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October October 2014 In This Issue
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>> Access ALL ISSUES Competition People + Opinion
Win Adam A7X monitors & Sub8 (UK, EU Inside Track: Jack White
ROW Only) Secrets Of The Mix Engineers: Jack White, Vance Powell &
Deadline: 2014-11-03 Joshua V Smith
Fantastic chance to Win Adam A7X monitors & For Jack White, analogue recording is not about
Sub8 in October's Issue Competition. looking back to the past, but choosing the ideal
medium for his art.
Win Dangerous Music D-Box (Americas Only) Sounding Off

Deadline: 2014-11-03 Why music is better now than it ever has
Fantastic chance to Win Dangerous Music D-Box in been. Alex Marten asks the question.
October's Issue Competition.
Why Pro Studios Matter

Reviews Leader
It is an unfortunate fact of life that the
ADK Z49
Valve Capacitor Microphone democratisation of music production, much of
which is down to the affordability of...
ADK have tackled Neumann’s M49 before — but in
this Custom Shop model, they’ve really pushed the
boat out. Is it worth the extra cost? Te c h n i q u e
Mix Rescue
Allen & Heath Qu24 Patrizio Cavaliere: Our engineer sorts out the
Digital Mixing Console bottom end of a deep–house track.
Allen & Heath’s Qu series goes from strength to
strength, with the latest offering more I/O, more
faders and more mix groups. Mix Rescue | Media
Patrizio Cavaliere
App Works Audio files to accompany the article.
Making Music On The Move
I do like a gadget, and at a show in Rome last week Package Deal
Pro Tools Tips & Techniques
something shiny caught my eye.
The plug-ins that come with Pro Tools are more
powerful than you might think.
Arturia Vox Continental V

Software Instrument
Arturia have pointed their considerable modelling Peak Performance
Sonar Tips & Techniques
experience at recreating a ‘60s classic.
Get the visual feedback you need with custom
metering in Sonar.
Audio Interfaces: The HDX Factor

Spotlight
There are many companies making interfaces for Pushing The Envelope
Ableton Tips & Techniques
Pro Tools HDX systems these days — but straying
from the Avid path carries some risks. We delve into the mysteries of envelope following in Ableton
Live.
Audio Spillage Drum Spillage 2

Drum Synth Software For Mac OS
Drum Spillage 2 lets you get down and dirty with
synthesized drum sounds.
Session Notes
Blue Rose Code: Recording A Live Show
We overcome unfriendly acoustics and
Crane Song & Dave Hill Plug-ins troublesome spill to capture a memorable gig.
Tape, Vinyl & Distortion Plug-ins For Pro Tools
When a top hardware expert turns his attention to
plug-in design, the results are a bit special.
Session Notes | Media
Blue Rose Code: Recording A Live Show
Audio files to accompany the article.
Dangerous Music Compressor
Dual-channel Compressor Side Effects
The Dangerous world of dynamics processing Digital Performer Tips & Techniques
promises plenty of pleasant surprises. Enjoy the creative possibilities of taking Digital
Performer’s bundled plug–ins out of their comfort
zone.
EBS ValveDrive DI
Bass DI & Overdrive Pedal Studio SOS
Could this thoughtfully designed bass front-end be The SOS team visit a community youth centre to
the answer to every bassist’s stage and studio help them make the most of their new studio setup.
prayers
Electro-Harmonix B9 Organ Machine The Beat Is On

The Beat Is On
Organ Emulation Effects Pedal Logic Tips & Techniques
Is EHX’s latest pedal perfect for guitarists who want Nine more ways to kick off your productions in
to pull out all the stops? Logic Pro X.
Business
Audiobooks
audible.com
Greiner Engineering Tools Sum.mation The Eightest & Greatest
Analogue Mixer With DAW Plug-in Control Reason Tips & Techniques
If you use outboard gear with your DAW, this could We look at what’s new in the eighth instalment of
be just what the doctor ordered. Reason.
Free Audiobook With 30

Day Trial. Listen On
Headway EDM1 & EDB2 The Mix Review
Smartphone Or Tablet! Acoustic Instrument Preamps Commercial Productions Analysed
Headway’s portable preamps are designed to help This month brush up on your listening skills with
you get the best possible sound from your acoustic Kiesza, Rita Ora, Duke Dumont, Kaiser Chefs and
instrument, whatever it may be. Mr Probz. Plus our Classic Mix - Santana's Smooth
featuring Rob Thomas.
WIN Great Prizes in SOS How Effective Are Portable Vocal Booths?
Competitions! Screen Tests Unfolding Events
Studio One Tips & Techniques
Win Dangerous Music D- Portable acoustic screens are designed to make it
Box (Americas Only) possible to record good–sounding vocals almost Studio One lets you apply effects directly to audio
anywhere. We put this claim to the test! Events, opening up innumerable creative WIN Great Prizes in SOS
Win Adam A7X monitors
& Sub8 (UK, EU ROW
possibilities. Competitions!
Only) Win Dangerous Music D-
How Effective Are Portable Vocal Booths? |
VCA OK Box (Americas Only)
Media Cubase Tips & Techniques
Screen Tests Win Adam A7X monitors
I’ve long wished Steinberg would implement true & Sub8 (UK, EU ROW
Audio files to accompany the article.
VCA–style automation in Cubase — it is, for me, a Only)
Korg SDD 3000 critical...
Digital Delay Processor
A modern stompbox take on a classic digital delay VCA OK | Media
unit — what’s not to like? Audio files to accompany the article.
The ZIP file of the Cubase 7.5 project template.
Music Business
Lewitt MTP 840 DM & MTP 940 CM
Hand-held Vocal Microphones
Can The App Save The Album?
Lewitt have made some fine impressions in the Power Apps
studio world with their innovative mic designs. Can Smartphones and tablets are giving artists new
they do the same for the stage? ways to explore multimedia. Does the ‘album app’
have what it takes to rekindle music sales?
MeterPlugs Meter
Metering Plug-in For Mac OS & Windows
Notes From The Deadline
An oft–overlooked advantage of analogue working TV Music From The Inside
methods is sensible metering. MeterPlugs’ K–Meter Credits can be a vital resource to the media
brings the same approach to your DAW.
composer, but they can also be a ball and chain.
Phoenix Audio Nicerizer Junior

