and destination and pedals. Some, like the Radial XTC, offer more facilities such as an effects loop, while
devices. Thus, ground loops other products including the Little Labs Redeye 3D Phantom put a DI and re-amper in
are a possibility if using the same box, so it can send to and receive from effects pedals.
any (Class I, grounded)
mains-powered pedals. Battery-powered
effects pedals or those powered from
Class II isolated PSUs shouldn’t present
any problems at all.
Cat Adaptors
The low cost, ubiquity and convenience
of Cat 5E (and related) cabling has, in
recent years, made it a popular option
Several companies offer adaptors that allow four channels of balanced audio to be sent over fairly long
distances using standard shielded Cat 5E (and above) cable — but they won’t all be compatible with each other!
for lightweight analogue and digital Catdusa another. Each of the four
audio ‘snakes’. The four separate twisted twisted pairs within the Cat cable is
pairs within a standard Cat cable can wired to one of the four XLRs, although
be employed to carry four separate note that since there’s no universal
balanced audio signals, and the cable’s wiring standard exactly, which pair is
characteristic impedance is also suitable allocated to which connector may vary
for conveying AES3 digital audio as well between manufacturers! For example,
as conventional analogue audio over input 1 on one manufacturer’s input box
distances up to 100m. might appear on output 3 on a different
manufacturer’s output box.
Some manufacturers offer
bespoke breakout boxes
which cater for two signals in
both directions. This allows,
for example, two mic feeds
to be sent to a preamp
rack and two headphone
monitor feeds returned into
the live room. Cranborne
Audio’s CAST system works
this way, for example. The
same Cat cable is used in
both applications; only the
break-out box wiring differs.
The cheapest standard
Cat 5E cable is ‘UTP’ or
‘Unshielded Twisted Pair’
but it’s not a great idea to
use unscreened cable when
working with audio signals
over long distances. Instead,
Cat 5 FTP (Foil-screened
Twisted Pair) cable, which
has an overall screen, is
the minimum standard
cable for use in this kind of
application. In practice, most
Cat adaptor systems employ
Cat 6 FTP cables, which
Most Cat adaptors are configured to have an overall screen, although Cat 7
send four independent balanced signals S/FTP (Screened Fully-shielded Twisted
in the same direction, so a matching Pair) cables are even better, as they are
pair of male/female Cat adaptor boxes guaranteed to have separate screens over
are needed. One box, as the input each individual pair — a feature which
unit, is typically fitted with female XLRs could be important to minimise crosstalk
and the other, the output unit, with between channels, such as when working
male XLRs. Radial’s Catapult system with multiple microphone signals of
is one such example, and Whirlwind’s different levels.  
www.soundonsound.com / January 2023 69
 COMPE TITION
Win! JZ
Microphones
Worth €2500
outique Latvian mic makers
B JZ Microphones recently
celebrated their 15th
anniversary, and to mark the occasion
they’re offering you the chance to
win two of their acclaimed premium
microphones, plus accessories, worth
a total of €2500!
Founded in late 2007 in the Latvian
capital city of Riga, JZ make their
microphones by hand, including the
all-important capsules. The company’s
debut mic was the striking Black Hole,
an innovative design in terms of both its
To enter, please visit:
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looks and capsule construction. It owed
its name to its unusual slotted design,
into which JZ cleverly fitted a bespoke
shockmount, while its capsule
was the first to employ what JZ’s
patented ‘Golden Drop’ technology. In
essence, this involves applying gold
to the diaphragm in circular ‘drops’ at
strategic points, rather than coating
the surface completely. This process
results in a lighter diaphragm, with the
advantage of improved transient and
high-frequency response.
Up for grabs this month are the BH2,
an updated, cardioid version of the
original Black Hole, and a V67, part of
JZ’s highly regarded Vintage series. In
the 15 years since JZ first made their
mark with the Black Hole, the design
has gained a number of high-profile
fans, including Grammy-winning
producer Rob Chiarelli and legendary
engineer Sylvia Massy, who has
70 January 2023 / www.soundonsound.com
described the Black Hole as “a work
of art”!
The Vintage models, meanwhile,
each pay homage to a classic
microphone, while maintaining JZ’s
distinctive looks and design ethos.
So, although inspired by Neumann’s
U67, the JZ V67 employs solid-state
electronics for cutting-edge noise
performance, and again makes use of
a Golden Drop capsule. That mic also
has its share of champions, including
rock production titans Romesh
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With both the modern BH2 and
vintage-inspired V67, you’ll be able
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scenario — and to help you get
the most from these outstanding
microphones, JZ are also supplying
a premium wire-mesh pop screen.
The JZ Pop Filter incorporates a flexible
gooseneck mount, and features
special contouring on one side to allow
maximum sonic transparency while still
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Finally, JZ are also including
a bespoke wooden box that can
house both microphones and the
JZ Pop Filter.
To be in with a chance of winning
this fantastic prize bundle, simply
follow the URL shown, and answer the
questions there, by Friday 3rd February
2023. Good luck!
Prizes kindly donated by JZ Microphones
W www.jzmic.com
 ON TE ST

Ingram
500-series EQ
Engineering EQ52 mixing; or 640Hz to 5kHz,
for more extreme tone
Its combination of ‘See-Saw’ shaping, or perhaps use
band and filters make this as an analogue side-chain
filter EQ — for example,
classy EQ really versatile. to focus a compressor’s
NEIL ROGERS
sensitivity more towards
the mids/high end.
ric Ingram runs Ingram Engineering Next, we have what
can otherwise end up being folded back into the audio
spectrum, and this obviously allows you to set the
turnover frequency anywhere within the range captured
by a 192kHz recording.
In Use
The gentle nature of the slopes used for the high- and
low-pass filters mean the EQ52 typically delivers a very
pleasing tone-shaping that feels like it’s doing something

E from his home in Atlanta, and it

draws on not only his passion for
audio electronics but also his background
as a musician. The company’s small
but steadily growing range of studio
hardware began with the release of
a two-channel preamp in the early
2000s, but in more recent years the
main focus has been on 500-series
modules. Ingram offer a few different
modules, covering preamplification, EQ
and dynamics processing. Under review
here is their latest release, which puts an
intriguing set of EQ tools into the compact
500-series format.
Features
The EQ52 has three different sections,
which are far more straightforward
to use than first appearances might
suggest. Working down from the top,
an EQ In control toggles the entire EQ
circuitry in/out of bypass. Next, a passive
high-pass filter provides a gentle 6dB/
octave cut over a wide, adjustable
frequency range. A push-switch selects
one of two frequency ranges: 45-600
Hz, which you might typically use to filter
out low frequencies when recording or
Ingram Engineering
EQ52
$400
PROS
• Passes the ‘nicer than software’ test!
• Low- and high-pass filters are
surprisingly versatile.
• See-Saw band delivers excellent tonal
shaping with minimum fuss.
• Good value for money.
CONS
• None.
SUMMARY
A nicely conceived collection of
simple yet high-quality EQ tools in
a single 500-series module, and at
a competitive price.
Ingram call the See-Saw
filter, and others describe
variously as a tilt or pivot
more useful than simply removing unwanted information.
I found it often encouraged me to be a lot bolder than
I might expect — I’d often end up with the high-pass
filter. The see-saw, with set comfortably above 100Hz on guitars and vocals, for
its straight-line lever example. The low-pass filter was similarly effective and
balancing and pivoting worked superbly for taming any source that was overly
around a fulcrum, is bright or whose transients seemed that bit too ‘spikey’.
a good visual analogy! You I think ‘tilt’ style EQs can be really under-appreciated
select the pivot frequency (simple solutions that just work are too often overlooked
and then, as you bring
one side of the frequency
spectrum up or down, the
opposite happens to the
frequencies on the other
side of this fulcrum. The
result can be a drastic
tonal rebalancing or
subtle enhancement, but it
always seems to be a very
‘natural’, smooth-sounding
tonal shaping. As with
all sections of the EQ52,
you can exercise this
filtering over a very wide
frequency range: the pivot
frequency can be swept
(in two stages; there
are two range options
selected with a button)
all the way from 110Hz
to 26kHz!
Finally, a High-Cut
Filter section offers
passive low-pass filtering
with a 6dB/octave roll
off, anywhere from
2.7kHz to a mind-blowing
200kHz! Why on earth
would you want to roll
off frequencies so far
above the threshold of
human hearing? I put that
question to Eric, and he
explained that he wanted
the EQ to be useful for
rejecting unwanted,
non-audio content that

72 January 2023 / www.soundonsound.com

by music makers today!), and in the
analogue domain in particular they’re
great: they enable you to shift a part’s
tonality in a very clean, natural-sounding
way, and to do so without having to
adjust too many controls. Being able to
‘sweep’ such a wide frequency range at
which the filter ‘pivots’ makes the EQ52’s
See-Saw section incredibly versatile:
I found a number of effective uses for it
over the review period, from dramatic
tonal reshaping of ‘out-of-whack’ guitars
in a mix project, through to adding a little
sheen and air to a vocal or bringing
out the snap and presence in a snare
drum. I’d have loved to have tried a pair
of these over a whole mix, and I think
the options available here could work
superbly for mastering jobs. Mastering
engineers can (understandably) be fussy
about using dual-mono units, especially
with unstepped controls as on the EQ52,
but the EQ options available here would
be fantastic in those situations where
you need to get a bit more involved
than is ideal. (If you are interested in
exploring a pair of these, Ingram have
some information on their website about
matching up a pair for mastering.)
Final Thoughts
Over all, then, I was really impressed
with the EQ52, and I wasn’t just ‘testing
it’ either: I was very pleasantly surprised
at just how much I found myself turning
to it for use in real projects during the
review period. I’m a big fan of putting in
the effort during the recording stage to
get sounds as close as possible to the
way I think they’ll need to be in the mix,
and the EQ52 is perfect for this way of
working: not only can you make sounds
leaner and more ‘mix ready’ by removing
unnecessary information at either end of
the spectrum, but you can also reshape
the tone of a source, sometimes pretty
radically, without much risk of it becoming
too difficult to readjust during the mix,
should you later change your mind.
The See-Saw filter is a wonderful and
wonderfully simple concept, and this
one sounds great. Yes, I know there are
tilt filters available in software, but it just
doesn’t seem to sound as good to me
ALTERNATIVES
Notable 500-series competition comes
from, among others, Tonelux, who
have built their Tilt EQ stage into their
MP5A mic pre, and Phoenix Audio’s
Pivot Tone Channel.
when performed in the digital domain.
I tried a couple of times to copy a setting
on the EQ52 using FabFilter’s excellent
Pro-Q 3 plug-in (which includes a tilt filter
option) and I just couldn’t recreate the
effect of the EQ52, for example in the
way I was using it to smoothly brighten
a snare drum and, separately, a vocal.
As much as I enjoyed using the three
sections of this EQ in isolation, though, it’s
what you can achieve by using them in
combination that makes this design really
clever. If you have a spare slot in your
500-series rack, I’m pretty confident you’ll
get some value out of the EQ52.  
£ $400 plus VAT, duty and shipping.
T Ingram Engineering +1 678 685 9838
W www.ingramengineering.net
m
 TALKBACK
Billy Nomates
WILLIAM STOKES
C
ombining deft production and fluent
beat-making with no-holds-barred
spoken word and singing, Bristol’s
Billy Nomates, aka Tor Maries, has enjoyed
a fast ascent to cult success. Her self-titled
debut album was released in 2020 on
Invada Records, the label founded by
fellow Bristolian and founding member of
Portishead, Geoff Barrow, and she has a new
album due for release in January.
At the moment I can’t stop listening to
At the moment, quiet is nice! I’m quite busy.
So, to be honest with you, my sound of
choice is... not a lot! There’s just peace and
quiet where I live. I’m quite lucky. I’ve got
a basement flat opposite an allotment, there
are birds’ nests… it’s quite quiet at the top
of the hill. So I like just having the window
open, hearing the birds in the daytime. It’s
very nice.
The project I’m most proud of
I think I’m most proud of my upcoming
record. Which feels like a cop-out thing to
say! But truly, I feel like it represents the
scope of what I want to be as an artist — and
it took a minute to write. It took about a year,
I nearly rushed into it last year, but didn’t.
I took some time with it.
I think the temptation, when you have
a little bit of success and you’re on a little
bit of a roll, is to be like, “Quick! Get the
next thing out!” And I’m really glad I didn’t
do that. I hope it just has some interesting
ingredients. I feel as though my first album
massively represented where I was in life
and represented what I could do in that
place. And now, three years later, we’ve
been through a bit of a fucking turbulent
one. I think maybe this has introduced a little
bit more emotion and a little bit more depth,
hopefully. And I’m really proud of that. It’s not
something that I originally thought was going
to happen for Billy Nomates. I wondered if
74 January 2023 / www.soundonsound.com
it was going to, kind of, carry on this fierce
trajectory. There’s a heap of vulnerability to
this next record... “My most personal work
to date!” God, I hate myself...
The first thing I look for in a studio
I look for those odd or old tools first.
I worked at Invada Studios in Bristol a little
bit on this next record, and their selection of
things really suited me well. If a studio has
a [Roland] Juno-60, that’s a good starting
point for me! I love them. I feel like there’s
nothing you can’t do with a Juno-60, if
you’re really open to it.  melodies. I just think it’s a timeless piece
But yeah, it’s mainly the strange things.
It’s the odd little analogue drum machines
that knock about or, you know, something
in the percussion trolley, maybe an odd
thing that just makes a strange sound, and
that could run through a whole track but
really quietly, so you don’t know it’s there
but it makes the song run. I really like those
things. Also, good coffee in the kitchen.
The person I would consider my mentor
I don’t know how he’d feel about it, but
in terms of understanding sound and its
capabilities, I would say Geoff Barrow. I’m
always interested in his take on things. He’ll
often find something that I haven’t thought
about, and make me think about it. It’s often
about what doesn’t need to be there. What
is that thing over there? What is it supposed
to be doing? He’s never too flowery with
feedback. He offers an honest opinion
about things in terms of my music and I’ve
always appreciated that, perhaps I haven’t
shown it in the moment!
I’m not overly familiar with Portishead’s
music but I’ve always understood and
recognised their pioneering impact. Now
isn’t the time, but I know a few years from
now I’ll go through their back catalogue.
I look forward to that. His score work on
soundtracks like Devs or Ex Machina...
odd and interesting. BEAK> are also
a hugely influential band sonically, and they
operate in their own way. Invada Records
specialises in soundtracks but it’s also
championing, I guess, the stranger corners
of music. I admire his approach and
sensibilities, both to songs and beyond.
My go-to reference track or album
I can understand if people wouldn’t
immediately make this connection, but
I always go back to David Bowie’s Low.
I’ve got Low on, like, all the time! It’s one of
those records where every time I listen to
it, I hear something new. I don’t tire of the
of work, and his most interesting one.
And it was the one that people weren’t so
keen on when it came out, so I love it all
the more.
For me, it’s a reminder to just do
something different. There’s still nothing
that sounds like Low. It’s totally mad, and it
has a theatrical side to it. I often think of
theatre in songs, in a strange way. I’m not
entirely sure what the correlation is. But
in some ways that makes sense to me.
One of my favourite songs of all time is
‘Always Crashing In The Same Car’. It’s
understated, and whether in the right or
wrong mood it floors me.
My top tip for a successful session
Again, I don’t want this to sound like
a cop-out... but, a good atmosphere! Look,
you’re not running a bank, you’re making
music, you know? So, it has to be fairly
laid back and human. And having good
chats about things before you start. I take
my shoes off a lot. Especially for vocal
recording, I’d rather not wear shoes. Like
you’re just at home, you know.
It’s all the stuff around the thing itself.
A good technician. If you’re producing
and working with a co-producer, a good
relationship with those people is key,
I think. A nice atmosphere. Get a nice
coffee and talk about the world! Just have
a human touch to the day. And then get on

with the song. It’s an intimate experience
and an exchange. I believe it’s easier and
makes the recordings more natural if you
like and trust the people you’re with.
The studio session I wish I’d witnessed
Low. That’s probably obvious. And it just
gets more intense, every year. It’s the ’70s.
He’s in Berlin with Iggy Pop — who is also
a great inspiration. I think it would have
been fascinating to witness what that time
was really like, you know. They were living
fairly frugally, by what I’ve read, they were
making music and they were getting clean
and getting away from stuff. I’m fascinated
by that. I don’t drink, and I don’t dabble
with drugs. I think it’s interesting what
happens when you’re in a clear mind.
To see, you know, how mental and deep
are you willing to go without too many
distractions — or aids.
I don’t shy away from classic, popular
records. The thing is, they’re classic
records for a reason. Heroes is another
good example. There’s so much going on
in there. And when you really tear it apart,
it’s the odd things that make it. It’s the
sounds that make you go, “That doesn’t
really make sense to put that there! Like,
there’s an odd scraping sound there that
just sits really low, underneath everything.
And you think, “Why is that there?” But then
you take it out, and you’re like, “Ah. Well,
that’s the atmosphere gone.”
The producer I’d most like to work with
That is a question. I’d like to work with Steve
Albini. He mentioned that he was a fan of
what I was doing on my first record, which
was very cool and nice of him. He brings
a really interesting and organic approach
to all of his musical endeavours and the
industry around it. I don’t know what he
would do with my stuff or what he’d really
make of it and thats intriguing. To have
somebody with that kind of experience and
attitude to music, similar to Geoff, I think it
makes for honest results.
The part of music creation I enjoy the most
I think it’s that burst when you’re first
demo’ing and you burn off that first demo,
and it’s in a crappy MP3 format, and it’s
in some headphones that are being held
together with tape because you’re too
cheap to buy the expensive ones. I’ll just
have a listen in my flat or go for a walk
in the park or around the harbour with it.
Nothing ever really competes with that
moment. The joy is at its biggest there for
me. From there, it’s a process. And don’t get
me wrong, it’s an interesting process, but
before that point, it’s my ugly creation. and
the world has no idea what I’ve just done.
And it might never, if I decide it’s crap!
It’s yours, flaws and all. No-one’s
dissected it, no-one’s gone, “Let’s look at
this. Let’s look at that.” It just exists, in all its
terrible state, primarily as a feeling that you
Photo: Immy Done
wanted to capture, because that’s what
I think a demo is. So much is up for debate
from that point. I’ve learned to not dissect
too much from a demo. Not to play with
it too much. Because I think you can lose
energy sometimes. We have to trust the
sound we’ve made. It’s totally OK to put that
that crappy GarageBand preset synth sound
somewhere, if that creates the feeling.
Geoff’s a big one on that. He will always say,
“Live with it.” I’ll create something and he’ll
just say, “Go and live with it.”
The advice I’d give myself of 10 years ago
Just make stuff! Ten years ago, I was 22.
I would have been very self-conscious
about being loud, being female... I probably
wouldn’t have had, for want of a better
word, the balls to do what I was doing.
I wish I’d just sketched more, sonically.
In my late 20s, not a day went by I didn’t
sketch, because I felt like I’d just not done
it enough.
Also, just to trust my ideas. To trust my
writing. good and bad. Because I didn’t.
I played in bands and I was very much
influenced by the idea that there are
parameters around music and what it can
look like. But actually, it’s like: No! There is
always another space. And then another
space, away from that other space! And
who really cares what everyone else is
doing? I didn’t feel that at 22. But I know
that at 32.  
www.soundonsound.com / January 2023 75
 ON TE ST
Rockruepel
Sidechain.one
VCA Compressor
This quality analogue processor offers precise and intuitive
control over your compression.
BOB THOMAS
ounded in 2004, Oliver Gregor’s
F Dusseldorf-based Rockruepel Pro
Audio now offer three products: the
Comp.two, which is a high-end, dual-mono/
stereo vari-mu compressor; the Limit.one,
a stereo/dual-mono analogue mastering
limiter; and, launched fairly recently, the
Sidechain.one. The last, which is slightly
more wallet-friendly than the others, is
a mono, analogue VCA compressor, and is
named for its interesting side-chain filter,
of which more below. In developing the
Sidechain.one, Gregor worked closely
with two Grammy-nominated mastering
engineers, Germany’s Stefan Heger and
Las Vegas-based Luca Pretolesi, whose
personal contribution to the project is
acknowledged by the presence of his
StudioDMI logo on the Sidechain.one’s
front panel.
Freq Out
As is the case for all Rockruepel products,
the Sidechain.one is designed and
manufactured in Germany. Its high quality
is evident throughout — in the mechanical
design, hardware, and electronic
components, of course, but also in the fit
and finish of its physical construction, both
76 January 2023 / www.soundonsound.com
internal and external. It all inspires precisely
the sort of confidence one should expect of
a device in this price bracket.
The most obvious clues to the
Sidechain.one’s headline functionality are
two horizontally-mounted faders on the
left-hand side of its front panel. The upper
fader sets the corner frequency of the
side-chain high-cut (low-pass) filter, and the
lower one sets that of its low-cut (high-pass)
filter. Each is a 12dB/octave filter, and its
frequency range can be switched to any
one of three bands, labelled High (1.5 to
20 kHz), Mid (150Hz to 2.3kHz), and Low
(20 to 200 Hz). A pair of tri-colour LEDs
indicates which band is currently selected
for each filter. This dual-filter arrangement
obviously gives you the option of applying
variable high-pass, low-pass or (when both
faders are in play) band-pass filters, but also
means the faders can be set very precisely
around the desired frequency.
The remaining four switches, each with
‘illuminate when active’ LEDs, form a line
at the bottom of the front panel. The first
takes the Sidechain.one in and out of hard
bypass, and the second activates the Listen
function. This routes the side-chain signal
directly to the outputs, which obviously
allows you to monitor the filtered side-chain
signal being sent to the compressor, but
also means the Sidechain.one can be
used as a high-quality high-pass/low-pass/
band-pass filter. If side-chain filtering is
not being applied, it also offers a means
of comparing the compressed and
uncompressed signals.
Next comes the Link switch, which
connects two Sidechain.one compressors
for stereo operation. Once linked,
whenever the threshold of either
unit is exceeded, both channels will
be compressed simultaneously. The
compressor and side-chain filter controls
are not linked and therefore should
normally be set to identical values to
avoid anomalies in the stereo soundstage.
Completing the row of four is the
three-position toggle switch that marks the
start of the compressor control section,
and selects between three compression
ratios (2:1, 4:1 and 10:1). Its orange LED
illuminates only when compression is
actually taking place.
A trio of chunky, ribbed, circular knobs
set the threshold (+15 to -44dB), attack (0.1
to 120 ms) and release (0.1 to 1.2 seconds).
A fourth such knob sets the output gain
(0 to 15 dB), allowing you to compensate
for any drop in level due to the action
of the compressor.
The sparsely populated rear panel
carries the balanced XLR audio in/out
connectors, the Link function’s quarter-inch
TS jack socket and the four-pin locking XLR
socket that connects to the unit’s outboard
switch-mode power supply. Presumably
due to the generous side-chain filtering
on the front, there’s no provision here for
an external side-chain input or side-chain
insert point.
More Flow, Less Work
Sonically, the Sidechain.one delivers
a clean, neutral and uncoloured musical
performance that doesn’t impinge on or

change the character of the source audio.
The characteristics of its compression
and the level of control available over the
threshold and attack and release speeds
are as you’d expect to find in a high-end,
modern, feed-forward VCA compressor.
In a situation in which a low frequency
is triggering the compressor and
causing unwanted overall compression,
a high-pass side-chain filter that offers
fixed frequencies in that area (60, 125
and 250 Hz are pretty common values)
can help prevent the bass and low
midrange component of kick and snare
drums, basses and large-bodied acoustic
guitars from dominating the compression.
However, filtering out those frequencies
in the side-chain can do little to prevent,
for example, the mid to upper midrange
components of a snare drum, or the higher
frequencies of cymbal splashes, sibilants
and acoustic guitarists’ finger squeaks
from causing problems.
The Sidechain.one’s extremely intuitive,
continuously variable, yet precise approach
to side-chain filtering enables you (to
continue the kick and snare example) to
high-pass the kick and snare in the low
band to prevent unwanted compression,
while also assigning the low-pass filter
to the mid band to create a band-pass
filter, which could for example be used to
put the emphasis on the snare’s mid to
upper midrange frequencies. While the
ability to focus compression on a specific
frequency bandwidth allows you to control
wayward frequencies, it also affords
you an opportunity to explore the more
creative aspects of midrange compression.
For instance, you can use compression
and make-up gain to bring the midrange
content of a signal more to the front, or
push it backwards, whilst leaving its treble
and bass relatively uncompressed and able
to breathe.
Good as it obviously is, when I first
began working with the Sidechain.one
I didn’t quite ‘get’ it, and that ‘aha moment’
arrived only once I’d studied some of Luca
Pretolesi’s many YouTube videos, which
gave me a much better understanding
of his approach, the equipment he uses
and how he uses it. Pretolesi is known,
amongst other things, for his use of
midrange and Mid-Sides compression,
and a video of his mastering setup (as
at June 2022: https://bit.ly/3DLVwYe)
demonstrates the extent to which the
Sidechain.one has been influenced by his
“hybrid digital, analogue, in-the-box and
summing” mixing and mastering workflow.
Check out the video, and you’ll notice that
a pair of Sidechain.ones is connected,
via a Dangerous Music Liaison switching
system, to an SPL Hermes mastering
router, which can route a dual-channel/
stereo audio signal through a chain of up
to eight hardware processors. The Hermes
can also assign single processors through
either of its two parallel mix stages to
allow comparison of, for example, the
effects of two different compressors. Also
present is an analogue SPL Gemini M-S
encoder/decoder that can be inserted
by the Hermes directly into a position
in the processing chain. In essence,
this configuration allows very flexible
push-button assignment of Pretolesi’s
Sidechain.one pair as an insert or parallel
compressor on the master stereo bus for
either stereo or M-S compression.
Having understood Pretolesi’s signal
path and workflow, the Sidechain.one’s
role in his very personal mastering
process really began to make sense. His
use of midrange compression to move
the centre-panned elements of a track
forward or back, and his utilisation of M-S
based processing to widen or narrow
a stereo sound stage by altering the
Rockruepel Sidechain.one
£1649
PROS
• Mastering-grade audio performance.
• Equally high-quality construction.
• Precise, versatile and intuitive
side-chain filtering.
• In Listen mode it doubles up as
a high-quality ‘bracket’ filter.
CONS
• Side-chain insert and wet/dry control
would extend its versatility as
a standalone unit.
SUMMARY
This single-channel mastering-grade
compressor has an intuitive high-pass/
band-pass/low-pass side-chain filter,
which that allows the precise tailoring
of the frequencies which trigger
compression. Best used in pairs!
relative Mid-Sides levels, and/or adjust the
dynamics of the Mid signal without affecting
the dynamics of the Sides (and vice versa)
is creative, highly-skilled, and often
impressively subtle.
Verdict
The Sidechain.one is a lovely device.
To me, it is, first and foremost,
a mastering-grade compressor, whose
intuitive side-chain filter implementation
allows quick but surgically precise control
over which frequencies do or don’t trigger
compression. A dual-mono pair, linked or
unlinked, could be used to great advantage
on the stereo output bus of, for example,
an analogue summing mixer or, for more
detailed enhancement and adjustment,
on a M-S encoded version of that bus. To
make the most effective use of it, though,
you’d need a patchbay or routing matrix to
combine it efficiently in series or parallel
with other hardware.
Personally speaking, I’m drawn to
outboard equipment that leads me to
explore new workflows, and the Rockruepel
Sidechain.one does that. If I earned my keep
mastering tracks in the analogue domain,
as Pretolesi does, a pair of these would be
high up on my list of potential purchases,
and there’s a very high probability that
the review pair would already be sitting
permanently in my rack!  
£ £1649 each or £3199 per pair. Prices
include VAT.
T MasteringWorks GmbH +49 2236 393731
E Info@masteringworks.com
W www.masteringworks.com
W http://rockruepel.com
www.soundonsound.com / January 2023 77
 ON TE ST

Spitfire Audio
Abbey Road Orchestra:
Low Percussion Sample Library

78 January 2023 / www.soundonsound.com

Spitfire launch their new
flagship series with an
intensively-miked drum
collection.
DAV E ST E WA RT

fter spending the last 15 years

A recording a prodigious number

of sample libraries at London’s
AIR Studios, Spitfire Audio appear to
have found a new spiritual home. Two
miles due south of AIR lies Abbey Road,
legendary haunt of the Beatles and
self-styled “world’s most famous studios”.
Spitfire recorded their Abbey Road One:
Orchestral Foundations collection here in
2020, subsequently returning to record
an ongoing series of themed Film Scoring
Selections mini-libraries. Though the
producers profess their undying love for
AIR Lyndhurst Hall’s sonorous acoustic,
the studio manager may be nervously
eyeing the bookings diary.
The latest library to emerge from
Spitfire’s new hangout is Abbey Road ARLP’s dedicated plug-in’s GUI. Keyswitches are marked in green and yellow, while the lilac-coloured keys
show the sample mapping, here duplicated two octaves higher on the keyboard to facilitate double-handed playing.

Spitfire Audio
Abbey Road Orchestra:
Low Percussion
£399
PROS
• A well-chosen, powerful and hugely
dynamic low drums collection.
• The drums are precisely played with
up to seven beater types.
• Recorded from over 25 microphones
in Abbey Road Studio 1’s magnificent
enveloping acoustic.
• The Abbey Road acoustic is
superb — did I already say that?
• It has a manual.
CONS
• Performance styles are largely
restricted to straight left- and
right-hand hits and sustained
non-crescendo rolls.
• Contains no timpani, cymbals or
metals.
SUMMARY
You want powerful, highly playable
cinematic low drums with a great
dynamic range? You want a fabulous,
clear and transparent reverberant
big-soundstage acoustic? You also
want a choice of beater types, up to 10
dynamic layers and more mic positions
than you can shake a stick at? In that
case my friend, I strongly recommend
you cast an ear over Abbey Road
Orchestra: Low Percussion, the classy
first entry in Spitfire Audio’s new
flagship orchestral range.
Orchestra: Low Percussion, a collection
of 20 drums recorded in Abbey Road’s
Studio 1. Though this enormous space
can accommodate a 110-piece orchestra
and 100-piece choir (ear plugs not
provided), ARLP (as we’ll call it) features
only one performer: Joby Burgess,
veteran of film scores such as Black
Panther, Rocketman, Ad Astra and
Mission: Impossible. This multi-faceted
percussionist also created the Virtual
Marimba Choir, runs his own Powerplant
and Pioneers Of Percussion solo projects,
is a first-call session player for alt-classical
recordings and has rocked out with the
Who at Wembley Stadium — the word
‘diverse’ comes to mind.
ARLP is the opening salvo in a new
flagship series described by Spitfire
co-founder Paul Thomson as “the most
detailed and comprehensive project
we’ve ever undertaken.” In an echo of the
company’s earlier BML series, the library
will be issued in modular chunks and
will eventually comprise a full orchestra.
Thomson explains: “We’re already
recording other sections and there’s a fair
amount in post production at present.
It’s a long and exciting journey we’ve
embarked on. Making stuff at this level
of detail isn’t fast, but it all locks into the
Abbey Road One sound immaculately —
same setups, same room, same engineer,
same musicians. This project will take
a few years to complete however; it’s
our most ambitious and complex set of
libraries to date!”
Overview
The new percussion library sees the
company come full circle: their first
public release was Spitfire Percussion,
an orchestral perc collection performed
entirely by Joby Burgess. Despite
its age, the library stood the test of
time and remains a go-to item in my
personal sample arsenal (you can
read my February 2011 review on the
SOS website).
While many Spitfire libraries include
a full percussion range, ARLP is the first
to focus exclusively on large unpitched
drums — the emphasis is squarely on the
well-worked ‘cinematic’ genre, comprising
powerful orchestral bass drums, taikos,
toms and a healthy complement of world
drums (see box below for a full list).
The real detail lies in the instruments’
timbral variations and dynamic range:
a total of 20 drums were played at up
to 10 dynamics and 12 round robins
with as many as seven beater types,
creating 13 ultra-lifelike, expressive and
playable presets.
Adding to the sonic depth, Abbey
Road’s Senior Engineer Simon Rhodes

www.soundonsound.com / January 2023 79

 ON TE ST
SPITFIRE AUDIO ABBEY ROAD ORCHESTRA: LOW PERCUSSION

ARLP’s Signal Browser holds internal reverb (based on a bespoke

the library’s 16 mic positions, six
of which are visible at a time.
recorded the drums
from a large number of
microphone positions
which were subsequently
collated and mixed down
into 16 signal options
(see the ‘Microphone
Positions’ box). Each signal has its own
mixer channel fader with pan, solo and
mute controls, so you can customise
and fashion each drum’s sound to your
heart’s content.
ARLP is 88.7GB installed and runs
exclusively on its own free plug-in that
you can load into your chosen DAW or
hosting application (I ran it in Vienna
Ensemble Pro 7 with no problems).
The dedicated plug-in doesn’t support
multitimbral use, so it’s not possible to
open multiple instruments within one
instance — however, you can freely
switch between and/or layer a single
instrument’s various articulations. The
plug-in GUI perpetuates the arty design
instigated in Spitfire’s Eric Whitacre
Choir in a funereal ‘none more black’
colour scheme. Not exactly cheerful, but
I suspect after hearing these drums your
mood will brighten.
Switching from the reverberant default
mic position to a close miking clearly
demonstrates the effect of the different
beaters. Thick, club-like wooden sticks
create the familiar sharp attack of a rock
kit, hard felt mallets soften the attack
but retain definition, while medium felt
mallets blur the initial hit while increasing
resonance. The ‘super soft’ mallet hits
sound exactly as described, but playing
them at high velocities with lashings of
IR developed at Abbey Road) creates
amazing low-end impacts and throbbing
volcanic pulses. By contrast, the plastic
poly beater gives a thinner, toppy tone,
while the light brush, rute (aka hot rods)
and Hawaiian bamboo puili stick hits
sound fine when served dry.
Taikos & Toms
No one seems to know who owns the
giant taiko (aka o-daiko) drum used in
this library — it’s not the kind of item
you can overlook, standing 60 inches
tall and producing a gargantuan,
trouser-flapping bass tone when struck.
Highly resonant and rich in overtones,
this Japanese drum excels at momentous
low booms, for which the hand hits are
particularly effective. A pair of smaller
taikos work well for rhythm patterns:
the lower-pitched drum’s precise attack

Bass Drums
With the exception of one gigantic item
which we’ll come to shortly, all the drums
in this library belong to Joby Burgess — in
which case, one has to feel pity for the
cab driver hired to take him and his gear
to the studio. Setting out the stall for
‘large’ is the percussionist’s 36-inch Gran
Cassa, a two-headed orchestral bass
drum. One loud hit straight out of the
box is enough to confirm that this drum
would definitely get a thumbs-up from the
thunder gods — big, spacious, booming
and voluminous, it evokes images of
Thor’s house band doing a soundcheck in
the Taj Mahal.
Featuring a 32-inch head mounted
on a shallow open shell, a second
orchestral bass drum combines a clean
attack with a pronounced resonance, and
mirrors most of its bigger brother’s mallet
variations. The less familiar Gong Bass
Drum also has a single head and packs
a tight, hard attack not unlike a rock
kick drum.

Vintage Tama floor toms maximise the boom.

