THE WORLD’S BEST RECORDING TECHNOLOGY MAGAZ INE 85 — 2022

TM
TM

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

INTEL
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AMD
WE BENCHMARK THE LATEST
CPUs FOR MUSIC

MIKE
Super producer and synth god!

Encanto
ow the hit soundtrack was create

Korg ARP 2600 M WOR TH $3282

A classic synth is reborn
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REVIEWS: APOGEE / PMC / BEYERDYNAMIC / CHASE

REVIEWS CHA BLISS / HIFIMAN / PRESONUS / WALDORF / UJAM / NEKTAR
USA $11.99 / Canada C$11.99
 L E ADE R

CENTRAL PROCESSING
One of the core skills of the audio engineer is
overcoming limitations. In the days of tape, that
might have meant squeezing additional overdubs
into the gaps where the bass wasn’t playing, or
giving everyone in the band their own fader to
move at mixdown. These and a hundred other
familiar compromises went out of the window
when computers replaced consoles. Practically
overnight, we could stop worrying about noise,
frequency response, track count, alignment,
maintenance and so on. And to compensate,
computer-based recording introduced some new
and confusing limitations of its own.
Some of these are inherent to the technology.
For example, a degree of monitoring latency is
inevitable in any digital system. But the limitation
that most of us have spent the most time fighting
is processing power. We’ve all experienced the
joys of trying to get a busy mix to play back
reliably on a maxed-out Mac or PC, doing battle
with buffer sizes and our DAW’s freeze function.
In the early days of DAW recording, other
computer resources were equally fragile. It was
easy to ask too much of a 5400rpm hard drive
or even a serial-port MIDI interface. Those days
are long gone, but overcoming the limitation of
finite CPU power has been more of a challenge,
because demand has kept pace with supply.
We don’t simply ask each new generation of
processors to do more of the same; instead,
we want to run ever more complex plug-ins
and software instruments. At the same time,
Sam Inglis
Editor In Chief
the exponential growth in computing power
characterised by Moore’s Law has slowed down.
Nevertheless, my sense is that CPU
performance is no longer a bottleneck for most
users. Even on my humble Mac Mini I can’t recall “When ‘cutting
the last time I needed to freeze a track, while edge’ means
the MacBook Pro I tested last year could run
a 300-channel Atmos mix with its eyes closed multiple streams of
and one hand tied behind its back. And, as Pete uncompressed 8k
Gardner’s definitive benchmark tests in this issue
show, the choices available to PC users are also video, audio starts
mightily impressive. When ‘cutting edge’ means
multiple streams of uncompressed 8k video, audio
to look like a solved
starts to look like a solved problem. problem.”
Sadly, this probably won’t mean we can jump
off the upgrade cycle. OS upgrades, software
compatibility and driver support will eventually
kick us onwards and upwards. But what it might
mean is that we can think differently about
upgrading. For as long as I can remember,
buying a new machine for the studio has meant
a big hit to the wallet, because it’s always been
essential to have the latest, fastest components.
If CPU resources are no longer at a premium,
perhaps we can think about using our financial
resources differently.

EDI T OR IA L ADMIN I S TRATIO N ADVER TISING

sos.feedback@soundonsound.com
SOUND ON SOUND NORTH AMERICA
Editorial Director Dave Lockwood
1083 VINE STREET #727 Executive Editor Paul White
HEALDSBURG, CA 95448, USA Editor In Chief Sam Inglis
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Reviews Editor Matt Houghton
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CAMBRIDGE, CB4 2HY, UK Production Editor Chris Korff
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 138 MIKE DEAN

INTHISISSUE www.soundonsound.com

April 2022 / issue 6 / volume 37

WIN
FEATURES
98 Basic Vocal Recording MILAB
VIP-60
These vocal tracking tips will help you achieve decent results
every time.

102 Get More From Your DAW

Learning to manage CPU resources effectively will make you
WORTH
a better mixer. $3282
C O V E R

122 Intel Vs AMD: CPUs On Test

Intel’s latest CPUs promise big performance gains. But can they
topple AMD in our audio benchmarks?

130 How I Got That Sound

Carl Bell and Ben Grosse tell us how they created the distinctive
guitar sound on Fuel’s ‘Solace’.

132 Inside Track: David Boucher
With six lead singers, an orchestra and more, ‘We Don’t Talk
About Bruno’ from Encanto was always going to be an epic
production challenge — and then lockdown happened.
138 Mike Dean: Super Producer
With an extraordinary track record as producer, songwriter,
musician, mixer and remixer, Mike Dean is a one-man
music industry.
144 Mixing Atmos: Andy Bradfield
Leading mix engineer Andy Bradfield shares some of the lessons
he’s learned in getting to grips with Dolby Atmos.
PAGE 76
148 Hildur Guðnadóttir & Sam Slater
HBO’s Chernobyl exposed the horrifying story behind the 1986
nuclear disaster — and made a star of composer Hildur
Guðnadóttir.
158 Q&A
Your studio and recording questions answered.
162 Why I Love... Making A Difference
Simon Glenister explains how music can change people’s lives.
 78 SONIC POTIONS/ERICA SYNTHS LXR-02

ON TEST
10 Beyerdynamic M90 Pro X 52 Waldorf M 95 Denise Audio Poltergate
& M70 Pro X Wavetable Synthesizer Gate & Transient Shaping Plug-in
Microphones
58 PreSonus Revelator io24 96 IK Multimedia T-RackS Fame
14 Toontrack Americana EBX USB Audio Interface Studio Reverb
EZbass Expansion Reverb Plug-in
62 Gainlab Audio Empress
Passive Valve Equaliser
16 Fracture Sounds Box Factory 156 Sample Libraries
Sample Library 64 UJAM Finisher Dynamo Heavyocity Mosaic Leads
Effects Plug-in
20 Chase Bliss Audio CXM 1978 Sonora Cinematic Poiesis Cello
Reverb Pedal 68 Red Panda Tensor Klevgrand Borsta
Looping Pedal
24 DryBell Vibe Machine v3 Best Service Elysion 2,
Guitar Effect Pedal 72 Nektar Impact GXP61 & GX Mini By Sonuscore
Controller Keyboards
26 Audio Imperia Solo
Sample Library 78 Sonic Potions/Erica Synths WORKSHOPS
LXR-02
36 ARP 2600 M Drum Machine 106 Cubase
Analogue Synthesizer
84 Apogee Duet 3 110 Logic
USB Audio Interface
42 Hifiman Sundara, Arya 112 Pro Tools
& HE-R10D 88 PMC6-2
Headphones Active Three-way Monitors
114 Digital Performer
118 Studio One
48 Tierra Audio Flavours 94 Boss IR-200
Inline Booster Preamps Amp & Cab Emulation Pedal 120 Live
EVERGROOVE
STUDIO
Atmos mix room,
Colorado
The mix room of Evergroove
Studio in Evergreen, Colorado
has been redesigned to
handle Dolby Atmos projects
featuring 11 Focal monitors in
an impressive 7.1.4 setup.
atmos@evergroove.com
Photo: Ethan Kotel
SM6 LINE

FOCAL | SM6 LINE

The SM6 line of monitoring
loudspeakers has been designed
for recording studios looking for
monitors capable of revealing every
detail of the entire audio spectrum
without any masking effect. The best
of Focal’s technology is the reference
in its price range owing to its extreme
neutrality, its precise stereophonic
imaging, and its ability to resolve the
most tiny sonic details.

Keep focus and mix on!

Focal.com
Distribution by Focal Naim America
Contact us for more information:
info@focal-naim.com
 ON TE S T
Beyerdynamic
M90 Pro X & M70 Pro X
Beyerdynamic’s new studio mics are as classy as they are versatile.
PAUL WHITE
F
or those unfamiliar with the brand,
Beyerdynamic started life in Berlin
in 1924, then moved to Heilbronn
in Germany after the second world war,
before adding a US subsidiary. The
company originally built loudspeakers,
but by the 1930s had designed and
manufactured the first ever set of
dynamic headphones. Their new Pro
X series, launched in 2021, currently
comprises two new headphone designs
(which we’ll also be reviewing), as well as
two microphones, one dynamic and one
capacitor model.
Starting with the latter, the M90
Pro X is an elegant, cardioid-pattern,
large-diaphragm capacitor microphone
priced to appeal to the home studio
operator. Its 34mm capsule is a true
10 April 2022 / w w w. s o u n d o n s o u n d . c o m
Microphones
capacitor (ie. externally polarised) design
residing in a suave, matte-black, tubular
body and protected by a single-layer
mesh screen. Beyerdynamic are
promoting the M90 Pro X as an
all-rounder, suitable for use on vocals,
drums and acoustic instruments such as
guitar, strings or piano.
Requiring a phantom power supply
of between 20 and 52 Volts, the
microphone is specified with a frequency
response of 20Hz to 20kHz, though
the character of a microphone is often
related to the subtle ways that the
frequency response varies, especially
in the upper-mid or ‘presence’ region of
the spectrum. Consulting the frequency
plot shows the usual proximity bass lift
when used close up, while at the high
end there’s a broad presence bump
starting at around 4kHz which then
tails off gradually above 10kHz. The
overall shape of the curve suggests
that the quoted frequency response
allows for a ±5dB deviation, which is
necessary to encompass the presence
boost region. The accompanying polar
plot shows a tight cardioid pattern
with a slight rear-facing lobe at higher
frequencies, as you’d expect from
a hypercardioid microphone.
With a nominal impedance of 180Ω,
the maximum SPL (at 1kHz) is specified
as a very loud 133dB, which may explain
why the designers didn’t feel it necessary
to add a pad switch. The self-noise is
creditably low at 5.6dB A-weighted
(equating to a signal-to-noise ratio of
88.4dB relative to 1Pa), while the open
circuit sensitivity at 1kHz (where 0dB
= 1V/Pa) is 36.3mV/Pa. The capsule
is mounted on a resilient structure to
Tiny Box,
Huge Sound.
This is the Grace Design m900 headphone amplifier, DAC and preamp.
Its incredible sonic detail combines with a rich musical character that
makes working on music easier and listening to music more fun.

Use it in the studio, at the office, at home, or on the road. Whether

recording, mixing, editing or just getting blissed out with the music you
love, the m900 is an affordable, must-have little piece of audio gear.

Proudly made by friends and family in Lyons, Colorado. Visit us at

gracedesign.com for complete details and purchase information.

Wi hout doubt the m900 is an excellent DAC/ headphone amp/ monitor controller,
d I was genuinely surprised to discover how affoSdable it is given the superb sound
quality and facilities. Its technical performance, monitoring versatility, com t s
and sheer practicality make it an absolute delight, and I recommend it highl
- Hugh Robjohns, Sound 0n Sound Magazine, February 2021

W W W.GRACEDESIGN.COM
 ON TE S T
BEYERDYNAMIC M90 PRO X & M70 PRO X
minimise vibration-borne noise. There’s
no low-cut switch so if you need to
eliminate the extreme lows, you’ll need to
use your preamp’s filter if it has
one, otherwise you’ll need
to EQ when mixing. For
those who worry about
such things, the overall
length is 197mm, the
diameter 52mm, and
weight is 296g.
The balanced
XLR connector is
attached directly to
the transformerless
circuit board, which
is populated in the
main by surface-mount
components, though there
are some discrete parts on
there too. A bespoke elastic cradle
shockmount is included, along with a soft
storage pouch for the mic, and a pop
filter with an integral gooseneck.
When used on vocals, ‘working’ the
mic distance allows you to effectively
control the amount of low end you
capture — moving closer brings the
proximity effect into play more strongly
to lift the bass. When working close up,
this is balanced by that broad presence
peak, yielding a ‘smile’ curve that boosts
both the lows and highs, adding weight
and clarity to the sound. In comparison
with some of my other similarly priced
capacitor mics, I found a lot of similarities,
the main differences being in that
all-important presence region, which is
why you should always choose a vocal
mic based on how it sounds on the voice
you plan to use it on, and not just on
how the spec sheet looks. The fact that
the presence peak is relatively broad
here helps keep the highs sounding
smooth as narrow presence humps can
sometimes sound peaky.
Testing the mic on acoustic guitar,
I quickly achieved a very usable sound,
though I did need to use a little EQ to
tame the 200Hz region for my particular
instrument and choice of mic position.
In this application the overall character
comes over as detailed, and without the
nasal honk that a narrower presence
peak, especially one lower down in the
spectrum, often imparts. Again then it
seems that the broad presence peak
works in its favour.
To summarise, the M90 Pro X is a very
capable mic given its attractive pricing.
It is very nicely engineered, it has a low
12 April 2022 / w w w. s o u n d o n s o u n d . c o m
Unlike the capacitor M90 Pro X, its dynamic
sibling the M70 Pro X is an end-fire microphone.
noise floor and, as the manufacturer
suggests, it is sonically adaptable, which
is important if you can only afford one or
two mics.
Beyer’s Dynamic
The dynamic M70 Pro X is similarly
styled to the M90 Pro X capacitor
model, but is slightly longer at
185mm, and also a little heavier, at
320g. It features eight wide vertical slots
protected by mesh, with the end of the
mic being protected by the same mesh.
Unlike the M90 Pro X, the M70 Pro X has
its capsule facing the top of the body,
rather than the side — this is an ‘end-fire’
microphone. Disassembly reveals that all
components, including the capsule, are
fixed to the lower body section, so the
majority of the mic body is essentially the
‘basket’ part of the microphone.
The M70 Pro X has a 40Hz to 18kHz
response when used at one metre
or more from the source, though this
changes to a 25Hz to 18kHz response
when used up close, again due to
the proximity effect. The M70 Pro X is
designed primarily for recording speech
and vocals, though many dynamic
vocal mics have secondary applications
including the likes of bass and guitar
speaker cabinets, while the lack of any
LF roll-off here means it should also
work well on kick drums. Its internal
resilient suspension reduces the
effects of mechanical vibration, while
the included gooseneck-mounted pop
filter suppresses plosive
popping sounds. The kit
also includes the same
elastic suspension mount
as provided with the M90
Pro X capacitor model.
As with many
microphones designed
for use with voice, the
frequency response is
by no means flat once
you get above 1kHz, and
because this mic has
a cardioid pickup pattern,
there’s a significant low-end
proximity boost when used
close up. There are two
distinct peaks to the presence
boost, one at 4kHz and one at
8kHz, with the response falling
away gradually above that.
The sensitivity is quoted as
1.8mV/Pa.
My first test with this mic
was to use it to record Hugh
Robjohns when we were
putting together a podcast. In
this role it provided excellent
vocal clarity, with a supportive
low end, and required little to
no EQ at the mixing stage other
than rolling off the extreme lows.
The mic can also work well on
guitar cabinets, especially if you need
a biting rock tone — in fact it can stand in
for any of the applications for which you
might use something like an SM57, but
offers a little more high-end extension
than Shure’s classic. Percussion also
fared well when recorded via the M70
Pro X, so although it is intended as
a vocal mic, it’s fair to say that it is
a capable, general-purpose microphone.
Summing up then, both of these mics
are classy, versatile affairs that should find
many uses in the studio. They’re keenly
priced too, especially given the included
quality cradle and pop screen.
summary
Two professional-quality microphones that
should suit many different sources. For
recording podcasts and the like, the M70 Pro
X does an excellent job, though the M90 Pro
X might be the better option if you also need
to record instruments.
$ M70 Pro X $299, M90 Pro X $349.
T Beyerdynamic USA +1 631 293 3200
E info@beyerdynamic-usa.com
W www.beyerdynamic-usa.com
 ON TE S T
Toontrack
EZbass Expansion
Americana EBX
Toontrack’s latest EZbass
expansion delivers a classic
American sound.
JOHN WALDEN
A
s with EZdrummer, Superior
Drummer and EZkeys, Toontrack
have generated a tempting stream
of additional content for EZbass since
it first appeared. The latest offering is
the Americana EBX and, while it might
be tempting to think that the option for
an additional vintage bass (the EBX is
built from sampling a late-’60s Fender
Telecaster Bass) might not be an essential
purchase given the vintage model included
within the core EZbass package, don’t
jump to any conclusions just yet; there is
musically more to this (virtual) instrument
than just Americana.
Americana EBX follows the format of
the previous EBX releases. You therefore
get Toontrack’s beautifully detailed
sampling of the instrument itself, plus
a collection of MIDI grooves that provide
performance options that fit within the
broad banner of Americana. The sampling
provides two versions of the bass; one
played via conventional fingerstyle while
14 April 2022 / w w w. s o u n d o n s o u n d . c o m
the other uses a muted thumb approach.
For both styles, you also get a range of
presets from clean DI through to gently
overdriven amp tones plus, of course, the
additional tone-shaping tools built into the
EZbass interface.
Bass Exploration
Compared to the core EZbass vintage
instrument, while in a similar ballpark,
the Americana model does feel different.
Describing bass tones is a subjective thing,
but you get a somewhat softer, less strident,
mellower sound here. Indeed, exactly the
kind of bass tone that you might expect
from a very professional session musician
wielding a vintage instrument and who
knows his performance needs to serve
the song, provide a solid foundation, and
not pull focus away from the other, more
prominent, elements in the arrangement.
The playing within the supplied MIDI
grooves suits the sound perfectly. The
grooves provide tasteful performances
that have plenty of character and feel but
without any flashy elements that might
pop out of the mix. As such, while it says
Americana on the ‘box’ (er... download),
these performances are hugely flexible
and would work in a diverse range of
musical contexts. If you like this groove
content, you can supplement what’s
included here with the additional purchase
of the Americana EZbass MIDI pack (which
complements similar Americana MIDI pack
add-ons for EZdrummer and EZkeys).
Having auditioned this alongside the EBX,
it provides more of the same — which is
to say it’s an excellent all-round addition
to your groove collection and suitable for
a whole host of musical styles.
Conclusion
While metal-heads might not feel the
Americana EBX is an essential addition
to their EZbass arsenal, I’d encourage
regular EZbass users in almost any other
style to give this expansion an audition. For
acoustic or electric pop, rock, folk, blues or
singer-songwriter — and, yes, Americana —
styles, this is exactly the kind of bass sound
and bass performance that will just work.
Solid, subtle, smooth and yet characterful,
Americana EBX is quietly, flexibly and
dependably brilliant.
summary
It might say Americana on the box, but this
is a very versatile addition to the EZbass
expansion series that is mix-ready for almost
any mainstream musical style.
$ $89
W www.toontrack.com
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 ON TE S T

Fracture Sounds
elevate the humble
cardboard box to
versatile percussion
instrument.
NICK MAGNUS

A
re you always
agonising over
whether to hold on to
those cardboard boxes from
Amazon, to recycle them or
because they might just come
in handy? The resourceful
chaps at Fracture Sounds
have found a most creative
solution: they’ve repurposed
theirs into a stonking 40-piece
percussion ensemble,
sampled at multiple velocities
and round robins. Huge, large,
medium, small and tiny boxes
FractureSounds
were variously played with
mallets, hands, sticks and
brushes, recorded in a concert
hall using top-of-the range
mics from three perspectives.
Box Factory
with the ‘Gritty’ patch, whose recordings
Sample Library
balance between initial transient and the tail of the
Whilst the natural recordings were processed using an “unusual” sound alters with varying amounts of Punch. Decay
sound jolly impressive, its processing chain (unspecified), including Trim is useful for shortening the hall’s ambience if it
pre-processed samples are a bass amp. The fourth Main Instrument becomes too messy during fast passages; the control
Box Factory’s superpower, patch, ‘Snappy’, takes the opposite works counter-intuitively — clockwise movement
taking the humble box to approach — shorter, brighter and more shortens the decay, with the full natural decay in the
corrugated heaven. transient, with a less roomy sound. fully anti-clockwise position. Saturate adds nice growly
The library comprises All four Main Instrument patches have distortion, as you’d expect — the clever part being that
seven NKI patches: the first identical controls that affect the entire every box articulation has its own Saturation module,
four ‘Main Instrument’ patches ensemble. Those providing perhaps the rather than applying the effect globally, the result
present the full ensemble most significant sonic variation are the being ‘clean distortion’ (I do love a creative oxymoron)
laid across the keyboard Close, Mid and Far mic levels. Below when everything is thundering away at the same time.
from low to high, culminating these is a horizontal Perspective slider,
with stick clicks at the very which alters the balance from all Close to
More Players Please
top. Each box is mapped all Far, the mid position favouring the Mid Ensemble Stack boosts the apparent number of
to adjacent pairs of white mics. You can of course set your own mix players, firstly with the Size control that layers
keys for easy two-fingered levels, and there are the usual solo, mute progressively more round-robin samples together.
playing. These four patches and unload buttons. The Mid and Far mics Looseness adjusts the timing spread between
share an identical feature set add the epic ingredient, revealing the ‘players’, whilst Spread applies randomised tuning
(discussed later), each patch hall’s powerful midrange and a substantial and panning to those layers for a more varied, wider
having a distinctly different bottom end — those boxes suddenly sound. This reminds me of similar features on Session
character. The ‘Natural’ patch become 10 times larger. Audio’s Taiko Creator and Drum Creator libraries,
is the least processed of the and it works well here. Spatialisation comprises
four, sounding closest to
Big Me Up Bigly a Stereo Width control (from mono to super-wide) and
the original recordings. The Further heft and sound shaping is Reverb amount, the reverb offering two algorithms,
samples for the ‘Aggressive’ applied with the Sculpt controls. Punch Hall and Room, accompanied by Size, Damping and
patch were pre-processed emphasises the attack transients; it Pre-delay controls.
with heavy compression sounds very much like Kontakt’s Transient
and saturation to bring out Master at work, although that effect was
It’s A Setup
the room ambience and not visibly evident on a quick inspection If the default keyboard layout of the Main Instrument
add “dirt and girth”. An even under the hood. I suspect some craftier patches doesn’t suit your needs, the Setup page
heavier-duty approach is taken process is going on, as the volume contains all you need to create your own. Any box’s

16 April 2022 / w w w. s o u n d o n s o u n d . c o m
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 ON TE S T
FRACTURE SOUNDS BOX FACTORY
The Rhythm Engine comprises two bar-graph step sequencers; the upper one sets each step’s velocity
whilst the lower one modulates the filter. The bars on the filter sequencer are bipolar, with the top controlling
the LPF and the bottom adjusting the HPF; combining both acts as a band-pass filter. In this example, we are
in Restart Sequence mode, so different notes triggered at different times (the yellow dots) chase each other
to produce more complex patterns. The number of steps for each sequencer can also be set independently.
articulation (mallet, stick, hand, etc) can be
mapped to any key — or a range (group)
of keys. A single articulation can even be
mapped to multiple key groups if required.
Why might you want to do that? Because
every key group boasts its own set of
parameters — Volume, Stereo Width, Pan,
Tuning, and most importantly, Low-pass
and High-pass filters — to further sculpt
whatever sound it contains. With these
tools a single box can be made to sound
like many different ones; the larger
boxes, for example, become Taiko-esque
monsters with a floor-shaking low end
by simply tweaking the tuning down and
attenuating their high ‘slappy’ frequencies
with the LPF. Conversely, squeezing out
all low frequencies with the HPF produces
a very convincing top-end clatter —
think hot rods on a blackboard. Custom
mappings can be saved as presets which
can also be loaded into any of the Main
Instruments patches.
Particular Patches
For the Designed Drums patch, Fracture
Sounds took the source samples and
mangled them with some seriously
weird processing to create an array of
distorted, squelching impacts that could
scarcely be identified as originating from
cardboard. Simultaneously aggressive
and cartoon-like, these can be sculpted
and customised with all the same features
as the Main Instrument patches, with
the exception of the mic mixer — this is
18 April 2022 / w w w. s o u n d o n s o u n d . c o m
a single-mic instrument.
The last two patches, ‘Booms and
Hits’ and ‘Whooshes and Risers’, deal
in cinematically flavoured one-shot
sounds, specially curated from the source
material by sound designer Karel Psota
and trailer composer Benjamin Squires.
The Booms and Hits deliver plenty of
low-end oomph and crunchy impacts,
certain to find their way into many a movie
soundtrack and OTT trailer. Whooshes and
Risers include some superb material just
perfect for those ‘shock’ moments — rolls
rising to an impact and weird, unnerving
reversed effects. The start and end points
and envelope contour of all one-shot
sounds can be individually adjusted via
a waveform display. Fractured Sounds
have even thoughtfully included a folder
of all these one-shot WAV files to drag
into your DAW as audio, if that method of
working suits you better.
Rhythm Ace
Perhaps the most intriguing and
creatively stimulating feature of the
Main Instruments and Designed Drums
patches is the Rhythm Engine. Based on
the ubiquitous Kontakt bargraph step
sequencer, this employs two separate
graphs, one for velocities and one for filter
values. The filter sequencer is unusual
in that the bars are bipolar; the top of
each bar controls the LPF, whilst the
bottom controls the HPF. The number
of steps (up to a maximum of 16) can be
set independently for each sequencer,
so the rhythmic and tonal combinations
are virtually endless; even a one-note,
one-step velocity sequence can sound
like a variety of different boxes with
creative filter sequence settings. The
more notes you play simultaneously,
the more instruments join in to make
a gloriously thunderous racket. The best
bit, however, is when playing notes in
‘Restart Sequence’ mode, where each
new note played starts at the beginning
of the sequence. With careful note timing,
different instruments’ rhythms become
offset against the others to produce
complex patterns and, as the filter
sequencer also resets to its beginning,
some beautifully complex interweaving
tonalities. Setting different numbers of
steps for velocity and filter makes for even
greater complexity, where the pattern
seems to be constantly evolving. Two
Humanise controls randomise velocity
and timing; in the case of timing, those
randomisations are applied differently to
each instrument, adding to the illusion of
individual players with their own subtle
timing variations. The overall dynamics of
a sequence can be further controlled in
real time via the modulation wheel, from
full blast down to the gentlest tickle. If
I could request two improvements to the
Rhythm Engine, they would be to increase
the number of steps to 32, and to add
a Swing control. Pretty please?
Final Thoughts
I can’t recommend this library highly
enough. Big Taikos and toms are great,
and have their place, but Box Factory
offers a unique freshness of sound that
delivers the same power, and finds its
place in a mix without bulldozing the
rest aside. There’s no PDF manual, but
if you need help the Fracture Sounds
website has plenty of useful information,
in particular two excellent walkthrough
videos by Box Factory’s inventor Will
Bedford that explain every detail more
succinctly than any manual. Box Factory
requires the full version of Kontakt or the
free Kontakt Player, v6.6 or later.
summary
The humble cardboard box — or rather
a whole host of them — becomes a dynamic
percussion ensemble packed with macho grit
and attitude.
$ $119
W www.fracturesounds.com
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 ON TE S T

ChaseBliss Audio
CXM1978
Reverb Stompbox
This modern take on the
Lexicon 224 may be a pedal,
but it has applications that go
way beyond guitar.
SIMON SMALL

T
he CXM 1978, a joint venture
between Chase Bliss Audio and
stompbox pioneers Meris, is based
on a classic studio reverb processor: the
Lexicon 224 which, along with its instantly
recognisable white LARC remote controller,
started to appear in recording studios in
1978. I work in a studio that has a vintage
224 and its appeal to me is obvious.
But while the CXM 1978 does set out to
replicate that sound, there are also some
welcome modern touches, including many
features not found on the original device.

Overview
Those familiar with the brand will recognise
Chase Bliss’s Automatone format from
their preamp collaboration with Benson
Amplifiers. It’s an elegant design, this
time with a white/cream body and
complementary side panels, and as with
all Chase Bliss Audio products I’ve used
it’s sturdy and has been built to a high
standard. As well as the footswitches, there
Chase Bliss Audio
CXM 1978
$899
PROS
• Sound quality is exceptional.
• Very reminiscent of the vintage unit.
• MIDI/Exp/CV control makes this a
versatile pedal.
• 30 preset slots.
CONS
• Not exactly an impulse purchase!
• Reverb drops when changing
between modes.
SUMMARY
The CXM 1978 offers all the lush
charm of the Lexicon 224 but adds
lots of modern niceties too — all in
a convenient box that works as well on
the desktop as on a pedalboard.
are plenty of finger-friendly controls and
it caters for balanced line-level I/O, with
separate TRS jacks for the left and right
channels. The whole package is only
marginally wider than a LARC too, and
it will just as happily sit on a desktop or
mixing console as it will on a pedalboard.
The pedal accepts a 9V centre-negative
DC input (250mA), but internally this is
stepped up to provide ±15V power rails.
Probably the first thing you notice is
the bank of six faders, for which there’s
an optional dust cover/protector. They
feel gorgeous, and are motorised and
fully automatable; they react to preset
changes, MIDI, control voltage and
expression pedal inputs, and give you
a helpful visual indication of what’s
happening behind the scenes.
Chase Bliss provide an extensive list
of CC messages for MIDI, control voltage
and expression, so that side of things
is incredibly easy to configure, and the
full-size MIDI In and Thru DINs make it
easy to hook up. I used an LFO on my
Empress ZOIA pedal to control the reverb
parameters, but it could just as easily
work with a MIDI switching system or be
controlled by your DAW. The pedal can
store/recall up to 30 presets spread over
several banks, all of which are accessible
via the footswitches or MIDI.
The CXM 1978 has an interesting
approach to controlling the reverb time.
Rather than offer a single decay control, it
adopts a split-band approach with EQ-style
and crossover controls. Adjusting these
faders does more than you might think: as
Chase Bliss put it, they allow you to “set the
absorption of the walls that make up the
reverb tank”. In other words, just as with
an acoustic reverb, the tone and decay
behaviour are intrinsically linked.
Cross sets the crossover point (from 0Hz
to 1.7kHz) for the Bass and Mids controls:
everything above the crossover is tweaked
by the Mids control and everything below it
by the Bass. In use this is fantastic: it makes
it really simple to tailor different decay times
and behaviours for the Mids and Bass.

20 April 2022 / w w w. s o u n d o n s o u n d . c o m
 ON TE S T
C H A S E B L I S S A U D I O C X M 1978
It’s very handy if you want to cut low-end
reverb on busier sections of a track, or
boost them to create a pad of bass reverb
without muddying up the mids. The Treble
control seems to be more than an EQ too,
and interacts with the Bass/Mids behaviour.
Set at its lowest, Treble will darken the
reverb (“increasing absorption in the reverb
tank”) and, as you’d expect, things get
progressively brighter as you bring it up.
But even with Mids and Bass at maximum
the decay time will remain fairly short while
Treble is turned down.
There’s also a Mix fader, which blends
between the 100-percent wet (reverb)
and dry (input) signals, and another for
Pre-Dly, which sets the amount of pre-delay
(the delay after the dry sound before
the reverb begins).
Clocking On
The overall length of the delay and fidelity
of the effect are determined by the Clock
switch which, like the others here, is
a stylish arcade-style metal button with
bevelled surround. This one offers three
Clock modes: Standard, Hifi and Lofi.
Standard is billed as the classic 224 mode,
and uses a 24kHz sample rate with 16-bit
conversion, and offers a maximum 168ms
pre-delay. Hifi is a cleaner, more up-to-date
variation on the same theme, using a 48kHz
sample rate and 32-bit conversion, with
a maximum 42ms pre-delay and, naturally,
a lower noise floor.
While those modes are great they’re
probably just the sort of thing you’d expect
of a ‘modern 224’, but Lofi offers something
a little different. As the name suggests,
this can be used to degrade your signal in
a deliberate way, and it works in a similar
way to a bucket-brigade delay: the Pre-Dly
fader acts like a BBD’s time knob: the
higher you set this the more your signal
will fall apart, with the sample rate varying
between 48 and 2.4 kHz. It operates at
16-bits and offers a maximum pre-delay of
1.7 seconds.
All three modes sound great. I’ve tended
to end up working most of the time in
Standard mode, though perhaps that’s
a matter of familiarity as much as anything.
The team really seem to have captured
the essence of the 224 in Standard mode:
with its warm tone, and imperfect musical
nuances from the converters; it feels to
me like it’s all there and is certainly very
reminiscent of the vintage 224 in the studio
where I work. But I’ve also frequently
wanted to use Lofi mode, which is great for
adding in glittering shards of degradation.
22 April 2022 / w w w. s o u n d o n s o u n d . c o m
This mode worked especially well for me
when I patched in a modulation source
to the Pre-Dly fader/parameter; using an
expression pedal for this was so much fun!
The Type button gives you access
to the three different reverb algorithms:
Room, Plate and Hall. With the tap of
another button you also have control
over the reverb’s diffusion (low, medium,
high); this sets how ‘smeared’ the attack
phase of the reverb is. Yet another button
is Tank Mod, which determines the rate of
modulation applied to various parameters
— I love modulation on reverb and the CXM
1978 did not disappoint. Low is dialled back
to the point where it just adds a feeling of
movement, whereas Medium and High take
you into territory closer to a chorus or even
Leslie speaker effect.
For guitar — this is a pedal, after all!
— I generally tend to gravitate towards
plate reverbs. But while the CXM 1978’s
Plate is lovely, I actually found that
I frequently preferred the Hall algorithm;
having the ability to sculpt that sound using
the ‘EQ’ faders made it really usable. It’s
easy to create dreamy, ethereal spaces
while controlling the low-end that can
otherwise overwhelm a guitar amp/cab.
The Room algorithm is fantastic for putting
your dry signal in a complimentary space
and blends right in, without any of the
obvious boxiness some pedals’ Room
settings create.
I want to make a point of mentioning
just how beautiful this reverb can be as
an aux effect for mixing. I recently mixed a
folk album that included a trumpet player,
and their instrument was recorded at close
proximity so sounded pretty dry. I sent it
out to the CXM 1978 on the Plate setting,
looking for a little ambience to fill out the
sound, and within moments I had a perfect
result dialled in — the faders make it so
immediate and intuitive to tweak — and
I found Standard mode suited this particular
The CXM 1978 can operate with stereo
balanced line-level I/O and supports MIDI, CV
and expression pedal control.
track, with its vintage feel. I was printing the
wet signal back to the DAW before I knew it!
One slight misgiving is that when you
change your reverb type, whether you
use a button on the pedal or a MIDI signal,
there’s an audible drop-out while the pedal
changes settings. Typically this isn’t a huge
problem when programming, recording or
practicing. However, one of the obvious
attractions of a compact pedal is that you
can take it on stage, and when playing live
this can be much more annoying — a ‘trails
on’ mode to allow the decay of one setting
to continue while you smoothly change
patches would be a big improvement.
Verdict
Overall, then, I am a big fan of the CXM
1978. When thought of as a guitar pedal
the price might seem pretty steep, and
I suppose that might put some people off.
But with stereo balanced line-level I/O
on board it is not at all limited to use with
guitar, and when you compare its price with
that of rack units of similar stature it makes
much more sense. There are now some
brilliant reverb plug-ins with Lexicon vibes
(Valhalla’s VintageVerb, Relab LX480 and
UA’s 224 emulation for the UAD platform,
for example) but most focus on recreating
that device’s sound and don’t tend to
try very hard to deliver much different.
Because of this, the CXM 1978 felt more
interesting to me, and its physical form
makes it quick and easy to use. If you are
willing to invest in this pedal, then, you will
be treated to exceptional sound quality and
flexible control over all parameters. It would
be a wonderful addition to any pedalboard,
synth or studio mixing setup.
$ $899.
W www.chaseblissaudio.com
 ON TE S T

DryBell Vibe Machine v3

Guitar Effect Pedal self-explanatory, as is the
Chorus/Vibrato switch, with
One of the best Uni-Vibe the latter just removing the
clones on the market just dry signal from the mix. The
got an update. tweak-pots are where it gets
interesting, giving you access
to the input impedance
DAV E LO C K WO O D
(match the Original circuitry,

O
riginating in Japan in the late Bright, or a Custom setting);
’60s, the Uni-Vibe Chorus chorus intensity/warmth; the
Vibrato was the first standalone ‘shape’ of the ‘throb’, in the
modulation pedal. In a world only just form of Range and Symmetry
getting used to fuzz and wah-wah pedals, controls; and the splendidly
it offered something truly different in named Grit, which seems to
guitar tones. An asymmetrical throbbing be a low-pass filter. That’s
amplitude and tonal modulation that is already a lot to play with,
a bit like a phaser and a bit like a chorus, then, but we’re not done
but without being identical to either, the yet because there’s also
Uni-Vibe sound will be known to many an external pedal input. You
as the effect used by Jimi Hendrix in his can calibrate a wide range of
iconic rendition of ‘The Star Spangled passive expression pedals
Banner’ at Woodstock. to operate the Speed
The Uni-Vibe pre-dates the invention control, like an original
of the bucket-brigade analogue delay Uni-Vibe, or use a footswitch
chips that would facilitate the creation of (latching or momentary) to
the first Chorus pedals, notably the Boss switch between two different
CE-1, so despite its Chorus and Vibrato modulation speeds, complete
setting options, its circuitry has far more with Leslie-style ramping up
in common with a phaser. The Uni-Vibe and down if you want.
may have been created to offer keyboard In an era of do-it-all
players a compact alternative to the modulation effects, not
bulky Leslie mechanical rotary speaker everyone needs or wants
effect, but it would be guitar players the very specific palette of
that would really exploit its potential Uni-Vibe tones. But when
to the full. extensive customisation via an array of miniature you do there may be no
The original Uni-Vibe used a lightbulb trim pots, as well as its fundamentally authentic more convenient way of
surrounded by four light-dependent sound. So, what could be special about a v3 in which getting there than with
resistors and a lot of discrete-transistor nothing much has changed? a Vibe Machine v3: the
circuitry, and any attempt to make The short answer is the sound. A less classic sound in a compact
a Uni-Vibe clone without a similar optical mid-focused modulation leaves the overall spectrum enclosure, with true bypass,
element tends to lose the unique shape less altered than with a v2, at the same time as remote control and almost
of the modulation and thereby an making more depth and ‘swirl’ available overall. infinite tweakability.
essential flavour of the sound. I prefer a Uni-Vibe placed before amp or pedal
Croatia-based company DryBell made distortion, particularly with this one, but with so much
summary
their first Vibe Machine photocell-based tweakability available, commendably all available
Uni-Vibe-style pedal just over 10 from the outside of the unit, it would be a mistake to When originals and clones
years ago, and recently celebrated assume that this version can’t be made to sound like can vary so much, choosing
a Uni-Vibe-style pedal is a highly
that milestone with the release of the almost anything you want in the Uni-Vibe spectrum.
personal and subjective matter
latest v3 iteration. Most major pedal Of course, the flipside of wide-ranging adjustability — but I’ve yet to hear one better
manufacturers and a number of smaller is that you can very easily have it sounding really than this.
boutique ones now have a Uni-Vibe-style not like a Uni-Vibe at all. It is a good idea to mark the
offering in their line-up, many of them positions of the tweak pots before you depart from $ $319.
very good, but DryBell’s v2 iteration the factory settings! E info@drybell.com
was certainly amongst the best, offering The two big controls, Speed and Intensity, are W www.drybell.com

24 April 2022 / w w w. s o u n d o n s o u n d . c o m
 ON TE S T
Audio Imperia Solo
Sample Library
A new collection adds
expressive solo instruments
and voices to the
company’s orchestral range.
DAV E ST E WA RT
N
ew from the team that brought us
the Jaeger orchestral collection,
Talos epic brass ensembles, Areia
cinematic strings, Cerberus percussion
and Nucleus symphonic essentials comes
a collection of solo instruments designed
to add lyrical expression and emotion to
your music with a minimum of fuss. The
library contains three solo strings, five
solo woodwinds, three solo brass and
two female sopranos, giving you a baker’s
dozen of melodic top-line timbres. In
26 April 2022 / w w w. s o u n d o n s o u n d . c o m
a piece of product labelling which should
satisfy the most stringent advertising
standards, it’s titled simply Solo.
This all-new collection was recorded
in conjunction with Budapest Scoring,
previously suppliers of brass and strings
for Audio Imperia’s production music
label AI Music. The sample recordings
took place over a period of a month
at Budapest’s Pannonia scoring stage,
a facility which specialises in orchestral
film score recordings.
It wasn’t meant to be this way.
According to Audio Imperia’s Jan
Hoeglund, the original plan was to spend
the time working on the keenly anticipated
choir project Chorus (another masterpiece
of honest product naming) in collaboration
with his pal Jasper Blunk of Performance
Samples, a company with a distinguished
pedigree in sampled choirs. Unfortunately
the pandemic buggered that up. Hoeglund
explains: “It would not have been
responsible to put that many people in
a room together at that time. And even
with the soloists, we took every precaution
possible; Covid tests, masks, social
distancing, etc. It was quite a challenge!
It just wasn’t safe to travel at that time, so
the sessions were all remote.”
Overview
Solo focuses on a specific expressive
lyrical style in which a slight crescendo
swell is combined with progressive
vibrato — in plain English, notes start out
straight, then vibrato is added after one
or two seconds and persists throughout
the note’s duration. This delivery was
captured in two true legato styles and
looped sustains, augmented by spiccato/
staccatissimo short notes, pizzicato and
tremolo (strings only) and woodwind trills.
Meanwhile, the singers keep it simple and
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 ON TE S T
AUDIO IMPERIA SOLO
The Advanced page shows the microphone
mixer and instrument transpose/range controls.
Real-life ranges are marked in blue with stretched
low and high notes marked in yellow. Note the
velocity-switch ‘combined legato’ patch in the top
slot — you can uncouple the two artics by clicking
on the small orange icon on the right.
stick to legato and sustained ‘ooh’ and
‘aah’ vowels.
Only first chair players and actual
soloists were booked for this project. The
musicians worked from scores and were
also given verbal instructions (beamed in
via Zoom from a safe distance, I imagine)
to emphasise the musical effect the
producers wanted to achieve. Happily for
all concerned, the soloists and studio staff
appear to have emerged unscathed from
the sessions, and their clear, unmuffled
sound suggests that the wind players
and sopranos removed their masks
before recording their samples. Solo
runs on Kontakt and the free Kontakt
Player version 6.5.2 or higher, and is
83.6GB installed.
Solo Strings
Played by the versatile Csongor Veér
(who took the role of doomed bandleader
Wallace Henry Hartley in the 2012
Titanic TV mini-series), the solo violin
epitomises Solo’s musical and technical
scope. The instrument’s ‘slurred legato’
28 April 2022 / w w w. s o u n d o n s o u n d . c o m
style features small, graceful portamento
slides between notes, while the rebowed
legatos offer a more distinct attack.
‘Legato combined’ multi-patches let
you switch between the two artics via
velocity or MIDI CC — alternatively,
you can decouple the patches and
use keyswitches to select the one you
want. Either way, switching between
these two legato styles on the fly
produces impressively organic-sounding
melody lines.
Mr Veér‘s violin has a robust tone with
plenty of body and presence. Played
with a fairly strong vibrato, its legatos
sound romantically expressive, great for
impassioned cadenzas, heart-tugging
sentimental melodies and wild Romanian
folk workouts. There’s no non-vibrato option,
so if you’re looking for a plain, astringent
violin sound you’re out of luck. Other artics
include superior biting spiccatos executed
with slide-rule precision, expressive sustains
and nicely tuned tone trills which sound fine
when played chordally — players often pay
insufficient attention to the tuning of the
upper notes in their trills, but these are right
on the money.
The excellent solo cello is a highlight
of the library, combining a dignified,
august and stately tone with a highly
expressive legato style and sustains
Audio Imperia Solo
$299
PROS
• A fine collection of sensitive,
expressive performance by 13
hand-picked musicians.
• Includes solo strings, woodwinds,
brass and two soprano singers.
• Features two highly effective legato
modes and excellent short note
deliveries.
• The simple layout of the GUI is
a pleasure to work with.
CONS
• No dedicated manual.
• There’s no way of controlling the
amount or timing of the vibrato.
• The vocal samples’ volume levels
need some work (NB not the
singers’ fault).
SUMMARY
Audio Imperia’s Solo adds a fine set of
solo performances to their orchestral
catalogue. Comprising 11 string,
woodwind and brass solo instruments
and two soprano singers, the library
avoids loud, overbearing sonorities
and focuses instead on lyrical and
expressive performances which
will grace quieter musical settings.
Featuring excellent true legatos and
zippy short notes, it’s a versatile,
easy-to-use collection capable of
adding subtle expression and musical
detail to your scores.
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 ON TE S T
AUDIO IMPERIA SOLO

Solo’s 13 pitched
pads share a majestic
hybrid orchestral sound.
On-screen controls
allow you to adjust their
start points and reverse
playback direction, as
well as automating filter
and volume settings.

which work
well for dark
atmospheric chords.
I occasionally
wished its top note
of C5 could be
extended, so was pleased to see Solo
allows you to edit instrument ranges —
however, when stretching their upper
extremities you might notice a quickening
of the vibrato caused by the sample
tuning hike!
I had fun playing melancholy autumnal
chords on the solo viola’s sustains patch,
and also enjoyed its cute pizzicatos,
emotional ‘slurred legato’ portamento
slides and vigorous spiccatos. The latter
short-note style is brilliantly performed
by all three solo strings: positive,
quick to speak, instantly responsive to
touch and ideal for rhythmic ostinatos,
these are a good supplement to
the excellent ensemble spiccatos in
Audio Imperia’s Nucleus.
Solo Woodwinds
Enter the woodwinds, headed by Anita
Szabó, principal flautist of the Hungarian
National Philharmonic Orchestra. Her
performances are another highlight,
featuring a gorgeous lyrical flute tone
which works wonderfully for slow, dreamy
melodies. Discreet expressive swells
add subtle motion to melodic lines,
tightly played staccatissimos are lightly
propulsive, and fast, liquid-sounding
trills create bursts of swirling orchestral
colour when you
play close-voiced
chords. A great
resource for
pastoral melodies
and lively, piping
piccolo-like parts,
the player’s sustains also work a treat
for three-part flute chords, a classic
orchestral sonority.
This stellar instrument is joined by
a super-smooth, mellifluous solo clarinet
played in the traditional vibrato-free
orchestral style. Its long notes
consequently sound unwavering and
somewhat dispassionate, with expression
limited to pronounced crescendo swells
which can get overbearing in the high
register. On the plus side, the player
performs perky staccatos and animated
trills which automatically end with
a staccato iteration, great for creating the
dancing woodwind figures heard in the
opening of John Williams’ ‘Adventures on
Earth’ (from the movie ET).
Normal vibrato service is resumed
by the library’s oboe. Performed with
a plaintive, subtly swelled delivery, it’s
a serviceable instrument for expressive
lead lines, and its unmistakeable
cutting tone is a useful texture when
constructing woodwind chords. The
player (a gentleman named Fruzsina
Káli-Fonyódi, oboe teacher at Budapest’s
prestigious Liszt Academy) also whips out
some rather alarming semitone trills.
While the oboe sounds lyrical
and a little sad (in a good way), the
lower-pitched English horn has a more
serious, pensive atmosphere. This is
a lovely instrument — mellow-toned yet
clear and tuneful, its highlights include
great staccatissimos and well-tempered
tone trills in the manner of the solo violin.
Accompanying these reedy top-liners is
a solo bassoon distinguished by a languid
legato delivery and an evolving slow
vibrato ideally suited to slow-moving
high-register melody lines. A pleasant,
reflective reedy timbre for low-pitched
sustains, this versatile bassoon’s fine
staccato short notes also tick the jokey
‘Tears Of A Clown’ box.
Solo Brass
In over 20 years of orchestral sample
reviewing this is the first time I’ve heard
a descant horn. A smaller version of the
French horn, the descant horn plays an
octave higher than its big brother and
is often brought into sessions by horn
players to cope with the unfeasibly
high parts inexperienced composers
sometimes throw at them.
With this unusual instrument,
high-pitched sustained chords take on an
eerie, almost distant quality. Its sound is
not immediately identifiable: on hearing
is very high top note (A5) you might

The library’s suite

of effects (available
only for its pads)
range from subtle
chorus and delay to
obliterative distortion
and bit-crushing.