Analogue Summing Mixer
Off The Record
The Nicerizer Junior aims to offer what analogue still Music & Recording Industry News
does best — subtle, yet pleasing warmth and Historic studios might hold emotional value for us,
distortion. but they are, after all, still only buildings.
Phoenix Audio Nicerizer Junior | Media

Studio File
Audio files to accompany the article. Teldex Studios Berlin
Playback Only a very few studios in the world are capable of

Reader's music reviewed accommodating the largest recording gigs.
This month we review releases from Dreamlake,
Jimmy Brewer and Phillip Sandifer.
Sound Advice
Q Are impedance mismatches causing my

Recording Music On Location vocals to sound harsh and aggressive?
Book Review Q Are impedance mismatches causing my vocals
Recording on location presents many challenges: to sound harsh and aggressive?
this book will help you meet them. I work in a facility where AES3 110Ω cables...
Q How do I emulate a real drum sound using

Roland System 1 EZdrummer?
Analogue Modelling Synthesizer
I’m currently trying to emulate the drum sound on
The System 1 is a fully fledged synthesizer in its the recent Lower Than Atlantis song ‘Here We Go’
own right, but it can also assume the identity of using...
Roland classics using its built–in ‘plug–out’
technology.
Q How does knowing the key of a song help
Sample Shop | Media when mixing?
Audio files to accompany the article. Q How does knowing the key of a song help when
mixing?Some audio forums are really keen on
Scuffham S-Gear 2.4 pitch/frequency charts as an aid to...
Amp Simulator For Mac OS & Windows
Two contrasting new virtual amplifiers and a reverb
add to the appeal of Scuffham Amps’ virtual amp Q Why has Paloma Faith's latest track been
simulator. recorded 'out of tune'?
Q Why has Paloma Faith’s latest track been
recorded ‘out of tune’?
Sonic Farm Silkworm
Microphone Preamplifier
This 500–series preamp boasts huge tonal
versatility.
Spitfire Audio Mural Symphonic Strings

Sample Library
Spitfire Audio enter deep symphonic waters with
their latest string library.
Steinberg UR44
USB Audio Interface
Could Steinberg’s mid–sized USB audio interface
provide the perfect package for the home studio and
music on the move?
Ueberschall Brasil Nova Primeiro & Brasil

Nova Segundo
Sample Library
Two reviews for the price of one here; Ueberschall’s
Brasil Nova Primeiro and Brasil Nova Segundo.
Ukulele Free Agent: Oldies Uke, Tropical Uke,

Folk Uke & Kalakaua Ave Sessions
Sample Library
Ukuleles galore!
Universal Audio UAD v7.8 Plug-ins

Thermionic Culture Vulture, Valley People Dyna-mite, Tonelux
Tilt & Neve 1073
Universal Audio’s catalogue of virtual vintage plug–
ins continues to grow.
Zero-G Extreme Environments

Kontakt Instrument
Created for Zero-G by producer and composer Si
Begg, Extreme Environments is primarily a Kontakt
instrument for soundscapes,...
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Sub8 worth £1200 What is the stated frequency
response of the A7X
In this month’s competition, we’re giving away a fantastic 2.1 monitoring system monitors? *
from Adam Audio consisting of a pair of A7X speakers and a Sub8 subwoofer.
42Hz to 30kHz
“Getting such a strong and transparent‑sounding mid‑range from a two‑way 42Hz to 40kHz
speaker is quite a challenge, but Adam seem to have managed to coax
42Hz to 50kHz
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can read his review here (http://sosm.ag/aug10adam) but it’s not only our
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A7X was voted by our readers as the best monitor of the year in both the 2011 Sub8? *
and 2012 SOS Awards. 5.5 inches
The A7X monitors offer a wide frequency range, with a quoted response of 42Hz 6.5 inches
to 50kHz. While much of this range is ultrasonic, a wider frequency response 7.5 inches
generally means a flatter phase response within the audible band.
8.5 inches
The upper frequencies are served by Adam’s X-ART ribbon tweeter, while the
lower frequencies are covered by a lightweight but rigid seven-inch bass/mid What kind of amplifier is used
driver made from a sandwich of carbon, Rohacell and glass-fibre. to drive the X-ART ribbon
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The A7Xs pack a serious punch too, with a 50W Class-A/B amp driving the
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Box worth $1499 What types of digital sources
can be accepted by the D-
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transition from two-inch tape and mixing desks to DAW recording and in-the-box
AES3 and S/PDIF
mixing presented some interesting new challenges. Living with and working
through the issues presented by this new studio paradigm led to the creation of MADI and S/PDIF
the Dangerous Music product line. The company’s ethos has always been to put ADAT and AES3
audio quality first, rather than to build to a price. This is demonstrated by their
ADAT and S/PDIF
Compressor reviewed in this issue, as well as this month’s competition prize: the
Dangerous Music D-Box.
Which inputs can be manually
In his March 2009 review (http://sosm.ag/mar09dbox) Sound On Sound panned? *
Technical Editor Hugh Robjohns called the D-Box “classy” and said that it gave 1 and 2
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monitors and has a pair of powerful cue headphone feeds with talkback The D-Box’s stated frequency
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“towards a hint of warmth rather than clinical sterility” and that he “detected no
noticeable sense of fatigue even after extended listening” — excellent traits in 1Hz to 10kHz
such a device. 20Hz to 20kHz
The 1U rackmountable D-Box has a series of useful front-panel controls such as 1Hz to 100kHz
a talkback switch, mono mode switch and an alternate speaker mode which can
be used for both A/B’ing or additive mixing. Tie-breaker *
What is the most dangerous thing
Then, as a summing box, it features eight inputs connected via the ubiquitous D- you’ve ever done? Answers in 30
sub connector. The first six are stereo pairs but inputs 7 and 8 can each be words or fewer, please.
individually panned. The mix bus boasts loads of headroom, a very low noise
floor and provides “that analogue magic that helps a mix gel together”, says
Hugh. It also has an extraordinarily flat frequency response, which is stated as
being a flat ±0.1dB between 1Hz and 100kHz.
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In this article:
In The Box
ADK Z49
Listening Valve Capacitor Microphone Buy PDF
Alternatives Published in SOS October 2014