80 January 2023 / www.soundonsound.com

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WHERE LEGEND MEETS FUTURE

EIGHT 1073® REMOTE-CONTROLLABLE PREAMPS IN A 2U RACKMOUNT UNIT

THE FRONT END OF CHOICE FOR MODERN DAW-BASED WORKFLOWS

DESIGNED AND CRAFTED IN THE UK BY NEVE ENGINEERS

NEVE.COM
 ON TE ST
SPITFIRE AUDIO ABBEY ROAD ORCHESTRA: LOW PERCUSSION
and beautiful resonance are a highlight
of the library and catapulted it straight
into my ‘favourites’ folder. In addition to
skin strikes, all the taikos have clacky rim
hits which sound great in Abbey Road’s
reverb-rich acoustic.
A pair of conventional floor toms
nail the requisite ‘cinematic’ (getting
a bit tired of that word) combative
pounding racket, covering the dynamic
spectrum from subtle, barely audible
quiet strokes to manic wallops. Though
Spitfire have labelled these toms ‘epic’
(ditto), when you solo their close mics
they sound surprisingly small and dry.
That’s not a criticism, but it does illustrate
a universal truth — room sound is a vital
factor when recording drums, and this
room’s reverb tails lend a heroic grandeur
to any instrument you care to play in it.
I enjoyed the precisely played mallet
hits of Joby Burgess’ vintage Tama
Imperial Star toms, which come in 10-inch,
12-inch, 13-inch and 14-inch sizes (the epic
pair are 16 and 18 inches). The 13-inch and
14-inch models sound almost timpani-like
in the lower velocity range, while all four
sound explosive at high dynamics —
even the loud brush hits take on a manic
battering quality.
World Drums
Moving into less familiar territory, two
Chinese ceremonial dagu drums (aka
dragon drums) introduce an exotic edge.
While these large two-headed drums
Microphone Positions
Captured with over 25 mics per drum, ARLP’s
signal options outnumber those in Spitfire’s
inaugural Abbey Road One: Orchestral
Foundations library. The 16 positions are laid
out on the interface’s Signal Browser, with
(annoyingly) only six visible at a time due to the
GUI’s vast expanses of wasted blank space.
The default position for each instrument
is Simon Rhodes’ Mix 1, based around the
classic omnidirectional valve Decca Tree 1 with
additional Overheads, Outriggers 1 and a blend
of close spot mics. This makes the most of
Abbey Road’s iconic Studio 1 ambience, placing
the drums in a reverberant space tailored for
big-screen orchestral productions. Used alone,
the Outriggers 1 miking produces a wide room
ambience with a massively powerful low end,
which when harnessed to a loud bass drum hit
threatens to blow the roof off.
As you’d expect, the Ambients position
is the most distant-sounding, while the two
vintage ribbon mic options (both placed near the
conductor) also contain a fair amount of room
sound. Vintage 1 uses a pair of ultra-rare RM-1B
82 January 2023 / www.soundonsound.com
A simpler view of the signal browser allows you
to quickly adjust the listening perspective.
have an agreeably long sustain, the only
mild concern is that the lower-pitched
one’s fairly distinct pitch (a rather flat G)
might clash with your bass line. Easily
fixed though — Spitfire’s sample player
has a tuning knob! More importantly, the
dragon drums’ stick and hard felt beater
hits sound superb.
Hailing from West Africa, a pair of djun
drums also turn in a meaty thump which
works nicely for insistent warlike rhythms.
The high drum’s distinctive attack shines
through in its roll performances; you can
emphasise the beater attack by turning
up the Close 2 miking and pulling down
the reverberant mix signal. Though it’s
a single instrument, the Indian dhol drum
produces two distinctly different sounds:
the larger head is played with a heavy
curved stick and has a powerful deep
tone, while the treble head is whanged
with a thin flexible stick and sounds not
unlike a clangy Latin timbale. Use this
mics first manufactured by EMI in 1949, while
Vintage 2 features classic RCA 44-BX microphones
of similar antiquity. Considering these mics’ simple
design (basically, a thin, flexible piece of tinfoil-like
metal suspended to pick up sound waves), they
sound surprisingly hi-fi.
For a more attacking, in-your-face sound you
can feed in the Close mics, which emphasise the
splatty tap of the beater hitting the skin. Close 2
has the driest sound, while Pop Close combines
a fierce attack impact with a resonant die-away.
If you’re working with a 360-degree sound
system you’ll have a field day deciding where
to place all these miking options. The Ambients
signal is an ideal candidate for rear surround
speakers, while Section Overheads (a set of
large-diaphragm condenser microphones placed
directly above the instruments) seems the obvious
choice to pan overhead. I haven’t yet figured out
what to do with the Spill signal, which combines
the sound of all the unused mics — maybe you
could route it out to the kitchen so the rest of
your household can enjoy hearing what you’re
working on?
drum to get your Bhangra grooves going.
Over to North America, home of the
buffalo drum. This 22-inch single-headed
frame drum has a long, ringing
semi-pitched sustain and rich overtones
— you can use its soft beater delivery
for doomy single hits or switch to hard
beaters to add definition to rolls. A great,
beautifully resonant sound.
We conclude this whistle-stop world
tour with two large folk instruments: the
Argentinian bombo is a 20-inch bass
drum tuned with ropes in the manner of
a military drum. It produces a powerful
deep, earthy thud which you can use
dry in intimate acoustic arrangements,
or steeped in reverb to beef up your
cinematic productions. If it’s power
you’re after, you’ll enjoy the surdo. This
metal-bodied 22-inch bass drum’s loud
resonant thump provides the heartbeat
of Brazilian samba music, and here adds
industrial-strength power to ARLP’s low
drum arsenal.
Articulations & Mapping
Rather than try to capture every playing
style under the sun, the makers have
focused on straight left- and right-hand
hits and straight rolls, with additional
brush sweeps and rim flams available for
certain instruments. Some drums also
have a damped option where the sustain
is reduced either by hand or by that old
standby, a tea towel.
The default two-handed mapping
layout places a duplicate of the samples
two octaves higher on the keyboard,
making it easier to play repeated notes
on the same sound — this feature can
be disabled if you wish. Different beater
types and damped versions are presented
as separate keyswitchable patches, while
‘All In One’ master patches map all of
a drum’s available beaters and playing
techniques across the keyboard. Left- and
right-hand hits are mapped respectively
Preset settings control dynamic response
and velocity range, and determine the behaviour of
the library’s played rolls. As there are no rhythmic
phrases, sync to tempo is not used in this library.
 Beatles To Blockbusters
It would be remiss to mention Abbey Road and
AIR Studios in the same article without saying
a few words about Sir George Martin. Martin’s
association with the Beatles is well documented
— after their astonishingly successful alliance
at EMI Studios generated megabucks and
helped spawn a global countercultural youth
movement, EMI cashed in and astutely changed
the studio’s name to Abbey Road, the title of
the last Beatles album. The studio has dined out
on the Martin/Beatles legacy ever since — well,
wouldn’t you?
Having torn up his miserly EMI contract and
secured a well-deserved producer’s royalty, Sir
to the adjacent white notes of C and D,
but if you activate the Right/Left Mapping
feature repeated hits on the C note will
trigger an alternating right-left sticking
pattern, a subtle but telling aid to realism.
A rare gong bass drum, played here with
a variety of beaters.
George set up his own Associated Independent
Recording company and opened AIR Studios in
Oxford Street, London in 1970. A sister studio on
the Caribbean island of Montserrat was opened
in 1979, but sadly destroyed by a hurricane 10
years later. Soon after, Martin closed his West
End operation and created AIR Studios Lyndhurst
in an imposing 19th Century Hampstead building
formerly used as a congregational church —
happily, the singing continues to this day.
Without Martin’s creative vision it’s arguable
that neither studio would still exist. In 2010
EMI made an abortive attempt to sell Abbey
Road but were stymied by a public outcry and
Though the library contains no
crescendo rolls, you can create your own
dramatic surges by using the mod wheel
to control the straight rolls’ dynamics —
alternatively, the ‘Soft Takeover’ feature
bypasses the wheel and makes the roll
keys velocity-sensitive. The rolls terminate
government intervention, and had the producer
overlooked its potential, AIR Lyndhurst would by
now likely be a collection of luxury flats.
At a time when other London studios are
falling like dominoes, there’s another compelling
factor keeping Abbey Road and AIR afloat: both
contain spaces large enough to accommodate
a full-scale symphony orchestra, thus enabling
them to accommodate anything from solo piano
to large orchestras, film scores and sample
recording sessions. I’m sure the broad-minded
Sir George Martin would have smiled at the
latter development, though he might have felt
25 mics on one drum was overdoing it a bit.
with a natural end hit on note-off. When it
comes to flams and grace notes, I suggest
you program them manually using the
two-handed layout — it’s not hard to do,
and a lot of fun when you get the knack.
Conclusion
ARLP is a promising first step in Spitfire’s
new collaboration with Abbey Road
Studios. On this evidence, the “world’s
most detailed orchestra” claim is
unproven — though there’s a welcome
abundance of beater types, a good
spread of drums, deep dynamic layering
and a startling profusion of mic positions,
other percussion libraries offer more in
terms of instrumentation and playing
styles. However, a library consisting
exclusively of large drums is no yardstick
by which to pre-judge an entire orchestral
series. Let’s wait until the strings and
woodwinds start to arrive, at which point
the full musical scope of the project will
become clear!
Is Abbey Road Studio 1’s acoustic all
it’s cracked up to be? If you search for
Christian Henson’s ‘Abbey Road One,
BBCSO and Albion ONE — What’s the
Difference?’ YouTube video you can
compare orchestral ensemble recordings
made at Abbey Road, BBC Maida Vale
and AIR Lyndhurst studios and judge for
yourself. Personally I found the opulent
natural reverb of these low drums
inspirational and look forward to hearing
more examples of this modular library. In
the meantime, if you seek thunderous,
high-quality, hugely dynamic, pristinely
recorded cinematic drums, Abbey Road
Orchestra: Low Percussion is hard
to beat.
£ £399 including VAT.
W www.spitfireaudio.com
www.soundonsound.com / January 2023 83
 ON TE ST
Waves Harmony
Harmony Generation Plug-in
Waves have a knack of making sophisticated plug-ins
easy to use, and this one is no exception.
PAUL WHITE
armony plug-ins and pedals
H are nothing new, but Waves
have gone the extra mile to
make Harmony easy to operate while
still offering enough control to make
artificially generated vocal (and other
instrument) harmonies sit in the mix
in a very natural way. In addition to
producing conventional harmony parts,
Harmony can also be used in a number
of creative ways when a natural sound
isn’t what you’re after. In fact, there’s
a lot of scope for unusual production
techniques. The source audio can be
pitch-corrected with hard or soft modes
before the harmony generation, and
there are various modulation options, as
well as formant tweaking.
Overview
When setting up, the Time and Cents
controls determine how cleanly the
84 January 2023 / www.soundonsound.com
harmony generator reacts to pitch and
timing variations in the incoming audio.
There are then essentially three modes
of operation, the simplest being an
automatic chord generation function
that follows the musical pitch of the
source and conforms to the key, scale
type and chord type selected. A chords
menu below the keyboard shows the
chord type options arranged by musical
genre, including a number of EDM
variations. The musical key is selected
at the top of the screen above the
note map keyboard.
Optionally, the user can again set
the key but can then add manually up
to eight harmony voices and select their
intervals relative to the incoming notes.
If you want to check out the presets
without the key and scale changing
there’s a lock option. The graphical
soundstage display in the centre allows
individual voices to be dragged left or
right to adjust their pan position, as well
as up or down to adjust their volume.
Waves Harmony
$149
PROS
• Can be used on a very basic level or
you can dig in deeply.
• Good sound quality with believable
harmonies, as long as you don’t use
excessive formant shifting.
• A choice of operating modes.
CONS
• None.
SUMMARY
Waves Harmony is a very advanced
harmony generator that can sound
very natural, but it doesn’t come over
as intimidating and you can get great
results right out of the box.
Harmony settings can be saved as
snapshots for later use.
Specific harmony notes may also be
played in from a keyboard in real time or
from a MIDI track. A keyboard note editor
lets you choose which new notes will
play when input notes are detected, and
there are options for dealing with notes
that don’t fit the scale. It’s also possible
to combine modes, by setting up
automatic harmony generation but have
a MIDI input override it when notes are
played. There are adjustments for how
widely the harmonies are spread, and
a Flatten option to create fixed ‘pedal’
notes. More adventurous users can
also use the note map editor to create
their own note mapping presets rather
than relying on the scale and chord
settings provided. There’s no perceptible
processing delay in any mode so
Harmony is also suitable for live use.
Parameters for the selected voice
are shown to the left of the screen, with
five modulation types arranged along
the bottom. Modulation assignment is
achieved simply by dragging a modulator
button to a control. The modulation
section offers up to nine modulation
sources with LFO, ADSR and sequencer
options, with or without DAW sync. Here
you can modulate pitch, harmony voice
positions, and engage Spreader to
create parameter variations, such as pan
position changes that are unique to each
voice and determined by the Spreader
Mode control. There’s user control over
the LFO wave shapes, courtesy of Warp,
Smoothing and Phase controls, as well as
an option to use the input signal envelope
or pitch deviations as modulation
sources. A Global section controls overall
pitch, formant compensation, pitch glide,
 MAKE
overall formant adjustment and stereo spread, with output
control to control the levels of the original sound source
and of the added harmonies.

MIDI
Careful adjustment of the individual harmony voice
parameters can really enhance the way the harmonies
sound but I’d advise not overdoing formant adjustment,
because more than a slight amount can make some

WORK
voices sound noticeably processed; a setting of one or
two in either direction is usually enough. Of course you
might want some artificial flavour for creative reasons,

FOR
and there’s enough range to push things into unnatural
territory if you like. Each voice can be individually
soloed and adjusted for pitch, formant (Coarse and
Fine), delay (time or note divisions), feedback, filter

YOU
(high-, low- or band-pass), pan and level, those last two
controls reflecting what is seen in the soundstage display.
A typical setting might see the left and right harmonies
treated to maybe 40 or 50 ms of delay, with a hint of pitch
detuning and perhaps a small formant offset in opposite
directions to give the harmony voices a slightly different
character from the main voice. You can also layer simple
detuned copies of the main vocal with different formant
and delay settings for a thickening effect.
Conventional wisdom has it that musically pleasant
harmonies comprise mainly musical thirds and sixths,
but adding in a fifth at a very low level and maybe even
a sub octave can also contribute to the overall sound
without becoming too obvious. Stacking harmony
voices using small formant, timing and pitch offsets is
an effective way to create a rich ensemble sound and, CONTROL
in the context of a mix, the harmonies can be made to Control CV synths from your
sound very realistic, fitting nicely into contemporary MIDI keyboard or sequencer.
productions, especially once a little delay and reverb
has been added. Once set up, Waves Harmony can
generate such harmonies automatically, and if your
song includes a key change or a section that requires MERGE
a different harmony, one way to deal with that is to Combine the data from a number of
separate the sections onto different tracks, each MIDI sources to a single output.
treated with its own instance of Waves Harmony with
appropriate settings. Alternatively you could simply write
the harmony parts for any unusual sections on a MIDI SPLIT
track and let that control the proceedings. Split a single MIDI source into
a number of identical copies.
Verdict
The quality of pitch-shifting is good; the formant correction
applied to the shifted notes helps to keep them sounding
CONVERT
as natural as possible. In my experience, some voice types MIDI IN and OUT for USB - only
keyboards and controllers.
react more favourably than others to formant shifting
— generally, the smoother the voice, the smoother the
end result. Voices that contain a lot of unpitched throat
sound artefacts tend to sound the most ‘processed’, RETROFIT
but keep intentional formant shifting to a minimum and MIDI for your vintage synth or drum machine.
the harmonies usually work extremely well in context. New MIDI Kits now available for
the Roland SH-1000 and SH-3, and
For the advanced user, there’s a lot of deep editability
miniKORG 700 and 700S.
and assignability here, but even if you use one of the
three modes at its most basic level, or simply explore
the presets, you can achieve excellent results. This is
one plug-in that I use Kenton on everything I do live. They are
£ $149 (discounted to $39.99 when is going to make R^LTYTKTWFQQYMNSLX2.).9TSXTK‫ܫ‬J]NGNQNY^
going to press). a lot of producers and so reliable. Perfect on any stage.
W www.waves.com very happy.   Matthew Carroll, Keyboards for Dua Lipa

Contact the MIDI Specialists

kenton.co.uk
 FE ATURE

the Goldfinger records, signed

two bands, and the Used Hear The Sound
were the third band I had
produced. In the studio I was W https://open.spotify.com/
track/5jZ1Z2GFTf2gwmFc3qiUxs
experimenting with a lot of
W www.youtube.com/watch?v=2_
different production styles J2HmI48N4
between two-inch tape, digital
tape, ADATs, and everything
else. I ended up using Pro
Tools for the Used self-titled
debut album which this track
comes from.
“I had purchased a vintage
AKG C12 and this very thin
pop filter. A lot of the plosives

John Feldmann: were coming through because

the pop filter was so very thin.
Every time Bert [McCracken]
The Used ‘The Taste Of Ink’ would accentuate a ‘p’ it
would just blow the mic up! Yet I used, was kind of pieced together.
J O E M AT E R A
I thought it fit the music very “Finally, I used a Softube compressor

W
hile having served as the founding
lead vocalist and guitarist for American
ska-punk rockers Goldfinger since the
early 1990s, John Feldmann has also etched out
a successful career as an in-demand producer and
mixing engineer. He has worked with the likes of
Korn, Blink-182, 5 Seconds Of Summer, Biffy Clyro,
the Used, and most recently with Canadian pop-punk
rocker Avril Lavigne. Here, he details how he got
his favorite vocal sound on the Used’s 2002 debut
single ‘The Taste Of Ink’.
With Intent
“I chose this track because of the unique vocal
sound. At that point in my career, I had only produced
well, so we did it with intent.
Bert was just millimetres away
from the vocal mic.
“Then I put that through
a Universal Audio 6176, which
has a 610 tube microphone
preamp and an 1176
compressor, so it’s a hybrid
of those two units. I had no
money at the time, being in
a band and still touring in
a bus. I had just gotten a small
publishing cheque, but every
penny went into buying the
microphone. Everything else
plug-in, which I love on vocals; I went with
a 4:1 ratio and went pretty hard on the
input so it would compress hard on the
takes. I didn’t want to leave it up to any
mixing engineer as I wanted to hear every
breath, like you do when you’re in the
studio. I also had a trash can next to the
mic because Bert screamed so hard, he
would throw up sometimes! It was a very
intense recording process.
“That was pretty much my chain for
that vocal sound, and I didn’t use any EQ
going into Pro Tools. It was basically just
the 610, the 1176, and in Pro Tools I had
the Softube compressor.”  

86 January 2023 / www.soundonsound.com

THE
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©2022 All Rights Reserved, Lindell Audio.

 TECHNIQUE
Keeping Synths & Grooveboxes In Time
Going DAWless presents many challenges — not least
keeping all your gear in sync...
CHRIS KORFF
D
esktop noise-makers have
never been cheaper. For almost
literal pocket money, a music-
production novice can pick up a drum
machine, sampler or groovebox and
start making a racket. It rarely stops
there, though: that first bit of kit can end
up being a gateway purchase, with the
next week’s paper-round money already
spent on a new device before the first
has been mastered.
Other than the ‘running before you
can walk’ issue, this can lead to another
problem: how to get multiple devices
to play in time. Unlike with DAWs and
USB peripherals, keeping separate
hardware boxes in sync with each
other without a computer is far from
an automatic process. In this article,
I’ll explain the various ways in which
synths, sequencers and grooveboxes
88 January 2023 / www.soundonsound.com
can be kept in step, using a few popular
machines as examples.
Jacking Up
I’ll start simple, with the ‘sync’ socket that
many affordable synths, drum machines
and the like now sport. There’s no magic
to it: sync is a simple analogue pulse
(around +5V) that plays in time with
whatever device it’s coming from, to
tell the next device in the chain when
to play its next notes. Fundamentally,
it’s a metronome, and its blips are
transmitted using a jack cable (usually
a mono mini-jack).
Here’s a nice easy example. I’ve
hooked up a Korg Volca Kick’s sync
out jack to the sync input on a Korg
Monologue. Now, if I hit Play on both
devices at the same time, they will both
play at whatever tempo is selected
on the Volca. Easy! Even better, their
synchronisation is an ongoing process,
so if I tweak the tempo on the Volca
Kick, the Monologue will follow suit. And
because both devices have both a sync
input and output, I can swap them around
so that the Monologue determines the
tempo rather than the Volca, if that’s
more convenient. (At this point, I should
mention that one reason this works so
seamlessly is because they are both Korg
devices, and therefore use the same sync
‘ppqn’ standard, as explained in the ‘On
The Pulse’ box.)
Now, this is about as basic as
synchronised setups get, and it has its
limitations. For one, you have to press
Play on both devices simultaneously,
because the sync pulse on these Korg
devices is pervasive; it begins the moment
you turn them on, it doesn’t start and stop
when the sequencer does, and there’s
no additional signal telling downstream
devices when to do so. Some devices,
like Behriner’s RD-series drum machines,
and Arturia’s MicroFreak, MiniFreak and
BeatStep Pro, can make use of a TRS
cable to send start and stop messages
to other kit alongsidse the sync pulse,
but the usefulness of that depends on
whether your other kit will heed those
messages. And, in any case, if you
accidentally hit Stop on the downstream
device, the lead device will continue
playing. To get them back in time again
you’ll need to carefully time your next
hit of the following device’s Play button,
otherwise the two will be offset.
This kind of transport-control
jockeying will be familiar to anyone
who’s ever DJ’ed, and the skill of it
(and occasional happy accidents when
you get it wrong) can be a fun part of
the DAWless experience — but if your
sync’ing needs are more complex, and
especially if you have a sequencer or
workstation with a proper ‘song mode’,
you’ll need to turn to...
MIDI Clock
MIDI is good for more than just shuffling
notes about. Part of its specification
BAck F rOM THE Dead...
Modern Korg gear is easy to
synchronise: just connect the
sync out jacks to the sync ins,
and your devices will maintain
the same tempo.
includes something called
MIDI Clock, which works
much like sync but can also
include other messages,
such as transport control
and Song Position Pointer
(SPP) data. It can be used
alongside other MIDI
messages to convey things
like time signature. The
transport messages, where
present, include start,
stop, continue, rewind and
fast-forward commands,
while SPP tells devices not
only what tempo to play at,
but where in the sequence
or song to play from. Note,
though, that not all MIDI
devices pay attention to the
extra data: simpler devices,
such as monosynths with
January 2023 89
Find a dealer @ nova-distribution.co.uk sales@nova-distribution.co.uk
tel: 0203 150 4446
 TECHNIQUE
DAW LESS SYNC
Modular gear can also join the sync party —
provided you send it the right signals...
no more sequencing ability than a basic
arpeggiator, may simply use MIDI Clock in
much the same way as a sync signal.
Robin Vincent’s recent article about
setting up MIDI (SOS November 2022:
www.soundonsound.com/sound-advice/
midi-hardware-studio) covered the
fundamentals of MIDI connections and
MIDI Clock, and is well worth a read if you
need a recap. I won’t add much except
to say that sync pulses and MIDI Clock
can happily be used at the same time:
I can slave my Monologue to a DAW, or
a hardware workstation like an MPC,
On The Pulse
Older readers will probably be familiar with
the term ‘ppqn’, which stands for pulses
per quarter note. In the days before MIDI,
different manufacturers had their own ideas
about how best to synchronise things, and
one of the primary differences between them
was the ‘resolution’ of their sync signals. As
the world adopted MIDI Clock as the de facto
standard, the specifics of who was using
what ppqn stopped mattering, but with new
analogue instruments being released almost
daily it’s back in the spotlight.
An instrument’s ppqn defines how many
pulses there are for each crotchet it plays.
So if you’ve got a drum machine playing
and it outputs sync at 4ppqn, there’ll be four
pulses for each quarter note (crotchet), or
16 pulses in a 4/4 bar. If the device you’re
sending your drum machine’s sync to is
also running at 4ppqn, then everything will
work as it should. However, if it runs at, say,
2ppqn (which seems to be the standard that
Korg have adopted since the Volca series),
then it will only play one crotchet for every
two that your 4ppqn drum machine does.
90 January 2023 / www.soundonsound.com
using MIDI Clock, and then send its sync
pulse out to a Volca, and everything will
play in time.
Modular
Sync and MIDI Clock are the two most
common ways of keeping the new
generation of standalone grooveboxes
and synths locked together, but what of
the murky world of modular? Anyone who
has thrown themselves down the Eurorack
rabbit hole will probably already be familiar
with the various sync’ing solutions, but
if you’re growing a DAWless setup and
wondering whether a modular system
could slot into it, the general gist is this: any
module that can be synchronised with other
modules or external gear — a sequencer or
arpeggiator, say, or a tempo-based delay
module — will have a ‘clock in’ socket
(and often a ‘clock out’ too). These ‘clock’
signals are very similar to the syncs found
on the Volcas, Boutiques, BeatSteps and
TR-style drum machines of this world, in
that they are both regular analogue pulses.
Indeed, these two ‘standards’ (such as they
are) are interoperable; you can often slave
a modular sequencer to a drum machine’s
sync signal, or vice versa, quite happily...
with a caveat.
The chief difference between sync and
modular clock is that the latter is essentially
a ‘play the next note now’ message,
whereas sync signals usually have several
blips (‘pulses’) per quarter note. To
illustrate, let’s say you’ve connected the
sync out from a Korg Volca (which runs at
2ppqn) to the clock input on a sequencer
module. Assuming you’ve programmed
the sequencer to play quarter notes, it will
essentially run at double time, because it’s
receiving two ‘play next note’ messages
for every quarter note that the Volca plays.
This can be worked around easily enough
— by spacing out your Volca notes, say, or
treating the sequencer notes as quavers
rather than crotchets — but if the device
generating your sync signal runs at 28 or
even 48 ppqn then your sequencer will be
in machine-gun territory, and unless you’re
exclusively into gabber then that probably
won’t be very helpful.
In that case, you’ll need what’s known
as a ‘clock divider/multiplier’ module,
which does exactly what its name
describes: it takes an incoming clock
signal, and then divides or multiplies
it so that, for example, a 4ppqn clock
becomes a 2ppqn one (ie. it’s divided by
two) or an 8ppqn clock (multiplied by two).
Such modules are available individually,
but some Eurorack sequencers,
like the Eventide Misha, have clock
divider/multipliers built in, allowing them
to conform to a wide range of standards
— as well as letting you get creative with
half/double-time sequencing.
Modular sequencers may also include
inputs for Trigger and Reset signals. The
former is much the same as the ‘start’
messages that you get from TRS sync
jacks (indeed you can use those signals
with modular, if you have a correctly wired
Y-cable), while the latter simply starts
the sequence at the beginning — useful
if you’ve got yourself out of step with
another sequencer, or you want to play
with polyrhythms.
A final word on modular sync’ing:
clocks, like sync signals, are nothing
 Triggers
Sometimes it isn’t necessary to keep two
devices in time per se, merely to get one
device’s sequencer to trigger notes on
another. In MIDI-land you do this by assigning
corresponding MIDI channels and note
numbers to both devices, but in the analogue
world it’s done using the aptly named trigger
inputs and outputs. To take the Behringer
RD-8 as an example, it’s possible to use the
sequencer lanes dedicated to the accent,
hand-clap and cowbell to trigger external
gear — for example a snare module in your
Eurorack setup, or any other synth that has
a trigger input. Indeed, trigger, gate and
clock signals are somewhat interchangeable,
depending on what you’re doing. Be aware,
though, that there are different standards
for trigger voltages among different
manufacturers: you may need to boost
a trigger using an amp module in order to get
the receiving device to register it.

Retrokits make a range of hubs and converters

more than regular analogue pulses —
which are precisely what you get out
of a square-wave LFO. So, it’s perfectly
possible to use one of your modular synth’s
LFOs as a clock source, with the LFO rate
essentially functioning as a tempo control.
Conversely, you can also use the clock
signal itself in other parts of your modular,
to trigger envelopes and so on.
I Sync, Therefore I Am
So, keeping your gear in sync can
occasionally be very straightforward,
and occasionally less so, depending
on what gear you’re using and what
your requirements are. If you find
yourself trying to sync to pieces that are
‘incompatible’, though, worry not: you’re
far from the first person to experience this
problem, and there are various gizmos
around that can help. Special mention
should go to Retrokits, whose RK-006
hub can help keep your USB, MIDI, sync
and modular gear all running in time.
Before the days of DAWs and USB
connections, knowing how to synchronise
your gear was the foundation of setting up
an electronic music studio. Indeed, it was
rather more complicated back then, with
analogue synth manufacturers employing
that can get your gear playing in time, whichever
sync standard it uses.
their own standards for sync signals, and
MIDI still in its infancy. And if you wanted
to record audio alongside your synths and
drum machines, you had to dabble in the
arcane art of SMPTE sync, sacrificing one
of your tape channels for timecode so that
your MIDI sequences and tape recordings
stayed in time.
So, no, setting up a studio without
a computer isn’t easy, and never has
been. But if you wanted an easy life you’d
just use a DAW... Wouldn’t you?  

www.soundonsound.com / January 2023 91

 ON TE ST
Roland Fantom-0
Workstation Synthesizer
The Fantom-0 squeezes the soul of its flagship big brother
into a compact and affordable package.
DAV E ST E WA RT
s intrepid TV ghost hunters terrify
A their viewers with dimly lit scenes
of bogus spooky happenings, there
nevertheless exists genuine evidence that
revenants walk among us. These lonely
creatures seemingly never die, but return
to haunt us again and again in a variety of
shape-shifting forms. As ghosts go, they’re
surprisingly amenable and house-trained
— in fact, you can buy one on credit from
your local music shop and take it home to
entertain your friends and family. I refer
to the Roland Fantom, a popular and
enduring keyboard which has restlessly
roamed the Earth for the last 20 years.
The latest manifestation of this
deathless apparition is the Fantom-0,
available in Fantom-06 (61-note),
Fantom-07 (76-note) and Fantom-08
(88-note) models. The instrument
is a direct descendant of Roland’s
2019 flagship Fantom keyboard
(aka the ‘big Fantom’), a pro-level
workstation-cum-synthesizer which
I reviewed in SOS November 2019.
92 January 2023 / www.soundonsound.com
Though impressive beasts, the big
Fantoms are expensive, large and
heavy: the 88-note version weighs 28kg
and is over 43cm deep, which would
overshadow anything placed beneath
it in a multi-keyboard stage rig. No
doubt conscious that this costly and
back-breaking unit might not be the first
choice for someone playing a Tuesday
night pub gig, Roland have created
a more affordable slimmed-down version.
The new Fantom-0 series combines
the original’s sound technologies and
open performance architecture in a light,
streamlined and competitively priced
instrument with potential appeal for
professional, semi-pro and hobbyist users.
For this review I examined the 61-note
Fantom-06, which is identical to its 76-note
and 88-note siblings in all respects bar the
number of keys and keybed style. Since
its design and functions largely replicate
those of the big Fantom, you might care
to read my review of that instrument
at www.soundonsound.com/reviews/
roland-fantom. Rather than repeat its
details, I’ll concentrate here on explaining
how the new Fantom-0 instruments differ
from their big brothers.
Overview
The main reason for the new Fantom-0
keyboards’ dramatic weight loss is the
substitution of a plastic chassis for their
predecessors’ tough metal body. The
Fantom-06 and Fantom-07 models also
shed extra ounces by utilising a standard
unweighted synth-action keybed. While
this light action is perfectly serviceable,
responds well to rapid note reiterations and
is relatively noise-free, I found it easier to
articulate fast runs on my old Fantom-G6,
rattly keys and all — by comparison, the
new keybed feels a tad spongy and
resistant. Maybe it’s just me? In any case,
strong-fingered keyboardists may prefer
the Fantom-08’s PHA-4 Standard weighted
keyboard, which simulates the ‘escapement’
bump and ivory feel of a grand piano. The
keys are full size and standard width, but
keyboard aftertouch is not implemented in
any of the models.
In the interests of affordability certain
features have been scaled down: the
Fantom-0’s colour touchscreen display is
5.5 inches from corner to corner (down
from the big Fantom’s 7 inches), the 16
patch-select/step-sequencer buttons are
smaller and there are no front-panel ADSR
envelope controls. The USB audio slider
has also been removed. Other than that, the
panel layout is essentially unchanged.
Back-panel audio connections consist
of a stereo headphone jack, main

output L/R balanced jacks, auxiliary
sub output L/R jacks, a sub out stereo
headphone mini-jack, a mono mic input
jack and line input L/R jacks. There
are no XLR connections (though the
online documentation claims otherwise)
and no CV/gate connections. For the
fashion-conscious, the big Fantom’s
somewhat incongruous bright red rear
panel has given way to a tasteful black,
though small flashes of red here and there
will help audiences identify the make
of keyboard.
Standard MIDI connections consist
of a single in and out (the latter doubling
as MIDI Thru, selectable in a front-panel
menu). You can send and receive MIDI
and audio to and from your computer and
other devices via USB external device and
computer ports. An additional USB memory
port is available for transferring sample
and patch data. Back in the pre-computer
world, there’s a jack socket for your sustain/
hold pedal (which supports half-pedalling)
and two further jacks for pedal controllers,
which can control MIDI volume, expression,
patch change, sequencer start/stop and
a host of other functions.
Lurking inside the product box you’ll
find a wall-wart mains adaptor (cue
massed booing) and a printed owner’s
manual, which reiterates its 35 pages
of text in eight languages, hopefully all
saying the same thing. This simple guide
is supplemented by several enormously
detailed PDF documents (including sound
lists), which you can download from
Roland’s website. Gigging keyboardists will
be pleased to know the Fantom-0 boots
up in 20 seconds, a welcome improvement
on its predecessor’s 40 seconds (though
less time to enjoy the psychedelic
flashing-lights display).
New Sounds
In musical terms, the most striking
difference between the Fantom-0 and
the previous Fantom is the absence of
the V-Piano engine, a modern recreation
of the 2009 Roland digital piano, which
weighed 38kg and cost an arm and a leg
(happy days!). In its place are two new
User Sampling
Roland implemented keyboard sample mapping
for the flagship Fantom series in September
2020, a year after its initial release. Since the
Fantom-G already supported the feature in 2008
the 12-month delay seemed strange, so I was
glad to see that the Fantom-0 incorporates user
keyboard sampling straight out of the box.
The Fantom-0 matches its big brother’s
maximum sampling time of 2 minutes 44 seconds
and supports a total of 128 multisamples. Both
keyboards have 2GB of storage for samples
played from the pads, an area of memory that’s
also used for loaded but inactive samples you’re
saving for later use. The maximum number of
Fantom-0 user samples is 2048 against the big
Fantom’s 8000, but the really bad news is that
only 256MB of sample memory is allotted for
all multisampled instruments, Roland model
expansions and EXZ expansions active in the
Fantom-0. On the big Fantom this area is 2GB,
Roland Fantom-0
From £1249
PROS
• Offers all the advantages of the
flagship Fantom range at a greatly
reduced price.
• Contains a huge range of infinitely
customisable sounds and effects.
• Supports user keyboard
multisamples.
• Optional Model Tones recreate the
sound of classic Roland synths.
• The light and compact design takes
the strain out of transport.
CONS
• User keyboard multisamples are
limited to 256MB.
• Too bad the V-Piano engine isn’t
included.
• Wall-wart power supply.
SUMMARY
Now incorporating a modelled
tonewheel organ, two new
SuperNatural pianos and an onboard
vocoder, the Fantom-0 keyboards
incorporate the sounds and features
of the flagship Fantom range in
a slimmed-down package that won’t
break your back, at a price that won’t
break the bank.
SuperNatural pianos originally released
as expansions for the earlier instrument.
(The ‘SuperNatural’ tag refers to Roland’s
proprietary behaviour-modelling
technology developed for emulating
acoustic instruments).
Though neither of the SuperNatural
pianos quite reaches the heights of
the V-Piano, both sound fine and play
beautifully smoothly and consistently
so you can fit in many more expansions and
user multisamples.
This miserly allocation seems
incomprehensible. Roland forum members share
my disappointment: while not everyone needs
to use a keyboard workstation to play their
mapped samples, those that do regard 256MB
as laughably restrictive and backward-looking.
Some users reiterate the longstanding complaint
that while today’s lush-sounding orchestral
libraries often stream hundreds of GB of sample
data, workstation manufacturers eke out their
sample space in relatively tiny amounts that are
not conducive to top-quality musical results.
The unspoken message seems to be
that players who rely on creating their own
multisamples for live performance will need to
stump up for a flagship Fantom model, but this
coercive approach may have the effect of driving
buyers into the arms of a rival manufacturer.
www.soundonsound.com / January 2023 93
 ON TE ST
ROL AND FANTOM-0

Fantom-0 keyboards have two dedicated USB ports for

sending and receiving MIDI and audio to and from your computer
and external devices, with an additional USB memory port
available for transferring sample and patch data.

tones. The big guy has 272 scenes,

across their full register and dynamic
range, an advantage of modelled
instruments over their conventionally
multisampled counterparts. I felt piano
number one’s big sound and strong
attack would suit loud pop/rock tracks
and dramatic orchestral settings, while
the softer, warmer tone of the second
piano works better for quiet ballads and
improvisatory playing.
A welcome addition to the Fantom-0
is the VTW (Virtual Tone Wheel) organ.
Introduced as an update in big Fantom
v2.50, this mimics the classic Hammond
drawbar layout with an interactive picture
of nine ‘harmonic bars’. You can use the
top-panel sliders to control bars 1-8, and
adjust number 9 (the all-important highest
harmonic) on the Tone Edit’s Wheel page,
or by using the touchscreen to drag it up
and down.
The organ (which works in Zone
2 only) uses four similar-sounding
tonewheel types: Vintage 1 and 2 are
modelled respectively on 1970s and 1960s
tonewheel organs, ‘Solid’ emphasises
Vintage 1’s low end and ‘Clean’ removes
the high-pitched ‘leakage’ noise which
affects old organs, considered by some
to be a desirable characteristic. The VTW
has all the usual Hammond bits and bobs
(second and third percussion, vibrato and
chorus) and three overdrive types, one
inherited from Roland’s discontinued VK-7
combo organ. It also has a rotary effect
which can be set to the ‘sweet spot’ speed
between fast and slow, a nice touch.
Verdict: a decent and highly customisable
Hammond clone.
Speaking of vintage, I was pleased to
see a vocoder included. My old Roland
SVC-350 vocoder is a wonderful machine,
and this modern-day emulation (originally
included in the big Fantom update
1.5) nails its tone and functions. The idea
is you select an onboard ‘carrier’ wave
(a bright, full string pad works well) and
trigger it by speaking into a microphone
(not provided), thereby producing the
classic Daft Punk/Sparky’s Magic Piano
(depending on your age) robotic pitched
vocals. You can shape the tone with a set
of 13, 20 or 30 band-pass filters (13 sounds
best) and get great effects by vocoding
the carrier with a drum machine, a unique
percussive racket.
Tone Trove
At the time of the big Fantom’s release
Roland were unable or unwilling to publish
a sound list, so your overworked scribe
had to add up the number of scenes
(combis/multis), tones (patches) and waves
(instrument multisamples or drum hits) on
his fingers. Seven updates of the original
instrument later, it can now be revealed
that both big and small Fantoms currently
contain 2108 waves, 3667 Zen-Core tones,
70 SuperNatural tones (not originally
included in the big Fantom) and 91 drum-kit
the diminutive upstart 283.
I must confess I haven’t listened to
all these sounds, but I did check out the
Fantom-0’s new material, including 30
or so scenes which take the place of the
former V-Piano-based setups. ‘Sure It
Is’ and ‘Jazz Café’ serve as a reminder
that this keyboard has one of the best
workstation Rhodes samples out there.
You can add effects and EQ to taste, but
the basic starting sound is playable and
authentic. The second scene features
a keyboard split with bass guitar at the low
end and a subtle octave vibraphone layer
at the top, both of which can be adjusted
to your requirements. Upholding Roland’s
reputation for strong bass sounds, the
Fantom-0 contains some excellent fretless
and slap-bass variants.
Moving swiftly from jazz to hip-hop,
‘Hop On Over’ contains a great, booming
sub-bass-cum-kick boosted by the built-in
Lo Fi Compressor effect, while ‘Remember
When’ brings back the ’80s with a stonking
Minimoog-inspired synth bass. The chaotic,
frankly mad ‘Wobble HYP’ confirms the
Fantom-0’s virtual-analogue credentials,
combining a screaming rave lead and
distorted dubstep bass with a tear-arse
overdriven single-note synth pulse and
Now equipped with a piano-roll
editor, the onboard sequencer enables
Ableton-style clip-based programming
and on-the-fly changes.