30 April 2022 / w w w. s o u n d o n s o u n d . c o m
 ON TE S T
AUDIO IMPERIA SOLO
Recordings took place at
Budapest’s Pannonia Studios.
think you’re listening to a clarinet, but as
the pitch descends a delightfully pure,
warm, classic horn tone emerges. Long
notes with built-in small crescendos are
delivered with unswerving precision;
the subtle distinction between fingered
and retongued legatos is an important
expressive asset, and the combined
legato multi-patch once again offers great
potential for enhanced melodic realism.
Also included in Solo is a full-sized
French horn. The player handles the
library’s trademark smooth, expressive
legatos well and maintains a good
performance standard apart from a slight
wobble on the front of his low D3 long
note. Though the library’s stylistic ethos
rules out over-the-top ‘epic brass’
sonorities, the satisfyingly brassy timbre
of the horn’s energetic loud staccatos is
a good bet for rousing chord stabs.
Rounding off the small brass section
is a solo trumpet played with a very
light vibrato, a break with orchestral
tradition. Its carefully controlled legatos
benefit from clean note attacks and
steady, well-tuned sustains. In keeping
with the house style, the delivery never
gets aggressive, so if you’re in the
Built-in
ADSR controls
give you control
of instruments’
attack speed and
release time.
32 April 2022 / w w w. s o u n d o n s o u n d . c o m
market for blazing cinematic fanfares or
a dominant lead voice in a hot soul brass
arrangement, this is not the trumpet
for you. Nonetheless, it’s a pleasant,
sensitively played instrument which
would work well in exposed, sensitive and
lyrical settings.
Solo Singers
Though it’s not entirely clear from
despatches, I’m guessing that Solo’s
so-called ‘angelic’ (hate that word) and
‘operatic’ sopranos are respectively
Orsolya (Orsi) Sapszon and Katalin
Vámosi. Although both are classically
trained, online videos show the first
singer working in a broadly pop vein
while the second seems to do more
Articulations
Solo’s instrumentalists and vocalists perform
a small menu of essential articulations.
• Strings: legato rebowed/slurred, sustained,
spiccato, pizzicato, tremolo, trill (half/whole
tone).
• Brass: legato fingered/retongued, sustained,
staccatissimo.
• Woodwinds: legato fingered/retongued,
sustained, staccatissimo, trill (half/whole tone).
• Singers: legato aah/ooh, sustained aah/ooh.
The library’s legato performances were played at
a single dynamic level — the makers’ explanation
is that crossfading between multiple dynamic
layers can introduce undesirable artefacts, so
to maintain continuity of tone they limited the
classical gigs. Apologies if I inadvertently
switched their credits, but rest assured
both achieve a high professional musical
standard regardless of the style tags.
Both sopranos perform a single legato
style and long notes using the vowels
‘aah’ and ‘ooh’. Orsi (if it is she) utilises
an agreeably plain, unmannered style
with very little vibrato, delivering smooth,
flowing legatos and steady, true sustains
which create an almost synth-like timbre
when layered in high-register chords.
There’s a huge volume disparity between
this singer’s piercing top C6 and delicate
bottom G3 note, which makes it very
difficult to use her full range — turn
down your speakers to prevent the high
note shattering all the glassware in your
legatos to one dynamic layer. By contrast, short
notes have up to five dynamics and incorporate
five round-robin variations.
In practice, the single velocity layer is not
a handicap, and it certainly avoids the tuning
and phasing issues which sometimes arise in
dynamic crossfades. Since the legatos’ and
sustains’ volume is controlled by the mod wheel,
you can program additional expressive swells
and diminuendos to your heart’s content. As
in Audio Imperia’s Nucleus, the legato patches
are super-smooth and agreeably playable, and
although the legato transitions are optimised
for slower, lyrical and expressive melodies, you
can adapt them to faster passages by setting
the sample start to around 60 milliseconds, thus
getting notes off to a flying start!
 ON TE S T
AUDIO IMPERIA SOLO
Solo Legato
Solo features two different legato styles:
rebowed or slurred for strings, and fingered
or retongued for woodwinds and brass. All
instruments have a dedicated patch for each
style, but for users’ convenience the makers
also combined the two into a single playable
unit with no need to keyswitch. Though it’s
sometimes hard to hear the difference between
the two legato artics, I found that judiciously
using both in a melody has clear benefits for the
expression, fluidity of line and realism of these
solo instruments, particularly the woodwinds
— once you’ve got the hang of the facility, you
won’t want to stop using it!
As is common in orchestral libraries, legato
patches default to monophonic playback
house, and the low G becomes inaudible.
I realise singers’ low register pitches are
a lot quieter than their high notes, but it’s
a simple matter for a programmer to even
out the volume of individual samples
within a patch. This massive mismatch
should have been corrected at the
programming stage.
Though not a big
opera fan, I enjoyed
“There will always be quiet passages where a solo
the purity of the instrument can make its voice heard. These are the
second performer’s
‘oohs’, which sound moments Audio Imperia’s Solo is designed for.”
wonderfully ethereal
in the high register. The developing
operatic vibrato kicks in at around two
seconds and gets quite strong, but unlike
with some classical singers, you can still
hear which note is being performed!
Some inter-sample low register volume
disparities are apparent, but you can
probably iron them out with CC1 volume
automation.
Sound Design & Pads
Solo’s sound design section contains
pads derived from each of its 13 sound
sources and a bunch of effects to apply
to them. Despite their disparate origins
the pads are surprisingly homogenous
— all are pitched, well-tuned and
benefiting from a grand, subtly evolving
quasi-orchestral sound, so you can
use them for virtually any style of
music. I enjoyed the ethereal, mystical
atmosphere of ‘Parlare’, the subtle
textural undercurrents of ‘Crepusculo’
and the denser, more threatening
‘Luminosa’. If I needed a bass drone I’d
go for ‘Ecliptic’, a heavenly horns pad with
a solid bottom end.
The sound design bit really kicks in
when you dial up the effects, which are
34 April 2022 / w w w. s o u n d o n s o u n d . c o m
(meaning that although the sound remains
stereo, you can only hear one voice at a time).
If you want to hear two simultaneous legato
parts operating in a single patch, you can
select polyphonic legato mode — this creates
a velocity split so that (for example) notes above
velocity 64 trigger one legato line, while notes
with a velocity less than 64 trigger a second
independent legato line. You can adjust the
velocity split point to taste.
One caveat: simultaneous legato parts are
CPU-intensive. I found that playing the two
soprano’s legatos in two-part harmony generated
loud clicks on note transitions which only
disappeared when I increased my soundcard’s
buffer settings from 512 to 1024 samples.
only available for the pads. Some are
frighteningly transformative — moving
swiftly past the usual suspects (delay,
reverb, chorus, rotary speaker, phaser,
flanger) we find a powerful distortion
effect introducing an overdriven
electronic edge. Add the lo-fi effect, and
you can bit-crush your mangled hybrid
orchestra into oblivion — who knew
a beautifully recorded bassoon could
mutate into something resembling the
snapping of twigs or the horrid crackling
and spluttering of a faulty pre-war
Soviet radio? Trent Reznor might enjoy
it (I certainly did), but if you’re trying to
snag your first Hollywood film scoring gig
it’s probably prudent not to include such
deranged sounds.
Space & Time
Though spaces such as Air Studio’s
Lyndhurst Hall add an impressive natural
reverb to recordings, the medium-sized
Pannonia scoring stage used for Solo
was chosen for its more intimate
‘close-up’ sound. The samples were
recorded from two close spot positions
and a Decca Tree, with outrigger and
far mics picking up more distant room
positions. While the latter mikings’
real-life ambience would be helpful
in a surround mix, they don’t sound
particularly reverberant.
In the GUI’s Basic page, a single Easy
Mixer slider allows you to quickly fade
between close and far mic positions, and
as in Nucleus (reviewed in SOS February
2020) you can choose Classic or Modern
mix presets — for this library, I opted
for the brighter, louder sound of the
latter. Additional on-board room and hall
reverbs are also available.
Another useful facility carried over
from Nucleus is the sample start feature.
In the interest of realism the producers
left the natural attack of samples
untrimmed, but when it comes to rhythm
parts it’s essential that string spiccatos
and wind staccatissimos speak quickly.
With that in mind the makers include
a ‘tight’ button which automatically trims
the sample fronts, a great help when
programming rhythmic ostinatos. The
degree of tightness can be altered in the
GUI’s Advanced page.
Solo’s focus on core melodic
instruments inevitably means that
some common orchestral timbres are
absent — there are no supplementary
instruments such as piccolo or bass
clarinet, and percussion was always
a non-starter.
The most notable
omissions are solo
trombone and solo
tuba. While that’s
understandable
in this context,
completists might bemoan the fact
that no Audio Imperia library currently
includes that particular pair of solo
brass instruments.
Conclusion
The crash-bang frenzy of action-oriented
media productions leaves little space
for reflection, but in the wider world of
music there will always be quiet passages
where a solo instrument can make its
voice heard. These are the moments
Audio Imperia’s Solo is designed for.
The emphasis throughout is on lyrical
expression, and the 13 musicians have
risen to the occasion with a fine set
of sensitive performances. As hinted
previously, if you’ve been hired to
compose the soundtrack for Death
Race 2030 it would be advisable to look
elsewhere for your samples, but when it
comes to more tender, nuanced musical
settings and delicate atmospheres
requiring a subtle expressive sensibility,
these soloists come through with
flying colours.
$ $299
W www.audioimperia.com
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 ON TE S T
ARP2600 M
Analogue Synthesizer
Back by popular demand: Korg have miniaturised the ARP 2600
but kept the spirit and sound of the original very much alive.
GORDON REID
I
’m sure that there have been millions of
words written about the ARP 2600, and I have
probably contributed more than my fair share
over the years. But the world never seems to get
tired of what might be ARP’s finest creation. It’s
not as impressive as the ARP 2500, nor is it as
convenient as the Odyssey, and it’s certainly not
as easy to use as the ProSoloist, but it embodies
a wonderful compromise of the impressive, the
convenient and the straightforward.
Now, having secured the blessings of David
Friend, co-founder and eventual President of ARP
Instruments and one of the designers of the original,
Korg have recreated the 2600 not once, but twice.
I reviewed the first version, the ARP 2600 FS, in
late 2019 and fell in love with it. I don’t think that
I’ve touched my original ARP 2600P since. This is
because, after I returned the review unit to Korg,
I was unable to stop myself from ordering a new
FS which, as well as being perfectly calibrated, is
clearer, brighter, quieter, and hasn’t suffered from
a crackly or intermittent connector in two years.
ARP 2600 M
$1729
PROS
• Yet again, it really is an ARP 2600.
• It sounds authentic, but simultaneously
cleaner and brighter than the original.
• It offers several improvements over
the original.
• It’s MUCH cheaper than a vintage ARP 2600.
CONS
• Authenticity cuts both ways: it has some of the
same deficiencies as its forebear.
• With no equivalent to the 3620 keyboard,
some important functions are missing or hard
to access.
• The MIDI spec is very basic.
• It uses an external AC/DC power supply.
SUMMARY
The ARP 2600 remains one of the most desirable
monosynths ever built, offering a degree of
flexibility that was mind-boggling in 1971 and
remains uncommon to this day. The ARP 2600
M recreates it with great accuracy, but also offers
a number of valuable improvements. You’ll have
to decide whether the loss of the 3620 is an
issue for you but, either way, there’s a great deal
to like here.
36 April 2022 / w w w. s o u n d o n s o u n d . c o m
It was perhaps no surprise when the FS sold
out almost immediately, nor when — a year
later — Korg announced that they would be
shipping a smaller, more affordable and more
obtainable version, the ARP 2600 M. This has
now arrived. So, has Emperor Zorg’s malevolent
Shrink’O’Matic harmed anything, or has it
simply made the 2600 more convenient and
more attractive to a wider spectrum of potential
owners? Let’s find out.
A Miniature 2600 FS?
When I opened the box from Korg, I had
a surprise. Rather than discovering the packaging
protecting the 2600 M, or perhaps an inner box
containing the packaging containing the 2600
M, I found myself removing a small suitcase that
contained the synth itself together with its power
supply, a manual, a sheet of ARP stickers, a packet
of patch cables and a Korg microKEY USB
keyboard (the last of which isn’t part of the M, but
an optional accessory) all ready for use. It’s a very
unusual approach, but I have to admit that I was
impressed. This was a good start.
The M lacks the wooden case of its larger
sibling, which means that it also lacks the lid that
protects an original 2600 or the main cabinet of
the FS when they’re not in use. Instead, it has
a wrap-round steel case with all of the controls on
the front and various connections on the left-hand
side. Nevertheless, you would have to look hard
to distinguish between the front panel of the M
and that of a shrunken FS because there’s just
one difference: the multiple has shrunk from four
sockets to three. It’s easier to spot the changes
when comparing it with the original — outputs
for CV2 (the uppermost note played) and the
modulation CV (MIDI CC01), the addition of the
ARP4012 (1971-1975) or ARP4072 (1976-1981)
filter selector switch, and the loss of numerous
calibration trimmer access points. In addition, the
M uses Odyssey-style slider caps whereas the
FS retained the larger caps from the original. This
makes perfect sense in the context of the smaller
panel but, given the 40-percent shorter throw
of the faders and the loss of the little pointers
on the original’s caps, this makes it harder to
draw and recreate sounds when using printed
patch sheets or the time-honoured method of
making chinagraph marks on the panel itself.
w w w. s o u n d o n s o u n d . c o m / April 2022 37
 ON TE S T
ARP 2600 M
Such differences aside, the M is clearly
the same synthesizer as the 2600 and the
main cabinet of the FS; even the elliptical
speakers have been retained, although
these are smaller than before. It would
have been great to see these replaced
with extra VCAs, mixers and multiples, or
even something akin to the excellent CMS
upgrades from Phil Cirocco, but maybe
that would be hoping for a bit too much.
Oh well... one can dream.
The biggest functional change
between the M and the FS is, therefore,
the omission of the 3620 keyboard. This
means that (in addition to the loss of the
pressure-sensitive keyboard itself) the
M lacks the FS’s LFO, its arpeggiator/
sequencer with its associated Latch switch
(which means that Gate Repeat has gone),
its two CV inputs, its single/multi triggering
switch, its two-octave up/down lever, and
the ability to control portamento.
The loss of the LFO places the M — like
the original 2600 with a 3604P keyboard
— firmly in Minimoog territory; unless you
jump through some significant hoops,
und, or
you can have a triple-oscillator sound
you can have a dual-oscillator sound with
a conventional LFO. Its loss also makes
m the
creation of common sounds that use cyclic
PWM almost (but not quite) imposssible
unless you employ external modules.
What’s more, the lack of a patchab ble VCA
means that you have to use the Ring
Modulator as a VCA if you want to control
the modulation depth using a wheel.
In contrast, while the M appears to
have lost its single- and multi-triggering
options, it hasn’t; they’re tucked awway
where you won’t find them unless you
read the manual. You select betwe een them
The Side Panel
Most of the M’s I/O is different from thhat of the
FS. For example, whereas the larger model
m has
balanced XLR audio outputs, the smaller has
unbalanced quarter-inch jack outputs. Alongside
these, the M has just a MIDI in socket, having
lost the MIDI out and thru of the larger model.
There are differences in the USB and keyboard
connections too. The FS has a single USB B and
a dedicated connector for the 3620 keyboard,
but the M replaces these with a USB B and a USB
A that allow you to connect any class--compliant
MIDI keyboard controller. Finally, the M has an
input for an external PSU, whereas the FS has an
IEC mains input for its internal power supply. The
one common feature here is the set of o five DIP
switches, four of which select the MID DI channel,
and the fifth of which determines wheether Auto
Power Off is enabled or not.
38 April 2022 / w w w. s o u n d o n s o u n d . c o m
by pressing sequences of the Manual
Start button while switching on the M’s
power. Similarly, the M still offers the FS’s
voice assignment modes, and different
sequences allow you to select between the
original 2600 behaviour (CV2 is retained
when you release the uppermost key so
any oscillators playing at this pitch don’t
drop to the lowermost) and the Odyssey’s
behaviour (CV2 drops to the lowermost
key when all others are released).
Although it’s not explained clearly in
the manual, portamento has also been
retained even though there are no physical
controls for it; you can switch it on/off using
MIDI CC65 and adjust its rate using MIDI
CC5. Unfortunately, these settings are
lost when you switch the M off and, even
if you set it up each time before playing,
it’s unlikely that making changes will be
practical during performance. Of course,
you can use the lag processor to give
yourself the ability to control a short glide
while playing, but this precludes its use for
anything else.
If I’m honest, I love the size and heft of
the FS,
FS but I accept that many musicians
nowadays view ‘compact’ as a benefit
rather than as an insult. Nonetheless, at
20.5 inches wide (around 60 percent of
the original) the M isn’t going to fit into
a 19-inch rack. Likewise, its height of a little
over 12 inches (again, around 60 percent)
precludes its use within any conventional
modular system. Yet there’s no doubt
that it’s ideal for use alongside modern
modular synths, to the extent that Korg
have even reduced the external voltage
required to trigger the contour generators
from 10V to 5V, which increases its
compatibility with (in particular) Eurorack
equipment. So, how should we view it?
Is it a self-contained synth that benefits
from the addition of external modules such
as LFOs, multiples, slew generators and
VCAs? Or is it a high-class colony of three
oscillators, dual filters, a noise generator,
two contour generators, a complex S&H
generator, a ring modulator, an envelope
follower, an electronic switch, a preamp,
four voltage processors, an audio VCA and
a spring reverb that extend your existing
modular synth far beyond its existing
capabilities? Of course, it’s both. How you
view a and use it is up to you.
The Sound
Despite the physical changes, Korg claim
that thhe M’s circuitry and components
are thhe same as those in the FS, and
thereffore sonically accurate to the
original. So I placed it next to my 2600P to
gauge e its authenticity and, whether I was
programming leads, basses, orchestral
imitations, percussion or effects, I was able
to creeate patches on the two that were
all but identical. I was also able to obtain
sound ds from the M that the original was
unable to produce, often because Korg
have improved (ie. extended) the miserly
attack k, decay and release times of the
original’s contour generators.
No onetheless, there are some
differe ences in the sounds of the oscillators
and th he filters, and the M’s main audio
signal path is noticeably quieter than
the orriginal’s. The same is true of its
spring g reverb, which Korg’s marketing
deparrtment describes as “re-engineered
and adapted to the new size body to
the sa ame lush effect”. They’re wrong.
It’s lussher. It’s also quieter, with a wider
bandw width and an increased maximum
reverb time. To be fair, some vintage
synths benefit from a bit of noise — for
example, the Emerson Moog Modular
would d have sounded very odd with
a pristine signal path — but the 2600 isn’t
one ofo them. This may be heresy but,
 ON TE S T
ARP 2600 M
Small(er) but perfectly formed: the M
has an all metal enclosure, rather than the
wooden case of the FS, and 3.5mm patch
cables are the order of the day, otherwise the
two are almost identical.
like the FS before it, I think that the M is
a clearer, brighter, quieter and — dare
I say it? — sonically better synth than my
50-year-old 2600P.
But it’s not perfect (no synth is) and
I found two deficiencies that require
mention. The first was that the M can
generate a low-level whine. For much
of the review, I had it connected to my
Mac via USB and to two keyboards: the
supplied microKey via USB and a second
via 5-pin MIDI, which was itself connected
to the Mac via USB. The outputs from both
Ch-Ch-Ch-Ch-Changes
Compared with the original ARP 2600P:
• You can select between 4012 and 4072
filter modes.
• The Upper CV is available from the
front panel.
• The modulation wheel CV is available from
the front panel.
• It offers 5-pin and USB MIDI.
• It has quarter-inch audio outputs.
• Auto Power Off will switch it off if you forget
to do so.
• You can’t connect a 3620 keyboard or its
equivalent.
• It has a three-way rather than a four-way
multiple.
40 April 2022 / w w w. s o u n d o n s o u n d . c o m
the M and the Mac were then connected
to the same mixer and, in addition, all the
devices were powered from the same ring
main. There were multiple loops here so
I suspected that these were the cause,
but it turned out that the microKey was
the culprit. Connecting a different MIDI
keyboard to the end of the same cable
eliminated the problem.
The second was that — like the FS —
the M’s MIDI specification is very limited:
just note on and off, pitch bend, modulation
CC01, and the programmable portamento.
Compared with the Korg ARP 2600 FS:
• It has unbalanced rather than balanced
audio outputs.
• It has a different reverb.
• There’s no immediate way to control it using
aftertouch.
• It has shorter faders with Odyssey-style caps.
• It has an additional USB A socket.
• There’s no 5-pin MIDI out or thru.
• You can’t connect a 3620 keyboard or its
equivalent.
• It has a three-way rather than a four-way
multiple.
• It uses an external PSU rather than an internal
power supply.
Bu l
e c rg o cts,
n o ownload
l n l d a bas a ic
c s plug-inss
, nd
’ s u les offere
e d
t r ariannts
t
t
t a l l
o ar
e am
e.
None of the sliders or switches transmit
messages, nor is anything received to
affect the settings that they control. When
I’ve raised this issue with manufacturers
in the past they’ve explained that adding
the VCAs necessary to implement a more
extensive MIDI specification might
compromise the authenticity of a clone, but
I’m not sure that I fully believe this. Either
way, the M offers what it offers.
To Buy Or Not To Buy?
Korg claim that the M is a “genuine ARP
2600 in a more convenient size” and, in
comparisons with my 2600P, it acquitted
itself very well. It’s not what one could call
a low-cost synth but, when you consider
what an original 2600 would cost and the
amount of work that you might have to
do even if you could find one — replacing
sockets, cleaning or replacing faders,
replacing capacitors, upgrading VCAs, and
so on — the M starts to feel like a bit of
a bargain. If the loss of the 3620 keyboard
isn’t an issue and you would like to obtain
the 2600 sound from something smaller
and more affordable than an FS... well,
here it is. 
$ $1729
W www.korg.com
 Spi r it
Black Bundle LDC without the
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 ON TE S T
Hifiman
Sundara,
Arya &
HE-R10D
Headphones
Hifiman are driving the
development of headphone
technology at an
impressive pace!
SAM INGLIS
E
stablished by Dr Fang Bian and
based in Tianjin, China, Hifiman
manufacture headphones and
in-ear monitors, along with associated
equipment such as DACs and portable
digital music players. As the name
suggests, their products have historically
been targeted at the hi-fi market.
However, it’s not unusual for headphone
and speaker manufacturers to cross over
into the studio world, and several Hifiman
models have become popular choices
for recording and mixing. At the time of
writing, there are more than 20 pairs
of over-ear headphones in the Hifiman
catalogue, ranging in price from the $129
HE400se right up to the $6000 Susvara
and beyond.
Planar Potential
The Hifiman range includes conventional
moving-coil and electrostatic designs, but
the large majority of their models feature
planar magnetic drivers. Like moving-coil
drivers, these are based on the principle
of electromagnetic induction, but
whereas moving-coil designs have
a moulded, rigid diaphragm with a coil of
wire attached, planar magnetic designs
use a flat diaphragm with thin wires
embedded in it or attached to it.
42 April 2022 / w w w. s o u n d o n s o u n d . c o m
This is a technology
that has some inherent
advantages over more
traditional designs.
Well-designed planar
magnetic headphones
can have incredibly
low distortion, and
their presentation of
musical dynamics is
second to none, as
is their low-frequency
response. Recent technological
developments have also mitigated the
historic downsides of planar magnetic
headphones, such as low sensitivity and The Sundara is one of the most affordable
poor high-frequency extension. High-end pairs of open-backed planar magnetic
planar designs like the Audeze LCD-X are headphones available.
now well established as monitoring tools
in the studio world, but they represent designs to get ever closer to the
a serious financial investment. Hifiman theoretical ideal of a diaphragm with
likewise offer plenty of choice in this no mass at all. In a planar magnetic
high-end bracket, but their range also driver, the magnetic field that pushes
makes the same technology available at the diaphragm to and fro is created
much lower prices. by having magnets either side of
Hifiman are also prolific innovators the diaphragm. Hifiman’s Advanced
when it comes to headphone design. Asymmetrical Planar Driver pioneers
New and proprietary technologies they’ve the idea of making the magnets on the
developed include an ultra-lightweight inside smaller than those on the outside,
diaphragm material said to be of reducing diffraction and obstruction of
“nanometre thickness”, allowing their sound waves passing from diaphragm
to ear. Sound escaping outwards,
moreover, can benefit from the
patented Window Shade louvre
design, which makes the outside
of the earcup more open and
less reflective. And whereas the
earcups on most headphones
are symmetrical about both the
vertical and the horizontal axes,
some Hifiman designs use an
asymmetrical cup that is intended to
follow the shape of the human ear
more accurately.
Going Deep
For this review, I was sent three
models from the Hifiman range.
Retailing at an extremely competitive
$349, the open-backed Sundara is
one of Hifiman’s most affordable and
most popular planar magnetic designs.
The $1599 Arya, meanwhile, is a more
upmarket open-backed planar magnetic
headphone that incorporates all four
of the new developments described
above. Finally, the HE-R10 is an intriguing
closed-back design with earcups
fashioned from wood. It’s available in
both moving-coil (HE-R10D) and planar
magnetic (HE-R10P) versions; I tested the
former, which retails at $1299.
I suspect that one of the reasons why
phones like the LCD-X sound so great is
the use of a very deep cushion around
the earcup, which positions the driver
quite a bit further from the ear than we’re
used to in conventional headphones.
This is also a design feature of HEDD
Audio’s remarkable HEDDphones and of
Avantone’s Planar model, and likewise of
all three of the Hifiman models I tested.
All three are also supplied in substantial
cardboard boxes, with the headphones
and their cables nestling in lavish velvet
drapery, but without any other sort of
bag or case. Beyond this, though, there’s
a surprising divergence in the physical
design of the three models.
Here Comes The Sundara
The Sundaras are perhaps the most
conventional of the three. This model
does use symmetrical earcups — in
fact, they’re perfectly circular — but the
cushions are less deep towards the
front, so the cups angle inwards slightly.
Connection to the audio source is made
by a Y-cable with TRS mini-jacks that slide
into the base of each earcup. Being only
1.5m long and rather inflexible, this cable
isn’t ideal for all situations.
The earcups are mounted on forks
which slide into the headband, allowing
them to rotate up and down but not
sideways, while a wide leather strap
spreads the load on the top of the head.
At 372g the Sundara are not as heavy
as some planar magnetic designs, and
I found them generally comfortable. Both
frame and earcups are made mainly of
metal and give the impression of being
robust. Hifiman’s specifications claim
a truly heroic frequency response of
6Hz to 75kHz, though no tolerances are
quoted. The nominal impedance is 37Ω
and sensitivity is given as 94dB.
Apart from the logo, the overall
black colour and some basic design
features that are common to almost all
headphones, the Arya are different from
the Sundara in almost every respect.
The earcups are deeply elongated and
also slightly asymmetrical front to back,
with the distinctive Window Shade slats
in place of the Sundara’s wire grille. The
cushions are once again tapered, but
are slightly deeper and faced with what
seems to be real rather than synthetic
leather. The earcups are attached to
a different fork system which, this time, is
also hinged at the point where it joins the
frame, allowing front-to-back rotation; and
this time around, it’s the headstrap itself
that adjusts to achieve a comfortable fit.
Even the cable is different, employing
“crystalline copper” wire with a more
flexible braided coating and a fixed
quarter-inch jack at the other end. (The
supplied cable did not work, though, so
I tested the Arya with the Sundara cable.)
I don’t think anyone will be terribly
disappointed to learn that the Arya’s
published frequency response is slightly
narrower than that of the Sundara, at 8Hz
to 65kHz, though again, no tolerances
are given. Impedance is listed as 35Ω
and sensitivity at 90dB. The Aryas are
also marginally heavier than their siblings,
weighing in at 404g, but once again
the soft cushions and broad headstrap
make them sufficiently comfortable for
long sessions. In practice, although
both the Arya and the Sundara are
Hifiman Sundara,
Arya & HE-R10D
$349/$1599/$1299
PROS
• Among Hifiman’s many technological
innovations, perhaps the most
notable is creating a planar
magnetic driver with an extended
high-frequency response.
• The Sundara offers excellent value
for money.
• Easy to drive loud from most sources.
CONS
• The Arya and HE-R10 would benefit
from software correction.
• No carrying cases supplied.
SUMMARY
With so many different models
available, and so many new
technologies implemented, an
exploration of Hifiman’s huge product
range is overdue for anyone seriously
contemplating a high-end pair of
headphones. The highly affordable
Sundara are a particularly attractive
option for studio use.
w w w. s o u n d o n s o u n d . c o m / April 2022 43
 ON TE S T
HIF IM A N S U N D A R A , A R YA & H E- R 10 D

less sensitive than typical moving-coil coating to the surface of the diaphragm planar magnetic designs. Compared
models, I had no problem driving them to control its texture. As you’d expect with the classic LCD sound, both display
loud from a good-quality audio interface from a moving-coil design, the HE-R10D a noticeable presence rise in the upper
headphone socket. is more sensitive than the other models, mids. In principle this is no bad thing;
achieving a specified figure of 103dB neutrality is both a moving target and
Into The Wood with a quoted impedance of 32Ω. It also a highly subjective goal in headphone
Both the Sundara and the Arya are, if not has a slightly more curtailed frequency design, and one person’s uncoloured
exactly conventional, at least obviously response, though since this is still given response may well be another person’s
recognisable as headphones. By contrast, as 15Hz to 35kHz, this is unlikely to be dull or soft.
the HE-R10Ds are truly extraordinary — a problem. Both models also inherit other key
and, once again, almost entirely different. strengths of planar magnetic phones.
I’m not sure to what extent the use of
Listening Tests Musical dynamics are put across with real
wood as an earcup material is driven Given how different the review force and power, and I was particularly
by considerations of performance or headphones are in other respects, it impressed with the stereo imaging.
appearance, but it’s certainly striking. was something of a relief to discover Obviously, headphone stereo will never
The wood has an attractive figure and that the Sundara and the Arya actually sound like loudspeaker stereo, but these
a bold orange tint, the cups are shaped have quite a bit in common sonically. headphones provide a convincing sense
smoothly and attractively, and despite My impression in both cases was that of width, and make it easy to locate
being the bulkiest of the three models on Hifiman have put a lot of effort into individual sources within the panorama.
test, the HE-R10Ds are also the lightest overcoming the subdued treble response And these two models both have the
at 333g. that is a shortcoming of traditional sense of clarity and endless transient
The ear cushions this time form headroom that I associate with really
a shallow oval, once again slightly angled good planar phones.
inwards at the front, and yet another One of the other things that’s so
different fork design permits remarkable about the Audeze phones
a small degree of forward/ is the low end. Bass instruments have
back movement. There is a sense of articulation, depth and balance
no separate headstrap as that you simply don’t get from most
on the other models, but moving-coil headphones. The low end is
achieving a comfortable less prominent on the Sundara and Arya,
fit is just as easy. Just and indeed the bottom octave or two
for variety, the HE-R10Ds sound a little shy to my ears. However,
connect to the outside there’s no doubt the information is there,
world via a single and some brief experiments with EQ
mini-jack socket in the The Arya’s distinctive suggested that it should be possible to
base of the left earcup. earcup is non-symmetrical restore a weightier balance if that’s what
No fewer than three cables and designed to follow the you like.
contours of the ear.
are supplied, all apparently of Inasmuch as I have reservations about
high quality and terminating the sound of these two models, then,
respectively in a mini-jack, they’re really to do with that presence
full-size jack and four-pin boost in the upper mids. As I already
XLR. You can also mentioned, the additional brightness
replace the cable compared to some other phones is not
with a miniature unwelcome in principle, but in order to
DAC called the be benign, this sort of treble lift needs to
Bluemini, which be broad and even, and I couldn’t quite
attaches directly to convince myself that this was the case
the mini-jack socket here. The Aryas, in particular, have quite
and receives audio a noticeable peak in the 4kHz region,
over either USB which can cause some sounds to jump
or Bluetooth. out of the mix in a rather unnatural way.
The drivers I would hope this could be tamed by
employed in the corrective EQ, but Sonarworks don’t
HE-R10D introduce currently have a generic profile for this
another Hifiman model, so I wasn’t able to confirm this.
technological What then of the HE-R10D? Once
innovation called the again, there’s also something of an
Topology Diaphragm. emphasis on the upper part of the
As far as I can mid-range, making the overall sound
understand, this involves a little ‘scooped’ and soft in the 1-2 kHz
applying a special area. I’m not sure what sonic benefits

44 April 2022 / w w w. s o u n d o n s o u n d . c o m
 ON TE S T
HIF IM A N S U N D A R A , A R YA & H E- R 10 D
Wood you believe it: the HE-R10
is unusual to say the least!
the wooden earcups provide, but they
offer somewhat less isolation than
a conventional closed-back design, and
are a little prone to amplifying mechanical
noise. The most noticeable feature
of these phones, though, is the bass.
It’s anything but shy, and is especially
prominent in the 100-150 Hz region. This
takes some getting used to, but I can
imagine it proving useful in genres that
require a particular focus on the low end.
Hi Fliers?
Even having tried three different models,
then, I’m reluctant to draw any universal
conclusions about Hifiman headphones.
Whereas other manufacturers’ product
ranges typically share many design
elements and a have ‘family sound’, that
just doesn’t seem to be the case here.
I can believe that there’s a Hifiman model
to suit everyone — it’s just a question of
finding it! Stylish and unique though they
are, I’m not sure I see an obvious role
46 April 2022 / w w w. s o u n d o n s o u n d . c o m
for the HE-R10D in a music production
context, unless you find that an epic
low end helps you get into the creative
zone; certainly, I’ve heard conventional
moving-coil headphones that sound
closer to my perception of a neutral
sound at a lower cost.
By contrast, the Aryas certainly do
have the potential to be a very capable
mixing tool. Achieving such an extended
and present high-frequency response
from a planar magnetic driver is no easy
feat, and I can’t think of another planar
model I’ve tried that comes close. To my
mind, however, they probably require
some form of software equalisation to
fully realise their potential. In their raw
state, that 4kHz peak gives the sound
a slight standing-in-front-of-a-Fender-Twin
quality that gets tiring fairly quickly.
They are also pitched price-wise against
impressive competitors, such as the
various Audeze LCD models, Focal Clear
MG Professional, Sennheiser HD800,
AKG K812 and HEDD Audio HEDDphones.
As always when there’s this much money
involved, I’d recommend extensive
auditioning before you make a choice.
Out of the three models on test,
then, I’d have to say that the relatively
humble Sundaras are the ones that really
hit the sweet spot for studio use. I’ve
heard the combination of good planar
magnetic drivers and big, deep earcups
enough times by now to be convinced
that it delivers a distinctive quality which
is both useful for mixing and hard to
get from other designs. Until now, the
Avantone Planars represented the most
affordable way to get that quality, but
these are significantly more affordable.
Both represent excellent value, and those
who prefer a crisper tonality won’t regret
choosing the Sundara.  
$ Sundara $349; Arya $1599;
HE-R10D $1299.
W www.hifiman.com
 ON TE S T
Tierra Audio Flavours
Inline Booster Preamps
Clean-gain ‘booster’ preamps are popular accessories for
ribbon and moving-coil dynamic mics — but what if that
gain were deliberately made more characterful?
NEIL ROGERS
T
here’s been a big resurgence
in the popularity of high-quality
passive ribbon and moving-coil
dynamic microphones in recent years,
both for music production and dialogue
applications such as podcasting and
YouTube videos. Consequently, we’ve
also seen more inline booster preamps
becoming available. These devices, such
as Cloud Microphones’ Cloudlifter and
the Triton Audio FETHead, are usually
intended to give low-output passive
mics a helping hand by way of applying
a fixed-level but low-noise boost before
that signal reaches the main mic preamp.
They typically draw power from the +48V
phantom supply on the main preamp.
To a large extent, Tierra Audio’s new
Flavours preamps perform just the same
task. They require +48V phantom power
from a mic preamp to operate but don’t
48 April 2022 / w w w. s o u n d o n s o u n d . c o m
pass it through to the attached mic (so
as not to risk damaging sensitive or
valuable vintage passive mics), and they
apply a boost to the signal. But in one
significant respect, they’re very different...
Each of the seven devices in this range
has been designed to ‘flavour’ the signal
in a different way, and their names are
all plays on a culinary theme. The options
range from the subtle enhancement yet
pretty clean gain of Salt through to full-on
distortion courtesy of Chilli and Truffle,
with a handful of gentler tone-shaping
options (Mint, Vanilla and Cocoa) filling out
the ‘menu’ along the way. Tierra explain
that the basic concept is to combine the
provision of the extra gain that low-level
dynamic mics often need with the
option of adding a sonic ‘flavour’ to the
clean-sounding preamps of a desk or
audio interface — rather in the way that
you might sometimes opt to use different
boutique mic preamps for their character.
Physically, the Flavours are rugged,
compact devices, about half the size of
a typical DI box, and they have no user
controls; there’s just an XLR on one end
for the mic input and another at the other
end for the output to your mic preamp.
Inside, they’re all completely different
designs (rather than being a case of
pairing the same booster with different
‘saturation’ circuitry), so as well as
stamping a different sonic footprint on the
signal they also apply different amounts
of gain. The maximum boosts range from
23dB, which is fairly conventional for
an inline preamp, to a whopping 53dB.
(Note that the precise gain figure for each
box depends on the impedance of the
attached microphone.)
Starter
When the Flavours first arrived at my
studio — Tierra kindly sent their whole
range to test and audition — I happened
to be in the middle of a drum recording
session, and thought I’d throw a couple
straight into action. But with the above
considerations of gain and power in
mind, I found that before I could really
get to grips with them I first had to spend
a little time contemplating how best to
implement them into my studio setup.
Using them initially on some more
low-risk ‘character’ mics seemed like
a fun way to start, but the only such mic
option I had available which didn’t require
phantom power became far too hot,
level-wise, for the main mic preamp when
plugged into one of the Flavours. I have
an Audient mixing console and various
outboard preamp options here, but none
of those are able to supply +48V phantom
power when they’re set to receive
line-level input sources (which would
be an obvious way to accommodate the
higher level). So with those preamps I had
to use an attenuator to use the Flavours
on anything loud, such as drums. With
my little RME Babyface audio interface,
however, I encountered no such problem:
the preamps in this (and various other
interfaces) can provide +48V while
padding the mic amps down to accept
line-level signals.
Once I’d got over these initial teething
issues, though, I ended up having a lot
of fun while trying out the different boxes
on various sources. They definitely offer
something different and useful in a variety
of recording situations. As well as
dropping them in on a few real recording
sessions I conducted some more
controlled tests, in which I auditioned
each of the Flavours on the same four
sources: a voice recording, an acoustic
guitar and a couple of different drums. In
the next section of this review, then, I’ll
give you a little explanation of how I got
on with the ‘main courses’ before offering
you some more general thoughts on the
range as a whole towards the end.
The Main Course
Salt ($150, up to +39dB): First up, we
have Salt, which is intended as the
cleanest option. Similar to a conventional
‘booster’ device, its main purpose is to
provide extra gain (as much as +39dB,
which is rather more than most such
devices) to passive ribbon or low-output
moving-coil dynamic mics like the Shure
SM7. Unlike some other Flavours, there’s
no expensive input transformer in this
design and that’s reflected in the lower
asking price. While it’s the least obviously
‘flavoured’, it can nonetheless add some
subtle enhancement.
For the drum recording tests, I had
a close kick drum microphone (Audio
Technica ATM25) and one channel
of a Royer SF-12 ribbon acting as
a mono overhead, about 1m (just over
a yard) above the snare. On both mics
Salt provided a helpful extra dose of
clean-sounding gain, without there being
any noticeable side-effects. Specifically,
when paired with a Shure SM7 on my
voice recording, the first unit I was sent
introduced some unwanted hiss, but
when I raised this with Tierra they said
they were aware of a few faulty units
from an early production run, and sent
a replacement which worked without
problem. I should point out that, even
with the faulty one, there were no such
issues when used on the acoustic guitar
with one channel of my Royer SF-12
— and on that particular source, I also
noticed a slight flattering of the sound;
it conveyed just a touch more clarity
and presence than when using the main
preamp on its own.
Pepper ($150, up to +36dB): Pepper
imparts a more obvious character.
Providing up to +36dB of gain this time,
Tierra describe it as having a “defined
sound, with a pinch of compression
and fast transients — a sibling of the
Chilli flavour [of which more below] but
less spicy”. Suggested applications
include live sound, and particularly rock
and other heavy styles, where those
characteristics are more likely to come
in handy.
In my tests, Pepper’s low-frequency
roll-off from 100Hz was clearly audible
and this meant it worked well on the
acoustic guitar, voice and drum overhead
test recordings. The same characteristic
made it less helpful on the kick drum,
of course, and it wouldn’t be the right
choice for anything where you need good
low-end extension.
Pepper definitely added a little
something around 3-5 kHz, leaving my
snare drum feeling slightly compressed
in a pleasing way; that bit more exciting.
While I’d never describe my own voice as
‘exciting’, I did find that using Pepper to
capture it through a Shure SM7 delivered
a nicely flattering sound. As with Salt,
the first review unit was noisy with this
particular mic. Apparently it was the same
issue, and again a replacement worked
fine; I experienced no noise problems
with any other Flavours.
Truffle ($225, up to +39dB): Tierra
describe Truffle as their most “daring
and exclusive sound”. It incorporates
a Lundahl input transformer, a design
decision which contributes to the higher
price. Like Salt, it applies up to +39dB of
gain but it’s intended much more as an
‘effect’ than as a conventional booster.
In my tests, Truffle’s high-frequency
roll-off (at 12kHz) was clearly audible, and
I liked how that gave my spoken voice
a more upfront, ‘radio ready’ sound. On
the drum recordings, it delivered some
heavy distortion, especially on the kick
mic, with the 1-3 kHz range in particular
‘coming alive’ in a very characterful
way. Truffle is certainly not a subtle
option, then. But if you think of it more
like a guitar stompbox it starts to make
sense, and I found that when I used it
in conjunction with an input attenuator
I could control the amount of distortion
pretty well. Used in that way it can be
surprisingly versatile and I wonder
whether perhaps such a facility could be
built into this box in the future?
Chilli ($260, up to +53dB): Chilli
is another more extreme-sounding
option, and Tierra describe it as having
a “sharp, crispy sound, that can produce
hard-sounding dynamics”. Featuring
a Lundahl input transformer, Chilli provides
up to +53dB of gain. That amount of gain
is getting into standalone mic preamp
territory and, while it’s generally fine when
working with quiet sources, the fact that
you have no control over the figure limits
it; frequently, I had to plug in my 25dB
passive attenuator so as not to overload
the next stage in my recording chain.
Excessive levels aside, Chilli provides
Tierra Audio Flavours
From $150
PROS
• Salt and Pepper are good options to
consider if in the market for a booster
preamp that adds a little ‘something’.
• Mint and Vanilla, in particular,
provide creative possibilities but
can also perform some useful
technical functions.
• Cocoa, Chilli and Truffle provide large
gain boosts and bags of character
and are surprisingly versatile if paired
with an input attenuator.
CONS
• Excessive fixed boost limits some
models’ applications.
• Two models are noisy with some mics
(Tierra say this is being addressed).
SUMMARY
The Flavours double as compact
booster preamps and character boxes,
offering a range of high-quality sonic
options from clean gain to full-on
distortion and saturation.
w w w. s o u n d o n s o u n d . c o m / April 2022 49
 ON TE S T
TIE R R A AU DIO FL AVOU R S
The full range of Tierra Audio’s Flavours. Each
model includes an aide-mémoire description of
its tonal characteristics on the top of its sturdy
metal housing.
some really appealing character,
delivering a hard, distorted sound on
close drum mics generally, and adding
a nice crisp bite to my snare drum in
a ribbon overhead. On the quieter
sources, which didn’t drive things
into such obvious distortion, it was
surprisingly subtle, though, adding
a touch more presence to my voice
and giving an acoustic guitar a slightly
compressed feel, seeming to ‘round
off’ the transients. There’s clearly
something good going on, and I wonder
if incorporating a pad into a Chilli Mk2, to
make it more versatile, might be a good
move for Tierra?
Mint ($235, up to +27dB): Mint was,
for me, the option that really showcased
the overall Flavours concept best, and
in real-world recording sessions this was
definitely the one I tended to reach for
most often. It offers a sensible amount
of gain, includes both a Lundahl input
transformer and applies a useful EQ
curve that can often help you capture
more mix-ready sounds — it’s a gentle
low-frequency roll-off, starting around
300-400 Hz, which I found was perfect
for counteracting the proximity effect
bass boost you encounter when using
a ribbon mic to close-mic a guitar cab or
an acoustic guitar. It could prove to be
50 April 2022 / w w w. s o u n d o n s o u n d . c o m
a great companion for something like
a Coles 4038 ribbon mic, enabling you
to make use of it on a wider range of
sources without much further processing.
Other than on the kick drum, Mint
worked very well in my more structured
tests too. With a drum overhead mic,
it added presence whilst also cleaning
out space for close mics to provide
the weight in the kick and snare. As I’d
observed on the recording sessions, it
enabled me to get a ribbon mic in much
closer to the acoustic guitar without
there being an excessive build-up in
the low end.
In short, if you have a decent clean
mic preamp, Mint offers a cost-effective
way of augmenting your input chain. It
could see plenty of use in any studio,
whilst also being a very handy, compact
device that you can take on location
recording sessions.
Vanilla ($260, up to +27dB): Working
our way into the dessert menu, Vanilla
features both low- and high-pass filtering
at 100Hz and 10kHz, respectively, whilst
providing up to 27dB of gain boost, which
seems a sensible amount. Featuring
a Lundahl input transformer again, Tierra
describe this option as having a “creamy
and controlled sound” and it certainly
seemed to add a nice sense of clarity to
my voice, the low-pass filter combining
nicely with a subtle sweetening of the
midrange. I missed a little bit of the top
end of my acoustic guitar, but it worked
well when combined with a ribbon mic
above my snare. Vanilla proved to be
a great companion for my Beyerdynamic
M160 ribbon mic, when using it
to close-mic a guitar cabinet; it shaped
the sound beautifully, whilst adding just
a touch more excitement.
Cocoa ($305, up to +48dB): Finally,
then, we come to Cocoa, which is
another on the ‘effect’ end of the Flavour
spectrum, its most immediately audible
characteristic being a high-frequency
roll-off that begins just above 2kHz. Tierra
chose a Carnhill input Transformer for this
model, which pushes the price up slightly
again — it’s a component you often find in
high-quality characterful mic preamps that
cost significantly more than this.
At the risk of sounding like a stuck
record, I’m slightly bemused at why so
much gain boost is required — this one
delivers up to 48dB, depending on the
mic’s output impedance — and while it’s
no doubt a consequence of the circuit
used to deliver this particular colour, it
feels like building in an attenuator would
make it much more versatile.
Cocoa does impose plenty of
character, though! On a kick drum, the
high-frequency roll-off and transformer
combined to create a lovely, ‘vintage
thud’ and it should be great generally at
taming any source which has excessive
top-end, or which would benefit from
thick, characterful lows. It was too much
on my own voice and on acoustic guitar,
but I can see plenty of uses — both
creative ones and more practical ones,
wherever the top end will benefit from a
little HF smoothing — for Cocoa.
Something Sweet?
Although it took me a little while to warm
to the whole Flavours concept, I ended
up feeling that these colourful little boxes
do a great job of changing the
personality of a sound, and I can see
lots of creative potential for using
them in the studio, on location or in
a live-sound setting.
I’ve mentioned a few practical niggles
and you should bear them in mind if
you’re considering investing in a Flavour
or two. Also, while it was great having
them all to play with, note that some of
the Flavours do a very specific thing that’s
more dramatic than you’d expect from
a ‘normal’ preamp, so it’s important you
choose the right one. Of course, most
prospective purchasers probably won’t
have a means of comparing them before
they buy, and to help inform that choice
I’ve prepared some audio examples
(https://sosm.ag/tierra-audio-flavours).
I mentioned above that the output
levels from some of the Flavours were just
too hot to make them of practical use with
my professional studio gear and, while
I do have devices like inline pads to hand,
I think it would be better if Tierra were to
consider incorporating a basic 15-20 dB
pad into these preamps; the extra switch
might bump the cost up a touch, but it
would solve most of the practical issues
I faced. Depending on where that pad was
placed, it could also mean much more
flexibility in terms of how much character
the more colourful options add to loud
sources such as drums or guitars; you’re
unlikely to need 40-50 dB of extra gain on
them, after all, but you may well appreciate
the sonic signature. And regarding the
(solved!) noise issue with a couple of the
early-batch review units used with my
SM7, Tierra were very responsive — if you
encounter any such issues I’m sure that
they’d be very happy to help.
I believe the core idea of the Flavours
range is to provide sonic options to
musicians and recordists who don’t own
or perhaps cannot afford the luxury of
having a range of boutique mic preamps:
a way, for example, to sweeten your
interface-based recording chain with
a high-quality input transformer normally
found in more expensive studio outboard.
They can do just that and, despite my
quibbles about practicality, it’s relatively
cheap and easy to acquire a passive
attenuator or external phantom supply
that would address them.
The most important ‘takeaway’ from
this review, really, is that all of the options
sound great in their own way when used
on certain sources. They do indeed
change the Flavour of your audio, in what
might be a creative or more practically
helpful way. It’s all in the taste, and a dash
of the right Flavour could well prove to be
an inspiring addition to any setup.
$ From $150 each, as detailed in the main
review text.
T M1 Distribution +1 248 556 4615.
E info@m1distribution.com
W www.m1distribution.com
W https://tierra.audio
 ON TE S T

Waldorf M
Wavetable Synthesizer

Waldorf return to their roots The PPG Wave was the perfect
marriage of digital oscillators and
with a modern take on the analogue filters, and the PPG Wave
classic Microwave. series sold well. Digital technology
moves fast, however, and by the
mid-1980s, synths like the Yamaha DX7,
RORY DOW Korg DW-8000, and Sequential Prophet
VS were offering the tempting delights
Microwave was released. Wavetable
synthesis became Waldorf’s signature
sound. The Microwave, Microwave II,
Microwave XT, Wave, Blofeld, Largo,
Nave, Quantum and Iridium all use
wavetable synthesis, not to mention the
various Eurorack modules Waldorf have
released. Ask any synth fan to name

W
avetable synthesis was
a revolution when it first arrived
in 1978. Pioneered by Wolfgang
Palm, the PPG Wavecomputer 360 was
the first synthesizer to use this new
technology. It had a few shortcomings,
chiefly a lack of filters, that stopped
it from becoming a big seller but its
successor, the PPG Wave, was a hit.
of digital without the hefty price tag.
PPG closed their doors in 1987.
At the same time, Waldorf Electronics
GmbH, a new company created by PPG’s
former German distributor, was forming.
Wolfgang Palm worked with Waldorf to
develop the integrated circuit chips for
their first product, based on the PPG
Wave. And so, in 1989, the Waldorf
a manufacturer of wavetable synths,
and, more often than not, Waldorf will be
the answer.
On a purely personal note, the
Waldorf Microwave XT, a later version
of the original Microwave, is one of
my desert-island synths. Even after 20
years of ownership, it still surprises me
with sounds I could never imagine. It