Reviews : Microphone
ADK Printer-friendly version
Z49 $3599 ADK have tackled Neumann’s M49 before — but in this
pros Custom Shop model, they’ve really pushed the boat out. Is
Emulation of classic M49 it worth the extra cost?
sound character, with rich Hugh Robjohns
mids and subdued high end.
I
Quality components recently reviewed ADK’s Frankfurt 49T, a solid–state (FET) large–diaphragm microphone designed to honour the sound of
throughout, including the legendary Neumann M49 (which was the first microphone in the world to feature remote–controlled polar pattern
Lundahl transformer.
selection). The 49T is part of ADK’s family of T–FET mics, which share common electronics and hardware, but emulate the
Supplied with alternative
valve options.
most popular classic European mics through the use of different custom–designed capsules. The subject of this review, the
Sturdy construction and nice Z49, is a large–diaphragm valve mic that plays an identical role but within the flagship Z–Mod range, which combines the
finish. expertise of ADK’s custom shop and their 3 Zigma brand. These Z–Mod microphones are assembled by hand in America
Impressive range of decent using premium components and are, in effect, customised versions of the original ADK TT microphones.
accessories.
cons As with the T–FET range, the Z–Mod collection comprises five mics that differ only in their body colour, capsule design and
Cost. output transformer (Sowter, Lundahl or Jensen, depending on the specific model). The Z49’s siblings include the red Z12,
Loose XLR bolts on review white Z47, blue Z67 and yellow–green Z251, which reflect the sonics of AKG’s C12, Neumann’s U47 and U67, and
model. Telefunken’s ELA M251, respectively. The Z49 is, logically enough, intended to honour the sound character of the Neumann
summary M49.
A high–quality large–
diaphragm valve In The Box
microphone which uses
modern design principles
The Z49 ‘Black Beauty’ microphone is supplied in a large aluminium briefcase with a comprehensive range of useful
and construction methods to accessories. A grey Hammerite–finished power supply unit features a chunky power on/off toggle switch and large lamp bezel,
approach the sound with a mains voltage selector for 115 or 230 V AC supplies and an IEC inlet socket. A winged knob on the opposite end of the
character of the classic unit selects the polar pattern in nine switch steps between omni and figure of eight, with cardioid at the mid–point. The
Neumann M49 while microphone connects to the power supply via a seven–pin XLR, while the audio output is presented on a three–pin XLR.
maintaining modern Unfortunately, on unpacking the power supply of the review unit, I noticed something rattling inside and discovered that all the
convenience and
bolts securing both XLRs were very loose, with one set having fallen out completely, allowing the nut and washers to float
practicality.
around inside the PSU. The bolt was lying in the case. Perhaps some thread–lock compound might be a useful addition during
information assembly!
$3599
ADK Microphones +1
503 296 9400
www.adkmic.com
A nice wooden box with fabric–lined foam padding protects the mic itself, which
measures 53 x 225 mm and weighs 750g. A basic screw–on hard–ring stand mount is
provided along with a large cat’s–cradle shockmount, and both a simple foam wind
gag and a metal–mesh fabric pop shield on a long gooseneck clamp are also
included. Premium silver Accusound HD7 seven–pin mic (5m) and three–pin (3.5m)
output cables are supplied, too. This is a generous collection of useful, good–quality
accessories.
A separate hardwood box with a foam base contains three brand new Russian–
made alternative valves as replacements for the selected tube installed in the mic
itself. There are no ‘new old stock’ (NOS) treasures here, but the supplied alternatives
allow some useful tonal experimentation. The review model was fitted with a modern
Russian ‘Mullard’ ECC83 (12AX7), and an identical spare was included in the ‘options’
box alongside a Tung Sol 12AX7 and a JJ Electronics ECC803. The ‘Mullard’ valves
are typical European ‘long–plate’ designs, while the JJ Electronics ECC803 features
an even longer anode plate with a gold–plated grid, which is claimed to give a thick
and harmonically complex mid-range. The Tung Sol tube is a remake of the early
American 12AX7 variant, with a much shorter plate, and is claimed to have a clear,
open high end with a warm, rounded mid-range. The aim, obviously, is for the user to
select different valves to provide the required tonality. Changing valves is quick and
easy, and only involves unscrewing the base of the mic, sliding the brass barrel off,
and slipping the valve out of its housing (with the power off and the valve cooled down, of course!).
The simple impedance converter electronics are housed on a printed circuit board mounted on one side of a metal frame
that extends between the capsule and XLR base. Full–size components are used throughout. A second circuit board on the
reverse of the frame has a cut–out to accommodate the valve, and a Lundahl LL1530 output transformer is fixed to the base
of the frame.
The capsule, fixed to the top of the frame, is protected within a dual–layer chrome mesh grille and is the same GK49D
capsule as found in ADK’s Frankfurt 49T. It features two 27.5mm–diameter gold–sputtered mylar diaphragms, each five
microns thick and centre–terminated. The overall capsule housing measures 24mm in diameter and the design is loosely
based on Neumann’s K47 capsule element, but it is voiced specifically to deliver an M49 sound flavour.
The technical specifications are much as would be expected for this kind of mic, with a sensitivity of 17.8mV/Pa (–32.8dBu)
and self–noise of 16dB (A–weighted). The maximum SPL is 129dB (for one percent THD) and, as there is no pre–attenuation
option on the mic, the Z49 may not be the ideal choice for close–miking particularly loud sources.
Listening
Like the Neumann M49, ADK’s Z49 changes in sound character audibly with different polar pattern selections, particularly in
terms of its high–end presence peaks — and this provides an additional option for tuning its tonality in different situations. I
wasn’t able to compare the Z49 with an original Neumann, but even if I had there were so many M49 variants over the years,
and so many have deteriorated away from the original sound anyway, that such comparisons would be largely pointless. The
important question is not whether the Z49 “sounds like an M49”, but whether it features those sonic characteristics that made
the M49 a popular and useful studio tool — and I’d say the answer to that is an emphatic yes.
The Z49 is primarily intended as a microphone for lead vocals, which it handles with great aplomb — especially male
vocals. It has masses of character, with a full–bodied bottom–end bloom and a rich and slightly forward–sounding mid-range.
The top of the voice range benefits from a slight presence lift which aids diction and clarity, but above that the response falls Audio-Technica
quite quickly to give a restrained or ‘dark–sounding’ character. In that respect it shares some characteristics with a vintage AT4047 MP
Multi-pattern
ribbon, as well as the classic Neumann M49 which it is trying broadly to emulate.
Condenser
Microphone
The Z49 is not technically as quiet as a modern solid–state mic, but it will not be perceived in any way as noisy in normal
Audio-
applications, and its relatively high output level won’t stress any preamp unduly, either. Different valves affect the tonality,
Technica
slightly altering the character, mainly in the mid–range richness and high end — but this really is pretty subtle in normal use have added
and I was quite happy to use the standard–fit Sovtek ‘Mullard’ for most of my review trials. However, the character of different multiple
valves does tend to be revealed more clearly when the incident sound level is very high, since that pushes the valve more polar patterns to one of
towards saturation and its distortion characteristics come into play. their already successful
designs, bringing
Electric guitar amps, solo strings (especially cello) and wind instruments like saxophones all come across well on the Z49, increased versatility in
and I found I generally favoured the character of the JJ Electronics valve for these louder sources. I didn’t have the the studio.
opportunity to try it on orchestral brass, but I would imagine it would fare well there too, provided the sound level didn’t push
Audio-Technica
the SPL limit too hard.
AT4047 MP |
The ADK Z49 is a pretty expensive microphone by any standards, but this is reflected fairly in the build quality, the supplied Media
accessories and — most importantly — in the sound quality. The mic feels solid and reliable, and the gloss–black paintwork Multi-pattern
Condenser
looks classy, combining vintage style and modern presence. At this price a microphone has to be chosen only after extended
Microphone
personal audition, but the Z49 is certainly an impressive performer and would be well worth adding to one’s shortlist if the
Audio files to
tonal character descriptions appeal. . accompany the article.
Alternatives Audio-Technica
Boutique valve–based M49 microphones are plentiful, some being faithful replicas and others less precise homages. The AT4050 ST
current Neumann M149 must be on the list of comparable alternatives of course, along with the Flea 49, Peluso P49, Stereo Condenser
Microphone
Soundelux E49, and Wunder Audio CM49.
There's
more to this
Published in SOS October 2014 variation on
Audio-
Technica's flagship
microphone than the
simple addition of a
second capsule...
Peavey Studio
Pro M2
Condenser
Microphone
Paul White
explores the
capabilities
of the
understated-yet-
powerful Studio Pro M2.
Schoeps VSR5
Microphone Preamp
Schoeps
make some
of the most
revered mics
on the planet, so when
they release a
commercial version of
the mic preamp they
use for testing, you have
to take it seriously...
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In this article:
It Takes Qu To Tango
Allen & Heath Qu24
Three Layer Game Digital Mixing Console Buy PDF
Screen Play Published in SOS October 2014