94 January 2023 / www.soundonsound.com

 ON TE ST
ROL AND FANTOM-0
The user sampling feature allows you to create your own keyboard-mapped multisamples.
Doctor Who sci-fi effects. On a calmer note,
there’s ‘Synth Utopia’, an addictive, driving,
tempo-sync’ed vintage synth pulse, which
can be improved by turning off the daft
novelty bass sound in Zone 1. Composers
of TV nature documentary soundtracks
will enjoy overusing the mysteriously
beautiful ‘Glacial Period’ soundscape. The
magic happens in Zone 3’s ‘Horror Pad’
tone, which skilfully weaves three waves
into a soft, haunting, utterly non-horrific
minor-chord texture.
All this reaffirms the fact that the big
Fantom and new Fantom-0 are immensely
programmable instruments. Though you
can’t alter the basic waveforms, you can
edit any of the tones (which contain up to
four waves) and scenes (which contain up to
16 zones, each of which holds a single tone)
and save your edits, for which purpose
a large number of empty slots are provided.
If you’re unused to synth programming,
I recommend you watch the tutorial videos
on Roland’s Fantom-0 YouTube channel,
in which the avuncular Ed Diaz patiently
explains the basics.
Expansions &
Technical Changes
A key selling point of the rejuvenated
Fantom is its potential for growth (you
listening, Liz Truss?). Roland’s Axial sound
library site offers 15 EXZ mini-libraries as
free downloads which can be added to
the Fantom-0 as expansions. The contents
range from drum kits, acoustic and electric
pianos, brass, strings and orchestral sounds
to dance loops and world instruments.
A more recent development is a set
of optional Model Tones instruments,
which recreate the sound of iconic Roland
analogue synths, each with its own unique
96 January 2023 / www.soundonsound.com
set of parameters. The models are
Jupiter-8, Juno-106, JX-8P and SH-101, all
available as downloads from the Roland
Cloud (the n/zyme wavetable synth model
currently doesn’t work with the Fantom-0).
These vintage timbres are a great way of
expanding the Fantom-0’s tonal range, with
the Jupiter-8 model leading the way in lush
string pads.
Certain under-the-bonnet technical
changes are worth noting: the Fantom-0
has less processing power than its older
relative (two BMC chips instead of four), less
internal memory (see the ‘User Sampling’
box) and a different D-A converter (type
unknown). The big Fantom’s analogue filter
has gone AWOL, but that’s no cause for
alarm: it was only featured in a few original
presets, and the Fantom-0’s regular filter is
as powerful as ever. Lastly, the Fantom-0’s
seamless-transition ‘Tone Remain’ feature
(now renamed ‘Scene Remain’) only works
with zones 1-8; when activated, zones 9-16
fall silent. On the plus side, polyphony
remains at 256 voices, the sequencer
(which can play MIDI files) has new piano-roll
editing and automation features, and the
DAW integration now includes Ableton Live.
Conclusion
If any of the economy measures
affecting the Fantom-0 cause you
concern, let’s consider what you get for
your money: a compact, contemporary
workstation-cum-synthesizer equipped with
a colour touchscreen, sample pads, active
front-panel synth controls with variable
filter types, a TR-808-style step sequencer
with illuminated switches, an Ableton-style
clip-based sequencer enabling on-the-fly
changes, an arpeggiator, a chord memory
feature and integration with DAWs and soft
Age Of The Fantom
Leading paranormal investigation magazine
Sound On Sound has tracked the Fantom’s
progress since Paul Nagle reviewed the
Fantom-FA76 back in February 2002.
The product line developed quickly: Nick
Magnus’ positive October 2003 appraisal
of the Fantom-S/S88 was followed by the
release of the Fantom-X (see SOS September
2004), which set the template for 61-, 76- or
88-note models and also boasted a rack
version (I miss those things). Progress then
slowed, and after Gordon Reid dissected the
scaled-down Fantom-Xa in SOS May 2005
there were no further confirmed sightings
’til the release of the Fantom-G, reviewed by
your humble scribe in January 2009.
Fast-forward five years, and Gordon
Reid’s back in action explaining the
ins and outs of Roland’s affordable FA
series keyboards, while debunking the
accompanying hyperbolic publicity blurb. The
Fantom trail then went mysteriously cold,
leading one to assume the brand had ceased
to be, popped its clogs, kicked the bucket,
shuffled off its mortal coil, expired and gone
to meet its maker (etc). But no — it turns
out the instrument was merely resting. With
little advance warning, Roland announced
a new flagship professional keyboard in
Autumn 2019. Titled simply Fantom, this
workstation-cum-synthesizer built on its
predecessors’ legacy with a brand-new
design developed from the ground up, as
described in my SOS November 2019 review.
synths over USB. Then there’s the large
library of acoustic and electronic sounds
including a good Hammond clone, two
modelled pianos, a vocoder, a collection
of iconic Roland drum machines, user
keyboard sampling, an expandable sound
engine with access to classic Roland
analogue synth recreations, 90 built-in
effects and thousands of presets, all
harnessed to deep programming facilities.
And have you seen the prices? Come on.
It was clear from the outset that the
Fantom-0 would be a popular keyboard
(a fact borne out by the inordinate amount
of time it took this writer to get his hands on
one). Some may be tempted by the slightly
superior specs of the flagship Fantom, but
for most buyers the £2k price increment
for each model will be the clincher. This
combination of musical versatility and great
value for money seems bound to ensure
that the new slimmed-down Fantom-0 will
never die.
£ Fantom-06 £1249, Fantom-07 £1499,
Fantom-08 £1599. Prices include VAT.
W www.roland.com
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 ON TE ST

Interphase Audio
EQ Modules Analogue Equalisers
M AT T H O U G H TO N

Compatible with both the 500-series and
the 51X modular formats, these classy
Allan Bradford-designed modules offer up
analogue magic in spades.
owards the end of 2017, Bruno Wynants, founder/owner of
T Belgium-based Interphase Audio, and gear designer Allan
Bradford, visited us at SOS Towers to demonstrate their new Ark
modular analogue console, a headline feature of which is its ability to host
500-series modules in line with its own channels. We found ourselves

98 January 2023 / www.soundonsound.com

impressed, and Hugh Robjohns remarked
that: “It appears to be a very well-conceived
and beautifully engineered console,
designed with considerable flexibility in
mind. The set of modules available at
launch is equally impressive and should
meet the needs of most clients, but there
are advanced plans for additional future
options.” Interphase have managed to sell
and install several consoles in a range of
different configurations since then, despite
the logistical difficulties presented by the
pandemic, and have recently started to
deliver on their promise of new modules.
The first trio, reviewed here, are all
equalisers, and there are plans in place to
follow these up with a mic preamp and an
intriguing multi-type analogue compressor,
amongst other things.
Overview
As supplied, the three EQs, named the
Iridium, the Carbon and the Helium, come
ready to plug into any 500-series chassis
but, unlike most such modules, they aren’t
limited to use in 500-series racks: a small
adaptor can be fitted onto the rear to
provide the additional connectors required
by 51X racks. Popular in the pro-audio DIY
community, the 51X format augments the
usual 500-series connections, adding ±24V
power rails, which allow module designers
to develop products with greater headroom
— they can accept and output higher-level
signals. It’s perhaps not a surprise to learn,
then, that one of the other products in the
Interphase pipeline is a chassis capable of
hosting 500-series and/or 51X modules; to
date I’ve only seen these as DIY projects or
built on a very small scale.
All three of these modules are
fully discrete Class-A devices, whose
beating heart is Allan Bradford’s Top
Amp, a low-noise, low-distortion discrete
op-amp that he developed specifically
to be more compact and better suited to
mass-manufacture techniques than the
other boutique options out there. He tells
me his design is guided by the ‘wire with
gain’ ideal and that, in developing the Top
Amp, he conducted extensive listening tests
of a range of op-amps, both IC-based and
boutique discrete; he says that on good
speakers it’s possible to discern a significant
difference between them. The chief
practical benefits of his compact low-noise
design are that it can be manufactured
at scale (great for building mixing
consoles!), and it’s easy to accommodate
several of them in small devices, such
as these modules.
The review units were all very nicely
constructed, exhibiting a solid feel and
a slick finish. The circuit boards are
populated with high-quality components,
of course, and are surrounded by a folded
metal case with a small slot at the rear
to allow the card connector through.
Considering the number of knobs on the
front plate, I was pleased with the amount
of space available, and never felt as though
my fingers would accidentally move an
adjacent control. The aesthetic is pleasing
to the eye too — very ‘hi-fi’. That said, while
the legends are clean and clear, the small
black-on-grey typeface could make it tricky
to read everything in a dimly lit studio, and
when the units are placed below eye height
much of the information is obscured by the
tall knobs and their shadows.
Three Elements
The Iridium is essentially the same EQ as
that created for Interphase’s Ark console
— a four-band inductor-based design.
Each band has a knob to switch the centre
frequency and another that turns a ±15dB
gain pot, so there are eight knobs in total,
and the only other control is a button
described on the panel with the words
Discrete Class-A. It’s an accurate description
of the electronics, but this backlit (green)
button is in fact an in/bypass control. When
the unit’s in bypass, the signal still flows
through the balancing/unbalancing circuitry.
This Iridium’s upper band can be set to 8,
9, 10, 12, 14 or 16 kHz; the higher mid band
to 1.5, 3, 5, 6, 7 or 8 kHz; the lower mid band
to 125, 250 or 500 Hz or 1, 2 or 4 kHz; and
the low band to 50, 80, 100, 150, 300 or
600 Hz. Each band is a peak/bell type with
a fairly broad curve (a shade over an octave,
the peak becoming narrower the more
you boost/cut), and it exhibits a pleasingly
vintage sort of vibe, that I’ll describe
subjectively below. Despite the coloration
harking back to days gone by, audible noise
is notable only for its absence.
The Carbon is another four-band EQ,
but this time all four bands are based on
resistor-capacitor (R-C) networks. The two
middle bands are bell/peak filters, while the
two outer bands are switchable between
bell/peak and high (top band) and low
(bottom band) shelving filters. There’s the
same bypass button, and two more such
buttons (orange-backlit this time) are used to
select the outer band types. There’s much
greater overlap between the bands than on
the Iridium and the frequency controls are
continuously variable rather than switches.
So while the Carbon may lack the other’s
Interphase Audio
EQ Modules
PROS
• All sound lovely, but the inductor
bands particularly so.
• Stylish and beautifully constructed.
• Great to have different options.
• Compatible with 51X and 500 series.
CONS
• Legends could be easier to read.
SUMMARY
This classy trio of EQ modules all
sound great, but the ‘analogue magic’
lies mostly in the inductors of the
Iridium and Helium.
silky charm, it is rather more versatile — and
it sounds gratifyingly clean too. The top
band can be set anywhere from 1.2 to 22
kHz, the upper mid from 0.5 to 8 kHz, the
lower mid from 100Hz to 1.8kHz and the low
band reaches from 440Hz all the way down
to 25Hz. Again, each band’s frequency
control is accompanied by a ±15dB gain pot.
Finally, we come to the Helium, which
is a sort of halfway house, with the middle
two bands being borrowed from the Iridium,
and the outer two being R-C types. The
latter offer bell filters, identical to those
of the Carbon, but this time they can be
switched to become 12dB/octave high-/
low-pass filters (rather than shelves). Given
their generous frequency range, you can
use these filters for all sorts of useful things,
from rolling off low-end rumbles, through
‘bracketing’ a source to sit it in a mix, to
taming the side-effects of generous boosts
using those wide mid-range inductor bands.
Impressions
I tested all three EQs on a range of different
sources in active projects over the course
of several days. These included lead vocals,
acoustic guitars, kick, snare and a drum
bus, as well as some commercial mixes
with which I’m familiar. I should note that,
since I had only one of each module for
review, most of these tests were performed
on mono signals (although I did partner
the Helium with the Iridium on occasion,
to give a sense of the inductor EQ on
stereo sources, albeit limited to two bands).
There’s currently no stereo version of any
of these modules (a double-wide version
with a single set of controls and more panel
space could be interesting!) but Interphase
do offer a discount for Iridium pairs.
My first impressions of all of them were
great. They didn’t seem to add anything
noticeable in the way of noise, and the
control layout is about as intuitive as
www.soundonsound.com / January 2023 99
 ON TE ST
INTERPHASE AUDIO EQ MODULES
possible. By this I mean that although I was
interested to note what settings I’d settled
on, with just two knobs per band I was more
than happy adjusting things and making
decisions by ear and intuition alone.
If you’re looking for the sort of richness
and sheen that most digital EQs can’t
supply, the inductor bands on the Iridium
and the Helium will definitely appeal. While
the Helium offers that bit more control,
courtesy of its continuously variable R-C
bands and high/low filters, I soon found
myself reaching instinctively for the Iridium.
Why? Well, there’s such a lovely fullness
and smoothness to the high and low bands
in particular, and this not only made gentle
rebalancing of sounds a joy, but it almost
always seemed to get me where I wanted
to go rather more quickly. This difference
was perhaps best revealed to me when
shaping a slightly dull-sounding, fairly
slowly strummed acoustic guitar. I was able
to bring out all the ‘sparkle’ I wanted with
no hint of grit or brittleness, and to tame
some unhelpful boom quickly without any
unpleasant side-effects. It’s not that the
Carbon or Helium sounded at all bad on
the same source (they’re really nice EQs
in their own right, the R-C bands sounding
very clean, yet somehow also ‘present’ and
‘lively’) but I found I needed to work that
bit harder to achieve a similar sort of result,
particularly at the top end.
Something the Iridium lacks is high-/
low-pass filters, but in many ways it doesn’t
really need them. Interphase’s philosophy
As shipped, the module’s rear card connector is
compatible with 500-series racks, but Interphase Audio
can also provide a clip-on adaptor to take advantage of
the 51X format’s higher-voltage power rails.
EQ bands (it is based on resistor-capacitor networks)
but it features Allan Bradford’s Top Amps, and has very
generous frequency overlap between bands.
100 January 2023 / www.soundonsound.com
is to
design gear
that performs one task
exceptionally well and, these being bell
bands, it’s not like you need to tame the far
reaches of a generous shelving boost. Also,
there’s not a huge amount of space left to
accommodate more features. But with up
to 15dB of gain available, it does limit how
radical you can be at the extremes, unless
you pair it with another device that does
have filters.
It also has switched frequencies for all its
bands, which the Carbon and the Helium’s
top/bottom bands don’t. This has pros and
cons, but for the most part I much prefer
the switched settings. It makes recall and
A/B comparison far easier and, with broader
curves like these, I usually enjoy working
with limited options anyway; I spend less
time tweaking things that make very little
difference so just find the nicest position to
cut/boost, commit and move on. Naturally,
though, the greater precision of the
continuous pots is sometimes helpful, and
not least when trying to massage acoustic
drums into shape. On the one hand, I could
often get a result I preferred using the
Iridium — that silky richness when boosting
the low end of kick drums in particular is
a joy — but on the other, I sometimes found
it tricky to pull out a boxy resonance that
poked its head above the parapet when
boosting. For example, a generous 60Hz
boost sometimes meant a little 200Hz
needed pulling back, and the Iridium can be
too broad-brush a tool for that job; in those
circumstances, the Carbon and Helium were
preferable, because I could centre that cut
right where it needed to be.
It’s easy to see why the Helium is the
‘factory choice’ for the EQ on the Ark
console: recording or mixing on a console
laden with these on every channel would
be a delight. But the Ark can accommodate
two 500-series modules in place of that
design (which is twice the heigth and uses
Unlike the other two, the Carbon has no inductor
The Iridium inductor EQ is derived from
the four-band EQ created for Interphase
Audio’s modular Ark console.
faders for the
gain),), and I can see
the attraction of putting
an Iridium and n one of the other
du on a handful of channels:
modules
tthey’d make a very powerful pairing, just
as they could in your 500-series rack. To
ssome extent, off course, that’s t precisely
what the Helium aims to deliver: the best
of both worlds. It largely succeeds in that
too. On many sources, not least lead vocals,
the two inductor bands on the mids deliver
a silky touch of midrange magic, while the
outer bands can be used to sculpt the top
and bottom ends. The generous frequency
ranges of its high-/low-pass filters make it
really versatile if you enjoy doing all your
mixing in the analogue domain. Still, there
were definitely scenarios in which I preferred
the Iridium’s high and low bands. Of course,
bank manager permitting, the two modules
could make powerful partners, combining to
form a six-band inductor EQ with high- and
low-pass filters.
Verdict
So, there we have it. This trio of high-quality
discrete Class-A EQ modules delivers
exactly what their manufacturers claim:
three different takes on the ‘console EQ’
concept which work well on their own,
but might also be useful in combination.
For me, the Iridium steals the show, with
the beautiful, effortless elegance to its
sound. Occasionally, I wished it had either
shelving or cut filters. I had the luxury of
achieving that by routing its output to
the Carbon or Helium, but it’s also worth
me mentioning that the same inductor EQ
circuitry is also available in three different
19-inch rackmount products — which
happen to include high- and low-pass filters,
as well as a Swing (in other words ‘tilt’) EQ.
If you’re in the market for a 500-series
EQ, and particularly if the idea of
a 51X-compatible EQ appeals, then I’d
heartily recommend checking out these
modules. I look forward to seeing what
Interphase release next!
£ Carbon €690 each. Iridium €785 each
or €1446 per pair. Helium €835. Prices
include VAT.
T Interphase Audio +44 (0)1626 650 017
E gear@interphase.audio
W www.interphase.audio
Only until 11 January

Save 20% on Live 11,

Push and Packs

Scan to go to the Ableton shop

 INTER VIE W

102 January 2023 / www.soundonsound.com

 Making This Is What I Mean
Britain’s most high-profile rapper has made his most
ambitious album yet — with a little help from his friends.
TOM DOYLE
“S
tudios are a funny one for me,”
grins Stormzy, as he considers
what makes him decide to
choose a particular recording space. More
than equipment, or even location, the key
factor is usually whether or not the rapper
feels instinctively comfortable in the room.
“It’s just a spirit, man,” he adds. “Like, even
walking in here, for whatever reason, I feel
like I could record here.”
Here being The Bridge Studio in London
SE1, where Stormzy and his team are
putting the finishing touches to a couple
of mixes for his third album, This Is What
I Mean. It’s the first time they’ve used
this particular studio, the latest in a line
of London facilities they’ve hired that
has included Metropolis, West Point and
Studio 13, the studio owned by Damon
Albarn which is not typically available for
commercial use. “People don’t really use
Studio 13,” Stormzy says. “I used to record
next door in The Bunker [writing room
owned by producer Stephen Street]. I just
had a couple of good experiences there.
So yeah, we ended up in Studio 13 and just
stayed there.”
However, at the heart of the sessions for
This Is What I Mean was a month spent in
a remote and unusual location: Osea Island,
situated in the River Blackwater in Essex.
There, in April 2021, Stormzy holed up with
executive producers Kassa Alexander aka
PRGRSHN and Tendai Senyange, engineers
Joel Peters and Calum Landau and featured
collaborators including Jacob Collier and
Debbie Ehirim, cutting themselves off
from the world in an intensive writing and
production camp.
“I always wanted to do a camp,” the
rapper says. “Wake up every day, and
The production team behind This Is What I Mean
at London’s The Bridge Studio. At front, engineers
Calum Landau (left) and Joel Peters; standing,
executive producer Tendai Senyange (front) and
Photos: Zebie
Stormzy; seated, executive producer PRGRSHN (at
piano) and #Merky A&R Manager Jermaine Agyako.
we’re on music.” In this way, the making of
the new album contrasted with that of its
predecessor, 2019’s Heavy Is The Head,
which was written with a legion of different
producers in multiple studios. “Especially off
the back of Heavy Is The Head,” Stormzy
stresses, “where I was all over the gaff,
I always thought, ‘If I make a concentrated,
conscious effort in one place, with all the
best musicians and producers and artists, I’ll
thrive there.’”
This Is What I Mean retains the original
grime spirit that propelled the 29-year-old
South London rapper from the streets of
Croydon to the cover of Time magazine and
his headline slot at Glastonbury’s Pyramid
Stage in 2019. At the same time, it finds him
venturing into new sonic territory, where
distinct echoes of classic ’70s soul can be
heard alongside the thumping beats. As he
raps on the album’s title track, “I think I’m
Kanye mixed with Donny Hathaway.”
But Stormzy’s plan wasn’t always to make
a soul album. “Nah, I didn’t have that in mind
at all,” he says. “I wanted to make kind of
like a hard rap album, if I’m being honest.
We went to LA just before the pandemic and
then we had to come back. Lo and behold,
a couple of years later, we’ve got this album.
Nothing about it was planned.”
Storm Building
Stormzy — real name Michael Ebenezer
Kwadjo Omari Owuo Jr — was only 12 years
old when he first recorded one of his raps,
using a desktop PC and Acoustica’s Mixcraft
software. “It would have been a £1 mic from
Poundland,” he laughs. “Literally, we used to
get one pound mics. Then just record, press
bounce, and you was good to go.”
Through his appearances with
Walthamstow collective the HeavyTrackerz,
his own self-released mixtapes and
collaborations with Chip and Lethal Bizzle,
Stormzy soon gained experience of
professional recording setups. He says he
didn’t feel intimidated whatsoever when
he first set foot into a studio environment.
“No, cause it wouldn’t have been like a solo
recording process. It would have been loads
of us. And it would have been a beat and
then everyone goes in and does their bit, and
then we might do a couple of ad libs. I think
maybe when I went by myself the first time,
that might have been more intimidating than
everyone just in the studio smoking weed,
having fun.”
Following independent hits ‘Know Me
From’ and ‘Shut Up’ in 2015, he began
licensing his own #Merky Records releases
through Warner Music, including his debut
solo album, 2017’s Gang Signs & Prayer,
made at Matrix Studios in Fulham with
producer Fraser T Smith. “That worked really
well,” Stormzy says, “’cause firstly Fraser is
incredible at what he does. Secondly, we
recorded it in one studio. It was like a hub.
I would go there, we’d have a whiteboard
there. Especially for my first album, it was
just good to know that I was going to one
place to create my album. And I’ve got this
phenomenal producer, this phenomenal
engineer, Manon [Grandjean]. That was
the great thing about working with Fraser.
If I wanted to make some soul, if I wanted
to make some grime, he would be able to
facilitate everything.
“Then the second album was just the total
opposite. That was kind of going out into the
big wide world and, like, different producers,
different studios, different files, different
engineers. When I look back on it, I think
that’s what needed to happen at that time.
Because I was doing Glastonbury and all that,
which was extremely overwhelming.”
But Stormzy decided to change his
operation again for This Is What I Mean, and
assemble a single, dedicated team. “One
thing about me is my spirit always tells me
what I need to do,” he says. “So, I knew that
I needed to just get my bearings straight.”
Opening Fire
“Two years ago,” says executive producer
Kassa Alexander, “Stormzy sent me a song
that he’d started writing with a pianist called
Dion [Wardle]. And he was just asking what
I thought. Then we kind of talked about
www.soundonsound.com / January 2023 103
 INTER VIE W
STORMZY
different things we could do to progress
the song.”
Alexander sent Stormzy what he calls
a “soundboard playlist” of songs he felt
could be used as reference points for the
album, including tracks by Kanye West,
Frank Ocean, Drake and Kendrick Lamar.
“It’s a bunch of songs, I think, that could
have been Stormzy songs,” Alexander
says. “He texted me back and he was like,
‘Aw, it’s like you’re in my head. You want to
executive produce my album?’”
Work began in late 2020 at Studio 13
where, encouragingly, the foundations of
the album’s opening track, ‘Fire + Water’,
were very quickly laid down. “It just set the
tone,” says Alexander. “In the sense that,
like, this album is going to be based on
truth, it’s going to be vulnerable, it’s going to
be honest. And it’s going to be soulful. That
was, I guess, the flag in the sand.”
Alexander had recently switched over
from Logic to Ableton Live. “I went to LA,
and I did this camp,” he explains, “and
everybody was making music faster than
me. I was like, ‘How?’ I’ve been using
Logic since Logic 4, so I was fast on it.
But everybody was able to make these
records just quick, quick, quick. I spoke
to a producer, and he was like, ‘You need
to use Ableton. It’s a lot more intuitive
and it always takes tempo and pitch into
consideration.’ Obviously, because people
use it for live shows and for DJ’ing, it’s really
good for that.”
Co-produced by PRGRSHN and KZ,
‘Fire + Water’ is an eight-minute-long epic
on the finished album. But it didn’t start life
that way. “That track never felt difficult to
make,” Stormzy stresses, “because it was
never like, ‘We are making this fucking
104 January 2023 / www.soundonsound.com
stonker of an intro.’ Me and PRGRSHN made
the first bit, just very naturally. He started
playing the chords. I just grabbed the [Shure]
SM7. The vocal you hear on the hook is the
original vocal I sang on that day, sat down on
the sofa.”
Engineer Joel Peters, who was trained at
SARM Studios by Trevor Horn, was brought
into the project by Alexander. “There was
a period when we were all kind of learning
what the record might be,” says Joel.
“There were lots of conversations. A few
tracks had kind of started to take form. But
it was really only that period at Osea Island
when we really got a sense of what the
album was going to be and the songs kind
of materialised. We did a few sessions in
Metropolis, a few sessions in West Point, late
2020. But then Osea was just the explosion
of having all the minds under one roof.”
Multi-threading
When Stormzy and the rest of the team
arrived at the manor house, they soon
built various programming and recording
set-ups in different rooms, away from the
control room, so that work on tracks could
progress continually. “Tendai would be top
floor,” Stormzy recalls. “There was a little
weird living room in the middle that Jacob
[Collier] or Owen [Cutts] sometimes would
be in. Then my room, P2J [Beyoncé, Wizkid]
has got a setup there, and then you’ve got
the main studio downstairs. There was also
a living room that anyone could set up in.”
On day one at Osea Island, inspiration
immediately struck. As filmed and later
posted by Stormzy to Instagram, the track
that is now ‘Hide & Seek’ was almost
automatically written. “That was the first
10 minutes, the first day we got there,” he
The majority of work on
the album was done during
a month-long residency at Osea
Island. A number of different
workstations were set up in
various rooms throughout the old
manor house.
remembers. “Kassa started
playing these chords.
Y’know how a session
goes... he plays some other
chords and then we’re on
a different idea and the day’s
gone, another day’s gone,
three days have gone.
“So, I was sat in the
studio with Joel, and I said,
‘What did we do on the first
day?’ Joel’s gone to the
very first thing. I was like, ‘I remember there
was some chord that resonated with me,’
and then he found it, and I started singing
to it. That was ‘Hide & Seek’. Then P2J, he’s
heard it and he’s like, ‘Yo, let me fuck with
this.’ Took it upstairs and he’s put the drums
over it. But then it went on such a fucking
insane journey.”
“P2J uses a combination of FL Studio and
Logic,” says Peters. “Everything’s coming
from in the box, really.” As with a lot of grime
and R&B, there’s an emphasis on rimshots
over snares in the rhythm tracks. “I guess it
must have come from a time of people using
snares, and then thinking, ‘What’s louder than
the snare?’” Peters reckons. “‘Hit the rim,
that’s got more impact.’ So, I guess it’s just
like the arms race of transient attack.”
Low & Heavy
Beats for the album tended to come from all
directions. “Kassa does a lot of programming,
along with Tendai on this record,” says
Peters. “Scribz Riley as well, and P2J.
Sometimes someone will send us something.
Sometimes we’ll work on it in the room with
them, and kind of sculpt as we go.”
Kassa Alexander’s go-to drum
programming resource is Splice, “along with
loads of samples out of banks that I’ve built
over the years,” he says. “There was a lot of
live percussion as well. So, for instance, in
the second section of ‘Fire + Water’ when the
drums come in, I played a lot of actual live
percussion as well, just to give it that human
feel too.”
Bass on Stormzy tracks tends to be
heavy and minimal. “A lot of the bass
is programmed,” says Peters. “Stormzy
isn’t a big fan of complicated bass parts.
Especially when the drums are hitting quite
 Island Load-in
Engineer Joel Peters was initially a touch nervous
about the logistics of the album sessions that were
planned for Spring 2021 in a 10-bedroom manor
house situated on Osea Island. “Going into it, I was
a bit skeptical,” he admits. “I’m going to turn up in
a new place and I don’t know what channels are
crackly, how’s the monitoring. All they had was
a pair of NS10s.”
The control room at Osea Island is based around
an SSL 4064 E Series console, but Peters felt that
bigger monitors were required for the Stormzy
project. Getting in touch with his friends at Quested,
he hired a pair of their Q212s, which sit neatly on
top of matching QSB118 sub units. Having the gear
delivered to the island was a challenge, however.
“To get to Osea Island,” Peters explains, “there’s
a causeway you have to drive across, and you
can only drive across it at low tide. You can get
a boat at high tide, but that’s not really going to
work when you’ve got huge flightcases! So, we
hard and they’re fairly busy. We’ve tried
on this album to keep the bass really quite
simple and not distracting. So, the vocal can
kind of shine and take priority.
“I think there’s a real trend at the moment
for simple kick patterns. Some of the kick
patterns on the record might just hit on the
one and then maybe there’s not a kick for
two bars. And if the bass is doing too much,
it almost feels a bit weird. So, save that
impact for just that hit.
“There’s a lot of tuned 808s from
Kontakt. And some kind of sine Moogs. Just
really smooth and full. And then on ‘This Is
What I Mean’, there’s quite a loud, distorted
bass. But apart from that, we kept the bass
quite simple.”
Miking the Feurich piano at East Point,
Peters used a pair of Neumann KM 184s.
At the suggestion of Owen Cutts, though,
he also experimented with putting a Shure
SM57, in a shoe, close to the soundboard.
“Owen, who produced ‘Please’, had this
mad idea about putting a mic in a shoe, for
some reason. It sounded extremely dull, but
that gave us a bit of clout in the bottom end.
I think if you got a nice setup, try one weird
thing, and just see if you get anything.”
Vocal Recording
When it comes to capturing Stormzy’s
vocals, the engineering team’s task is often
like bottling lightning. On this new album in
particular, the rapper wanted to avoid the
vocal booth, preferring to perform in the
control room, using one of two different
mic setups: a Shure SM7B handheld, or
a Sony C-800G hanging over the console,
which were individually routed through the
arrived a couple of days before everyone and Ollie
[Shortland] from Quested was a little bit ahead of
schedule. When we go over the brow of the hill, we
can just see this Quested van halfway across the
causeway, and he’s going for it... the tide isn’t fully
out, the causeway isn’t fully exposed. We were like,
‘Is it gonna be all right?’
“We had a van behind us with a piano in.
Luckily, we all made it across the causeway. But
it’s really a nice entrance and a unique experience.
So, we set up these Questeds in the house. I like
speakers that kind of give the artist confidence
when they see them. I think that’s an important
part of a big set of speakers. Because we don’t
listen that loud that often, but we want to play
back at the end of the day, to vibe out, to get
excited about what we’re doing. The piano was
a Feurich 122 that we’d heard at West Point. It was
just a really good-sounding upright. It was kind of
tight, poppy.”
two-channel Neve 1073DPX preamp/EQ unit
and into a Tube-Tech CL1B compressor.
“When we’re kind of in writing mode,”
says Peters, “I’ll probably just have the
SM7 recording the whole time, even if
nothing’s going on. Just in case someone
says something in the room or has an
idea. Jordan [Boza], the videographer, is
always recording. So sometimes when we’d
stopped and there was no track playing,
someone might do something, and then it’d
be like, ‘What was that?’, and then we could
get it off the camera.
“But we like laying stuff down initially
on the SM7. Stormzy’s a big fan. He uses
the warmth of it. We’re all on headphones
and the Pro Tools rig’s usually on the
trolley, so I can get fairly close to Stormzy
in a place where he can see the screen
as well. He might point at some region,
or whatever. If he’s in the booth, it’s a bit
harder to communicate, so we both like to
be by the screen.”
Their desired recording method is: the
fewer takes, the better. “For a rap verse,
it’s very few takes. He will write it in his
head. And we’ll have the track on loop for
a bit and when we come to lay it down,
there will be few comps in it as well. He’s
pretty amazing like that. But yeah, it’s just
a case of having everything there and just
being ready. Because you don’t know what
magic’s gonna appear really.”
Out Of The Booth
Stormzy is clearly happier with this new,
freewheeling way of recording vocals than
with the more traditional methods he’s tried
in the past. “I used to be a booth guy once
Quested Q212 monitors with their matching
subwoofers: not something to transport in
a small boat!
upon a time, but yeah, I’m not any more,”
he laughs. “What happened to me was,
basically, over the period of Heavy Is The
Head, I didn’t have the best experiences
recording in the booth, and then coming out
and hearing what I’ve done and having to
comp it.
“I really dislike getting into the booth,
because I’m so particular with my vocal
and how I want things to sound. When
we started working on this album, I got
my spark for recording back. If I go in
[the control room], and I do 64 bars, and
I listen back, and I don’t like the 12th bar,
Joel can just cut me in and then I get it. It’s
a seamless vocal.”
“We’re favouring not using the booth
for vocals,” agrees Peters. “Just from the
point of ease, where he can stand up and
deliver the vocal. He doesn’t have red light
phobia in any way, but some of the vocals
on the record are really quite intimate,
and we were quite conscious of him not
delivering them in the booth with us staring
at him. For some of the vocals, we cleared
everyone out apart from me and him.
Like ‘Please’, for example. It was such an
intimate, emotional vocal that it was just me
and him in the room.”
On the new album, Stormzy sings just
as much as he raps, revealing a melodic
confidence that he admits he’s had to
strive to achieve. “I’ve been rapping for
20-odd years now,” he points out. “I’m very
comfortable doing that. Whereas singing is
something that I’ve had to work on for the
past seven years. I wasn’t very good at it.
I was bad at it, and I’ve gone on this journey
of training myself and learning. So, if I go in
www.soundonsound.com / January 2023 105
 INTER VIE W
STORMZY
Stormzy’s vocals were tracked in the control room, either on a Sony C-800G or, as here, on a handheld
Shure SM7B. A&R Jermaine Agyako and engineer Joel Peters are also visible in this photo from East Point.
with Joel and Calum and we’re going to nail
a vocal like ‘Firebabe’, line by line we are
trying to deliver the best vocal. Because I’m
still gradually improving. I’m improving with
rap, but I’m fucking incredible at that [laughs].
With singing, I’m still really working at it.”
As one of the executive producers, Kassa
Alexander was instrumental in encouraging
Stormzy to sing more on his third album.
“One hundred percent,” Alexander nods.
“I’ve always known that he has a love for
melody and lyric, and he loves singing.
But I guess sometimes, as a rapper, it’s
a confidence thing. Because it takes time.
This Is What I Mean is Stormzy’s third
full-length album.
106 January 2023 / www.soundonsound.com
I was able to work with him and really push
to get the best vocals out. And he’s singing
with conviction. I think that that’s proper
important in his growth and the evolution
of him as an artist, being able to express
himself in whichever way he feels he needs
to, to get the message across.”
When it comes to the mix, Stormzy’s
vocals are usually left almost entirely dry,
especially the raps. “I do like a dry vocal!”
says Joel Peters. “There might be a tiny bit
of room, probably no plate. Maybe a really
short delay. We use a lot of Valhalla and
Altiverb. The standard Pro Tools delays are
all good. Sometimes we’ll have a tape delay
for a little warmth or making it a little bit
more saturated. The CL1B is doing the most
sculpting in the vocal chain, compressing
on the way in. We try and keep the vocal as
natural-sounding as possible.”
Fire To Water
The atmosphere at the Osea Island studio
was a creatively vibrant one, and Stormzy
and the team found that they made major
songwriting and production breakthroughs
in their month together there. One day, the
others overheard Debbie Ehirim working
on a song named ‘Pour Me Water’, which
Stormzy instantly felt could work for the
ending of the opening track, ‘Fire + Water’.
“We were all in the next room,” he says,
“and we could hear it and we came in and
it was like, ‘This is fucking cold.’ There was
an idea to put them together. I went on tour,
and I was listening to the album a lot. I was
listening to the intro, and I would listen to
‘Pour Me Water’, and I was like, ‘This can
work.’ And we were trying to merge them
together. But obviously ‘Pour Me Water’ is
quite a lot slower.”
In the end, Joel Peters had to work on
a gradual tempo slowdown of the ‘Fire’ part
of the song to match it to the closing ‘Water’
section. “I can’t remember how many bpm
it is,” he says. “Maybe 15 bpm between
the two. We were kind of taking inspiration
from those Timberland/Timberlake records
where, quite often, you have a tight radio
edit of the song, and then when you listen to
the album version, they’ve found some nice
segue into some mad jam at the end.”
Engineer Calum Landau thinks that the
ending of ‘Fire + Water’ captures on record
the spirit of the Osea Island sessions. “It was
late night in the living room, everyone was
chilling. It sounds like a party. You can feel
it. You can hear Debbie’s vocal in the guitar
[channel]. It’s part of an open room with
people playing.”
Jermaine Agyako, A&R manager at
#Merky Records, credits Stormzy himself for
maintaining a breezy and artful atmosphere
at Osea Island. “There were a lot of
incredible musicians there. But I think what
Stormzy was very, very good at was making
 Collier Calling
A whirlwind presence during the sessions for
This Is What I Mean was Jacob Collier, whose
stacked harmonies, recorded alone in a room
at Osea Island using Logic and a laptop, are
a recurring sonic feature throughout the record.
“When we were working on something
else,” remembers Joel Peters, “we could hear
him upstairs singing all these different notes
that out of context sounded bizarre. And then
he brought his laptop downstairs, plugged in
and played us this huge vocal arrangement of
‘Firebabe’. We were like, ‘Woah, that’s amazing.’
The chord choices and the voicings are just so
tasteful. But yeah, he typically layers it all up
himself, because he knows exactly what he
wants, and it’d be a lot slower if I was pushing
the buttons for him. He goes away and 10, 15
minutes later, you’ve got a masterpiece.”
“There were so many things I learned from
Jacob,” says Kassa Alexander. “And there were
sure that the ego in the room was six foot
deep. The culture wasn’t, ‘Yo guys, we’ve
come here to make a hit album for Stormzy.’
So, everybody felt free in their own way to
throw their paint at the canvas.”
Lucky Number
After Osea Island, the sessions returned
to Studio 13 in West London, with its Neve
VR60 console and huge array of hardware
keyboards and synths. “One thing about
Studio 13,” says Tendai Senyange, “is
because there’s so many instruments,
there’s so much — in my head — freedom.”
“They have a really good-sized control
room,” Peters says. “The old Neve V in
there. And it’s like, hidden away, so we can
be in there and work and no-one knows
we’re there. We were kind of left to our own
devices.”
“Exactly,” Alexander says. “So, in some
ways, it has a similar vibe to Osea in the
sense that when you get inside there, you’re
almost shut off from the world in a similar
way. You’re in this cocoon. Thirteen was
equally as prominent to the process as
Osea was. It was the post process.”
One memorable highlight for all
involved came when Sampha arrived and
improvised his contribution to an alternate
version of ‘Please’, titled ‘Sampha’s Plea’.
“He’s a hard man to get a hold of,” says
Peters. “We had the session in the calendar
for a while. And I thought, ‘Yeah, he might
not come.’ He came, but I thought, ‘We’re
not going to get him in the booth. We’ve
only got him for a few hours. We’ve just got
to get a mic in front of him and whatever he
does, that will be it.’ We put up a [Neumann]
so many things he learned from us in the way
that we work. So, for instance, I was giving him
808 samples, cause he’d never used them.
I just watched him stack harmony, harmony,
harmony, harmony. Then you have all of these
other little countermelodies. To see how
quickly his brain is making all this happen is
mind-blowing.”
Collier also played the Feurich piano on
‘Please’. His performance was recorded in
one take only 10 minutes before he left Osea
Island, and required much in the way of audio
clean-up later since he was wearing a massive
coat that made unwanted noises every time
he moved. “The performance was amazing,”
Peters recalls. “But as soon as he played the
last chord, he was out in the car. He just didn’t
have time for another take. And we spent days
with [iZotope] RX trying to get the coat out of
this recording.”
U47 really quickly in front of him, and just
hit Record.”
“What’s so funny is with that demo,
I was so ready to say, ‘That’s fine,’” Stormzy
says. “There was no lyric in it, he was just
humming it out. And I was like, ‘Yo, if you
don’t come back, I’ll use this.’ But then he
went and put a lyric to it and came back.”
Nothing Radical
For the mixing stage, different tracks were
farmed out to different mixers, including
Leandro Hidalgo (see elsewhere in this
issue) and Mark ‘Spike’ Stent. “We’re always
trying to make the roughs as good as they
can be,” says Joel Peters. “We don’t ever
want to send anything out that’s too quiet
or doesn’t sound good. We’re always really
focused on making a set of roughs that
sound amazing. Because you’re always kind
of mixing in a way now.
“Our rough mixes are so detailed, we’d
kind of laid out the path for the next people
to work on. There wasn’t anything where
when we sent it to mix, really, we were
going to get something radically different
back. Sometimes, someone’s got a better
perspective than us, and they’re like, ‘That
does need to come up. Despite what you lot
feel.’ But it’s a fine line, man.”
All agree that the toughest mix part
to nail was the dynamic moment in ‘Fire
+ Water’ when the drums kick in. “I spent
days and days and days in the early part of
the project trying to get those drums hitting
right,” Peters says. “Because I knew they
wanted to come in with a real bang and
knock you back. There was a period where
we’d do a session and Stormzy would leave
Jacob Collier
made a big impact
in a short time!
at, like, 4 or 5 pm, and I’d just spend four or
five hours on those drums. Just really trying
to wring the most out of them.”
“For a long time, for about four months,
I hated the drums,” Stormzy says. “They
never moved me. ‘Fire + Water’ goes on such
a journey, and I love the journey, and then the
drums dropped, and I said, ‘Nah, you’ve lost
me.’ It’s probably been a two-year journey
to where it was literally last week where we
finally got the drums, and I was like, ‘Yeah,
we’re slapping that.’”
Feeling Good
Much like his instinctual feelings about
recording studios, Stormzy tends to focus
on the emotional effect that a finished mix
has upon him, along with a simple desire for
sonic clarity. “Sometimes my initial emotional
reaction is, ‘This don’t feel right,’” he says, as
the others chuckle knowingly around him.
“But I just know there’s a feeling I’m meant
to feel. And there’s meant to be a warmth.
There’s synergy I want between the volume
of the vocal, and the volume of the music,
and the richness. If I’m rapping, I need it to be
present and clear and concise.
“A mix is meant to be a perfect marriage,”
he concludes. “I’m hearing my voice and I’m
getting all the music and it’s nice in my ear.”
Mission accomplished, then. This Is What
I Mean is punchy, and tough-sounding in
parts. But, due to its soul and R&B crossover
appeal, it’s also very likely to end up being
Stormzy’s biggest album yet. Ultimately,
though, the process of making his third
record has taught him one important lesson.
“I don’t think I’ll ever go in the booth
again!” he laughs.  
www.soundonsound.com / January 2023 107
 ON TE ST