52 April 2022 / w w w. s o u n d o n s o u n d . c o m

was with great excitement then that
I received the Waldorf M for review.
What Is M?
The M is a modern recreation of the
first Microwave, which has become
a cult classic thanks to its combination
of crunchy 8-bit digital oscillators and
smooth analogue filters. Subsequent
revisions — the Microwave II and
Microwave XT — moved to digital
filters. Many felt they lacked the
warmth of the OG. However, they did
enhance sound design possibilities
with a proper modulation matrix,
lots of digital filter options, and
enhancements to the synth engine
like oscillator sync, ring modulation, FM
and effects.
For better or worse, most of the
Microwave II/XT improvements have not
made it into the M. There are a couple
of exceptions. Oscillators can work
in Microwave I or Microwave II mode
(more on that shortly), and sync and ring
modulation are available (although only in
the Microwave II oscillator mode). There
is no FM, modulation matrix, or effects.
Like the Microwave, the M is an
eight-voice synthesizer. An optional voice
expansion card will double the count
to 16. It has two wavetable oscillators,
a 24dB/oct filter (an SSI 2144 ladder
design replaces the original Curtis SEM
chip), analogue VCAs (stereo), four
envelopes, and two LFOs. The M is
also four-part multitimbral (the OG was
eight-part, but with only eight voices;
perhaps that was considered overkill).
You don’t have to look far to confirm
that the M is more than just a reboot. The
Microwave’s single-knob 2U rackmount
design wasn’t particularly fun to use, but
the M’s spacious 440 x 305mm front
panel offers plenty of hands-on control.
There’s a 3-inch colour display and
plenty of buttons to speed up navigation.
Wavetable import, patch backup, and OS
updates are all done via SD card. And
along with the traditional MIDI DIN in, out,
and thru ports, there’s a USB Type B port
(for MIDI only, no audio).
One welcome improvement is that
each oscillator can now load a different
wavetable instead of sharing one.
Wavetables contain 64 waveforms, and
there are 96 factory wavetables, plus an
extra 32 slots for importing your own. The
OG wavetables are all present and will
sound pleasingly familiar to anyone who
has ever owned a Wave, Microwave I, II
or XT.
As I mentioned earlier, wavetables can
work in one of two modes: MWI or MWII.
These emulate the digital compression
of either the Microwave I (8-bit, 240kHz,
no anti-aliasing) or Microwave II (16-bit,
40kHz, band-limited). There was not
as much difference in these modes as
I had expected. High-pitched sounds will
exhibit different aliasing characteristics,
and the 8-bit mode does imbue some
extra crunchiness on bass sounds, which
is very welcome. But I doubt you would
spot the difference when the synth is
buried in a mix.
In MWI mode, you can even enable
‘ASIC bug mode’, which emulates a bug
Waldorf M
$2499
PROS
• A classic synth given a new lease
of life.
• Spacious front panel with excellent
build quality and a handy screen.
• That nostalgic sound.
CONS
• A few missed opportunities to bring
Microwave II/XT features along for
the ride.
SUMMARY
The M is a recreation, with loving
enhancements, of the classic Waldorf
Microwave. Whilst it doesn’t have
exactly the same circuitry or sound,
the effect of 8-bit wavetables through
analogue filters remains as beguiling
as it ever was.
in the Microwave I that causes distortion
if oscillator levels are above a certain
threshold. This attention to detail is
vital because the M is compatible with
Microwave I SysEx files — great news for
anyone with a sound library they want to
migrate forward.
Navigation
Getting around the M’s interface is
simple. There are dedicated controls
for key parameters, but to reach others,
you tap one of the Mode buttons above
the screen. For example, tap the LFO
button, and you’ll see everything LFO
related. Each screen combines helpful
graphics (the LFO shape and frequency
in this case), some information about the
currently loaded patch, and labels for
the four ‘soft controls’ underneath the
screen. These four knobs give you access
to further LFO-related parameters. You’ll
need to navigate further sub-pages to
access additional parameters.
This navigation system works well
and is commonly used by other synth
manufacturers. I do wish that, like many
of the recent Sequential synths which
use a similar approach, the screen would
automatically move to the relevant page
when you move a dedicated control. For
example, move the filter frequency knob
and automatically see the filter pages
on screen. I find this speeds up sound
design on the Sequential synths, and
I think it would work well here too.
Presets & Wavetables
The M packs in a whopping 2048 presets,
arranged in 16 banks of 128. Every
w w w. s o u n d o n s o u n d . c o m / April 2022 53
 ON TE S T
WALDORF M
preset slot can be overwritten. I can’t
imagine needing that many presets for
a wavetable synth, but it’s nice to have
the space available. With such generous
capacity, it’s a surprise that Waldorf
didn’t choose to increase the number of
wavetables beyond 128 (and only 32 of
those are user writeable). I would have
gladly traded half the preset slots for
a doubling in wavetable slots.
Vladimir Salnikov, Waldorf’s DSP
guru and the man behind much of the
M’s operating system, gave me some
information on the technical specs of its
wavetable format. The first 65 wavetables
are taken directly from Microwave I. In turn,
the first 28 of those were taken from the
PPG Wave. All wavetables are 16-bit, but
the Microwave wavetables are upsampled
from 8-bit, giving them an effective 8-bit
dynamic range for authenticity. Any new
tables are true 16-bit.
User-imported wavetables must be
8-bit (Vladimir tells me they hope to
support 16-bit user wavetables soon) and
256 samples per waveform. Although
the M stores 64 waveforms per table,
the import algorithm can cleverly handle
wavetables of different lengths either
by skipping periodic waves with longer
wavetables or duplicating them for
shorter ones. I successfully imported
a few of my own. If you need to convert
from another format, opening wavetables
in Xfer Records’ Serum and saving them
as 8-bit .256 files works very nicely.
One peculiarity of the Microwave
factory wavetable format is that the last
three waveforms in every wavetable are
reserved for sine, triangle and square
54 April 2022 / w w w. s o u n d o n s o u n d . c o m
Round The Back
Joining the 12V DC 1.8A PSU socket are
a stereo quarter-inch headphone jack with
volume control, main outputs on L+R mono
TS jacks, four auxiliary outputs on stereo
TRS jacks, MIDI in, out and thru on five-pin
DIN connectors, a USB 2.0 port (for MIDI
only), and an SD card slot (for firmware
updates and import/export of sounds and
wavetables) plus a Kensington-compatible
security lock slot (to keep anyone from
wandering off with your precious).
waveforms. Both oscillators shared the
same wavetable in the Microwave, so the
user could use one oscillator as a true
wavetable and the second for a simple
‘analogue’ waveform. With the M, I’m
not sure it makes sense anymore as you
can select different wavetables for each
oscillator. The real problem with having
these waveforms at the end of every
wavetable is that if you use, say, an LFO
to sweep the wavetable position, it is
easy to end up with a glitch whenever
the LFO pushes the position into the top
three waveforms. The Microwave II/XT
introduced an intelligent parameter called
‘Limit’, which stopped any modulation
from selecting these three waveforms,
resulting in guaranteed glitch-free
modulation. Although it wasn’t in the
OS version I was sent to review, Waldorf
tell me that this Limit function will be
coming in a future update, which is
excellent news.
Dial M For Modulation
Modulation remains very similar to the
original. There are two ADSR envelopes
(for VCA and VCF), an eight-point
loopable Wave envelope (for wavetable
position), a four-point sustainable Free
envelope, and two LFOs. Dedicated
modulation slots are available in most
sections of the synth. For example, in
each oscillator, there are two slots for
pitch modulation and two for wavetable
position. The filter has two slots for
frequency and one for resonance.
Usually, where two slots are available,
one of them can be multiplied by
a second source, allowing you to, for
example, control the LFO amount with
a modulation wheel.
In practice, this system does offer
plenty of modulation possibilities. But it
does mean a lack of overview because
the modulation is scattered over many
sub-pages. A modulation matrix would
give a central place to see everything in
a patch. Instead, we have to investigate
each page and sub-page of the synth to
see what is assigned.
Modulation sources include the usual
suspects: the envelopes and LFOs, and
MIDI sources like mod wheel, velocity, key
tracking, aftertouch and poly pressure.
There’s a few fun ones, too: Inverse (a flip
flop), CoinFlip (random 0 or 1), and Random
(a new random value for each note).
The ADSR envelopes allow for
modulation of all stages. The filter
envelope has a delay option that can also
be modulated. The more complex Wave
and Free envelopes work using multiple
pairs of level and time parameters, which
string together to make multi-point
envelopes. The Wave envelope has
adjustable loop points that can be placed
at any of the eight points. There are also
modulation slots to adjust all levels or
times so it can double as a complex LFO.
The Free envelope works similarly, with
three time/level parameter pairs. It has
 ON TE S T
WALDORF M
The M measures 440 x 305 x 85mm and weighs in at a reassuringly weighty 5.7kg.
sustain level after the three pairs and
release time/level, and can even loop in
two different places.
The LFOs offer plenty of tweakability
too. Alongside the usual sine, triangle,
pulse, random and S&H waveforms are
options to adjust delay, symmetry and
humanise (randomising the frequency).
The LFO speed ranges from 0.01Hz up to
62Hz via a switchable range parameter.
There is a Global LFO too, which
mirrors the setting of LFO2 but doesn’t
retrigger with every voice. This LFO (but
curiously not the others) can sync to
MIDI Clock, with an impressively wide
frequency from 0.000065Hz (4 hours 15
minutes per cycle) to 160Hz (extending
into audio range) depending on your bpm
and clock division settings.
Filters
The M has more filters than you
might imagine. Its principal filter is an
analogue 24dB/oct low-pass SSI 2144
ladder design. It sounds excellent, very
analogue, with a strong and stable
resonance that can keytrack over about
five octaves if required. And it will
self-oscillate. It does suffer from bass
reduction when resonance is increased,
a ‘feature’ of ladder designs. The
analogue filter, of course, is one of the
56 April 2022 / w w w. s o u n d o n s o u n d . c o m
key selling points of the M, but Waldorf
have included more.
A second state-variable digital
filter offers low-, band- and high-pass
options. This filter is a 12dB/oct slope
with resonance but doesn’t self-oscillate.
Frequency can be modulated via one
modulator. For anyone who wants to keep
their signal path purely analogue, this filter
can be disabled, but it is handy to have.
Multi Mode
The M is four-part multitimbral. That
means you can play four sounds
simultaneously. There are still only eight
voices (or 16 with the expansion card),
but it does mean you can sequence
perhaps a bass sound, a pad sound, and
a lead at the same time. Each of the four
parts in a multi can have a MIDI channel,
voice limit, pan settings, output, key span,
velocity span and transpose setting.
Sadly, arpeggiator settings are per-synth,
not per-part, so using multi mode for
layered polyrhythms is out.
Multi mode is flexible enough to
achieve several things. You can layer
two or three sounds together or create
keyboard splits. There is a handy filter
lock function — you can lock parts
together so that when you adjust filter
settings on one part, any other locked
parts will also change. The four stereo
aux outputs also mean you can assign
each part a separate MIDI channel and
have each sound on a separate output,
effectively giving you four synths in one.
Because the M has no internal effects,
there is no dilemma on sharing those
effects in a multitimbral setup. I rather
admire Waldorf for resisting the urge
to add effects. They’re great for sound
design, but the classic dry sound of the
M benefits from decent outboard delay
and reverb, and having the luxury of four
parts mixed on separate channels in your
outboard mixer is hard to beat.
Mmm
The big question, of course, is how
does the M sound? Wavetable synthesis
has come a long way since the PPG
Wavecomputer 360. Software instruments
like Massive and Serum have pulled
wavetables into the modern age. The M is
not that. It is unashamedly vintage, which
is the main reason to own it. It’s a familiar
sound, especially if you grew up in the ’80s
and ’90s. It can sound pure and sweet or
gritty and harsh. It excels at glassy pads,
bell plucks, gritty basses, otherworldly
textures, vocal pads, sharp leads and zany
effects. There are plenty of PPG Wave
tribute sounds in the factory presets too.
 Arpeggiator THE MIDI
There was no arpeggiator on the original Microwave, but one
was added to the Microwave II/XT. On the M, arpeggiator settings
are saved with the preset and include bpm, clock division, play
direction and range. The user can also specify a pattern from
SPECIALISTS
a choice of 16 presets. These patterns turn specific steps of the
arpeggiator on or off, creating more interesting rhythmic variations.
The screen shows a nice graphical preview of the pattern.

Comparing the M’s sound to the original Microwave,

or even to the PPG Wave, is probably unhelpful. The M
isn’t built with original Microwave parts, and it doesn’t
use the same chip types (ASICs). Instead, it is a digital
simulation of these old chips. The analogue low-pass
filter is of a completely different design too. Therefore,
it is best to say that M uses the Microwave sound as
a target. I don’t have a Microwave to compare to, but
from my investigations online listening to demos, it’s
close enough.
PRO SOLO MK3
Indeed, it sounds fantastic. I was particularly struck
MIDI CONTROL FOR YOUR
by how well the M fits into a mix. Unlike many analogue VINTAGE & MODULAR SYNTHS
synths, it’s not massively heavy in the bottom end. A must for vintage and modular synth fans, the Kenton Pro Solo
That is not to say it can’t do thundering bass, but the Mk3 enables your MIDI keyboard or sequencer to control CV
harmonically rich wavetables mean that many sounds are synths with V/oct, Hz/V & 1.2V/oct scaling systems.
complex in other areas of the frequency spectrum. As
such, the M is an ideal companion for analogue synthesis.
NEW MIDI FOR YOUR USB
Conclusion ADDED KEYBOARD OR CONTROLLER
ROUTING
These days, I am constantly agog at the breadth of MODES
choice we have when buying synthesizers. It feels like MIIDI USB HOST mk3
every taste is catered for. Vintage synths are reinvented Got USB but need MIDI? You need
for a new audience (not to mention the old one) and the a Kenton MIDI USB Host to provide
MIDDI In & Out sockets for keyboards
reasons to own genuine vintage synths seem to diminish and
d controllers which have only a
with every year that passes. USBB B type socket.
Kudos to Waldorf for keeping their legacy alive by
reimagining the Microwave whilst simultaneously pushing
THRU-5, MERGE-4
the boundaries with new synths like the Quantum/Iridium. 12 & 25 &8
AVAILABLE AVAILABLE
The M is a powerful synth and a unique one. Nothing else
sounds quite like it. This reboot is packed with valuable
improvements whilst keeping the essential core of what
made the Microwave great.
If I have one regret, it’s that the M doesn’t use a few
more of the excellent features that Microwave II and MIDI Thru Boxes MIDI Merge Boxes
XT brought to the table. Notably the modulation matrix, Plug in your sequencer or master It takes a lot more than a splitter
modulation modifiers and FM. Then again, I’m sure keyboard and get 5, 12 or 25 cable to merge MIDI signals properly.
there are plenty of people who will rejoice at the lack of identical MIDI copies with no Kenton Merge boxes combine 4 or 8
audible latency for your sound MIDI sources into a single MIDI data
a modulation matrix.
modules and other MIDI devices. stream, with all MIDI data including
The Waldorf M is a fantastic synth. The price is on the clock, time code and SysEx just as
high side, but you get a well-built, large-format (at least they should be.
by modern standards), ‘vintage’ synth that won’t need
maintenance, and has plenty of modern conveniences
Keep all your synths, sequencers and
like USB MIDI and an SD card for backups and imports.
controllers talking with Kenton MIDI products
Its solid, all-metal construction should last a lifetime or
two, and with 2048 preset slots, you won’t run out of
sounds any time soon.   Find out more at kentonuk.com CELEBRATING

35
YEARS
$ $2499.99
W www.waldorfmusic.com

w w w. s o u n d o n s o u n d . c o m / April 2022 57
 ON TE S T
PreSonus
Revelator
USB Audio Interface
io24
PreSonus’s Revelator io24 is packed with
useful features for podcasters, streamers and
even common-or-garden recording musicians.
ROBIN VINCENT
E
verything about the Revelator is
a little bit different. In a departure
from both the 1U boxes and the
desktop units the io24 is intriguingly
wedged shaped. It has only one knob on
the front, and it seems to poke out a lot
further than most. You’ve got to stick on
the rubber feet yourself and there’s a great
big warning sticker over the USB-C port
which makes you feel you’re about to do
something wrong. But in the hand it feels
good and weighty with a metal enclosure
PreSonus Revelator io24
$199
PROS
• Clear interface design.
• Great display.
• Fat Channel DSP processing.
• Versatile software mixing solution.
• Two virtual stereo channels.
• Detailed and useful documentation.
• Live-streaming made easy!
CONS
• Digital interface is not for everyone.
• Couldn’t the channel display stay up
just a little longer?
• Small I/O count.
• Limited preset selection from the
front panel.
SUMMARY
PreSonus’s interface for live-streamers
is a perfectly judged and innovative
box that’ll please anyone who only ever
needs two inputs.
58 April 2022 / w w w. s o u n d o n s o u n d . c o m
and plastic front panel. I like the angles
already and when you turn it on the little
screen is handsome enough to pull an
audible ‘ooooh’ out from me.
The PreSonus Revelator io24 is a USB
Audio Interface with Loopback and effects.
It’s aimed at podcasting and live streaming,
although that doesn’t exclude it from being
a recording interface for music making,
a fact supported by the presence of
genuine old-school 5-pin MIDI I/O ports on
the back. You have two XLR combi inputs
using low-noise XMAX-L mic preamps on
the front and two jack outputs on the back
with a separate socket for headphones,
and that’s your lot. There are no gain
controls, phantom power switches or
monitoring options to be seen. Everything
is controlled via the little screen, knob
encoder and a couple of buttons.
Screen Control
While the Revelator does make full use of
the PreSonus’ Universal Control app, which
we’ll come onto in a minute, I wanted to
focus first on how much that little screen
can do right there on the front panel.
The glowing knob is your main point of
contact and it’s impossible to miss because
the blue halo is a little overwhelming. It’s
a free-moving encoder that you can also
push in to make selections. By default, it
controls the master output level. Push it
once to switch to headphone level and
push it again to control the blend between
direct and software monitoring. The icon
on the screen changes to show what
you’re controlling and displays the knob
position. On the left of the screen, you
have some nice input monitoring indicators
with peak lights, and on the right, you get
output meters. The screen is nicely lit and
clear so you’ll have no trouble seeing from
the other side of the room and at all angles
in dark or light conditions.
For individual control over the inputs,
you press the ‘1’ or ‘2’ button. You get
a larger view of that channel’s input
monitoring, a gain reduction meter is
added and the encoder now controls the
input gain from zero to 60 dB. You can
also enable +48v phantom power and
a high-pass filter here by pushing the
encoder to cycle from gain to power to
filter, and then turn the knob to enable or
disable it. Having to turn the encoder to
operate a switch is a little strange but it’s
no big deal. If you look away for a moment,
you’ll find that the display has returned
to the main screen, making you press
the number button again to get back to
the channel controls. It’s both a handy
and slightly annoying automatic function
depending on what you’re doing at the
time. I tend to make adjustments to the
instrument I’m recording as well as the gain
on the input when setting something up
but I’m constantly losing what I’m looking
at. There is a dedicated button to return
you to the main screen just in case it hasn’t
automatically switched back quick enough
for you. I’ve never had to use it. Perhaps
 Round the back we find a USB-C port, a pair of proper, steam-powered five-pin
MIDI ports, two quarter-inch audio outputs and a quarter-inch headphone socket.

there’s an option to turn off the automatic
return or set the length to a much longer
time so it wouldn’t annoy me as much — it’s
always the little things.
Under each channel button is another
button labelled ‘Preset’, which is there to
switch between effects presets that you
set up in the Revelator app. Tap the button
to switch between two different presets
or hold the button for a few seconds to
bypass them. The last button is a big, red,
bold and very welcome master output
Mute button.
Despite my annoyance at the
auto-return, the display, the button and
encoder combo work perfectly well. While
I enjoy having a single knob or button
per function it can often be very fiddly on
small audio interfaces. The widened and
angled front panel definitely helps in this
respect. The single knob approach offers
up more space and accessing the controls
feels very easy and uncrowded. Overall it’s
a positive experience.
are some simpler channels; one for
Playback from your DAW and two for the
Loopback drivers.
Loopback is a vital component for
anyone trying to live-stream or podcast
from one PC and we’re seeing it on almost
every audio interface these days. The
basic idea is that you can have multiple
pieces of software playing audio set to
different Loopback drivers and route them
as virtual inputs to your host software,
whether that’s a DAW or some streaming
software like OBS or Vmix. Revelator
has two stereo Loopback channels
which, rather than going directly to your
host software, are mixed through the
Universal Control along with your other
inputs and playback to a choice of three
stereo outputs. This gives you much more
control over all your hardware and virtual
inputs before it reaches your DAW or
streaming software.
For each stereo output you can create
a different mix within the mixer. You’ve
got Main that goes to your speakers and
then Stream Mix A and Stream Mix B,
which are routed to separate virtual stereo
outputs. So, you can set a mix for what
you’re monitoring, set a different mix for
streaming and another one going back
to a guest on Zoom. You can route each
mix to your headphones with the click of
a button. This is some brilliantly versatile
stuff that audio interfaces without monitor
mixer apps tend to miss.
DSP Effects
The secret sauce I mentioned is poured out
all over your inputs as some tasty onboard
effects. This is the ‘Fat Channel’ DSP

Software Control
While the box gives you plenty to fiddle
with, the secret sauce to the sound and
features of the Revelator can be found in
the Universal Control app. PreSonus use
the same driver/configuration/mixer app for
all their hardware interfaces but the layout
and features of the mixer varies depending
on what interface is plugged in.
It’s a good-looking mixer. It has shades
of Studio One about it but has a little bit
more visual flair. There’s a channel strip
for each input with both input monitoring
and gain reduction monitoring if you’ve
engaged the gate, compressor or limiter
that we’ll come onto in a minute. There are
gain controls, panning, mixer level, mute,
solo, phantom power and a send for the
reverb. Alongside the two input channels The Universal Control app — PreSonus’s version of the one ring.

w w w. s o u n d o n s o u n d . c o m / April 2022 59
 ON TE S T
P R E S ON U S R E V E L AT OR IO 24

processing that uses the same State-Space

Modelled technology as we find on the
StudioLive Series III mixers and as plug-ins
within Studio One. They are a revelation.
The Fat Channel effects are presented in
two different ways: Preset-led set-and-forget
for beginners and advanced access for
people who want to look a little deeper.
The Preset side is very welcome and
gets you to good results very quickly.
I imagine it’s aimed at people who just want
to sound great with the minimum of fuss.
The last thing a beginner needs is to be
faced with complex compressors, EQ and
signal chains when they just want to record
their first track or live-stream on Twitch.
In the Universal Control software there’s
a big knob with which you can select from
eight permanent presets. They come
with helpful names like Broadcast, Vocal,
Electric Guitar and so on, so you can
choose the appropriate one for each input.
To the right of the knob is a list of the Fat
Channel component plug-ins with some
individual presets and the ability to turn
them on and off. You get options like Low,
Moderate, High on the Gate and Limiter,
a couple of source-specific ones on the
Compressor and a choice between 40,
80 or 160 Hz on the HPF. That’s plenty of
control for most of us in most cases. If you
do make any changes, there are six slots
for saving your own presets.
You can allocate two presets to input
channels 1&2 that can then be switched
between on the hardware via the Preset
button. It’s slightly odd that the big preset
knob in the software isn’t tied to the big
hardware knob on the Revelator. Why
couldn’t you dial in a preset from the front
panel rather than only giving you the ability
to switch between two with a button?
Fat Channel
Within the Fat Channel you get
a High-pass Filter, Gate, Compressor, EQ,
Limiter and zany Vocal effects because
you’re such fun as an outrageously
entertaining live-streamer. When you
opt to view the Fat Channel it becomes
a row of plug-in GUIs along the top of
the mixer and gives you full access to all
the parameters. It will all be very familiar
to anyone who’s used the Fat Channel
plug-in in Studio One because it’s the
same. The Compressor gives you the
option of Standard, Tube or JFET and
the EQ can offer Passive and Vintage
alongside the Standard version.
The quality of these processors is
excellent and having them available, not
Vocal Fun
Along with the professional seriousness of the
Fat Channel there are some cool/fun/cringy vocal
effects available exclusively on Channel 1. You’ve
got a Doubler, De-tuner, Vocoder, Ring Modulator,
Comb Filter and Delay. The Doubler is actually
quite interesting, and the Delay is fine, but the
rest are for kids showing off to their mates.
as CPU-draining, latency-inducing plug-ins,
but as DSP-run processors on your inputs,
is pretty awesome. This means that these
plug-ins run when direct monitoring. Turn
the blend knob all the way down and
you can enjoy separate processing on
both channels direct to your speakers
or headphones.
There’s also a reverb that sits outside
the Fat Channel on a channel strip of its
own, which isn’t saved as part of a preset.
Both input channels have a send control
to route to the reverb. Interestingly the
reverb is not present in the recorded
audio when using the main two inputs
to your DAW directly, it’s just there for
monitoring exactly, as it should be.
However, it is there if you record via one
of the two stereo Stream Mix inputs. So, it
works both in a recording situation where
you wouldn’t necessarily want reverb
printed on the input and in a streaming
situation where you would want the
reverb present in the stream. That’s good
and thoughtful design.
I plugged directly in with a condenser
mic and electric guitar, dialled in the
appropriate presets and added a touch of
reverb and it sounded really lovely. The
There’s one other effect-based facility that
could be useful in your super-fun podcast and
that’s the Hot Key. You allocate this button
a preset, it defaults to having a slap-back echo,
and you can drop it into your vocal stream
to add a cool, crowd-pleasingly temporary
special effect.
dynamics are fresh and deep, the vocals
crisp, the guitar had just enough crunch,
the gate did its job and the reverb put me
in a believable space. This is great stuff
that has the potential to radically improve
the signal for recording.
Streamers
For streamers, live performers and
podcasters the Revelator io24 has a lot
to offer. The Universal Control software
is immensely powerful and becomes
the focus of your mix regardless of the
other software you’re using. You can
bring in audio from Zoom or a Discord
conversation while running game audio
and the playback from your DAW all at
the same time with no shenanigans, tricky
setups or workarounds. You can get good
professional-sounding vocals, with great
levels and processing to keep it there while
giving yourself and virtual collaborators
a different monitor mix.
The fact that it’s a box with a great
display and controls that can take any
microphone you want to use and has the
versatility to let you run another mic or
an instrument makes it a far more useful
proposition than a USB microphone. And

60 April 2022 / w w w. s o u n d o n s o u n d . c o m
you have the added bonus of using it as
a bona fide recording interface for music.
Headphone Interference
One blight on an otherwise impressive
resume was some noise interference
on the headphones. It was that washy,
scratchy sort of noise you get with earth
loops on laptops that would change as
I moved knobs in the Universal Control
app. It wasn’t being recorded, it wasn’t
killed by the noise gate, it wasn’t present
in my speakers, it was just there buzzing
quietly in the headphones. Searching
online I found a few people with a similar
issue. It’s not overwhelming or intrusive
when recording but it is noticeable. One
suggested solution was to try headphones
with a higher impedance, so I swapped
my 32Ω cheap phones for a pair of AKG
240DFs at 600Ω and the problem did
indeed go away. I also tried a 10-quid
ground-loop noise isolator, and it did the job
as well, so it’s definitely a shielding issue of
some kind, but you don’t necessarily need
to buy expensive headphones to solve it.
I spoke to PreSonus, and they are aware of
the problem and have fixed it in the latest
production run. However, be aware that if
you’re buying an older or secondhand unit
you might run into this issue.
Conclusion
PreSonus have put a lot of thought into
the functionality of the Revelator io24
and it achieves that very rare thing of
bringing some stand-out features to the
crowded marketplace of two-input audio
interfaces. The digital front end gives a lot
more information than you’d usually see at
this level, and you can operate it without
referencing the software. But then the
software is superb and sets the stage for
whatever application you want to apply it to.
It’s been designed to serve the needs of the
podcaster and live-streamer and it achieves
that with aplomb, but it’s also a decent
recording interface with just enough DSP to
make a real difference.
I am completely sold on the concept
of a wedge-shaped audio interface. The
angle of the front panel makes it easier to
see, easier to plug things in and makes the
sort of ergonomic design sense that the 1U
format interface never has. It also adds to
the stability of the unit, letting it sail through
my cable hang test where an interface fails if
it tips over when a couple of mic cables are
plugged in. The Revelator is rock-solid with
all the forces acting in helpful directions. I’d
like more inputs, please, but that’s hardly
a criticism.
There are small elements that don’t fill me
with joy, like the blue band on the encoder
being way too bright for a dimly lit studio or
having to turn a knob to switch on phantom
power, or not being able to select all the
presets from the front panel. I experienced
some crackle when changing presets or
moving parameters in the Fat Channel, and
the vocal effects are made more ridiculous
because you can’t use the vocoder effect
like a vocoder or the ring modulator in
the way you should, but these are minor
inconveniences next to the versatility of
this particular hardware and software
combination. It’s a great audio interface for
streamers and musicians alike.
$ $199.95
W www.presonus.com
w u o so d.co / il 022 2
 ON TE S T

Gainlab Audio Empress

Passive Valve Equaliser

Hungarian company Gainlab impressed us with their Dictator compressor, so we were

looking forward to putting their EQ through its paces.
NEIL ROGERS
like — and not only because I considered All three bands offer more
the price to be very keen for what was frequencies than most EQs of this

G
ainlab Audio are a Hungarian
company, formed fairly recently
by the engineers at Budapest’s
Gainlab Studio — after years of servicing
and modifying the equipment in their
own studio, they decided to develop
their own designs. I reviewed their
Dictator stereo valve compressor back
in SOS December 2021 (http://sosm.ag/
gainlab-audio-dictator) and found a lot to
Gainlab Empress
$2090
PROS
• Excellent build quality.
• Extra mid band adds some
substantial flexibility.
• Tube Boost facility adds an extra dash
of colour.
CONS
• Dual-Mono design can make
recall a little fiddly when used on
stereo sources.
SUMMARY
The Empress is another impressive
early release from Gainlab Audio.
A dual-mono EQ, it provides the
qualities of a classic passive tube
EQ whilst adding extra features that
successfully provide some meaningful,
extra flexibility.
on offer. For review this time is another
of Gainlab’s early releases: after my
positive encounter with the Dictator
I was very much looking forward
to spending some quality time with
the Empress!
Overview
The Empress is a dual-mono valve EQ
that attempts to combine the attributes
of a passive, Pultec EQP-1A-style EQ
with some additional features. The
headline addition is an extra mid-band
that you wouldn’t traditionally find on
this style of EQ, but there are also a few
other features that attempt to add more
flexibility to a studio classic that has
been around in one form or another
since the 1950s. Gainlab explain that
they were keen to maintain the heart of
what people like about passive EQs of
this kind — that curious simultaneous
cut-and-boost of the highs or lows being
perhaps the most obvious example
— whilst also including a valve amplifier
stage that’s intended to allow the user
more control over their EQ moves,
and over the saturation effect of the
electronics. I’ll talk a little more about
these features as I explain how I got on
with the Empress in my studio.
type. All the controls are stepped (ie.
switches), the cut and boost controls
operating in 1.5dB increments, and the
unit generally feels good to play with.
A very small niggle was that, due to the
combination of extra control features, the
dual-mono control set and some quite
small switches/labelling, I often found
recalling settings a little time-consuming
(you have to do everything twice when
used on a stereo signal, of course).
Electronics-wise, the Empress features
the sort of high-quality components
you’d expect to find on this sort of
processor at this sort of price range,
such as military-grade vacuum tubes and
nanocrystalline transformers (which, as
I understand it, have lower losses and a
higher saturation level than conventional
transformer cores, as well as being
more expensive!). Overall, it exhibits
a confidence-inspiring air of quality.
Magic In The Mids?
The first thing I wanted to know about
the Empress was how well it did ‘that
Pultec thing’, so I spent a little time trying
to fill out a limp-sounding kick drum.
Adding a generous amount at 60Hz
(with the gain set well past the 12 o’clock
position) whilst simultaneously cutting

62 April 2022 / w w w. s o u n d o n s o u n d . c o m

the same frequency worked superbly
to add all the weight I wanted in a very
focused way.
One of the additional features found
on both the low and high bands in this
design is a Shift
option, which
allows the
mid-frequencies
of the
simultaneous
cutting and
boosting to be
shifted relative
to each other.
Gainlab decided
on this option The two channels’ audio inputs and outputs are presented on rear-panel XLRs.
at the expense
of any other kind of bandwidth controls,
which is an interesting choice and
something worth consideration if you’re
comparing models in this genre. Playing
with this facility on the aforementioned
kick drum, I found the difference was
quite subtle and perhaps even a little
difficult to hear. But the difference
was much more pronounced when
experimenting with the high band on
an acoustic guitar; I was cutting and
boosting at 5kHz to add some definition
and, while I do rather struggle to put
the difference into words, the effect
was certainly audible (I preferred the
sound when the Shift setting was
engaged, which seemed to push the
high-end boost into a nicer area for this
instrument). When using the high band,
you also have the option of choosing
a bell or shelving filter, which adds
another useful layer of flexibility.
The mid-band was a very welcome
feature, and on that same kick drum job
I described earlier it was great to be able
to cut a little around the 300Hz region
at the same time. I tried running a few
different vocals through the Empress
during the review test period, and
I found that the three EQ bands gave
me a surprising amount of flexibility
when experimenting. Adding some high
‘air’ frequencies worked beautifully at
opening up the top end, and I was able
to remove some troublesome boxy tones
around 500Hz, while experimenting with
adding a little weight with the low band.
After a period of experimenting with
individual elements in a mix, it was time
to see how it fared on the mix bus. When
used on the stereo bus, I’ve always
found Pultec-style EQs to be very much
the kind of tool you have to ‘mix into’
rather than add at the end, as a kind of
finishing/mastering tool. This is largely
because the EQ bands are quite broad,
and so might not suit every mix. The
extra controls available on the Empress,
however, allowed me more possibilities
in this scenario, and the mid band
in particular helped me make more
obviously beneficial moves, typically
involving me boosting in the upper mids
at 5-6 kHz or cutting around the 200-300
Hz area to remove some muddy build-up.
This combined with the traditional ‘smiley
face’ boosting of the lows and highs
achieved some great, satisfying results
that helped give a mix a much more
‘finished’ sound.
Push The Colour
A significant factor that contributes
to the enduring popularity of passive
valve EQs is the subtle coloration they
impart, whether you’re boosting/cutting
or not. It’s a difficult thing to articulate
in a written review, but the Empress
generally seemed to add that little
special something when audio was run
through its high-quality components.
To allow you to push this effect a bit
further, Gainlab have incorporated a Tube
Boost facility, which can introduce more
obvious tube ‘saturation’, its nature and
audibility depending very much on the
signal levels you’re pushing through
the Empress: I often found it a little too
much applied to a whole stereo mix,
but it worked superbly to add weight
to individual sources. It imparts that
pleasing sense of ‘analogue warmth’ and
it’s a great tool to have alongside the
core EQ functions. For electronic sources
such as drum loops or sterile-sounding
synth parts in particular, it proved to
be a very effective means of adding
weight and character.
Empressive?
The Empress is another great-sounding
piece of hardware from Gainlab Audio.
It has been nicely designed and put
together beautifully, using high-quality
components. It performs the functions
you’d expect a Pultec-style EQ to
perform very
well indeed,
as you’d
rightly expect
at this price.
But there are
several other
high-quality
Pultec-alikes
out there and
what might set
the Empress
apart from the
alternatives is that the thoughtful design
makes it more versatile. The Mid band
makes it substantially more flexible
than most, and while I did often find the
extra focus controls to be rather too
subtle for many applications, they’re
still welcome additions. The Tube Boost
is a great enhancement, though, and
completes a really well-conceived set of
features that will almost certainly mean
that the valves in this unit remain warm
in your studio on recording, mixing and
mastering sessions alike.
$ $2090.
T Front End Audio +1 888 228 4530
W www.frontendaudio.com
W https://gainlabaudio.com
w w w. s o u n d o n s o u n d . c o m / April 2022 63
 ON TE S T
UJAM Finisher Dynamo
UJAM add sonic movement to the themes explored in
their popular Finisher series.
JOHN WALDEN
A
longside their range of ‘virtual
musician’ guitar, bass and drum
instruments, UJAM have in recent
years been establishing an interesting
Finisher range of effects plug-ins, all of
which are available for Mac (10.14 or later)
and Windows (7 and above), in VST, AU
and AAX formats, and authorised via an
emailed password.
The idea of these plug-ins is that they
all offer processing to suit a specific
theme and, while there is absolutely no
compromise in terms of the quality of
the processing on offer, the UI prioritises
ease-of-use and instant results over
the ability to dive deep into extended
parameter lists. The latest in this series is
64 April 2022 / w w w. s o u n d o n s o u n d . c o m
Effects Plug-in
Finisher Dynamo, reviewed here, and this
multi-effects plug-in has the strapline ‘The
Magical Movement Machine’. The effects
are all intended to combine animation
with sonic/rhythmic manipulation.
Keeping It Simple
Dynamo shares the basics of its UI and
engine with other titles in the Finisher
range, and sports a control set that’s
deliberately kept simple and uncluttered.
The input and output level controls
are joined by five macro-style knobs
(including the large Finisher knob) in the
bottom half of the interface, while the
upper half provides access to the preset
system (nearly 300 presets are helpfully
organised in eight categories) and 50
of what UJAM call Modes. Each Mode is
essentially a pre-configured multi-effects
chain, and for each one the five macro
knobs are linked to one or more key
parameters. The Finisher knob acts
as a wet/dry control but, like the other
macro controls, it’s also linked to other
parameters. Each of the presets is built
from a combination of one Mode and
different starting-point settings for the five
macro knobs.
This stripped-down control set might
seem too simple for some, particularly
those who like to dig into the nitty gritty
of their effects chains, but for most of
us there’s a lot to like here: Dynamo
allows you to tweak the behaviour
of the underlying effects quickly and
easily, while preventing the user from
getting bogged down in parameter
paralysis or needing to consult the
manual. That said, you really should
read the manual, as the list describing
the style of processing offered by each
of the 50 Modes is very informative,
and the nerdy music-production jokes
make it enteratining (and suggest the
development team might be good fun to
hang out with!)
Dynamic Dynamo
Under the hood, the 50 Modes are
evidently built from some pretty complex
effects combinations, with some fairly
routine effects, such as reverb, delay and/
or EQ, joined by things like sweeping
filters, distortion, pitch-shifting, various
glitch/stutter options and, in many of the
Modes, host-tempo-sync’ed modulation.
The Modes are arranged in five
groups (Remix, Solo, Rhythm, Sustain
and Vocal), each designed and/or
optimised with a particular type of
source in mind. Remix is intended for
an instrument group or master bus, and
there are some cool options for creating
broad-brush strokes on a full mix that
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might suit an ear-catching, mid-track,
drop. The Solo Modes are designed to
work better with single sounds, such as
a bass or a melodic instrument, while
Rhythm Modes have drum or percussive
sounds in mind and will often add further
rhythmic interest or complexity. The
Sustain Modes aim to add movement and
interest to sounds such as pads, while the
Vocal Modes provide options for creating
chopped and glitchy results on vocals. Of
course, there’s nothing to stop you from
applying a Mode to something other than
its obvious target, and I suspect the good
folk at UJAM would actively encourage
you to experiment in that way.
Three of the five macro knobs retain
their main function across all of the
presets. I’ve mentioned the Finisher
knob already, and the last of the four
Vari controls acts as a sort of global Tone
control (I suspect it is tweaking more than
just a conventional EQ in most cases,
though). The first Vari knob, Pattern, offers
five different rhythmic styles (half-time,
polyrhythmic, standard, triplet and
double-time) and these each interact with
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Followers Followers
138,817 66,749
UJAM Finisher Dynamo
$99
PROS
• Great combination of creative effects
ideal for electronic music production
and sound design.
• As with all the Finisher line,
super-easy to use.
• Comes with a sense of humour
included.
CONS
• None, though some might prefer
more tweakability.
SUMMARY
Dynamo manages to combine
sophisticated and creative sound
mangling with ease-of-use. UJAM’s
Finisher line continues to impress.
whatever rhythmic element is built into
the effects chain that underlies the Mode;
there’s plenty of fun to be had (and often
a good dollop of chaos to be caused!)
by twisting this dial away from whatever
setting a preset has supplied.
The other two Vari knobs are
pre-configured to adjust the key
w w w. s o u n d o n s o u n d . c o m / April 2022 65
 ON TE S T
UJAM FINISHER DYNAMO

parameters (often multiple

parameters) in the Mode’s
effects chain, but their
function varies according to
which Mode is in use. Usefully,
the knob’s labels change
where appropriate to reflect
what their influence might
be. Labels such as Stutter,
Echos, Filter and Doubler are
fairly self-explanatory, but
others, such as Hallucination,
Smurfbass, Kraftwerkness or
Skrillexness, will perhaps be
less obvious to many users.
Still, it’s a lot of fun to find out
what more Smurfbass can do
to your track, and if nothing
else such labels do encourage
you to experiment. As with
the other Finisher plug-ins,
all the macro controls can
be automated.

In Use
Dynamo is not an effects
plug-in that is aimed
at those who want to
make conventional
singer-songwriter ballads! But if your
focus is on the sort of contemporary
dance/electronica styles where you need
every bit of sonic trickery you can lay your
hands on, or electronic scores for drama,
sci-fi or horror, then you could well find
Dynamo both fun to use and very useful.
I was particularly impressed with
what it could do to transform drum and
pad sounds. For example, applied to
a fairly simple electric drum loop, the
Action Loop preset (found in the Drums
category) splits the original signal into
three frequency bands and uses some
form of resampling to resequence each
band. You can adjust the rhythmic feel
using the Pattern knob, change the pitch
of bands with Vari knobs 2 and 3, and
blend between the dry and processed
sound via the Finisher knob. Add a sweep
of the Tone knob and you have an ‘instant
ear candy’ drum break.
With a simple, static, synth pad, I found
that almost any of the Synth category’s
presets could inject something interesting
to liven it up. For example, the A Bit
Nervous preset adds a rhythmic pulse
and a tempo-sync’ed modulated filter.
The Tone knob lets you adjust how
aggressive the filter sweep can get, while
the Confuse knob (Vari 2) adjusts the
character of the pulse in some interesting
ways. In contrast, taking exactly the same
synth source and applying the Calm
House Sequence preset instantly turns
the sound into a perfect tension bed,
creating a filter-shifted drone mixed with
subtle rhythmic elements. Haunted Thing
takes the same synth and adds creepy
overtones (courtesy of the aptly named
Ghost 1 and Ghost 2 Vari knobs) — ideal
for that part of the scene when you know
something horrible is about to happen but
it hasn’t happened quite yet!
Those are just a few examples, but
whatever your sound source Dynamo
always seems to be able to offer
something that will make it pop out of
a mix (or your mix pop out of a speaker),
giving it extra rhythmic elements or
a sense of tonal motion through some
combination of conventional spatial
effects, modulated filters or distortion and
various resampling tricks for added chop
and stutter. The results can be super-cool
and, thanks to the streamlined workflow
Finisher forces you into, it’s easy to end
up somewhere cool very quickly: just pick
a preset, tweak a knob or two and see
what happens. Repeat as required.
The Glitch List
Of course, Dynamo is not the first
effects plug-in to provide this kind of
The Mode presets and global preset systems
are well stocked and can be access in the upper
half of the UI.
experimental sound mangling. For
example, Sugar Bytes’ Turnado or
Effectrix provide a similar sort of sound
processing. Output’s Movement, or
Lunatic Audio’s new Narcotic (review
in the pipeline!), are also in the same
sort of sonic ballpark. But while all
those plug-ins provide some amazing
processing options and ship with some
instant-gratification presets, things can
get pretty deep pretty quickly. That may
be exactly what some users want, but in
preventing you disappearing down the
rabbit hole, Dynamo offers something
a little different from those.
For those that want their sonic
mangling to come with a minimum of fuss,
enough control to keep things interesting,
and a dollop of good humour, then, UJAM
may just have delivered the perfect
combination: Finisher Dynamo sounds
great, is easy to use and competitively
priced and, at the very least, I’d
encourage you to take advantage of the
free 30-day trial. 
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66 April 2022 / w w w. s o u n d o n s o u n d . c o m
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 ON TE S T
Red
Panda
Tensor
Looping Pedal
Embracing both the charm and limitations of open-reel tape,
the Tensor is a unique and creative looping pedal.
SIMON SMALL
R
ed Panda have been making
innovative DSP-based effects
pedals for over a decade now
and their popular releases, including
the Particle and the Context, usually
delve into the more experimental side of
stompbox effects processing. The Tensor
continues in that vein. A wildly creative
tool, it can of course be used on guitar
but it’s also fun with other sources in the
studio. While it offers most of the features
of a typical loop pedal it gives you
access to so much more besides, and
it’s perhaps best to think of it as a digital
‘tape effects’ pedal: it boasts a collection
of open-reel-tape-style effects that can
be applied either in real time to the
incoming signal, or to recorded loops.
Overview
The Tensor features effects such as live
reverse, tape stop and pitch shifting,
68 April 2022 / w w w. s o u n d o n s o u n d . c o m
but isn’t bound by the limitations of
tape — you also have the ability to
stretch or compress time with no pitch
change. You can record loops, overdub
(sound-on-sound) and randomly splice
recordings of up to 9.4 seconds long
(4.7s for stereo). Having used actual
tape to achieve these sort of effects I’m
familiar with the sound and process, and
the Tensor not only delivers all these
wonderful sounds but it also makes them
very quick and easy to produce.
It’s quite a ‘deep’ pedal: it can be used
as a basic looper with record, playback,
loop and overdub functions, but you
would be doing a disservice to the
Tensor by using it exclusively in this way.
It took me a few sessions and a good
read of the (excellent, 34-page) manual
before I completely understood what
it was capable of, and I suspect some
of the more casual guitar pedal users
may be less keen. I have to say that it’s
totally worth the time investment! It’s
not that it is hard to use, just that a little
understanding is necessary if you’re to
get the most out of it.
Despite its depth in terms of features,
the front panel is mercifully simple. It’s
not spacious, exactly, but everything is
accessible and easy to adjust, and the
pedal has a small footprint so it sits nicely
on a pedalboard or on the desktop.
Speed controls the tape speed; at the
12 o’clock position the ‘tape’ is stopped,
while clockwise from there it plays
forward and anticlockwise in reverse.
Red Panda Tensor
$299
PROS
• Sounds great.
• Wonderful tape-like pitch and
speed shifts.
• Inspires creativity.
• Versatile.
CONS
• Reading the manual is a must!
SUMMARY
Not your typical digital looper, the
Tensor focuses on open-reel-tape-style
looping effects — which are a lot of fun
and inspire creativity!
You’ll notice that adjusting the Speed
control bends your pitch but it doesn’t
stretch the time of your playing. If you
depress the On switch, however, the
buffer will play back in Drift (stretched
time) mode. Time controls your time
stretch/compression (1:4 to 4:1 for
looping and up to 4:1 for real
time). Pitch gives control
over pitch shifting up to ±2
octaves. Rand controls the
amount of randomisation
applied to your sound;
lower settings add stutters,
set around 12 o’clock it
will randomly repeat short
segments, and turned fully
clockwise it slices up your
sound and shuffles it. It also
randomises the values of
each knob, depending on
how they are set. A little bit
of randomisation can go
a long way, but this Rand
control can really help
to breathe some life into
a sound. Blend provides
a wet/dry control. All of
these features can be
controlled simultaneously
or individually by an
expression pedal.
Looping
The Tensor has three
configurations for Hold
(ie. looping) mode, and
you can switch between
modes while playing and
your loop will carry over.
To record a loop, you use
the Hold footswitch. The
toggle switch above this allows you
to set the footswitch to momentary or
latch behaviour. When operating as
a momentary switch, you hold it down to
record, and a short press can be used
to create glitch and stutter effects; latch
can be triggered with one push to begin
recording and another to end.
Rec mode records in real time and
is played back based on the pedal’s
settings. You cannot overdub in this
mode, so if you arm record a new loop
will be written, erasing the previous one.
Ovr mode behaves in a similar manner to
Rec mode but this one allows you to do
sound-on-sound overdubs, as you might
with an analogue tape machine. The first
loop you record will set the loop time
and, while overdubbing, the Tensor’s
current control settings will be applied to
the recording. For example, if you have
the Pitch control set an octave up, this
will be reflected in the recording. But
upon turning Pitch to unison, the original
loop will return to its original pitch and
the overdub will be an octave down.
Red Panda have done well to keep the control
set so simple for what is, beneath the hood, a pretty
sophisticated pedal.
If you’re not used to this way of
working, it can sometimes seem a little
frustrating. A good example would be
while working on a reverse loop... You’ve
set Tensor in reverse mode using the
Speed control and would like to overdub
another part while playing along with
the reversed loop. If you record with
Speed still in reverse, then you’ll find
upon playback that your overdub will be
playing forwards rather than backwards.
It’s not a problem, as such, but a good
reminder that this is not a standard loop
pedal! Rather, it’s a digital interpretation
of a tape machine, so you really have
to ‘think in tape’! Get into that mindset
and it works well, offering plenty of
charm. This style of overdubbing has its
advantages, such as committing to your
effected sounds and overdubbing with
time compression/expansion. Things
can get infinitely weird, with glitching
and drones too. There’s no ‘undo layer
recording’ feature, of course;
it’s very much a mode for
old-school experimentation!
In NXT mode there’s only
half of the recording time
available in the other modes,
since it splits the record
buffer in two. One half listens
while you play and the other
plays it back, with the current
control settings being applied
to that signal. This means
playback and recording times
can be completely different.
A perfect example of a use
for this mode is a tap-tempo
delay: using the Hold switch
in latch mode as a tap lets
you input a tempo. Every
time the Hold LED blinks
red, Tensor is swapping
the live real-time buffer
with the playback buffer,
creating a delay.
Next to the Hold mode
toggle switch is another
labelled Dir, which controls
the playback direction of the
loop; options are Forward,
Reverse, and Alternating,
which flips between
forwards and reverse.
It’s worth noting that the
Speed knob and Dir switch
combine to determine the
playback direction; for example, if Speed
and Dir are both set to reverse then the
loop will play forward.
Ins, Outs & Controllers
The Tensor can be run in stereo or
mono. To cater for this, the single input
and output jacks are both TRS types.
While some users may not be a fan
of this decision, since interfacing with
other pedals in stereo with separate left/
right jacks requires a Y-cable, this allows
a more compact pedal footprint and I think
many more will appreciate that. Stereo
mode reduces the maximum recording
buffer time by half, of course, but it’s still
long enough to be perfectly usable.
Another TRS jack caters for an
expression or tap-tempo pedal, TRS
w w w. s o u n d o n s o u n d . c o m / April 2022 69
 ON TE S T
RED PANDA TENSOR

An online browser-based editor app allows you

to configure presets and access some of the pedal’s worked well with Google Chrome, which control down one notch from Forward,
deeper functionality. Red Panda recommend. raising the Pitch by one notch and
lowering Time by one. In stereo, this
MIDI in, or a control voltage (CV) input
Sound Opinion sounds so wide, and on a vocal effects
of 0 to 3.3 Volts. Red Panda’s own There’s such a lot you can do with the bus it can work very well — almost like
Remote 4 control can also be used Tensor that the above description should an Eventide-style widening effect. (By
to access presets (the Tensor has 31 already make fairly obvious, so I’ll focus the way, when using the Tensor as an
user customisable presets, which are here on a few favourite tricks. I love aux send effect like this, there’s a handy
accessible by a controller or ‘kill dry’ feature). The Tensor
by MIDI) and other controls also allows you to specify
too. I found mode three the maximum input level
on the Remote 4 to be for the pedal, which helps
excellent for the Tensor; this when changing between
turns the remote control into low-level instrument signals
a classic looper interface, and higher-output sources
which will feel very familiar such as audio interfaces and
to users of the Line 6 DL4. mixing consoles.
All these external controllers After an initial period of
are easy to set up and the getting my head around all
manual covers the process the features the Tensor has
in great detail; there are to offer, it has proved to be
even details on how to build an exciting and wonderfully
your own DIY controller. It interactive pedal, with
would have been nice to bountiful possibilities. It does
have a full-form MIDI DIN have limitations, similar to
port too but, again, the those you’d encounter with
The audio I/O are on TRS jacks, which cater for
pedal would need to have mono or stereo operation. A multi-function MIDI/CV/ an analogue open-reel tape
been much larger for that. controller pedal TRS input nestles between them. machine, but that’s part of
The Tensor requires a 9V DC power its charm — and of what makes it a good
supply capable of delivering 250mA keeping Speed set to fully reverse and creative tool. The occasional clicks and
— it will not function correctly without tweaking the Blend control, so that I have the noise that can be introduced with
that, like most digital pedals — and the reverse effect sitting just behind my time compression are generally pleasing
I found it worked well with my Cioks playing in real time. Set like this, the Drift to the ear and musical, but if you plan to
and Strymon supplies. There is also function ends in a glorious tape-stop buy a Tensor you need to be aware you
a micro-USB port on the rear panel, and effect, which is a lot of fun to include in are getting a digital tape machine and
this is used for updating the firmware as reversed arrangements. that it has these personality traits!
well as connecting to Red Panda Lab’s The Speed control also opens up
Web Editor. This browser-based app is a world of detuned effects, and a patch
used for editing presets and accessing I now use frequently for a chorus/warble $ Tensor $299. Remote 4 controller $99.
under-the-hood adjustments, and it effect is created by knocking the Speed W www.redpandalab.com