Reviews : Mixer
Effects
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Join The Qu Allen & Heath’s Qu series goes from strength to strength,
Alternatives with the latest offering more I/O, more faders and more
The Capture Effect mix groups.
Paddy Power
Mike Crofts
Going Soft
T
How Solo Can You Go? he Qu24 is a 32–input console with a fader count of 25 (including one fixed master). There are 24 mono inputs on the
Allen & Heath back panel with on–board preamps, plus a further three stereo input pairs and a stereo USB playback facility. Despite
its compact size the Qu24 manages to house a full local complement of inputs and outputs on the back panel (there are
Qu24 $2799
no fewer than 24 analogue outputs on this thing!) making it a direct drop–in replacement for an existing analogue mixer if
pros
desired. A comprehensive complement of DSP including EQ, dynamics and effects is controlled by a combination of top-panel
Compact, neat and very controls and the LCD touchscreen, and mix control is enhanced, compared to the smaller Qu16, by the addition of sub–groups
functional mixer.
and a matrix facility. The Qu24 offers moving faders for total recall, four effects processors, USB audio streaming of 32 tracks
High-quality audio with great
effects. direct to PC- or Mac–based DAWs, an integrated direct–to–drive 18–track QuDrive recorder, and it is fully compatible with
Flexible audio streaming Allen & Heath’s dSnake and ME1 personal monitor system, both of which can be programmed from the desk . As if all this
and control options. isn’t enough, you can also control all the mix parameters over a Wi-Fi connection using the QuPad iPad app and a wireless
Fast, easy and largely router.
intuitive operation.
Great entry product for It Takes Qu To Tango
digital mixing.
cons When the Qu24 arrived I was surprised at how small the box was, and when unwrapped it was apparent that a lot of thought
No analogue inserts. has gone into producing a functional but very compact unit. It looks great, with its all–black surface, nice obvious control
Surface labelling difficult to blocks and clearly grouped functions. There’s no element of gimmickry or flashiness in the external design, and the console
read in dark venues. has a very purposeful ‘pro’ look to it that I find ergonomically very pleasing and effective. Those more used to their mixer
summary lighting up like a Christmas tree may initially wonder where all the bells and whistles are, but when you get around the surface
An elegant, well made and everything you need is there, and the uncluttered layout is helped by appropriate and intelligent use of the colour touchscreen.
powerful mixer that is easy The printed labelling is on the small side, but it’s clear enough and the use of white text on a blue background stands out well
to use, and would make an from the black surface, even under coloured lights. The Qu24 has a strong all–steel casing that is unusual in that it looks, from
excellent replacement for an the side, like a very thin mixer that has been bent around into a three–sided shape with the connectors at the rear. The space
existing analogue mixer.
directly below the control surface is consequently empty — there are no side panels so there’s no need for cooling fans as the
information
air can circulate freely underneath. However, this space is actually really useful for keeping your tape, Sharpies and lunch
items out of sight. The casing is very well made, as with every other A&H product I’ve come across, and I can personally
$2799
endorse the makers’ claim that it’s strong enough for a large person to stand on without causing any damage. I mean, they
American Music &
Sound +1 800 431 2609 can’t go around saying things like that without expecting someone to actually try it, can they?
info@americanmusicandsound.com
www.americanmusicandsound.com
www.allen–heath.com
When fired up, the Qu24 was ready for action very quickly
(about 10 seconds) and the surface layout and I/O connections
are so obvious that you can get your hands on those sexy
motorised faders straight away.
Three Layer Game