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SAM INGLIS

aving invented

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the release version incorporates not only not available. This ‘early 2000s’ vibe is this arc belongs. The Threshold slider in
George’s innovations but the insights of carried through into the user interface, DRC2-Native actually sets the position of
many studio legends. which has none of the slick graphical this rotation point — meaning, unlike the
One word that MDW don’t use in features you’ll find in, say, FabFilter’s threshold control on most compressors, it
association with DRC2-Native, though, Pro-C 2. It’s not much to look at, but in interacts with the knee setting.
is “compressor”. They are careful to practice, it is fast to use and gives you The most important innovation in
call it a “dynamic range controller”, and all the visual feedback you need. I could the design of DRC2-Native, though, is
it’s perhaps best thought of as a fresh wish that it was resizable, though, and something that Massenburg Design
attempt to tackle the same issues for apparently it soon will be. Works are referring to as Variable
which compressors were originally One of the novel concepts employed Exponent Averager. A simplistic
invented. Some of its controls employ in the design of DRC2-Native is that of description of a compressor is a device
similar terminology to those found on the ‘rotation point’. Plotted on a graph that can detect when audio is above
a conventional compressor, but their that shows input level on the X axis and a certain level, and turn it down. The

108 January 2023 / www.soundonsound.com

turning down part is straightforward,
but the detecting aspect much less so.
Should our hypothetical compressor
simply respond to the instantaneous level
of the signal? Or should it respond to an
average level integrated over a period of
time? And if so, how should that average
be calculated?
VEA is an attempt to increase
the sophistication of this calculation,
designed to make the level detection in
DRC2-Native more akin to the complex
response of the human ear. The idea,
in other words, is that gain reduction in
DRC2-Native is triggered not by the level
rising above some arbitrary point, but by
events that we would perceive as being
subjectively louder.
This is achieved first by optionally
filtering the side-chain using the BS1770
filter, which anticipates the dynamic and
timbral sensitivity of the human ear, and
also by combining two separate detector
timing algorithms. The Main detector
uses a sophisticated RMS algorirthm to
identify loud events on a relatively broad
scale, but there’s also a second detector
labelled Peak. This uses a different RMS
calculation and would typically be set
up using faster time constants, the idea
being that it has the power to override
the Main detector when a sharp transient
event is detected. It’s important to note
that these are not two separate dynamic
control devices placed in series, like for
example the compressor and limiter on
a Neve 33609. They’re separate methods
Massenburg
Design Works
MDWDRC2-Native
$399
PROS
• Can apply startling amounts of gain
reduction on most sources without
any artefacts being apparent.
• Extremely versatile.
• Although the range of the controls is
unusually wide, so is the sweet spot.
CONS
• Basic, non-resizable GUI.
• There is a learning curve, though
I hope the imminent availability of
a manual will help negotiate this.
• Depending on what you compare it
to, it could be considered expensive.
SUMMARY
DRC2-Native is a genuinely different
dynamics plug-in that has a unique
ability to iron out level variations
without sounding like it’s doing
anything at all.
of analysing the side-chain signal which
are combined to trigger gain reduction in
one device.
Most of DRC2-Native’s controls
are at least superficially familiar from
conventional compressors, but many
of them operate rather differently here.
As previously mentioned, the main
Threshold and Knee controls interact, and
so do the time constants. Release time
is calibrated in dB per second, as on the
old dbx 160, and Attack simply runs from
1 to 8. ‘Real’ attack and release times are
derived from their relative settings and
reported. The range of most controls is
also unusually large. It’s possible to set
a ‘real attack’ time ranging from over 10
seconds down to 0.3ms, while the Ratio
control runs from 1.2:1 right up to 100:1
and the Knee range is equally expansive.
The optional Auto-release setting is also
highly configurable.
Sliding Scales
At the time of writing, DRC2-Native
includes no presets and has no manual
(though one should be available by
the time you read this). As a result,
I did my share of floundering around
blindly before getting a sense for how
everything works. One feature that takes
a bit of getting used to is the provision
of both a Threshold slider and a Main
Compressor Gain level slider. At first
I wondered why both were necessary,
but I think the answer has to do with the
fact that the Threshold adjusts ‘rotation
point’ and not threshold as it’s usually
understood. Because this interacts so
much with the Knee control, it’s useful
to have a separate gain slider, but it
would be nice to have the option to
link the two; as it is, changes to either
affect the overall output level and mean
you’re constantly having to adjust the
Output fader.
The threshold for the Peak section is
set relative to that of the Main section,
and can be anywhere between 0 and
18 dB above it. On a signal with lots of
transient content, setting this slider to
zero and the Peak Knee setting to very
soft means the Peak detector overrides
the Main detector much of the time;
turn up the slider and/or reduce the
Knee setting and it will take over only
on the most violent peaks. All of this is
indicated simply and effectively using
colour-coding: gain reduction triggered
by the Main detector shows up blue,
while the Peak detector turns it orange.
The stated design goal behind
DRC2-Native is what might be called
‘invisible’ compression, and Massenburg’s
blurb makes great play of its ability to
apply lots of gain reduction without
obvious artefacts. It’s worth noting,
though, that you don’t have to use it
like that. It’s quite possible to obliterate
sources if you push it hard enough, and
I had fun recreating that early-2000s
drum sound whereby you mix the kick
and snare far too loud and then use
a bus compressor to push them back
down again. I should also point out
that although the controls are both
wide-ranging and unfamiliar, they are
also quite forgiving. Putting the settings
somewhere in the middle, as a general
principle, sounds good on most sources.
And yes, if it’s invisible — or rather
inaudible — compression that you’re
after, DRC2-Native is almost eerily good
at it, and nowhere more so than on sung
and spoken vocals. You can apply 15dB
or more of gain reduction without any
pumping or other side-effects. In fact, in
many cases, it’s the presence of room
tone or valve mic self-noise that limits
how much gain reduction can be applied,
rather than artefacts directly related
to compression. There’s no sense that
DRC2-Native is flattening out the sound
or obviously changing its tone. It just gets,
well, more even. I don’t think I would
even describe it as ‘smooth’, because
that implies imposing a character, which
isn’t happening here. To deploy a slightly
laboured analogy, some compressors act
a bit like zoom lenses: they enable you
to inspect things more closely, but you
always know that you’re looking through
one. Using DRC2-Native is more like
moving nearer the thing you’re looking at.
It will, I think, be a particularly
attractive product for anyone working
with dialogue and voiceover, but it’s
hardly limited to those applications.
Mastering engineers will find many uses
for this plug-in, and once you’ve got to
grips with its unusual controls, it makes
a deluxe option for general-purpose
compression at the mix, especially as
it seems to have a pretty light CPU
footprint. Yes, it’s expensive, and can
feel counter-intuitive if you’re used to
conventional compressors, and no, it’s not
pretty to look at — but if you demo it, you
might find it hard to live without it.
£ $399
W www.massenburgdesignworks.com
www.soundonsound.com / January 2023 109
 ON TE ST
IK Multimedia
iLoud Precision MTM
IK’s affordable iLoud range has grown into a serious
monitoring tool. We put the new flagship to the test.
PHIL WARD
ith their iLoud range, IK
W Multimedia have carved out
a niche for ultra-compact yet
high-performance active monitors created
using injection-moulded plastic enclosures
and, in the case of the iLoud MTM, with
integrated ARC room optimisation. The
110 January 2023 / www.soundonsound.com
Active Monitors
iLoud Precision MTM subject of this review
marks a significant departure, because
while the integrated room optimisation
remains, ultra-compact dimensions
and plastic enclosures have given way
to more traditional proportions and
wood-panel construction.
The iLoud Precision MTM is the
largest of a new range that also includes
the smaller iLoud Precision 5 and
iLoud Precision 6. Each monitor ships
with an IK Multimedia MEMS (Micro
Electro Mechanical System) ARC 3
measuring microphone and a set of four
monitor isolation pads. Simultaneously,
IK Multimedia have also launched
a standalone X-Monitor Windows and
Mac OS app. X-Monitor provides monitor
configuration functions (so no more
reaching around the back to fumble with
EQ switches), manages firmware updates,
provides a range of default monitor
EQ profiles, and also plays the role of
front end for ARC 3 room optimisation.
So, together with the monitor itself, I’m not
short of things to write about. I’ll start with
the monitor.
Precision Engineering
As its name suggests, the Precision
MTM is a D’Appolito-format monitor
that incorporates twin 130mm bass/mid
drivers located above and below a 38mm
shallow-waveguide-loaded textile-dome
tweeter. The system is reflex loaded by
a rear-mounted and generously flared
port. The Precision MTM cabinet is of MDF
construction and not entirely rectilinear,
flaunting some subtle curves. The cabinet
is finished in a conservative ‘studio black’
textured paint. It feels substantial, and
its relatively large and slightly curved
side panels respond to a knuckle rap
with a satisfying dull and non-resonant
thunk. Precision MTM amplification
comprises 145 Watts and 30 Watts
of Class-D power respectively
for the bass/mid drivers and the
tweeter. The big difference in rated
power specification between the
drivers serves to illustrate how
much more sensitive the latter is
than the former.
The bass/mid drivers
feature a coated diaphragm
that IK Multimedia describe as
 Screen 1: The X-Monitor app
guides you through the process of
measuring the monitors’ response
in your room.
“ultralight”, and the tweeter
features a chamber behind
the dome that serves
both to lower the driver’s
fundamental resonance
and to attenuate energy
radiated from the rear of the
diaphragm. Beyond that, IK
Multimedia offers no particular
details on the drivers that
might suggest they’re
anything but conventional.
The crossover between
the drivers is at 1900Hz
and has extremely steep,
eighth-order (48dB/octave)
filter slopes. These steep
slopes are made feasible
by the Precision MTM being very much
a monitor of the DSP age. Its DSP runs
internally at 96kHz, and along with the
crossover functions, it implements the
ARC 3 features, the rear-panel EQ options
and the USB-enabled integration with
the X-Monitor app. The DSP also enables
wide-band time-domain correction that
helps ensure that, on its reference forward
axis, the Precision MTM’s phase response
stays within ±20 degrees from 150Hz
upwards. Control of phase response
should help ensure that, on-axis at least,
the Precision MTM will produce notably
tight time-domain performance. This ought
particularly to benefit stereo imaging, as
our ears rely significantly on timing clues
in order to identify audio source position.
Considering the all-digital internals of
the Precision MTM, it’s slightly surprising
that it offers only an analogue input, on
a balanced combi XLR/TRS jack socket.
There’s no digital input of any type. The
rear-panel EQ switches, duplicated within
the X-Monitor app, offer low-frequency
extension and shelf adjustments, desk
EQ options, midrange boost and cut,
and high-frequency shelf adjustment.
X-Monitor additionally provides a range
of preset EQ profiles, using the five filter
elements and designed for different
installation scenarios: ‘Small Desk Wall’,
or ‘Large Desk Corner’, for example.
Alongside the rear-panel EQ switches
are an input sensitivity adjustment
control, an input socket for the ARC 3
measuring microphone, and a CAL button
that enables the integrated ARC 3 room
optimisation to be managed from the
monitor rather than from the X-Monitor
app. There’s also 3.5mm jack socket
present to connect an optional wired
remote control, which offers a non-app
means of switching monitor EQ profiles.
ARC De Triomphe
Speaking of the X-Monitor integrated ARC
3, if my memory serves me correctly, ARC
was one of the first room-optimisation
products, and is now on its third major
iteration. If you’re unfamiliar with the basic
principle, it is to measure the frequency
response that the combination of monitor
and room acoustics generate, at and
around the primary listening position, and
then to use the power of DSP (sometimes
within the DAW computer, sometimes
within the monitor) to equalise the
monitor’s inherent frequency response
such that the room effects are corrected.
Early versions of room optimisation
routines were, I think it’s safe to say,
relatively unsophisticated, but the field
has developed rapidly, ARC 3 being an
example of that development, and the
technique has matured to become both
effective and useful.
Critics of room optimisation, of which
I have been one, sometimes point out
a fundamental flaw, which is that it
sacrifices the tonal accuracy of the direct
sound of the monitors (ie. the sound
that reaches the ears directly and hasn’t
yet bounced around the room) in order
to correct flaws that might be more
appropriately fixed by room acoustic
treatment. This, especially in the case
of nearfield monitoring, can result in
a skewed direct sound tonal balance
and, potentially, compromised stereo
imaging if the two monitors of a pair have
significantly different room optimisation
EQ applied. It’s ironic that monitor
manufacturers such as IK Multimedia
go to great lengths to ensure that pairs
of monitors are closely matched for
frequency response (matching speakers
consistently within, say, 1dB of each
other is perhaps the hardest of speaker
manufacturing challenges), and then have
us intentionally degrade that pair matching
by applying different room optimisation EQ
to each monitor of the pair. Perhaps this
is telling us that stereo pair matching isn’t
IK Multimedia
iLoud Precision MTM
€2399
PROS
• Fine, well-rounded electro-acoustic
design.
• Great bass.
• Strong stereo imaging.
• X-Monitor and ARC 3 add some very
cool extras.
CONS
• None.
SUMMARY
With the Precision MTM, IK
Multimedia’s compact iLoud has grown
up to become an extremely capable
full-size nearfield monitor. X-Monitor
and ARC 3 add significant extra
capability and appeal.
www.soundonsound.com / January 2023 111
 ON TE ST
I K M U LT I M E D I A I L O U D P R E C I S I O N M T M
quite as important as we’ve always been
led to believe?
Using ARC 3 to optimise a pair of
Precision MTMs for the room is quick
and intuitive. The procedure first requires
the supplied MEMS microphone to be
plugged into the appropriate socket on
the rear panel of one of the monitors.
A standard XLR mic cable is used.
The mic is then positioned on a stand
at the listening position. Clicking on
the X-Monitor Calibrate button for the
mic-connected monitor then opens
a dialogue box that confirms the presence
of the mic and instructs on the four
measuring positions required. Clicking
on Capture Point, as illustrated in Screen
1, triggers a series of short sine sweep
signals. When the response at the first
measurement position has been captured,
X-Monitor instructs you to move the
mic to the next position. The process is
repeated for three further mic positions,
following which an optimisation curve for
the first monitor is generated. Repeating
the whole process for
the second monitor
results in a stereo pair of
room-optimised monitors.
Screen 2 shows the before
and after optimisation
response curves for one
channel (the right) in my
Diagram 1: The Precision
MTM, measured using X-Monitor’s
‘smartphone’ profile (red trace),
and the response of a real iPhone
in the same room (blue).
112 January 2023 / www.soundonsound.com
studio room. I’ve shown just one channel
for clarity because the other channel was
reasonably similar.
The ARC 3 process is very slick
and apparently works well, but I was
particularly intrigued by a couple of
aspects. Firstly, there’s no specific
calibration of the MEMS measuring mic.
Professional measurement microphones
are usually supplied with serial-number-
linked calibration curves (and data) so
that they can be relied on to represent
a known reference, but this is not the
case with the ARC 3 MEMS mic. However,
a MEMS mic is supplied with each
Precision MTM, so to investigate how
similar the two mics in my possession
were, I sequentially measured a frequency
response curve for one monitor using the
two review microphones. The result was
reasonably reassuring, with the maximum
difference between the two mics being
less than 1dB. The ARC 3 MEMS mic is
not a professional measuring mic, but it’s
adequate for the job.
Screen 2: The pre- and
post-correction responses of a single
Precision MTM.
The second intrigue is simply
that X-Monitor can treat each
monitor of a pair independently.
In my review of the Neumann
KH 150 [elsewhere in this issue
— Ed.] I mention that Neumann’s
MA 1 room-optimisation app
specifically doesn’t allow
significantly different EQ to
be applied to each monitor of
a pair, because doing so would
risk compromising the stereo
performance. With ARC 3,
however, it’s perfectly possible
to apply different EQ profiles
to left and right monitors, or
even have room optimisation
engaged on one monitor but not
on the other. I can confirm doing
any of these things can make a pair sound
anything from slightly to extremely weird.
Speaker Profiles
In addition to enabling ARC 3, X-Monitor
also enables the selection of EQ profiles
that IK Multimedia say are designed to
mimic the frequency response and phase
characteristics of a variety of different
types of speakers — from portable
Bluetooth boxes and TVs to generic hi-fi
speakers and classic pro monitors. There’s
a total of 30 speaker profiles to choose
from. In terms of both technical validity
and operational utility I’ve personally
never been entirely persuaded by this kind
of speaker modelling. The idea behind
monitoring for me is to provide a reference
standard on which all downstream
reproduction of the material in question
is built. And part of the skill set of mix or
mastering engineers is to know how to
use the reference monitoring such that
translation to downstream playback will
work appropriately. Adding the option for
 ON TE ST
I K M U LT I M E D I A I L O U D P R E C I S I O N M T M
Diagram 3: The on-axis response (red) versus
the response measured 30 degrees above and
below (blue and yellow, respectively).
30 alternative performance profiles, which
may, or perhaps may not, accurately reflect
the performance of imagined downstream
devices seems to me likely to provide
a mechanism for doubt and prevarication
in a mix or mastering session. You might
find that the facility X-Monitor provides to
make a pair of Precision MTMs pretend
to be something they’re not is right up
your boulevard, and while I can report that
the functionality of selecting alternative
monitor profiles works very well, and that
listening to them is superficially interesting,
it’s not something I think I’d use very often.
Phone Home
To illustrate X-Monitor’s ability to model
the response of different species
of speaker, I used FuzzMeasure to
generate a Precision MTM in-room
frequency response curve with the app’s
‘smartphone’ profile engaged. Then
I measured an actual smartphone (my
ancient iPhone 7; FuzzMeasure enables
this by usefully providing the facility to
import and analyse ‘field recordings’) in the
same room, located at a position between
the monitors. The result is displayed in
Diagram 1 and, apart from a few localised
differences, the two curves are remarkably
114 January 2023 / www.soundonsound.com
similar. Furthermore, despite my
philosophical doubts about the inherent
value of making a pair of high-performance
monitors sound like an obsolete iPhone,
my subjective response aligned pretty
much with the objective measurement
— the Precision MTMs did sound like my
iPhone. It seems an oversight, though,
that engaging the X-Monitor smartphone
profile doesn’t force it into a mono
mode, because despite its two speakers,
a smartphone is effectively a mono source
unless you’re a bee, or maybe a small
spider. Stereo combined with restricted
smartphone bandwidth from the Precision
MTMs sounded decidedly incongruous.
Continuing with FuzzMeasure analysis
of the Precision MTM, Diagram 2 shows
its axial frequency response along with
second- and third-harmonic distortion.
The monitor output level was 90dB at 1m.
The frequency response is impressively
flat, with a hint of high-frequency down-tilt.
Distortion is also well controlled, with the
more troublesome third harmonic typically
at between -50dB (0.3%) and -60dB (0.1%)
below the fundamental above 100Hz.
While not quite as impressive as the KH
150, which boasts particularly impressive
distortion performance, the Precision
MTM nonetheless demonstrates distortion
numbers that are highly competitive.
Diagram 2: The Precision MTM’s
on-axis frequency response (red), and
harmonic distortion at 90dB SPL at
1m (second harmonics in green; third
harmonics in blue).
Diagram 3 illustrates the
Precision MTM axial and
30-degree vertically off-axis
frequency response, and it
provides a textbook example
of the power of the D’Appolito
format to narrow dispersion
through the upper midrange,
and similarly how the output
of the relatively large tweeter
dome falls away off-axis above
about 12kHz. Having pointed
that out, I’d suggest it’s of more
academic interest (no, really)
than subjective significance. The
D’Appolito midrange suck-out
is significant, however. It’s
subjectively noticeable and
makes the Precision MTM tonal
balance a little variable with
head position. Midrange balance
dulls appreciably with head
movement away from the central
axis, which might sound like an
issue — but if you’re aware of the effect,
it can be factored in to working practices,
and the midrange suck-out probably
reduces desk, floor and ceiling reflections
in the critical midband.
Diagram 4 looks somewhat of a jumble,
but stay with me and I’ll explain. Each
curve represents a frequency response
captured with the measuring microphone
very close to its specific subject. The
green curve is the close-mic response
of one of the Precision MTM bass/mid
drivers, and the dip at 40Hz reveals the
system’s reflex-port tuning frequency. The
red curve, meanwhile, is the port output
captured with the mic at the port mouth.
Apart from the port output peaking, as
expected from the green curve, at 40Hz,
the red curve reveals some output from
500Hz and above that really ought not
be present. It appears to be midrange
energy ‘leaking’ out through the port, but
of course it could potentially be ‘crosstalk’
from the forward output of the bass/mid
drivers. So in order to check that, the blue
curve was generated by leaving the mic
at the port mouth but blocking the port
itself (with a duster, now you’ve asked). It
confirms that at least some of the output
revealed in the red curve, especially the
peak at 900Hz and above, is port output,
because it’s attenuated when the port is
 Diagram 4: Close-mic
measurements of a bass/mid driver
(green), the port exit when left open
(red) and when blocked (blue).

blocked. Stopping midrange

energy inside a cabinet from
making its way out through the
port is a significant challenge
in two-way ported speakers.
Using an ABR (auxiliary bass
radiator) rather than a simple
port can be a solution, as can
locating the port in the rear
panel of the monitor, as is the
case with the Precision MTM,
where unintended output is
made less troublesome, but
port output such as displayed
by the Precision MTM will
potentially contribute to
the subjective character of
the monitor.
The final FuzzMeasure
analysis, in Diagram 5, shows
the Precision MTM step
response. IK Multimedia
display a step-response
curve on their website
in order to promote the Diagram 5: Precision MTM step response.
monitor’s phase-corrected time-domain fundamentals of monitoring. The Neumann
characteristics, so I thought it would be KH 150 and has set a particularly high time-domain performance — but low levels
interesting to see if I could duplicate it. The bar for nearfield monitoring, but despite of distortion and coloration help too.
answer is yes: the Precision MTM does being less expensive, the Precision Adding ARC 3 and X-Monitor to the
display a notably clean step response, MTM was not remotely outclassed. The mix uncovers a whole extra layer of
especially in terms of its well-damped Precision MTM displays slightly more performance and utility from the Precision
low-frequency ringing. I’ve displayed the character in terms of tonal balance than the MTM. As with any room optimisation
step-response data further along the time unfailingly neutral Neumann; there’s a hint system, the benefits it brings will tend to
axis than is shown on IK Multimedia’s own of midrange coloration and I found the be related to the inherent acoustic quality
curve because, just for interest, I wanted Precision MTM tweeter occasionally slightly of the room. If a room is already good,
to include the arrival of the first room attention-seeking. The low-frequency room optimisation will add little more than
reflection at 37ms (22ms after the monitor performance, though, is very strong. It a final polish, but if the room is poor, it may
output arrives). displays usefully extended bandwidth, well rescue the situation (although the best
combined with an impressive ability to advice would always be to try to improve
Final Thoughts start and stop and securely reproduce the room). In my room, ARC 3 helped
The Precision MTM, along with its pitch. High volume levels also are no clean up the low-frequency performance
partnering X-Monitor app, packs in an problem for the Precision MTM. It can play and smooth out some inherent midrange
awful lot of monitoring technology, but satisfyingly loud without strain, and without character, but even so, the Precision
of course none of it would be worthwhile appearing to change much in terms of its MTMs were more than usable without it.
if it didn’t do a competent job of the tonal character. And while I’ve expressed my uncertainty
As I mentioned earlier, the Precision concerning the true value of all the
MTM’s D’Appolito format is noticeable alternative monitor profiles that X-Monitor
ALTERNATIVES subjectively, and that leaves the monitors provides, I can’t help but acknowledge the
I’ve already mentioned the more
somewhat sensitive to setup in the room: way they have been so smartly integrated
expensive Neumann KH 150 in the
review, but there are also monitors such ear position, especially for genuine into the X-Monitor user interface. I may
as the PSI A14M, the Genelec 8340A, nearfield monitoring, really needs to be wonder if 30 monitor profiles is a truly
the Focal ST6 Solo 6 and the Unity on the tweeter axis. Stereo imaging is useful facility, but it is undoubtedly a very
Audio Rock MkIII to consider — although convincing, with good focus and portrayal cool one.  
they would require extra expense on
of depth. I suspect this is partly a result of
room optimisation to be truly equivalent
to the Precision MTM. the D’Appolito format and partly a result £ €2399.98 per pair including VAT.
of the Precision MTM’s phase-corrected W www.ikmultimedia.com

www.soundonsound.com / January 2023 115

 ON TE ST

Eventide Misha
Eurorack Module

C
onsidering how wide-ranging the art of
sequencing is, rewarding all manner of
approaches and philosophies, it’s no surprise that
sequencers themselves come in all shapes and sizes.
Sequencing is such a fundamentally simple process, but
one that somehow continues to prompt new discussions, a balanced audio out, whose range. User buttons 3 and 4 also lend
and yield new techniques to this day. Eventide Audio sine wave can’t be edited further ‘shift’ functionality.
have now thrown their hat into the ring with Misha. An at all, has no envelope and Press the Record button and punch in
“interval-based instrument and sequencer for Eurorack” doesn’t generate chords even a few intervals, and once you’ve reached
is what it says on the tin. While this is certainly correct, it’s when Misha is in that mode. the end of your tone row the sequence will
but a hint of the possibilities contained behind Misha’s It is fairly useful as a type of automatically start playing. More often than
sleek 28HP faceplate. test tone for tuning purposes, not, I should say, this took me by surprise;
To understand what Misha is best at you need to though, or for generally by which I mean that it started playing
understand the concept of a tone row. Pioneered by the checking your external sooner than I instinctively thought it
serialist composers a century or so ago — specifically oscillator is playing what it’s would, and by extension that I wanted the
Arnold Schoenberg with his 12-note tone row — a tone supposed to. Three additional sequences to be a little longer. A setting
row is (at least, in simple terms) a sequence of notes sets of CV inputs labelled to either limit sequences to a tone row or
with equal emphasis on each, in which no note can X, Y and Z can be assigned allow them to extend beyond this would
be repeated until all the other notes in that sequence to create various musical be welcome, but perhaps that’s (ahem)
have been played. Schoenberg called it a “method of gestures, such as slides, and for a future firmware update. In any case,
composing with 12 tones which are related only with one also to trigger the interval keys this would be more of an issue were it
another”. It follows, then, that Misha is a sequencer whose themselves if desired; which is not for the way Misha can then augment
every movement is based on the relationship between the particularly handy if you want sequences with all manner of clever
current note and the note before. Sure, Eventide haven’t to use Misha with gates from octave and key transpositions, triplets,
religiously stuck to the tone row principle, but for that elsewhere in your system. playback order changes and individual
I daresay we should thank them. The screen’s main page interval shifts. Don’t be fooled into thinking
So, Misha’s row of multi-colour-lit buttons do not presents a ring of notes (your the work is done once your tone row starts
represent notes, they’re intervals. From left to right it’s -4, first clue that no one note is playing back — it’s just the beginning.
-3, -2, -1, 0, +1, +2, +3 and +4. +4 what? And from where? more important than any other) The potential length of a sequence
It’s +4 notes of one of Misha’s 100 (yes, 100) factory scales; alongside indications of the also depends on what scale you’re
and as for the latter question it’s from the previous note. If currently active key and scale. working with, and the amount of scales
you’re in the key of C major and 0 is a C, then +4 is an F. Two push encoders are used on offer here is frankly staggering. We’re
Hit 0 again and you’ll get another F. Hit +4 again and you’ll to navigate around Misha in not just talking major, minor or modal: just
get a B. Hit -2 and you’ll get G. Confused yet? Don’t worry, the first instance, but there’s in case you needed it repeating, Misha
it gets easier. a lot under the hood here: offers 100 unique scales; far more than
Misha’s interface is based around the aforementioned a reset/undo button below the you could name, from Blues to Raga
row of buttons, as well as four user-assignable buttons screen is a welcome addition, and beyond. There are also two banks
(octave jumps, semitones, undo and the like) and while dedicated ‘shift up’ and for user-definable scales on top of this,
a generously-sized central colour screen. It has three ‘shift down’ buttons at the so there’s no real limit to what you can
sets of CV and gate outputs (which are either identical or bottom of the faceplate cycle do tonally. The factory selection even
arranged into chords with that mode switched on), and between pages to adjust bpm, extends to microtonal scales, the most
a TRS Type A MIDI output. There’s also, for some reason, presets, chords and note subdivided of which offers 48 intervals