70 April 2022 / w w w. s o u n d o n s o u n d . c o m
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 ON TE S T
ROBIN BIGWOOD
T
hese two MIDI controller keyboards
from Nektar are representatives
of two new product ranges for the
American company. The 49-, 61- or 88-note
models in the Impact GXP range all have
full-size semi-weighted keys with velocity
and channel/mono aftertouch response,
transport-oriented DAW control features,
a ‘repeat engine’, a DIN MIDI out, pedal
input sockets, and buttons dedicated for use
with Nektar’s VST plug-in host Nektarine.
Other than one programmable knob there
aren’t any extras like drums pads, knobs or
faders, making for a sleek appearance and
shallow dimensions.
The GX range is simpler still: this time
with synth-action velocity-only keys and no
connections to the outside world except
a USB socket and a sustain pedal input.
Pitch and mod wheels are replaced with
a miniature sprung joystick, but DAW control
features survive intact. Models are available
with 49 or 61 full-size keys, or with the 25 mini
keys of the GX Mini on test here. This gets an
intriguing ‘Part 2’ feature: something we’ll get
on to in a moment.
In terms of where these controllers fit
in with the rest of Nektar’s output, they’re
at the more affordable end. Above them
sit the LX+ controllers with faders, knobs
and pads, then the Panorama Ts with their
ambitious plug-in control system, and at the
top of the pile the Panorama Ps, which add
a motorised fader into the mix. There’s one
lower tier, the SE range, which are simple,
straightforward entry-level controllers with
just rudimentary features.
72 April 2022 / w w w. s o u n d o n s o u n d . c o m
Nektar Impact
GXP & GX Mini
Controller Keyboards
Nektar keep doing what they do best: making solid,
capable, no-nonsense MIDI controllers.
The controller market is one in which
it can be hard for products to stand out,
and it’s all too common for complicated
DAW-control schemes to sound great in
press releases but to be largely useless
in practice. So have Nektar hit on a good
balance of features here?
GXP A To Z
Let’s start with the big one, the GXP61. The
construction is all plastic, at least outwardly,
but there’s a nice (and surprising) 5kg
heft to it. The keyboard action lacks some
luxury: it’s pretty noisy on key-release,
key-front corners are not smoothly finished,
and visually the gaps between keys were
slightly uneven fresh from unboxing.
However, in use it’s absolutely fine, and
there’s a useful extra sense of swing and
momentum over and above the most basic
synth actions. It’s visually stylish too, with
black keys significantly higher at the front
than the back. Velocity response is nicely
controllable and repeatable, and aftertouch
a distinct cut above the average: easy to
trigger and modulate, and actually the best
I’ve experienced on a controller at this
price point. Depth of touch of the white
keys is about 10mm, with 3mm more travel
for aftertouch. Pitch and mod wheels are
small (45mm of them is showing) and quite
strongly sprung, but they’re more than fit
for purpose.
As you’d expect, power is provided
through the USB B socket, which also
handles the data connection to your DAW.
But there’s also a 9V 600mA DC inlet that
could come into its own in hardware-only or
iOS-based setups. Slightly disappointingly
it’s for tip-positive PSUs, so not compatible
with old-faithful guitar pedal adaptors.
The presence of both footswitch and
expression pedal sockets is great to see.
The former auto-senses the switch’s
polarity at power-up, and the latter worked
beautifully with a Moog EP-3 pedal I used to
test. Unexpectedly, you’re able to connect
two pedals (or some dual pedals) to the
footswitch socket via a TRS plug to dual-TS
socketk splitter
l .
The single DIN MIDI output is also
unexpectedly versatile, transmitting local
messages by default, but with an option to
source its data from a USB host, essentially
turning the GXP into a USB MIDI interface.
And, while it’s such a boringly functional
thing to get excited about, I was delighted
to see a power switch, which comes into its
own shutting off any downtime power draw
in a laptop-based studio. Controllers without
one sometimes have to be laboriously
unplugged to prevent unwanted battery
drain in some setups.
Taking a butcher’s at the front panel now,
I’ll save us all the tedium of going through
every button individually. But suffice to
say many are colourfully backlit, and the
pairs for octave up/down and semitone
transposition are well placed above the
wheels. Those semitone buttons can
be remapped to MIDI channel, program
change and patch down/up (in a supported
DAW) functionality, by the way. There’s Nektar Impact
a button cluster dedicated principally to
DAW control (for which, see the ‘DAWs
GXP 61 & GX Mini
& More’ box). And most of the rest have $229 & $69
a role in two very specific onboard features, PROS
which I’ll cover right now. • Simple yet purposeful, with
First, there’s a ‘Repeat Engine’. I said, well-chosen and robust DAW
a ‘Repeat Engine’. (Sorry). This causes control features.
• Some unexpectedly handy
held notes to be repeatedly re-struck, as it additional abilities.
were. Like the similarly named Note Repeat • Good documentation and setup
feature on Akai’s MPCs, it can play a useful walkthroughs.
role in rhythmic programming, or creating • Better pedal connectivity than
some competitors.
pulsing melodies or chords. There’s an
internal clock whose tempo you can adjust CONS
with a knob, or via numerical buttons, but it’ll • Programming scheme offers little
also follow and sync to DAW tempo. You get in the way of feedback or overview.
• Nektarine (on the GXP) takes some
to choose the rhythmic division, from quarter
work to master.
to 64th notes (with triplets available via
a neat double button press), and five levels SUMMARY
of swing. Velocity variation of held notes A pair of affordable and effective
can be generated via aftertouch or the mod keyboard controllers, suited to
slightly different roles (and desktop
wheel, and accents dialled in momentarily or real estate), that prioritise robust
automatically in patterns (with a ‘hi-hat pair’ DAW control over knobs, sliders
the useful default). I was just sorry that the and pads.
w w w. s o u n d o n s o u n d . c o m / April 2022 73
 ON TE S T
N E K TA R IM PACT G X P & G X MINI
DAWs & More
Despite the obvious difference in size and design,
the GXP and GX (and all other models in their
respective ranges) share almost identical DAW
control features, and a useful modicum of general
programmability.
For Nektar’s supported DAWs (namely Bitwig,
Cakewalk, Cubase/Nuendo, Digital Performer, FL
Studio, GarageBand, Live, Logic, Reaper, Reason
and Studio One) you need to install a software
package available from the Nektar website to get
some system-level DAW-specific support files on
your Windows 7/8/10 and Mac OS 10.7 (or later)
computer. Pro Tools is also supported, via some
easy manual configuration, and potentially all
manner of other software is compatible (with
a more restricted set of commands) via a MIDI
Machine Control (MMC) mode.
I used Studio One 5.4 on Mac OS Big Sur for
testing, and thanks to a handful of ‘user device’
files written to my Mac OS user preferences folder
the GXP and GX Mini were automatically set up in
and seen by the DAW, and seamlessly switched
into their DAW control modes. Impressively
smooth. It’s the same story with the freebie
bundled DAWs (Cubase LE 10.5 for the GXP, and
Bitwig 8-Track for the GX mini).
Despite there only being eight DAW control
buttons, 21 commands are potentially available,
thanks to a shift key that can be either toggled
or held. Functionality is not identical between
DAWs, but all enjoy basic transport operations
along with, in most cases, click/metronome and
loop playback toggles, previous/next track and
patch (for virtual instruments), undo, some locate
functions, and shortcuts for various application
views or panels. In Studio One I had options to
enable pre-count and input quantise, whereas
Repeat functionality doesn’t extend to even
simple arpeggiator shapes.
Second, there’s a bunch of dedicated
buttons for Nektar’s plug-in host and patch
librarian software, Nektarine. Available
to download from GXP owners’ Nektar
website accounts, this runs standalone
or as a plug-in in VST/VST3/AU and AAX
DAWs, and itself hosts VST, VST3 and AU
plug-ins: up to 16 instruments in a single
instance, with four insert effects per slot and
four send effects available to all. Slots can
be constricted to specific pitch and velocity
ranges, and to individual MIDI channels.
In this respect it’s conceptually similar to
something like Apple’s MainStage, but with
the advantage that it’s cross-platform, and
with the added ability to operate inside
your DAW.
Why bother with Nektarine? Well,
you’re certainly not forced to, but there
are reasons some users might find it
useful. First, and simply, it gives you the
possibility of using your GXP to play virtual
instruments on your computer without the
74 April 2022 / w w w. s o u n d o n s o u n d . c o m
in Logic (for example) you get to toggle the
arpeggiator and overdub on and off. Ongoing
success with systems like these are very much
dependent on Nektar’s firmware and software
releases keeping pace with DAW and OS
developments, but the firm have good form in this
area, and it’s perhaps reassuring that they don’t
do much else except controllers.
The GXP61 adds one useful additional feature
over its little brother: the top-left knob controls
the currently selected track’s level (and, in some
DAWs, the master fader level too). Potential
(actually, make that inevitable) mismatches
between knob and on-screen fader position are
nicely handled in most of the officially supported
DAWs, with a system using two LED indicators.
When you switch to a new track they show you
which direction you need to turn the knob before
you’ll get a benign ‘soft takeover’ of the fader.
And that, folks, for DAW control, is it. There’s
no attempt to try and control your pans, sends or
plug-ins, and no displays, meters, or other more
sophisticated niceties. It’s a much simpler scheme
than some other controllers (including Nektar’s
more expensive models) but perhaps, in the typical
enthusiast studio, all the more useful for that.
There’s a final trick up the GXP and GX
Mini’s sleeves in the form of straightforward,
generic MIDI control. It’s only workable when
DAW integration isn’t active, and there are slight
differences between the two units, but essentially
you can generate various MIDI messages of your
choice from the DAW control buttons, joystick/
wheels/knob, and pedal inputs. Buttons can send
program change messages (and, on the GXP,
bank messages too) or momentary full/zero-
value CC toggles. From the continuous controls
faff of opening them in a DAW. A decent
one, Steinberg’s Retrologue 2, is in the
software bundle. Then, more specifically,
instrument plug-ins can be instantiated
from GXP buttons, and explored via
imported patch lists, with user-configurable
tags and tag groups allowing you to sort
sounds by various criteria. This, incidentally,
explains the otherwise incongruous
printed blue instrument names underneath
the number keys. In this respect a GXP/
Nektarine combo is a little like Native
Instruments’ NKS/Komplete Kontrol system
or Arturia’s Keylab/Analog Lab combo.
I stress a little... The differences (and some
potential drawbacks) include the GXP
not having any displays, no purpose-built
‘ecosystem’ of wholly compatible plug-ins,
and differences in patch handling and
accessibility between VST and AU formats
on the Mac. I didn’t always find the
software interface and associated GXP
button presses to be altogether intuitive
either. However, there’s no question that
Nektarine can offer a really meaningful
Nektar’s installers for DAW integration in
Studio One not only made all the necessary MIDI
configuration but also added dedicated icons for
the hardware: neat.
you can generate a full range of a CC values. In
the absence of any displays or menu systems
Nektar have employed the old-school type of
programming system that uses key-presses to
access different functions. On the GX Mini values
are entered using black keys, and on the GXP with
the dedicated number/value buttons. All fine —
this is a tried and tested system — but for some
parameters with specific ranges of options you’ll
need to keep the manual handy. Thankfully these
are printed, clear and informative, and available
online (of course) too.
level of additional functionality to the GXP, if
it fits your way of working.
How Many Elephants?
Turning to the diminutive GX Mini now, this is
obviously a different (and decidedly tiddlier)
kettle of fish, for different use scenarios that
probably involve a smaller desk... But it clearly
shares a lot of DNA with the GXP range.
The 25 keys, with a mere 136mm octave
span, are velocity sensitive only and at
first glance appear alarmingly toy-like. The
action is certainly light and springy, but in
use perfectly fine, with enough key dip —
7mm — for things to feel positive. Velocity
response is as good as you could reasonably
expect, but there are fixed minimum velocity
values in place: 16 for white keys and 13 for
black, for the standard velocity curve. Those
values vary slightly for the other two available
curves. This looks like a design choice, to
improve the sense of expressivity with the
small keys, more than a technical limitation,
and in typical use it’s all but unnoticeable.
Despite the tiny size there’s the same
complement of DAW control buttons as the
GXP range. Otherwise, though, the GX Mini
is pretty straightforward. On the left end
there’s just a USB Micro-B socket for power
and computer connection, and a 3.5mm
socket for a breakout dongle (provided) that
lets you connect a footswitch. There’s also
a Kensington lock receptacle on the back,
and five rubber feet on the bottom of the
plastic chassis.
About the joystick: it’s a sensible
alternative to really small wheels, which
might have taken up too much room. From
the factory an upwards push generates
CC1 (modulation wheel) and a backwards
pull CC4 (foot controller). Left and right are
pitch bend down and up. Two things to be
aware of: first, that it’s sprung on both axes,
so you can’t ‘park’ your virtual mod wheel in
an above-zero position; second, because of
a physical limit in diagonal movement, you
can’t achieve max modulation value at the
same time as max pitch bend. One element
has to give, slightly.
Finally, the unusually named ‘Part 2’
feature. Using a dedicated pair of keys
to the left of the keyboard, you can dial
in modifications to the GX Mini’s MIDI
output. Momentary transposition by one or
more octaves (up or down) is one option,
supplementing the dedicated octave keys’
latching behaviour. It can also transpose
in semitones, or transmit note and other
messages on another user-selectable MIDI
channel. Or a combination of all three. In
practice all this proves more useful than
I expected it to be, but more interesting still
is that the temporarily modified output can
add to the normal output rather than just
replace it. Amongst other things that makes it
viable, and easy, to play thick two-hand pad
textures or high multi-octave string lines that
might be impossible to achieve otherwise
with the very limited keyboard compass. The
implementation is intentionally robust too,
so you can (for example) hold a bass note
transmitting on MIDI channel 1, hit the Part 2
button, and knock out a quick brain-melting
solo several octaves above it, in fifths, on
MIDI channel 3. Like you do.
Honey I’m Home
MIDI controller keyboards are a bit like
shoes, or bicycles: no one type or model
cuts it for every conceivable task. These
should be just the ticket for DAW-centric
home use and (in the case of the GXP)
more modest hardware setups. As
reliable, class-compliant all-rounders
they’re pretty much rock solid. I loved the
way that everything worked exactly as
advertised, and is so well documented:
neither is a given with some other
manufacturers’ products. But then, should
you want to explore it, there’s a swathe
‘value-added’ functionality in the form of
the nifty but not over-complicated DAW
control, GX Mini’s Part 2 scheme, the
GXP’s Repeat Engine, expression pedal
support, and hardware integration
with Nektarine.
Competition in the controller market
is almost mind-bogglingly extensive and
varied, and fierce with it, but these stand out
as quietly capable keyboards that should
prove nice to work with long-term, and that
offer some individuality to boot.  
$ GXP 88 $299.99, GXP 61 $229.99, GXP
49 $189.99, GX Mini $69.99.
W www.nektartech.com
 C OM PE T I T ION

Win!
Milab
VIP-60
Worth $3282
M
ilab’s pioneering 1984 microphone, the VIP-50, is
aptly named: its advocates include some of the most
prestigious names in popular music, from production
legends Bruce Swedien and Quincy Jones to singers Ray Charles,
Michael Jackson, Jeff Buckley, Tori Amos and many more. But
it’s not just the distinguished user base that sets it apart. While
the vast majority of microphones are based around circular
capsules, the VIP-50 employs a Milab-designed rectangular
affair; a geometry that both minimises undesirable resonances
and results in a useful degree of polar pattern asymmetry. In
practice, this means that the VIP-50 has a wider pickup pattern in
the horizontal plane than the vertical. This makes it an unusually
flexible mic, allowing the engineer to be highly selective about
what sounds they pick up, and what they reject.
Thirty-seven years after its introduction, Milab have released
an update, called the VIP-60, which brings the original’s unique
attributes and acclaimed sound up to date with a more modern
(but still rectangular!) capsule, and improved self-noise specs.

To enter, please visit:

https://sosm.ag/milab-comp-0422
In all other respects, though, it’s the same mic that has graced
countless hit records, with the same choice of five polar patterns,
two pad settings, and two high-pass filter turnover frequencies.
When we reviewed the new VIP-60 a couple of months ago,
SOS Editor in Chief Sam Inglis found it a joy to use, describing it
as “never less than a good vocal mic, and frequently an excellent
one,” capable of delivering “a lifetime of high-quality recordings”.
All of which is excellent news for the winner of this month’s
competition! To be in with a chance to own this outstanding
microphone, simply follow the URL shown, and answer the
questions there, by Friday 6th May 2022. Good luck!  

Prizes kindly donated by Milab

W www.milabmic.com

76 April 2022 / w w w. s o u n d o n s o u n d . c o m
 Engineer
Emm
win

An equalizer is probably the tool you use most while mixing and
mastering, so you need the best of the best! With FabFilter Pro-Q 3,
you get the highest possible sound quality and a gorgeous,
innovative interface with unrivalled ease of use.

Distributed by Music Marketing Inc. To find a dealer visit www.musicmarketing.ca

 ON TE S T
Sonic Potions/Erica Synths
LXR-02
Drum Machine
Sonic Potions and Erica
Synths team up to create
a truly unique drum machine.
RORY DOW
T
he Sonic Potions LXR was an
affordable DIY project released
back in 2014. For those with
the requisite soldering skills, it offered
a six-voice FM/subtractive sound engine
with a sequencer, four different synth
algorithms, MIDI, and open-source code.
Sonic Potions sold the component kit to
build the machine and a separate acrylic
enclosure. Erica Synths subsequently
offered an all-metal enclosure.
Not everyone has the time or
inclination to solder their instruments,
so Erica Synths and Sonic Potions have
teamed up to create the LXR-02. It is not
available as a DIY kit, and the operating
78 April 2022 / w w w. s o u n d o n s o u n d . c o m
system isn’t open-source, but otherwise
the LXR-02 doesn’t stray far from the
original concept. It’s a six-voice digital
synthesis drum machine with an onboard
sequencer. There’s plenty of tweaks and
improvements, of course, the addition
of volume faders for each voice being
the most obvious. Overall the general
design philosophy seems to be one of
accessible tweakability. Rarely is anything
more than a click or two away, and you
can edit everything without stopping
the sequencer.
Synthesis
The LXR-02 uses a combination of
subtractive and FM synthesis. There
are four main algorithms: Drum, Snare,
Cymbal/Clap and Hihats. The synthesis
parameter pages are cleverly grouped so
that no matter which type of sound you are
editing, you can hit a button and go to the
relevant page for that algorithm. No matter
whether you’re editing a snare or a hi-hat,
if you hit the Filter button, you always go to
the filter page for that algorithm.
The first three channels use the
Drum algorithm, a mix of simple FM and
subtractive synthesis. The primary use is
probably kicks and basses, but they can
easily make toms, percussion, snares,
laser zaps, etc. The Snare channel is more
focused and uses subtractive oscillators
and noise to create snares and percussive
hits. The Cymbal and Hihats channels
use a more complex FM setup to create
metallic, enharmonic sounds. The Open
and Closed Hihat channels use the same
synthesis parameters except that each
can set a different amplitude decay time.
They can then be sequenced separately
but will choke each other.
Mostly, the synthesis modules used
in each drum algorithm are similar.
The oscillators create classic analogue
waveforms: sine, triangle, sawtooth,

rectangle (aka square), noise and PWM.
The filters are a 2-pole state-variable
design with low-pass, high-pass,
band-pass, unity gain band-pass, notch,
and peak filter types.
The envelopes are a simple AD
design but offer variable exponential and
logarithmic curve controls and a repeat
parameter that causes rapid retriggering
— useful for synthesizing clap sounds.
The Drum and Snare algorithms offer
two envelopes, one for amplitude and
one for pitch modulation. The pitch
envelope can also automate a long list
of destinations with optional velocity
sensitivity. The Cymbal algorithms offer
only one amplitude envelope but still
provide velocity modulation over many
possible destinations.
All algorithms offer a Click page where
you can alter the transient using either an
additional short pitch envelope, adjusting
the starting phase of the main oscillator or
mixing in a transient sample. The samples
are baked into the firmware and cannot
be replaced but offer variants such as
acoustic kick, rim-shot, hi-hat, clap, tom
and finger snap. The samples are short
but valuable for adding a consistent
transient to any sound.
Each voice has an LFO, which can
free-run or sync to an internal or external
clock. It can only control one parameter,
but that can be any parameter (including
effects) on any voice. Effectively then,
you have six freely assignable, global
LFOs. Possibly my favourite detail is that
you can assign any voice to reset the
phase of the LFO. It means you could
have a kick drum retriggering the LFO on
a percussion track which is controlling
some wild modulation. This interaction
between sequencer tracks makes for
some exciting and dynamic patterns.
A big part of the LXR-02 sound is
overdrive. Both the filter and the output
can be overdriven for each drum voice,
resulting in sounds ranging from clean
and polite to downright filthy. If that still
sounds too pleasant, you can add some
bit-depth reduction, which will cause
anything from a touch of aliasing to total
annihilation. Bit-depth reduction is also
available on the master mix outputs. In all,
there is a ridiculous number of ways to
roughen up the sound.
Voice & Performace Modes
The LXR-02 works in two modes: Voice or
Performance. Voice mode is where you do
synthesis and sequencer programming.
Performance mode is for live tweaking.
Voice mode focuses on one voice
at a time. Choose the voice using the
buttons underneath the sliders, then pick
a synthesis page group such as Osc, FM
or Filter. The screen will show up to four
parameters which can be altered using
the four endless encoders underneath. If
more than four parameters are needed,
the data encoder can access more pages.
Because the screen is relatively small,
the parameters have short three-letter
codes. COA is coarse tune, FRQ is filter
frequency, DEC is envelope decay, etc.
Most of these are easy to figure out,
but if you find yourself scratching your
head, a quick press of the data encoder
will toggle to a single parameter screen,
where the full title is displayed.
The 16 buttons will show the sequencer
pattern for the current voice. If the pattern
is longer than 16 steps (up to 64), you can
move between pages of 16 steps using
the Bar forward and backward buttons.
This method of keeping everything tied
to the selected voice is intuitive and easy
to grasp. My one gripe here is that the
busy layout of the buttons makes muscle
memory challenging to develop. You have
to look carefully at the panel legends
before pressing a button. Luckily, the
white-on-black colour scheme is easy to
read, and the button LEDs aren’t so bright
they’ll blind you, even in a dark room.
Performance mode is for live
jamming. The Voice buttons below the
sliders become mute buttons. The first
seven pattern programming buttons
become trigger rolls, allowing you to
play quantised rolls, at various rates,
over the playing pattern. Other global
effects such as sample-rate reduction,
pattern-shuffle, and sequencer tempo are
all accessible in this mode. You can still
access individual voice parameters in this
mode, so tweaking sounds on the fly is
still achievable.
Perhaps the most enticing Performance
mode feature is Morphing. Every pattern
can load a second kit, and morphing will
smoothly transform every parameter of
every voice from the pattern kit to the
morph kit with just a single knob. The
possibilities here are mind-blowing. You
can load two completely different kits
and enjoy the radical chaos of blending
between them, or save a copy of the
current kit with subtle changes and use it
to tweak a few different sounds at once.
Live editing of a pattern and its sounds
might feel dangerous in a live setting,
but the LXR-02 has a handy trick up its
sleeve. You can hit the Reload button at
Sonic Potions/Erica
Synths LXR-02
$589
PROS
• Four robust and flexible FM/
subtractive drum synthesis engines.
• A capable little sequencer.
• Four outputs (configurable as two
stereo, four mono or a combo).
• Good MIDI spec.
• Solid construction.
CONS
• Some buttons are a bit ‘clacky’.
SUMMARY
The LXR-02 is a pure synthesis drum
machine with a massive personality.
It combines FM synthesis with clever
sequencing and well-designed features
to create a compelling package. It can
go shoulder to shoulder with any of the
big boys and stand proud.
w w w. s o u n d o n s o u n d . c o m / April 2022 79
 ON TE S T
SONIC POTIONS/ERICA SYNTHS LXR-02
The LXR-02 measures 230 x 145 x 70mm and weighs in at a mere 810g.
any point, and the saved kit and pattern
will reload from the current project.
So you can deviate from any sound or
pattern safe in the knowledge that you
can return to a known state.
Outputs & Effects
Each voice can be routed to any of
the four outputs or a digital effects
processor. The audio outputs can
be used in stereo pairs, mono, or
a combination if required. If configured
in stereo, each voice has a panner that
can be manually set or automated using
LFOs, velocity or sequencer automation.
There are four digital effects too.
drive, ring modulator, compressor, and
delay, although only one can be active.
Any voice sent to the effects bus will
be processed then sent to an output of
your choice.
Drive offers three different algorithms:
tube, wave-folding and clipping, all
with adjustable drive strength, wet/dry
ratio, tone, and make-up volume. The
ring modulator offers robot beatboxing
by multiplying the input signal with
an oscillator capable of various
waveforms and an adjustable frequency.
The compressor provides simple
compression with ratio, threshold, attack,
decay, and make-up gain.
80 April 2022 / w w w. s o u n d o n s o u n d . c o m
The delay effect, I suspect, will be the
most used. Mono and stereo ping-pong
modes are available, and the delay times
are set in two ranges: Low for modulation
effects like chorus and flanger (1ms to
20ms) or High for more traditional delays
(20ms to 0.7s).
I particularly enjoyed the drive and
delay effects — the only pity being they
can’t be used together. The Clip mode on
the drive effect can introduce delicious
feedback that interacts with the input
signal in lovely ways. And the delay effect
has a very analogue sound, especially
when the delay time is automated using
LFOs or sequencing.
Sequencing
The LXR-02 has everything you need
to make sophisticated sequences. Each
project can contain 64 patterns and 64
songs. Each pattern can have up to 64
steps and its own kit (optionally loaded
when switching patterns).
Every pattern step can contain pitch
information, velocity, trigger probability,
a flam multiplier, time shift and two
automation values. The automation values
can be any parameter from the synthesis
algorithm of any voice. You can program
these individually (if you want different
parameters for each step) or record
tweaks as the sequencer plays.
Every voice track within a pattern
can have a different length. And for
quick inspiration, there’s a Euclidean
pattern generator. Just enter the length,
number of steps and rotations value,
and the pattern will be auto-generated
for the current voice. You can even use
the pattern generator in a live context,
reverting to the saved pattern whenever
things stray too far.
The LXR-02 can transmit MIDI so that
you can layer up other sounds from an
external sound module. There are no
dedicated sequencer lanes for this, so
you have to use one of the voice lanes.
The LXR-02 will also send program
changes when you change patterns
(and respond to them if required). All of
this is optional, of course. You can turn
off transmission of MIDI notes, control
changes, program changes and clock
individually in the settings.
In terms of sync, there are MIDI and
analogue clock options, and the LXR-02
can act as clock source or destination.
The resolution of the analogue clock is
adjustable, so it should work with just
about any configuration.
Song mode allows you to build
a playlist of patterns. It’s a simple list.
You can insert more patterns anywhere
 Want to make your vocals
An entire shine, your guitars dance,
your drums smack, and your
master bus raise the roof? 22

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 ON TE S T
SONIC POTIONS/ERICA SYNTHS LXR-02
Round The Back
Despite its small size, the LXR-02 is packed
with connectivity. There’s a 12V DC power
input (adaptor is included) with on/off switch,
analogue clock input, reset and output,
Micro SD card slot (for saving Projects
etc), MIDI in and out on USB and DIN, four
unbalanced quarter-inch TS outputs, and
a 3.5mm stereo headphone output.
in the list, and there are a couple of
‘special’ pattern types: End, for the end
of the song, and Loop, to loop back to
the beginning. I cannot imagine anyone
seriously writing a song with it, but
combining a few patterns into a longer
chain is a valuable function.
External Control
For those who prefer to sequence
externally, the LXR-02 has
a comprehensive MIDI implementation.
Each voice could have a MIDI note on MIDI
channel 10. Or, if you choose, each voice
can have a separate MIDI channel that can
play chromatically. For those who want to
use an external MIDI keyboard to program
the LXR-02’s internal sequencer, you can
set up a global MIDI channel to control
whichever voice is currently selected.
ALTERNATIVES
For digital FM drum synthesis, there isn’t
a massive number of options. Roland’s
TR-8S got an FM synth engine in an
update back in mid-2020. The Twisted
Electrons Blast Beats is a six-voice
drum machine based on the OPL3 FM
chip used in old Soundblaster cards,
which looks fun. The Sonicware Liven
XFM is a four-track FM groovebox with
interesting morphing features. Lastly, the
Nord Drum 3P might be worth a look as
it has many synthesis types, including
FM, but no sequencer.
82 April 2022 / w w w. s o u n d o n s o u n d . c o m
A detailed MIDI CC chart offers
external control over almost every
synthesis parameter. MIDI CCs are
received on the global MIDI channel,
which does present a problem if you want
to sequence externally because you have
to send your note triggers and MIDI CCs
on different channels.
The LXR-02 can also route MIDI
between its USB and MIDI DIN ports,
turning any port output port into a thru
port. For example, you can route USB
input to MIDI DIN output. Or MIDI DIN
input to MIDI DIN output. Or even merge
both the USB and MIDI DIN inputs and
pass them to the MIDI DIN out. This
flexibility is a nice touch that allows you to
fit the LXR-02 into just about any setup.
Conclusion
I like the LXR-02 a lot. Its physical size
hides a monster drum machine capable of
tearing your productions a metaphorical
new one — if you’ll pardon the expression.
As a purely digital device, it manages to
avoid the analogue clichés that are so
common whilst still managing to sound full
and fat.
FM synthesis, in particular, is
brilliantly suited to electronic percussion
generation. The synthesis algorithms
are complex enough to offer a broad
palette of sounds, but the LXR-02 still has
a unique signature. It’s a ‘highly electronic’
sound, by which I mean that you won’t be
synthesizing any realistic drum sounds
here. Overall, the sounds are dynamic
and transient. It’s crisp and clear when
you want it to be, or filthy and muddy and
swimming in feedback. I’d recommend
the LXR-02 to any electronic musician
who embraces pure synthesis. There are
no samples to hide behind, but it simply
doesn’t need them.
The internal sequencer is impressively
feature-packed. Probability, flams,
different track lengths, automation
per step — these are things you might
not expect to find on a machine of
this size and price point. Having four
outputs configurable in any way is also
fantastic. There are drum machines at
twice the price with only a single pair
of outputs (shame on them!). The MIDI
implementation is equally impressive, with
USB, MIDI DIN and enough config options
to satisfy almost any situation.
Of course, nothing is perfect. The
compact front panel squeezes a lot of
buttons into a small space, and I found
myself pressing the wrong button too
often. The larger round buttons also
have a clacky feel, which I wasn’t fond
of. But, these minor imperfections don’t
matter. I haven’t been this excited by
a drum machine for years, and I can’t
recommend the LXR-02 enough. If you
make any genre of music that benefits
from synthesized drum sounds, whether it
be techno, electro, electronica, ambient,
or a billion others I’ve yet to discover,
then you should give the LXR-02 some
thorough consideration. It’s a beast.  
$ $589 including VAT.
W www.ericasynths.lv
 ON TE S T
Apogee
Duet 3
USB Audio Interface
Apogee’s long-serving desktop audio interface gets a thorough makeover.
SAM INGLIS
W
hen desktop audio interfaces
first appeared, they were
usually marketed as budget
options for hobbyists. Apogee overturned
that perception with the Duet, a stereo
interface that majored on sleek design
and superior sound quality. The original
Firewire version was superseded by the
USB Duet 2 in 2012, and so successful
has this been that it’s only now being
retired in favour of a new Duet 3.
Third Movement
The Duet’s positioning as a premium
product has always depended in part
on its attractive industrial design, and
the Duet 3 has been reinvented on
cutting-edge aesthetic principles. The
aluminium body is topped by a sheet
of obsidian-like black toughened glass,
beneath which various LEDs are invisible
84 April 2022 / w w w. s o u n d o n s o u n d . c o m
when not illuminated. The only moving
part is a large rotary control, which sheds
a mysterious purple glow from its nether
regions. The overall effect is striking and
very much of a piece with recent Apple
consumer products, or possibly the
interior of a spaceship.
The Duet 3 still connects to the host
computer or iOS device using USB,
but employs the now-current Type C
connector rather than the older Type
B. It no longer comes with an external
power supply; instead, a second Type
C socket is built in to allow powering
from a dedicated USB source if your
laptop can’t provide the necessary juice,
or if you want to use it with an iPhone.
Apogee describe the Duet 3 as a two-in,
four-out interface; in other words, it’s
a stereo device where the headphone
output can be addressed independently
from the main output.
As before, the only conventional audio
Apogee Duet 3
$649
PROS
• Elegant, minimalist design.
• Excellent sound quality.
• Good-quality carry case supplied for
portable use.
• Optional Duet Dock as an alternative
to the flying cables.
• Built-in DSP allows you to record with
EQ and compression.
CONS
• Limited iOS support at present.
• Loses the Duet 2’s touch buttons.
• Native version of the Symphony ECS
Channel Strip plug-in is a cost option.
• Balanced line inputs on XLRs only.
• DSP can only be used in the
record path.
SUMMARY
Apogee’s Duet 3 makes a visual and
sonic statement. It may not be the
most affordable desktop interface, but
it’s undoubtedly the most stylish!
connector actually built into the Duet
3 is a single mini-jack headphone
socket. There are two ways of
accessing the main inputs and outputs.
One is to use the supplied flying cable,
which locks securely into a compact
multi-way connector on the back of the
interface. The other is to purchase the
optional Duet Dock, a plastic desktop
chassis that mates with the Duet,
rather spoiling its elegant minimalist
appearance, and breaks all of its
I/O out onto dedicated connectors.
Adding the Duet Dock is also the only
way to get a full-sized headphone
socket. This mirrors the mini-jack on
the front panel so you can connect
two sets of headphones at once,
albeit without any control over their
relative levels.
A welcome improvement,
compared with the Duet 2, is the
provision of separate XLR and
quarter-inch sockets for the inputs,
both on the flying lead and the Dock.
This doesn’t allow you to record more
than two sources at once, but does
mean you can leave mics plugged
up at the same time as guitars. Note,
though, that the quarter-inch sockets
are unbalanced and designed for
guitar use; if you want to get the best
from balanced line-level sources, you’ll
need to connect these to the XLR
inputs. I’m not awfully keen on this
arrangement, as it makes it possible
to apply phantom power to things
that shouldn’t receive it, as well as
requiring jack-to-XLR cables for most
synths, which many people won’t have.
Controlling Influence
Many desktop interfaces are designed
to be used without any form of
software control panel, making all
of their functionality controllable in
hardware, but that’s not the case
with the Duet 3. It’s class compliant,
so requires no driver on Mac OS, but
top-panel control is restricted to input
gain, headphone and main output
volume, and mute. Most of its features
must therefore be controlled from the
attached computer.
All of Apogee’s USB interfaces
are now handled by a single package
called Apogee Control 2, which
detects what interface is attached,
invites you to update firmware
when necessary, and displays
the appropriate control panel.
This is clearly laid out and mostly
self-explanatory. It’s also described
clearly in the PDF manual, which is
lucky, because the Hover Help window
does not yet offer any hovering help.
Other than that, it ran flawlessly on my
Mac; I did not test it on Windows.
Apogee Control 2 makes visible
VIP-60
the mixer built into the Duet 3, which is
fed by two mono (but linkable) inputs
derived from the mic or guitar sockets,
“A GORGEOUSLY
and two stereo inputs derived from
your DAW’s outputs. The output of
this mixer can be routed to the main
SILKY SMOOTH
speaker outs and/or the headphone
output; alternatively, if you prefer,
these can be set to pick up one of
AND PRESENT
the DAW playback channels directly.
The mixer output can also be routed
VOCAL MIC”
to the DAW software inputs, allowing
software-generated sound to be – Recording Mag #2/22 –
re-recorded. Selecting this option
brings up a stern but very apposite
warning about the possibility of
generating feedback loops.
The more you work with this control
panel, the more you appreciate the
attention to detail that has gone
into the design of the Duet 3. For
example, the output volume control
is mirrored by the Mac OS volume
level, so you can adjust it using the
volume up/down keys on your Mac,
and it’s possible to switch this so
that the system volume controls the
headphone level while the Duet’s
knob adjusts speaker level, or vice
versa. Both the headphones and
speakers have independent mute, dim
and mono controls, and you can even
adjust the brightness of the Duet’s
LEDs from Apogee Control 2. What
is sadly missing in action, however, is
the Duet 2’s pair of assignable touch
buttons. This means it’s no longer
possible to engage the mono or dim
options directly from the Duet itself.
Audio quality in general, and mic
preamp design in particular, have
always been big selling points for
Apogee. In line mode, the XLR inputs
can be switched between +4dBu
and -10dBV sensitivity levels, while in
mic mode, they provide up to 65dB
of gain, adjustable in 1dB steps. This
is actually a smaller gain range than
was available on the Duet 2, but is still
wider than you’ll find on most desktop
interfaces, and perfectly adequate
for most of the Duet 3’s intended independentaudio.com
applications. You also get the bonus of
w w w. s o u n d o n s o u n d . c o m / April 2022 85
 ON TE S T
APOGEE DUET 3
a switchable soft limiter, polarity reversal
and a link button which allows the gain on
the two mic preamps to be ganged whilst
preserving any existing offset between
them. A further button in the input
settings area provides access to what is
probably the biggest new development in
the Duet 3.
In Print
Ever since Steinberg launched the VST
protocol back in 1998, people have been
telling me that the days of DSP built into
audio interfaces are numbered. Someone
forgot to tell the manufacturers, though,
because in recent years it has become
more rather than less ubiquitous. It’s
a major selling point of Apogee’s larger
Symphony Desktop, and, in a much more
streamlined implementation, is on tap in
the Duet 3 too.
There are various ways in which
onboard DSP can be presented to the
user. It can be normally be placed in the
monitor path, allowing performers to hear
a low-latency foldback signal processed
with compression and so on whilst the
engineer tracks an unprocessed signal.
Additionally, it can sometimes be made
available to DAW software to run plug-ins
at mixdown. In the Duet 3, however, the
onboard DSP is referred to as Print FX,
and as that name suggests, it can be used
only in the record path. In other words,
if you activate the DSP, it will always be
processing the signal that’s going to disk
as well as that going to the performer’s
headphones.
Like many other manufacturers,
Apogee package their DSP functions in
a plug-in-like interface. However, whereas
86 April 2022 / w w w. s o u n d o n s o u n d . c o m
the Symphony Desktop offers a choice
of half a dozen plug-ins, only one is
available here. Clicking on the Print FX
button in the Apogee Control 2 interface
opens a cheerful grey window with
purple controls, which is described as the
Symphony ECS Channel Strip. This was
developed in association with legendary
mix engineer Bob Clearmountain, and
he was responsible for the 25 or so
presets that are supplied. It occupies
an interesting middle ground between
the sort of utilitarian signal conditioning
that you find on most digital mixers, and
the colourful vintage-style processing
we effect from companies like UA.
A variable high-pass filter is bolted onto
an EQ section that is clearly designed
for broad-brush tone shaping in the
Pultec vein, while the compressor offers
only ratio and threshold controls, but
with the option to blend in the dry signal
to taste. The order of the two sections
can be switched, and the high-pass
filter can also be switched into the
compressor side-chain. Finally, there’s an
analogue-style Drive control as well as
a standard output level control.
It’s not possible to access the Duet 3’s
DSP directly from within your DAW, so you
can’t use the DSP version of Symphony
ECS Channel Strip as if it were a native
plug-in, UA-style. However, if you like the
sound, but not the thought of tracking
through it and having it printed on all your
recordings, you can invest in a native
version which is available in all the usual
plug-in formats and comes with the same
presets. This isn’t particularly expensive
as plug-ins go, but it is a cost option, and
I can imagine some Duet 3 purchasers
The Duet 3 can be
connected to instruments,
mics and loudspeakers
either using the supplied
breakout cable or the optional
Duet Dock.
feeling resentful at
having to pay to get
the native version of
something they already
own in DSP. In practice,
though, I actually rather
liked the way Apogee’s
DSP implementation
forces you to commit
to a sound at the
recording stage. As
you’d expect, Bob
Clearmountain’s
presets are very usable, and the generally
streamlined control set makes it easy to
home in on the settings you want.
As a bus-powered desktop interface
with built-in DSP, the obvious rivals for
the Duet 3 are UA’s Apollo Solo and
Antelope Audio’s Zen Go Synergy Core.
Those products both give you the option
to track through an almost endless
variety of simulated guitar amps, mic
preamps and vintage studio hardware,
as well as letting you record dry and
monitor wet. The DSP capabilities of the
Duet 3, though certainly worthwhile, are
obviously rather modest in comparison.
If you’re a podcaster or voiceover artist
who just wants a good ‘set and forget’
vocal treatment, Symphony ECS Channel
Strip will do the job nicely. If you want to
explore the outer reaches of guitar tone
or authentic emulations of vintage studio
hardware, it won’t get you there.
In Harmony
Apogee interfaces have always been
premium products, and the Duet 3 is no
exception. That price premium buys you
some tangible benefits. One I haven’t
mentioned so far is the supplied carry
case, which safely houses the Duet
and its breakout cable in a tough,
no-nonsense padded environment.
Another is the sound quality. The Duet
2 was already impressing in this respect
10 years ago, and the new model
improves on nearly all its specifications.
Dynamic range on the inputs and outputs
is now 119dB and 124dB respectively,
and the Duet 3 features a new “zero
Ohm” headphone amp that can deliver
much higher output levels than most
 Apogee Control 2
and the Symphony ECS
Channel Strip.

bus-powered
designs. If
you’re looking
for a portable
interface with
the kind of
sound quality
that’s usually
available only in
mains-powered
units, options are
relatively limited,
but the Duet 3 is
certainly one. It’s
also uniquely and
distinctively stylish, at least until you add also has an unwelcome tendency to spit have kept the Duet in the best-seller lists
the Dock. out thumps and splats when you shut the — and the Duet 3 is the best-looking and
However, there are also a few computer down. best-sounding Duet yet.
respects in which the Duet 3 is lacking All things considered, though, these
compared with its predecessor. I’ve are minor wrinkles, and some at least will
$ $649; Duet Dock $179; native Symphony
already mentioned the narrower gain be dealt with in future updates. In the ECS Channel Strip plug-in $49 when
range and the loss of the touch buttons, bigger picture, it’s always been classy bought with Duet 3.
which makes the Duet 3 less powerful looks and exemplary sound quality that W www.apogeedigital.com
as a hands-on monitor controller,
and another limitation concerns iOS
compatibility. At present, the Duet 3
will work with iOS devices, but the only
control available is that offered by the
encoder. Settings that, on a computer,
are only available in Apogee Control
2 cannot yet be adjusted from an iOS
device. Until the promised control app
is released, this probably makes the
Duet 3 usable as a playback device but
of limited value for recording to an iPad
Get the skills
or similar.
I also feel that the minimalism of the
Duet’s physical design undermines its
to produce
serious results.
intuitiveness in some small ways. For
example, the purple LEDs around the
wheel do double duty, showing you
which parameter is currently assigned
to the wheel and flashing if something
Explore Berklee Online’s vast offerings in music production
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are some important settings that are
visible in Apogee Control 2 but not Online Degree Programs 12-Week Online Courses
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use, this means it’s quite often necessary
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tab away from your DAW to Apogee Learn more at online.berklee.edu 1-866-BERKLEE
Control, which doesn’t always make for
a seamless user experience. The Duet 3

w w w. s o u n d o n s o u n d . c o m / April 2022 87
 ON TE S T
PHIL WARD
D
espite writing about monitors in
Sound On Sound for longer than
I care to remember, PMC have
never before found themselves the subject
of my witterings. That is about to change
because, dear reader, I bring you the new
PMC6-2. The 6-2 is the middle model of
a range of three new active monitors from
PMC that covers nearfield to all-but-main
monitor applications, the latter particularly
when paired with one of the two equally
new XBD active subwoofers.
PMC describe the 6-2 as a nearfield
monitor, but in dimensional terms I feel
it falls into the overlap between midfield
and nearfield. That’s a convoluted way of
saying the 6-2 is biggish for a nearfield
monitor, but not so big that you’d take one
look at it and immediately conclude it won’t
fit in your studio space. At 21.3kg each,
however, the 6-2s are undoubtedly heavy,
so any mounting hardware will
88 April 2022 / w w w. s o u n d o n s o u n d . c o m
PMC6-2
Active Three-way Monitors
If anyone knows how to make a transmission-line speaker,
it’s PMC...
very much need to be up to the job. To
that end, PMC offer dedicated stands and
a hanging yoke.
In terms of its driver complement, the
6-2 is a three-way system comprising
two nominally 150mm (6-inch),
extended-excursion, paper-diaphragm
bass drivers; a 55mm (2-inch) textile
midrange dome; and a 27mm (1-inch) textile
dome tweeter. As perhaps you will have
ascertained from glancing at the pictures,
while the bass drivers and tweeters appear
relatively conventional, the midrange dome
is a particularly unusual-looking device. I’ll
get around to explaining the reason for its
appearance a little further down. The active
amplification of the 6-2 comprises ICE
Power Class-D modules rated at 400 Watts
for the tweeter, midrange driver and each
bass driver. The 6-2 definitely is not short of
amplifier power.
As with all PMC’s pro monitor offerings,
the look is businesslike black with an
almost complete absence of any aesthetic
frippery. Only the polished aluminium
circumference of the bass driver chassis
and the honeycomb motif embossed in the
plastic moulding that surrounds the tweeter
and midrange drivers offer any contrast
to the no-nonsense look. And even the
honeycomb motif is a visual reference to
PMC’s ‘Laminair’ transmission-line vent-flow
conditioner. Speaking of transmission lines,
not having written about PMC in these
pages before means that I’ve not had the
opportunity to turn my thoughts towards
the company’s Advanced Transmission Line
approach to low frequencies. So before
I move on to describing the 6-2 in more
detail, I’ll take a detour...
Loading Time
There are five basic techniques for
managing the low-frequency output of
a moving-coil transducer: open baffle, horn
loading, closed box (sometimes called
‘infinite baffle’), reflex loading (in which I’d
include ABR and coupled-cavity systems),
and transmission line. In order of popularity
among designers the ranking goes: reflex,
closed, horn, transmission line (although it’s
a close run thing for third place I think) and,
bringing up the rear, open baffle.
The problem that all of these techniques
aim to fix is that, while speakers have to
fit into human-sized spaces, given the
atmosphere we humans evolved to thrive in,
the wavelength of sound at low frequencies
is very long (6.9m at 50Hz). The upshot is
that when a naked moving-coil driver tries
to reproduce a sound with a wavelength
greater than its physical diameter, the
reverse-polarity acoustic energy radiated
from the rear of the diaphragm will cancel
the energy radiated from the front. The
cancellation gets more effective as
frequency falls but the result is typically
low-frequency silence, or as near to silence
as makes no difference. Reflex, closed-box
and most horn-loaded speakers aim to
solve the problem by isolating the driver’s
PMC6-2
$8500
PROS
• Immense monitoring ability at all
volume levels.
• Striking midrange performance.
• Extended yet accurate and
punchy bass.
• Soundalign constitutes a huge
advance in monitor configuration.
CONS
• None.
SUMMARY
The PMC6-2 without doubt joins
a select group of monitors in
the nearfield/midfield sector
that currently define the best of
electro-acoustic design.
rear-radiated energy from its
front-radiated energy through
the addition of a box over the
back of the driver. Thanks
to the box, the rear energy
can’t now escape to cancel
the front energy and — voila!
— audible bass is the result.
The fundamental problem of
using a sealed box to ‘hide’ the
driver’s rear energy is that doing
so adds the springiness of
the air in the box to that of the
driver’s suspension, and that
pushes the driver’s fundamental
resonance, below which its Diagram 1:
efficiency plummets, up into Cross-section
musically important frequencies. of the PMC6-2’s
transmission lines.
The bigger the box, the less
this is a problem, but of course
that brings us back to the need to make
speakers fit in human-sized spaces.
The trick that reflex-loading adds to the
concept of isolating the rear radiation in
a box is a Helmholtz resonance. It comprises
the ‘slug’ of air in a pipe (the port) bouncing
on the compliance of a contained volume of
air (the cabinet), and it works by reversing
the polarity of the driver’s rear radiation
over a narrow band of frequencies (around
the tuning frequency). The reversed-phase
port radiation now reinforces rather than
cancels the driver’s forward radiation and
a useful gain in low-frequency bandwidth
(or power-handling or efficiency, or
a combination of the three) results. While
reflex loading, all things being equal (well,
box volume and efficiency), will result
in more extended bandwidth than the
closed-box option, it has a snag in that both
the reversal of phase and the resonance
result in time-domain issues. Phase reversal
implies a minimum time delay of however
long half a cycle takes at the frequency in
question (40Hz, for example is a typical port
frequency, and half a cycle is 12.5ms), and
the nature of a resonance is that it rings
on after its stimulus signal has stopped. So
a reflex-loaded speaker typically has trouble
following the ups and downs of a stimulus
signal as accurately as a closed-box
speaker — it’s always playing catch-up
or overshooting.
Transmission Accomplished
The transmission-line solution to the
problem of bass, as employed on the
PMC6-2, might look much like a reflex
system — architecturally, it’s a box with bass
drivers and holes — but it’s conceptually
a different beast entirely. Behind each
bass driver is a folded, foam-lined labyrinth
created by dividing panels within the
enclosure. The two labyrinths (one per bass
driver) end in the twin exit mouths on the
front panel.
The function of the transmission-line
labyrinth in terms of its contribution to
low-frequency performance has two
elements. Firstly, the labyrinth and its foam
lining aim to fully absorb the rear radiation
from the bass drivers as it travels down the
line, stopping it from interfering with the
forward radiation (or bouncing back through
the bass driver diaphragm). The concept of
dealing with the rear radiation by making
it disappear is of course not dissimilar to
that of reflex and closed-box speakers.
The difference is that a transmission line
presents an entirely resistive load to the
driver, whereas closed and reflex boxes
present a reactive load — the air spring
in the former case, and the air spring and
reflex port air mass in the latter case.
However, absorption in a line of finite
length will only work down to a specific
frequency, and that’s where the second
element of the transmission-line labyrinth
comes into play. Below the frequency
where absorption stops, the labyrinth sets
up a quarter-wavelength pipe resonance
that acoustically loads the bass drivers,
significantly reduces their diaphragm
excursion, and results in output through
the line exits that reinforces the forward
radiation. The frequency of the resonance
is defined by the length of the labyrinth
— in the case of the PMC6-2 that’s 1.3m.
However, a secondary (and really rather
useful) effect of the labyrinth absorption is to
slow the speed of sound in the line, which
results in a longer effective length and,
w w w. s o u n d o n s o u n d . c o m / April 2022 89
 ON TE S T
PMC6-2

Diagram 2.

consequently, a lower
quarter-wave resonance
frequency. In the case
of the PMC6-2, the
quarter-wave resonance
frequency is 43Hz, which
means the effective
length of the labyrinth is
actually around 2m and
that the speed of sound
in the labyrinth drops
from the free-air 343m/s
to approximately 223m/s.
The electro-acoustic result of all this on the technique is expensive to implement and ‘lid’ off a transmission-line monitor like the
PMC6-2 is low-frequency bandwidth that fiendishly tricky to optimise in practice (one PMC6-2 and, as illustrated in Diagram 1,
extends down to 33Hz at -3dB combined PMC employee described it to me as “like the internal components and layout look
with low distortion, impressive volume-level juggling water”). The expense is significant relatively simple, but they hide significant
capability and very well-controlled due to the complexity of cabinet work complexity and result from huge design
time-domain performance. that construction of an internal labyrinth effort and technical know-how.
And with regard to time-domain demands, and the technical difficulty is due
performance, while the purely resistive load to the number of variables and degrees
Diagrams
presented by the transmission line in the of freedom inherent to the technique. For I’ve endeavoured to illustrate the principles
fully absorbed portion of the low-frequency example, balancing the labyrinth dimensions of transmission-line loading with some
band implies no additional time-domain and internal damping so that upper-bass PMC6-2 FuzzMeasure data displayed in
effects beyond those inherent to the drivers rear radiation from the driver is absorbed Diagram 2. The orange curve shows the
(ie. no extra group delay), the introduction while the quarter-wave resonance isn’t 20Hz-1kHz frequency response data from
of a low-frequency quarter-wave resonance over-damped is not a trivial task. Take the a measuring mic positioned very close
sounds like it ought to be more of an issue
in that respect. After all, the time-domain
effects inherent to the Helmholtz resonance
of reflex-loaded speakers are well known
and often discussed. The transmission-line
difference is in the detail. In order to work
effectively, the port resonance of a reflex
speaker needs to have a high Q, which
means its frequency response contribution
is limited to a relatively narrow bandwidth
and that its time-domain effects can be
significant. Peak group-delay numbers well
past 50ms are not unheard of. A skilfully
designed transmission line, however,
employs a low-Q quarter-wave resonance
that contributes over a far wider bandwidth
than a reflex Helmholtz resonance, and
has significantly lower impact in the time
domain in terms of group delay (which
peaks in the 6-2 at 15ms) or overshoot and
ringing effects.
Evangelists for transmission-line loading
argue that the technique offers a superior
solution to the low-frequency problems
of moving-coil speakers than either reflex
or closed boxes. If you look around the
world of high-performance speaker design,
however, in either pro or hi-fi sectors, you’ll
see that transmission-line designs are very
much in the minority. The reason for this is
not that transmission-line evangelists don’t
have a strong case — they do — it’s that the The Soundalign web interface gives you comprehensive control over the monitors via an Ethernet connection.