The faders operate in two factory–preset layers; one for the
mono input channels and one for stereo inputs, effects returns
and mix buses, with orange or blue LEDs telling you whether the
fader is controlling a channel or a bus. There is also a third
custom layer available, which contains a convenient pick–and–
mix combination of inputs and buses configured for any
particular user and/or live application. Setting up a custom layer
is very easy using the touchscreen, by choosing Setup / Control /
Custom Layer and simply touching one of the 24 fader slots on screen, then using the rotary encoder to dial up what will be
assigned to that fader slot. As far as I can see, anything that can be controlled by a fader can be assigned to any fader in the
custom layer, and multiple assignments are allowed — you can set them all to control channel 1 if you want to, and all the
faders will move together! More sensibly, there’s a useful incremental facility where the next channel or bus in ascending order
is automatically assigned to the next fader in sequence, by repeatedly tapping the ‘plus 1’ button on screen, saving the trouble
of dialling them all up one–by–one. I really value any kind of user–customisation, especially when it comes to fader
assignments as I (like everyone else, I guess) have my own ideas about what I want on my main fader layer during a
performance. The Qu24 even allows a ‘custom layer only’ setting where, for example, a guest user will only be able to control
what’s been assigned within this layer and cannot get at anything else.
Upon pressing the appropriate button (or both buttons together for the custom layer), the faders fly quickly, smoothly and
pretty quietly to their new positions — they’re not silent but they don’t chatter or clatter like some I’ve used. As the Qu24 has
24 faders and 24 mono inputs, there’s no need for input patching, so input 1 is controlled by fader 1 and so on. I took a little
while getting used to the ‘second’ layer as I’d have liked the mix bus masters to start at the left–hand side, or at least right
after the stereo inputs and before the effects sends, where I could locate, say, stage monitor feeds in a hurry. As the fader
assignment for the two default layers is fixed there’s no need for an illuminated ident strip as with the more upmarket GLD
series, consequently getting your finger to the correct fader quickly means you have to look where you’re aiming until you get
used to the surface. There is space for tape or magnetic labels if needed.
Although most controls are ‘shared’ and accessed by one channel at a time, as is standard practice with small–format digital
mixers, each channel fader does have mute, select and P/AFL buttons, and a small three–segment signal level meter, which is
useful for a quick glance at the surface to check that nothing’s in danger of clipping and that the expected incoming sources
are in fact present at the board.
Screen Play
To access the huge range of processing functions available within the Qu24, a single set of hardware controls and the
touchscreen display are laid out across the upper part of the surface. This is called the SuperStrip and it controls only the
currently selected channel or bus (whichever one has focus), with the touchscreen and its associated access buttons also
used for configuration, metering and other control options. If the fader layers are like the Qu24’s flight deck then the
SuperStrip is the engine room, where all detailed adjustment is made and processing functions are controlled. There are
dedicated controls for the main parameters — preamp level, high–pass filter frequency, four–band parametric EQ, dynamics
and pan — and a greater degree of control over individual parameters is possible via the touchscreen. One of the most useful FOR SALE: MIXERS in SOS
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features of the touchscreen control is the little tab which appears in the lower left corner, and allows access to whatever
further level of control is appropriate to the currently displayed process, changing its function according to what information is GLOSSARY: technical terms
on screen at the time. Take the preamp adjustment, for example: touching the preamp part of the main channel screen brings explained
up another screen that allows adjustment of things like phantom power, polarity and channel linking. The little tab at the
bottom now says Source, and touching this brings up a really easy–to–follow screen where the channel source can be chosen
Competitions!
as local input, dSnake or USB, and the individual channel assigned. Very neat, very clever and, although you are stepping
through a couple of screen pages, it doesn’t have a menu–like feel and manages to provide low–level access that is obvious Win Dangerous Music D-
and very nicely displayed. Only ‘legal’ options are ever presented to the user, so there’s no danger of wreaking havoc by Box (Americas Only)
making unfortunate choices. Win Adam A7X monitors
& Sub8 (UK, EU ROW
Only)
There’s a lot of precise control available in the SuperStrip, and