116 January 2023 / www.soundonsound.com

within an octave. Add this to the fact that it can also walk
chords through any scale’s tone row, and the result is
a bona fide tonal powerhouse, as improvisatory and
expressive as it is mathematical and cerebral.
Now, as we know, sequencers are commonly used for
far more than just melodic patterns. They might just as
well be patched to control a filter, to rhythmically control
the pulse width of a square wave, to dial noise in and out
— the list goes on forever. Misha is undoubtedly designed
for melodic patterns, and not much else. You could
certainly choose to patch it into something else — and it
Modular Profile:
Finlay Shakespeare
T
he founder of Bristol-based developer Future Sound
Systems, Finlay Shakespeare is a certified maverick
creative. The most recent module brought forth
by Future Sound Systems is the FIL4 Timbral Sculptor,
a wild beast of a module combining multimode filtering,
wavefolding and distortion, which you can expect to see
reviewed in these pages soon. Future Sound Systems has
collaborated with some icons in the world of electronic
music, from Throbbing Gristle’s Chris Carter on the
Gristleizer series to Mute Records’ modular-loving founder
Daniel Miller on two modules, Makrow and Stumm.
The company also recently lent their expertise to new
company PWM, designing the analogue architecture of
their debut semi-modular synth, the Malevolent.
On his entry into modular
I got into modular synthesis after deciding to try building
my own equipment. During my teenage years, the Internet
was beginning to get to a point where people could easily
share designs and ideas, so it started to become relatively
easy to find circuits to build, and the necessary electronic
know-how to vaguely understand how these circuits
operate. After a small foray into the 5U format, I got into
Eurorack just around the time that brands such as Make
Noise, The Harvestman and Intelljel began to form. This
was really at the beginning of Eurorack’s domination of the
modern world of modular synthesis.
On his go-to modules
The two modules that really drew me towards the
Eurorack format were Livewire’s Audio Frequency
Generator and The Harvestman’s Tyme Sefari. I still use
these frequently after purchasing them over a decade
ago. I found the prospect of both unusual oscillators and
voltage-controlled delay, looping and recording intriguing,
and I still often think of those two modules — even when
designing my own equipment. More recently, I built
a Westlicht PER|FORMER and Intellijel Rainmaker; two
very different modules but incredibly deep in what they
do. Structuring songs on the PER|FORMER and designing
delay lines on the Rainmaker is a bit of a dream come true.
On working with artists to develop modules
I find that designing modules with artists is a really
interesting experience. It gives a glimpse into their
would certainly perform a function
of some sort — but the finely tuned
complexity of this module would
be completely lost on the result,
which would essentially resemble a
stepped random voltage. You might
see this as a severe limitation,
considering how Misha’s price
places it in stiff competition with
some exceptionally multifunctional
sequencers such as Intellijel’s
artistic practice, perhaps on a more
technical level, with the aim to
solve a problem they’ve run into in
their modular system. It’s extremely
gratifying to know that we’re giving
something back to those artists,
many of whom have been heroes
of mine for a long time. One of the
most exciting results is seeing how
customers then use those designs
in ways that we wouldn’t have ever
expected. This reflects how any
modular system, whether that be
a synthesizer or an entire studio
setup, can present a multitude of
creative possibilities.
On the Gristleizer
The Future Sound Systems Gristleizer
series was born out of a temporary
gap in the market. There had been
commercially available Gristleizers
produced before ours, but somehow
the stock flow had seized up and
there was still a demand. I had been
working with Paul Smith — Chris and
Cosey’s ex-manager — for a good
while then, and I raised the idea of
doing a FSS Gristleizer with him.
Paul passed the idea on to Chris.
Chris was on board once we started
www.soundonsound.com / January 2023
Metropolix, but there’s a satisfying
level of purity here as well. I couldn’t
escape the fact that not only
does Misha do what it does so
exceptionally well, it’s essentially
the only sequencer of its type in
existence. Bravo. William Stokes
£ £649
T Source +44 (0)20 8962 5080
W www.sourcedistribution.co.uk
W www.eventideaudio.com
talking, and he also introduced me
to Roy Gwinn, who had designed
the original circuit which Chris had
extracted from [the July 1975 issue
of] Practical Electronics magazine.
After a couple of years of hardware
revisions, we released the modules
to a pretty huge fanfare. It was
quite unexpected! FSS owes a lot
to Chris and Roy. We wouldn’t have
the following we have now without
the Gristleizer modules — and
their patience!
On the culture of modular
It’s been incredible to see how
modular synthesis has become
such a commonly seen thing now.
When I started experimenting with
electronics in my parents’ garage,
I never expected it to evolve into
a day job. I simply hope that with its
dramatic rise, the community feel
of the ‘early days’ can be somehow
retained. Nevertheless, it’s great
to see that artists have a way of
constructing their own electronic
musical instrument so easily
nowadays. Perhaps we’ve reached
peak synthesis? Only time will tell.
William Stokes
117
 ON TE ST
Modular Round-up: Cases
T
hey may seem like an unromantic part of
modular — even a frustration, since they can be
as expensive the actual modules — but having
the right case can make all the difference to your
system, from power to portability to workflow, and
ultimately impact the most important thing: your
creativity. While it can be tempting to simply opt for a
plain, box-style case every time your system requires
another 3U of rack space (particularly if you’re on
a budget), it’s worth remembering there are a host
of manufacturers out there, each offering inventive
takes on the humble Eurorack case.
Intellijel Palette
Intellijel’s range of sturdy and portable compact
cases promises to facilitate “powerful but focused
mini-systems and control surfaces”, and they’re not
wrong. Available in 62HP or 104HP sizes in either
silver or black, the Palette has rows for both 1U and
3U modules, as well as an array of useful patch
points and sockets at the top. The 62HP version
has two buffered mults, expansion sockets for both
USB and TRS MIDI and balanced quarter-inch audio
input and output sockets. Intellijel also offer lids and
padded gig bags, sold separately, for the complete
Palette package.
www.intellijel.com
118 January 2023 / www.soundonsound.com
4ms Pod
The Intellijel Palette isn’t the only
compact case out there: 4ms’ Pod
cases range in size from 64HP all
the way down to a minuscule 20HP,
small enough to encourage modules
to be thought of and played as
standalone tabletop instruments.
This can be incredibly useful in the
modern era of Eurorack, where some
modules, such as the Make Noise
Morphagene or Intellijel Atlantis, can
capably work by themselves. Made
of sleek, black anodised aluminium,
Pods are available either powered or
unpowered, and can be daisy-chained
together as your setup expands.
www.4mscompany.com
Arturia Rackbrute
It can be difficult to know how best
to configure your system if you’re
integrating it with a semi-modular
hardware synth, keeping all the patch
points of both within easy reach of
one another. With the Rackbrute,
Arturia have come up with a clever
solution called the Link fastening
system; a means by which the case
can be fastened snugly to the side
panels of either the Minibrute 2 or
Minibrute 2S synthesizers to hover
over the top, moveable to whichever
angle is most comfortable. Don’t own
a Minibrute 2 or 2S? No problem:
Rackbrute cases, available in 3U
or 6U, can stand up clear of the
tabletop by themselves and are
also linkable with one another for a
flexible, portable and expandable
powered case.
www.arturia.com
Tiptop Happy Ending Kit
If you’re needing a no-frills,
budget-friendly case solution then
Tiptop Audio’s Happy Ending Kit may
be for you. A simple set of 84HP rails,
it’s definitely not for travelling, but in
the safety of the studio it can either
stand up on its own or be mounted
into a 19-inch rack alongside your
other studio hardware — preferable,
in fact, if you’d rather save table
space. It includes a 4HP uZeus PSU
with two flying bus boards, too.
www.tiptopaudio.com
William Stokes
www.heritageaudio.com The sound of yesterday.....
® built for tomorrow.
 ON TE ST
Sonicware
ROBIN VINCENT
Liven Bass & Beats
onicware hope to bring the same
S intense attitude to wavetables
as they did with FM synthesis on
the Liven XFM I reviewed in SOS June
2022. This time, with the Liven Bass
& Beats, there’s a fierce split between
the dual-oscillator wavetable engine
and the sample-based rhythm engine.
It’s an in-your-face EDM machine that
loves to bang out those beats and get
you dancing.
The general idea is that Liven
Bass & Beats is a portable and
performance-oriented groovebox with
hands-on control over the party that’s
about to kick off. Deeper editing of both
the Bass and Drum engines is available
via some handy overlays which transform
the front panel into a detailed synth
or drum editor. It’s affordably priced
for a groovebox and sits somewhere
between a Korg Volca, Novation
Circuit Tracks and Elektron Models,
but is altogether louder, trashier and
more exciting.
The hardware is identical to the XFM
but with a different paint job and two
120 January 2023 / www.soundonsound.com
Groovebox
Sonicware’s Bass & Beats
has a lot to offer, but you’ll
have to work hard to get it.
function-changing overlays. It has the
same clunky, plastic, velocity agnostic,
clangy piano keys and buttons, a range
of knobs and an inadequate speaker
for the full-on sound that tries to escape
through it. Sonicware still don’t include
any batteries, power supply or any kind of
guide and so the out-of-box experience
is one of silent miscomprehension.
However, like the XFM, there’s lots to
discover behind the pink and blue facade.
Bass Overview
Liven Bass & Beats has two independent
wavetable oscillators fed by 64
wavetables. It has a sub-oscillator and
a noise generator with 16 different
flavours. There are two LFOs and two
envelopes for modulation, and a separate
one for the amplifier. A unison function
lets you stack and detune the oscillators,
and a distortion effect adds some bite.
There is a multimode filter, but it seems to
take a bit of a back seat. There are no filter
controls at all on the front panel, which
seems very odd to me considering the
intention and emphasis of this machine.
However, there are a pair of assignable
knobs that could take on that role.
We’ll get into the synth editing side
in a moment but in terms of front-panel,
hands-on access, anything in pink is to
do with the Bass. You have six banks of
presets with 16 sounds in each and like
everything else in this machine the presets
are fierce and intense. Two more banks
are left for user presets and are currently
all set to basic square waves, which gives
you a good starting point.
There are eight central knobs that
give you some control over the sound.
Most knobs and buttons have a dual
function that are run by the Shift and
Function keys that you will definitely have
to make friends with. The first knob does
bank and preset selection. The next two
are assignable and could be anything
depending on the preset. They also take
care of the range and speed of the quirky
Sweep function, which is an adjustable
glide that applies to every note regardless
 The Bass & Beats in its natural, overlay-free state. The unit measures 297 x 176 x 48mm and weighs 790g.
of the note that went before. The fourth
knob offers Dimension control, which
is essentially the width of the unison
detuning. Its Shift function sets the note
probability in the sequence.
The second row of four knobs offer
pitch modulation, release and gate
length, and then the Machine Gun effect,
which is a kind of tremolo, ratchet-type
affair. That’s about it for the Bass. As I say
there’s no filter to play with unless it’s
been mapped to the assignable knobs
in the Editor and that doesn’t seem to
be the case for most presets. The bass
presets are excellent though and really
show off the potential of the synth engine.
Bass Sequencing
There are two ways to record a sequence
of up to 64 steps. In Step mode you
select a step with the Value button and
press a key to assign the note, or you
can play in a sequence with or without
a metronome and your notes will be
pulled into the nearest step. There are
many of the usual things you can do with
sequencers like change the length, add
some swing, adjust gate length, transpose
and such like, but all the fun really starts
when you engage the Parameter Lock.
With Parameter Lock on you can
hold a step and change more or less
any parameter and it will be burnt into
that step. This goes as far as selecting
a completely different preset, so each
step can have its own sound if you
wish. Your monophonic bass line can
very quickly become filled with all sorts
of changes, switches, and unexpected
happenings. And this is not some sort of
clever multi-lane automation, it’s anything
and everything recorded per step. Which
also makes it less editable, but then Bass
& Beats is all about the here and now and
embraces the idea of the do-over.
One note I should make about saving
— you need to save. Also, make use
of the Copy/Paste of patterns so you
can have various stages and variations
available along the way. Because there
are many ways you can lose what you’re
doing. Accidentally hit the Pattern button
and your unsaved pattern is gone. Move
to a different pattern and all the unsaved
Parameter Lock data is gone. Step away
for a cup of tea and Bass & Beats powers
itself off losing everything you didn’t
save. And because the workflow is quite
challenging when starting out, it is really
easy to go to a wrong function and lose
what you’re working on. Some of it seems
to be down to poor design, for instance,
you can’t turn steps on and off. You can
Clear steps to remove the note allocation
but then the note is gone, and to put it
back in you have to go into Step mode
and re-enter it. There are Undo and Redo
buttons but they seem to be another way
to lose everything you’ve done so far
rather than just the last thing.
Drums Overview
With Drum mode engaged Bass
& Beats becomes a drum machine. Your
previously saved bass line will continue
to play if you’re adding drums to the
same pattern but none of the front-panel
Sonicware
Liven Bass & Beats
£262
PROS
• Enormous fun once you get the hang
of it.
• Comprehensive wavetable synth.
• Excellent collection of drum kits.
• Good value for money.
CONS
• Steep learning curve.
• Challengingly unintuitive.
• Odd workflow.
• Clacky and velocity free keyboard.
SUMMARY
Liven Bass & Beats delivers frantic
bass lines and in-your-face beats with
depths hidden behind a challenging
interface.
www.soundonsound.com / January 2023 121
 ON TE ST
S ONI C WA R E LI V E N BA S S & BE AT S
controls will affect it. Knowing which
mode you’re in is one of the secrets to
success in these Liven boxes.
In the drum library we have 350
sounds carefully selected to be both
modern and timeless. The six banks of 16
kits are infused with special effects, risers
and tension smashing bangs, zaps and
crashes. The vast majority are in-your-face
dancefloor fillers, the intensity of which is
emphasised by the lack of velocity on the
keys and buttons and the non-existence
of any sample layers. Consequently,
these are huge-sounding kits that are
enormous fun to play and will blow back
any remaining hairs on your head.
With your kit loaded, the middle eight
knobs give you some useful hands-on
control over the first four drum sounds
and the first four sound effects. For kick,
snare and hats you have level and decay,
and then with the Shift button you get
tuning on these four and the four effects.
That’s about it for the performance
controls over the drums. Again, I’m
feeling the absence of a dedicated filter
and this time there are no assignable
knobs to map to a cutoff. However, the
effects engine does have a filter in its
arsenal so perhaps all is not yet lost. Lots
more can be done in the Kit Editing page,
which we’ll get to in a minute. But first
let’s look at programming drums.
Drum Sequencing
To get your rhythms into the machine you
can either play them in live or program
them like an 808. The two modes are Pad
for playing and Select for step entry, and
if you find that the keys and buttons have
stopped functioning on the drum track the
reason will be because you’re in Select
mode. Nothing tells you that you’re in that
mode, you just have to know you are.
Effects
There are 13 types of effect covering your
usual chorus, flanger, delay and reverb, plus
bit-crushing, distortion, Isolation (which is
compression, I think) and a Tilt EQ. We also
have a low-pass and high-pass filter, which
makes me very happy. However, they don’t
have much in the way of control. You have
a send amount, and you have an FX Amount —
that’s it. The FX Amount knob varies in what it
does from effect to effect. So for low-pass filter
it’s the cutoff control, but that’s all you get.
The effects do certainly add something to
what you’re doing but seem to work best as
drop-ins during performance. You can only
run one at a time and that be quite limiting.
122 January 2023 / www.soundonsound.com
So, in Pad mode you hit record with or
without a metronome, with or without your
bass line, and bash the buttons until the
end of the pattern. Quantisation is always
in play and so you shouldn’t find yourself
off the beat. If you do or you prefer
the 808 style of note entry, then swap
into Select mode and you get 16 steps
displayed at a time. You can program each
of the 16 sounds in the kit individually.
Parameter Lock works in a similar way
but it’s only really the tuning, decay and
level that you can change for each drum
sound. You can’t seem to swap kits for
different steps like you can the presets
on the bass, but then you can make your
own kits — more on that in a moment.
Bass Editing
There’s a deeper side to the fun and
games of Bass & Beats and that’s in the
Bass or ‘Memory’ Editor. This is a whole
other mode where the groovebox
is stashed away and the front panel
becomes a synthesizer editor for the
wavetable engine. This is physically
reinforced by a pink overlay that drops
over all the knobs and buttons.
The editing is completely separate
from the patterns, so much so that
when you press the required keyboard
combination to enter the Memory Edit
mode, you find yourself starting with the
default preset rather than the sound you
were currently using and perhaps hoping
to edit. However, it is all in there — you
just have to learn a new interface to find
out how to get to it.
The top four knobs become the ADSR
amp envelope, the next row lets you select
wavetables for each of the two oscillators
and their Dimension, or unison width.
Below the knobs we use the top row of
buttons to select the part of the synth we
want to interact with, and the second row
gives sub-sections and parameter choices
that spill over to two pages.
The options are remarkably
comprehensive with detuning, phase
shifting, cross-modulations, sync,
oscillator shaping and blending, but some
of the interface choices seem really odd.
That top row of knobs seems completely
wasted as an ADSR especially as it’s
duplicated over in the envelope section.
Sonicware allocate an individual knob to
Dimension for each oscillator that does
nothing until you dig into Page 2 of the
oscillator sub-section to enable it and
increase the blend. The most interesting
aspect of the wavetable engine, for me,
is selecting the wavetable, morphing
through it, detuning and modulating.
Selecting the waveform is front and
centre but the rest is buried under
sub-buttons. It would make more sense
for those eight main knobs to be split
between the two oscillators and give
you wave select, wavetable position/
morphing, detuning and shaper on each,
because these are the most interesting
and immediate sound-design parameters.
Similarly, there are no dedicated knobs
for the filter and it’s off by default.
And when you turn it on it’s still off
because the filter type has a completely
superfluous and frustrating ‘off’ type. But
if you go to Filter / Page 2 / Filter Type
and set it to an actual filter and go back
to Page 1 to swap between cutoff and
resonance, then you are in business.
These workflow anomalies and
annoyances detract from an otherwise
detailed and interesting wavetable
synthesizer. You’ve got two LFOs and two
modulation envelopes that are easy to
assign and can be pushed at anything.
You’ve got a sub-oscillator with its own
waveforms and detuning. You’ve got
different types of noise you can mix in,
a distortion effect, and you can assign as
many parameters as you like to the two
assignable knobs for use in the patterns.
There’s plenty to do, and plenty of
sound design to be had in here if you can
overcome the particular way Sonicware
have set it all up. You can even import
your own wavetables if you can work
out how. The manual is your friend here,
but while it lists every function it fails to
tell what they are or what you might use
them for. What are the Oscillator Shapers?
I don’t know. There’s a list of their names
in the manual like Squeeze, Level Comb
and Threshold RT but no clues as to
what they do. So, I would say there’s
some room for a more tutorial-orientated
approach to the documentation.
Kit Edit
Swapping to the blue overlay we enter
the Kit Editor. This is all about samples
and where you want them. The step
buttons represent the 16 sounds in your
kit and for each one you can choose one
of the many samples from the library
of Kick, Snare, Hat, Cymbal, Perc 1&2
and SFX 1&2. Despite each step being
labelled as part of a kit you can put in
whatever you want and build your own
custom kit. So, if you want your kick to be
a cymbal, that’s no problem.
 There’s quite a lot you can do to the
sounds in the kit and Bass & Beats likes
to treat them all as individuals. The main
eight knobs give you a Total EQ section
which contains cutoff and resonance, and
then a three-band EQ which is decent
enough to give your samples a bit of
character. You have controls for tuning,
decay, slope, panning and level. You can
also reverse the sample. Each sound can
have its own effects send amount but the
effect is the same across the whole kit.
The kit editing does the job nicely and
is probably the most straightforward thing
I’ve found on this machine. Sadly, you
can’t import your own samples.

In Performance
Once you’ve designed your sound,
customised your kit and have your Bass The Memory Edit overlay reveals controls for the wavetable synth within...
& Beats pattern or chain of patterns
written then you are free to let loose with
some performance features. The eight
central knobs can be tweaked during
playback to add variation. Just remember
to hit the Bass or Drum button depending
on what you want to do. If the button is
red then that’s the one that’s engaged,
if it’s green or orange then it’s on but
not currently in use, or something — it’s
different for other buttons but you’ll get
the hang of it.
The Stutter effect is good fun: you
hold a step and the step repeats until
you release. This is separate for the
bass and drums, which works really well
although I’d quite like the option for it to
be both. There’s also a Random option
on the same button which will activate
a randomisation of the pattern dependant ... while the Kit Edit overlay helps you to edit kits and sounds.
on another setting you’ve set elsewhere.
This is something you really have to set you do when starting up a groovebox is and cool groovebox. It comes with
before you start getting into it, which press play. Normally, on other machines, an exciting library of drum kits that
doesn’t lend itself to improvisation. you’d fiddle with knobs and buttons, perfectly match its EDM vibe and plenty
For drums you can mute the individual things would happen, and you find your of in-your-face bass presets. Behind
channels, but there’s no mute for the happy place. With the Bass & Beats the scenes the wavetable synthesizer
bass, you have to turn it down. They do nothing appears to do anything very is deep, interesting and has a wide
have separate level controls and there’s much on your first visit. Only after careful scope for clever sound design. The
also a line input for external sources. reading of the manual, identifying button workflow can be odd in places and
The fun in any groovebox is not in the combinations, a lot of mistakes, lost as much as I’d like to say it’s just a bit
editing of sounds or the programming of patterns and perseverance does Bass quirky it has a tendency to become
beats, it’s in the sheer joy of what comes & Beats start to open itself up to you. frustrating and annoying until you
out of it when you press play. Liven Bass Once you’ve got the hang of it then you have memorised the way it functions.
& Beats delivers that in spades. It’s bold, can very quickly bang in a bass line, drop There’s nothing wrong with having
brash and gives you just enough to fill in some drums and happily go to town deep and complicated functions but
a dancefloor and have a really good time. on the Stutter button. But it will take you I wish it was more immediate out of the
However, it’s going to take a while a while to get to that point of ease. box. Sometimes you have to work at it
to get there. When I reviewed the Liven to get to the fun part.
XFM I mentioned how I didn’t find it
Conclusion
very intuitive. Bass & Beats has a similar Liven Bass & Beats certainly fulfils it’s £ £262 including VAT.
level of impenetrability. The first thing brief of being a portable, affordable, bold W www.sonicware.jp

www.soundonsound.com / January 2023 123

 INTER VIE W
Queen’s Keyboard Player
For Spike Edney, touring with one of the biggest bands
in the world has meant keeping up with the very latest in
stage keyboard technology.
TONY BACON
I
t’s 1984, and Spike Edney is just a few
gigs into his new job with Queen as
their touring keyboard player and
occasional guitarist. Among the keys set
out discreetly to one side of the stage
at the Birmingham NEC is his Roland
Jupiter-8, and just as the set is about
to move to that year’s big hit, ‘Radio Ga
Ga’, Spike accidentally knocks one of the
Jupiter’s controls.
“It was the slider that adjusted the
arpeggiator’s tempo,” he recalls, “and it had
only about an inch of travel, from zero to
a million — very sensitive. Hung around my
neck on that tour I’d have a little Casio thing
with a metronome that I’d use to manually
set the arpeggiator for ‘Radio Ga Ga’,
and normally this would take quite a few
minutes to do. But my heart sank, because
there wasn’t time for any of that. Roger
Taylor points at me, goes, ‘Two, three...’ and
I had no choice but to press the key.”
Instead of the relaxed arpeggiating
groove that 40,000 fans might have
124 January 2023 / www.soundonsound.com
expected, out came a manic blur of
machine-gunned notes. All four members of
Queen glared at their new keyboard player.
“It’s the only time in my career that I held up
my hands and said, ‘I don’t know what on
earth happened — carry on without it!’ And
that was my third gig. I was really expecting
shit after that,” he says, laughing, “and to
be fair, eyebrows were raised. So in order
to save my arse, I deflected. I said that
something went wrong with the machine.
And since then, I’ve had this motto:
Technology will fuck you up.”
Just Another Gig
It all began for Spike in the usual teenage
bands in his home town of Portsmouth,
which led to gigs in the ’70s playing
keyboards for the likes of soul legends
Ben E King and Edwin Starr, and on to ’80s
tours playing trombone with everyone
from Duran Duran to Thomas Dolby and
Dexys Midnight Runners to the Boomtown
Rats. In the ’90s he worked in Brian May’s
and Roger Taylor’s solo groups, and also
formed Spike’s All Star Band (SAS Band),
which has featured a host of special
guests over the years, including Roger
Daltrey, Jeff Beck, Kiki Dee, Ian Anderson,
Jack Bruce and Ronnie Spector.
Since his two-year live stretch with the
original Queen line-up, which stopped
touring after the Magic outings of 1986
and ended upon Freddie Mercury’s death
in 1991, Spike has toured with all the later
versions of the band. Alongside Brian
May and Roger Taylor, at first in 2005
this was with ex-Free and Bad Company
singer Paul Rodgers, bassist Danny
Miranda, and guitarist Jamie Moses; since
2012, the line-up has featured vocalist
Adam Lambert, bassist Neil Fairclough,
and percussionists Rufus Tiger Taylor or
Tyler Warren. The most recent Queen
gig — Spike describes it as “a last-minute
dash” — was at the Taylor Hawkins tribute
concert in September 2022.
Back to 1985, though, a year after the
Jupiter mishap, and the chance came for
Queen to play at something called Live
Aid. Spike remembers the band thought it
no big deal. “It was another gig,” he says.
“Another short gig.” Twenty-one minutes,
in fact. Twenty-one minutes that would
make Queen even bigger stars than they
were already.
“I was in Bermuda doing a gig with
a mate, a gig holiday, and I had to come
back,” he remembers. They rehearsed

Queen on their 1986 Magic tour, their last
with original frontman Freddie Mercury.
for two days at the Shaw Theatre in
London and worked out a medley to fit
the allotted time. “Queen were always
big on medleys, because there’s so much
material. It’s the way to give an audience
more bang for their bucks. We sat out
around a wooden picnic table, a beautiful
day, I had my Casio thing that also had
a stopwatch in it, and it took us a few
minutes to decide the songs to include.”
Then they pieced them all together,
chopping and adapting to suit. “We
tapped it through and I timed it. So OK,
we’ll start with ‘Bo Rhap’, up to the end of
the guitar solo, then straight into ‘Radio
Ga Ga’, a shortened version of that. Once
we’d figured out the timing through to
‘We Are The Champions’, we went back
in, said right, let’s play ’em and join ’em
up. But it was not a big deal.”
It helped that Queen understood what
was required and that they delivered.
“Play your biggest hits, and play as many
of them as you can fit in, and you will
please more people,” Spike explains. It
sounds obvious, but it wasn’t a scheme
widely adopted at Live Aid. “I was
gobsmacked by how many on that show
did one or two songs with extended guitar
solos forever. Or played their new single.
I think we gauged it right — and I was still
very much the new boy, learning all the
time about how they put things together
and their viewpoint on stuff. I learnt an
awful lot, which stood me in good stead
later when, among other things, I was
musical director for the Nelson Mandela
46664 benefit concerts for AIDS.”
Come the day of Live Aid, Spike recalls
a great sense at Wembley of everyone
being there for the right reasons, and
that the band’s main worries — without
a soundcheck, naturally — concerned
getting on, hearing everything, and getting
through unscathed. “That was the order of
the day: play what we know; don’t cock it
up. We came off after our six songs in 20
minutes, and it was a case of OK, job done.
Simple as that.” Except?
“Except, of course, the hysteria started
the next day,” he says, smiling at the
memory. “Queen this, Queen that. And it
just grew and grew over the years. Now
you get, ‘Oh, the greatest gig ever!’ and all
this kind of stuff. Well I think that’s a load
of old bollocks, myself. It was a band at
the top of its game doing what it does, and
doing it well. I do understand that people
want it to be some magical thing where we
all descended from the heavens, but no. It
shows you what experience and a sense of
occasion can produce.”
The DX5 & The Wardrobe
Spike’s gear had taken a few shifts in the
meantime. A few months before Live Aid,
Photo by Denis O’Regan, copyright Queen Productions Ltd
in May ’85 at the conclusion of the Works
tour, Queen played some dates in Japan,
and Spike took advantage of some time off
in Tokyo to go tech-hunting. First stop saw
him bag a Sony twin-cassette Walkman, and
then he moved on to the Yamaha building.
“The DX7 was all the rage then. So
many records at the time were using
its very clean sounds. I used one in the
studio now and then, but it had this very
peculiar digital sound to it, and I was still
in love with big fat analogue noises. So
I was not much of a DX7 fan. But I went
into the building, and in their showroom
was the glorious DX5 — which was more
or less two DX7s in one. I played that, and
it changed everything — all of a sudden it
had some real depth and width to it.”
He asked where he could get one, but
it was a few months away from release
and he was told this was demo only. “I said
you’ve got to get one to me, and told
them what we were doing. I said ‘I need
it!’ So I think I was one of the first people
to get a DX5, and that’s what I used on
the ’86 Magic tour. By that time my setup
was the DX5 as my main keyboard, plus
an Emulator III, which I’d use for its rather
boxy-sounding sampled piano — certainly
no match for Fred’s Steinway or whatever
it was — and the old Jupiter-8, which was
wonderful for its thick, analogue, paddy
sounds, plus a Roland vocoder for ‘Radio
www.soundonsound.com / January 2023 125
 INTER VIE W
SPIKE EDNEY

Ga Ga’. That took me through the whole of

the Magic tour up to Knebworth in August
’86, which was Fred’s last gig, although of
course we didn’t know that at the time.”
Spike worked with Roger Taylor’s
Queen offshoot, the Cross, from 1988
through to 1990. It was around this time
that he rebelled against the X-frame stands
that were de rigueur at the time, and
developed something more substantial.
“I refused to use those stands. They
really turned me off. I thought ‘This is not
good for the soul, boy.’ So I had my first
B3-type cabinet built. It was known as the
wardrobe, because that’s what it looks like.
We had a cinema props carpenter copy the
Hammond, and he measured up my three
keyboards — the DX5, a two-manual Korg
organ, and the Vocoder — so it was four
manuals in a box. I’ve still got it today and
use it with my SAS Band.
“We had a lot of fun,” he says of his
time with the Cross. “Got slagged from
beginning to end, apart from a clutch of
very loyal fans who loved us and still do,
but the press hated us and the records
didn’t sell. Looking back, I do think they
were good records.”
From there, he worked for a while in the
other Queen offshoot, Brian May’s band, Spike Edney and his ‘wardrobe’ keyboard stand, which was originally built to house a Yamaha DX5,
playing on a number of records and tours a dual-manual Korg organ and a Roland Vocoder.
through the ’90s. The wardrobe didn’t
make it for that, because the idea was be using four keyboards, and would Korg After some more of this to-and-fro,
to have a lighter setup, and Spike used be interested in sponsoring this? And he Spike, who admits his sight reading isn’t
a Roland A-series keyboard controller said ‘Yes indeed!’” great, figured out what the chart was
MIDI’d to a number of Roland modules. “It Once he got home, Spike wondered if doing, and said it was nothing like the way
was the first time I ever used a MIDI system in fact he could bear life as a pit musician, Fred played it. He concluded that he wasn’t
with stuff built into a rack,” he remembers. but he was told he could do it for six going to last in this environment, not least
weeks and then leave if he wanted. because it was all going to sound wrong
Sheet Music “I agreed to that, and then I said about the to him. So at the end of the six weeks,
There was also the hugely successful deal with Korg I’d lined up. ‘Oh no, no, no, he prepared to leave. He was advised,
Queen jukebox musical We Will Rock we only use Kurzweil, and we’ve already however, that he could continue and put in
You, which opened at London’s Dominion bought them.’ You’ve bought them? I was deps if he wanted to have days off.
Theatre in 2002. Spike had been asked getting this stuff for nothing! But we “I said really? I didn’t know about
to take part, and he figured it presented ended up using the Kurzweils. I never that.” Spike pauses for yet more laughter.
a good opportunity to upgrade his thought much of them, and I thought the “I then went about finding as many deps
instruments. A chance meeting with Korgs would have been a better bet.” as I could — in the end I think I had 14
product manager Alan Scally at a NAMM His misgivings about the theatre deps, and at any one time I had half
show in Anaheim started a long-term love job began to grow. The prospect of full a dozen on the go. And then I would
affair with Korg keyboards. employment for a while was certainly work whenever it suited me. I would take
The We Will Rock You organisers had attractive, but there were other worries. months off at a time. And I kept a lot of
told him there would be three keyboard In rehearsal, Spike and the band began piano players working for a long time.
players plus a keyboard-playing conductor. to play ‘Somebody To Love’, but the I was gobsmacked by that skill — very
“I thought that was bloody ridiculous,” conductor called a halt after a few bars. respectful of the people who can do that.
Spike recalls. “Four keyboard players at “He looked at me and said, ‘That’s not So anyway, a six-week engagement at We
a Queen gig? There’s barely enough for how it goes.’ I assured him that it was. Will Rock You turned into 14 years.”
me to play!” But anyway, he got to talking I said ‘This is how we’ve played it since
to Alan about the Triton workstation that 1984, and I was taught it by Freddie
Backup Plan
was on show at Korg’s stand. “I told him Mercury!’ He said, ‘No, you have to read Spike’s setup for the post-2012 version
who I was and that I’d been asked to do the the chart.’ I repeated that I knew this stuff, of Queen, which features Adam Lambert
musical, that I believed we were going to that I’d played it a great deal.” fronting the band, has been consistently

126 January 2023 / www.soundonsound.com

 INTER VIE W
SPIKE EDNEY

This year, Spike swapped out

his Korg Kronos setup for a pair of
Nautilus keyboards, pictured here in the
Steinway-style housing used for Queen’s
Rhapsody tour.
Korg-based. He reckons the story
of how that came to be provides
a good insight into the life of the
touring keyboard player. Picture
a large stage somewhere in
Australia, in the middle of a Queen
tour, in 2014.
“I had the wardrobe, my cabinet
meant to resemble a Hammond B3
— although it actually looks more
like a post-office counter. People
used to say I was half man, half
desk. So anyway, I’m there looking
at my Korg Kronos, and I said to
my tech at the time, ‘If anything
ever went wrong — touch wood,
heaven forbid — do I have any kind
of backup or spare?’ He said yes.
‘Oh, so where is it?’ He said ‘Well,
at the moment it’s out in the truck.’
I said ‘Hmmm, we should maybe
have it a bit closer than that. Why don’t During Queen’s Rhapsody tour, which quick data transfer, and there I was, newly
we try having it right here next to us?’ ran in various chunks from 2019 to 2022, converted to the Nautilus after a crash
I was thinking yet again of my motto: he began to use “something that looks course. And I must say it’s been fantastic.”
Technology will fuck you up. It’s going to like a Steinway grand piano case”, which After the Glasgow gig in June ’22,
happen one day, and probably at a really was originally planned to enclose the two with the new Korg installed, the next date
important moment. As they’re counting Kronos keyboards, main and backup. was something of a big one. “We were
in ‘Bohemian Rhapsody’, let’s say.” “And for the first time ever in my opening at the Queen’s Jubilee in front
Spike asked the now worried-looking career, I played sitting down,” he says. of Buckingham Palace,” Spike reports.
tech how long this backup keyboard “I’d never done that — I’ve always “So you can imagine the kind of tension
took to be ready to play from cold. “He been a stand-up man. But, you know, I was going through, thinking ‘Oh my God,
has a think, says ‘Oh, about seven or I turned 70 this year, so I guess it’s quite is this going to be the greatest cock-up
eight minutes.’ I said ‘What! You mean I’m dignified for me to sit there. Also, I think of all time?’ But the Nautilus was good.
looking at you getting that keyboard out they wanted to give it a grander, more And it saved us. Which was just as well,
of its box, putting it into the post office operatic vibe, because the Rhapsody because from there we went on to 10
counter, and seven or eight minutes stage set was like you’re in an old opera days of sold-out shows at the O2.”
later I can play something?’ I said ‘How theatre. So I understood it. And it was He likes the ease of use of the
do you think that’s going to go down at miles better than the post-office counter, Nautilus, and the main hurdle he
the beginning of “Bo Rhap” when Roger I have to tell you.” faced was to overcome the inevitable
counts me in and I point at you? What During a rehearsal for the 2022 leg, differences between what he was used to
do you think’s going to happen?’ And he however, Spike’s Kronoses started to with the Kronos and the way the Nautilus
said, ‘Never really thought about it.’ I said, misbehave. They had lived a long and operated. “It was things like slight
as calmly as I could, ‘Well, shall we think hard life, and Korg had already suggested changes to where the pages come up,
about it now?’” that Spike might like to retire them and how you get into edit mode, stuff like that.
Spike gathered together Queen’s try their new Nautilus. “Being an old guy I’m ageing and I use glasses, so I would
tech folk and they agreed that a second and resistant to change, I’d said no. Why love if they would allow it to connect or
Kronos, plugged in and ready to go would I change? I’d have to learn a new Wi-Fi to my iPad, which I use for notes
with duplicate sounds and settings, was system. I’m happy with the old ones, and various other bits and pieces.”
quite a smart idea. It meant another I know how to fix things instantly. But now
level added to the post-office B3, and I had little choice. By about the fourth gig
Splitting For Ga Ga
it meant that Spike could relax a little in, I was in Glasgow, and my roadie had to Any teething problems Spike had with the
behind it. go and commandeer a Nautilus from the Nautilus were vastly outweighed by its
local shop, under much protest because positives, and one of his favourite things
The Good Ship Nautilus it was the only one they had — I think we is its set-list feature. “On tour, for example,
More recently, Spike has been enjoying bribed the guy with some tickets for the we might make changes if we get bored,
a much posher housing for his keys. show and he released it. We did a really moving songs around, seeing what they

128 January 2023 / www.soundonsound.com


Spike Edney (far right) takes a bow at a Queen + Adam Lambert show in 2014.
feel like, and it’s great to be able to just
copy the set list, edit it, and there you
are — and you’ve still got your original, so
I can go way back to set lists for the past
God knows how many years. It’s a great
facility. I normally have eight combos on
the screen, and that leaves me two-thirds
of the screen to put notes, like ‘Wait for
drummer’ or ‘Do not count this song in
until singer is ready’, or lyrics that I cock up
because I can’t remember the difference
between verse one and verse three.”
He also loves the ease with which
he can use keyboard splits, which
are essential for some of the songs in
a Queen set. “Still the most complicated
song I have is ‘Radio Ga Ga’. On the left
side of the keyboard I’ll have an octave
of that arpeggiated bass thing that starts
it off. Then there’s a regular piano and
strings, layered, with the strings brought
in with a volume pedal. And up at the high
end of the keyboard I have a vocoder, so
I can do the ‘ray-dee-oh’ bit. And that’s
the whole song right there.”
Sometimes he’ll split the Nautilus keys
to provide piano, strings, and maybe
an octave of organ to one side for, say,
‘Under Pressure’, where the organ only
appears in the middle eight and doesn’t
require a whole keyboard. Generally with
Queen it’s pretty straightforward. And
this is no bad thing, he says. Allied to his
technological motto is Spike’s admirable
inclination to keep things relatively simple.
“I’ve always said that you should be
able to do a gig with a decent piano,
decent strings or pad, and a decent
organ sound,” he reckons. “That should
get you through most things. That’s
what I always start with: what must
I absolutely have? I’ve never been one
for having little dingly digital bells and
effects and stuff, because I find that they
complicate things and get in the way.
And with a loud guitar band like Queen,
all of that stuff would just be lost. In the
studio, of course, you can have endless
combinations of nuanced bits and
pieces. But in front of 50,000 people in
a stadium? I’m afraid a lot of that stuff will
just be fighting to be heard.”
The Synths Of Our Fathers
So what’s it like still to be playing with
the latest incarnation of Queen nearly 40
years since he started with the original
line-up? “It’s always a challenge, and it’s
Creative Commons licence 2.0: flickr.com/bobnjeff
always about not cocking it up,” he says,
smiling once more. “And when you’ve got
quite a few thousand people out there,
it’s such a blast and it’s such fun. The
quality of the playing was always great,
and still is. To be part of that is wonderful,
and that’s never gone away.”
To conclude, we’re back in Portsmouth
in the very early ’70s, and the 18-year-old
Spike is off with his bandmates to see
Robert Moog give a talk at a local venue.
Basically, Robert was in the UK to tell
the story of the Moog synthesizer so
far. Spike found it all interesting, even
if a lot of what was said was technical.
But at this time, very few people really
knew yet what a synth was or could do.
“And then at the end, Robert Moog said,
‘I can see a time when every band will
have a synthesizer, in the same way that
it has a guitar now.’ And the whole place
burst out laughing. It was like him saying
‘I believe every band will have a hot-air
balloon.’ We laughed all the way to the
pub — we were saying that all these
synths really do is make space noises,
wee-yow wee-yow, that you could do that
with a swanee whistle!
“And of course, within a couple of
years we all had one...”
www.soundonsound.com / January 2023 129
 ON TE ST

Electro-Voice
Portable PA Speaker

130 January 2023 / www.soundonsound.com


With its powerful built-in
battery and water-resistant
cabinet, E-V’s new compact
PA will deliver the goods
whatever the weather.
PAUL WHITE
t is almost as thought the music
The built-in mixer can
accommodate two mic/line inputs,
with input 1 capable of providing
phantom power and input 2
switchable to high-impedance mode.
A stereo input can be accessed
via either mini-jack or Bluetooth,
and the mix output is presented
on XLR. The Everse 8 also features
a USB-C port for charging, and a 12V
DC output for use with wireless
systems, or any other 12V gear.