90 April 2022 / w w w. s o u n d o n s o u n d . c o m
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 ON TE S T
PMC6-2
Diagram 3: Midrange driver and waveguide construction.
(<0.5cm) to one of the 6-2 bass drivers.
The most interesting feature to note is the
shallow dip in the driver’s high-pass roll-off
centred just below 40Hz. This dip, if you
examine it closely, actually extends from
about 25Hz to just under 60Hz — more
than an octave. I’ve marked this in the
diagram with a grey region. It reflects
the band over which the quarter-wave
resonance of the transmission line results
in output and loads the driver, reducing its
diaphragm displacement. Close-mic driver
measurement of a reflex-loaded speaker
will show a similar kind of dip at the port
resonance, but the dip will typically be
much sharper (higher Q) and extend over
a significantly narrower band. The blue
curve of Diagram 2 shows the 20Hz-1kHz
frequency response data from a measuring
mic positioned 50mm down one of the 6-2’s
honeycomb-shaped port flow conditioner
holes. It very clearly shows the broad
hump of the quarter-wave transmission
line resonance centred on the same
frequency as the dip in the orange driver
curve. The blue curve also shows a hint
of the change in transmission-line mode
at around 150Hz, where it becomes fully
absorptive and its output drops. The rise
in output towards 200Hz is the leakage
of output from the forward radiation of the
bass drivers into the measuring mic (These
sorts of close-mic measurements tend to
be usefully illustrative only over a relatively
narrow band of frequencies. Sadly, you can’t
drive the transmission line without having
the bass drivers working, so ‘crosstalk’ will
always happen).
The Rest Of It
I’ve devoted many words to the 6-2’s
transmission-line loading, so it’s high time
to move on, not least because the 6-2
also sports a couple more particularly
unusual and interesting elements. Firstly,
as I mentioned, the 6-2’s 55mm-diameter
midrange dome is an unusual-looking
92 April 2022 / w w w. s o u n d o n s o u n d . c o m
driver. In constructional terms, behind the
dome, the driver is relatively conventional
(apart from the fact that dome midrange
drivers are generally outliers), however
the plastic moulded waveguide in which
the dome sits is anything but. Waveguides
appended to the front of dome drivers have
two functions. First, they help to control and
define the driver’s dispersion characteristics,
and second, they can provide a degree of
horn loading that increases the radiation
impedance the driver ‘sees’, which
consequently increases its efficiency.
That extra efficiency can be deployed as
increased power handling, and/or to allow
the driver to operate to a lower frequency
(a more efficient driver run at an
attenuated level can be used over a wider
bandwidth). The diameter, depth and profile
of a waveguide are all significant in terms
of the effect on dispersion and efficiency
— both of which may also vary significantly
with frequency.
In choosing to run with a relatively
small midrange dome in order that it
can reach further up the audio band, the
6-2’s designers needed the midrange
waveguide to offer good efficiency gain at
lower frequencies so that the driver could
also reach down to a sub-400Hz bass/mid
crossover. They found that a hyperbolic
waveguide profile provided the required
lower-frequency efficiency gain, however
it didn’t offer the right high-frequency
dispersion characteristics. Conversely, an
exponential waveguide profile gave the
required higher-frequency dispersion but
not the efficiency gain at lower frequencies.
The solution was to design a waveguide
that incorporated both hyperbolic and
ALTERNATIVES
If the PMC6-2 is on your list of possibles
you’ll probably also want to think about
the Dutch & Dutch 8C, Kii Three,
Ex Machina Pulsar, Genelec 8351B,
Amphion Two18 (plus amp), ATC
SCM25A and PSI A23M for a start. You
might also want to think about just how
fortunate you are!
exponential elements, and that explains
the unusual shape. The eight inclusions
around the dome add hyperbolic regions to
a base exponential profile. It’s an unusual
and innovative solution that PMC calls
‘N-compass’. The detail and complexity of
the waveguide shape can be seen clearly in
Diagram 3.
Panel Show
Describing the second unusual element
of the PMC6-2, and in fact all the 6 Series
models, is best done by first describing
its rear panel. So, around the back of the
monitor there’s the traditional panel that
carries connection sockets, mains socket
and configuration controls. As far as signal
connection facilities go, the 6-2 offers both
balanced analogue and AES3 digital inputs,
and an AES3 output for daisy-chaining.
The 6-2 internal signal flow operates in the
digital domain at 96kHz.
Controls fitted to the 6-2’s rear panel
comprise a couple of rotary encoder
knobs along with an associated LCD
panel that displays a menu structure
comprising a variety of configuration and
EQ options. However, the second unusual
element of the 6-2 is that it can also be
configured over a TCP/IP network via an
Ethernet connection to a rear-panel RJ45
socket. Plugging each 602 of a pair into
a network switch and browsing to their
IP address opens PMC’s new Soundalign
web interface. And once you have access
to Soundalign, there’s really no need ever
again to look at the 6-2 rear panel LCD
display or touch its rotary encoders.
The Soundalign interface is great to look
at but also easy to use. The UI designers
at PMC have, I think, found a good balance
between intuitive accessibility and aesthetic
appeal. All of the functions available
from the 6-2’s rear-panel controls are
duplicated within Soundalign. There’s input
selection and level gain/trim, room position
compensation, desk filter, orientation
(landscape or portrait on the two-way
PMC6), subwoofer integration filtering,
polarity inversion, and low/high-shelf
filtering, followed by five discrete stages
of parametric EQ. Configuring a monitor
through Soundalign is simply a matter of
selecting the required one by clicking on
its Select button (this will usefully switch
the colour of the monitor’s front-panel
indicator to the same colour as displayed
in Soundalign), and adjusting the required
options. The first three Soundalign
screenshots show all the configuration
options available.
 In addition to providing a full suite of
configuration options, Soundalign also has
the ability to go far beyond a simple stereo
monitoring system. PMC have, over recent
years, been particularly busy equipping
Dolby Atmos studios, and Soundalign
very much reflects how significant
multichannel audio has become. The fourth
Soundalign screenshot shows how multiple
network-connected monitors can be added
to the system and selected. Monitors can
also be grouped and their configuration
parameters copied and pasted. The final
Soundalign screenshot shows the shelving
and parametric EQ stages. The settings
displayed reflect those developed for
the 6-2s working in my monitoring space
following its acoustic measurement using
Room EQ Wizard. Although it doesn’t
include any automatic monitor/room
optimisation functionality, Soundalign
reminds me of Genelec’s GLM software
approach in terms of its look, feel and
overall sense of professional competence.
It’s a huge advance on reaching around the
Soundalign offers
control over multiple
speakers (an Atmos
array, for example), as
well as a visual parametric
EQ interface.
back of a monitor and fiddling with some
switches, and its ease of use I think makes it
much more likely that a monitor installation
is likely to end up working optimally in any
given room.
In Use
Speaking of working optimally in any room
brings me to how the PMC6-2 worked
in mine. The answer is that they worked
fantastically well. The overwhelming
subjective character of the 6-2 for me is
a feeling of wide bandwidth accuracy and
dependability at any volume level, all the
way up to very loud. Any monitor the size
of a 6-2 that clearly aspires to a midfield
role has to be able to make that significant
jump towards main monitor volume levels
without losing its cool and starting to sound
rattled, or more subtly changing in character.
A genuine midfield monitor is not simply one
that’s a step bigger than a nearfield monitor.
It has to back up the scaled-up dimensions
with an equivalent scaling up of volume
level capability. The 6-2 pulls that off without
any question. I’m reminded of PMC’s roots,
as I described in the Golden Gear feature
on the KEF KM1 a couple of months ago
(https://sosm.ag/kef-km1). Having become so
familiar with the KM1 in his role at the BBC,
Pete Thomas’ motivation to found PMC was
built on a desire to design monitors able to
reach very high volume levels while at the
same time achieving genuine tonal accuracy,
and combine that with wide bandwidth,
low distortion and time-domain accuracy
too. The PMC6-2 to my ears achieves all of
that, but I don’t want to give the impression
that it’s a one-trick pony, only good at high
volumes. The 6-2 also does subtle and
informative at everyday monitoring volume
levels, with a finely focussed portrayal
of stereo, and detailed, insightful high
frequencies. In those terms it’s an extremely
capable mix tool, it just happens also to be
able to play genuinely loud.
While I’ve skewed the content of this
review towards describing how the 6-2
plays low frequencies, it was actually the
midrange performance that most often
caught my attention subjectively. Said
midrange is particularly notable for a great
sense of realistic, unhyped accuracy,
especially on naturally recorded human
voices, where it quickly reveals if something
isn’t quite right, or fools you into thinking
there’s another person in the room. Once
again the 6-2 demonstrates just how much
of a performance step skilfully designed
and engineered midrange drivers can bring
to monitoring.
I’ll finish with some final thoughts about
transmission-line bass as implemented in
the 6-2. To my ears there’s little doubt that
the way the 6-2 plays bass is right up there
with some of the best monitors I’ve had
the pleasure of hearing in my own listening
space. 6-2 bass is extended in terms of
bandwidth so well able to reveal those little
quirks of recording that bandwidth-limited
nearfield monitors often miss — the thud
of an accidental foot tap on a microphone
stand, or the low-frequency thump of some
acoustic guitar techniques. But the 6-2
combines extended bandwidth with great
dynamic, pitch and temporal accuracy. It
could be said to offer the bandwidth of
reflex loading with the accuracy and punch
of closed-box loading, and in doing so it
has something of a rare mythical beast
about it.
$ $8500 per pair.
T PMC USA +1 949 861 3350
E info@pmc-speakers.us
W www.pmc-speakers.com
w w w. s o u n d o n s o u n d . c o m / April 2022 93
 ON TE S T
Boss IR-200
Amp & Cab Emulation Pedal
The IR-200 is a compact, DSP-powered
pedal that pairs an amp simulator with
a speaker cabinet impulse-response engine.
Conceptually this is rather different from the
company’s GT-1000 or GT-1000 Core, both
of which integrate both amp and speaker
modelling with a multi-effects collection
and comprehensive routing. The target
market, then, is the sort of player who either
doesn’t use additional effects or prefers to
use discrete pedals rather than all-in-one
solutions. It’s also ideal for studio use,
though, a scenario in which plug-ins can
supply any additional effects you require. As
well as the speaker and cab, there’s some
ambience processing built in, with a choice
of three types, so the resulting sound never
The inbuilt effects loop is a nice touch that
makes integration into your pedalboard easy.
94 April 2022 / w w w. s o u n d o n s o u n d . c o m
seems too dry — even when you’re not
using additional effects. Power comes from
an included 9V DC adaptor.
On the technical side, the IR-200’s
credentials are suitably impressive,
with 32-bit A-D/D-A conversion, 32-bit
floating-point processing, and a 96kHz
sample rate. Handling the processing are
two custom DSPs. There are 128 mono or
64 stereo user IR slots, and these accept
(via a free IR Loader app) mono/stereo
WAV format impulse response files with
a maximum time of 500ms. They can be
44.1, 48 or 96 kHz, and 16- or 24-bit linear
or 32-bit floating point. While it’s great that
you can ‘roll your own’, there’s also a very
generous selection of speaker-cab-plus-mic
configurations to get you started. With the
144 Boss IRs and 10 from Celestion Digital,
I can’t imagine many cabinet types that
aren’t covered here.
In addition to a mono input jack and
a stereo output (on two separate jacks),
there’s a stereo mini-jack aux input and
a headphone output, which includes
processing to provide a more immersive
sound. So as well as having a low-latency
feed over headphones, you can also play
along to music from a mobile device.
A driver for Mac or Windows also allows
for direct USB recording or loopback to the
DAW for reamping with routing to the ins and
outs selectable via the master settings.
Your choice of cabinet emulation can be
used with one of eight guitar amps: Clean,
Tweed, Class A , Vintage and Modern
high-gain as well as three bass amp types.
Each amp has controls for gain, bass, mid
and treble, plus a noise suppressor, which is
very useful in cleaning up high-gain sounds
or for taming the hum from single-coil
pickups. The default speaker cabinets are
appropriate for the amplifier type but you
can switch to any you like.
If going large is your thing, you’ll
appreciate the pedal’s ability to run two
IRs at once, to produce either a mono
or a stereo output. I mentioned the 128
memories for storing user setups, and that’s
way more than I can imagine needing,
particularly since, in addition to the amp and
IR settings, there are two global EQs, with
a choice of parametric or graphic types to
help tailor the final sound.
To facilitate integration into
a pedalboard, there’s an effects
The aux input and
headphones allow you to play
along to an external source
— but this pedal can also be
used over USB with a computer.
loop with a mono send and a mono/stereo
return that comes after the amplifier section.
This can be switched to pre-cab emulation
or post-EQ. The Micro USB port is used for
firmware upgrades, communication with the
IR Loader and for the audio interface, which
shows up as a four-input, four-output device
for Mac/Windows, and basic stereo use
with iOS devices is also supported. External
control is available via a jack that can take
an expression pedal or two footswitches,
while MIDI In and Out is also provided on
TRS mini-jacks, which means adaptor cables
are required if you need to connect to
conventional five-pin MIDI devices. These
MIDI connections make for easy integration
with the Boss ES-8 and ES-5 loop-switchers.
My first impression was promising: it’s
all so easy. The lower row of knobs is set
up like a guitar amp panel, with Gain, Bass,
Middle, Treble and Level controls, while the
three knobs below the LCD window are
used to select Amp, Cabinet and Ambience
types. A Memory turn-and-press encoder
selects patches and serves as an Enter
button with two buttons below for accessing
Menu and Exit. Pressing both together
functions as Write. The factory settings give
a good idea of the range of amp/speaker
sounds that can be coaxed from the pedal,
and it’s very easy to swap out cabinets,
amp types and ambience types as well as
tweaking the gain and EQ.
Digging deeper, both the sound
and playing feel of the amp models is
convincingly realistic, with lively cleans at
one end of the spectrum, via some gentle
bluesy overdrive to full-on distortion and
modern, hard-edged rock at the other. The
ambience options really help to take away
that sense of ‘sterility’ that often afflicts
DI’d guitar sounds and, unlike with many
modelling solutions, those elusive ‘almost
clean’ sounds come over really strongly.
Switching cabinets makes a huge difference
to the sound character so, while there could
perhaps be a few more amp types on offer,
having only eight to choose from really isn’t
a limitation in practice. I tended to dial back
the highs on some of the factory presets
when using a Telecaster but, again, that is
very easily done either with the ‘amp knobs’
or the master EQ.
If you want an approachable,
good-sounding pedal that does both amp
and speaker emulation but you don’t want
the cost or complexity of additional effects,
the IR-200 really does have a lot going for it.
Paul White
$ $399.99.
W www.bossus.com
Denise Audio Poltergate
Gate & Transient Shaping Plug-in
Poltergate is a combination tool for use
with drums or other percussive sounds
and has both creative and problem-solving
capabilities. Supporting the usual-suspect
desktop plug-in formats, it combines
gating with bleed-reduction and transient
shaping, and it can be applied to recorded
drums, drum samples, drum loops or even
to non-percussive sounds, provided that
the notes have a recognisable attack.
Even the gate has features you might
not expect. In addition to a threshold
and comprehensive parametric
side-chain EQ and side-chain
monitoring, there’s a Spike
“The Flip button reverses the action
control that makes the gate more of the gate so you only hear what’s
sensitive to transients. Controls
for Attack, Hold and Release allow
the gated drum sound’s envelope
to be shaped in a fairly conventional way,
while Floor sets the maximum amount
of attenuation applied when the gate
is closed, and Return sets the gate-off
threshold, effectively adding hysteresis
by making the gate close level lower
than the gate open level. This is often
necessary to avoid unwanted ‘chattering’
or multiple triggering. Lookahead of
up to 10 milliseconds is available. All
the controls are available on four tabs:
Side-Chain, Curve (amount), Dynamics
and De-bleed, which are used to access
the various sections.
A waveform envelope display in
the upper part of the screen features
thresholds and shaded grey areas to
show the effect of the processing. The
side-chain EQ curve is also visible here,
in a more subdued colour. An external
side-chain is also available, opening up
a lot of creative possibilities, especially
when used with the Flip button, which
reverses the action of the gate so you
only hear what’s normally gated out.
To attenuate spill that is in a different
frequency range from the drum sound
being gated — for example, cymbals
bleeding onto a kick or tom track — the
De-bleeder section includes high- and
low-pass filtering, each with a choice
of de-bleed Mode (soft, medium or
heavy) and Release time. De-bleed is
applied after the gate threshold and time
constants have been set, and it works on
the same threshold as the gate. This has
normally gated out.”
the effect of applying EQ to the signal
during its decay to minimise the subjective
effect of spill. In effect, then, this applies
a form of frequency-selective transient
shaping, by modifying the spectral content
during the note’s decay.
For use in melodic recordings or loops,
the side-chain EQ can be used to target
the sounds you wish to keep, with Spike
again helping to target transients. The
release time of the gate can be sync’ed
to the host DAW tempo for the creation
of rhythmic effects, and Curve can be
adjusted to change the feel. Depending on
the type of material, it’s often possible to
use the gate in combination with surgical
side-chain EQ to target specific notes.
In the dynamics section, there are
controls for Punch, Fat and Clip along with
Attack and Duration. Punch enhances
the note transient while Fat affects the
sustain and is adjusted using both the
Attack and Duration controls to set the
ramp up and subsequent decay times of
the added sustain. Clip can be used to
add aggression to things like snare drums.
In the v1.1 update, a make-up gain control
was added to the dynamics sections too.
It’s important to understand what
Poltergate is and what it is not. It isn’t
really designed to separate sounds from
an already mixed drum kit — even if, with
careful side-chain filtering, you might
be able to split out the kick or even the
snare if the kit is balanced in a suitable
way. Rather, what it is really about is an
enhanced form of gating. The
parametric side-chain EQ and
Spike control allow for more
forensic operation than those
gates that have only high- and
low-cut side-chain filters, and
once the gate has done its
thing you can use the additional tools
to finesse the transient character of the
sound and help disguise any spill. How
effective Poltergate is on melodic sounds
depends very much on the character of
those sounds — you’re still relying on
the gate section to handle the initial note
recognition, though you can often come
up with interesting treatments when
working with loops, and the external
side-chain option and invert functions add
further flexibility.
To sum up then, Poltergate can be
thought of as a frequency-conscious gate
with additional superpowers, optimised to
work best on material with clearly defined
transients, notably though not limited to
drums and percussion. Paul White
$ Regular price €99 (about $112). Discounted
to €47.43 ($55) when going to press.
W https://www.denise.io
w w w. s o u n d o n s o u n d . c o m / April 2022 95
 ON TE S T
IK Multimedia T-RackS Fame
Studio Reverb
Reverb Plug-in
IK Multimedia’s Fame Studio Reverb
emulates the sound of the legendary
FAME studio in Muscle Shoals, Alabama.
It can be called up either as a single
plug-in or from within IK’s T-RackS 5 shell,
and it serves up eight different spaces
(two live rooms, five isolation booths plus
an echo chamber) as well as an EMT
plate reverb. To create this plug-in, IK
helped FAME’s Rodney Hall restore the
disused echo chamber back to its 1960s
condition, and kit it out with its original
equipment. They also worked with the
studio’s engineers to capture the mic
placement used on their most iconic
albums, along with the location of the
performer within the space. Thus, once
you select the type of source to process
(the options change according to the
space you’ve selected) the microphones
and sources are positioned automatically
at the appropriate positions within the
room or booth. If the plate reverb is
selected, the instrument type dial turns
into a decay time control.
As well as exploiting FAME’s extensive
vintage mic locker, IK put their classic
mixing consoles to good use. Studio A is
equipped with a Neve 8232 console,
while Studio B houses the restored SSL
6056 E which was used on Stevie Ray
Vaughan’s best-known albums, and both
virtual studios can instead make use of
FAME’s Universal Audio hand-built 610
console — the mixer that was used in the
first few decades of the studio’s 60-plus
years. The 610 is accessed in this plug-in
using the Vintage input option.
The GUI is suitably retro-looking, and
includes an inset 360-degree image of
the selected studio. You can rotate, so as
to view it from all angles, and on selecting
a different space the image changes
accordingly. On the left is the input
section, with large knobs for setting the
input and output levels. These are joined
by a Mono/Stereo input toggle switch
and the aforementioned Modern/Vintage
switch for choosing the console.
In the centre of the GUI,
a four-position rotary switch sets the
Source Position, with labels suggesting
the sort of application each setting might
best suit. Moving down this section,
there’s a pair of buttons to set the Spread
amount to Broad or Focused — the
96 April 2022 / w w w. s o u n d o n s o u n d . c o m
manual recommends using Focused
for single-source sounds and Broad for
wider sources such as drum kits. Below
these are Pre-Delay and Decay time
knobs for the reverb, and a four-band
EQ. The latter offers broad control, with
four gain knobs for a pair of high- and
low-pass filters (10Hz and 20kHz) plus
a pair of low- and high-shelf filters
(200Hz and 5kHz).
To the right, a mixer section offers
a level fader, pan pot and solo button for
each of: the two mono room mics, and
the Wet Left, Wet Right and Dry signals.
The mono mics (one ribbon and one
capacitor) are blanked out for the Plate
and Booth selections. A Link button
allows the left wet and dry channels
to be moved together. And at the top,
four different setups can be saved and
recalled using the A, B, C and D buttons.
The T-RackS environment in
which you can host this plug-in offers
a dedicated metering plug-in covering
Peak, RMS and Loudness metering with
lots of user customisation options, plus
a phase meter and a real-time spectrum
analyser. For that reason, other than
a variable-brightness, multicolour lamp
below the input control knob there’s no
dedicated metering built into the Fame
Studio Reverb plug-in.
Sonically, the EMT plate reverb sounds
warm and dense, which is just as it should
be. The echo chamber has a wonderfully
lively feel to it with a choice of dynamic
or capacitor microphones set up as a
spaced stereo pair. By contrast, both
booths impart a coloration and ambience
to the sound, endowing it with a sense
of dimension without adding any obvious
reverb tail.
The two Studios are also more of an
ambience effect than a long reverb, and
they differ in character considerably.
Studio B delivers a more weighty sound,
particularly at the low end, but in both
cases you can increase the sense of
roominess by dropping the Wet fader
levels and bringing up the room mic
faders. These two rooms are great for
guitar and drums, as they add a definite
sense of place and space to the sound
without washing it out in unwanted
reverb. In most cases I got the best
subjective results for electric guitar when
adding relatively little dry sound, while
for drums I found that mixing more of the
dry sound with the room mics and then
adding just a touch from the wet faders
did the job.
Depending on the sound you’re aiming
for, the booths can also be impressive in
those applications. The two booths add
their own unique coloration, with less
emphasis on the low end than either
room. Juggling the balance of the mics
with the dry sound is the key to getting
a good result from these, whatever
source you’re running into them.
So is it worth the price of admission?
Well, for a typical modern music
production, you can probably get by just
fine by using your favourite algorithmic
and convolution reverbs. But there’s
something endearingly vintage-sounding
about the Fame sound that just helps
recorded tracks sound ‘comfortable’. If
you hanker after the sound of the classic
records made in that studio — there are
far too many of those to list them here
— then the T-RackS Fame Studio Reverb
will certainly help you get those few steps
closer. Paul White
$ €149.99 (about $170).
W www.ikmultimedia.com
 Revelator™
Professional USB mic for live streaming
musicians, podcasters, and more.

Revelator™ Dynamic
Professional dynamic USB mic for
recording and streaming vocalists,
podcasters, and more.

Revelator™ io24 Revelator™ io44

USB Audio Interface with integrated
The ultra-compact recording
loopback mixer and effects for
and broadcast studio.
streaming, podcasting, and more.

Meet the Revelator family.

Every Revelator-series product features and live streaming. And it all happens onboard the Revelator,
powerful onboard processing that so your computer doesn’t have to do any extra work. Plus,
provides compression, EQ, reverb – even they’re all bus-powered, so no batteries or power supplies are
VoiceFX like delay, chorus, and more. required. All Revelator products include Studio One™ Artist and
You also get a built-in mixer that lets you combine audio from access to the Studio Magic software suite with over $1000 USD
multiple applications and create custom mixes for monitoring of music production tools. Visit presonus.com to learn more.

www.presonus.com

©2022 All Rights Reserved, PreSonus Audio Electronics. PreSonus is a registered trademark of PreSonus Audio Electronics, Inc.
Revelator is a trademark of PreSonus Audio Electronics, Studio One is a trademark of PreSonus Software Limited. All other trademarks are property of their respective owners.
 T E CH NI Q U E
These vocal tracking
tips will help you achieve
decent results every time.
M AT T H O U G H TO N
I
n one sense, recording a singer is
incredibly easy: put a mic in front them
and hit Record. But if you’re fairly new
to recording, and particularly if you record
at home, you might be wishing you could
do better. Your recording might sound
boxy or roomy, dull or boomy, brittle or
thin. You might cringe at sibilance, lip
smacks, bumps, thuds, distortion, hiss,
popping plosives... and where those birds
and dog barks came from, who knows?
I’ve heard every one of those things
compromise home-brew recordings — and
whatever problems you’ve encountered,
I promise you’ll get better results more
quickly if you head them off at or before
the recording session!
Rehearsal & Warm Up
You can’t have a good vocal recording
without a good vocal performance, and
while we’re not all blessed with the voice
of an angel there’s plenty you can do to
make the most of any voice. If you’re the
singer, practice how you want to deliver
the part before you think of recording it,
98 April 2022 / w w w. s o u n d o n s o u n d . c o m
and when the time comes to record try
some vocal warm-up exercises. You’ll find
plenty of exercises online, and I’d suggest
finding a handful to help you control your
breathing, articulate lyrics clearly, control
the resonance in your voice, and loosen
up your muscles more generally. If you’re
the engineer/producer, encourage singers
to adopt a similar approach, and allow
them time to warm up before cutting
takes. Don’t overdo the warm-ups, though,
or the voice will tire very quickly!
What you eat or drink can also affect
vocal delivery. Dairy products and diuretics
(basically anything that either leaves some
sort of coating/residue in your mouth or
dries you out, so things like coffee, tea,
cola, alcohol and sugary drinks) can be
unhelpful. Water will keep you hydrated
without unwanted side-effects, and I find
a few sips of lemon and ginger tea or
a similar infusion can reduce lip-smacking
and tongue noises.
Mood & Feedback
One of the most important jobs for an
engineer is to coax the best performance
from the artist, and making sure they’re
at ease is a good place to start. Consider
basic facilities such as heating/ventilation,
seating so they can relax between takes,
and having somewhere within easy reach
to put down their drink or place a lyric
sheet. Maybe also put some thought into
the general vibe of the recording space
too: keep it tidy, and figure out whether
changes in lighting might help or hinder
the singer — I like daylight, but I’ve worked
with singers who insist on singing barefoot
in the dark; I try to go with their flow!
Once recording, you need to appraise
each take. I like to have a lyric/song
structure sheet to hand, on which I can
write brief notes. (If you do this in the same
room as the singer, don’t let your pen/
pencil/paper noise ruin a recording!) You
then need to give constructive feedback,
and this can be tricky. First, if you become
engrossed in an engaging performance,
or you’re focused on technical issues, you
might forget to check they’ve hit all the
right notes, articulated every word clearly
and not fluffed any lyrics (those two things
go wrong on pretty much every session!).
Second, perceived criticism can really
dent a singer’s confidence. So don’t be
negative, but also remember that a singer
can infer a lot from your silence. Make
a point of saying something after each
take, even if it’s just “Well done, shall we
do another while you’re warmed up?”
and try to stay positive, or phrase what
you have to say as a question in order
to gauge the singer’s feelings — there’s
a certain amount of ‘reading the room’
involved here.
 Always think how much you’re asking
of a singer: voices, particularly untrained
ones, can tire very quickly, so while you
might need five or six takes don’t aim for
dozens, and remember that you don’t
have to record the whole song every time!
Learn to use your DAW marker navigation
and transport controls to make ‘punch-in’
overdubs easy. It can also be an idea to
make sure you get that difficult high note
or big ‘payload’ chorus nailed down fairly
early on, before the voice starts to tire.
The Mic
The classic mic choice for vocals is
a large-diaphragm capacitor mic. Most
have a cardioid or switchable polar
pattern; if the latter, best set it to cardioid
unless you have a specific reason not to.
Cardioid moving-coil dynamic mics like the
Shure SM58 or SM7 and the Electro-Voice
RE-20 are also a popular choice, and
ribbon mics have their merits too, though
most of those have a figure-8 pattern.
For more detail about the different
mic polar patterns check out our
March 2021 article (http://sosm.ag/
using-polar-patterns), but essentially
cardioid means the mic is most sensitive
at the front, so it picks up more voice
and less sound from other directions,
while figure-8 means it’s equally sensitive
to the front and rear but rejects sound
arriving at the sides. Speaking of ‘the
front’ I’ve sometimes seen inexperienced
vocalists sing into the wrong side of
the mic! For most moving-coil dynamics
you sing into the end of the mic, while
most large-diaphragm capacitors are
‘side-address’ mics, with the ‘front’ usually
being the side with the manufacturer’s
logo on it.
Generally, capacitor mics offer the
most open, airy, detailed sound, with good
high-frequency extension. They may be
deliberately ‘voiced’ to sound flattering
(for example, with a broad frequency
boost around 5-10 kHz) though they can
sometimes sound a bit brittle or ‘spitty’,
depending on the voice. Moving-coil mics
tend to sound a bit more ‘compressed’
and throw the focus more onto the
midrange, which can be just the ticket for
some voices and some styles. They’re also
less sensitive than capacitor mics and are
usually designed to be worked a bit closer.
Ribbons can sound lovely and smooth:
in its raw form the sound can be a bit
dark/dull due to a natural high-frequency
roll-off, but for reasons I won’t dwell on
here they tend to ‘take EQ’ very well
Drinks can have an impact
on vocal delivery: water is
generally good, but sugary or
caffeine-heavy drinks, alcohol
and dairy can be unhelpful.
so you can usually address that with
a high-shelf boost.
How High?
As important as the mic choice is where
you put it in relation to the singer. I prefer
mounting a mic on a conventional stand
than on a desktop one, since this makes
it easier to alter the position to suit the
singer, avoids unwanted reflections
from the desk and makes it less prone
to accidental bumps. Start with the mic
at a height that allows the singer to feel
relaxed when singing, but from there
you can adjust it to nudge their tone and
delivery in a certain direction — if the
singer’s comfortable with that!
For instance, if a singer has a tendency
to hunch over the mic, that can restrict
their throat and make them sound
a bit muffled. If (and only if) you hear
a problem, raising the mic height a touch
can encourage them to lift their head
a little, opening up their throat. Setting
a mic slightly higher up, ‘looking down’
at the vocalist’s mouth, can also reduce
the amount of the more directional
high frequencies, sibilance and wind
blasts from plosives reaching the mic.
As you bring the mic lower down you’ll
eventually hear more of the singer’s chest
resonance. There’s no right or wrong
here; listen for what sounds good/bad
and adjust accordingly.
I like to use a stand with a boom arm
rather than the height-adjustable-only
stage types. For one thing, singers
seem less inclined to grab the stand like
a stage prop (causing thuds and bumps),
but it also allows you to put the base of
the stand further from their feet, again
minimising accidental bumps. A boom arm
also makes it easier if you want to hang
the mic upside down, pointing towards
the vocalist — which I find makes cable
dressing neater and leaves a bit more
room to allow for dramatic arm gestures!
Up Close & Personal?
How far from the mic should the singer
be? The optimum distance depends on
several things, including the singer, the
room sound, the type of mic and the song
style/genre. In essence, the closer the
singer is to the mic, the drier the voice
will sound — more voice, less room —
and the more the recorded sound will be
focused on what’s actually coming from
the mouth rather than, say, the chest
cavity. Miking from further away can sound
more ‘open’ and natural, but will bake
more room sound into the recording. Most
contemporary vocal styles demand a fairly
‘dry’ recording, free of room sound. You
might well add reverb and delays later,
when mixing, but the dry recording gives
you the option of having a very up-front
vocal sound, and it also makes editing and
processing (particularly pitch correction)
easier to do later. What’s more, a drier
voice may be the only feasible option
if recording in a typical domestic space
(more on that below).
There’s more to it than dry versus
wet, though. Almost all directional mics
(cardioid, figure-8 and so on) exhibit
proximity effect, where low frequencies
are disproportionately boosted as the
singer gets closer to the mic. This can be
helpful if a mic is voiced too brightly or
w w w. s o u n d o n s o u n d . c o m / April 2022 99
 T E CH NI Q U E
RECORDING VOCALS

the singer has a thin voice, but not if you

want a drier sound without changing the
natural balance! Also, a directional mic’s
sensitivity changes as you move off axis,
so if a singer moves from side to side the
level will vary slightly; the narrower a mic’s
polar pattern and the closer the singer,
the greater those fluctuations will be. The
frequency response also changes as you
move off axis, especially at the top end.
Finally, singing very loud and close
up can occasionally produce levels that
overload a mic. You might be able to
counter this with a mic’s built-in pad or
high-pass filter if it has one, but backing
away from the mic or singing a touch
quieter is the easiest solution!
As a rough guide, I recommend starting
a small hand-span (15-16 cm, or 6-7
inches) away from the mic, then refining
the position if required. I generally find it
unhelpful to be closer than 10cm/4 inches,
or further back than about 30cm/12 inches.
A strategically placed pop filter can help
put the singer in the best spot, and for
extreme cases where a singer tends to
‘eat the mic’ a dummy mic can provide
a useful focal point — just raise the more
Hanging a mic upside down from a mic
distant ‘real’ mic and point it down so stand with a boom arm can leave more space
the dummy one doesn’t get in the way. for things like music stands or a singer who
If a singer moves forward/back during moves their arms, and can reduce the risk of
a performance, you’ll have a recording with a singer inadvertently bumping the stand.
an inconsistent room sound, so try to avoid
changing the distance dramatically during are facing something absorbent (usually possible, then, switch noisy appliances off
a performance; the kind of exaggerated a bass trap or acoustic panel) and that the (if it’s a fridge-freezer remember to switch
‘mic technique’ some singers use in live mic isn’t picking up any horrible noise or it back on again!). You can also exploit
performance isn’t always helpful here! reflections to the rear, and if I really want your mic’s polar pattern to reduce noise:
to dry things up I might have an absorber point a figure-8 mic’s side nulls towards
Controlling The Space suspended above the singer. If you have the recording computer and you might
You can also manage the dry/wet balance acoustic treatment, it’s usually fairly easy reduce any cooling-fan noise, for instance,
of a voice by manipulating the recording to arrange the panels differently for the and if you place an absorber behind it, it
space’s acoustics. First, it makes good mix and recording sessions, but I realise will pick up less of the room sound too.
sense to choose the best space you have not everyone has a dedicated studio Most sound that gets in from outside does
available, and which one that is might not space. A cheap but reasonably effective so through the air gaps around doors and
always be immediately obvious. Small alternative is to erect a sort of temporary windows, so you can start by closing them
box rooms (even with foam on the walls!) ‘booth’ out of duvets: drape them over and sealing any gaps up (temporarily with
or clothes closets may have similar inner mic stands, or anything else you have to tape is cheap, but removable secondary
dimensions to a professional vocal booth, hand — I’ve used a long-handled brush glazing or a custom acoustic panel to fit
but they’re not ideal: untreated, they’ll mounted on two shelving units before! snugly over a window while recording
sound boxy and, since there’s only space is a longer-term option). Potentially, you
for minimal treatment, the ‘treated’ space
Niggling Noises could also choose a time of day when
will still have natural reverb skewed If recording at home, chances are that there’s less ambient noise around!
towards the low end. This makes it hard the sound isolation isn’t great, meaning Note that you and the singer can also
to balance the captured sound without you potentially have unwanted sounds be sources of unwanted noises. Listen out
artificially adding back in a good chunk of from outside (traffic, birdsong, barking for things like squeaks, creaks and rustles
reverb back in higher up the spectrum. dogs...) and inside (heating, air con, from clothing, chairs, the turning of lyric
It’s better in my opinion to use a larger fridges, computer cooling fans) to contend sheets, watches ticking and so on. Some
space and put some effort into controlling with. Your ears do a great job of filtering singers will tap their feet, causing a thud
the acoustics. Rather than attempt to build out such sounds but they can be really that’s transmitted up the mic stand, and
a booth in my 6x5m home studio, I just annoying on a recording, especially once while I don’t always find mic shockmounts
make sure both the mic and my back you apply processing when mixing. Where essential for vocal recordings, they help

100 April 2022 / w w w. s o u n d o n s o u n d . c o m

here. For them to be effective, though,
ensure the mic cable isn’t pulled tight
between the mic and stand, and note
that thicker, more rigid mic cables
transmit more sound than thinner, more
flexible ones.

Gain, Pads & Filters

Whether you’re using a standalone mic
preamp or one built into your audio
interface/digital recorder, you need to
set the gain to bring the signal up high
enough above the noise floor without
clipping your A-D converters. This isn’t
hard to do if you record at 24-bit, and
there’s really no need to record super
hot. Do a quick soundcheck, and try to
get the singer to sing the loudest bits of
their performance at performance levels.
If you adjust the gain so that the average
signal is around -18dBFS or the peak level
around -10dBFS that should leave some
headroom in case they belt things out a bit
louder when recording.
If you hear distortion and the meters
on your interface or DAW input channels
aren’t hitting the red, you could be
overloading the microphone. It’s not
a common problem — far more often,
distortion is due to cranking the preamp
gain too far — but lots of mics have
a switchable pad and/or high-pass filter on
board and both can potentially help. The
pad lowers the sensitivity of the mic, so
if you engage it you may want to change
the gain on the preamp accordingly.
A high-pass filter can sometimes achieve
the same thing by removing energetic
but almost inaudible ‘rumbles’ that push
up the overall level. The same facility
on a preamp or interface might help to
remove low frequencies and stop the
preamp going into overdrive, but it won’t
prevent a mic from overloading.
The Cue Mix
The singer will usually need a backing
track to sing along to, mixed with
a ‘foldback’ of their voice. You’ll get
the cleanest recording if they use
headphones, and preferably closed-back
ones so sound from them doesn’t spill
onto the mic. Some singers like to slide
a cup off one ear so they can hear their
natural voice in the room, in which case
make sure that cup is directing sound in
towards their skull, not blaring sound out
into the room. If you’d rather leave the
singer alone, you can use your DAW or
interface routing software to put the mix
Closets and box rooms are not like professional vocal booths, so if recording at home it’s usually better
to choose a bigger room and improvise some acoustic treatment. A polyester duvet hung behind the singer
can do wonders. Shockmounts will reduce the vibrations from tapping feet reaching the mic, but make sure
the mic cable is not pulled too tight. And as well as protecting the mic from plosive blasts, a pop filter is also
a great tool for showing a singer how far you want them to be from the mic!
only in one ear. Some hate singing while audio interfaces allow you to mix the DAW
wearing headphones, though, and while playback with the signal coming from
it’s possible to work without them you the mic, and this side-steps the problem
might need to tread more carefully; see our of latency. That said, you can use the
October 2015 feature for more detail on DAW to add a ‘comfort reverb’, as any
that approach (http://sosm.ag/tracking-voc latency is just perceived as ‘pre-delay’.
als-without-headphones). (Some singers seem to find reverb helpful
Most DAW software allows you to create in judging pitch when monitoring on
a dedicated ‘cue mix’, with a different headphones, while others don’t like it.)
balance from the one at your master stereo
bus. You can use this facility to refine
Playback Time
the balance to the singer’s satisfaction. Finally, at some point you’ll probably need
A single fader each for drums, bass, to play recordings back to the singer and
backing/guide vocals and everything else discuss which bits of which takes they like.
should give you plenty of control, and as For this sort of playback, it can be good
well as delivering what the singer wants to have a basic processing chain ready
you can manipulate this mix to nudge their to roll even though you don’t use it in the
performance into place: you might bring cue mix. There’s no need to overcook
up just the kick and snare, or use EQ to things here, but some gentle compression,
emphasise the kick’s click, for example maybe some gentle shelving EQ to
if you think they need to feel the rhythm compensate for any proximity effect bass
a bit more. boost or a ribbon mic’s natural HF roll-off,
I don’t recommend using your DAW to and just a touch of reverb or delay can be
apply processing to the incoming voice. enough to make it all sound that bit more
For one thing, it’s often unhelpful if the ‘polished’ and keep those confidence
sound in the headphones is very different levels up. Again, what works best there
from that in the room. For another, most depends on the singer.  

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

 F E ATU R E
STEFAN W YETH
T
he amount of audio processing
power on tap in the latest
laptop and desktop computers
is astonishing these days, and this has
certainly expanded the capabilities of
the software we use. We’ve all seen
masterclasses where our favourite
engineers break down hit mixes that
have 10 or 20 plug-ins flexing away on
each channel. However, not everyone
has access to the same resources, and
many of us can’t justify the expense
of a cutting-edge machine for working
on projects at home. It’s always useful
to know how to get the most out of
a modest system, and even if you’re
lucky enough to have the latest ARM
processor or an external DSP interface,
there are still benefits to working in
a way that conserves resources. These
are approaches to using a DAW that are
rooted in old-school analogue signal-path
sensibility, and their advantages
are universal.
Monitoring During Tracking
Minimising latency is usually a priority
when recording audio, because you often
want to monitor the input signal through
the recording path. Too much latency will
mean that the artist hears a noticeably
102 April 2022 / w w w. s o u n d o n s o u n d . c o m
Get More From Your DAW
Learning to manage CPU resources effectively will make
you a better mixer.
delayed version of their performance,
which is distracting and makes it hard to
play or sing well. Unfortunately, the actions
you can take to bring down latency, such
as reducing audio interface buffer size or
using features like Logic Pro’s low-latency
mode, usually place significant demands
on your computer. This means that, without
resource-friendly management of your
project, you can easily overload your CPU
at this point, especially if you’re trying to
monitor with effects such as compression
and reverb during tracking.
Bear in mind that you are not usually
making a permanent commitment when
choosing these plug-ins. They can always
be changed later, so it is worth exploring
your plug-in folder to find the ones that
don’t impose either a heavy CPU load
or additional latency of their own. As
a general rule of thumb, the plug-ins
supplied with your DAW are likely to be
pretty economical in this respect. Note
also that if the tracks you’re recording
to are routed through group, master or
monitoring channels that have their own
plug-ins, any processing delay introduced
by those plug-ins themselves will be
added to the latency. It may be worth
temporarily disabling them, or routing
the tracks you’re recording directly to the
output during tracking.
Bouncing Benefits
Virtual instruments and processing
plug-ins can consume significant CPU and
memory resources, as well as sometimes
adding to the latency of the system. The
load of each one may not be significant
in isolation, but as your project grows,
the cumulative burden can compromise
performance during playback and
recording, until eventually your system
becomes sluggish or refuses to play back
altogether. The solution is to bounce or
render some of these instruments and
plug-ins, generating new audio files
which have the plug-ins’ output printed or
‘baked in’. The instruments and plug-ins
in question can then be disabled to free
up resources.
This will give you more power to
dedicate to the task at hand, but it can
benefit other areas of your workflow,
too. When you’re working
on a collaborative project or
sending files to an engineer
when it’s time to mix down,
bouncing your tracks ensures
compatibility across DAW
systems and between plug-in
libraries. This process also
commits you to the sounds and
channel strip settings you’ve
created, and provided you aren’t
bouncing anything with plug-ins
requiring automation at a later
stage, represents a firm step
towards the completion of your
project.
Rather than using your DAW’s
batch export feature to output
all the tracks as rendered audio
files, it’s better to get into the
habit of bouncing tracks as you
move through each section of
your project. As you work,
the bounced audio
tracks become
way markers
showing you

External DSP Hardware
If your computer just isn’t up to the job,
another option is to offload some of the work
to an external DSP system, such as a UAD
card or another networked computer running
Vienna Ensemble Pro or AudioGridder. This
might seem like a logical way to save CPU
resources, particularly if you have another
computer gathering dust, but there are some
factors to consider before taking this route.
Dedicated external DSP systems have been
around for decades, and they remain at the
pricier end of the market. And even with the
best software, using multiple computers can
seem a little clunky and mean more cooling
fans whirring in your studio. So if saving CPU
resources is your only goal and you have funds
available, it’s worth asking whether a faster
which sections have been processed.
Adding bouncing to your workflow during
the arrangement phase encourages you
to think proactively about the progress
of your project. Facing up to decisions
like which plug-ins to leave active and
which to bypass when bouncing tracks to
audio can be a positive step in becoming
a better engineer and a more effective
DAW user. You’ll also need to decide
which channels to process in mono or
stereo depending on instruments you’re
working with and their orientation in the
mix, bearing in mind that some might
require panning at a later stage. Thinking
about what flexibility you’ll need at a later
stage in the mix can help you make
decisions that move the project forward.
computer might give you more bang for the
same buck — if your computer is over about
four years old you might be surprised by how
many more plug-ins a latest-generation one
can run.
Having said that, there are certain features
of DSP systems that no native system can
match, perhaps the most significant in this
context being guaranteed performance levels.
Each plug-in in a DSP system typically has
a fixed system overhead, meaning that there
is no ‘grey area’ or uncertainty as to whether
the system can run a given number of plug-ins.
DSP systems also offer very effective copy
protection, which contributes to the fact that
there are many high-quality DSP plug-ins that
are not available in native form.
of reverb on a per-channel basis by
changing the send level. Where effects
busses are being used mainly for
a single group of instruments — such
as, for example, a reverb that is being
applied to multiple drum tracks — you
can route both the source tracks and the
effects return to a single group channel
for processing with compression and
EQ. This can help make the effects feel
like they are part of the sound of that
drum kit.
The point where your CPU can be
tested the most is often towards the
end of a project, when mixing down
the multitrack or finalising tracks to
send elsewhere. With arrangement and
tracking complete, the project’s channel
count will be high, and the final mix
process will require even more system
resources for tasks such as automation.
Summing your channel outputs in groups
will make the job easier, both for you
and your CPU. Individual tracks should
be bounced to audio by now, with any
additional effects applied using sends.
Working with groups such as drums,
vocals and guitars, and subgroups within
these such as backing vocals or rhythm
guitars, can help you achieve a holistic
and cohesive mix faster. Instead of
working with individual channel levels on
80-120 faders, you can focus on moving
seven or eight group faders to achieve
the right overall mix balance.
Asset Management
Each audio file bounced within a project
demands storage on your drive, and

Start Making Sends

When mixing in the analogue domain,
channel counts are always limited, and
you can’t simply conjure up additional
effects units wherever they’re needed.
Getting the most from the desk and
outboard you have involves imaginatively
sharing resources. For this reason,
reverb, delay and even modulation
effects are typically addressed from
auxiliary sends rather than channel
insert points; this allows a single reverb
or other effect to be applied to multiple
sources simultaneously.
The same approach is equally valid
in DAW-world, so make a point of using
effects plug-ins on a send/return basis
where possible, instead of loading up
your channels with inline effects. This
reduces CPU load, and having the same
reverb applied to many instruments
can help your mix feel more coherent
— it’s still possible to vary the amount

w w w. s o u n d o n s o u n d . c o m / April 2022 103

 F E ATU R E
MANAGING CPU RESOURCES

The Commit
function in Pro Tools
is a neat alternative
to conventional freeze
and bounce-in-place
options. Here, three
virtual instrument tracks
have been rendered to
stereo audio files; the
original tracks are still
present with all their
MIDI data intact, but they
are deactivated.

file counts will often increase as you workflow is an ongoing project rather no point in saving a few CPU cycles if
work, until you reach the final phases than a quick fix, so be patient, and it means doing something that hinders
where the stems are the entire length don’t be dogmatic about it — there’s your creativity.
of the song. Giving your files detailed
names can avoid any future confusion,
both for yourself and anyone you work
with. Before entering each phase
(arrangement, tracking, mix) of your
project, save alternate versions of your
project and consolidate its assets to
ensure that no unused files are taking
up unnecessary space. Keeping these
alternate project versions backed up
on the cloud and a separate drive will
avoid duplicates being found when
opening the project. This way, you only
have the latest, optimised version on
local storage, giving you the fastest load
times and maintaining the availability of
your system’s resources.

Summing Up
Every DAW has its own idiosyncrasies
regarding system resources and plug-in
management, and the ‘pain points’
won’t always come in the same place
for every project. For example, music
styles like rock and folk might use
predominantly live recorded audio,
whereas electronic music can be
entirely comprised of soft synths and
sample-based sound sources. The
greater the understanding you have
of the tools you use and the job at
Here, a conventional multitrack drum recording has been organised according to time-honoured principles.
hand, the easier it is to find hacks and Two kick and two snare mics are each combined on separate mono groups, which are then routed along with all
loopholes that will help you out in a given the other drum tracks to a single stereo group. The same reverb is being applied to multiple drum tracks using
situation. Implementing a CPU-efficient auxiliary sends, and this too gets returned to the main drum group for processing along with the dry tracks.