I think the feature set within the processing section has
everything you are likely to need from a console of this type. The
dynamics and EQ work very smoothly and with an excellent
correlation between the on–screen display and what you hear,
which I think is an important point as this can give confidence to
the user, who may under certain conditions need to trust the
display. The dynamics section is very comprehensive for a small
The Qu24’s vast array of local I/O makes it a suitable ‘drop–
console, with four different manual and automatic operating in’ replacement for an existing analogue mixer.
modes. The channel processing settings can be stored in
internal libraries, either as separate presets for graphic EQ, parametric EQ and so on, or as a combined channel preset. There
are some factory presets available, which can be useful as a place to begin, but the real value of this function is the ability to
store, name and recall any particularly effective settings that may be needed again in future. Overall I found the level of
control, the ease of access and the audible results very rewarding, with no apparent quirks or problems. The way in which
different EQ designs work is obviously a matter of personal taste, but I’d be very surprised if anyone can find much not to like
about the processing within the Qu24.
Effects
The integrated effects are inherited directly from the Allen & Heath high–end iLive consoles and emulate various industry
standard processors. Four stereo effects processors can be loaded into a virtual effects rack; a library of effects and presets is
available for loading into the four rack slots, and levels are controlled using a ‘sends on faders’ approach. The most commonly
used effect for live sound will be reverb, and there is a good range of ready–made settings available in the library list. Basic
parameters are accessed directly from the effects screen, and there’s an ‘expert’ button which allows detailed editing. The
effects return channels have a four–band parametric EQ available for further control. A nice touch with all the effects is the
graphical display, which makes different processors and settings more memorable, and there’s a very neat button which flips
the graphic between front and rear panel views — basically the processing is controlled from the front and patching the effect
is accomplished from the back panel. I found something in the preset library for every situation I encountered, and I really liked
the quality of the reverbs — and made use of the classic Symphonic Chorus a couple of times too! I got the feeling that the
effects section within the Qu24 is a bit of a hidden treasure just waiting to be explored and exploited...
One of the main attractions of going digital is the ability to store and recall mixer settings, either for individual performances
of bands and venues, or for a number of different scenes within a live show. The scene memory within the Qu24 has 100
available slots, which can store all the mixer operating settings, or selectively bypass certain parameters if needed — for
example you may want to capture and recall different fader positions but you may not want to revert to an earlier master fader
setting, so it’s essential to be able to exclude certain elements from your scene recall.
Join The Qu
For me, the Qu24 is just about ideal in every respect as a great little workhorse console with a big enough feature set to cover
most small–to–medium live work. About the only thing I could find missing was analogue inserts, which I myself hardly ever
require, but a much–loved outboard unit can always be connected using the analogue outs and returned to, say, one of the
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£516,050
There’s a lot to the Qu24, and what I like most is that it works really well on a number of different levels: if all you need is
basic mixing, you can just plug it in and get going with only a few minutes of getting used to the surface, whereas if you need
a lot more than this it’s all there waiting to be used. The Qu24 is a go–anywhere solution for live sound and I’d be more than
happy to own one for my work. On my wish list would be a little bit more user–customisation for the mix-bus fader layer, and
maybe a slightly wider space for the custom layer scribble strip. Apart from that, I love it! .
Alternatives
Check out also the consoles in the Soundcraft Si Expression, PreSonus StudioLive and Behringer X32 ranges, as well as
the Tascam DM3200 and Yamaha’s 01V96i.
The Capture Effect

The ability to make live recordings at a gig is a great additional feature, and the Qu24 allows direct multitrack streaming
through the USB port to any suitable DAW application — and it’s bi–directional, so multitrack returns from the DAW can
also be routed back into the live mix. If this isn’t enough, there’s also a built–in QuDrive recorder function, controlled from
the touchscreen, which captures multi–channel audio direct to a USB hard drive. In other words, the Qu24 is also a
complete multitrack recorder needing only an external drive to make it fully functional.
Paddy Power
It’s becoming very cool to run remote control of lighting and audio consoles, especially in theatre venues, and the Qu24
lets you do this by using a free iPad app called QuPad. With a wireless router plugged into your Qu24 you can enjoy
remote access to all the live mix parameters, and the surface controls can be used at the same time, so one person could
be stage side looking after foldback levels, while the operator at the board works the front of house. Even if you don’t
really need the remote capability, connecting an iPad will give you a larger screen display.
Extensive wireless control of the Qu24 is possible via the

QuPad iOS app and a Wi-Fi router.
Going Soft
The Qu24’s soft keys are very useful and can be programmed to perform various functions — for example, I used one to
act as a kill switch for the effects sends by setting up a mute group for this purpose and assigning a soft key to it. The
Qu24 has no fewer than 10 of these keys so all those often–used functions should be well covered.
How Solo Can You Go?