I industry looked into the future, saw

the prospect of rolling power cuts
due to energy shortages, and decided
to build more equipment capable
of running on batteries. Combining
Electro-Voice’s loudspeaker know-how
and Dynacord’s electronic expertise, the
battery-operable Everse 8 is a two-way
speaker, employing an eight-inch woofer
alongside a waveguide-mounted,
one-inch, titanium-dome tweeter, cooled
using ferrofluid. Its moulded, ported
polypropylene cabinet is available in
either black or white and is shaped so
as to allow it to be used horizontally as
a floor monitor, vertically in a ‘kickback’
position, or as an upright pole-mounted
speaker. As the dispersion is symmetrical
at 100 x 100 degrees, the performance is
essentially the same in either landscape
or portrait orientations. A carry handle is
fitted to the top of the cabinet and there’s
a standard pole-mount socket in the
base. Dynacord are responsible for the
amplifier design and for the effects.
The housing, which has
a powder-coated steel grille, meets IP43
requirements for resistance to light rain
and splashes. As shipped, a plastic cover
protects the rear-panel connection, and
an optional slip cover is available for
Electro-Voice Everse 8
£829
PROS
• Clear sound at an impressive level
given the size of the speaker.
• Can run on battery power for long
periods.
• Onboard facilities include basic mixer,
effects and feedback suppression.
• IP43 rated.
CONS
• The inclusion of a high-capacity
lithium-ion battery adds to the cost.
SUMMARY
The Everse 8 is eminently portable,
a strong performer and offers good
value if you need the ability to run from
battery power.
enhanced weather resistance.
This speaker is ideally suited
to portable use, with a weight
of only 7.6kg and a compact
size of 400 x 275 x 272mm.
In common with many
recent portable PA speakers,
the Everse 8 incorporates
Bluetooth music streaming
and digital signal processing.
At its heart is Electro-Voice’s
QuickSmart DSP, which
looks after all the usual
housekeeping, protection and
crossover (2.25kHz) duties, as
well as providing a four-channel
mixer with effects, an automatic feedback
suppressor (AFS), compression and
ducking. Its battery pack provides over
six hours’ running time at maximum level,
and double that at more modest levels.
Thoughtfully, there’s a 12V DC power
port for running an RE3 or R300 wireless
receiver, or any third-party device that has
the same power requirements. There’s
also a USB-C port for charging mobile
devices. When no signal is present,
a battery-preserving power-saving mode
kicks in.
Appy Days
Remote control is possible over
Bluetooth, using the Electro-Voice
QuickSmart mobile app for iOS and
Android devices. Up to six speakers can
be controlled via the app, and two Everse
8s can be paired for stereo playback
over Bluetooth. All functions can also
be accessed and edited using a single
turn-and-push encoder that also doubles
as a master volume control. A small LCD
shows parameter values and battery
status, and there’s an illuminated button
over each input to select it for editing.
If no buttons are lit, then the master
settings are selected for editing. A safety
limiter helps keep peak levels in check
and when this is triggered a ‘LIMIT’
warning is shown on the display. The
battery charges whenever the speaker
is plugged into the mains, even with
its power switch in the off position. As
the battery is a lithium-ion type, it is not
necessary to allow it to fully discharge
before recharging it as was the case with
the older NiCad batteries.
By way of specifications, this
surprisingly compact speaker can
produce a maximum SPL of 121dB
at one metre and it operates over
the 50Hz-20kHz range (-10dB), or
65Hz-18kHz (-3dB). Its amplifier pack
is rated at 200 Watts peak. No other
details are provided for the amplifiers or
power supplies, though given the size
and weight of the speaker, it seems likely
that it uses a switch-mode power supply
and a Class-D amplifier pack. The battery
capacity is 91 Watt-hours.
The mixer features two combi XLR/jack
inputs for mic- or line-level signals with
the option of 48 Volt phantom power on
input 1 and high-impedance instrument
signals on input 2. A 3.5mm mini-jack
accepts stereo signals from an external
source, and there’s a balanced mix output
on XLR. Effects provision is very practical,
with 30 preset effects, the most useful for
live use being the delays or reverbs, the

www.soundonsound.com / January 2023 131

 ON TE ST
ELECTRO-VOICE E VERSE 8

quality of which are excellent, and there

are also a few chorus, chorus/echo and
doubling effects. A jack socket allows
connection to an optional footswitch, for
turning the effects on or off.
The selected effect can be added
to channels 1 and 2, with independent
send levels for each channel. The
mixer’s three-band EQ offers a +6/-12dB
range per band, and there are also
application-specific channel presets for
Flat (no EQ), Voice, Acoustic Guitar and so
on, which variously engage low-cut filters
and apply a suitable EQ curve. A main
parametric EQ with eight bands can be
set up to shape the overall sound. Mixer
setups can be stored (there are five preset
slots available), and as is increasingly
common, there’s a master setting to tailor
the sound for Music, Live, Speech or Club
applications. Music is active by default,
though Live looks to be flatter, especially
at the low end. Less common is a further
range of settings that take account of
the speaker’s position, the options being
Tripod, Monitor or Kickback. There’s also
The Everse 8 can be controlled locally, or from the iOS/Android-compatible QuickSmart app.
a setting for using a separate subwoofer,
with a choice of high-pass frequencies. gigs could get by with just one or two The QuickSmart App is clearly set
The feedback suppressor operates Everse 8s on their own, and because of out, with arrowed tabs to move between
in the usual way, deploying a series of the battery option, there’s no reason to pages in Mixer Edit mode, allowing
very narrow notch filters when feedback stop playing just because there’s a power access to all available controls for Main,
is detected. There are 12 filters, so cut. The speaker is also ideal for garden Effects and Channels as individual pages.
a good tactic is to ring out the system parties, busking and so on, all the more Once a knob is selected, a rectangular
by gradually increasing the gain to so because of its weather resistance! outline appears around it and its value is
provoke feedback until around half the My practical test was carried out at changed using a wide slider at the right
filters are in use, then leave the others a gig in a pub’s covered courtyard, where of the page, which is much easier than
free to counter feedback that might it was used as one side of the PA handling trying to operate a small virtual knob
occur during performance. Ducking and mainly vocals but with additional feeds directly. The AFS and effects sections
compression are available on channels 1 from a separate mixer comprising cajon, can also be turned on and off from the
and 2. In ducker mode, channel 1 or 2 is acoustic guitar and electric guitar. For test Effects page. In the Basic view, the
designated to control the overall level of purposes I ran the speaker from battery master volume and a master three-band
the other inputs. rather than the mains, and it behaved EQ can be accessed along with the
just as well as when mains-powered. master EQ mode, speaker orientation
In Use Its anti-feedback system also came in and sub crossover frequency. Overall
The first thing I should say is that the very useful to cut out offensive ringing the app is very slick, and makes for
quality of sound available from this and to claw back a few dB of extra level easy control as long as you stay within
compact speaker is impressive. If without compromising the overall sound. Bluetooth range.
you want to handle a lot of low end, There was still plenty of charge left in the Overall the Everse 8 ticks all the
then using a pair with a sub would be speaker at the end of the gig too. boxes for a compact speaker that can
a good idea, but for general guitar/voice Equally impressive is the quality of the also run from batteries. It is versatile,
amplification or music playback at effects, and though many of the longer weather-resistant, reasonably powerful
moderate levels, a single Everse 8 reverbs wouldn’t see a lot of use outside and it offers good sound quality with
will do the job very well. It would also an Enya covers band, it’s good to have a useful range of good quality effects,
make an effective stage monitor or the choice. If delay is your thing, then plus automatic feedback suppression,
extension speaker for a PA trying to cover there’s plenty to choose from including compression and ducking. The
oddly shaped venues. I’m not always a nice slapback-style rockabilly delay, and QuickSmart App is simple to use, clearly
a fan of two-way box speakers, especially a tap-tempo delay that you can set from set out and provides deep access to all
those with large woofers, but in this the app. If you are using an external mixer the speaker’s parameters.  
case the eight-inch woofer gives good and don’t need the effects or feedback
coverage of the important midrange. suppressor, you can simply turn them off £ £829 including VAT.
Solo artists and duos playing typical pub in the master settings menu. W www.electrovoice.com

132 January 2023 / www.soundonsound.com

One Plugin.
Endless Effects.
HWYakf'*[ii[dj_WbIekdZjeoi[\\[Yji_ded[
fem[h\kbfbk]#_d$
Effect Rack is the most powerful plug-in in the
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built around 14 full-featured Soundtoys effects. The result
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sound of our most-loved and creative effects with a fun,
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Effect Rack’s global controls give you even more power.
Use the Recycle knob to create instant feedback and the
Mix control for easy parallel processing. The extensive
library of presets makes it easy to find new creative
inspiration and to save your own multi-effect masterpieces.
Free 30-day Trial at soundtoys.com
 ON TE ST
iZotope RX10
Audio Repair Software
What’s new in the latest version of iZotope’s
market-leading audio restoration app?
HUGH ROBJOHNS
hristmas comes but once a year
C and, almost as reliably, so does
iZotope’s annual update to
their ever-more-impressive audio repair
software, RX. The latest incarnation,
RX10, is available in the usual three
versions: Elements, for those with fairly
basic audio restoration needs; Advanced,
aimed at demanding professional audio
post-production users; and Standard,
for those with slightly less challenging
requirements. RX9 presets can be
imported and used in RX10, of course,
and the software has been updated with
native support of Apple M1 processors.
Previous RX users will feel right
at home with RX10, as very little has
changed on the home screen and the
vast majority of modules and features
134 January 2023 / www.soundonsound.com
are unchanged from RX9. However, as
we’ve come to expect, RX10 introduces
several new features (seven in all), with
one available to all three RX variants, four
more for the Standard version, and two
exclusively for the Advanced package. I’ll
document these below.
Repair Assistant
The Repair Assistant, already present in
RX8 and RX9, has been updated for all
versions. This analyses the material and,
using machine learning, identifies a variety
of common ‘faults’ (clipping, clicks, hum,
and noise), and then offers a set of
processing module suggestions intended
to rectify them. Clearly, this facility is
aimed primarily at the less experienced
user, but that hasn’t stopped the previous
versions from often providing a good
starting point — even for RX gurus.
In previous versions of RX, the Repair
Assistant has been fast and pleasingly
effective in most typical cases, but in
RX10 it has been rebuilt completely, from
the ground up. It features an all-new
user interface, as well as a radically new
machine-learning analyser function. Once
launched, the first job is to select the type
of audio material from the centre-top box,
with options for voice, music, percussion,
or sound effects (the previous incarnation
offered only dialogue, music and ‘other’).
Doing this enables RX to preselect
relevant module options.
With the appropriate audio type
selected, a Learn button starts the
analysis. It is no doubt more advanced
but, sadly, is achingly slower than
the previous version. On my current
Windows machine, a 45-minute podcast
track took less than 30s to analyse in
RX9, but well over 20 minutes in RX10.
The computer obviously plays a part in
this time, as a colleague using one of
the better-specified M1 MacBook Pros
completed a similar task (a 45-minute
dialogue track) in 10 minutes, but it’s
significantly slower than the older version.
It’s also worth noting that the plug-in
version of Repair Assistant analyses the
audio in real time, rather than through
 iZotope RX10
PROS
• Small but worthwhile upgrades to
four processing modules.
• New speech-to-text feature improves
navigation of long speech tracks.
• Native Apple M1 compatibility.
CONS
• Better results from updated
processes take more time.
SUMMARY
One of the world’s best audio
restoration packages has been updated
to make it even better, although the
upgrades this time are relatively small
and won’t be relevant to everyone.
a Learn pass of the whole file or a partial
selection. Once the offline analysis is
complete the Repair Assistant panel offers
a wider range of processing modules,
including de-clip, de-click, de-noise,
de-reverb and de-ess (for voice mode, or
de-harsh for the others). It also includes
de-hum, apparently, but I couldn’t find
any control options for that — perhaps
my sample track was hum-free! Each
module has an on/off button, a listening
(solo) button, and a single ‘amount’ control
in the form of a circle with up/down
arrows and a coloured ‘fill’ level. The
bottom-centre de-noise module enjoys
an oversized amount circle dominating
the centre of the panel, which I initially
found confusing and distracting. All of
these amount displays change colour
depending on the selected audio mode
(blue for voice, cyan for music, green for
percussion, purple for sound effects).
In the RX10 Advanced an extra facility
at the bottom of the Repair Assistant
window opens the recommended
processing as a conventional module
chain, and this obviously grants the user
access to more module parameters for
finessing the result, as well as enabling
the recommended/tweaked processing
to be saved as a preset chain. Clearly,
the RX10 Repair Assistant is a much
more sophisticated version than its
predecessor, with enhanced capabilities
and, presumably, far more accurate
machine learning. However, the sluggish
offline analysis stage is disappointing
— there’s considerable room for further
optimisation there, I think!
Text Navigation
When working with a clean speech
recording of more than 10s duration, the
new Text Navigation mode (not available
The Repair Assistant, available for all versions of RX10, benefits from a significant overhaul.
in Elements) can be activated through an
icon resembling a speech balloon bubble
on the bottom left of the spectrogram
display. This uses a speech-to-text
translator (US English only) which
analyses the audio track at roughly 8x
speed and then generates a series of
tabs above, and synchronised with, the
corresponding spectrogram revealing the
spoken words.
The idea is to make it much easier
to navigate a long speech file visually
via the text translation, to locate edit
points, retakes, and so on. Like most
speech-to-text systems it’s not 100%
accurate, and heavy background noises
or music, as well as strong accents, tend
to reduce accuracy further. If necessary,
though, the text can be corrected by
clicking in the tab and overtyping the real
words, and it’s a very useful navigation
aid, particularly in long speech tracks
such as interviews and podcasts. I found
it very useful and, much to my surprise,
quickly grew to rely upon it!
When the Text Navigation is active,
an adjacent button (looking like a menu
list) opens a side window that enables
a search of the translated text. By typing
a word or phrase into the window’s
search box, every occurrence is listed,
along with its entry time. Clicking on
a particular entry selects it for playback,
and left/right arrows allow quick
scrolling through the listed words.
These text-based navigation and search
functions make it remarkably fast to
find specific parts of a long speech, so
locating retakes or other edit points
becomes much faster and more accurate.
Although I’ve managed to edit dialogue
tracks on tape and in DAWs for nearly
40 years, just by scrubbing through and
listening, RX10’s Text Navigation is one of
those features that, once used, will leave
you wondering how you ever coped
without it.
Another new feature, Multiple Speaker
Detection, is an extension of this Text
Navigation function for RX10 Advanced
users only. It can automatically detect
up to eight different voices within
a mixed speech track, identifying their
contributions through different colour
text tabs. Within the Search box, each
identified voice has a different coloured
radio button and can be uniquely
named, if required. Clicking on a button
automatically selects all of that speaker’s
contributions to help target text searches
or to apply processing to a specific voice.
De-hum Dynamic
Adaptive Mode
Standard and Advanced users also
benefit from an upgrade of the Dynamic
Hum Filter, first introduced in RX9:
clicking a tick box activates a new
Adaptive Dynamic Mode. The idea
is to analyse the audio signal and
automatically adjust the dynamic filter
parameters accordingly, without needing
A ‘Feather’ tool now allows you to specify
crossfades for your edits, in both the frequency
and time domains.
www.soundonsound.com / January 2023 135
 ON TE ST
I ZO T O P E R X 10
The new speech-to-text feature: while the
transcription isn’t always accurate, it makes
navigating long dialogue recordings much speedier!
to run through a Learn pass first, as we
had to in RX9. (The Learn feature is still
available for both the static and dynamic
modes, if preferred). However, a note in
the user manual states that, because of
the inherently high latency involved in
this Adaptive mode, the feature “is not
optimised for real-time use” and iZotope
recommend that the Adaptive mode is
only employed for stand-alone and offline
plug-in processing, where the analysis
latency is irrelevant.
Another minor change on this module
is the removal of the high- and low-pass
filter parameter boxes for the Static filter
mode. This is simply to declutter the user
interface, though, and the facilities are
still present but accessed through sliding
buttons on the frequency graphic display.
Clicking one of these buttons causes the
hum filter’s frequency and Q sliders to be
renamed as the corresponding high- or
low-frequency filter parameters. All very
straightforward and logical.
Other Improvements
Selection Feathering is a new feature for
Standard and Advanced, and is selected
by a feather icon squeezed between
the Select Harmonics tool and the View
Clip Gain button, just under the main
spectrograph display. When selecting
areas of the spectrograph for processing,
136 January 2023 / www.soundonsound.com
this Feathering button brings up two
new sliders which adjust a crossfade
duration on both the time axis (from zero
to 1000 ms) and the frequency axis (zero
to 1.5 octaves). The idea is to help blend
modified sections of the spectrograph into
the existing material more subtly, helping
to remove any abrupt discontinuities and
making impossible edits possible... or at
least, even more subtle!
Somewhat bizarrely, the new First-time
User Experience function is not included
in the Elements version, which, of course,
is inherently aimed at first-time users!
For Standard and Advanced users, this
new feature provides a semi-interactive
eight-page animated tour of RX10, giving
the unfamiliar novice user a gentle
overview of how it all works.
The Spectral Recovery function
(Advanced only) has been treated to an
upgrade. This process resynthesizes
any audio frequencies that are lost when
working with online communications
technologies such as are common for
remote meetings, video calls, and so
on. The technology basically recreates
absent signal components lost through
the data compression process, essentially
resynthesizing speech frequencies above
4kHz, as well as lower frequencies lost
through system high-pass filtering. The
RX10 update uses a more advanced
algorithm to improve the quality of the
restored signal even further.
Impressions
RX10 is a recognisable
step up with a few genuinely
useful updates, and the
package as a whole remains
outstandingly powerful, versatile,
and remarkably easy to use.
For most, the Standard version
will offer the best value for
money — it is a phenomenally
well-specified package, and
it’s a very big jump in price for
the Advanced version, most of
whose additional modules relate
to processing dialogue from
The Spectral Recovery facility (for RX10
Advanced only), which synthesizes new
information to ‘restore’ compressed and
filtered recordings, has been improved.
location recordings, as might be needed
by professional film and TV sound
editors. iZotope provide a very handy
feature comparison list if you need to
check whether the Standard (or the even
more affordable Elements) version will
meet your needs: www.izotope.com/en/
products/rx/features.html#comp.
However, from the point of view of an
end user who already owns RX9, I have
to be honest and report that there’s little
in the new additions and improvements
that would have me running to a retailer
begging them to take a selection of
internal organs or my first-born child so
that I may possess RX10. In the past, the
capability jumps between some versions
of RX have been huge and genuinely
game-changing, and despite there being
several worthwhile improvements, I don’t
feel that’s the case this time. Yes, the
Spectral Recovery, De-Hum and Repair
Assistant modules are all incrementally
better than their forebears, but not to
such a degree that they render RX9’s
older modules instantly redundant.
The new Selection Feathering option is
a notable upgrade, and it will undoubtedly
appeal to users who work a lot with RX on
high-end professional projects. Likewise,
the Text Navigation and Multiple Speaker
Detection features could certainly be very
convenient time savers for those working
with long speech files to tight deadlines.
But for a lot of RX9 users, neither of those
features will seem essential.
That said, behind the scenes, I’m
quite certain that the iZotope boffins
have worked very hard over the last
year to develop the underpinning
machine-learning technologies, that
RX10 is genuinely an engineering
step up from RX9 — particularly with
its native compatibility with Apple M1
platforms, which will inevitably be the
must-buy feature for many. And I’m sure
it stands them in good stead for the next
version too!  
£ Elements £135.35, Standard £418.63,
Advanced £1257.99. Prices include VAT.
(Standard and Advanced also available
in some bundles, including subscription
options.)
W www.izotope.com
3 - O S C I L L A T O R E N H A N C E D L A D D E R F I L T E R A N A L O G P O LY S Y N T H

sequential.com
 ON TE ST

NovoNotes Ambience Enhancer

Multi-channel Ambience Plug-in
Ambience Enhancer is a clever 64-bit VST3/AU/AAX
plug-in for Mac OS 11.0 or above (there’s no Windows
version), and it runs on both Intel and Apple silicon
processors. Authorisation is by a licence key, which is
emailed on purchase.
Essentially, Ambience Enhancer analyses the incoming
signal and separates it into two parts: ambience and
direct sound. This allows you to control the ambience
as you wish, while leaving the direct signal unaffected.
NovoNotes say they achieve this by analysing the phase
differences between channels, based on the principles
behind M-S encoding — and that means, of course,
that it can’t do anything for mono recordings. There are
currently two versions: the standard version (reviewed
here), which can handle multi-channel sources, and
a Lite version, which is stereo only and free to download.
A third ‘multiband’ version is also planned. was more ‘around me’. The electric piano, with some reverb ‘baked’
The GUI is pleasingly simple and the controls main kit sound was coming into the recording, and it really shone:
self-explanatory; you won’t need a degree in manual from my front L, C and R I could open the sound right out and
reading to get instant results. On the left are four speakers, with the closer bring up the reverb without drowning the
horizontal slider controls, and on the right a pretty visual stereo room mics assigned dry signal or compromising the piano’s
representation of what’s going on. Top left, the Ambience to my side speakers and attack. Pushing in just a millisecond or
Gain slider allows you to apply up to ±12db of gain to the far mics to the rears. two of Ambience Delay opened it out
adjust the amount of ambience already present in the Again, using Ambience even further, all the while retaining the
sound. Next comes Ambience Low Pass, a variable Enhancer across the 7.1.2 integrity of the original sound. Going the
(3-22 kHz) low-pass filter applied across the Ambience drum bus, I could bring other way, backing the Ambience Gain off
signal path only. Ambience Delay is, effectively, those room mics up nicely, a little dried the signal in a very natural
a pre-delay: it can apply a 0-50 millisecond delay to the without affecting the drier way, making the piano sound a little
Ambient signal path. Finally, there’s an overall Output close mics. more forward and prominent without any
Gain control (not included with the Lite version), which Of course, in these sonic detriment.
allows you to compensate for the overall level changes specific examples I could As with M-S balancing, things can start
inherent in this process. simply have raised the to sound a bit weird if you push things
Double clicking any slider will reset the position of that level of the room mics too far. Too much Ambience Gain can
slider back to its default flat setting. Also, hovering over in the drum bus, but seem to leave a hole in the middle, while
a slider’s name will give you a detailed description of when working on Atmos backing off the Ambience too much can
the function of each control should you need it. A button mixes it’s not always give a mono effect. Used judiciously,
accesses a Settings pane, where you’ll find your license possible to access the though, Ambience Enhancer is a subtle
info and licence type (you can purchase a month-long individual channels — I’m way of bringing out or reducing the
licence or a perpetual one), and if running the plug-in on often sent stereo stems natural ambience of any multi-channel
anything with more channels than stereo, you have the from someone else’s sound source.
option to bypass channels that are assigned as LFEs. stereo mix, so I can see While very useful, it’s probably not
I first tried Ambience Enhancer on the stereo drum bus Ambience Enhancer a plug-in you’d use every day, which is
of a song I’d mixed a few months before. The drums were being very useful. It would perhaps why NovoNotes offer a very
recorded in a medium-sized (roughly 8 x 12m) room, with be possible to make reasonable monthly licence. But if you
the kit close-miked at one end of the room, a mono room a copy of the stereo drum work with stereo stems in immersive
mic about 1m in front of the kit, a stereo room configuration stem, add an instance of audio, with bounced stereo mixes where
halfway down the room, and another stereo pair at the far Ambience Enhancer to you wish the mixer had backed off the
end of the room. These drums sounded natural, fat and crank up the ambience on reverb just a touch, or if you’ve been sent
punchy with a nice bit of room ambience mixed in with that channel, and follow it a mix project in which the only drums
them. With Ambient Enhancer, I was able to dial in a little with your upmix plug-in of are stereo loops and samples, Ambience
more room sound on the stereo bus without losing clarity choice: you’d then have Enhancer offers a very quick, easy and
or weight in the close mics. Obviously, bringing up the the ability to spread the sonically satisfying way to dial in just the
ambience made things brighter — there was a lot of cymbal natural ambience of the right amount of ambience and space.
sound in those room mics — but the Ambience low-pass stereo drum stem around Trevor Michael
filter made it easy to compensate; I could ensure things the other channels in your
£ Lite version: free. Standard version perpetual
sounded natural and in-keeping with the original sound. immersive environment. licence: $68. 30-day licence: $6.80.
Next, I opened my Dolby Atmos mix of the same song, I also tried Ambience Multiband version will be $128.
with the same kit running through a 7.1.2 bus — the sound Enhancer on a stereo W https://novo-notes.com

138 January 2023 / www.soundonsound.com

 a trim pot inside the
TC Electronic Bucket Brigade Delay case. To access this, four
Analogue Delay Pedal small crosshead screws
It seems that bucket-brigade delay pedals are making must be removed, along
something of a comeback, and it’s almost certainly due with the in and out jack
to the unique way in which their delays degrade as they retaining nuts. That’s a bit
repeat — something which helps the delays ‘sit behind’ of a fiddle, perhaps, but
the dry sound in a pleasing way. Early models were you should only need to
plagued by excessive noise or restricted maximum delay do it once, if at all. A red
times, but while this type of delay circuit can never be LED indicates whether
entirely noise-free, modern devices can perform a lot the pedal is active, while
better. TC’s Bucket Brigade Delay can go from a fast the footswitch leaves
slapback up to a maximum 600ms delay time, without the dry signal intact
intrusive levels of background hiss — I didn’t notice any when operated.
significant noise when using it normally. Sonically, this pedal
This pedal is not a part of TC’s TonePrint series, and captures everything
so has its own distinctive, folded-steel enclosure. It that’s desirable about
has a small footprint but is a little deeper than a typical the bucket-brigade delay
cast-alloy pedal, measuring 95 x 57 x 56mm. Weighing sound, with just the right
300g, it also feels very solid. A 9V power supply, capable amount of warm-sounding strong chorus-like effect. Perhaps the
of delivering 40mA or more, is required since there’s no signal degradation to nearest competitor to this pedal is the
battery option. The power jack is located between the the repeats but not so recently released EHX Nano Memory Man
in and out jacks at the back end of the case to make the much top-end loss that it Deluxe, which has a 550ms maximum
most of pedalboard space. On the top panel are familiar sounds dull. The feedback delay time and also features adjustable
controls for Delay time, delay Volume and Feedback, but control can go all the modulation. There’s also the Maestro
there’s also a Depth knob, which controls the amount of way to self-oscillation, Discoverer Delay and that too includes
LFO-controlled pitch modulation when the small toggle which should please the a modulation section, but it looks like the
switch in the middle of the control section is set to Mod. dub fraternity, and the TC will be the least costly of the bunch so
The dry sound always comes through at unity gain. modulation can go from very much worth a test drive. Paul White
Interestingly, TC have made it possible for the user to an almost unnoticeable £ £65 including VAT.
change the speed of the modulation LFO by adjusting worn-tape ‘wow’ to a very W www.tcelectronic.com

Warm Audio Centavo in being an almost exact replica, with

Overdrive pedal
Launched in 1994, the
original Klon Centaur
Professional Overdrive
pedal was surely the
product that paved the
way for today’s vibrant
‘boutique’ guitar-pedal
market. Created by guitar
player Bill Finnegan in
collaboration with some
MIT electronics engineers
(who apparently had no
previous experience of
guitar-pedal design!) the
Centaur would go on to
become, for some, the
most sought-after pedal
that the market would
ever see. Warm Audio’s
reproductions of classic
mics and studio gear often
give more than a nod to the
aesthetics of the products
that inspired them, and this
one goes a little further
the Texas-based manufacturer’s new
Centavo pedal sporting precisely the same
custom-cast, oversized housing, correctly
proportioned Oxblood knobs and similar
‘mythical creature’ graphic, as an original
Klon Centaur.
Inside, the Centavo is stated to be
a faithful recreation of the original circuit,
too, employing the same TL072 op-amps,
the famed 1N34A Germanium diodes,
carbon resistors and the essential charge
pump voltage regulator in the power
supply that makes an 18V supply available
internally from a 9V PSU or battery
source. The 1N34A diodes are a big various versions of each component to land on the right
part of the mythology of the Centaur, recipe to get the true Klon Centaur sound”.
but that component designation itself Although never explicitly marketed as such, the
covers rather a lot of diodes that are not Centaur was designed primarily just to push the
necessarily exactly the same. The ones front-end of Fender tube amps, to allow them to deliver
Bill Finnegan and his collaborators chose a bit more of the sustain and feel that they offer when
for the Centaur had a very particular turned up to their optimum level: something that was
forward-voltage characteristic, without becoming increasingly difficult to do in the ever more
which you, allegedly, don’t quite have volume-restricted clubs and bars of Bill’s local area. That
a Centaur. I can’t say exactly which 1N34A was a job most often undertaken by various incarnations
diodes are in the Centavo, but the Warm of the Maxon/Ibanez Tube Screamer circuit, but Bill
Audio production team are said to have wanted something that could retain more of the sound of
“spent may hours meticulously comparing the original amplifier. To assist with this, the Klon circuit

www.soundonsound.com / January 2023 139

 ON TE ST

ended up being designed with a clean
path in parallel with its distortion process,
with the clean signal getting progressively
faded out as the distortion is increased.
The concept of the ‘transparent overdrive’
originates here, even though the effect of
the circuit is actually far from transparent.
The ‘classic’ setup for a Centaur is
with the Gain at a very conservative
nine o’clock, the Treble more or less
neutral and the Output somewhere
above unity to push the front-end of the
amp hard. Running the Centavo into
a ’60s Deluxe Reverb, with the volume
on four, so still mostly clean, I could
immediately hear that this does what
a Centaur is supposed to do in that
mode. There’s no obvious distortion or
limiting of the dynamics, but the sound
just seems ‘curated’ and polished. Now,
I have to admit that I last played a real
Centaur in about 1996, but I have used
a couple of the better ‘Klones’ since,
and that has always been the common
experience. What you get is still very
much the sound of the amp, just nicer
and more focused, as if trimmed a little
at the top and bottom of the spectrum, to
leave you with more of the bits you want
and less of anything that might otherwise
get in the way.
One thing I remember very clearly
about working with the Centaur circuit is
that the more you turn up the Gain, getting
into the zone where the pedal itself is
generating distortion rather than the
amp, the less special I think it becomes.
And that is exactly the same for me with
the Centavo, which perhaps attests to
a level of authenticity in the circuit. It is
still unusually ‘fast’ and articulate — that
18V internal supply adds headroom and
increases the slew rate of the op-amp,
allowing it to preserve transients better
— but it is mostly op-amp distortion to
begin with, at least until you start to work
the ‘special’ diodes, which are deployed
in a hard-clipping configuration, like
a RAT pedal or a Boss DS-1, rather than
the gentler, diodes-in-the-feedback-loop,
soft-clipping effect of a Tube Screamer
and its many derivatives.
The Centavo incorporates a switchable
popular mod for Centaurs that subtly
extends the bass response, which I find
makes for a more complete sound when
using high-gain settings, but there are
plenty of players who reject the ‘low-gain
Klon’ orthodoxy altogether and love the
standard Centaur circuit just as a full-on
distortion pedal, with or without mod.
Warm Audio’s Centavo has buffered
bypass like the original Klon model, and
joins a long list of historically faithful
‘Klones’ that all do ‘the Klon thing’ with
only quite subtle differences. This one
does it, too, ‘fabled diodes’ or not, and if
you are keen to have the look as well as
the sound, you might also want to bear
in mind that you could buy about 20 of
these for less than the price being asked
for the occasional original that comes up
for sale! Dave Lockwood
£ £189 including VAT.
W www.warmaudio.com