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

 T E CH NI Q U E
We offer some tips for processing strummed guitars.
JOHN WALDEN
I
work in a lot of different musical
genres and something that appears
in almost all of the mainstream ones
is strummed acoustic guitar. Really, to
achieve a good strummed acoustic
sound, the main battle is won or lost at
the recording stage: right guitar, right
performance, right room, right mic and
right mic technique. If you need some
advice on that front, I’d suggest checking
out Mike Senior’s April 2010 article (www.
soundonsound.com/techniques/how-rec
ord-great-acoustic-guitar-sound) and an
earlier one by Paul White (August 2001:
www.soundonsound.com/techniques/
recording-acoustic-guitar-0).
But if you have a good recording or
even choose to program your parts using
a virtual guitar instrument, you still need to
get that part working well in your mix and
to that end, as Mike outlined in his August
2020 SOS Podcast (www.soundonsound.
com/techniques/troubleshooting-a
coustic-guitar-recordings-podcast),
some processing of such guitar parts is
almost always required at the mix stage.
Thankfully, Cubase has a lot of tools to
help you on that front, and in this article
I’ll explore some strategies that Cubase
Elements, Artist or Pro users can try.
Trim The Fat
For almost any strummed acoustic guitar
recording, a good first step is to try
106 April 2022 / w w w. s o u n d o n s o u n d . c o m
applying a high-pass (low-cut) filter. This
can easily be done in the Pre section of
Cubase’s Channel Settings panel or its
MixConsole, using the Low Cut control. The
slope of the filter means it reaches gently
up from the specified turnover frequency,
and if you set it around 30-40 Hz it should
take care of any unwanted bumps and
rumbles that can eat up headroom and
trigger dynamics processors such as
compressors, without impacting negatively
on the guitar sound itself. You might or
might not want to roll this up higher as you
get further into the mix, to carve out space
for other instruments (that all depends on
the mix in question) but this is generally
a decent place to start — even if you can’t
actually hear any difference!
A further low/low-mid EQ move might
help too, and be rather easier to hear. I’ll
often try ‘cleaning out’ some frequencies
in the 200-500 Hz range. Don’t get me
wrong, this sort of decision is all about
context, of course. But lots of instruments
have energy within this range, and if it’s
left unchecked on all sources you’ll often
hear a build-up of ‘mud’. Since strummed
acoustic guitars are rarely contributing
much of real musical importance in this
region, I usually find that cleaning them up
in this region allows more prominent parts
the space they need.
For such cuts, Artist/Elements users
can turn to the Channel Settings EQ
section — I suggest experimenting with
a static cut of a few dB somewhere in this
Cubase
Cubase’s Channel Settings window provides
all the options needed for you to apply basic tonal
sweetening EQ to your strummed acoustic guitar
parts. As described in the main text, this might
include cutting some of the low end, adding a little
high-end ‘zing’, and/or managing any low-mid mud
or other problematic resonances.
frequency range, with a Q value around
1-2 being a good starting point in terms of
bandwidth. You’ll often audition the result
in solo but make sure you also listen to it
in the context of the mix — depending on
the style of the production, you might be
surprised by just how far you can go with
cuts in this area before you perceive the
guitar as sounding ‘thin’. If you do choose
to be fairly aggressive, though, make
sure you listen to it in the parts of the mix
where the guitar part is most exposed.
Cubase Pro users have a more
sophisticated option for this ‘mud
removal’ job: a dynamic EQ band can be
applied using the Frequency 2 plug-in.
Actually, Frequency presents you with
a number of potentially useful options,
but the most simple is often enough: you
configure Frequency 2 to apply a cut only
when the guitar displays particularly high
energy in the target frequency band. As
shown in the Frequency 2 screenshot,
you can take this further by adjusting the
Start control. This way, you can apply
both a static (always on) cut, in this case
of 1.7dB, with some additional dynamic cut
when there’s more energy in this band.
The combination of the Threshold, Ratio
and Gain controls determine the extent of
the latter, and as you’re only cutting when
it’s really needed you can, in theory at
least, be left with a more natural tonal
balance most of the time.
Ching, Ching
Having brought the lower frequencies
under control, you can turn your
attention to the top end. Again, you
can’t EQ by numbers since what’s
required here always depends on the
tone of the original recording and the
instrument’s role within the mix. But
if you want to enhance (or tame) the
‘zing’, a good starting point is to adjust
the 10-15 kHz frequency range, and
perhaps even higher.
Elements/Artist users can configure
band 4 of the Channel Settings EQ
as a high shelf; I tend to use the type
IV shape, but pick what sounds best
to you. Set it to start somewhere
around 10-15 kHz and adjust the gain
to boost/cut to
taste. Again, Pro
users might do “Frequency 2 can be
something similar configured to apply a cut
with Frequency
2, and investigate only when the guitar
the additional
potential of
displays particularly
dynamic boosts high energy within the
or cuts. As shown
in the screenshot, target frequency band.”
and like with our
low-mid cut, the Start control lets you
apply a fixed boost/cut if required,
and any dynamic element can be
‘superimposed’ on this static setting.
Balancing Act
While the low end and high end are the
obvious places to start, there may well
be other frequency ranges that could
benefit from some TLC. Overprominent
resonances can be caused by various
things, such as the room within which
the recording was made, the specific
instrument being played, or the skill/
consistency of the player. Whatever the
cause, though, such resonances might
mean some notes or note ranges tend
to sound louder/quieter than others.
That’s rarely desirable in a strummed
chord, so how should you tackle it?
Compression can undoubtedly be
part of the solution, but it can also be
worth applying a little EQ first to target
any specific frequency ranges that
appear to be problematic. Pro and
Artist users can put the SuperVision
metering plug-in to good use here
— inspect the frequency curve for
obvious humps and hollows. Elements
users can use the frequency spectrum
display in the Channel Settings
window to similar effect.
Whether by ear or by eye, if you
find a very specific frequency range
that needs a little attention – perhaps
associated with a single resonating
pitch — a narrow-band cut/boost can
be applied, whether via the Channel
Settings EQ (Elements/Artist) or
dynamically via Frequency 2 (Pro; just
cutting/boosting when the resonance
is most problematic). However, care
is needed if you end up with two or
more cuts at similar frequencies as
the possible overlap between the
filters can result in some unpleasant
artefacts. Frequency 2’s Linear Phase
mode can help in this regard, although
the processing involved does add
additional latency
and can’t be
combined with
dynamic EQ in
the specific band.
You should also
listen out for the
effect on notes in
other parts of the
performance —
EQ a high E note
and it will likely
impact on the harmonics of a low E, for
example (if it does, then dynamic EQ
can be a useful tactic).
Pick Control
Depending on the recording and
performance, pick noise may be
apparent. If it’s too loud it can be
a distraction but the ‘clicky’ (rapid
transient) element of the pick noise
(generally above 5kHz) can also
contribute usefully, enhancing the
percussive nature of the guitar part.
While a single band of a multiband
compressor (focused on the required
frequency range and with fast attack/
release settings) can often be put to
good use here, Elements and Artist
users don’t have that option in their
stock plug-in collection. Thankfully,
for toning down a too-clicky pick,
the De-esser plug-in provides an
alternative. In the screenshot example,
I’ve focused on the 5-10 kHz range and
set a fast release (you can’t change the
attack time). In dialling in the required
frequency band, the Solo and Diff
w w w. s o u n d o n s o u n d . c o m / April 2022 107
 T E CH NI Q U E
ACOUS TI C GUITA R S Cubase
Frequency 2 offers Cubase Pro users
the option of using dynamic EQ. This allows
you to control when cuts or boosts are
applied, using compressor-like controls for
each EQ band.

buttons are particularly useful as

they let you hear exactly what
the De-Esser is taking away from
the signal.
Pro users could use the
Multiband Compressor but, having
come this far with Frequency 2, I’d
happily suggest using a dynamic
band focused on the required
frequency range to either boost or
reduce the pick attack as needed.
Here, you can control both attack
and release times on the dynamic
EQ, and the Listen button (when
Side-Chain is set to Internal)
allows you to solo the band to
hear exactly what elements of the
sound you are processing.

On The Spot
Noise generated by finger
movements across the fretboard
often find their way into acoustic
guitar recordings. In some cases, these
‘spot’ noises add a sense of realism to
a performance (indeed, there are pop
tracks where said squeaks often seemed
to be emphasised for effect) but, usually,
they’re an unwanted intrusion.
Conventional EQ can address them to
some extent but there are usually ways
to achieve a better result. Elements users
can try detailed level automation to duck
the level of the performance by a few dB,
to mask the offending noises, whereas
Pro/Artist users can turn to a dynamic EQ
band in Frequency 2 or attempt some
spectral editing using SpectraLayers One.
SL One doesn’t offer all the selection
options of SL Pro, but finding finger
noise, selecting the required frequency/
time range, and then adjusting the gain
of that selection, is fairly easy. If your
performance has a lot of finger noise it’s
somewhat time consuming but, short
of re-recording, this is likely to be the
method that enables you to achieve the
most satisfying results.
other instruments shine. Either way,
whether you’re using Elements, Artist
or Pro, Cubase 11 has all the tools you
need to get good results. And once
your EQ-based tonal balancing is done,
you can move on to some additional
dynamics control... but that’s a discussion
for another day.  

For Elements or Artist users, in the absence of
a multiband compressor, Cubase’s DeEsser plug-in
can be used to tame some frequency-specific
issues such as pick click.
Splendid Isolation?
I’ve provided some audio examples
(available on the SOS website:
https://sosm.ag/cubase-0422) to
accompany the suggestions made here.
In the majority of the audio examples,
I’ve provided just the isolated guitar part
to make auditioning easy. With any EQ
moves made when mixing, though, while
you might use the solo function, you
should always make sure you audition
them in the context of the other parts
in the mix too. It’s an iterative process,
and in a busy arrangement you may
well find that moves that sound quite
radical when soloed can seem more While EQ or clip-based volume automation
can be used to subdue spot noises such as finger
subtle in the mix. Provided you don’t
squeaks, spectral editing via SpectraLayers (the
rob the guitar of its overall impact, the One or Pro versions) is a very effective alternative.
more of it you can surgically chip away, It allows you to adjust the gain over very tightly
the more sonic space you have to let specified time/frequency ranges.

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

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 T E CH NI Q U E
Logic
STEPHEN BENNETT We explore some of Logic’s lesser-known features.

L
ogic Pro was designed
to be a flexible and controller and System Exclusive (SysEx) features that
open-ended piece of make them eminently accessible via MIDI (you’ll find
software. But with this flexibility this information in the hardware’s manual).
comes complexity. It doesn’t You can easily create ‘virtual’ knobs and faders in
help that many of its features Logic to control these devices, and again you’ll need
are not obvious in use and, to open the MIDI Environment to do this. This time
while the basic functionality navigate to the Create layer menu item and rename
as a DAW is very easy to get the new Layer accordingly. From the New menu,
your head around, getting to create a Multi-instrument and assign it to the MIDI
grips with (and finding) all the Ticking the Use External Instrument Plug-in box port and channel your hardware is on. You’ll need
features you may find useful is allows you to define an audio input alongside a MIDI to know which controller number/SysEx commands
very difficult — even reading port when using external synths in Logic. adjust which parameter on your hardware, but Logic
the whole manual might not Pro’s on-screen controls can send both types of data.
help! In this workshop, we’ll that the fader turns yellow. Now the pan Create your faders and knobs from the New menu and
go through a few of these control affects just the send and the yellow assign them the required output controller number (or
more obscure features, many fader sets the send level. You can assign SysEx string) in the Inspector. Finally, patch them to
of which I have stumbled this to a Key Command to make swapping the Multi-instrument so that when adjusted, their data
across when trying to achieve between these modes easier. passes through to your connected hardware.
something that’s musically out Most of these old digital synths and effects have
of my comfort zone.
External Instruments internal patch memories, so wouldn’t it be useful if
Logic Pro’s MIDI features were second to you could change these patches directly within Logic
Pan Sends none when the software was developed Pro? And wouldn’t it be even better if you could select
When you send a signal from and while most of us have moved from them by name? The good news is you can! To do
a stereo track to a stereo bus, using dozens of external MIDI devices this create an External Instrument track (with the Use
the signal panning matches to plug-ins, there is a resurgence of External Instrument Plug-in button unchecked) and
the channel’s settings. interest in hardware instruments. Logic select the MIDI port and channel that your instrument
Sometimes this is not what you Pro’s External Instrument Track can send is connected to. Then give the track a useful name in
want — I’m very fond of having MIDI information to external MIDI devices the Inspector (if one isn’t automatically created from
different effects in different such as synths or hardware effects. Here your Audio MIDI setup program). Next open the MIDI
places in the stereo image. you can choose the MIDI channel and Environment from the Window menu and Navigate to
I’d been setting up complex port required to communicate with said the MIDI Instruments pane via the pull-down menu at
routing to achieve this before instrument. However, to record the audio the top of the Inspector. You’ll see a Multi-instrument
I had a Eureka (or D’oh!) from that Instrument you’ll also need to object has been created. Now we’ll need to find
moment and discovered that set up the audio inputs that the hardware a text-based list of the patches that are on your
Logic Pro allows you to do just is patched in to. For years I was creating MIDI hardware. If the patch names are in ROM or
this right from a send itself. If extra audio tracks for these audio inputs, still at default, you might be able to find a list of
you choose Independent Pan until I noticed that a little box named Use them on the Web, but if you’ve edited your patches,
from the Send menu, and then External Instrument Plug-in had appeared. you’ll need a piece of software such as MidiQuest
Sends on Faders, you’ll see Checking this opens an extra audio input
section where you can define both the
MIDI connection and audio input for
recording your hardware synthesizer.

The Independent Pan option
allows you to pan a source and its aux
sends independently of one another.
Virtual Controllers
While prices for old analogue gear are
soaring, there are real bargains to be
found in the keyboards and modules
created during the great knob and slider
drought of the 1990s. Some of these
instruments are almost as sophisticated
as newer plug-ins, but with the added
character of their digital-to-analogue
converters — EMU’s Proteus series, I’m
looking at you. However, trying to adjust
a filter in something like a Proteus really
is akin to painting a hallway through Logic’s MIDI Environment lets you create ‘virtual’ controllers
a letterbox. Helpfully though, many that can communicate with even the most arcane ’90s
of these devices have extensive MIDI rackmount devices.

110 April 2022 / w w w. s o u n d o n s o u n d . c o m

(https://squest.com) to extract them. As
a bonus, MidiQuest can also export them
as text files in the correct ‘one patch per
line’ format that you’ll need for the next bit.
Double-click on the MIDI object you that
It’s also possible to extract patch names from
old hardware, which you can then select from the
Program section of the Inspector.
you created and you’ll see a bank of patch
names displayed alongside a pull-down
menu where you can select different patch
banks (MIDI patches are often in banks
of 128). If you use the Options pull-down
menu, you can paste in the patch names
from the text file. Now, when you select
this track in the Arrange page, you can
change banks and patches using the
Programme section of the Inspector — just
remember to make sure the Program box
is checked! You can also manually type
in your own names into Multi-instrument
to make it easier to select patches with
hardware that has numbers rather than
names for its internal programs.
Quick Sampling
While you can import audio files into the
main window to
create audio tracks,
Dragging audio files
into the track list will give
you a variety of options
for what to do with it,
rather than simply placing
it into the timeline.
dragging them from the Finder
(or Browser) to the track list
will present you with several
options as to how you want the
audio to be used. This feature
allows you to quickly create
simple or complex Sampler
instruments, or add audio to
Drum Designer or Alchemy (in
its various forms).
Reuse & Recycle
It may be that you have
a plug-in chain or audio or
MIDI regions in another Project
that you’d like to reuse in
a new one. To do this, open
the Browser and select the
little camera icon at the top
right and choose the All
Files tab. From here you can
navigate to the folder where
Logic’s Plug-in Manager lets you create sub-folders, into which
you can organise your plug-ins and virtual instruments.
the Project is that you wish to import material from.
Select this Project (and any Project alternatives) and
you’ll see a display of all the content in that Project.
You can decide which elements to import — for
example, if you just want a Plug-in chain, select just
these. Once you have chosen these elements, clicking
on Add will place them in your current Project.
Plug-in Manager Sub-folders
Logic Pro’s Plug-in Manager is brilliant for organising
your virtual effects and instruments. It’s easy to create
folders to organise your most-used Plug-ins, but you
can also add sub-folders to these by using a colon
between the names of the folder and sub-folder. If you
create folders (use the + at the top of the list) called
Synth:Analog and Synth:Digital, for example, then drag
the required instruments into these folders, they’ll
appear as sub-folders in the plug-in track menus.
Fine-tune Your Automation
The Event list is a great place to finely adjust Note
positions, velocity and controller information, but you
can also tune your Automation here. Use the Key
Command to open the bespoke Track Automation
Event List. Here you can precisely adjust all the
relevant parameters relating to Automation events.
I find this very useful when dropping levels to limit the
effect of plosives on vocals, for example.
Take Notes
Adding text notes to the bottom of a channel strip is
a useful way to keep track of your, um, tracks; it’s a sort
of digital version of the masking tape we used to stick
on the bottom a mixing console. These notes are also
available on the track itself — but you must explicitly
make them visible by right-clicking on a track and
selecting Notes from the Channel strip components
menu in the same fashion as you do in the Mixer.
Speaking of notes (the other kind!), if you’re
producing ambient music and your MIDI data consists
of long notes, it can be really irritating if you start
playback when the Playhead is in the middle of that
note as, by default, no sound is heard. However, if
you check the Notes box in the Chase section of the
Project’s MIDI Project settings, the notes will play back
perfectly wherever you start playing the Project from.
This is a pretty brief dip into the murkier waters of
Logic Pro and I hope you found something in there
that you hadn’t known about, or tried, before. Logic
continues to surprise me with its depth of features and
by the time this is published I’ll probably have found
another dozen obscure (to me) way to do things in
Logic Pro!  
w w w. s o u n d o n s o u n d . c o m / April 2022 111
 T E CH NI Q U E
We explore some handy
tools for making the
most of macros.
JULIAN RODGERS
A
rguments about which DAW is best
have always raged online, and aren’t
likely to subside any time soon.
For the army of working professionals who
are quietly getting on with their work, one
of the biggest strengths of Pro Tools is its
speed. There are some areas where other
DAWs are faster — I can create a tempo
map faster in Studio One or Logic than
I can in Pro Tools, for example — but for the
things I do over and over again in a typical
project, Pro Tools wins for me. Part of this
is the UI, which is mature and streamlined,
and part of it is the comprehensive set of
keyboard shortcuts, which crucially aren’t
customisable. While lots of people dislike
this, a key part of the ubiquity of Pro Tools
is the consistency of the interface and
the shortcuts.
It only takes an intense session of vocal
editing or drum tweaking for the value of
a five-second gain in an operation you
perform hundreds of times a day to become
painfully apparent. Small incremental gains
can add up to very significant time savings.
But there are other options for speeding
things up besides the keyboard shortcuts.
Take EuCon, for example. An S6 is an
expensive item but they still sell. There is
a reason for this...
Avid Control
Anyone who wants to extend their control
over Pro Tools has a very capable free
option available to them in the form of the
Avid Control app. Paired with an Avid Dock
and as many S1 control surfaces as you can
afford, this makes an amazing control option
for Pro Tools, but even on its own the app
is useful. The Tracks page is quite literally
a gift, the app is free, and for speedy track
operations it’s very easy to get used to, but
it is the Soft Keys which I’ll focus on here.
The Soft Keys section of the Avid Control
app contains 150 pages of Soft Keys which
offer single-button control of any of the
hundreds of ‘EuConised’ parameters in Pro
Tools. This is the term for controls which
have been added to the very extensive set
of parameters controllable by EuCon.
There is an understandable consistency
of approach taken by Avid here. In the same
way as the UI and keystrokes are consistent
across all Pro Tools systems, so are the
112 April 2022 / w w w. s o u n d o n s o u n d . c o m
Pro Tools
The Soft Keys page in the Avid Control app.
a much more accessible and
Soft Keys, and if you know where to find a particular mainstream product. The tech
soft key in Avid Control on one system, it will be in the remains largely unchanged,
same place on another. That isn’t to say there is no but it’s now much easier to use.
customisation possible, though. The last four pages Prior to version 5 the biggest
of the Soft Keys section are available for the creation issue with it was that what you
of custom commands. Any EuConised parameter can were buying was, in the words
be mapped, and combinations of commands can be of SoundFlow, “like buying
combined into custom macros as you would expect. a complex synthesizer with
The experience of using the Soft Keys from a tablet no presets”. You had to learn
can be disorientating and, rather than learning how to use it before you could use
to navigate the huge library of preset Soft Keys, I’d it. For busy professionals this
recommend setting up a page of custom Soft Keys you isn’t great — buying something
know you’ll use all the time, and getting the intended to save you time, which costs
benefit rather than trying to perform all tasks using Soft you time to learn. It’s not much
Keys just because you can, when you can perform them better for the aspiring amateur
faster using the old techniques of keyboard and mouse. for whom time is arguably less
Soft keys and macros are supposed to make things precious, as it’s a complex tool
faster and more convenient, but if a learning curve is which you have to learn to use
involved they can make things slower! before you can use it.
I can think of many products in the Pro Tools universe This is why I was so taken
which suffer from this learning curve disincentive. One with the new version 5. Very
day I’m going to learn UVI Falcon properly but I’m not specifically targeted to address
taking bets on when that day will be... the ‘learning curve penalty’,
which discourages both
State Of Flow tech-savvy professionals and
One third-party product which has always offered novice users alike, SoundFlow 5
potentially huge time savings for the busy Pro Tools user is made accessible through
is SoundFlow. This very clever scripting/macro app is a guided setup process, which
usable with all sorts of Mac software, but it started life as takes under 10 minutes, and
a Pro Tools solution and has remained Pro Tools-centric a pre-installed tranche of
throughout its development. It started life in 2017 as over 1700 Pro Tools-specific
a rather nerdy solution which allowed people who were macros over 30 pre-built
comfortable with scripts to semi-automate laborious control schemes. I tried the
or ‘click intensive’ operations and to benefit from the new version as someone who
incremental time savings which result. had previously passed on
Subsequent versions have taken a tool which was SoundFlow precisely because
written ‘by computer people for computer people’ into of being in too much of a hurry

to learn a new tool. I can confirm that even
someone as impatient as me managed
to get useful results and a real-world time
saving in Pro Tools in under 30 minutes.
Whereas the Soft Keys in the Avid
Control app rely on a parameter or action
having been EuConised by Avid, SoundFlow
allows almost any action to be recorded as
part of a macro. The huge collection of over
1700 Pro Tools-specific actions means that
for the majority of users, if they can’t find
what they are looking for in the pre-installed
layouts, they can build their own without
needing to get technical.
Macro Macro Man
Macros are sequences of tasks which can
be triggered via a keystroke, MIDI or OSC
trigger, or even a window being active on
screen. While you can use them to create
keystrokes you feel are missing from Pro
Tools (my vote is for a keystroke to toggle
the colour palette), it is in combination
with Elgato’s Stream Deck hardware that
SoundFlow becomes most appealing.
These relatively inexpensive peripherals
present a few illuminated LCD buttons and
combine the tactile experience of hardware
with the visual feedback of a software
UI. Stream Decks work seamlessly with
SoundFlow, but SoundFlow’s commands
can be displayed and navigated via an iOS
or Android device too. A perfect opportunity
to repurpose that old smartphone.
As for downsides, SoundFlow is Mac
only, and subscription-based. If either of
these are show-stoppers for you then the
Avid Control app is free and, if you engage
with the custom soft key features, can
SoundFlow running on an iPhone.
SoundFlow ships with over 1700
pre-configured macro actions for Pro Tools.
perform similar tasks. If you like the idea
of creating custom shortcuts for Pro Tools
then alternatives like Keyboard Maestro
are available, though they involve more
know-how to set up with Pro Tools than
SoundFlow 5 does.
Muscle Memory
The strength of Pro Tools is in its ubiquity
and consistency of operation: once you
know Pro Tools well you can walk up
to any Pro Tools workstation and start
work with no more to worry about than
remembering to reverse your modifier
keys depending on whether you’re using
a PC or a Mac. Would learning to operate
it via a highly personalised set of custom
controls isolate a regular SoundFlow user
from one of Pro Tools’ key advantages?
SoundFlow would argue not, because
your SoundFlow settings are stored in the
cloud and if you move to another machine
you can simply log into SoundFlow and
load up your settings — though that does
assume SoundFlow is installed, and that the
new machine is connected to the Internet.
However, as a new and enthusiastic convert
to SoundFlow and Pro Tools my concern
is that I may find it too useful... In the days
of landline phones I used to have my
most-used phone numbers memorised, but
these days I don’t know any phone numbers
at all apart from my own. Will the same
happen to my Pro Tools Shortcuts? I really
hope not — they took years to learn! 
w w w. s o u n d o n s o u n d . c o m / April 2022 113
 T E CH NI Q U E
Digital Performer
a row of tabs. You could choose between the horizontal zoom controls at the
We explore the many Soundbites, Loops, Beats, Tempos, bottom right, or DP’s regular zoom
tools available in DP’s Pitch and Volume. Under each tab was
a pull-down menu, which contained
key command: Command+right arrow
(Control+right arrow on PC). Then use
Waveform Editor. actions available for that Mode. It was the Pencil Tool, from DP’s main Tool
easy enough to find what you were Palette, to draw in your changes.
MIKE LEVINE looking for, but with all those tabs and
pull-downs, it took up a lot of real estate
Beat It

R
ecently, we’ve been focusing on in the top part of the window. The Beats Layer is integral to many
the new features introduced in In DP10, MOTU rearranged the types of actions and processes that are
DP11. But one of the highlights architecture of the Waveform Editor, available in DP. Thanks to the upgrade
of the earlier DP10 release was the changing the Modes to Layers. Now, you of the Beat Detection Engine in DP10,
complete redesign of the Waveform can access them from a single pull-down Beat placement is even more accurate.
Editor. It’s got a spiffy new look, and called the Edit Layer at the bottom of To make sure DP is analysing Beats
more importantly, a streamlined the screen (Screen 1). When you select automatically, go to DP Preferences
workflow and more editing options. a Layer, DP automatically changes the (Digital Performer / Preferences /
Here, we’ll take a look at some of the actions available in the Edit Layer menu, Background Processing) and make sure
important changes and additions, and at which is another pull-down just to its that you’ve checked the Automatically
the various types of editing available. right. DP switches to the appropriate Analyze Beats and Tempo options
First, though, why would you want tools automatically when you choose for both In This Project and Default
to use the Waveform Editor, when the a Layer.  For New Project. Also, confirm that
Sequence Editor
has many of the
same features?
The answer is
that it depends on
what you’re trying
to accomplish.
For example, if
you wanted to
pitch-correct a vocal
track in a mix or
automate Bite
Volume on a track,
you’re probably
better off doing it in
the Sequence Editor
where it’s in context
with other tracks.
The Waveform
Editor, on the other
hand, lets you work
on audio that may
not be placed in The Layers menu gives you access to specific
a track yet. For example e, you may want MOTU also changed tools for performing particular tasks.
to make loops for the Clips
C window the functionality and
from an instrument or vocal
v recording, names of some of what you’ve checked the Show Beat Grid
or extract individual samples.
mples You could were formerly Modes Lines in the Edit Windows section of
also transpose your sample in the Pitch and now are Layers, and added some the Preferences (Digital Performer /
Layer and specify loop behaviour in the powerful new capabilities. Preferences / Edit Windows).
Loops layer. You can easily do these The Audio Layer replaces the In the Waveform Editor’s Beats
sorts of things in the Waveform Editor, Waveform Mode of earlier versions of Layer, Beats show up as green lines
where edits apply to all instances of DP. It’s the layer where you can select with movable handles, as in Screen 2.
the Soundbite throughout the project. and destructively edit the actual audio The handles represent velocity and
It also provides better focus and more waveform of a Soundbite. You can choose are essential when extracting grooves,
precision than track editing. sections to edit or the entire Soundbite converting Beats to MIDI and more.
and, using the commands in the Edit Layer In the Beats Layer, you can select
Tab Fab menu, choose to Normalize the audio, add Beats, delete Beats, move Beats,
In earlier versions of DP, the Waveform a fade in or out, or apply plug-ins. change their velocity and more. If
Editor featured different Modes that you If you need to repair a click or pop, you’re extracting individual samples (for
could switch between by selecting from zoom down to the sample level using example a snare or kick) you can easily

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

 You can manipulate Beat placement and
velocity in the Beats Layer.
do that in the Beats Layer. Select a Beat,
which you do by clicking it, and then
opening Edit Mode Options and choosing
New Soundbite from Active Beat. DP
will select the audio from that Beat until
the next, and it will appear as a new
Soundbite in the Soundbites window.
There are also several new commands
in the Beats Layer. Copy Beats as MIDI is
one; it lets you create a MIDI version of
your Soundbite (minus pitch information).
Another is Play Beats as Clicks, which
creates a click track based on the Beat
placement in the Soundbite. You can
use that functionality to check the clicks
against the Soundbite’s audio to make
sure the Beats are correctly placed.
Screen 3). If you hover over an anchor
point, you’ll see a little arrow icon, which
lets you drag the anchor point to adjust
the boundary of the stretch region.
If you want the stretched audio to
sound natural, don’t stretch it too far.
That said, you can change the rhythm of
an audio file with careful stretching.
If you don’t like the results, select all
and hit Delete, and the Soundbite will
revert to its original state. Again, if you’re
working on a track in your mix, and want
to make a couple of rhythmic tweaks,
you’re probably better off using the
Stretch Layer in the Sequence Editor.
Spectral Views
Another new feature in the Waveform
Editor is the Spectrogram display (Screen
4), in which the vertical axis shows
frequency, the horizontal shows time, and
colour (and brightness) indicates harmonic
content. (You can also see individual
tracks in Spectrogram Display from the
Sequence editor.) The Audio Layer is the
only Layer in the Waveform menu where

Stretching Out
DP’s new Stretch functionality allows you
to stretch audio in several ways, all of
which are based on the detected Audio
Beats. In the Waveform Editor, you can
open a Soundbite and stretch Beats to
change their timing.
Choose the Stretch Layer, and
you’ll see the Beats as vertical lines
superimposed on the Waveform display.
Hover over one of the Beats until you
see the hand icon and use that to drag
the Beat. You’ll see the waveform move
along with it.
As you drag, notice the segmented
pink line on either side of the beat. The
vertical segments of this line are the
anchor points for the stretched beat (see Here, a Beat is being stretched. The vertical segments of the pink line are the anchor points.

w w w. s o u n d o n s o u n d . c o m / April 2022 115

 T E CH NI Q U E
T H E WAV E FOR M E DITOR Digital Performer
The Spectrogram
view lets you see
your Soundbite’s
frequency content.
Layer), but there
might be times
when you want
to place it after
the start of the
Soundbite. If you
do, the Quantize
Soundbite function
will look at the
Sync Point rather
than the first Beat
for positioning.
You can
manipulate
Soundbites in the
various Layers
you can see it Layer has some handy functions. For of the Waveform editor, but only in
in colour. In the other Layers, example, if you want to transpose er can you move
the Soundbites Layye
you only see a monochrome a whole Soundbite, use the Relative a Soundbite directly into the Tracks or
view when you’ve selected the Pitch option. Then you’ll see just Sequence Window ws. Open one of those
Spectrogram
p g Display.
p y a single
g continuous line representing
p g windows above or below
b the Waveform
To switch to the Spectrogram the audio, rather than the ones that editor in the Consolidated Window, click
view, pull down the View Mode menu, are created for individual notes in on the Soundbite from its header and
which is just to the left of the Edit Mode Absolute mode. drag and drop it where you want it in
menu, and choose Spectrogram. From the timeline.
the same menu, select Spectrogram
Take A Bite
Colors and choose from one of the The Soundbites Layer is where you can
Even More
options in the submenu. manipulate Soundbites. You can create The Bite Volume Layer makes it possible
a new Soundbite from a selection, to adjust the Bite Volume of the selected
Pitch & Time audition it and set its Sync Point. The Soundbite. Use the Pencil Tool from
In the Pitch Layer, you can apply Sync Point is critical for determining DP’s Tool Palette to draw in volume
non-destructive (ie. non-permanent) the positioning of a Soundbite when it’s changes, just like you would with volume
pitch-correction or pitch changes, similar quantised or moved. automation or other continuous data. If
to doing so in the Pitch Layer for audio For the most part, you’ll want to you want to create some unusual volume
tracks in the Sequence Editor. set the Sync Point at the first Beat of effects, you could use the Reshape Tool
If you’re working on creating the Soundbite (which is one of the and choose a shape for redrawing the
Soundbites, loops or samples, the Pitch commands available in the Soundbites Bite Volume, as in Screen 5.
The Loops Layer lets you create
and manipulate Sampler loops. Open
a Soundbite in the Waveform Editor,
do whatever editing you want in the
various Layers, then open up the Loops
Layer, and you can turn it into a Loop by
selecting it and choosing New Loop from
Selection in the Edit Mode Options menu.
With the loop selected, right-click it
and open the Sound File Information
Window. There you’ll see an Audio
Loop section at the bottom where you
can specify its behaviour (such as play
forward or backward) when it’s triggered
in a sampler, change its tuning and more.
Those are some of the essential
functions of the newly redesigned
Waveform Editor. Explore it yourself, and
you’re sure to come up with plenty more
You can use the shapes available in the Reshape tool to draw in volume effects in the Bite Volume Layer. ways to use it!  

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

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ROBIN VINCENT
W
ith the ridiculous range of
cinematic sounds we have at
our fingertips I think we’d all
relish the opportunity to score a movie.
But when the call from [insert favourite
director’s name here] comes in, is Studio
One up to the task of generating the
soundtrack to a film and putting you on
the road to your first Oscar? Perhaps the
better question is: how does video work
inside Studio One?
Studio One has no pretensions to take
on Premier or Davinci Resolve when it
comes to video editing, but it does have
some rudimentary video support to allow
you to write and compose in perfect sync
to moving pictures. We’re assuming here
that you’ve been sent a digital video file,
and it’s on your computer. We’re not going
to be tackling sync’ing to an external
video machine or capturing film ourselves.
Great Import
You’ll find the Video Player under the View
menu, but the easiest way to get video
rolling is to drag in a video file from the
browser. All supported video files appear
in the file list with a film strip icon next to
them. Supported formats include MP4,
3GP, MOV, AVI, MPG, M4V, MKV and WMV.
Drag the video file into your timeline and
it will automatically bring up the Video
Player ready to go. The video won’t exist
in your timeline like an audio track; it only
appears in the Video Player window.
You can use a couple of modifier keys
during the drag-and-drop to get a bit more
value out of that single action. Hold Alt
(Windows) or Option (Mac), and you can
extract the audio from the video at the
same time and place it in the timeline. Or, if
you hold Ctrl (Windows) or Cmd (Mac) you
can pull over the audio by itself,
although in this case, we really
do want the video.
You are unlikely to want to
extract the audio if the video has
come from a studio that wants
you to provide a soundtrack.
However, if you want the video’s
Importing a video is as simple as
dragging the file into your timeline.
118 April 2022 / w w w. s o u n d o n s o u n d . c o m
Studio One
audio to appear mixed with your music
when you render the project, you’ll need
that audio on its own track.
According to the Studio One
manual, when you drag in a video it will
automatically be offset to the point on
the timeline that you dropped it. This
has not been my experience. Whenever
I drop some video in it is always sync’ed
from the start of the project. If I’m
extracting the audio at the same time
then I do get a solid option to place it on
the timeline, and the video starts at the
same point as the audio. These things do
not have to stay linked, and if you hit the
little clock icon in the Video Player you
can set an offset so that the video starts
at a specific point on the timeline. If you’d
rather not have anything in sync, there’s
a button to turn the video offline (more
on that later).
On Screen
The Video Player window always stays
on top of your project as a floating
window. It would be a common
practice to plonk it onto a secondary
screen where you can blow it up to full
screen. Otherwise, it’s very resizable to
accommodate your screen real-estate.
You can only have one piece of video
in your project. You cannot queue them
up or have them scattered around the
timeline; the Video Player can have
a single video loaded into it at one time.
If you drag in another video it will simply
replace what’s already there.
On the player itself you have buttons
for loading and unloading a video (if
you didn’t want to drag one in from the
browser), audio extraction, offset and
some settings where you can change
the size of the video playback. The audio
on a video is muted by default but you
can change that if you hit the speaker
icon. As I mentioned earlier, if you want
the audio to be present in your exported
video file, you’ll need to extract it onto
the timeline. It’s also worth noting that
if you extract the audio to a track you’ll
want to keep the Video Player muted to
avoid any doubling up, because the audio
is still embedded in the video. Finally,
there’s a little chain link button that sets
the video online or offline. When online
the video remains frame-accurately in
sync with the project. When offline it is
unsync’ed, flowing and fancy-free.
In Sync
While the Online button is lit, the video
will remain in sync with the playback of
the project. It will start at the right time
and maintain sync throughout. However,
the latency in the playback of video is
greater than that of audio and so your
video may take a moment to buffer before
playing back. You’ll experience this as
a little skip at the beginning of playback,
which is simply the video catching up in
order to remain in sync.
Video will often come with visible
timecode stamped into the video stream.
To ensure the smoothest experience it’s
best to set your project to the same frame
rate as the video. You can do this in Song
Setup from the Song menu. There are 13
different frame rates to choose from.
Once in sync you can do useful things
like place markers on the timeline that
represent key points of the video. This
makes it much easier to compose music
to meet those events. If you enable
Cursor Follows Edit Position in the toolbar
you can drag a marker in the Marker
Track and effectively scrub the video to
get to exactly the right frame.
Now there are some scenarios where
you may be dropping a video onto an
existing project and so rather than writing
sound and music to the picture you
need the video to match up to whatever
is there already. I don’t mean you’ll be
time-stretching the video or anything like
that, but maybe you’ve had the video of
a live performance sent to you that you
now need to match up to the audio that
you recorded separately. This is where
offset comes in handy. There are no
special magic tools for sorting this out,
 The Video Offset tool lets you sync the video to your audio.
instead you’re going to have to use your
eyes and ears to get them in sync. So,
find something that happens in the music
that’s easily identifiable in the video.
Vocals is good because bad lip sync is
always the thing people are most likely
to helpfully point out. So, find a point at
which the vocals start, or maybe a big
Studio One’s Export Video options.
drum hit, and place the timeline song
pointer on it. Then use the offset box to
scrub the video until it hits on exactly the
same spot. You could use the embedded
audio to help you at this point if there
is any. You could also extract the audio
to a track and use the shape of the
waveform as a guide. Whatever way you
do it, it’s not difficult to match it all up with
a bit of trial and error.
Video Export
While Hollywood is unlikely to need you
to export a video with audio embedded, it
could be useful for smaller projects or as
a way of sending a draft composition to
a director or producer. You export video
in a very similar way to how you’d export
a regular audio mixdown. Select Export
Video from the Song menu and it will
generate a video that includes a mixdown
of any audio on the timeline within the
video’s length. If you are using external
effects or instruments, you should select
Realtime Processing to capture those in
the mixdown. For video codecs you can
use any that are installed on your system.
So, while the video options in Studio
One are rather basic they do offer
enough to give you the ability to compose
sound-to-picture all within the one DAW. It
would be nice to be able to lay video on
the timeline and perhaps do some simple
start- and end-point trimming and fading,
but before too long it becomes a very
different piece of software. 
w w w. s o u n d o n s o u n d . c o m / April 2022 119
 T E CH NI Q U E
LEN SASSO
T
his month we’ll use
Live’s Device Racks
to get more out of
Live devices and third-party
plug-ins. We covered the
basics along with a couple of
applications in the December
2011 Live column and explored
using Racks for complex
audio effects processing in
the February 2021 Live Tech.
Both articles are available on
the SOS website. This time
we’ll focus on using Racks to
create complex instruments and
MIDI processors.
Fender Bender
One thing I like to do is layer
other instruments with an
electric piano. That’s easy
with an Instrument Rack.
Start by creating a new MIDI
track (Command+Shift+T/
Control+Shift+T) and dragging
an electric piano instrument
to the track (the ‘E-Piano MKII
Bright’ Electric preset is a good
choice for layering). Select the
instrument and enclose it in an
Instrument Rack (Command+G/
Control+G). Reveal the Rack’s
Chain List by clicking the
bottom button along the left
edge of the Rack. Now select
the instrument you want to layer
with the piano and drag it to the
Drop Instrument area to create
a second chain. The Operator
preset ‘Camels in the Desert’
is an arpeggiated preset that
works well for this. Simpler’s
‘Aroura Pad 1’ preset is a softer
and less active sound that also
compliments the electric piano.
Screen 1 shows how to give
yourself both alternatives.
With the three Instruments
in separate chains of an
instrument Rack, I’ve configured
the Rack’s Chain Selector to
fade from the pad through the
EP to the arpeggiator, mapped
it to the Rack’s first Macro knob
and assigned the pitch-bend
wheel to that knob. I used the
Less Macro Controls button (‘-’)
to hide the other Macro knobs.
Pitch-bend’s default position is
120 April 2022 / w w w. s o u n d o n s o u n d . c o m
Live
What can Live’s Device Racks do for you?
1. Pitch-bend controls the Chain Selector (top) to crossfade between the pad, EP and arpeggiator chains of the Rack.
centred, which leaves only the EP audible. Pulling the In screen 2, I’ve Grouped MIDI Pitch,
wheel backward fades in the pad, pushing it forward Note Echo, Random and Velocity effects
fades in the arpeggiator, and releasing the wheel jumps into a MIDI Effect Rack. The coloured arrows
back to the solo EP. A MIDI keyboard’s wheels can only indicate the Rack’s Macro knob mappings
be mapped if Remote In is enabled for the keyboard in shown at the top. The Pitch effect serves
Live’s Link/Tempo/MIDI preferences. The downside is two purposes: it sets the pitch range of
that, once pitch-bend is mapped it is no longer sent to incoming notes and transposes the notes
Live tracks — you’ll need to send it from a different MIDI before they are passed to Note Echo. The
device to bend pitch. Random effect either randomises Note
Echo’s output (Rnd mode) or generates
Note After Note a cycle round-robin sequence (Alt mode).
Live’s Note Echo MIDI effect repeats incoming MIDI Turning its Macro knob to zero mutes the
notes. In Sync mode, numerical buttons across the top Note Echo chain. The Velocity effect keeps
set the repeat interval in 16th-note increments. The the output note Velocities between its Out
Feedback knob sets the rate at which the Velocity Hi and Out Low settings. With the settings
declines as the notes echo and feedback stops when shown in screen 2, incoming notes generate
the Velocity reaches zero. With Feedback at 100 percent three, four or five repeats depending on
the echoes repeat forever, but the knob acts in real their Velocity, and their pitch is randomised
time — turning it down starts the Velocities declining and within an octave above the incoming
turning it to zero percent stops the process immediately. note. Try it with the electric piano from
The Feed Delay knob alters the
Velocities of incoming notes,
thereby also affecting the number
of repeats. The pitch interval set
with the Pitch numerical is applied
to each succeeding echo. Thru
and Mute buttons determine
whether the incoming note is
passed through the effect, and
you may want Mute if the source
notes are also feeding another
track. In this example, I’ll assign
Macro knobs to only the Delay
Time and Feedback settings and
then use other Live MIDI devices
to process the results.
2. The Note Echo Rack at the bottom
processes incoming notes using the settings
shown top-right.
screen 1: play sparsely with a bit
of pitch-bend to blend in the
pad and arpeggiator effects.
In Four Parts
In screen 3, I’ve Grouped four
of the Note Echo Racks to
create the Quad Note Echo
Rack. Each of the Note Echo
Racks (Low, Low-Mid, High-Mid
and High) displays its first four
Macro knobs, which set the
Rack’s lowest note input, input
range, transpose and velocity
range. Those are typically ‘set
and forget’ parameters. The
remaining four Macro knobs
— Echo Rate, Echo Feedback,
Rand/Alt and Random Chance
— are hidden because they
are mapped to the Quad Note
Echo Rack’s Macro knobs
shown at the left. Those Macro
knobs are grouped by function
and colour coded to match
the source Note Echo Rack. If
you use a MIDI control surface
with eight Macro knobs, such
as the Akai APC40 MkII, the
first bank of those knobs will
control the top row (echo rate
and feedback) and the second
bank will control the bottom
row (random settings). The
settings shown let notes from
the C1, C2, C3 and C4 octaves
trigger respectively the Low,
Low-Mid, High-Mid and High
chains of the Quad Note Echo
Rack. With all Random Chance
knobs set to 0.0 percent the
notes played will be the notes
output. Other Random Chance
values will produce random
pitches within an octave of the
played note. Setting the inputs
of four separate MIDI tracks to
come from the different Quad
Note Echo Rack chains will
let you route them to different
instruments as well as capture
the results. Insert Scale MIDI
effects on those MIDI tracks
when you want to pitch-correct
the captured notes.
Four In One
For a more hands-on approach,
combine all four parts along
with scale effects and target
instruments on a single track as
3. The Quad Note Echo Rack holds
four Note Echo Racks in parallel. The
Macro knobs provide easy access to the
essential settings. Each capture track
holds a Scale effect and one instrument.
shown in screen 4. Start with the Quad Note Echo Rack settings will target 48 drum pads (C1 through
from screen 3 and use the input settings shown at the B4). For 64-pad Drum Racks, set the Xpose
bottom of screen 4: C1 triggers the low instrument (bass), Pitch knobs to multiples of 15 instead of 11.
C#1 the low-mid (horns), D1 the high-mid (strings) and You’ll find a variety of 64-pad Drum Racks
D#1 the high (choir). The chains’ Xpose Pitch settings in Live’s Drum Booth and Drum Essentials
restrict the four parts to successive octaves. That Rack Packs as well as in the Mad Zach 64 Pad
is followed by a new MIDI Effect Rack with eight chains Lab Drum Racks, which are available at no
(shown at the top). Its first chain is empty, whereas Scale charge from the Ableton website. You can
MIDI effects in the other seven chains map incoming also stack 16-pad Drum Racks by dragging
notes to various scales. Those chains are selected by them to separate chains in an Instrument
MIDI notes E1 through B1. MIDI note range C2 to B2 is Rack and then inserting a Pitch effect before
mapped to each Scale effect’s Base knob to select the each Drum Rack to restrict and transpose its
scale root. You can now trigger Quad Note Echo parts input. Because the Scale Rack in screen 4
with C1 through D#1, select the desired scale correction now blocks or swaps kit-pieces rather than
with E1 through B1 and select the root of the corrected correcting pitch, you may want to create
scale with C2 through B2. Alternatively, you can use custom Scale effect settings as well as
MIDI clips to trigger each of the Quad Note Echoes, and modulate the Scale and Base knobs.
you can play notes or
use clip automation
to change the Scale
Rack’s Scale and
Base settings. The
eight-note clip at the
bottom of screen 4
triggers combinations
of Quad Note
Echo chains.
For a different
approach, replace
the four-channel
Instrument Rack in
screen 4 with a Drum
Rack. The screen
4 Quad Note Echo
Rack’s Xpose Pitch
4. The Scale Rack (top) selects
a MIDI Scale effect chain (E1-B1) and
the scale root (C2-B2) to feed the
four-part Instrument Rack below. It
follows a Quad Note Echo Rack, whose
input settings are shown at the bottom.
w w w. s o u n d o n s o u n d . c o m / April 2022 121
 F E ATU R E