It’s possible to customise the way in which P/AFL operates in a number of useful ways, including a very handy ‘LR to
P/AFL’ function, which routes the main left/right mix to the monitor bus if there is no individual channel or bus P/AFL switch
active. This is useful for identifying and isolating individual sounds, for example if you can hear something in the mix but
you’re not sure where it’s coming from, or you want to compare a single pre–fader source with how it sounds in the mix
using headphones.
Published in SOS October 2014

Reviews Information
SOS Site Analytics
In this article:
Mel Lab Cel4 & TX3
App Works
EasyBeats LE Making Music On The Move Buy PDF

Reviews : Software: ALL
Jon Burton
Mel Lab Cel4 & TX3
Reference Microphone
I
do like a gadget, and at a show in Rome last week something shiny caught my eye. The sound system engineer had a
tablet PC, and from the side was poking a small microphone the size of my little finger. It turned out to be one of the Mel
Lab reference microphones, the Cel4. This small Italian company is building a good word-of-mouth reputation for their
various reference microphones, but this was the first I had seen of their miniature series.
The Cel4 is a reference microphone with a built-in 3.5mm four-pole connector. It has been optimised for use with laptops,
tablets and similar devices. The mic comes in a watertight steel tube designed to keep it safe in your bag or even your pocket,
and its stainless steel body has been treated to make it corrosion proof. The microphone and container weigh a mere 40g and
the tube is only 8cm long. The plug is screwed into the body, so it’s cheap and easy to replace if you happen to sit on it!
The response of the microphone has been made practical rather than totally flat. The low end has been rolled off to avoid
saturating any hand-held devices it may be used with; the rest of the microphone’s spectrum is surprisingly flat. The Cel4 is
not intended for lab work, but rather for everyday use, taking measurements in the field.
The web site quotes a very respectable 20 to 20000 Hz (-3/+1.5dB), with the
range only being +/- 1dB above 100Hz due to the deliberate bass roll-off. With less
than 3 percent total harmonic distortion at 115dB this microphone is made for
music.
Mel Labs make a similar-sized microphone fitted with a TA4F connector, the TX3.
This can be used directly with many wireless packs, a handy tool if you are making
lots of remote measurements. You can also use it with a cable and readily available
power supply. Between this pair of microphones most engineers can find a handy
pocket reference microphone for their kit bag, and I think at this price that’s
something of a no-brainer. Highly recommended! Jon Burton
Cel4 €100, TX3 €100.

www.mellab.it
EasyBeats LE
Beat Creation App For iOS
E
asyBeats LE is the free version of EasyBeats and is compatible with all
iOS devices running iOS 6.1 or later. I tested it on an iPad 3 and it worked The Mel Lab Cel4 alongside its 8cm
flawlessly. Being free it has certain limitations, but still does everything you carry case.
need to get a taste for what the full version can do. Essentially EasyBeats LE
combines a set of 808-inspired electronic drum sounds with the ability to record a sequence of up to four measures (16 steps
per measure), either directly by tapping an MPC-style pad page or via a familiar drum grid. This free version only includes one
kit of sounds and you can’t chain patterns together to create a complex arrangement as you can in the full version, but it is still
pretty flexible. It shouldn’t take you long to decide that you need the full version, and why not as it costs peanuts?
The grid and pad entry systems can be used as necessary to shape your
pattern. Both have the ability to add or remove notes, giving you the ability to
combine real-time and grid programming. There’s a variable level metronome
that automatically quantises your hits on the pads. The tempo and amount of
swing of the pattern can be varied, as can the pitch and velocity of individual
steps. There’s also a neat effects section where any combination of high-pass
filter, reverb and distortion can be applied and controlled by running your finger
across a virtual XY pad to control the effect amounts in real time. These effects
actually sound really good and can turn a sterile loop into something with real
character. Your XY movements may be recorded so that they play back along
with the pattern, so dynamically changing sounds are easy to achieve.
If you want to be able to export your finished beats you’ll need to splash out
$4.99 for the full version, which also allows you to create arrangements from
up to 16 patterns with quantisation of up to 64 steps per measure. If you need
additional kits to use with either version you can purchase them from within the
app (though the paid version includes some extra kits anyway) while the paid
version also allows you to import your own sound samples, to set other time
signatures and to save your work as exportable WAV files.
EasyBeats certainly lives up to its name in making it simple to create some EasyBeats familiar MPC-style pad screen.
very usable rhythmic loops, the sounds of which are easily good enough to use
in a serious composition. The bottom line is that if you like tinkering with old-school drum machines, you’ll love this. If you’re
too mean to buy the full version you could record your endeavours via the headphone jack, but given that the full version of
the app costs less than a beer, there’s really no reason not to buy it. Paul White
In this article:
Overview
Arturia Vox Continental V
Extra Continental Software Instrument Buy PDF
The Effects Published in SOS October 2014