Warm Audio Warmdrive

Overdrive Pedal
Reviewed alongside Warm Audio’s Centavo
(see above) is their version of designer
Alfonso Hermida’s famous Zendrive pedal.
Unlike the original Centaur, Zendrives are
still being made, albeit under licence by
USA manufacturers Lovepedal. But circuits
by themselves cannot be protected by
patent, so there are numerous ‘unofficial’
versions available from other makers,
too. There is some symmetry to these
two Warm Audio pedals being launched
together, as one of the models that
inspired them, the Klon Centaur, was
conceived out of dissatisfaction with the
ubiquitous Tube Screamer, whilst the other,
the Zendrive, actually is an ‘improved’ Tube
Screamer circuit. The latter contention
might seem heretical to some, but any nomenclature of the lower gain to avoid sounding too congested, whilst you
overdrive pedal that uses ‘soft-clipping original Zendrive exactly, can get away with a lot more gain when trimming off
diodes in the feedback loop of an op-amp’ with Volume, Gain, Tone more bass. The Tone control then lets you filter how
is to some extent building on the heritage and Voice controls. This much treble makes it through to the output, which works
of the Tube Screamer. Where the Zendrive last control sets how rather differently to the Tube Screamer’s active Tone
circuit differs most significantly, however, much bass makes it control, offering both cut and boost of highs after a fixed
is in giving the user access to two of the through to the clipping high-frequency roll-off has already been applied.
really influential parameters that are fixed stage, and simultaneously In contrast to a standard Tube Screamer’s
characteristics in the Tube Screamer, affects the amount of simple symmetrical pair of silicon clipping diodes,
specifically the amount of bass cut prior gain available, making the the Warmdrive, like any proper Zen circuit, uses
to the distortion stage and the amount of two controls interactive. a combination of 1N34A Germanium and Schottky BAT41
high-frequency roll-off after distortion. Lower Voice settings silicon diodes plus 2N7000 MOSFETs in an asymmetrical
The Warmdrive mimics both the that allow more bass clipping arrangement. That slightly unusual configuration
appearance and the control layout and through generally need seems like it might have been something arrived at by

140 January 2023 / www.soundonsound.com

 ear, and adds complexity to the sound
without creating the kind of harshness
that a wide asymmetry can generate. The
MOSFETs here are actually just being
used as clipping diodes, as they are in an
OCD or MOSFET Fulldrive, for example.
You don’t need to know or understand
any of that, of course, to appreciate the
Warmdrive, but when so many pedals
are essentially based on the same few
original circuits, it may be useful to
know which ones offer something a little
different, and why.
A lot of users run a Zendrive (or
clone) into the clean channel of a tube
amp to create a ‘singing’ lead tone, with
the pedal gain somewhere up around
12 o’clock. The Voice setting is then
very much guitar-dependent: lower for
single coils, higher for humbuckers, in
general. But it is also very much amp- and
cab-dependent, too. Used that way, the
Zen design gained its reputation as the
go-to pedal for the Robben Ford/Larry
Carlton ‘fusion lead’ sound, and the
Warmdrive will certainly give you a nice
version of that. This is not a high-gain
pedal design, being famed more for its
touch sensitivity, fundamental tonality
and the quality rather than the quantity
of its distortion voicing. But, with the
Gain up anywhere around halfway or
more, I’ve always felt that the Zen circuit
doesn’t clean up as much as I’d like with
the guitar’s volume control backed off,
and the Warmdrive is exactly the same.
It seems also that the bridge and neck
pickups would, often, ideally benefit from
different Voice settings.
The absolute best of the Zen design
in my experience, equally true of the
Warmdrive, is to be found running it
into a slightly higher-gain channel, with
the pedal’s Gain kept lower, around
nine o’clock or just past the point where
you hear the louder gain begin to kick in.
With Volume around or just above unity,
the combination of the pedal and the amp
gives enough drive and sustain for the
sought-after ‘singing’ lead tone, while the
pedal itself seems to regain the ability to
clean up really well. There seems to be
even more touch sensitivity, and whatever
Voice setting you settle on for either
pickup will work just fine with the other.
That is where the real touch-sensitive
‘note-bloom magic’ in this design is to be
found, for me.
Unlike the situation with the Centaur,
which has been out of production in
its original form for many years, some
people may not be totally comfortable
with so exact a physical replica of
something that’s still very much a live
product, but the market itself ultimately
tends to be the judge of such things, and
Warm Audio are far from alone in taking
this approach. Their Warmdrive certainly
goes a long way towards replicating the
sound of the original, ‘classic’ version
of the pedal design that helped create
the whole ‘Dumble-in-a-box’ concept. Of
course, no overdrive pedal, on its own,
can fully replicate any amp, still less
a really special one, but in combination
with the right rig, with the right settings,
this one can certainly get you into that
area, and I have to say I like how it does
it! Dave Lockwood
£ £159 including VAT.
W www.warmaudio.com

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 ON TE ST
Heavyocity
Vocalise 3
Kontakt Instrument
++++
Vocalise 3 is Heavyocity’s eighth
release in their Gravity Pack series,
a line of products that are built
upon the full Gravity engine (but not
requiring the user to own Gravity
itself) and with each title focused on
a specific instrument/sound group
for the sample base. No prizes for
guessing the theme is vocal-based
here; Vocalise 3 provides 7GB of
vocal-inspired sample data and
runs in a dedicated front-end within
Kontakt 6.7.1 (Player or full version).
In total, nearly 200 presets
are supplied. These are split into
five categories within Kontakt.
The first of these is a collection of
vowel-based, evocative, melodic
phrases. These presets are divided
into multiple keys, with major,
minor and harmonic minor options
provided, and each preset also
includes keyswitch options for
transposing the phrases if you need
to match a specific key. While the
content doesn’t perhaps break
any massively new ground (there
are plenty of libraries that provide
atmospheric, vowel-based, melodic
vocals), they do sound fantastic.
The phrases themselves are sung
at slower tempos and many are
several bars long so you can easily
weave them together to create your
own variations.
The second category provides
Interval presets. The concept
is simple, with a single, simple
phrase that moves legato-style
between two pitches. Again, you
get multiple key options and, within
each preset, there are a range
of intervals, transitioning both up
and down in pitch and mapped
across the keys within each preset.
142 January 2023 / www.soundonsound.com
They also sound great and can, for example,
be used to complement the more extended
melodic phrases.
Category 3 provides a huge collection of
Rhythmic Pedals. These are organised into
a number of sub-categories but, in each case,
you essentially get a selection of rhythmic vocal
chants, each of which stays on a specific root
note. Whether used individually, or stacked by
playing multiple notes, these can add some
interesting rhythmic elements to a Vocalise
3-based composition. And, as Category 4 then
provides you with an impressive collection of
vocal-based Scrapes And Drones, you can also
add sustained textures to your project. The
final category rounds out the collection with
the (compulsory?) selection of vowel presets
and includes plenty of mod-wheel-based vowel
shifting options.
The Gravity engine offers plenty of additional
processing features to spice things up further
(the Motion engine is particularly impressive).
Media composers might want to blend in a few
additional sound elements, but Vocalise 3
can deliver score-ready textures, rhythms and
vocal top-lines to be the star of the cue. As
commented above, while there are other vocal
libraries out there that cover similar territory, if
you are a composer looking to add the sorts of
evocative/ethereal/atmospheric vocal textures
(as opposed to the more intense/bombastic vocal
elements) you might hear in parts of film scores
from Gladiator to Dune (the 2021 version), then
Vocalise 3 would make a worthy addition to your
vocal library options. John Walden
£104
www.heavyocity.com
Vegas Audio
Drumslot
Plug-in Instrument
Fledgling company
Vegas Audio’s new
virtual instrument
tempts you to take
a chance on a roll
of the fruit machine,
through their uniquely styled drum plug-in called
Drumslot, offering a vast number of drum sounds
geared toward electronic production.
The Drumslot concept begins by placing
random drum-voice selection right at its heart.
Much like a fruit machine, three ‘reels’ provide
a sample each. By hitting the Spin button, you
generate a random selection of samples, with
subsequent options to hold or nudge a sample to
the next sample in line. It’s a strangely addictive
concept, which can generate some impressive
sounds by chance, in part thanks to the 600 or
so samples which form the included library. If
that isn’t enough for you, there’s also capacity to
add user samples, although the diverse nature of
the included sounds will take you quite a while
to exhaust, not least thanks to Drumslot’s two
modes of operation.
The first, and most obvious mode, is a layered
multi-mode, where each sample is triggered at
the same time. Each layer is equipped with its
own ADSR envelope control, as well as see-saw
filter and volume gain. It is also possible to invert
the polarity of a sample, reverse its playback,
or deactivate the layer altogether, should you
feel that you only want to work with one or two
samples at a time. When selecting an instrument
category, spinning the reels (samples) will collate
categories together, so the resultant patch
would consist of, say, three snares or three kicks,
maintaining coherence but delivering some
surprisingly convincing randomised results. You
can however manually combine different drum
elements, combining a hi-hat with a kick for
example, by merely using the Hold function and
clicking through the categories and samples. The
included content clearly has a leaning toward
electronic music production, being impressive
and generated from stalwarts such as the 808,
as well as substantial floor-shaking kicks from
modular systems. The three sample layers
are summed to a master section, whereupon
a further set of mirrored editing controls await.
The second operational mode
is Cut, which allows the splicing
of the three samples together,
to create a beginning, middle
and end. The resulting edit can
sometimes contain glitches; there
is no crossfading here, but it’s
possible to dial back any click
or glitch noise with the ADSR
envelopes. There is an argument
that those noise colours become
part of the plug-in’s overall charm,
though, as a form of happy accident.
This is a rather surprisingly cool little plug-in.
I can’t lie; I was slightly dubious about the
concept, upon initial explanation, but it does
sound huge and very usable, and could quickly
find itself in numerous production environments,
with a heavy leaning toward the
commercial. Dave Gale
$54.99
www.vegas-audio.com
Fluffy Audio
Timedrops
Kontakt Instrument
++++
Timedrops is a Kontakt (full
version required) instrument that
uses granular processing to take
sections of a sampled waveform
and then turn it into a new sound
that can be played from a keyboard.
Its library samples offer up anything
from a rattling bell or piano note to
a long section of classical music,
and by changing where in the
sample the granular extracts are
taken from, and by adjusting the
granular parameters, the same
sample can yield a wide range of
usable sounds. Timedrops comes
with a library of preset instruments,
though users can also import their
own samples.
The controls are fairly
straightforward with a waveform
view at the top of the window. This
is equipped with A and B sliders
where the A slider adjusts the point
at which grains are generated. The
pointer position may be adjusted
manually or varied via an LFO, or it
can be assigned to a custom CC.
The B slider comes into play a little
later when we get onto Modes.
The grains are shown as moving
vertical lines imposed over the
waveform display.
Granular synthesis can get
quite complex so our Fluffy friends
have provided a set of Modes to
simplify things. These dictate where
the grain generation starts, at or
between the A and B markers with
some useful morphing and fading
options. In Key Span mode, the
pitch is fixed to the chosen root key,
though every key in the range has
a different starting point allowing
different sounds to be generated
from the same sample. Velo Span uses
velocity to set the start point, while Key
Slide forces the starting point to slide when
playing legato. Pitchscan mode is useful for
interesting atonal results.
The Modulation tab is where you set
up parameter control using the LFO (sine,
saw, triangle or pulse with designation in
a drop-down menu) and/or MIDI CC. Where
a parameter is being modulated, a moving
yellow line shows the rate and range of
modulation. Key range can also be adjusted
in this bar along with the maximum voice
number and an auto CPU limit.
Presets are organised into categories:
Atmos, Music, Noise, Phrases, Shorts,
Sustains, Synth and User. Many of the
sounds have an obviously granular
signature, often lending them a diffuse,
evolving character with a slightly unearthly
timbre. However it is also possible to coax
very synth-like sounds out of Timedrops,
some with a very Blade Runner kind of
vibe. The ability to load in a long sample,
such as a slice of classical music, and then
to dive in and manipulate small sections
is also impressive and is a good way to
create shifting backgrounds and drones.
I was a little disappointed that the designers
haven’t explored Kontakt’s drag-and-drop
for adding your own samples, but because
of the different ways in which you can
process the factory library, there should be
enough there to keep you busy for a long
time. If you need something to help you
cook up dynamic drones, creepy cues or
surreal synths, Timedrops has a lot to offer.
Paul White
€69
www.fluffyaudio.com
Sampleson
Alienize
Plug-in Instrument
++++
While retro sci-fi has been a thing since...
well, since it wasn’t actually retro, the
blockbuster success of Stranger Things has
certainly bought it right back into the film/
TV scoring mainstream. Sampleson’s latest release,
Alienize, is pitched at this genre and they describe
it as a cinematic sci-fi instrument with mouse
gesture-based modulation (although you also get the
option for MIDI CC control of parameters).
An Alienize preset is built from three sound
elements and these are drawn from 41 underlying
‘sound modules’. Under the hood, these modules
utilise a combination of sample playback and
granular, subtractive and wavetable synthesis.
However, the vast majority of this is kept well
away from the user as the streamlined UI (with its
alien-themed graphic) contains a fairly minimal
control set. The key elements of this are the preset
browser (where the 120 presets are categorised via
some suitably useful tags) and the three sound slots,
each of which can draw from a different subset —
Pedals, Generators or Textures — of the 41 sound
modules. Module names such as Retro Bass Arp,
Simple Flute, Arp Violas, Alien Speech, Metal, Hit,
String and Tuned Bass Drum perhaps give a sense
of what these sonic building blocks provide, and
they span both playable (melodic) elements, pad/
sustained sounds and sound effect elements. And,
yes, there is a Theremin sound module — with the
pitch controlled by mouse motion — if you want to go
fully ’50s sci-fi B-movie.
The upper half of the display is where you can use
left-right motion of your mouse to modulate some
pre-configured elements of the sound, whether ‘live’
or by recording the mouse motion as automation
data alongside your MIDI note data (this worked
a treat in Cubase). This real-time mouse-based
sound modulation is very nicely implemented and,
as a means of easily sync’ing crescendo-style
sound design to any visuals, it’s very intuitive.
A good number of the modules offer further settings
accessed via a small pencil icon. For example, in
the Retro Bass Arp module, you get a mini step
sequencer, plus speed (sync’ed to your host tempo)
and a filter cutoff control. As appropriate, these offer
MIDI CC control (and a MIDI Learn function) allowing
further sonic modulation.
While there are plenty of sound possibilities to
be had from combining 41 sound modules, sonically,
Alienize is perhaps not the most diverse sound
palette you might encounter. That said, whether you
want to create a complete musical cue, or layer it
with other sounds, it has plenty of suitably retro sci-fi
character. A PDF user manual would be a useful
addition to the package but, that aside, at a very
accessible price, I suspect busy media composers
will happily take a punt, even if only for occasional
use. Quirky, fun to use, and sonically right on the
money for the retro sci-fi target. John Walden
$59
www.sampleson.com
Audio examples of this month’s libraries are
available at www.soundonsound.com.
www.soundonsound.com / January 2023 143
Q Why are my recordings using
an iRig and iPhone so quiet?
I’m planning to get a Cloudlifter to plug
into an iRig Pre HD, which I connect to
an iPhone for recording my acoustic
songs with a Shure SM7B mic. Right now,
I have to set the iRig preamp almost at its
maximum gain to get a clean sound, but
then it clips very easily on loud songs.
Someone said that with the Cloudlifter
you only need the iRig Pre gain to be set
about halfway, which is plenty. I looked
it up and the Cloudlifter adds 25dB of
gain or “clean gain” (though I’m not sure
what that means). And the iRig Pre can
add 40dB of gain (I guess that’s when
it’s maxed). So that would mean the iRig
would have to be almost all the way up
to get to 60, right? But everywhere it
says SM7B needs 60dB of clean gain to
run at its best. How come everyone says
that setup is perfectly fine? Should I get
a Rane MS1 or something so I can get
to 60dB without having to go up much
on the iRig? Also, I was wondering if
a Focusrite interface is the same thing as
this iRig, or if it is better somehow.
SOS Forum post
SOS Technical Editor Hugh Robjohns
First, there isn’t any need here for
a Cloudlifter or other preamp: the SM7B
is a fine mic, and the iRig is capable of
making clean recordings. The second
thing to point out is that context is
everything when you read advice on the
Internet! In a professional studio setup,
the A-D converters typically need a huge
signal level before clipping, and to get
the signal from an SM7B to peak around
-10dBFS at the converter you really will
need at least 60dB of gain, and maybe
more, depending on how loud the source
is. However, your recording system is very
different, being based around an iPhone
and iRig, which has a much more sensitive
A-D converter. Consequently, it needs
a much lower input level to reach -10dBFS
and it’s clear from what you say that the
available 40dB in the iRig is more than
enough in that context.
Note that clipping occurs when the
signal level is too high for the electronics
to handle. This might be because the
signal coming into the preamp is too big,
or it might be because the preamp is
boosting the signal too much (the gain is
too high) and it is either overloading itself,
144 January 2023 / www.soundonsound.com
The Shure SM7B
may have a reputation for
needing lots of gain, but if
you’re already clipping your
converters then a lack of
gain is not the problem!
or the subsequent
element in the signal
path (such as the A-D
converter in the iRig, which
turns your recording into the 1s
and 0s needed for digital recording).
If there’s no clipping with a low gain
setting, the input signal is not too high.
If it clips when you turn the gain up, the
clipping is being caused by setting the
preamp gain too high — you’re sending
too much signal into the A-D converter.
If you have enough gain on hand in the
preamp to drive the signal into clipping —
and you clearly do — then you don’t need
to boost the input signal any further with
a Cloudlifter (or any similar device). All that
will happen if you add a Cloudlifter is that
you’ll get clipping when you turn the iRig
preamp gain to 5 (for example) instead
of 10. And you will turn it to 5 because
the signal still won’t be loud enough in
your headphones!
You say that “the thing is it will not clip
if I keep the gain lower around 8 or 7” so
you already know how to make a good,
clean recording: just set the gain to 7 or
8 (there’s no need to spend money on a
Cloudlifter). The real problems that remain
are that: (1) the mic signal doesn’t sound
loud enough to you through your local
monitoring while recording at a sensible
level; and (2) neither does it sound as loud
as commercial recordings to your friends
when they play the files. It’s important to
understand that these are separate issues
which need to be addressed separately.
To improve your local monitoring
volume, you either need more sensitive
headphones, or an external headphone
amplifier of some kind. That way, you’ll
be able to hear yourself at your preferred
headphone level when the iRig preamp
gain is set to 7 or 8, and the mic signal
is peaking at -10dBFS or so in your
clean recording.
To make the recording sound
like other commercial files, for your
audience, you need to process the raw
recording like commercial recording
companies do. Basically, that means
it needs to be compressed, and
possibly limited, and you do that
using Digital Audio Workstation (DAW)
software of some kind. Most people
do that in a computer, but there are
phone apps that can help too.
GarageBand or Cubasis might
be good places to start, for
example, though there
are plenty of other
options for the iPhone.
Start by setting the
compressor ratio at 2:1
or 3:1, with a medium
attack, 300ms (or auto)
release, and adjust the
threshold until you’re seeing up to 4-6 dB
of compression on the meter at the louder
points in your recording. This will make
the loudest bits quieter, and you can then
adjust the makeup gain to raise the overall
level to just below 0dBFS. A limiter is just
a very assertive compressor, and after
initial compression one of these can help
you bring the average level up a bit higher
— though if you push it too far you’ll start
to hear side-effects.
Q How should I use other
pedals with the Caveman
Audio AP1?
I enjoyed the review of Caveman’s AP1
acoustic pedal in the October issue [www.
soundonsound.com/reviews/caveman-
audio-ap1]. I had the same question:
whether it would be possible to have
a send/return installed as an upgrade
and a surplus price. I tried to mail two
times to the company (Caveman) but did
not get any answer. I intend to use the
Origin Effects Cali 76 pedal and an LR
Baggs Acoustic EQ pedal. Would placing
those two pedals before the Caveman’s
AP1 work, without compromising the
high-quality signal? Is there a better
configuration? Thanks.
Leo Janssens via email

SOS Reviewer Bob Thomas
Hi Leo. I’m glad that you enjoyed my
review of the Caveman AP1 — many
thanks for your kind comment! I’ve
passed your feedback on to the
manufacturer but I can offer a little advice
too, since I often use a similar setup to
yours. In my case I have an Effectrode
PC-2A compressor running into a 1980s
vintage TC Parametric EQ pedal. To
get the best from of the Caveman AP1
preamp, you would have to put it before
the compressor and EQ. If, instead, you
put the Cali 76 and the Acoustic EQ
in front of the AP1, then your pickup is
going to ‘see’ the Cali 76’s input stage
as its preamp, and that isn’t going to
sound as good as putting them after the
AP1 preamp.
Personally, I would ‘drive’ the Cali 76
(with the dry signal set at zero) from the
AP1’s Amp Out, then run the output of
While there’s no send/return loop on the Caveman
AP1, its versatile outputs mean it’s still possible to
make good use of other pedals alongside it.
the Acoustic EQ and the AP1’s DI output
into two separate channels of either
a mixing console or into a little submixer
on your pedalboard. This would allow
you to run a parallel effects setup with
the AP1’s DI out as the dry signal and
the Cali 76/Acoustic EQ as the wet
signal. I hope that helps!

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 TECHNIQUE
Reaper
With a bit of fine-tuning, Reaper can (a domestic room or office) to
a separate file using Zencastr.
help you edit long podcasts at pace! I’ll start by normalising
each person’s recording to
M AT T H O U G H TO N
the same Integrated LUFS,
both to put them in the same
’ve been doing a fair bit of podcast production recently and ballpark level-wise and ensure
I while editing tasks are usually pretty simple, most need to be
performed many dozens of times for each speaker. With two
presenters and a guest speaking for 45 minutes or so, this can
they’re hitting the thresholds
in my default processing chain
roughly where they need to
add up to a dizzying number of key presses, mouse clicks and right from the off.
drags, and previewing time. So even the tiniest efficiency tweaks Some people prefer not
can save you a lot of time on each production. to start processing until the
In this article, I’ll take you through some of the approaches editing work is complete but
I’ve adopted and Custom Actions I’ve created which help me I prefer to work the other way
to edit dialogue-based podcasts quickly. Combined with some around, since the processing
judicious processing decisions, they’ve reduced the time I spend often emphasises certain
on post-production dramatically. But before I dive in, let me things I wished I’d attended
offer credit where it’s due: Jon Tidey’s The Reaper Blog is a rich to with edits but didn’t
resource of video tutorials for Reaper, and I’ve drawn much notice at the time! As for the
inspiration for my approach from them; I strongly recommend processing itself, I’ll generally
checking out his https://reaperblog.net site. use iZotope RX10 to tackle
some of the inevitable noise
Overall Approach issues, and lean on Accentize
For one of my regular projects, a typical podcast must last no DeRoom Pro to tame the
longer than 45 minutes. The raw recordings are often circa room sound (reviews of both
50 minutes, so edits are needed to bring the overall time can be found elsewhere in
down, as well as to attend to problems and improve the flow this issue). The processing
of the conversation. Each episode has two or sometimes chain in Reaper also includes
three speakers, each of whom records in a different location some basic stuff like EQ and
146 January 2023 / www.soundonsound.com
compression, a de-esser
(two, actually!) and Oeksound
Soothe, and I’ve created
presets for each regular
speaker in each frequently
used location. There are
de-breather plug-ins available,
but I prefer to tackle breaths
with edits, as described later.
When editing, I’ll look to
remove or truncate silences,
tame breath sounds, cut out
or filter bumps, bangs, pen
clicks and the like, delete
unnecessary repetitions,
reduce the number of filler
words and so forth. A dual
aim all the while is to respect
each person’s natural pace
and delivery style, and
make the whole thing seem
like a natural conversation
to the listener.
Most work will be carried
out in the ‘Ripple editing all
tracks’ mode: when engaged,
if you cut or move a section
on one track, everything to
the right on all tracks moves
left accordingly. You can set
 Increasing the project play rate can be a great way to speed up your
editing — just don’t forget to reset it to 1.0 when you’re done!

this using the grid icon, top left of the project page, but I prefer to Inverting the right-click-drag behaviour in the Arrange view can make edit
switch to/from this mode using steps in my Custom Actions, since selections really easy to perform.
this prevents me carelessly leaving it on or off (and so prevents
me making mistakes off screen when I’m zoomed in).
Some detailed individual edits are done in Reaper, while
others require RX10, which I’ve set up in Reaper as my primary
External Audio Editor (you can have two), with a keyboard
shortcut to open a copy of the selected clip in RX10. That way,
I hit one button, do my RX work, then save over the original
(duplicate) file and it’s job done; the original remains as an
alternative (but muted) take.

Template Basics
I’ve created a Reaper project template for regular jobs, with
a separate folder track for each person, onto which I drag and
drop the recordings. Automatically routed to this is a child track
immediately below, which I use for breath/mouth-noise control.
A two-fingered click-swipe of my MacBook Pro’s trackpad selects
a region, and then I trigger a Custom Action with a mouse
modifier to cut each breath out to the child track. Not only can I My Custom Action for moving breaths to an adjacent track.
then process the breaths separately (filtering can sometimes be
useful) but I can also adjust the overall breath level whenever behaviour for the Arrange This turns the breath into
I want using the track fader. Since this track’s output is routed view to ‘Select items and time’ a separate clip, selects it, and
directly to its parent, the same processing chain is applied to the and that command’s default deselects other parts. Then
breaths as to the rest of the part. I’ll detail this Action shortly. Cmd modifier to ‘Marquee three actions paste the breath
I also have a dedicated track for the music, including select items’. Why? Because clip, at the correct point on
automated fade-outs/ins, as well as a reverb send, all set up and I need to do this operation the child track: (4) Go to start
ready to go. All the voices are sent there, and the return track’s more frequently than the of time selection, (5) Track:
fader is pulled down so that, if I feel I need a dash of reverb to tie default and it cuts out a key Go to next track, and (6) Item:
them together, all I need do is bring up that fader to taste. press. Now, I can perform Paste Items/tracks.
One of the biggest potential time-savers is one I picked up a two-fingered/right-click and This completes the main
from Mr Tidey. In the default theme, adjacent to the transport drag to select a breath (or any ‘move breath’ job, but I’ve
bar Reaper has a global play rate control, whose default value is other section I wish to edit). added a few more stages.
1.0 (meaning playback happens at normal speed) but which can With a selection made, my First, (7) Go to previous track,
safely be increased a touch while you’re making edits. I’ve found Custom Action is as follows: ensures I’m back where
I start to lose sight (sound?) of some editing artefacts if I go much (1) Set ripple editing off, I started should I want to
higher than 1.25x, and in any case you need to allow sufficient a safety move that ensures enact different actions. (8)
time to make selections without having to rush, but even that edits to trim or move the Xenakios/SWS: Move cursor
setting can save a huge amount of time when working with long breath don’t interfere with left configured seconds,
files. (Just don’t forget to double-click and set it back to the other tracks. (2) Track: Select and (9) Play start playback
default before your final previews and rendering!) track under mouse. This from slightly earlier, so that
allows me to create a precise I can preview the edit and
Custom Action 1: Breath Control selection by dragging across continue on. (The free SWS
Let’s look at how my approach to ‘de-breathing’ works in the waveform on the track extensions must be installed
practice, which all starts with an efficient way to select the I wish to edit. (3) Item: Split for this to work: https://www.
breath. In Mouse Modifiers, I changed the default right-click-drag items at time selection. sws-extension.org).

www.soundonsound.com / January 2023 147

 TECHNIQUE
SPEEDY PODCAST EDITING
I assigned a shortcut to this Custom
Action, so all I need do is swipe right
and hit Opt+Ctrl+B: the breath is moved
and the track plays from just before the
breath. It might sound a painstaking
approach compared with a de-breath
plug-in, but it offers so much more control
and really takes no time at all as you go
through; you soon learn to recognise
a breath ‘blob’ in the waveform.
One thing I’ve found is that if I’m
struggling to hit my 45 minute maximum
time, I can often quickly run through the
breaths in the ‘Ripple editing all tracks’
mode, and slice a bit out of each breath
to make up a good chunk of time. And
to do that, and much more,I use another
Custom Action...
Custom Action 2:
Delete, Slip & Crossfade
This one is designed to simply remove
an unwanted section on all tracks, and
apply crossfades that ensure smooth
transitions. It’s easy enough to do this
sort of thing manually in ‘Ripple editing
all tracks’ mode, but my Custom Action
reduces the amount of mousing around
and keystrokes. As before, to use this
one you first need to right-click drag to
make a time selection. Then, we have
10 actions. (1) Item: Select all items in
current time selection. This selects any
items in the time selection, regardless of
which track they’re on. (2) Go to start of
time selection. (3) Set ripple editing on
all tracks. I find it’s better to embed the
desired edit mode in these actions, as
it saves you having to look and switch
things manually; while ripple editing is
a Godsend, you don’t want to make such
edits inadvertently or, sooner or later,
you’ll find yourself wasting time unpicking
all sorts of problems. (4) Options:
Enable auto crossfades, which ensures
148 January 2023 / www.soundonsound.com
Reaper
The SWS extension
settings window, where
you can specify how
far the Custom Actions
move the playhead left to
auto-preview your edits.
a smooth-sounding
result. (5) Time
Selection: Remove
contents of time
selection (moving
later items).
This deletes the
unwanted section,
and moves all that
follows forward,
while the ripple
editing mode keeps
the other tracks
in sync. (6) Grow
left edge of items.
This is necessary because the cut a Mouse Modifier — I can simply do a
method doesn’t result in the same sort two-finger swipe, then Ctrl-click, and the
of overlap and crossfade as the regular unwanted section is gone, with all tracks
Split command does — it expands the moved accordingly (whether there’s
currently selected item to overlap the media at that point on those tracks or
right of the trimmed section, which not), and Reaper resumes playback from
automatically results in a crossfade. a second or so before.
That’s the main edit done, but there’s Incidentally, if you were wondering
some tidying up to do. First, comes (7) how to specify where these Actions
Item: Unselect (clear selection of) all move the playhead to preview your
items. This ensures we can’t later apply edits, this is done in the SWS settings
further actions to this section by mistake. dialogue, under Extensions / Command
Then (8) and (9) form the same preview parameters... You can also call this
mechanism as in the de-breath Custom up with an Action and assign a key
Action we looked at earlier. Finally command if you wish to be able to
comes (10) Set ripple editing off, which change the amount quickly.
ensures we don’t make ripple edits by
accident — something that’s all too easy
More?
to do when other parts are off screen. There are plenty of other things
This particular Custom Action is you might want to think about
the one I use most frequently in my (semi-)automating, such as creating
podcast projects. With a little practice, a command that splits an area and
it becomes almost second nature to applies a mute, to leave things in place
zoom in on waveforms, scanning ahead and visible as a reference. Or an Action
of what you’re that splits out a section and applies,
hearing, so that the say, a 5dB boost or cut. You might also
moment you hear find that you prefer to use splits at the
something that cursor position than the right-click drags
needs zapping, I find so convenient on the MacBook.
you can swipe to But hopefully my walking through these
select and run the examples will help you think your way
Action. Given how through how you might approach that.
frequently I use Whatever you do, one thing’s for
this one, I decided sure: while these four tweaks (a simple
to assign it to template, two custom actions, and
a small play rate increase) might not
Another Custom
constitute the most sophisticated setup
Action following similar
principles, this time in the world, they save me an enormous
to remove unwanted amount of time when editing 45 minutes
sections of audio. of multitrack dialogue!  
 The Console That
Changed Mixing
V IDEO DOCUMEN TARY ORIGINAL S

IN ASSOCIATION WITH

After Solid State Logic introduced the E Series console in 1979, music would never be the same again. In
this month’s SOS video feature, we head to Echotown Studios in Dorset with SSL expert Karen Down for
a hands-on demo of the revolutionary automation and recall technology that took the 1980s by storm.

www.youtube.com/soundonsoundvideo
 TECHNIQUE
Studio One
Your Studio One
productions can now
have integrated lyrics.
ROBIN VINCENT

e have an interesting

W integrated lyric engine with

the shiny new version 6 of
Studio One. It’s peppered all over the
place, finding itself in its own track, in
a lane under other tracks, attached to
notes in a score and running karaoke
on the Show Page. As I mentioned in
my review of Studio One 6 last month,
it’s not a unique feature, but the way it
threads itself about the DAW is really
well thought out. So, for this workshop,
we’re going to look at how you can add
lyrics to your songs in Studio One.
Global Lyric Track
The best place to start is with the
Global Lyric track. This resides at the
top of the arrange window where you
also find the Chord track, Markers,
Tempo track and so on. Within this
narrow band of space, you can
double-click, whereupon a box appears
into which you can type a line of text.
All that matters is the starting place,
which will probably be the start of the
bar where you’d start singing. Pressing
Return closes the box and opens a new
one for a new line at the next bar. Tab
does the same but without starting
a new line in the Lyric Display.
The words you’ve entered are held
in a box which you can move about
on the timeline. You can add boxes for
every line or every word if you wish,
although that can get a bit unwieldy.
The best practice here is to use it to
roughly sketch the words alongside
the arrangement. Your level of
zoom will have an impact here,
as it does on other tracks, where
word boxes will overlay each
other. There’s no problem with
that other than not being able to
see all the words when zoomed
out. There’s a certain amount of
connection between the boxes. If
you move one down the timeline
past another box it will move
along with it. So while you can
Lyrics can be typed directly into the Global track.
change the position of a line, you can’t
actually rearrange their order on the
Lyric track. If you try to bring a line in
from a later point, Studio One won’t let
you drop it in front of another lyric.
One cool lyric-arranging tip is that
you can position the song pointer
where you want the lyric to start and hit
Alt+Return to send the selected lyric to
that spot. It then auto-selects the next
line. So you can play the song through
and sit there hitting Alt+Return to tidy up
the placement of every lyric.
Lastly, you can drag lyrics in from the
browser in the form of text or MIDI files.
They appear as a string of lyric blocks,
separated by bars, one for each line in
the text file. You can drop them in and
then use the Alt+Return function to put
everything where it needs to be.
 An important thing to realise is that
the lyrics in the Global track are not
bound to anything. So, if you move audio
and MIDI events around, the lyrics will
not automatically follow them. You also
can’t select audio/MIDI events and lyric
events together and move them that
way. However, this is where the Arranger
track comes to the rescue.
The Arranger track is extraordinarily
useful for making huge, sweeping
changes to your songs. Once you’ve
defined the sections of your song (which
is simply done by placing a box and
calling it something like ‘verse’, ‘chorus’
or ‘bridge’), you can rearrange these
sections, and it takes everything with it,
including the lyrics. If you’ve never used
it before, now is a good opportunity to
start because it’s incredibly efficient. It’s
also the only way to include the lyrics in
your rearranging.
Lyric Display
Hit the big ‘L’ on the Global Lyric track,
and the Lyric Display will pop up. This
displays your lyrics line by line and
situates them in the context of the
timeline. As the song plays, each line is
highlighted so you always know exactly
where you are. It’s like lyrics are being
used as markers. And the best thing
is that if you click on a line, your song
will jump straight to that point. This is
perfect for when a singer wants to go
from a certain line, and it’s a terrific way
to navigate your song.
You can edit the lyrics here and you
may find that some lines you typed
into the Lyric track didn’t go to a new
line — that’s an easy adjustment. Hit the
pencil button, and the lyrics are shown
in the Lyric Display with a timeline. You
can’t move or rearrange text directly,
but you can copy, cut and paste to get
everything in order. Under the spanner
icon, you can set text alignment, offset

Alternatively, lyrics can be dragged

and dropped from the Studio One browser
in the form of text or MIDI files.

150 January 2023 / www.soundonsound.com

the highlight so it runs ahead of the actual song
position, and increase the font size up to 60 point.
The editing in the Lyric Display is not particularly
sophisticated, but what it’s doing is enormously useful.