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

CPU Audio Performance Tested
Intel’s latest CPUs promise big performance gains.
But can they topple AMD in our audio benchmarks?
PETE GARDNER
C
omputers underpin most music
production today, and at the
heart of each computer is a CPU.
While some audio producers’ computing
needs are relatively modest, plenty of us
need more power — but when you want
to assess a chip’s capability for real-world
audio applications the headline specs tell
you only part of the story. As well as the
maximum single-core speed and number
of cores, factors include the price,
power consumption, memory handling,
heat, and the measurable real-world
performance for different use cases at
various audio interface buffer settings.
This article aims to help you understand
what the latest crop of CPU chips are
capable of when working with audio:
how many signal processing or virtual
instrument/sampler plug-ins you can
realistically run at a range of audio buffer
settings before you run into problems.
I’ve included some charts which show
how Intel’s new 12th-generation (Alder
Lake) chips compare with the previous
series and AMD’s current range, as
well as how RAM might influence the
performance. But I’ll start by bringing you
up to date on the main manufacturers’
progress since we last assessed CPU
performance, in SOS August 2020:
http://sosm.ag/core-wars-0820. I’ll
then explain some other significant
developments in the computing
world, before discussing some useful
‘takeaways’ from our tests for anyone
considering buying or building an audio
machine in the near future.
Catching Up
If you read that previous article, you’ll
recall that AMD were in a strong position.
Although Intel had dominated the
audio-computer world for the best part
of a decade, AMD’s 2000 series had
already begun to level the playing field
and by this time their well-specified and
affordable 3000 series had left Intel with
no option but to bring their prices down.
Meanwhile Apple’s decision to develop
their own chips had further ramped up
the pressure on Intel, and I predicted that
we may be on the verge of another ‘CPU
arms race’. So let’s quickly catch up on
what’s happened since.
Early in 2021, Intel released their
11th-generation (I’ll call it ‘gen’ from here)
Core processor range, which met with
a mixed reception. The Skylake desktop
chip architecture was replaced with
Cypress Cove-based cores borrowed
from the 10nm-based solution already
found in various laptop models. It was
certainly a step in the right direction,
delivering a significant improvement
to single-core performance; the fastest
model offered a base speed of 3.5GHz
and a maximum Turbo frequency of
5.3GHz. But new-gen chips usually
also benefit from a ‘process shrink’ (or
‘die shrink’; a physical scaling down of
the semiconductors) and this wasn’t
achieved. So the 10nm design had to be
fitted into a 14nm-chip package, and the
performance increase came at the cost
of relatively high power consumption
and heat. And that limited the maximum
number of cores...
For applications where single-core
speed is of paramount importance,
then, the 11th-gen chips held their own
despite stiff competition. But where
multicore processing is more important
they proved less impressive. The
top-of-the-range Core i9 11900K, for
example, had eight cores compared
with the previous-gen Core i9 10900K’s
10, and in several performance tests
involving tasks that efficiently exploit all
the available cores the 10900K scored
better, hence Intel continuing to offer
that chip.
AMD weren’t resting on their laurels.
Their much higher core-count chips
had already proved suitable for audio
workstations before their 5000-series
CPUs arrived at the end of 2020, several
months before Intel’s 11th-gen chips.
AMD’s 5000 series showed that the
company were able to remain a step
ahead of Intel, in terms of both release
schedule and CPU performance. AMD
issued an interesting note ahead of
the 5000-series release, explaining
that they’d not only delivered higher
single-core speeds but had also
w w w. s o u n d o n s o u n d . c o m / April 2022 123
 F E ATU R E
CPU BENCHMARK TESTS
Diagram 1: The DAWBench DSP test compares
the performance of different CPUs when running
effects plug-ins.
reworked the memory handling, with
a new, larger, shared level-3 cache
— a more tightly integrated design
that eschewed multiple CCX cores in
favour of a single-block arrangement.
Putting this in more musician-friendly
language, the performance of the earlier
generation Ryzen chips often suffered
at the smallest ASIO buffer settings
(which are needed for low latency) and
this could be partly alleviated by using
faster-than-specified memory, effectively
overclocking to solve the problem.
The new system promised to eliminate
cross-CCX latency (a result of the data
being passed back and forth between
CCX cores) and thus to enable better
performance at low latencies.
Windows 11 & Alder Lake
Windows 11, released towards the end of
2021, a few weeks before Intel’s 12th-gen
hardware, included some optimisations
that Microsoft developed in tandem with
Intel to exploit their new chips. As well
as benefitting from the belated process
shrink, the 12th-gen chips are the first
widely available PC-based hybrid-design
CPUs. ‘Hybrid’ means that the Alder
Lake CPUs employ two core types, one
geared towards higher performance and
124 April 2022 / w w w. s o u n d o n s o u n d . c o m
the other power efficiency. (Similar in
concept to Apple’s M1 chips, this is often
referred to as a ‘big.little’ configuration
after ARM’s big.LITTLE design.) As with
the 11th-gen Core i9 11900K, the new
flagship Core i9 12900K offers eight
hyper-threaded high-performance cores,
capable of exceeding 5GHz. But it also
includes eight ‘efficiency’ cores, which
aren’t capable of hyper-threading and
top out at 3.9GHz, so this CPU supports
24 threads in total.
Although Apple’s move away from
Intel silicon seemed to go relatively
smoothly, early adopters inevitably
encountered some problems: lots
of software had to be updated and
recompiled to work on the new platform;
Apple provided the Rosetta application
compatibility layer in Mac OS Big Sur
but its success with audio applications
was a bit hit and miss. As more software
has been reworked for native M1
compatibility and users have settled on
alternatives to legacy software, such
problems are quickly evaporating. But
the experience nonetheless prompted
observers to wonder if similar disruption
might plague early adopters of Windows
11 and the Intel 12th-gen platform.
Microsoft’s big change to support
Intel’s hardware is a reworking of
Windows 11’s scheduling, with something
they call Thread Director technology.
Historically, because audio applications
look to spread their workload across
all the available cores, ‘staggered’
core performance (different speeds
on different cores) could reduce the
maximum attainable performance.
Older Intel chips often staggered their
maximum Turbo speeds — the slowest
core could end up clocked about 10
percent slower than the fastest core
— and for audio handling this could result
in one core overloading early, potentially
causing the project to overload long
before the CPU’s theoretical capability
was reached. Past suggestions to
address this have been either to disable
the Turbo entirely or to improve cooling
and run all cores at the same maximum
speed, overclocking the slower ones for
the best possible performance. But over
the past few CPU generations, Intel have
moved away from this staggered-Turbo
approach and given each chip
a maximum Turbo rating for applications
limited to running on four cores or fewer,
while offering a slightly slower all-core
maximum — all cores now operate with
the same upper limit.
Other important advancements in
Intel’s Alder Lake architecture include
support for PCIe 5.0 and DDR5 RAM.
The PCIe 5.0 lanes offer significantly
greater bandwidth than the PCIe 4.0
implementation introduced on AMD’s
 F E ATU R E
CPU BENCHMARK TESTS
Diagram 2: The DAWBench VI test assesses
the capability of different CPUs when running
sample-based virtual instruments.
X570 range in 2020. The key devices
to have benefited from this so far have
been a small number of high-end GPUs
and solid-state storage drives, with the
latter being of more obvious interest
to music makers. NVMe SSDs are
a compact storage format that normally
mounts directly on the mainboard,
and they’ve been around for a number
of years now. The original PCIe 3.0
versions offered an upper transfer limit
of 3500MB/s, while the fastest PCIe 4.0
drives reach in the region of 7000MB/s.
PCIe 5.0 drives have a theoretical
limit of 14,000MB/s — twice that of
PCIe 4.0 — and early announcements
in the storage world suggest that
they’re on course to become available
commercially in 2022.
The new DDR5 memory may be
of even greater interest. Memory
has tended to have a limited impact
on the DAWBench DSP test for
effects plug-in handling, since that
focuses on the CPU performance.
But NI’s Kontakt library has benefitted
from faster memory. Initially, DDR5
offers speeds of up to 4.8Gbps, so
126 April 2022 / w w w. s o u n d o n s o u n d . c o m
promises to exceed DDR 4’s 3.2Gbps
limit by 50 percent. It’s a very promising
development for those using lots of
instruments in large DAW projects.
Benchmark Test Results:
Key Points
Before we consider the performance
tests of specific processors, it’s worth
pointing out that although some AMD
users reported performance hits when
switching to Windows 11 immediately on
its release, these were addressed very
quickly with a series of updates from
both Microsoft and AMD. We started
our own tests a few weeks later and,
with those initial problems already
resolved, we noticed that performance
had changed for both Intel and AMD
chips, and mostly for the better. Across
the board, both companies’ CPUs
made gains on the 64-sample buffer,
showed less variance on the 128 and
256 settings, and tended to lose a small
amount of performance on the 512
setting. The older Intel chips tended to
exhibit a greater swing than the AMD
hardware but, ultimately, the change
in OS looks to be a positive one for
recording, no matter your choice of CPU.
Moving on to specific processors,
Intel’s 12th-gen CPUs responded well
in our tests, with Microsoft’s Thread
Director approach apparently delivering
the hoped-for improvements in
performance. Scaling was consistent
across the 12th-gen CPUs, with all
chips offering even load balancing and
capable of being fully maxed out at all
ASIO buffer settings. Our DSP test (for
effects/processor plug-ins) provides
a measure of raw CPU performance and
entails repeatedly loading instances of
the ReaXComp multiband compressor
plug-in until the ‘breaking point’ is
reached. (There are thirstier plug-ins, but
using ReaXComp makes results more
comparable with those from previous
tests). As you can see from the charts,
while the 24-thread Core i9 12900K
demonstrates the 12th-gen platform’s
strong potential, it clearly comes in
second place, behind the 32-thread
AMD 5950X.
For the VI test (virtual instruments),
we keep increasing the polyphony
in use in NI’s Kontakt sampler, so
as to raise the load on the system
until ‘breaking point’. As Kontakt
can benefit from both RAM and CPU
 Diagram 3: The DSP test again, but comparing
the same Intel processors when used with DDR4
and DDR5 RAM.
performance, the test tends to favour
higher memory-bandwidth platforms, and
Intel’s X299 quad-channel platform has
historically tended to prove the strongest
performer with memory-intensive loads
like composition and orchestration. This
test allowed Intel’s new DDR5 memory
support to flex its muscles as we stepped
through the ASIO buffer settings. With
the smallest buffers the older Intel Core
i9 10980Xe held on to first place, but the
new, dual-channel DDR5-based Core
i9 12900K outperformed it once we
increased the buffer past 128 samples.
The AMD memory handling discussed
above also seems to have paid
dividends: using the same mainboard
and memory, we measured latency
as being 10-15 percent lower than for
previous equivalent models. In practical
terms, this means smoother performance
with a 64-sample buffer, with the CPU
now capable of being placed under full
load at the tightest of settings. This is
where the older generations often fell
short, with audio breakup occurring
earlier than expected.
Choose A Chip
Both Intel’s and AMD’s latest platforms
have their strengths, then, so which
should you opt for if you’re specifying
a new system? Well, at the current point
in time, the best option for most users

Intel’s 12th-gen Core processors benefit from

a so-called big.little architecture, which overcomes Intel’s 12th-gen chips support DDR5 RAM, which far
problems in previous generations when running outperforms DDR4, but at the time of writing the largest
different cores at different speeds. kits theoretically supported aren’t yet widely available.

w w w. s o u n d o n s o u n d . c o m / April 2022 127

 F E ATU R E
CPU BENCHMARK TESTS
Diagram 4: The VI test, comparing the same Intel
processors when used with different RAM.
could well be ‘either’! I’m not sitting
on the fence — it’s a reflection of the
fact that while our tests highlight the
strengths of each CPU in particular
usage scenarios, real-world projects
typically require a machine to do a bit of
everything, with the balance changing
from project to project.
For general, all-round performance,
the publicly available Passmark CPUMark
benchmark results
offer a quick, rough
comparison against
older chips. These
“The Intel 12th-gen CPUs clearly exhibited
suggest that even
the lowest-model
current-gen chip
on test, the AMD 5600X, performs
comparably to the Intel Core i9 10900K
(which was top dog as recently as 2020)
while costing about half as much. But in
our own tests, the Intel 12th-gen CPUs
clearly exhibited the fastest single-core
performance, with the Core i9 12900K
scoring roughly 10 percent higher than
the (higher core-count) AMD 5950X.
Scenarios that require strong single-core
performance, such as real-time effects
processing whilst recording with small
buffers, are likely to respond well to that
higher speed. In-the-box production,
particularly intensive mixdown processing
128 April 2022 / w w w. s o u n d o n s o u n d . c o m
at more relaxed buffer settings, is likely
to make better use of the additional
cores of the AMD or older Intel X299
chips, despite the lower headline speed.
Another deciding factor could be
memory. Intel’s older X299 platform
supports the most RAM (up to 256GB)
and the AMD AM4 platform will handle
up to 128GB, which will still be more
than ample for most users. Both use
DDR4 RAM kits, which are widely
available. Intel’s newer 12th-gen CPUs
the fastest single-core performance.”
run on the Z690 chipset, which will
eventually offer up to 128GB of faster
DDR5 RAM. However, DDR5 was only
released recently, and is currently
in short supply — see the ‘Supply
Problems?’ box for more about that.
Note that all the platforms tend to
perform at their best when used with
memory that’s one step up in speed
from the base specification, though
going higher than this will tend to offer
smaller incremental gains, and may
also affect stability; we run our systems
with a minor speed bump, to ensure we
reach a performance and stability sweet
spot. In these tests, the AMD models are
running with 3600MHz RAM, the Intel
10th-gen are running with 3200MHz
kits, and the Intel 12th-gen are on
5200MHz kits.
It’s worth noting that as well as the
DDR5-capable Z690 mainboards, we’ve
also seen a number of DDR4-based
models that support these 12th-gen
chips. These could allow someone to
upgrade the CPU and keep using their
older memory. That sounds a tempting
cost-saving
measure but
we noticed
a negative impact
on performance.
The DSP test,
with its CPU
focus, predictably shows a very small
variance, with little to differentiate the
two memory standards. In the more
memory-intensive VI test, though, we
saw instances of performance jumping
by around 40 or 50 percent, for various
ASIO settings. In short, if you can get
enough of it, DDR5 is easily the best
choice for anyone considering this
12th-gen Intel platform for those kinds
of projects.
On The Horizon
Looking ahead, there are plenty of
interesting developments on the
 Supply Problems?
DDR5 launched at the same time as the Intel
Z690 platform and, so far, supply has been
hampered by component scarcity issues.
Memory manufacturers Micron stated, during
the final week of 2021, that the delays were
down to a shortage of the VRM and PMIC
(power management integrated circuits)
components that are required for DDR5
production, and that delays are likely to
continue well into 2022. Furthermore, many
manufacturers are focusing initially on the
most popular kits: 32GB (packs of 2x16GB
sticks) is the only widely available size at the
time of writing, although 64GB kits are slowly
starting to appear. Early compatibility issues
have seen problems with mixing these packs
into quad-stick arrangements, but this too is
beginning to be addressed with supporting
BIOS updates. Some factory-matched quad
packs have been fully validated by a number of
mainboard manufacturers and listed as suitable
on the current qualification lists, but with the
continued focus on meeting the demand for
the popular smaller packs, their availability may
remain limited for some time.
Supply chain problems are not limited to
memory; semiconductor production generally
horizon. Intel’s highest-end chips are
currently only represented on the older
X299 platform in our testing round-up,
and its performance is starting to be
eclipsed by Intel’s newer midrange
chips. A long-awaited update to the
platform due later in 2022 is likely to
feature many of the advancements
currently supported in the Z690
line-up. Should it get the same core
design further scaled up, along with
the new RAM support we’ve come to
expect with the current generation, it
should prove a very worthy successor.
There’s talk of AMD refreshing
their 5000 series some time early
this year, with a new 3D cache design
which they suggest could potentially
deliver a 15 percent performance
gain in some applications. It’s also
suggested that the refreshed chips will
drop straight into current-generation
AM4 mainboards — they may well be
the final upgrade option for anyone
already running this generation of
board. AMD also recently announced
plans for a new 6000 series,
though this seems to be focused
on mobile platforms, rather than
high-performance desktop chips.
For desktop users, they’re looking to
release the Zen 4-based 7000-series
Ryzen chips later in 2022, along with
is struggling to keep up with demand. The key
fabrication plants tend to provide hardware
for more than PCs, with microprocessors
for cars, washing machines, TVs... and just
about anything else you can think of. The
main players are rushing to expand and open
new facilities that can meet demand but, with
reported wait times being up to 12 months
for new production runs over the course of
2021, it’s clear that until these new facilities
come online hardware supply may remain
limited throughout 2022. For computing, this
shortage was initially felt most strongly in the
GPU market, where models old and new have
been in short supply since late 2020. High
demand for crypto mining and limited supply
runs have severely impacted availability for
the last 12 months, and we’ve thus seen
prices rocket for the small number of cards
making it to market. At the time of writing,
this has placed Intel in an interesting position,
since many of their 12th-gen chips feature
a GPU suitable for a DAW machine. AMD have
continued to bring a steady supply of chips to
market, but their need for a separate graphics
card has sometimes been an obstacle for the
keen self-builder.
a new AM5 chipset-based platform
that should see AMD catch up with
Intel on DDR5 and PCIe 5.0 support.
Apple are, of course, continuing
to roll out their own hybrid CPU
range. The first-gen M1 chip has
been very well received, and both
CPU Benchmark and GeekBench
results suggest the M1 10-core found
in the Pro models ranks somewhere
between Intel’s 11900K and 12600K
in terms of raw performance,
achieving this on a maximum of
40W — that’s roughly 25 percent of
Intel’s Turbo power requirement at
full load! It’s a strong chip for more
modest workloads, and it will be very
interesting to see how this technology
scales up in future iterations.
We’re now seeing great-value
CPUs that match the very best from
just a few generations ago — anyone
with an ageing system who is maxing
out their current setup should be well
served with a computer hardware
upgrade at this time. The past 18
months have seen the arrival of ever
more powerful chips and feature-rich
mainboard solutions from each of
the key players, so it’s going to be
interesting to see how they continue
to build upon these platforms over the
coming generations. 
u
 F E ATU R E

Hear The Sound

W www.youtube.com/
watch?v=5iBlEoTREAg
W https://open.spotify.com/
Carl Bell in his studio.
track/0iWoLKWzC864JQx0B5Yif2

Mars, to name but a few. Their favourite of

the many sounds they’ve created together
is a solo on ‘Solace’ from Fuel’s second
album, Something Like Human (2000).
Carl Bell: “In the middle of the song
‘Solace’ there’s this solo section where
you hear this strange ‘wo-ah, wo-ah’ kind
of folding sound. That was created using
a small DigiTech Talker pedal in a very
unorthodox way, where it was kind of
imitating a voice box. Instead of speaking
into the microphone, it had an option where
you could get a microphone and plug it in
and with that, you could change the sound this being now more than 20 years ago,
with the microphone. And so what we did I can’t remember the specifics.
was, instead of changing it with a voice, “The DigiTech box we used for this
we turned it around to face the nearfield particular track had a mode where you
monitors in the studio and we just let the could switch it to become a vocoder. I ran
track kind of mess with the sound and the guitar signal through the box before
that’s how we achieved it. It was a very it went to the amp. Then I took a Shure
cool technique and we got some weird SM57 and plugged that into the voltage
Carl Bell & Ben Grosse sounds happening.”
Ben Grosse: “As far as I remember this
control input and had the microphone
placed on a boom, and as I swung the
Fuel ‘Solace’ track was done in Los Angeles at my studio boom around towards the studio monitors,
The Mix Room, but we did do some solo it would envelope in a really great way,
guitar tracking at Sear Sound in New York and it also would respond to the volume
J O E M AT E R A
as well, so there’s a small chance that’s peaks, for example from the drums, and

C
arl Bell is guitarist and songwriter
from American post-grunge
rockers Fuel, while producer Ben
Grosse has worked with the likes of Dream
Theater, Disturbed and Thirty Seconds To
where it was done — I always have my
guitar boxes with me when I travel. That
said, I seem to remember swinging the mic
around in Studio A at The Mix Room for
this. I’m sure I did some EQ’ing, but with
give the envelope some pulse. As I moved
the mic closer it would envelope up almost
sounding like a wah-wah pedal. That way
I could control the envelope, along with the
pulse of the song adding to it as well.”

130 April 2022 / w w w. s o u n d o n s o u n d . c o m

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 IN SIDE TR AC K

culled. At the time of writing, the song

was at its third week at number one in the
Billboard charts and its second week at the
top of the Billboard Global 200 chart, with
the album enjoying a fifth week at number
one in the US. The single and album also
held number-one positions in the UK, while
other songs from the musical, like ‘Surface

David Boucher Pressure’ and ‘The Family Madrigal’ were in

the singles top 10 in both countries.
With six lead singers, an orchestra and more, ‘We Don’t Talk The soundtrack blends Latin music
styles, particularly from Colombia,
About Bruno’ from Encanto was always going to be an epic like vallenato, cumbia and bambuco,
production challenge — and then lockdown happened. with hip-hop, pop, rock and Broadway
influences. Unusually for a featured Disney
PAUL TINGEN
he created the feel of all the characters, with song, and in fact any hit song, ‘Bruno’
them stepping on each other’s lines and features six main lead singers, aided by the

“L
in-Manuel Miranda’s Logic interacting with each other, was incredible!” movie’s entire cast.
demos were always great road Also incredible is the enormous success
maps,” recalls engineer and of ‘We Don’t Talk About Bruno’, and the
Growing Pains
mixer David Boucher, “but his demo for ‘We soundtrack for Walt Disney Animation David Boucher was at the heart of the
Don’t Talk About Bruno’ was insane! How Studios’ Encanto movie from which it is process of developing Miranda’s demos for

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


the Encanto soundtrack, working closely
with soundtrack music producer Mike
Elizondo (Eminem, 50 Cent, Fiona Apple,
Maroon 5, Muse, Ed Sheeran, Rag‘n’Bone
Man), as well as movie directors Jared Bush
and Byron Howard. Boucher’s work on the
soundtrack began as far back as May 2019,
and once 2020 hit, the project was greatly
affected by the pandemic.
“Lin would send his demos to the
directors,” explains Boucher, “and if they
liked the song, they would send it to Mike.
He’d work on the demos at his studio
in Nashville, adding bass and further
developing the arrangements, and then
Mike and I would discuss who would play
on the songs. We started with recording
some accordion and percussion for the
opening track, ‘The Family Madrigal’,
in the basement at Disney, fleshing out
the arrangement to see if it worked. The
demos were then given to [original score
composer] Germaine Franco, and she would
arrange the orchestral elements for a proper
orchestra.
“We recorded more bits and pieces
during the rest of 2019, and when the
pandemic hit, it was a big deal. Everyone
switched to working remotely and using
Zoom and Audiomovers. I was often
the only one in the control room, and it
was really lonely! During the pandemic
I only did the sessions in Los Angeles, at
EastWest Studios, United Studios, and the
Fox sound stage. The rest of the sessions
took place in New York, Colombia, Atlanta
and tons of other places. The number of
studios we ended up using was shocking!
“Initially I was set to record the score as
well, but we ran out of time, and I brought in
Greg Hayes to record at the Fox Newman
Recording Stage, while Alvin Wee continued
recording at the Village, and he then mixed
the score. Because of Covid restrictions,
‘We Don’t Talk
About Bruno’
Written by Lin-Manuel Miranda
Produced by Lin-Manuel Miranda
& Mike Elizondo
we could not have the entire orchestra in
one room, so we had separate recording
sessions for strings, woodwinds, brass
and orchestral percussion. This meant
that some Pro Tools sessions grew to
600-700 tracks. Throughout the whole
process I kept an overview of what we had.
Everything was 96kHz, so I had about 8TB
of data to maintain.”
Getting It Done
Keeping track of a two-and-a-half-year long
project, recorded in many different studios
around the world, with 14 lead singers
and track counts in the hundreds, sounds
insanely complicated. “Yes, it was hard,”
agrees Boucher, and if he says so, it must
have been, because he is one of the most
experienced film-music engineers and
mixers in the industry, and has worked on
many Disney projects.
Originally from Atlanta, Georgia,
Boucher studied music engineering at
the University of Miami, and then cut
his teeth as assistant to the great Bob
Clearmountain. Boucher went independent
in 2001, and has since worked regularly
with producer Mitchell Froom, gaining
credits like Crowded House and Rufus
Wainwright. Boucher’s movie soundtrack
credits include Moana, Coco, Monsters
University, Toy Story 3 and 4 and The
Princess And The Frog, and he was
featured in the April 2014 issue of SOS
talking about recording and mixing the
w w w. s o u n d o n s o u n d . c o m / April 2022 133
 IN SIDE T R A CK
DAV ID BOUC H E R • E NC A N TO

soundtrack for Frozen.

Boucher needed every bit of his
vast experience to keep the Encanto
soundtrack project on track. One crucial
aspect of his work was making sure that all
final vocals were recorded and delivered
to the animators as early as possible.
“Usually the order of operations is: get
the demo, make sure the arrangements
fit the film, record all the vocals as early
as you can. You then replace the demo
arrangement with the final instruments,
or add them. If there suddenly is a crowd
or a percussionist in the street, you can
always add that later.
“When we’re working on the vocals and
the compositions, the movie is still in the
storyboard process. The film-makers want
the lead vocals as early as possible, so
the animators can get to work. The story
often changes, the character’s emotional
viewpoint changes, sometimes there are
lyric changes. Changing animation on
a lead vocal when you have lip-sync is

Photo: Jake Dahm

Lockdown recording is
time-consuming and expensive, so to do a lonely business! David Boucher
that is not a decision that is taken lightly. at Sunset Sound Studio 2 in LA.
Once the animation is complete, all vocals
are locked for timing.”

No FX
After vocal recording comes vocal
production, and Boucher did most of this
as well, though with expert help. “When
working remotely, we made comp notes,
and then the engineers sent me all the
takes, and I comped them in my studio.
Mike [Elizondo] was focused on the music
production, so Andrew Page, who is VP
of Music Production at Disney Animation,
helped me with the vocal production,
because it was an insane amount of work!
He did this via Zoom and Audiomovers.
Nothing gets past him!
“After the first comp, Jared, one of
the directors, would come in from time
to time via Zoom, and he’d go shopping
for lines that told the story best from
a directorial standpoint. One of the nice
things about the Zoom way of working
is that I get more time with the directors.
They have a lot of work on their plate,
but in the Zoom world, they can go from
Zoom meeting to Zoom meeting and at
six o’clock have dinner with their family,
because they’re at home!
“The process of vocal production is
that we first comp the vocals, and then
I time and tune them. I use Melodyne for
this and do everything by hand. I take
a first-do-no-harm approach. I look at the Engineer Shawn Coleman meticulously noted his mic placements and gear settings when recording
page and listen to Lin’s demo vocal, and Dianne Guerrero, in case further vocal sessions were needed.

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

then I simply listen to the melody and make
sure it’s intact. I’m not looking for perfect
tuning. I am only looking for the actor to
represent the character’s emotions and
Lin’s composition.
“People are more relaxed about tuning
today, but in the film world things have
changed over time. There was a time
when it was almost taboo, and then there
was a period when you couldn’t tune
too much. Now it’s in between the two.
I’m fortunate that when I work on pure
audio albums, I don’t tune at all. On the
Andrew Bird album that I just finished,
and with Crowded House, and when
working with Rufus Wainwright, I don’t tune
anything. But of course, if there’s a great
performance, and one note a little blue,
and the artist wants that fixed, I’m happy
do to that.”
Vocal production is also about volume
riding, adding EQ and compression, but,
when working for film, says Boucher, very
little else. “I always use a Massey de-esser,
and the Massenburg Designworks EQ5,
and I now use the FabFilter Pro-Q 3 instead
of the Waves F6. The Waves Kramer PIE is
my favourite plug-in vocal compressor.
“But the immediate reaction from the
directors and music personnel is that
vocal effects are persona non grata.
They don’t want record-style reverbs
and delays. When somebody’s telling
a story, and especially with Lin where
there’s a lot of words in there, delays and
Vocal Recording
Vocal recording is perhaps the single most
important part of the production process, and in
the case of Encanto, the pandemic added an extra
layer of complexity, as David Boucher recalls:
“When a singer is in Colombia, or somewhere else
in the world, it takes more time to schedule things.
And in general it’s hard to feel collaborative when
everybody is remote! So I was really happy to do
the Los Angeles sessions in person. The singer and
I would both test the day before, and we made
sure we had the test results before we met. I did all
the recordings with Stephanie Beatriz, the actor for
Mirabel, at a dialogue studio in Glendale under the
Disney umbrella.
“I go to these vocals sessions with a couple of
different mics, and it turned out that my AKG C24
stereo microphone worked best for everyone.
Once you pick a mic for someone in an animation
production, you stay with it, in case you need to
re-record the singer later. You want to keep the
same tone. I also bring my Undertone Audio MPDI-4
mic pre, and Chiswick Reach valve compressor. The
Undertone sounds fantastic, and the compressor
also is really beautiful-sounding on vocals.
“I like using the C24 because the idea is to
Compared with a typical pop or rock track,
vocal effects were used very sparingly, with almost
no reverb. This was a typical vocal plug-in chain.
reverb just mess with the intelligibility,
and if you can’t understand the words,
you’re in trouble. They also want to
give the vocal a sound that matches the
picture. So once we’re done with the
vocal production, I deliver the vocals LCR,
with a completely dry vocal in the centre
have a main pickup and a backup microphone in
case the main microphone overloads. Especially
with dialogue, the dynamics are extreme, and
sometimes they are extreme in movie songs as
well, because singers can go from a very quiet
whisper to almost shouting. I never liked the sound
of a more distant microphone, and prefer to record
close up, so I used my C24 with one capsule as
my main pickup, and the other capsule is padded
down, and for backup.”
Boucher also supervised the remote sessions,
with suggestions for what mics to use. “Before
the remote sessions I was sent video or audio
examples of the actors singing, and I looked at the
microphone list of each studio where they were
going to record, and picked what I wanted as
primary mic. For example, at Reservoir Studios in
New York they have a great Neumann U67 that we
used a lot.
“There was a buddy of mine, Shawn Coleman,
who recorded Dianne Guerrero in Atlanta with
a Neumann M149. I needed people who could
record a singing lead vocal, not dialogue, which
is a different art form. After the session Shawn
sent me pictures of the mic height, using a tape
and a bit of reverb in the left and right
channels, just to add some sustain. I also
send the film-makers an instrumental
stem and a percussion stem.”
Work In Progress
After Boucher delivered the film crew the
vocals and music stems, he continued
to record instruments, and of course
new vocals if and when they were
measure, and pictures of the settings of the gear,
so I could match the tone if new vocals had to
be recorded with Dianne. I later did a session
with her in LA, and put the 149 at 57 inches from
the floor and the same outboard settings, and it
matched perfectly. The safety mic in this case was
a Neumann KM84 with a pad on it.
“Finding people who can really capture
a vocalist’s personality was one of the biggest
challenges. You can’t just throw up a microphone
and set it so it doesn’t clip, because you’re unlikely
to get the juice. It’s why I used engineers like
Shawn and Joey Raia and Kevin Harp. There also
were two engineers in Bogotá, Colombia, at Fox,
who did a great job recording.
“The more we could take off the plate of the
recordist, the better, because I was desperate to
keep them focused on getting a great vocal sound,
and not on the technical side. For this reason we
sent Shawn and the other recording engineers
a Pro Tools session for them to record into. These
sessions had a stem of the instruments, a stem of
the drums, and all the engineers had to do was set
things up for Audiomovers, and adapt the I/O for
the place where they were recording.”
w w w. s o u n d o n s o u n d . c o m / April 2022 135
 IN SIDE T R A CK
DAV ID BOUC H E R • E NC A N TO
When the time comes to create the stereo mix
for the soundtrack album, David Boucher will add
a small amount of mix-bus processing.
needed. “The drums were played by Matt
Chamberlain, who has a great recording
setup. As I mentioned earlier, Mike
would play bass, and the MIDI orchestral
elements were replaced, and we added
instruments that were recorded in New
York, guided by Mike and the amazing
Edmar Castañeda. Some electronic
elements were not replaced, like the
MIDI trumpet in ‘Bruno’, because it was
supposed to be a synth-like sound.”
During the entire process, Boucher
would keep making rough mixes of
everything he had, moving between what he
calls the “multitrack” — ie. original Pro Tools
sessions with all tracks separate — and the
stems. “I always had a working mix going,
with rudimentary stems. I never reduce the
original multitrack. The final mix is always
derived from the multitrack, because things
change all the time, and when you’re locked
into stems, you are dead.
“My sessions are the master sessions,
for everything. I always try to keep it pretty
mellow in the plug-in world and, if I can,
also with the track count, but with this
project I couldn’t do the latter. Things were
just recorded in too many different places.
136 April 2022 / w w w. s o u n d o n s o u n d . c o m
The fortunate thing is that I could keep
the amount of plug-ins down, because we
recorded things the way we wanted to hear
them, and this meant that we didn’t need 10
plug-ins on every track.”
Beyond Stereo
Adding to the complexity, Boucher was not
only dealing with huge track counts and
session material from all over the planet,
but today’s surround-sound requirements
for the re-recording mixer. For Encanto he
delivered stems to David Fluhr, who was
tasked with mixing all music, singing and
dialogue to picture.
“I started the final mixes at the John Ford
mixing stage at Fox. Mike flew out to LA for
that, and we finished the mixes together,
and got them approved by the directors, Lin,
and music executives. It is during this stage
that I spend the most time on getting the
final sounds I want and add most plug-ins.
Everything is in the box, because things
change all the time, and there’s no room
for recall.
“As I mentioned earlier, we did not
overuse plug-ins. With regards to the
vocals, once you’ve lived with their level
of dryness, it starts to become the sound
of the mix. I’m not going to add a bunch
of effects. Sometimes Lin will have some
effects in his demos that we like, and I’ll try
and imitate those. But in general, it remains
pretty dry. When I do add reverb, I lately
like to use the LiquidSonics [reverbs], for
example Cinematic Rooms, Lustrous Plates
and Seventh Heaven. I have a Bricasti, and
I know all my presets on that, and you can
just put that right into Seventh Heaven, and
it’s close.
“I deliver these film mixes to the
re-recording mixer as composite mixes,
which are the sum of the stems. When you
zero all stems, they are the same as the final
composite mix and will phase-cancel out.
For some films I deliver straight 5.1 stems,
but that method creates big track counts
very quickly when there are on-screen
performances of multiple vocals. So I tend to
move to a format of varying track widths.
“On ‘Bruno’ there are 12 lead vocal
stems, and four group vocals stems, so that
alone is 68 tracks, because the lead vocals
are LCR, and the group vocals are 5.0. The
instrument tracks are in a variety of formats,
like stereo for some of the programmed
drums, because they originated from
a stereo or mono keyboard, all the way up
to 5.1 for the orchestral elements.
“The instrument stems are all in
folders. Every stem is a routing folder. As
I mentioned, I will print my composites mix
from the multitracks, and I will commit to
those folders, which are now my stems. The
drums are LCR and LFE, ie. low-frequency
extension. Pro Tools does not have a 3.1
format, which is a pain because you need to
have an LCR plus a mono LFE.”
Mixing The Soundtrack
After Boucher has delivered his film stems,
he creates the stereo and Atmos mixes for
the soundtrack album. “Once my film stems
are approved, I can make the album mixes,
knowing that this is the shape these songs
are going to take in the film. That is when
I crash them down to stereo. But I don’t just
blend the stems. I still have access to all the
multitrack tracks. I don’t reference the movie
mixes for my stereo mix, because I don’t
have the final dub mix yet. I also don’t deal
any more with reverbs and compression,
and other effects, because the stereo mix
needs to be very similar to the film mix. This
is based on what I call the MacDougall rule,
which refers to Tom MacDougall, President
of Walt Disney Music. He wants people
to remember the film when they listen to
the soundtrack.
“So I can’t turn a song that was in the
film and that the audience got attached
to into a pop track, as these tend to be
 Stills from Encanto during the ‘We
Don’t Talk About Bruno’ scene.
over-compressed, unnatural-
sounding and Auto-Tuned. I don’t
try to make it hit any harder either,
because I am already trying to make
my mixes hit as hard as I can in the
film mix. To push them harder in the
soundtrack world would divorce
them from the picture.
“I may change minor things in
the soundtrack mix. Sometimes I do
a little bit more riding and sometimes
I add a little more compression.
Usually I try to enhance the reverb
that is already there, just to make it
a little more supportive. By the time
I get to the stereo mix, I have seen
the movie, and there are moments
that suggest that I might make
something louder or softer. Then I try
to make it stick together more like
a stereo mix, as opposed to the full
dynamic range of a movie theatre. In
a movie theatre you have a captive
audience and an enormous dynamic
range, but I need to make it sound
a little more contained, so it will work
in a car or on a set of headphones.
“One of the main changes I make
on the stereo mix is the stereo bus
treatment. I don’t really need much,
because I have done the amount of
processing that I want to the stems
already. The stereo bus chain adds
a little extra glue. The stereo bus
chain in Encanto has the Acustica
Sand2Bus4 [SSL-style compressor]
and the Dangerous Bax EQ. This is
very similar to the hardware stereo
bus chain in my own studio. There’s
also a FabFilter Pro-L 2, but there’s
no gain on it. It’s there for some mild
peak-limiting and to monitor the
integrated level.
“After that I do an Atmos mix for
spatial audio for the consumer, on
streaming platforms and for Dolby
Atmos playback of the soundtrack.
Sometimes I use stems, sometimes
I use multitracks, sometimes
a combination. It depends on what it
needs.”

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 IN TE R VIE W

Super Producer

Photos: Jason Martinez

138 April 2022 / w w w. s o u n d o n s o u n d . c o m


With an extraordinary track record as producer,
songwriter, musician, mixer and remixer, Mike Dean
is a one-man music industry.
PAUL TINGEN
“I
could not imagine being just
a mixer, and all you do every day is
to mix, mix, mix. I would go crazy.
Instead, I mix a little, I produce a little,
I engineer a little, I master a little, and so
on. It’s good to keep things interesting.
It keeps me from getting Alzheimer’s. It
keeps my brain active!”
While Mike Dean’s list of his activities
may already sound quite impressive
here, a cursory glance at his track record
makes clear that he’s in fact dramatically
understating what he is up to. Wikipedia
simply has him down as a “record
producer” to distinguish him from Mike
Dean, the Premier League referee, while
others have called him “hip-hop’s biggest
stoner”. Some would argue that being
constantly stoned makes it harder to keep
one’s brain lucid and well-balanced, but
in Mike Dean’s case, the only thing that’s
crazy about him is the spread and scale of
his activities.
In addition to the above, he’s also
an artist, songwriter, guitarist, keyboard
player, composer, remixer, plug-in
designer and more. Dean’s credit list
on Allmusic has an astonishing 1750
entries across a wealth of different
genres, including hip-hop, trap, pop,
computer music, heavy metal, prog
rock, film scoring, symphonic rock and
so on. In short, Mike Dean may be the
single most multi-faceted and busy
man in the American music industry. He
manages all this because, he admits, “I’m
a workaholic, a little bit, yes. But actually,
I work really smart. For example, I mix
quickly. I may later go back a couple of
times for 30 minutes each, but that’s it.
And I don’t use templates. I start with
the session as it comes. You don’t want
to reinvent the song. As a producer
I understand that.”
West Worlds
When asked what he considers the
highlights of his long and varied career,
Dean selects just two projects out of
thousands. “I particularly like what we did
on Scarface’s The Untouchable album
[1992], and on Kanye’s My Beautiful
Dark Twisted Fantasy [2010]. We were in
Electric Lady Studios for a few months
for the latter album, working around
the clock. Those were some of the best
sessions I can recall.”
Mike Dean had his hands all over
both albums, with his credits on Dark
Twisted Fantasy consisting of bass, cello
arrangement, composer, engineer, guitar,
keyboards, mixing, piano, producer
and soloist. The album was enormously
influential, winning a Best Rap Album
Grammy Award, coming in at number
one in dozens of Album of the Year and
Decade polls, and going platinum in the
UK, the US, and several other places.
Dean is best-known for his long
association with West (now Ye), which
began in 2002, when, recalls the protean
producer, “Kanye first came to my house
in Houston, where I had my studio, to do
his first mixtape. In those days everybody
flew there to work with me.” The mixtape
was called Get Well Soon, and Dean has
played a part in the making of almost all
West albums to date, including 2003’s
The College Dropout and Graduation,
Late Registration (2005), Watch The
Throne (with Jay-Z, 2011), Yeezus (2013),
The Life Of Pablo (2016), ye (2018), Jesus
Is King (2019) and Donda (2021). In many
cases Dean had a very central role, and
he earned all his five Grammy Awards
through his work with West, including Best
Rap Song for ‘Ni**a In Paris’ (2011).
Dean comments: “I’d say that Twisted
Fantasy and Pablo are my favourites of the
albums I did with Kanye. However, I can’t
discuss how I work with Kanye. His process
is his story to tell.”
The fact that West does not allow his
collaborators to talk about their work with
them is a loss to the world, as there surely
are endless fascinating stories to disclose.
At the time of the interview, West and
Dean were still hard at work on Donda 2,
but all the latter will say is, “We’re back
at full speed! Firing and hiring engineers
every day!”
Out Of Nowhere
The story of how Mike Dean went from an
unknown Texan to someone who helped
reinvent hip-hop, trap and pop music, and
who now is on his way to becoming the
next synth god (see box) starts in 1965,
when he was born, in his own words, “in
the middle of nowhere”. It was, in fact,
not far from Houston, Texas. He tried his
hand at saxophone, piano and bassoon at
a young age, became a serious classical
piano player and was part of the school
orchestra. In his teens he graduated to
playing hard and prog rock in school
bands, and also soul, funk, blues and
country in cover bands.
Still a teenager, Dean was invited to
join Parliament-Funkadelic when their
keyboard player Bernie Worrell left to join
Talking Heads, but instead he choose to
play for local group Selena y los Dinos,
because, he recalls, “It made more
money!” He became the group’s musical
director, and also acted as a producer for
singer Selena while she was still a young
girl, before quitting in the mid-1980s.
Dean continued working in piano bars,
and in the late 1980s was a member of
a local group called Def Squad, in which
he tried samplers and drums machines,
and started making hip-hop. He connected
with the new and local label Rap-A-Lot, and
ended up being mentored by the label’s
producer John Bido. Through the label,
and using his ever-growing skillset, Dean
ended up becoming one of the key people
in the development of Texas rap, helping
develop what’s called Southern Rap, aka
the Dirty South sound, and working with
artists like Scarface, Geto Boys, Do Or Die,
w w w. s o u n d o n s o u n d . c o m / April 2022 139
 IN TE R VIE W
MIKE DEAN
Mike Dean’s B room
is kitted out with an
impressive Dolby
Atmos mixing setup.
5th Ward Boyz, UGK, Tha Dogg Pound
and many more. Two albums by Scarface
that he co-produced, The Diary (1994) and
The Untouchable (1997), went platinum.
Around the turn of the century, Dean
moved to New York, where he opened
a studio called Dean’s House Of Hits.
This, he says, “was in my apartment,
and kind of crazy. But all my studios
have always been in my house, where
people come to work with me.” In
2002, while both working on Scarface’s
album The Fix, Dean and Kanye West
met and it ignited their long-standing
collaboration. Dean has since opened
a studio in Los Angeles, and has worked
with an astonishingly wide variety of
artists, including Travis Scott, Kid Cudi,
Beyoncé, Jay-Z, Madonna, the Weeknd,
Frank Ocean, Travis Scott, 2 Chainz,
Nas, Christina Aguilera, Selena Gomez,
Gorillaz, Pop Smoke, Burna Boy, Lana Del
Rey, FKA Twigs and many more.
Dean operates under the motto,
“Producers should never get stuck in time.
They should always be time travellers
living in the future.” He also often uses
his DJ name MWA, an abbreviation for
Mexican Wrestling Association, and he
started a label under this name.
Drop Ins
Today, Mike Dean has a studio in Houston
and two in Los Angeles, where artists
140 April 2022 / w w w. s o u n d o n s o u n d . c o m
apparently like to visit. “Madonna drops by
every couple of days. She is always fun.
I just mastered a compilation of 51 of her
number-one dance hits. We’re also working
on some remixes of ‘Frozen’ and we do
remixes of remixes. And we’re writing.”
Spreading his wings still further to
collaborate with mostly female pop
singers, Dean previously worked with
Madonna on her albums Rebel Heart
(2015) and Madame X, and with Lana Del
Rey on the song ‘Wildflower Wildfire’ from
her most recent album Blue Banisters
(2021), which he co-wrote, co-produced,
mixed, mastered and appeared on.
“Lana lives close to my studio, and also
drops by regularly. The session we did for
her album was interesting. Lana came over
for a day or two, with a rough idea of what
she wanted to do, and we worked on the
song, on the chords, and got it all together.
We worked in a room upstairs where
I have a piano, and she sang through the
karaoke machine.
“We then recorded her at the same
time with my Sony C800 and Shure SM7B,
with no windscreen. The latter sounded
better on her than the Sony did. In fact,
Lana’s whole history has been about using
the SM7B. My mic pres here are mostly
the Neve 1073, either vintage or BAE, and
I have the Neve Shelford Channel in my
B room. My compressor on the way in is
the Tube-Tech CL-1B. That’s it. I keep it
pretty simple. Everything else is synths
and guitars.”
No Tuning
After recording Del Rey, Dean did vocal
production, which, he says, was minimal.
“She really did not need much vocal
production. She sang the song through
two or three times, no punch-ins. I think
we comped two or three lines. There
also was some very minimal tuning with
Melodyne. And that was it. In a situation
like this, where you just do some minor
pitch-corrections, you want Melodyne.”
Suddenly animated, Dean elaborates:
“Have you heard of Christine And The
Queens? We worked on her new album
here. She does not use any Auto-Tune or
Melodyne. That’s really nice. There is stuff
out of tune here and there, but it is cool.
Marvin Gaye was always 15-percent flat.
Imagine if he was in tune! He’d sound like
a robot! Tuning can make singers sound
very sterile. Sometimes I have two or
three rappers on one song, and I can’t tell
the difference!
“I do a lot of vocal production myself.
It’s all part of the job as a producer. But
vocal production has become a separate
discipline now, and this means that many
producers are just beatmakers at this
point. They have to hire a vocal producer.
But I’ve been doing vocal production
since back in the old days. If a note was
 Becoming God
At the age of 56, Mike Dean is aiming for a career
change. He wants to become, in his words,
“a synth god”, similar to Jean-Michel Jarre or
Rick Wakeman, playing solo concerts to huge
audiences the world over. Dean famously is the
owner of 50-something hardware synthesizers,
and, as evidenced by YouTube videos, likes
to play several at the same time, with a joint
between his lips, smoking so furiously that he
creates his own theatrical smoke effect. The
music is smoking as well. His first major releases
in his ‘synth god’ persona were his studio albums
4:20 (2020) and 4:22 (2021) and his live album
Echoplex (2021).
The first two of these, recorded at his Los
Angeles studio, were in part the result of the
pandemic, as they contain compilations of some
of his Instagram live-streams, with Dean mostly
improvising. Dean calls the music “a psychedelic
symphony” and it’s best listened to in a very
relaxed, or, one could say, very stoned state
of mind.
“Yes! Becoming a synth god has always been
there,” enthuses Dean. “It’s something that has
been building over the years. I’m also a guitarist,
and I play guitar and synths live with Kanye and
Travis Scott. Lots of my records have guitar, with
solos and crazy tones. I play lead and rhythm.
When I did a show with Kanye last month, the
guitar was screaming through the whole thing.
But with every album, I turn the synths up a little
bit more, and now with my solo albums especially
there are tons of synths, even as I also play some
guitar. My ambition is to be a solo artist more than
a producer, even as I’ll continue to do production
as well.”
Dean now has too many classic keyboards to
fit in his studio at once. What’s the attraction of
having that many synths, given that they are heavy
to lug around, take up space, can go out of tune
and malfunction in other ways, may need MIDI
fitted, and in general require a lot of maintenance?
(Dean’s keyboards are maintained by Rosen
Sound, “LA’s pro shop for analogue synth repair,
restoration.”)
“I have about 50 keyboards. All of them have
been upgraded to work with MIDI. They give me
real sounds. They sound far better than soft synths.
I just spent $15,000 on an ARP 2600, and it sounds
so good, it paid for itself in a week. It has nice,
warm distortion and wonderful oscillators. I have
the Behringer clone and it sounds terrible, and the
Arturia version, and while it sounds good, it doesn’t
sound like the real thing.
“I also just spent $50,000 on a Yamaha CS80,
and it sounds great, and nothing plays like it.
Nothing has the same polyphonic aftertouch. I also
bought a Roland Jupiter 8 this year, for $20,000.
People talk shit about these instruments. When
you have one, you can hear the difference with
the imitations. The fact that the real synths are
sometimes out of tune is what makes them so cool.
It’s funky. It gives them character! I also use soft
synths a lot, I’m not against them. But soft synths
are like singers with Auto-Tune. They sound very
sterile.”
Dean is particularly a fan of Moog synths, and
explains: “The Moog synths I use the most are the
Matriarch, the Grandmother, and the Voyager for
my signature bass sound. I have been to the Moog
factory a few times, looking at prototypes and trying
things out. The Voyager, Juno 106, Korg Triton and
two laptops are my core setup live. One laptop is for
playback and the other for sound sources, like my
guitar sounds and stuff like that. But when I work
in my studio in the box, I use mostly samplers, like
Native Instruments Kontakt, and I also use a lot of
Arturia stuff, for organs and so on.”

out of tune, we’d use an Eventide H3000
to pitch it down a few cents and punch it
back in. It was manual vocal tuning, and it
was fun. You had to really use your ears!”
There’s a tone of disdain when Dean
talks about modern vocal tuning, which
is a little surprising as he was at the heart
of some of the Auto-Tuned recordings
that had a major impact on the industry.
“I used what we call ‘the Heartbreak
sound’, which is Auto-Tune and distortion
[and a reference to West’s 808s &
Heartbreak album], for the first time when
John Legend came in to do backing
vocals on ‘Jesus Walks’ [a track on West’s
The College Dropout album]. John was
really pitchy, and we put Auto-Tune on
him, set to zero, which sounded really
cool and turned it into a talkbox kind
of thing. That sound was a big step for
everybody. It became a strong influence on
the way singers use Auto-Tune today, and
also on the kind of chord structures that
everybody uses.”
From A To B
Dean’s misgivings about this aspect of
modern technology, as well as his obvious
love of analogue keyboards, suggest that
he may be a bit of an analogue purist,

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 IN TE R VIE W
MIKE DEAN

but in fact he’s clear that he prefers

the minimalist modern approach to the
hardware studios of old. “I like it more
simple. I’m using the two new SSL UF8
controllers, in both my rooms, and I like
them a lot. But that’s it. Apart from the
recording gear that I mentioned earlier,
my hardware consists of the Neve 5060
Centerpiece 24x2 desktop mixer, SPL
MixDream summing mixer, and I have two
units of the McDSP APB16, and four units
of the UAD Satellite [two Thunderbolt
3 and two USB Quad Core], and the
Antelope 10MX Rubidium Atomic Clock.
“I have two Mac Pros with 384GB of
RAM and 8TB hard drives, running Pro
Tools, Ableton, Logic, FL Studio and
sometimes Reason. I use Pro Tools for
recording vocals, mixing and mastering.
I don’t do any creation in Pro Tools,
because it has the most clunky MIDI
environment in the world. I don’t see how
people can create in Pro Tools. I tend to
do drums in Ableton, or in FL Studio if you
want the particular sound of that DAW.
“Why do I use Pro Tools if I think it’s so
clunky? Well, the automation in it works
a lot like that of an SSL desk, which I am
used to. I also like the fact that the HDX
stuff takes a lot of load off the computer,
like plug-ins and the playback engine.
But I don’t really use any Avid interfaces.
I have one Antelope Goliath in Studio B,
and two in my main studio, so I have 64
channels going out to my summing, and 32
channels of keyboards coming in.
“My main room and my B room have
identical gear, so we can easily move from
one to the other. The main difference is that
my A room has my keyboard, and different
monitors, because my B room has an Atmos
set up. I have Tannoy 215 monitors and A synth god in his synth Valhalla. Mike Dean’s
Yamaha NS10s in my main room, and my collection of modern and vintage synths is so huge
B room has all Amphions, with Two18’s at that they won’t all fit in his studio at once, but
the front, One18s at the back, and 418 subs. among the classics visible in this shot are multiple
I use my main studio to make beats and Moog instruments including a Memorymoog,
mix, and then I can take that to the B studio, Minimoog and Voyager, a Yamaha CS80, ARP 2600
and Roland Jupiter 8.
and when I load it up the mix comes up the
same. I then make the Atmos mixes in there
from the stems. My monitor controller is the sound like the NS10s with a couple of into a studio for a few days with Kanye
DAD MOM Monitor Operating Module. extra octaves at the low end.” recently. I just go in and catch the vibe. I’ll
“The Neve 5060s are at the core watch him track vocals, and then I grab
of my setup. They give you that Neve
Home Comforts the files and bring them home, and I do all
sound. They sound very similar to a Neve During his entire career, Dean has preferred the additional production and mixing and
8080 desk. You can push things through to work in his own studios, but this is now mastering here.
a half-million-dollar 8080 and then more the case than ever. “Mixing and “When I work on vocals, I use plug-ins
through the 5060, and the 5060 wins mastering in other studios is fucked,” he like the Avid D-Verb and Waves H-Delay.
every time! The NS10s are also crucial to explains. “Nobody’s rooms are tuned right I particularly like the D-Verb. It makes stuff
my workflow. I have eight sets of them, any more. They are not maintained like they sound like it’s in a room, what else do you
so I can replace woofers and tweeters. used to be. This started in the 2000s. But want? The Valhalla reverbs also are go-tos.
Plus I have six sets of Tannoy 215’s. They I still occasionally go to other studios. I went They are nice, and simple to use. I like