Show Page
Before we get into the Lyric lanes within editors and
instrument tracks, we should visit the Show Page.
Lyrics here refer to the Global Lyric track and are
displayed in the same way as on the Song page. You can also add lyrics to the
Any additional words in Lyric lanes will be ignored. a word up into syllables, use the ‘-’ key. Score view, using the same keyboard
Otherwise, in the Show Page, the Global Lyric track If you use space for syllables they will shortcuts as in the piano-roll editor.
and Lyric Display, lyric editing is exactly the same. appear as gaps in the Lyrics Display,
You can place the Lyric Display window on another whereas the ‘-’ signs will be ignored, and they are harder to identify in
screen for your vocalist, but at the moment, lyrics do you’ll see a complete word. the lyrics lane.
not appear on the Perform page. Hopefully, that’s You can copy and paste, and even Once you’ve written your
something that will come along in a future update. drag in lyrics from a text file, but it’s a little lyrics into a lane, you can
weird. If you copy a phrase and then display them in the Global
Lyric Lanes double-click to open the first lyric field track. In the header of the
Another, more versatile, option for entering lyrics is and paste it, it will just run from there on track is a drop-down menu
in the Lyric lanes in the piano-roll editor. This lets you that one note. To paste it across multiple where you can select from
get down to the juicy details of attaching words and notes, which is what you want, you a list of all the instrument
syllables to actual notes, rather than floating them mustn’t have the field selected. Instead, tracks in your song. If they
about in the Global track. have the editor and lyric lane open and in have any lyrics in lanes, they
You can add lyrics to any instrument track, but as focus, and then hit Ctrl+V to paste — it’ll will be displayed. Now, these
you’re attaching them to notes, it makes most sense work! This is one way of pulling the lyrics lyrics will move with your MIDI
to use a melody track. It occurs to me that this is not from the Global track into a lane. There’s track because they are part of
something I’ve ever created, because my melodies no special relationship or tool; you just that track’s note information.
are most often sung. In which case, the Global Lyric double-click the lyric in the Global track No need for the Arranger
track seems like a much better idea. But assuming and copy the text. track in this instance.
you have mapped out a melody, let’s dive in. Dragging in a text file doesn’t seem
You open the Lyrics lane the same way you would to work quite right. It puts the words very
Score
the velocity lane or automation lane in the editor: nicely into the lane, but ignores the notes The Lyric lanes all translate
with the tiny mountain range icon beneath the piano completely. So typing them in is probably beautifully to the Score editor.
keyboard. In there, you’ll find the Lyrics tabs. Under the best approach to get everything There’s a new symbol called
each note is a little space that’s patiently expecting exactly where you want it. What’s nice ‘Lyr’ which you can use to
your poetry. about all this is that the Lyric Display enter text under notes if
Double-click in that first space, and you can begin window will now highlight each word in you would rather add your
to type your lyrics. If you press Space it will jump to turn rather than each line. You can also lyrics in the Score view. It’s
the next note for the next word. If you want to break put in line breaks and corrections here, as all the same action and key
commands as in the piano-roll
editor. You would expect to
be able to move the words
around on the page, but you
are restricted to deleting,
moving to another note and
re-entering.
Overall it feels like the
Global track acts as a lyrical
notepad for ideas and
overviews, whereas the Lyric
lanes really get down to the
nitty gritty of matching your
lyrics to the melodies in your
song. So, get friendly with
the Arranger track, and when
exporting to the Show page,
remember to select the right
track for lyrics in the Global
track. Now, go and write
The Lyrics lane lets you assign specific words and syllables to MIDI notes. some songs.  

www.soundonsound.com / January 2023 151

 TECHNIQUE
Reason
SIMON SHERBOURNE Join us as we delve into the hidden mysteries
e’ve covered the mechanics of Reason’s of the Kong drum machine.
W Kong drum machine in the dim and distant
past, but I’ve recently been exploring
some of the interesting sound creation features of its
integrated synth and sample modules, and wanted
to share.

Boss Drum
Kong is Reason’s primary drum instrument, coexisting
happily with the classic Redrum drum machine. Kong
is much like an MPC or Maschine drum program,
with a 4x4 pad grid triggering independent channels,
each with its own synth or sampler-based module
and effects chain. Redrum, on the other hand, is
modelled after classic beatboxes with an internal
pattern-based sequencer.
It’s quite easy to miss the complexity and versatility
of Kong’s design when looking at its primary (and
slightly drab) view in the Rack. There’s a generic
settings panel for the selected pad over to the left,
and some complicated looking options over on the
right. You’d be forgiven for assuming Kong is a simple
sample-based drum kit. However, if you click the
Show Drum & FX button at the bottom to reveal the
inner workings of each pad you’ll see there’s a lot
more going on.
Each pad can be powered by one of nine different
sound modules, some sample-based and some
synths. Each channel also has two insert FX slots
that can be populated from the pool of nine effects
modules. Two additional slots house Bus and Master
FX modules. Seasoned Reasonistas will probably be
familiar with this not-so-secret part of Kong, but did
you know there are two additional mini-synth modules
that can be used in any pad, or that you can set up
trigger variations for many of the modules?

Inner Voices
Screen 1: Kong’s internal
If you drop a sample onto a Kong pad it automatically there are four synths that model classic
modular instrument structure
loads into a mini-sampler instrument called NN-Nano. analogue drum-machine voices, offering has many hidden gems that
But any pad can instead run a drum-synth module. kicks, snares, hats and toms. you can use to build authentic
These are chosen from the Drum Module pop-up We could probably dive into a rabbit analogue-style drum machines.
menu at the top left of Kong’s foldout instrument hole going over the details of these
panel (Screen 1). In fact the modular setup of each fantastic sub-modules, but I promised Tone is a sine-wave
drum channel and its FX slots can be saved as ‘secret’ features and I’m going to assume oscillator with a wide pitch
a preset file, called a .drum file in Reason. These can you’ve spent some time playing with range and a Shaper control
be loaded from the file browser in the pad settings these already. that dials in more harmonics
area, or dragged onto a pad. Reason’s Factory (and I think is probably a wave
Sounds Refill has plenty of examples in the Kong
Supporting Roles folder). There’s a simple
Sounds and Samples folder. At the top of the FX slot selectors you’ll Attack/Decay envelope which
At the top of the drum module list you’ll see the find two modules separated into their own appropriately has a percussive
two sample-based engines: NN-Nano, and Nurse section. These aren’t FX at all: they are decay slope that can go down
Rex Loop Player. Nurse Rex can load and trigger REX in fact sound-generating synth modules to very short times. The other
loops and slices: we covered the cool things you can called the Support Generators, as they are section is for pitch mod; apply
do with this, like quickly juggling and resequencing mainly meant to supplement the primary a pitch envelope with instant
breaks, back in the June 2017 issue of SOS. The rest drum modules. You can however use them attack and variable decay as
of the list comprises drum synths. There are three without a drum module. They remind me set by the Bend Dec knob.
physical-modelling engines that dynamically recreate of the percussion/drum-synth voices found The other knob, Bend, sets the
the sound of bass drums, snares and toms. Then in classic old Reaktor groovebox patches. depth of this pitch sweep.

152 January 2023 / www.soundonsound.com

 With settings in the vicinity of those in Screen 1 you
can get some lovely fat classic kicks, with the satisfaction
that you made them yourself. Layering the tight, punchy
sound of this synth with the softer dusty tones from the
analogue bass-drum synth is extra effective. I like to add
a little of the Overdrive FX module with the resonance
turned right down. If you place this in the Bus FX slot
and dial up the bus send from the pad parameters you
get a parallel processing structure. This keeps the clean,
clicky punch of the original with nice warm saturation.
If you run it in the usual way as an insert it can lose its
impact and sound tubby.
There’s loads more the Tone module can do. With
some knob twiddling you can reach other classic
synth drum sounds, blocks, taps, tweets and zaps. It’s
essentially what would typically be the ‘body’ part of
an analogue drum voice, which would be blended with
a noise source. You can see this approach used in the
primary analogue drum modules.

Make Noise
The Noise module has a similar layout to Tone, but uses
a white noise source as its starting point. The Pitch Screen 2: As well as the drum modules, there are two synth voices hidden in the FX section.
control sounds like it is controlling a band-pass filter that
this noise passes through. In place of the bend controls
you have Sweep and Reso, which apply an envelope and
resonance to the filter. The Click knob controls the level
of an extra transient sound that provides some attack.
Noise can be used to create a range of classic synth
snare sounds, and can be deployed in tandem with the
Tone generator as in Screen 2. Of course you can add it
to any of the other modules to add some of that Roland
snap. (The Rattler effect is also good for this). Again,
this module is quite versatile and responds to velocity,
getting brighter for louder notes. With a little application
of short delay you can even get close to a crude synth
clap or fingersnap.

Hit Types
The other little known trick up Kong’s sleeve is the Hit
Type system. This feature is used in conjunction with
Drum Assignments, allowing you to trigger a single
sound module differently from different pads. Back in our
2017 Kong sample chopping tutorial we looked at how
this can be used by the Nurse Rex module to configure
Screen 3: Using Hit Types you can set up
different pads to trigger specific sections or hits within to a module that supports a single drum module to respond differently to
a sliced loop. Other modules use Hit Types to vary their Hit Types, this section will triggers from multiple pads.
sounds dynamically. show different variations.
Analogue drum machines old and new often share For the Synth Hi-Hat module with the Physical Snare Drum,
synth circuits between more than one sound. A typical this is Closed, Semi-Closed, where you can hit different parts of
example would be the hi-hat, where the closed and Semi-Open and Open. I’ve set the virtual drum. NN-Nano has its
open hat sounds use the same voice but with different each of my three hat pads to own implementation of this feature
decay times. You can model this behaviour in Kong a different variation (skipping too, with four different sample slots
using the Synth Hi-Hat module. In Screen 3 I’ve chosen Semi-Closed), and can now corresponding to the Hit Types. Each
this module on pad 7, then used the Drum Assignment play authentic analogue hats. hit can have multiple velocity layers too.
grid on the right of Kong to also assign pads 9 and 11 to The three sounds auto-choke And you probably spotted that the Tone
this channel. You can see that these pads (which map to (as they are the same source) and Noise mini-modules have a Hit Type
the cluster of three black keys in the C1 octave of a MIDI and share a single signal path. selector on their panels, so you can set
controller) now all have 7 on them. For further experimentation up variation or accent pads for the main
At the bottom right of Kong’s panel you’ll see the Hit I’d also highly recommend module with the supporting generators
Type section. When you select a pad that’s assigned exploring using this technique only triggered by some.  

www.soundonsound.com / January 2023 153

 TECHNIQUE
Pro Tools
Pro Tools has some powerful
features for quantising both
MIDI and audio.
JULIAN RODGERS

hether your projects are mostly MIDI or

W audio-based, whether you play your parts

in in real time, draw in your MIDI with
a mouse, or program your beats by dropping samples
on the timeline, at some point you’ll want to quantise
them. Here’s what you need to know to get things in
time in Pro Tools.
‘Quantise’ means to restrict to a limited set of
values. We do this when rounding decimal numbers
to integers, for example: numbers between 1, 2, 3 and
so on are nudged up or down to the closest ‘allowed’
value. This is most commonly applied to music with
regard to pitch and to timing. When applied to timing,
quantisation forces notes to precise note time values.
But there’s more to quantising than 100 percent ‘on
the grid’ music. In fact, a lot of quantising involves
putting human feel back into the music.
If, for example, you are a producer who draws in
MIDI and drops samples with your DAW’s snap-to-grid
function enabled (in Pro Tools that would be working
in Grid Mode), you might wonder why you would
need to quantise at all. After all, your music is already
snapped to the grid and ‘in time’. The answer lies
in the difference between ‘on grid’ and ‘in time’.
I remember hearing a story of a producer explaining
to someone that if the snare is late once, that’s out
of time. If it’s late by the same amount on every hit,
that’s groove.
Pro Tools’ quantise features allow much more
than moving note starts onto a regular and potentially
robotic-sounding grid. They make possible the
introduction of swing and shuffle; quantising can be
used selectively, for example to only move notes
falling on downbeats onto the grid; it can just nudge
notes closer to the ‘right’ location by a certain amount.
Power users can even take the off-grid ‘feel’ of an
audio recording and impose that onto MIDI and audio.
There are a few ways to quantise in Pro Tools.
Here’s how to get the results you want with the
minimum of fuss.
The Quantise Cluster
Introduced in Pro Tools 2022.9 is a new cluster of
quantise controls with which you can select clips
or MIDI notes, and quantise them by clicking the
big Q button. Simple. This cluster is available in the
Edit window, and also in the MIDI editors. It allows
easy access to the most commonly called-upon
quantise features. Here you can change the quantise
grid resolution, and a great tip is to use the Follow
Bars|Beats Grid setting at the top. When displayed on
the timeline, the grid lines are a useful visual guide;
if you can’t see them click on the word Grid (above
Nudge) in the toolbar and the lines will be displayed.
Using the Follow setting you can change
both your grid setting and your quantise
value using a keystroke. On a Mac it is
Ctrl+Opt and the plus/minus keys on the
numeric keypad.
From the quantise cluster you can
dial in Swing, which shifts every other
beat or sub-beat a little later, and the
often-overlooked Strength parameter,
which moves selected events closer to,
rather than fully onto, the quantised value.
These controls address 90 percent of my
quantise needs, particularly when I’m using
quantise selectively. You don’t have to
apply it to all the events in a performance,
and often shouldn’t anyway. If you need
more control, clicking on the gear icon will
take you to the Event Operations window
(which is where you’d access quantise
functions in earlier versions of Pro Tools).
Event Horizon
The Event Operations window can also
be accessed from the Event menu, or by
using the shortcut Opt+0. Although there
are nine pages to this window, it opens on
Quantize, which offers all of the features
of the quantise cluster with the addition of,
amongst other things, a useful Randomize
option for ‘de-quantising’, and a selection
of options including ‘Exclude within’ and
‘Include within’ parameters, which can
be used to target your quantising only
at notes which are either close to, or far
away from, the grid. Used with care, these
can tidy up a performance without killing
the feel.
If you want to check out the other
eight pages, rather than clicking the
drop-down menu, use Cmd+up/down to
navigate from the keyboard. One to pay
particular attention to is Input Quantize,
which does exactly as the name suggests.
This is a good area in which to try the
various options rather than defaulting
to a hard quantise. For example, rather
than quantising to a 16th-note grid, try
The new ‘quantise cluster’ of
controls lets you perform most
routine quantising jobs directly
within the Edit window.
a quarter-note grid with
‘Include within’ at 20 percent;
that way only the downbeats
will be quantised, allowing
looseness on the notes
in between.
MIDI Real Time
Properties
There is an alternative to
‘baked in’ quantising. MIDI
Real Time Operations are
applied on playback. The
difference is a little like the
difference between printing
effects using AudioSuite and
applying them on playback
using plug-ins, and to extend
the simile, Input Quantize
would be recording through
hardware effects. The Real
Time MIDI Properties menu
offers basic quantise with the
option of swing; the tools for
For more detailed control over
your quantising, the Event Operations
window is the place to go.

154 January 2023 / www.soundonsound.com


MIDI Real Time Operations are applied to your
MIDI data upon playback, and leave your original
note positions unchanged.
finessing the quantisation found in the
Event Operations window are absent,
but something worth investigating here is
the use of Groove templates, which are
custom grids that can be used to inject
a little more human feel. These are also
available in the Event Operations window,
but their use in a real-time environment
can be more immediate and more fun.
Check the MIDI Preference ‘Display Events
As Modified by Real Time MIDI Properties’
box if you want to see as well as hear
the results. If you want to dive deep
on this subject, check out the process
for extracting Grooves from existing
performances using Beat Detective.
Quantising Audio
So far we’ve only discussed quantising
MIDI, but audio can be quantised using
Elastic Audio — and with the introduction
of ARA 2 integration, Melodyne
can now be used to quantise
the timing of audio as well
as in its better-known role as
a pitch-correction tool.
Elastic Audio is possibly best
known as a way to conform
loops to the tempo of a session.
This kind of time-stretch
algorithm was groundbreaking
as a tempo-manipulation tool
in early incarnations of Ableton
Live, but it is useful for quantising
too. Once enabled in the track
header, and after the necessary
analysis has completed, you can
Melodyne offers a powerful
alternative to Elastic Audio for quantising
your audio recordings.
quantise in a very similar way
to using quantise with MIDI.
Elastic audio is file-based,
so even if you’re only using
a small segment of a long
audio file in your clip, it will
analyse the whole file. When
complete, you can use Elastic
Audio to fix individual timing
errors by selecting Warp in
the track header and grabbing
Warp Markers with the Grab
tool, but make sure you
double-click the neighbouring
Warp Markers to fix their
positions or you’ll end up
moving everything!
If you want to experiment
with quantise operations
across entire clips, you can
treat Elastic Audio much like
MIDI, using the same tools.
Get a tambourine or shaker
loop, hard quantise it using
the Q button in the Quantize cluster and you’ll have
a ‘straight’ loop. Dial in some swing from the Quantize
cluster and hear the effect. Dial in some Randomize
from the Event Operations window (because no
shaker is completely in time), pull the downbeats back
into time using the quarter-note quantise trick detailed
above, and you’ll have just the right amount of groove.
Quantising With Melodyne
This used to be the extent of your quantise options,
but if you are on Pro Tools 2022.9 or higher you might
check out Melodyne as a quantise tool. If you have
Melodyne Essential, you’re limited to monophonic
pitch manipulation; the semi-magical DNA polyphonic
workflow options of the higher tier products isn’t
available. But something which is overlooked by
some is that Melodyne offers an alternative, and many
think superior, method for timing manipulation and
correction.
The choice between Elastic Audio and Melodyne
is made on a per-track basis, so both can’t be used at
the same time on the same track. If you try to switch
from Elastic Audio to Melodyne you’ll be prompted
that you’ll need to either commit or abandon your ‘live’
Elastic Audio processing. But if you’ve never checked
it out, open up a drum loop or similar and see what can
be achieved with the Quantize Time Macro. It’s simple
but very effective, and sounds great. If you’re finding
you can’t manually adjust the length of notes using the
Main tool, try clicking the note icon in the top right of
the Melodyne Editor to enter the Melodyne equivalent
for Slip Mode, or adjust the Time Grid Settings to
a suitable value.
There’s much more to quantising in Pro Tools than
conforming MIDI and audio to a regular grid, and using
whichever combination of one-click quantise from the
new Quantize cluster, the extensive features found
in the Event Operations window, the Real Time MIDI
Properties and whichever combination of Elastic Audio
and Melodyne best suits your purpose, you’ll find
there’s not much you can’t do. And for those things,
there’s always Beat Detective...  
www.soundonsound.com / January 2023 155
 TECHNIQUE
Cubase’s Sampler Track is
capable of making surprisingly
playable backing vocals.
JOHN WALDEN
ot everyone can afford top-flight vocal
N sample libraries, and if you all you need
are some basic vowel-style vocals parts
then the Sampler Track, included in the Pro, Artist
and Elements versions of Cubase 12, offers a way
to roll your own. The steps involved are pretty
straightforward too, so it’s easy to explore the idea,
even if the parts you create will only be used as
placeholders to be replaced with a singer later on.
Vowel Play
In order to create a playable vowel-based vocal
instrument, we obviously need a sample as a starting
point and, to make the subsequent steps easy, two
key things are worth considering when selecting
or recording a suitable sample. First, ensure that
you know (or can work out) the pitch of the original
sample. Second, consider whether the performance
you wish to create requires sustained sounds or
shorter, staccato sounds (for example ‘ooooh’ and
‘oh’, respectively). If possible, pick or create a source
sample that matches this style — while the Sampler
Track does let you create a sustained sound from
a shorter sample, for vocal samples in particular it’s
not always possible to make the necessary looping
sound completely seamless.
Drag & Drop
Execute the Project / Add Track / Sampler command
to create a new Sampler Track. By default, this opens
the (initially empty) Sampler control panel in the
156 January 2023 / www.soundonsound.com
Project window’s Lower Zone.
Then, simply drag and drop
your sample into this panel.
Hey presto! You now have
an instrument, made of that
single sample mapped across
the full MIDI note range.
To get the best from it
some tweaks are needed, and
the first screenshot shows the
settings I used for my working
example; a sustained ‘dooo’
vocal sample. I’ll focus on the
key settings I adjusted, but
don’t let that stop you from
experimenting with the other
controls if you wish! I’ve also
created some audio examples
so you can hear what’s going
on, and you can find these
on the SOS website (https://
sosm.ag/cubase-0123).
A few minor modulation tweaks within the Pitch, Filter and Amp panels can help add
a touch of sonic movement when triggering your vocal sample.
Cubase
It’s worth spending some time refining the
loop region if you want a natural end result from
a vocal sample.
Within the waveform display, the
first task is to adjust the sample’s start
and end points, by dragging the small
‘S’ markers. If there’s a lot of unneeded
material, the Toolbar’s ‘[ ]’ button can also
be used to trim the sample. For smoother
playback, it’s worth applying a short
fade-in/out by dragging the solid white
squares associated with both S markers.
Next, in the bottom-most mini-keyboard
display, we need to check that the root
note of the original sample (highlighted
in blue) is set correctly to ensure the
most natural-sounding playback; you can
simply drag and reposition as required.
To Loop Or Not To Loop?
If we wish notes to sustain longer than
the original sample we need to configure
a loop region. If you select an appropriate
option from the Loop Mode drop-down
selection list (I selected Until Release
in this case), the green loop (L) markers
appear ready for adjustment.
Achieving a seamless loop using
the two L markers, even with the Zero
Crossing buttons active and using the
crossfade options for the loop start/
end points, can be a bit hit and miss. As
a guide, it helps to try and get as close
a match between the original waveform
(shown in blue) and the waveform of the
looped section (shown in green). Usefully,
if you hover the mouse directly between
the two L markers, a green horizontal bar
appears, allowing you to reposition the
loop section without changing the length
of the loop. Of all the steps involved, this
is the one most worth spending time on,
as our hearing is very sensitive to what
sounds ‘natural’ in the human voice;
patience will usually reward you with
a smoother end result.
A Bit Of A Stretch
In the Playback panel, I’ve enabled
AudioWarp. This applies time-stretching,
so playback is the same length at
different pitches. However, a quick
tinkle up and down the keyboard from
the sample’s root note will reveal two
things. First, the sound is a little static,
and, second, beyond a few semitones
from the root note, even with AudioWarp
engaged, the vocal quickly strays into
unnatural-sounding territory.
To address the first issue, we can dip
into the Pitch, Filter and Amp panels and,
in each case, apply some parameter
modulation to increase the sense of
movement. Clicking on the ‘mod’ label
for any of these panels superimposes
a modulation envelope upon the
waveform display. As an example,
the screenshot shows some of the
possibilities offered by Pitch modulation.
I’ve applied a short upwards scoop
into the note and a drop down as the
note decays. In anticipation of playing
sustained notes, I’ve also extended the
pitch curve over about three seconds
and added some minor pitch drift. You
could make this as detailed as you
might like but it makes for a slightly less
robotic sound when notes are held. Note
that you have to adjust the AMT slider
(far left) to a non-zero value before this
pitch modulation curve will have any
effect, but it does provide a very easy
means of scaling the magnitude (and
direction) of the modulation. Another
strong modulation candidate is the
filter frequency (using the envelope or
Keyfollow controls in the Filter panel)
as this can add pleasing tonal variation.
If you have longer source samples and
want to create a more gentle, pad-like
fade-in, the Amp modulation envelope is
where to start.
Make It A Multi
In terms of the second issue, we really
need a feature that the Sampler Track
doesn’t currently support: multisample
instruments. While that would be great to
see (pretty please Steinberg!), the very
streamlined nature of the Sampler Track
workflow does make it easy to construct
a simple workaround. Imagine we want
to be able to play our vocal sample
instrument over a single octave note
range but our current Sampler Track only
really sounds natural over a range of four
or five semitones around the root. What
we need is multiple Sampler Tracks.
For example, if we used three Sampler
Tracks based on different ‘oooh’ samples
originally recorded at C3, F3 and A3, and
configured using the steps described so
far, we could, in principle, achieve natural
playback over a full octave. Then, for
each track, we can constrain the active
note range using the mini-keyboard
display at the base of each Sampler Track
panel, ensuring that they don’t overlap.
For playback, if we then select all
three tracks within the main Project
window track list (so all three become
highlights and record enabled), any MIDI
notes you play will be sent to all three
tracks, but each track will only trigger
sample playback within its own limited
note range. The end result is, essentially,
a multisample instrument created from
three Sampler Tracks and a much more
natural playback of our vocal sample over
a full octave range.
And on a practical front, it’s worth
noting the Toolbar’s ‘lock’ button. Having
carefully created your first Sampler Track
with all the appropriate settings, if you
then duplicate it, and active the lock
button on the duplicate, you can drag and
drop a new sample onto this track without
it zapping your carefully crafted settings.
You can then go in and manually change
the settings for each sample as required.
And There’s More
Of course, having got into the swing of it,
you don’t have to stop at three Sampler
Constrain the active note ranges in each of
your Sampler Track instances, and you can achieve
a more natural sound over a wider note range.
With all three tracks selected, any MIDI note
input will reach all three Sampler Tracks but, with
the active note range of each track limited (as
shown in the previous screen), each track will only
respond to a limited range of MIDI pitches.
Tracks. How about a second set of three
Sampler Tracks based on three different
sampled ‘oooh’ performances — with
each set of three panned to opposite
sides of the stereo field? Select all six
tracks and play for a nice, wide vocal pad.
Equally, you could create a second
set of Sampler Tracks for a different
vowel-based vocal (I’ve done this in the
audio examples with a ‘baaah’ sound).
With careful use of the mini-keyboard
to create suitable mapping, you can
trigger this second sound in a different
part of your MIDI keyboard, allowing you
to mix and match a performance with
the two different vocal sounds using left
and right hands.
And, if you route the output of all
these Sampler Tracks to a suitable
audio bus you can add some global
reverb and delay, while Pro users could
also try an instance of Cubase’s Cloner
plug-in to add some further depth to
the vocal ensemble.
Finally, if you think you might like to
reuse these new DIY vocal instruments
in a future project, simply select all of
the Sampler Tracks you have created,
right-click on any one of them, and
choose Save Track Preset. Provided the
original samples used don’t get moved
or deleted, you can then load this Track
Preset within a new project and your
various Sampler Tracks will be fully
restored and ready to play.  
www.soundonsound.com / January 2023 157
 TECHNIQUE
Logic
We explore
a few ways
to share your
projects with
collaborators.
DAV I D R I CA R D

hile Logic

W Pro provides
multiple
means and methods for
sharing projects and files,
choosing the right one
depends on a range of
variables. It may surprise
you that there is no clear
one-size-fits-all approach, and each solution about what’s happening on each take. This project was shared by five musicians (me,
has its pros and cons. Rather than describe Singers might give you several different alto sax, tenor sax, trombone, and trumpet). The
mix at the top of the session was provided by the
all of them, let’s look at a few scenarios and approaches for a vocal (more aggressive,
band, the MIDI horn tracks are disabled, and each
decide which route is optimal. falsetto, laidback) and, while it’s not a big horn player added their part one at a time.
nuisance to have this labeled on the track
Howdy Partner name or even the file name, it’s much To prep your session, create a new
Say you have a production partner with cleaner to have this information in its Alternative (File / Project Alternatives /
whom you collaborate. You write, arrange, own box, especially if the notes are more New Alternative...) and give it a name or
and produce your songs as a team, albeit extensive than just a few words. version number. This way, no matter what
from different locations. Each of you is
adding to the other’s contributions and the
Logic session is continually being updated.
The main obstacle with this type of
collaboration is getting the Logic session
files to open easily and completely intact on
each producer’s system. It might be wise
in this case to limit your instruments and
effects to Logic’s offerings. If you indeed
have some of the same third-party extras,
feel free to include them. Something like
Kontakt might be tricky since it will look for
samples on drives that may not exist on one
of the systems. But if you are a team that
collaborates regularly, you can set up your
drives to match each other’s and employ
other strategies to render your systems
effectively the same.
One excellent feature for sharing
project files is the Notes section. You can
paste your lyrics or general directions
and ideas into the Project Notes, and
your partner can edit them too. It’s such
a simple and easy to overlook benefit but
I love not having to refer back to an email
or a text document once I’m working in
Logic. I’ll usually copy and paste all of the
information (the brief, notes, emails from Here I’m prepping files to send to a trumpet player to record some overdubs. Not every element of this
project needs to be separated out (or even included) in the stems and that is indicated by only a few tracks
clients) into the Project Notes even when
selected to be exported as stems. Notice that I included markers to inform the player where his overdubs
I’m working on my own. are. I also export an audio click track, so the other musician doesn’t even have to adjust the tempo on their
Additionally, the Track Notes section is DAW. One more thing to note: the trumpet, trombone and saxophone stacks are routed to the Horns track. This
where your collaborator can provide details allows me to bounce the horn mix without separating out each horn group onto its own stem.

158 January 2023 / www.soundonsound.com

 ELECTRONIC ELECTRONIC
MUSIC MUSIC

Robert Henke Electric Indigo

Co-founder of Ableton
Part electronic visual music artist and part software
developer as co-founder of Ableton, Robert
Henke talks to Caro C and delves into the overlap
between artist and engineer and how software
needs to be accessible in order to encourage and
not hinder the creative process.
female:pressure
Susanne Kirchmayr, also known as Electric Indigo,
is an Austrian music producer, techno DJ and
founder of female:pressure, an international
network of women in the electronic music
industry. Here she chats to Caro C about her
career beginnings through to her current projects.

PEOPLE & RECORDING

MUSIC INDUSTRY & MIXING

Novation - 30th Anniversary Recording Vocals

Grooveboxes, Controllers and Synths
This year marks the 30th birthday of one of the
world’s leading manufacturers of synths, samplers
and hardware controllers. Chris Calcutt chats with
Sam Inglis and looks back over three decades
of innovation and explores the key products,
technologies and partnerships behind a British music
technology success story.
Pro Techniques Explained
Mark Mynett, sound engineer and senior
lecturer presents part four of his recording
series. Here he details how to prepare a
vocalist for studio recording and how to
select and set up your recording equipment
for optimal results.

Follow our channels by subscribing to the shows on Apple Podcasts,

Google Podcasts, Spotify or wherever you get your podcasts.

All shows are mastered to the highest quality the podcast channel will support and are in stereo.

Check out our website page for further details

www.soundonsound.com/podcasts
 TECHNIQUE
SHARING YOUR PROJECTS
happens, you can always revert back to
your previous version.
Next go to File / Project Management
/ Consolidate, and select all the elements
that apply to your project (or, to be safe,
check them all). This process is similar to
the dialogue you’re presented with when
saving a file in Logic for the first time. In case
you didn’t anticipate needing your arsenal
of sounds saved on your various hard drives
included with your project at its onset,
Consolidate gives you another chance to
incorporate them.
And finally, before sending your Logic
session along with all of its resource files,
right-click on the folder that the project
resides in and select Compress. This will
ensure that every element of the project
remains unharmed on its journey.
While this working scenario may
sound a bit extreme, there are definitely
scaled-down variations of it that are quite
commonplace. I’m often hired by bands to
write horn arrangements. They provide me
with a mix that I load into my Logic horn
arrangement template, and, using MIDI
instruments, I create the full arrangement.
After I receive the band’s approval,
I proceed to the recording phase. I write out
the parts in a notation program (although
I have used Logic’s notation for this in the
past). Because all of the horn players I hire
all use Logic, I can send them the session
as-is. I will even have the tracks labelled
for them to record onto. I disable the
MIDI regions and hide them. The session
includes all of my routings and plug-ins but,
because nothing is going through them at
this point, it doesn’t cause any issues with it
being opened on a new system.
Bounce To This
Here’s another scenario. You’re working
on a full album with 60+ track sessions
and system-crashing amounts of native
and third-party plug-ins chugging through
it. Your drummer uses Logic, so what are
the options? For one thing, you don’t need
to put too much weight on the fact that
you both use Logic. If you bounce out
basic stems, the drummer will have all they
need. They may even be okay with just
a stereo track. However, it’s not uncommon
for them to need more control. Maybe the
tambourine track is throwing them off or
the vocal is too distracting or the bass is
too low. A decade ago, outputting stems
by instrument group would be tedious but
these days, it’s pretty simple.
Select each track or stack that you want
a separate stem for (by Command-clicking
160 January 2023 / www.soundonsound.com
Here I’m importing all of the Global Tracks into
a new Logic session. Once imported, I will add stems
and send this off to the drummer.
them), go to File / Export / X Tracks as Audio
Files. Confirm that X is the number of tracks
you have selected. If you have four tracks
selected and that option is reading ‘47
Tracks as Audio Files’, you probably have
selected tracks that are hidden.
Ensure that your locators are set to cycle
the entire song and, additionally, confirm
that Export Cycle Range is selected in the
export dialogue.
Once your drummer has the files, they
can load them into Logic or any DAW and
get to work. Most DAWs can use the audio
file’s tempo, but just in case it doesn’t,
indicate the bpm in the folder name.
Markers are embedded into the audio files
so take advantage and add any that may
help your collaborator navigate your song.
Without additional instructions, the
drummer should send you back one file per
track that all line up at the beginning of the
cycle range.
Tempo Tantrums
That’s all very well, but what if the song
has multiple tempo and meter changes
throughout? In this scenario, we need to
find a solution that allows us to export stems
while still using some of the tools that Logic
affords us. A quick and dirty fix is simply to
include a MIDI file along with your stems.
Create a track with any software
instrument on it. Play a few notes at the
beginning of the cycle range, then jump
to the end and record a few more notes. If
you have just one region that fills the cycle
range, great. If you have two regions at
the start and
end, select
both and hit
Command+J to
join them.
Now, with
the region
Consolidating
a project before
sharing it ensures
that all of the
project’s resource
files are included.
Logic
selected, go to File / Export / Selection
as MIDI File. You would then instruct the
drummer to open the MIDI file in Logic
(or any DAW) before loading in the files. If
everything was done correctly, all the tempo
and meter changes should load into the
new session.
Now let’s suppose our drummer would
love it if we gave them a Logic session
with all of the tempo and meter changes,
but they only need the stems. Here’s what
I would do: create a new Logic session, then
go to File / Import / Logic Projects...
Locate the song that we just got the
stems from and, from the long list in the
media browser, Shift-click to select Marker,
Signature, Tempo and Notes (if applicable),
and click on the Contents box. Now hit the
Add button on the bottom. At this point we
should have an empty song with all of the
markers, tempo changes and so on from our
masterpiece. Now, go ahead and drag in the
stems you created earlier.
I would consider this to be the
best-of-all-worlds approach. The musician
only has to deal with a few tracks of
audio, without any software instruments
or plug-ins to gum up the works, yet they
have a session with notes, markers, and
meter and tempo changes included. They
can then send the entire Logic session
back to you (no need for them to bounce
anything), and you can then use the Import
Logic Projects process to bring the drum
recordings into your session.
What makes this method ideal is that
the drummer can take advantage of track
alternatives and include numerous takes
without having to export each batch, one
at a time.
Sharing projects in Logic is a bit of
a balancing act. The scope of the project,
the role and needs of the collaborator(s),
and their proficiency with Logic must all
be considered when deciding on the
ideal approach.  

 
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COMPACT DISCS
DAV I D B A E R
Horowitz or Elton John, and prices. That’s only one of many range. Mahler symphonies call
have every transient start of reasons I find the resurgence for colossal forces. My informal
bought my first CD player in every note be pristine, entirely of vinyl incomprehensible. measurement of the dynamic

I 1984. I had been lusting for

one since I first saw/heard
the new format three years
earlier, when CDs were first
introduced. There were many
reasons for that lust. No more
requisite cleaning, no getting
up every 20 minutes to put
on a new record, and most of
all, no more intrusive noises
of any kind. LPs (aka records),
which is what we called vinyl
back in those days, were truly
a fragile medium. It was a
given that the more precious
to you a record was, the more
certain it was to incur very
audible damage.
Not so CDs, which are
comparatively indestructible.
No matter how many times a
CD is played, it sounds just as
good as it did the first time. It
was heavenly to at last be able
to listen to a piano recording,
be it of Keith Jarrett, Vladimir
free of scratchy grunge and
other unpleasantness. CDs
have a blissful lack of noise
in general. The surface
noise of even never-played
LPs is easily audible. The
background noise of CDs
played on even low-end
systems is considerably
quieter than the ambient noise
of the listening room.
It took me three years to
take that initial plunge, in part
because it took that long for
the major stereo component
manufacturers to add CD
players to their line-ups. By
1984, though, it was a safe bet
that the technology was here to
stay. But CDs back then were
quite expensive – they cost two
to three times what the same
LP cost, due to insufficient
manufacturing capacity.
Today, it’s just the opposite
in terms of CD versus vinyl
And I do mean off-the-charts
incomprehensible, in some
cases. Consider the magnificent
collection of Mahler orchestral/
choral works released by the
San Francisco Symphony on
SACD between 2001 and
2009. Those discs are among
my most highly prized. SFS
also decided finally to release
the collection on vinyl, in a set
costing $750 — about triple the
cost of the SACDs.
But this is Mahler we’re
talking about. What a horrible
idea vinyl is, irrespective of
price. The most one can safely
put on one side of an LP is
about 23 minutes. Mahler
symphonies have a number of
movements that easily exceed
that limit. So what did they
do? Fade-outs/fade-ins, or
abrupt ecstasus interruptus?
I honestly don’t want to know.
And then there’s dynamic
range in one of the symphonies
(No. 5) is 45dB. That’s no
challenge for 16-bit digital as
found on CDs, which gives us
90dB to work with. But on vinyl,
with its dynamic range of about
60dB, those passages 45dB
down will be given serious
competition by the surface
noise. So, hey, why not fix
that with a little compression?
Again, this is Mahler... so,
sacrilege!
Thank you, but I think I’ll
stick with digital. You might
respond: why not just go
with digital streaming then?
I have a number of very good
reasons why that has little
appeal to me, but that rant will
have to wait for another time.
I’ll happily continue to just
listen to my substantial library
of music on CDs. I truly love
music, so, of course, I love
CDs... a lot.  

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