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

Slate stuff, I like UAD, I like SoundToys,
Plugin Alliance, Waves, and more. I have
every plug-in that you can buy, just so I can
load everything that people bring me, and
if I don’t have a plug-in, I will buy it. But
I don’t like stuff where I don’t know what it
is doing. And in general I like regular, stock
plug-ins. I don’t use a lot of weird stuff.
“I push things through the Neve board
when I’m mixing, which makes it sound
really good, and I keep everything digital
from there on. As I mentioned, I work very
quickly, and when I mix, I stay very true to
the demo. Producers sometimes work on
demos for years. For a mixer to take off
the effects is just a waste of time. It is like
buying a new car, and saying: OK, I am
going to take the transmission out and then
put it back. Especially with the rap stuff,
everybody has very specific vocal effects,
which are part of their sound, and you want
to respect that. And yes, I have specific
plug-ins for mastering, but I don’t really
want to get into them!”
Having arrived perhaps too close
for comfort to discussing some of
his trade secrets, Dean decides to
sign off, but not before mentioning
his a new solo album, which will be
released in April this year, just like
4:20 and 4:22 in the previous two
years. Tantalisingly, more news was
not forthcoming, and fans will have to
practice patience. The bong-shaking
synth god clearly likes to keep his
fans’ brains active as well…
w w w. s o u n d o n s o u n d . c o m / April 2022 143
 IN TE R VIE W
Mixing Atmos:
Andy Bradfield
Leading mix engineer Andy Bradfield shares some of the
lessons he’s learned in getting to grips with Dolby Atmos.
SAM INGLIS
D
uring more than 30 years in the
business, Andy Bradfield has
established himself as one of the
most versatile mix engineers around. He
trained at Olympic Studios and became
an in-house engineer at Olympic and
Townhouse Studios before going freelance.
He now works mainly from his own studio,
which was recently upgraded to Dolby
Home Entertainment specifications, and
is seeing increasing demand for Atmos
144 April 2022 / w w w. s o u n d o n s o u n d . c o m
mixes. Andy’s earlier experience mixing in
other surround formats has stood him in
good stead here.
“I’ve seen a lot of change over my
career, because I started on analogue
tape and SSLs. Then gradually Pro Tools
started coming in, and I embraced it quite
early on, but it was always in conjunction
with other things. And eventually, that
morphed into what we have now. But also,
over that same period of time, I started
dipping my toe into movie soundtrack
work. So, I was working on score mixing as
well as traditional stereo work, and that’s
continued ever since. So for me to leap into
Atmos was perhaps less of a jump than for
some other people, because I was already
quite comfortable working in 5.1 or 7.1.”
On The Level
For a studio like Andy’s to be certified
by Dolby, it has to meet fairly stringent
technical requirements. Obviously, it needs
a surround monitoring setup that includes
ceiling-mounted speakers. Less obviously,
the output from that setup also needs

to be regulated. “In the post-production
world, all the spaces are calibrated. Dolby
go in regularly and check the setup, and
they work at a fixed, calibrated level. For
someone who’s a straight-up music guy
like me, that put my head on backwards,
because I didn’t fully understand the
concept initially. Richard and Myles at
Dolby were really great, and enormously
helpful in getting me set up and
configured. Avid were super helpful too.
“I started working at a calibrated level
a good few years ago now, when I was
doing score mixes, and actually, I found
that really helped in terms of getting
things to translate to the dub stage, and
also integrating with the dialogue. But
the thing that I found quite early on was
that I struggled to work at 85 [dB SPL],
because I’m in a smaller space than most
post-production studios. Working at 85
when the speakers are 20 or 30 feet away
is OK. But when they’re as close as they
are in a regular music studio, that’s much
harder, and if you’re not careful, without
realising, you calibrate yourself. So, what
happens is, you mix quieter.”
On the basis of this experience, Andy
has settled on a calibraton level of 79dB
SPL in his studio; and now that he’s
acclimatised, he finds that his instincts as to
when things are too quiet or too loud are
usually sound. This is crucial when mixing
for Atmos, because the specification insists
that mixes should not exceed -18 LUFS, and
any that do are liable to be rejected.
“I’m sure a lot of music people will be
thinking: why are Dolby dictating all these
things? But actually, the more you read
the spec and you start to understand what
they’re trying to do, the more it makes
sense. It is a little bit of a steep learning
curve — and you’ve still got the added
complications of having the renderer.”
Render Me Speechless
The Atmos renderer is a piece of Dolby
software that integrates with the DAW
and acts both as monitor controller, virtual
audio interface and master recorder.
In essence, the renderer maps Atmos
beds and objects to the specific speaker
configuration in use, and when the time
is ripe, generates the ADM file that will
eventually be used for Blu-Ray authoring or
sent to Apple, Tidal, and other streaming
services. For Andy, most of the wrinkles
associated with moving from conventional
surround mixing to Atmos relate to learning
the ways of this Dolby utility.
“It’s actually not too difficult once you’ve
used it a bit and you’ve understood it, but
initially the renderer is a bit of a shock —
because although it’s a monitoring box,
it’s also ultimately where you output your
files to. It’s sort of half a workstation, except
it doesn’t have any waveforms. It’s a bit
strange to get your head around when
you’re so used to working with all these
different tools that have waveforms. And of
course, on top of that, you’ve also then got
the binaural rendering for headphones.
“I’m using the renderer the way that
most people do, on the same computer as
Pro Tools. You set your Pro Tools interface
to be the Dolby Audio Bridge, and that
gives you 128 paths out to the renderer,
and then they can be a mixture of what
they call beds and objects. Beds are up
to 7.1.2, so if you’ve got, say, a bunch of
stuff that’s already mixed in surround, and
you’re not really bothered about getting
that much height coverage, you can just
leave it in a bed, and it will pan to all the
correct places in the regular surround
plane. Objects can either be mono or
stereo, although you can’t send those to
the low-frequency channel.
“The Dolby Atmos panner is very
clever, because it’s kind of like auto-pan
on steroids. The idea is that you put it
on an object channel, and it creates the
panning metadata instead of the Pro
Tools panners. And you can do all sorts
of crazy sequencing things which would
w w w. s o u n d o n s o u n d . c o m / April 2022 145
 IN TE R VIE W
MIXING ATMOS : A N DY BR A DF IE LD
be quite difficult to achieve with the Pro
Tools panner.”
Space Balls
In total, an Atmos mix can contain up to
128 channels. If 10 of these are used for
a single 7.1.2 bed, that leaves 118 available
for objects. “You may think 128’s quite
a lot, but if you started getting really crazy,
it can get out of hand quite fast. But I’ve
ended up only really needing one bed,
which is usually a 7.1.2. I tend to start by
just putting everything into the bed. Where
it becomes a little bit limiting is that the
height on a 7.1.2 bed is only two channels.
So, generally, if it’s something that I want
to add height to, I’ll make it into an
object, but it’s also good for more precise
movement through the 360-degree space
that Atmos gives you to mix in.
“One of the other things that’s a bit
annoying is that the numbering drives
you a bit crazy. The objects are numbered
from 11 upwards, yet you have to have
at least one bed channel, and that uses
inputs 1-10. So, your first stereo object
pair is 11 and 12. What you quickly learn
is some little tricks, like you suddenly go,
where’s that? So then you solo it, and if
you look up on the renderer, it’s got loads
of little circles, and if it’s coming in, they
go green. They’re like little sort of input
LEDs. So you go, ah, yeah, that’s on four,
or 24. All right, cool. And then carry on.
Once you route to an object you can then
basically pan that anywhere in the 3D
space, and it shows up on the renderer as
a ‘ball’ when it’s got signal — or pair if it’s
a stereo object — so you get a dynamic
view of the object panning or movement
while you are mixing.
“One of the things that Atmos is
intended to overcome is translating to
different systems, so if you play on a 5.1.2
system or a 5.1 system, the hardware
decoding the file will deal with it and
play it in a meaningful way, which is
almost impossible on traditional surround
formats. In Atmos the objects get rendered
depending on the playback configuration.
“I wish Apple would make us a real-time
tool to monitor their headphone playback
[the Apple Music Headphone Decoding],
because their headphone rendering is
slightly different than Dolby’s, and Apple’s
position is like, well, you just render in
QuickTime and put it on your phone.
And I’m like, yeah, but mixing is real time.
It’s always been real time. Whereas the
Apple way is that I’ve got to print into the
renderer or bounce it to an ADM file, then
146 April 2022 / w w w. s o u n d o n s o u n d . c o m
Budget For Success
For the time being, at least, mixers like Andy
Bradfield are still delivering separate stereo and
Atmos mixes. Andy is optimistic about the future
for the new format, but his concern is that unless
clients understand and budget for the additional
work involved, quality will suffer. If labels
choose to treat the Atmos mix as just another
‘deliverable’, then either the Atmos mix will be an
afterthought, or the stereo mix a hasty fold-down
from the surround mix.
“What seems to be happening in music
currently is that people are mixing in stereo,
then rendering stereo stems, and making an
Atmos mix from that. This is good as it keeps the
sonic vibe of the stereo mix which can then be
expanded into a more immersive Atmos mix. But
it’s a time-consuming process. Content creators
have to have a budget to do this properly, as you
export it to a QuickTime MP4, I’ve got
to get it on my phone, and then if I want
to make a change, I’ve got to do that all
again. It’s just really too time-consuming.
If Apple want to push this format, they
need to give us the tools to do it, even if
it was a piece of software that had to be
run on another machine. Apple playback
can still render a full-speaker Atmos via
an Apple TV 4K and an Atmos receiver or
soundbar as well.”
Bottoms Up
Not only does mixing in Atmos
mean learning the ways of the Dolby
renderer and adjusting to a calibrated,
loudness-normalised mixing environment;
it also involves unlearning some familiar
techniques. “A lot of people, myself
included, for many years have used
what you would call a top-down mixing
approach. That could be as simple as
having a stereo EQ on your mix bus, or
it could mean having quite elaborate
compression and tone-shaping tools. And,
of course, that affects how everything
sounds. In this world, it’s much more
difficult to do that. You can’t really heavily
compress the stereo bus, because you’re
dealing with discrete sources a lot of
the time, as the objects are in effect not
a traditional bus!
“Being someone who’s made records
for a long time, I’m not adverse to using
limiters, and I think used well, they’re
a very good tool. But sometimes on
stuff that’s perhaps a little bit more
extreme-sounding, it becomes part of
the sound. Certainly on a lot of pop
records, there’s a lot of detail, and part
of that smacky, middly driving vibe is the
are mixing in a totally different format, and often
a separate additional mix from the stereo one. If
they don’t then I think it will be tempting to just
do a quick headphone-only thing and it won’t
really work that well. You need to hear the mix in
a proper speaker setup and a headphone setup
to do it justice.
“Overall Atmos is a really exciting new format,
and the creative control it gives you to develop
an immersive mix that you’re part of is amazing!
I think surround was always seen as a ‘film’
format and Atmos has really changed this now.
Pop music and rock bands are really exciting in
Atmos! As the format develops and becomes
more widely adopted, especially in things like
cars — there are some already coming to market
— it could become the format. Who knows? The
future is bright, the future is Atmos!”
processing and the limiting. And when
you come to doing Atmos, it’s much more
difficult to achieve that effect, because
that’s something that’s put on as an overall
thing, and it doesn’t work in quite the
same way when some of it is separated.
“Something else that I’m aware of
when mixing in surround is that I always
want to try and make it sound like it’s
coherent, and of course, to some degree,
when you’re trying to make it immersive,
you’re also splitting it apart. It’s like having
a ball of dough, and then just pulling
it out into loads of bits. It doesn’t look
like a big coherent piece any more, you
know? When you’re working in Atmos,
you’ve got lots more places to put things,
but also then, by definition, if you do
that, sometimes you’re exposing things.
And that can be good and bad, because
sometimes things have artefacts, or it’s
a chopped-up sample, suddenly you can
hear it. When it was in a stereo mix, there
may be some masking going on of where
it sits, so you don’t hear some of those
things. So, some things like that have to
be a consideration. But again, I think, once
you’ve got an Atmos setup, you’ve got
a good monitoring environment that’s very
revealing, then you have to make those
calls as and when they come up.
“The other thing that I’ve been very
aware of is that stuff that’s been mixed in
stereo, if it’s quite rhythm-heavy, it’s quite
difficult sometimes to pull it too far into
the room, because what tends to happen
is the energy drops, and it doesn’t have
that drive. And so, you kind of almost
have to pin it to the front. If you’ve got
something that’s big and orchestral and
very lush, you could probably spread it

right out, and it would sound amazing.
Whereas if it’s something much more sort
of tight-knit, like a drum & bass thing or
a band, you sometimes want it to be much
more focused.”
Stuck In The Middle
Lead vocals are usually the most
important element of any mix, and in
stereo, we nearly always position them
in the centre. It seems natural to do the
same in Atmos, but as Andy points out,
that idea can be realised in multiple ways.
A vocal routed only to the centre channel
is ‘centred’ in a very different way from
a vocal routed to both the left and right
speakers, or to all channels equally. “In
the cinematic world, they tend to want
the vocal in the centre channel, as that’s
where the dialogue is too. That’s partly
to do with having massive theatres, and
the fact that if you are over on the far
right or the far left, you still get the vocal
because it’s in the middle, because it’s
got its own channel. And I get that, but
I know that some people find that a bit
uncomfortable, because they’re used
to listening to phantom stereo, which is
what we’re all doing when we’re mixing in
stereo. So, some people tend to still use
left and right and put a bit of the vocal
in the middle, or you could even do the
funny sort of triangle thing where you
could put the vocal in the middle and the
effects behind you, if you wanted to. But
again, it depends on the track, because
sometimes you do that, and then the vocal
ends up turning dry, and it doesn’t really
sound right.”
The key, for Andy, is to make these
decisions in context. He cites as an
example a ballad from Fish’s album
Weltschmerz. “It started very small, and it
was just piano and vocal, and I was like:
do you know what, this would be quite
cool if the vocal was kind of centre, but
the effect was kind of around. So I actually
used a surround reverb, because there’s
a surround version of the Liquidsonics
plate, and it’s fantastic. And it was
absolutely the right thing, because there
was enough space to have it. On a much
busier track, it’s possible that that would be
lost on you, because you wouldn’t really
hear those back/side channels, anyway,
certainly not a reverb return back there on
a very dense mix. So, it depends on the
arrangement and what’s going on.”
Head Space
“Mixing has always been a technical
exercise, but it’s also a creative, musical
exercise,” concludes Andy. “And at times,
those things are slightly in conflict with one
Andy Bradfield’s studio has recently undergone
a revamp to meet Atmos Home Entertainment
specs. The speakers in the surround array are
Neumann KH120s, with the Kii 3 and good old
NS10s still present for stereo mixing. Note the Dolby
renderer on its own screen, top left.
another. One of the things with Atmos is
that initially, it’s a bit more of a technical
exercise than it is a creative exercise. Once
you get past that, it’s great, but I’m always
very aware that I’m trying to not let the
technology dictate to me, so that I can be
creative. And so, in a way, I almost have to
split my brain into two parts. There’s the
creative part that’s Creative Andy. ‘Let’s
do a mix, let’s make it really amazing.’ And
then, there’s Technical Andy that has, ‘Oh,
we’ve got to print this now, and we’ve
got to output it, and we’ve got to make
a QuickTime so I can send it to the client.’
And those two things are very different
parts of the process. But unfortunately,
they’re necessary parts of the process.
“I’m certainly very excited about what
Atmos has to offer, but it’s a big rabbit
hole! You start going down it, and you go
‘Oh my god, what have I just started?’ And
I think part of it is fear of the unknown.
In terms of the technical aspects and the
equipment, there’s a lot to it. But it’s not
insurmountable, and once you get going
it’s great, and I’m really enjoying it.”
W www.andybradfield.com
w w w. s o u n d o n s o u n d . c o m / April 2022 147
 IN TE R VIE W

148 April 2022 / www.soundonsound.com

 Hildur Guðnadóttir
& Sam Slater
Composing With Field Recordings
HBO’s Chernobyl exposed the horrifying story behind
the 1986 nuclear disaster — and made a star of
composer Hildur Guðnadóttir.
TONY BACON
“W
e were in a room so
vast that when you
made a sound in there,
you wouldn’t hear reverb,” says Sam
Slater. “The sound just dissipated.”
He’s talking about the Ignalina nuclear
power plant in Lithuania, where he
and Hildur Guðnadóttir went to record
sounds and atmospheres for their
Grammy-winning score to the 2019 HBO/
Sky mini-series Chernobyl.
The successful duo likewise won
Grammys for the score to the 2019 movie
Joker, which also bagged Hildur an Oscar,
a Golden Globe and a BAFTA. Hildur was
born and grew up in Iceland, where she
learned cello from an early age. She then
studied and worked in Berlin, establishing
a space in a studio built by a crew of
experimental/scoring composers in the
city’s Kreuzberg district. Sam is British,
and studied Experimental Composition at
Leeds University before moving to Berlin
in 2013 to work with Jóhann Jóhannsson
at the Kreuzberg studio. Appropriately,
that’s where the two met. After all, they
agree, studios are where most of the
good collaborations start.
The Collaborative Urge
Hildur reckons there’s much to be
gained from a close dialogue with your
collaborator. “Specially big projects
like the scores for films or TV series
Photo: Camille Blake
or computer games that we’ve been
working on,” she says. “There’s so many
different threads and angles and roads
that have to lead in the same direction.
There are so many places you have to
look in the corners of. By their nature,
these types of productions are huge
collaborations between so many different
departments. The sound and music is just
one small part, because you also have
the visuals, the costumes, the script, the
cinematography. So to work in these
kinds of projects, you really don’t have
much choice — you just have to be able
to collaborate.”
“We’re married, so we collaborate
on life as a whole,” explains Sam. “But
that wasn’t the case when we started
collaborating together musically, and as
we get better at that, I think sensitivity is
the key word. It’s about understanding
what the benefits of a collaboration can
be — and whether it’s even needed.”
By way of example, he mentions some
projects that Hildur’s working on currently.
“And while I always love to be supportive
if I can, in the current shape that they’re in,
I’d just get in the way. In those cases it’s
important to be sensitive to that and not
let your ego dictate what to do. You have
to know when to step back and recognise
that the way to collaborate is actually not
to participate this time.”
Their working collaboration began
as a sort of composer-producer
relationship. “That was more to do with
Hildur having all of these wonderful
melodies and harmonies in her head,
and me having a bunch of ideas to do
with sound design, and how conceptual
ideas for sound design can be used. We
found some shared ground, specifically
with Chernobyl, and then recently it’s
been more of a composer-composer
relationship. But, you know, it’s a recipe
for disaster if you insist on collaborating.”
In their collaborations, Sam handles the
production and Hildur, as she puts it, has
the space to sit and use her ears. They
balance each other out, she explains.
Not that Sam has neglected his own
compositional work, which includes the
score for the 2022 movie Guerrilla and his
2018 solo album Wrong Airport Ghost.
Hidden Sounds
For Chernobyl, says Hildur, “We had really
strong ideas when we started the project
that we wanted to go to the Ignalina
power plant and capture its sound.
We intended to use the building as an
instrument.” Built along similar lines to the
infamous Ukrainian plant where disaster
struck in 1986, Ignalina is currently in
the midst of a lengthy decommissioning
process that began following permanent
shutdowns of its reactors in 2004 and
2009. “We worked at Ignalina with Chris
Watson, the field-recording master,” Hildur
recalls, “and we had a fantastic time with
Chris. He’s a deep listener, which you
don’t encounter that often. It’s a huge
pleasure to go and listen to anything with
him, really, let alone to record.”
She says they tried to capture the
sound of the spaces at Ignalina without
“activating” them too much. One
particular star was an enormous door that
provided some magical sonic gifts. “It
had a whole symphony of sounds coming
from it,” she says with a smile. “It had all
these harmonics and things that were
resonating wonderfully.”
Hildur, Sam and Chris had just six hours
to complete their mission at the plant,
much of which was taken up completing
w w w. s o u n d o n s o u n d . c o m / April 2022 149
 IN TE R VIE W
Steinberg’s Cubase is the main DAW
used in the compositional process.
security clearances and putting on
multi-layered protective clothing. So
the trio made sure they had open ears
throughout their time there, as Chris
deployed his double-Mid-Sides Schoeps
array and a selection of contact mics. “So,
there we were in the reactor room, and
I’d never been in a space so large,” Sam
says. “You don’t really have much of an
opportunity to influence that kind of space
interestingly. You need to be receptive to
what’s going on in the space.”
They moved next into Ignalina’s vast
turbine hall. “It made the reactor hall
seem tiny in comparison,” Sam continues.
“It was so long and so large that there’s
a crane to move things from one end to
the other. And all the while there were
several thousand decommissioning staff
in this factory, so there was so much for
us to explore. There were tones, there
were noise sounds, there were little
melodies hidden in overtones, there
were percussive ideas, there were
things that acted as impulse responses.
There was so much sound available if
you got yourselves out of the way and
concentrated on finding the right things.
Really, we were sound hunters.”
No Dramas
Once they got back home to their base
in Berlin, they began to sort methodically
through the riches they’d captured.
150 April 2022 / w w w. s o u n d o n s o u n d . c o m
The Right Tool For The Job
Hildur Guðnadóttir and Sam Slater use different
DAW programs depending on what they’re
trying to achieve, and where they are at in the
production process. “If it’s more sound-design
based, we work in Ableton, and live as well,” Sam
reckons. “We do a Chernobyl live show, which
is a 10-channel immersive concert — pretty fun,
very loud! That all runs off Ableton Live.”
But if the work involves more traditional
scoring, picture, timecode and MIDI, they’re
both pretty much into Cubase. Hildur explains:
“I find that when you have to work with sample
libraries, for example, when you’re working on
a timeline, to picture, I find it most comfortable
to work on a fluid timeline, and for that I find
Cubase more intuitive, somehow, than Pro Tools,
for example.”
“Sam and I went gold digging through
the material,” Hildur reports. “We
microscopically listened to all of these
recordings, many times over, and dug out
what we could mould into musical clay.”
It took a good deal of patience and
time to sit with many of the sounds in
order to understand how they might use
them in a musical context. “We began to
realise we had enough material to build
the whole score on,” she says, “and that
was really exciting. I’d be sitting there
with headphones and I’d be thinking ‘OK,
now we’re at the fifth hour, 46 minutes,
20 seconds in, and right! There it is!’ So
we’d dig that little sound nugget out and
manipulate it — and very often it would be
so high that we needed to pitch it down
to an audible range, stretch it out, and
mould it.”
Photo: Camille Blake
Sam too finds the Cubase experience better
suited to free-flowing composition. “Whereas
Pro Tools, we’ll see an error message, and it’s
aghhhh! You’ll be sidetracked trying to sort all
that out — and the idea’s gone. ‘Would you
like to save a detailed report?’ No! I would like
to make some music! ‘Sorry, this is a mono
MIDI channel, and you’re going to stereo.’ Just
go away!
“But all of those are the reasons why Pro
Tools is perfect for doing the end recording,
because it’s precise. When things are actually
being recorded in a proper studio, everything is
moved to Pro Tools. So all projects end up in Pro
Tools at the end — and that’s the sign that you’re
about to finish, because you can’t go backwards
once you’re in Pro Tools.”
In this way they built up a huge bank of
sounds from the material and started to use
them to compose the music. “It took a lot of
time and patience to put together,” Hildur
adds. “The story of Chernobyl needed
this kind of approach, this much attention,
so that we could marry the real events
that have influenced so many people that
are still alive today and who remember
it happening, or have lost loved ones,
or were somehow affected. It was really
important that this reality was engraved in
the music, that the music wasn’t just there
to dramatise the events, with a conventional
string orchestra or something. We really
felt it had to be factual music more than
fictional music, so that you can really feel
the radiation in the music.”
A TV production like Chernobyl is a big,
complex affair. Hildur says an important
 CUTTING
VIDEO DOCUMENTARY ORIGINAL S
IN ASSOCIATION WITH
VINYL
AT ABBEY ROAD
We visit the world’s most famous studio to discover how the ones and zeros in a digital recording
get turned into grooves on a vinyl record. Masters of mastering Miles Showell and Geoff Pesche fire
up their vintage Neumann lathes and demonstrate the delicate process of cutting a lacquer disc. It’s
a skilled business that requires man and machine to work together in perfect harmony — and the
slightest mistake will make the resulting album unplayable. No pressure...

www.youtube.com/soundonsoundvideo
 IN TE R VIE W
HILDU R GU ÐN A D Ó T TIR & S A M S L AT E R
For radical speed changes, Sam Slater finds
that tape is more forgiving than digital.
consideration for the composer is to
determine the point in the process at
which you actually start working. “Often
with TV productions, the composer
comes in really, really late, and maybe
has something like three weeks to score
a whole series. Whereas Chernobyl took
a year, and I much prefer to start very
early and for the music to be a bigger
part of the DNA of the whole production
and the whole story. I just find that way of
working more interesting and artistically
fulfilling — but the downside, of course, is
that this takes way longer. And you don’t
get paid any extra!”
Their sound-gathering trip to the
Ignalina plant in Lithuania happened
right before the crew started filming
the Chernobyl series there, because of
Ignalina’s physical similarity to the Ukraine
station. So Hildur and Sam felt they were
working more or less in parallel with the
whole production.
“While they were getting the visuals
together and beginning the edit, we were
already starting to sketch ideas and see
what sounds might work,” Hildur says,
adding that the overall intention for the
score from all sides was to not dramatise
the events. No one wanted a traditional
drama score. No pianos and pads, Sam
says with a smile. “We wanted the events,
152 April 2022 / w w w. s o u n d o n s o u n d . c o m
as horrific as they are, to really be able to
just stand, undramatised, on their own,”
Hildur clarifies.
Nonetheless, there’s quite a distance
between a score that doesn’t dramatise
the events, and one that uses an actual
nuclear reactor in order to create
a strange sound-design-oriented score.
“Yes,” Sam says, “it was very unusual for
a production as large as Chernobyl to
have that much faith in a composer. Bear
in mind that at that point Hildur hadn’t won
the awards she’s won subsequently, and
the director, Johan Renck, was a first-time
director in the series context. He just had
so much confidence and faith in these
crazy ideas that we were coming out with,
and I think without that faith it would have
been very different.”
An Open Door
As they dug through the sounds they’d
collected at Ignalina, they came upon
their recording of the door, which they’d
both earmarked as a dead cert for use.
It’s a good example of something that
started life as a field recording and then,
through processing, became musical.
Minus an edit here and there, it formed
the percussive core of the track that
opens the Chernobyl soundtrack album,
appropriately titled ‘The Door’. It might
sound as if there’s a keen drummer
playing perhaps with some brushes on
a tin can. In fact, it started life as a drone
sound, captured with contact mics on that
Ignalina door.
“There was lots of tonal information in
the mids,” Sam says, “but then up around
16k there was this tiny and incredibly fast
fluttering that came from... something. But
there it was. So, we’re in the studio and we
go to isolate this fluttery sound. We were
using a lot of old tape machines, basically
to slow down this sound and move 16k to
8k to 4k and down into a useable domain.
And the tape maintains enough structural
integrity in the sound. I often find the same
process digitally gives you something very
brittle, but every time you slow something
down with a tape machine, it’s adding
noise on top, and if you then subsequently
process that, the noise becomes part of
the sound.” He used transient shapers and
similar tools to accentuate the dynamics.
“We ended up with this very broad, very
beautiful-sounding thing that’s chugging
away in a very rhythmic pattern. It was an
awesome groove!”
Sam contrasts this approach with what
he calls the synth fetishism prevalent in
Berlin, where someone might turn up their
nose at your TB-303, say, because, “‘Oh,
that’s like the second generation version
with the crappy filters, whereas I’ve got the
real 303.’ A 303 or a drummer playing the
same thing as that groove on ‘The Door’
doesn’t give the same effect, somehow.”
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 IN TE R VIE W
HILDU R GU ÐN A D Ó T TIR & S A M S L AT E R
Hildur agrees: “The process you use
to get the result you want when you’re
writing music or recording music is as
important as the end result itself. Because
the way you were feeling or what you’re
experiencing through that process is
hugely influential on what the final result
will be. That’s something I find very
important — enjoying the process as much
as you enjoy the outcome.”
Heart Of Glass
Another huge project that Hildur
Guðnadóttir and Sam Slater have worked
on together is Battlefield 2042, the
Electronic Arts video game introduced
in 2021 and staged in a near-future
climate-disaster resource war. And while
they planned again to travel to various
suitable locations to find and record
sounds for the score, the only place
they managed to get to, thanks to Covid
restrictions, was a World War II submarine
base in Saint-Nazaire in France. They
brought hydrophones and contact
mics, and went into the submarines and
warships to capture strange sounds.
They took the results to Kerwax, a studio
in an old boarding school in Brittany
run by Christophe Chavanon and Marie
Kernéüs. Sam reports that Christophe has
a 40-channel valve console there that
he built on his kitchen table. “It’s nuts,
really fun,” he says. “And basically, it’s
a 40-channel distortion machine.”
However, back in Berlin with lockdown
under way and no opportunities for
sonic outreach. they had to modify
their approach, staying put and making
sounds in their studio, using all manner
of sources and techniques. “We were
interested in the idea that we could take
the materials of the various levels of the
game and create the music from that,”
Sam explains, “so that the level comes to
life sonically around the players. There
are no real, traditional instruments on the
whole score.”
One level of Battlefield 2042 is
a cityscape where the humans are
all gone but the buildings remain. It
seemed to Hildur and Sam that there
might be some mileage in creating
a musical dialogue between people
and glass. To do this, Sam says, they
built a machine-learning algorithm with
a coder in Berlin. “You feed it material
sounds, so in this case, we worked with
a percussion wizard, Robyn Schulkowsky,
and recorded all these incredible ringing
glass and textural glass sounds. Then
154 April 2022 / w w w. s o u n d o n s o u n d . c o m
you ask the algorithm to essentially learn
the properties that it thinks are present
within glass. So it makes a sort of glass
machine. And then you take a voice, enter
it into this algorithm, and it will stylistically
transfer the sound of one thing on to the
sound of the other.”
When you ask the algorithm to process
these sounds, he says, it might spurt out
24,000 files in a single process over the
course of many, many days. “And then you
have to sit and go through the outputs of
this machine. It can be so beautiful, and
it can be chaotic — and it can be totally
empty. It’s a strange technology, and there
are already many examples of algorithmic
generative machines that go further
than what we’ve used by a million miles.
But ours felt pretty punk, and we really
enjoyed that.”
Doom Without Gloom
Composing and recording the score for
a game is of course quite different from
working on a movie or TV, thanks in part
to its nonlinearity. “That’s what interested
me from the beginning,” Hildur says.
“It’s an interesting way to write music
when you’re not on a timeline. You’re
working in a time-fluid narrative, while
the narrative itself can also change —
and the person playing the game can
influence how the music is working.
It was interesting to work outside
a story-based narrative, and go more
with this narrative of the surroundings
rather than the emotional narrative. In
fact, the changes that happen in those
surroundings are the things that affect
you the most emotionally.”
It’s also important to consider the
long-term potential of the job. When is
the music for a game completed? Hildur
is working on more levels as we speak,
and the game’s producers are constantly
responding to feedback. Players might
find the footsteps here are too quiet, say,
or that the music there needs bolstering,
and so on — and EA will release a patch,
alter the audio engine and remix things,
as appropriate. The music for a game
becomes a dynamic object, with its
relationship to other sounds within the
game constantly evolving. “The whole
process is nonlinear,” Hildur says.
At one point in the process, they were
talking to the audio director, and someone
said how sad it was that they’d just died
as a player. The director was unfazed. He
told them it was essential to remember
that within the context of the game, dying
as a player is one of the least important
things. Sam laughs at the memory, but at
the time he was surprised. “I’m like, what?
The guy says ‘Yeah, you just get reborn
and chucked back into the game.’”
This is in complete contrast to the
thinking when you’re producing material
for a movie. “If the protagonist dies in
a film, then you’re really in trouble,” Sam
says, still smiling. “It’s a different film,
suddenly. But within this context, the
context of a game, the narrative somehow
sits slightly outside of the action itself.
So it’s a really different perspective. You
can charge all these landscapes and
these environments with atmosphere and
meaning and excitement, but ultimately
that creates a kind of cloud or a space
in which the play takes place. It’s a very
different way of thinking.”
Hildur comes back to the thought that
the score they made for the apocalyptic
Battlefield 2042 came out of their
explorations around the concept of
materials. “And also, it came out of an
overwhelming sense of fear and dread
as we sat there in the middle of this
pandemic. We’d be thinking ‘Oh, my God,
is this really where we’re heading?’
“I was trying to turn that feeling
into music, but I hope in a way that is
motivating rather than doomy. Because
when you start thinking about this,
especially in times like these, it’s really
easy to lose hope and to lose faith. It’s
easy to think, well, why should we even
bother? But it’s important to us that we
don’t lose hope, that we keep working
towards a better future. I think that’s the
point of this game, and it’s why we chose
to be a part of it to begin with. We hope
that people will get that feeling as well and
hopefully be motivated to take action. Not
just to take action to play this game, but
also to take action for our planet.” 
From One Studio To Another
The Kreuzberg studio where Hildur Guðnadóttir
and Sam Slater met was an inspirational shared
space in the middle of Berlin, Sam says. “Lots of
great musicians and composers came through.
It was the kind of space that could exist in Berlin
20 years ago but just isn’t possible to find now.
We’d all meet up in the central kitchen area — part
kitchen, part social area, part therapy space. If you
look down the credit lists on Hildur’s scores, if you
look at the credits on Jóhann Jóhannsson’s scores,
on Dustin O’Halloran’s scores, and various other
people’s scores during the years we were there,
they will mostly consist of the people who were
there in those rooms, the people who worked in
this great, interwoven space.”
When the Kreuzberg studio closed down,
Hildur and Sam had to hunt through Berlin for
a fresh place. It turned out their friend Francesco
Donadello had some spare capacity at his Vox-Ton
studio, in an old piano factory in the north of the
city. Hildur and Sam built two small production
rooms into a couple of spaces there — they look
about 20 square metres each — and they have the
benefit of backing on to the larger main studio.
“I worked with Francesco for over a decade,
and he’s engineered most of my work,” Hildur
says, referring for example to her 2014 album
Saman or the score she composed with Jóhann
Jóhannsson for the 2018 movie Mary Magdalene.
“So it seemed a logical and extended family
situation for us to go to his place to work.” Having
two rooms means that Hildur and Sam can be
isolated when they need to be, and when they
work on larger sessions — or what Sam calls their
strange sound-gathering experiments — they can
Sam’s space is a hybrid affair with some choice outboard available.
link all the rooms together. “And then,” he says,
“we really can have a lot of fun.” Trained cellist
There’s plenty of gear available, too. Hildur Guðnadóttir
“The beautiful main live room here has an has her own unique
old 24-channel Cadac console,” Sam says. instrument at hand.
“Francesco bought the history of analogue gear
when everyone thought analogue was lame and
digital was cool. Now, obviously, the tide has
turned. He bought the famous Klein+Hummel
mastering EQs, things like that. People were
like, ‘Take both of them for a grand,’ and they’re
worth, what, 10, 15 times that now?”
Sam describes his space as a sort of hybrid
environment involving ATC SCM50ASL Pro
monitors, an Apogee Symphony MkII (“the king
of converters”), and outboard gear including
a Tube-Tech SMC 2B stereo compressor, an
Audio & Design F760X-RS limiter, an Aphex
204 Aural Exciter, a Shadow Hills Equinox rack
mixer, a Zahl EQ1 equaliser, a pair of BAE 1073D
mic preamps, a Thermionic Culture Vulture
saturation processor, an Overstayer 8755DM
stereo input channel, an early release of
Teaching Machines’ Wellspring reverb system,
and a Moog Minitaur bass synth.
“I think the rooms crystallise our working
relationship very well,” says Hildur. “My room
is dedicated to thinking and receiving music,
where Sam’s room is dedicated to the knobs.
Not that he doesn’t think of the music as
well, but our different thought processes are
crystallised right there.” Her work space thus
looks positively relaxed, with just a piano, some
string instruments, a small desk, and some
HEDD Type 20 Mk2 speakers.
w w w. s o u n d o n s o u n d . c o m / April 2022 155
 ON TE S T
Heavyocity
Mosaic Leads
Kontakt Instrument
+++++
I’ve been very
impressed with the titles I’ve reviewed from
Heavyocity’s Mosaic series over the last couple
of years. While each brings a specific theme
in terms of the core style of the sound source
— Keys, Vocals, Tape, Pluck and Bass — they
share a common Kontakt-based front-end. As
well as some very flexible effects and sound
modulation options, it’s the ability to combine
three individual sounds into a preset, and the
excellent arpeggeiator/pattern features, that make
these titles instantly inspiring for the modern
media composer.
Heavyocity have now released Mosaic
Leads. The focus here is very much on analogue
synth-based sound sources (in contrast to Keys
which features electric piano sounds) but you
get the same three-sound-slot front-end. The
instrument includes over 100 individual sound
sources, and these are combined into over
200 snapshot presets. The sounds themselves
are excellent and, while the underlying library
is a fairly modest (by modern standards) 3GB
in size, by making full use of the ability to
combine three sounds into a preset, and the
impressive filter, effects and modulation options
within the engine, Heavyocity’s sound design
has created a huge range of synth tones here.
Indeed, Mosaic Leads can do classic rich,
warm, old-school analogue, more aggressive,
cut-through-the-mix leads, delicate and subtle
tones for lighter melodies, moods and noise/
sound effects.
The snapshot presets are divided into three
categories. The Playable category lets you
play your own melodic/harmonic parts like any
regular virtual instrument. However, it’s the Arp
and Rhythmic categories where the most instant
gratification is to be found as these make full use
of the powerful arpeggiator/pattern-sequencing
engine. For example, with an Arp category
preset, you simply play a few notes, adjust the
mod wheel (which acts as a macro-style control
for the sophisticated sound modulation engine)
and it’s instant cue time. The Rhythmic category
also uses the internal sequencing features but
emphasising rhythmic elements rather than
pitch-based shifting. Whether as a low-end pulse
156 April 2022 / w w w. s o u n d o n s o u n d . c o m
or high-end top-line, these can
easily provide your cue’s rhythmic
starting point.
Given these different preset
types, combining a couple of
instances of Mosaic Leads can take
you a long way towards a finished
piece of music. And, given the
combination of the synth-based
sound set and the powerful
modulation/performance options,
if you are a media composer
wanting to generate score cues
that span the stylistic ballpark from
Terminator to Stranger Things,
Mosaic Leads ought to be right up
your street. Individually, the Mosaic
titles are very sensibly priced and
ought to be within the reach of any
working composer. There is also
a bundle offer if you want to buy
the whole set. If Mosaic Leads falls
within your compositional brief, this
is top-notch stuff and a joy to use.
John Walden
$119
www.heavyocity.com
Sonora Cinematic
Poiesis Cello
Kontakt Instrument
++++
Composer Alessandro Mastroianni’s
Sonora Cinematic operation are
known for their Dronar collections
of complex evolving textures, pads
and soundscapes. Poiesis Cello, the
first title in the company’s new solo
string series, expands the frontiers
with a large set of atmospheric
articulations beautifully performed
by Alan Black, distinguished
principal cellist with the Charlotte
Symphony Orchestra. Mr Black’s instrument was
originally commissioned and owned by Yo-Yo
Ma, so as you might expect, it sounds superb.
The library is 4.92GB installed and runs on the
free Kontakt Player (v6.5.3 and up), as well as the
full version of Kontakt.
As everyone knows, the ancient Greek
word ‘poiesis’ refers to the act of bringing
something into being that didn’t exist before,
and that’s certainly true of some articulations in
this unconventional library. Rather than dutifully
slogging through the usual menu of longs,
shorts, plucks and tremolos, the player freely
varies his bowing to produce timbral shifts and
shadings, octave harmonics, bow bounces
and reiterations, creating mobile textures
which range from tender and reflective to
nervily unsettling.
The main Textures patch contains 43 long,
sustained articulations, any of which can be
loaded into either of the patch’s ‘A’ and ‘B’ layers.
You can create mobile, subtly evolving textures
by using the mod wheel to crossfade between
the two layers, with the added satisfaction
of knowing the number of possible sound
combinations is practically infinite.
I was taken by the delicate flautando
and sul tasto styles (the latter of which has
a nice tremolo variant) and loved the dreamy
harmonics. A set of ‘normale’ straight notes
played with a restrained and graceful vibrato
work well for melodies and pads, while an
additional shorts patch contains spiccato,
staccato and pizzicato (including Bartok snap)
artics. There are also ‘motions’ note reiterations
played in half-, quarter-, eighth- and 12th-note
divisions. All sustained styles have a muted con
sordino option which you can crossfade with the
unmuted version to great effect.
In a less traditional vein are extended
techniques such as the ‘behind the bridge’
bowing (which introduces classic horror-film
dissonances) and a mad circular bowing
technique where the player continually
changes the bowing point to produce
a constantly evolving, somewhat unnerving
one-man soundscape. More experimental
‘textural’ performances range from solemn,
harmonium-like drones to agitated and jittery
rebowings. While the deliveries generally lack
aggression (no Mad Max bombast here), the
edgy sul ponticello performances are more
imposing and threatening.
The cello was recorded over a C2-C#5 range
in a dry studio acoustic using three high-end
mics — each has its own mixer channel, with
a hard-panned stereo mix of the three also
available. Unlike Sonora’s Dronar titles, these
samples are unprocessed and effects are
entirely created by the performer, thus offering
composers an organic and intimate human touch
suitable for sensitive drama and subtle
psychological thriller scores. Dave Stewart
£82.50
www.sonoracinematic.com
Klevgrand
Borsta
Plug-in Instrument
++++
Borsta is a brushed percussion instrument
plug-in available in AU, VST and AAX
formats for PC and Mac, and also available
for iPad. Its audio engine is designed to
simulate natural-sounding brush playing
and it offers 18 different instrument
sounds, including snare drum, frame
drum, cajon, djembe, rusty bucket, chains,
cardboard pipe, spade and so on, including
multisampled one-shots and round robins.
Different pattern types are loaded for the
various preset menus but it is possible to
use any pattern with any sound menu.
At the heart of Borsta are two
sequencers that run together: the Event
sequencer and the Envelope sequencer.
The Envelope sequencer, which is viewed
as a set of editable curves shown in blue,
sets the brush intensity while the Event
sequencer determines when and how
the brushed elements start and end.
Note that brush sounds set up in the
Envelope sequencer only sound where
there is a corresponding Stroke in the
Event sequencer.
There are actually three Event views,
as selected by the red Strokes box or
‘1’ or ‘2’ buttons, the number buttons
corresponding to the one-shot types
available from the current sound type
menu. Buttons at the bottom of each hit,
represented by vertical lines that also
reflect their velocity, select between their
open and closed sounds. There are tools
for adding and deleting events, snapping to
grid etc, as well as sliders to adjust shuffle,
delay, time randomisation and velocity
randomisation amounts.
Each sequencer has independent
sync settings for Host Beat, Direct and,
in the case of the Envelope sequencer,
External, which hands over brush control
to an external MIDI CC. The length of each
sequence can also be adjusted to add
more steps.
The Global tab brings up a mixer that
allows adjustment of the levels of the
Brush, Brush On, Brush Off, and both
One Shots (open and closed), as well as
a master level, a master reverb, master
room ambience with a choice of IRs,
a three-band EQ, and global controls for
pitch and added harmonics.
While this may seem like an oddball
instrument, it is actually capable of some
really natural-sounding rhythm patterns,
many with a distinctly ethnic flavour. The
brushed drum sounds are very realistic
while things like the metal spade sound
almost as though they are being scraped
or rubbed. For some musical styles Borsta
may provide a complete rhythm part,
though I can also see it being used to add
‘toppers’ to more basic drum loops to give
them interest. If you’re after something
that sounds really organic, Borsta is highly
recommended. Paul White
$59.99 desktop, $19.99 iOS.
www.klevgrand.com
Best Service
Elysion 2, By Sonuscore
Kontakt Instrument
+++++
The original Elysion release got a glowing
review from Nick Magnus back in the April
2020 issue of SOS. It took The Orchestra’s
very impressive performance engine
and fine-tuned it to work with a range of
synthetic and ‘organic but processed’
sounds. The result was just as impressive
as The Orchestra and the two products
undoubtedly complemented each in terms
of generating scores with a hybrid flavour.
Now, just as happened with the Orchestra
2, Elysion has received an upgrade;
Elysion 2 The Encounter.
All the original Elysion content and
features (including MIDI export) are
included but Sonuscore have increased
the number of individual instruments (now
over 200) and the number of animated
themes (for many, the star of the Elysion
show and now over 400 available). The
rather quirky (but very effective) theme
categories based upon the names
of planets (and now including other
interstellar objects) continues. However,
it’s not just more of the same; the ambient,
organic and cinematic moods that
dominated the original sound sets are now
supplemented by some more aggressive,
powerful
and driving
options. The
new additions
are excellent,
considerably
expanding the
sonic palette,
especially in
terms of darker, more sinister tones.
Extra sounds aside, some of that
additional aggressive element is delivered
thanks to new distortion and saturation
effects options and, in many of the presets,
these can be dialled in via the mod
wheel as part of the overall performance
dynamics. It’s very effective indeed and,
for composers working in modern drama
or sci-fi genres, some of the new Animated
and Pulse (rhythmic) presets are going
to provide plenty of inspiration for new
cues. Also new is the option to define
two envelope or arpeggiator patterns
within the performance engine for each
of the five sound slots, and the ability to
morph between them. This is really cool
in use, with the mod wheel (CC1) allowing
you to morph between the two states,
although you can also specify CC2 or CC11
as alternatives.
Also new in v2 are single-instrument
NKIs. While the performance engine
is undoubtedly one of Elysion’s star
attractions, these single instruments are
a very useful addition, especially when it
comes to adding a playable melodic top or
bass line alongside something like one of
the Animated presets. These are organised
into three categories: Ambience FX (pads
and textures), Hits (melodic sounds with
short sustains and some drum/impact
sounds) and Sustains (melodic sounds with
longer sustains).
If you’re a regular user of the original,
the upgrade to Elysion 2 is an absolute
no-brainer. For potential new users, Elysion
2 provides both instant gratification and
inspiration via the various animated
presets. However, that instant gratification
also hints at the long-term potential for
those media composers willing to dig
a little deeper. Sonuscore absolutely know
their stuff when it comes to designing
performance engines for multitimbral virtual
instruments; Elysion 2 The Encounter is
powerful stuff. John Walden
$249
www.bestservice.com
Audio examples of this month’s libraries are
available at www.soundonsound.com.
w w w. s o u n d o n s o u n d . c o m / April 2022 157
H ow do balanced signals
work in audio gear?
Could you please explain why audio
gear uses balanced signals and how
they work? Are impedance-balanced
signals the same thing and if not, what
advantages or disadvantages do
they offer?
SOS Forum post

Technical Editor Hugh Robjohns

A balanced audio system (it is the whole
output-to-input ‘system’, not just the
cable) is primarily used to minimise
unwanted external interference making
its way into an audio circuit, but it also
has the advantage of being ground-free,
by which I mean the audio signals
passing across it are not referenced to
the system ground, and that helps to
avoid ground-loop noise problems.
A balanced interface comprises two
signal wires and, usually, an overall
(grounded) screen, although the last is
not technically essential. The balanced
input receiver, whether in the form of
a transformer or active electronics, works
‘differentially’. This simply means that
the signal voltage on one input ‘leg’ is
subtracted from that on the other. If the
signal voltages are exactly the same on
both legs (what we call a ‘common mode
signal’) they cancel each other out, so
that common-mode signal is rejected. In
contrast, the voltage difference between
the two legs will pass through. So, for the
interface to work as intended, the wanted
audio has to be sent as a differential
signal, and unwanted interference has to
be made common-mode.
To act as a common-mode signal,
external interference must generate
identical voltages on both connection
wires, and this requires the two lines to
have exactly the same impedances to
ground. That’s what makes the interface
‘balanced’: it’s the impedance balance
that matters. A lot of balanced interface
explanations wrongly assert that there
must be symmetrical and inverted audio
signals on the two wires, but actually the
wanted signal just has to be presented
differentially (different signals on each
wire), and there are a couple of equally
valid ways of achieving that.
Arguably the most common approach
is to split the input signal voltage (V) to
send half on one wire and half on the
A balanced interface using symmetrical and inverted output signals.
An impedance-balanced interface puts all the wanted signal on one wire, and nothing on the other.
other, but with the latter polarity inverted:
the classic symmetrical-and-inverted
format. Mathematically, that’s V/2 on
one wire and -V/2 on the other. When
subtracted at the differential receiver
(balanced input), the output is: V/2-(-V/2).
Simplifying that equation, you get (V+V)/2
= 2V/2 = V. In other words, you get out
exactly what you put in! Any interference,
though, will appear as +U on both wires
and because +U-U = 0 the interference
doesn’t appear at the output at all. Hurrah!
The other approach, known as the
‘impedance-balanced’ format, sends
the whole input signal on one wire and
nothing on the other (it is effectively just
grounded). For this technique, we have V
on one wire and zero on the other, so the
receiver gets (V-0) = V. Again, you get out
exactly what you put in, and interference
is still rejected (since U-U=0).
So, if both approaches deliver the
same result why have the two formats?
Impedance-balanced outputs are
less costly to implement and have
the useful advantage of being fully
compatible with both balanced and
unbalanced destinations, which is why
they’re popular in semi-pro equipment.
However, in professional applications
the symmetrical-and-inverted signal
format offers other advantages.
First, lots of audio equipment uses
transformers, and transformer-coupled
outputs inherently generate that format,
making it convenient. Second, it offers
a 6dB headroom and 3dB system noise
advantage. The third advantage is that
the output circuitry (or transformer)
feeding each signal wire is identical, so
there are inherently identical impedances
to ground and this ensures excellent
rejection of interference.
In contrast, an impedance-balanced
output has only one line driven actively,
while the other is simply tied to ground,
typically through a resistor whose
value is selected to match the notional
output impedance of the active driver.
However, the output driver’s impedance
might vary with frequency, especially
at radio frequencies, and this can
result in the source impedances of the
two lines being imperfectly balanced
so interference may not be rejected
quite so effectively. Of course, most
impedance-balanced outputs are
carefully optimised to minimise this
potential problem, but it is a fact that
there’s a risk of being slightly less
effective at rejecting interference than
symmetrical-and-inverted outputs.  

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

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w w w. s o u n d o n s o u n d . c o m / April 2022 161

 MAKING A DIFFERENCE
appearances. All the stuff those kids lives that most need when you hear that they’ve
SIMON GLENISTER
you wanted as a kid, except it: those in care, excluded from re-engaged with school or

M
y school attendance the cash, obviously (though school, etc — and we’re doing their mental health team
was, ahem, erratic emotionally rich, right?). it by making bleepy noises because you’ve changed how
to say the least. The What really changed with kids in studios. they feel about themselves.
only place I felt I was any though was my own sense of I say that last bit slightly We’re saving local authorities
good at anything was hitting self. Music had changed me, flippantly, because it’s more and families about a million
drums in the music block at made me believe in myself. complex than that. We know pounds a year through
lunchtime. The only thing I could be good at something. exactly what we’re doing and reduced use of services and
I left with was an internalised I had a concurrent role as how to do it and why it works. improved life outcomes.
feeling of educational failure. a youth worker (a boy’s gotta That’s because in a bizarre The reason I wrote this
After school finished, I noticed eat), and after juggling touring twist of fate I got invited to piece is that I wanted to say
that all my friends were off and youth work in 2009 I set do a Masters at Cambridge that there is a whole world
to this mysterious thing no up Noise Solution, a social University to work out what it of potential opportunity out
one had mentioned called enterprise that fuses youth was we were doing that was there for producers like you,
University, and I was stuck with work with music technology having such dramatic impacts in community music projects
no qualifications in a dead-end mentoring. I figured if music on kids. (Take that Mr “You’ll like www.noisesolution.
post-industrial town (sorry could change me, it could never amount to anything, org. Your options aren’t
Kidderminster). change others. Glenister” Form Tutor.) just restricted to studios or
Then I met this guy in a pub That organisation is now It turns out that teaching being a pop star. You can
who said come to London and nationally award-winning and kids to use a DAW is a perfect use your skills to both have
join a band. Well, I grabbed has grown into one of the UK’s way to engage them. It’s fun and make a difference
that ticket with both hands. top 100 best-performing social amazing seeing a kid have in helping people. I love
What followed was a career, enterprises. It’s a community that ‘get it’ moment as they using my production chops
multiple record contracts, music project that is having layer their first drum loop to make a difference. You
international travel, festival transformational impacts on and bass line. It’s even better might too. 

AD INDEX To Advertise in Sound On Sound please contact Paul DaCruz t: (707) 569 6021 e: paul.dacruz@soundonsound.com

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