Reviews : Software: Instruments
Thumbs Up?
Vox Continental Arturia have pointed their considerable modelling
Models
experience at recreating a ‘60s classic.
Extended Mode
Gordon Reid
Arturia
I
Vox Continental V $99 f I’m honest, I’ve never fully understood the appeal of Vox Continentals. Coming to prominence in the second half of the
1960s, their brash, reedy sounds neither emulated the tonewheel organs of the era, nor were they as suitable for
pros
psychedelia as the Farfisa Compact models. Nevertheless, they’re often ranked as the most important and influential of all
It’s simple, it’s flexible and it
sounds like a Vox.
the combo organs, and when a family of instruments is adopted (no matter how briefly) by the likes of the Beatles, the
If you ask nicely, it also Monkees, the Animals, Terry Riley, the Doors and the Grateful Dead, you’ve got to respect its heritage.
sounds like a Jennings.
It weighs nothing, and the Unfortunately, many surviving Voxes have been poorly (if ever) maintained and, as a consequence, they can be unreliable
keys trigger every time. beasties. If I leave mine untouched for a few months, it becomes prone to mistriggering, with intermittent notes, a scratchy
cons key–on noise, and a high probability that assorted harmonics on various notes will go walkabout. Playing it then becomes an
It’s a shame that Arturia exercise in avoiding the notes that go ‘peep’ instead of ‘parp’. What’s more, the tone generators have a tendency to stop
removed the pedalboard dividing, so that lower octaves of a given note (and their related contributions as harmonics) can also disappear into the
used on Wurlitzer V. aether. If you’re lucky, these are not difficult repairs but, for serious players, maintenance is vital if a ‘Connie’ is to remain
The Leslie effect doesn’t reliable and consistent over long periods. Clearly, in this era of software synths, organs and pianos, there’s a place for a
pass muster. modelled Vox Continental. The only question should be, ‘does it do the job?’.
There are a couple of small
bugs to be swatted. Overview
summary
One of two things is Installing Continental V was a doddle and, because it’s a model rather than a sample library, it takes up less space and loads
happening. Either I’m much more quickly than you might imagine. Having got it up and running in a handful of minutes, I connected two Arturia AE
becoming less grumpy in my USB controllers to my Mac and directed these to the upper and lower manuals, and was ready to start. The next thing to do
old age (unlikely) or soft was to link the physical knobs and sliders on the keyboards to the controls on the GUI and save this as a configuration. You
synths and other ‘soft’ can assign a MIDI CC to every control within Continental V, and the software not only allows you to constrain the MIDI CC
instruments are becoming
values to sensible ranges, but also to reverse the effect of your controllers’ sliders so that they act as drawbars (toward you to
damn good. In the past
couple of years I’ve
increase values, away from you to decrease them) rather than in the normal fashion. Arturia’s standard preset management
threatened to retire my system is retained, so it’s easy to create, save and recall your own setups and registrations, even down to the colour of the
Wurlitzer EP200, Philicorda Tolex for a given preset.
and Clavinet in favour of
software instruments, and
now I have to add my Vox
Continental and Jennings to
the list. Continental V isn’t
always indistinguishable
from its inspirations, but I’m
not sure that I care enough
about the differences to
continue maintaining and
carrying the originals
around.
information
$99.
info@arturia.com
www.arturia.com
Unlike some hardware emulations of recent years, the GUI for
Test Spec Continental V follows the form and function of the original
instrument (in this case, a Continental 300) closely. The original
MacBook Pro 2.5GHz Core
i7, 16GB RAM. OS 10.7.5. offered drawbars for 16’, 8’, 4’, and mixtures II and III on its
Continental V Version 1.0.0 upper manual, drawbars for 8’, 4’, 2’ and mixture IV on its lower
(64–bit). manual, drawbars for 8’, 16’ and a release option for the bass,
Digital Performer 7.24. and additional drawbars that controlled the contributions of the
Plogue Bidule 0.9726 and Flute (sine) and Reed (filtered square — essentially, triangle)
0.9737. wave generators for each. Upper manual ‘percussion’ was also
available, and all of this appears to be recreated correctly.
Encouraged by the accuracy of the GUI, I compared the sound

of the software’s raw waveforms (8’ drawbar, no effects) with
those generated by my Continental II, and found the two to be
only slightly different, perhaps due to degeneration in the
components in my old beastie. A more significant difference lay
in the nature of the V’s vibrato. This is an important element in
the ‘Vox sound’ and Continental V’s was initially shallower and
slower than the original’s. In short, my Connie sounded brighter,
brasher, and had an altogether stroppier attitude than Continental V.
Happily, a solution was close to hand. Opening the top panel of the GUI revealed the ‘tone generators’, which allowed me to
dial in all manner of tweaks, including detuning of the individual divide–down oscillators, adding background noise, and
causing notes (or footages within registrations) to mistrigger to a greater or lesser degree. There are also controls to adjust
the vibrato speed and depth to emulate the aggressive effect on the original. I spent a few minutes adjusting all of these and
found that a bunch of noisy, unreliable and mistuned notes that wobbled to an unpleasant degree was sure as heck starting to
sound like a bag of bolts that’s approaching its 50th birthday. If I have to find a fault (and there is a small one) it’s that, while a
key is depressed, there’s a low–level but permanently sustained note when using the percussion, even when no drawbars are
extended. Perhaps this was a characteristic of the Continental on which the software was based.
Extra Continental
Expanding beyond its emulation of the original organ, Continental V also offers an Extended Mode that inserts additional
controls into the GUI and adds a third waveform option to the sonic palette. (See the ‘Extended Mode’ box.) Called ‘S’ (for
‘String’) the new waveform is described as a triangle wave, but it’s much brighter than that, and an oscilloscope revealed that,
at low frequencies, it’s closer to a ramp wave, which has the same harmonic content as a sawtooth wave. This is as it should
be: you almost always start with a sawtooth wave to create string sounds.
Now, forgive me if I stray from the main narrative for a moment. Following long–term financial difficulties and a buyout in
1967, Tom Jennings (the founder of Vox and the co–designer of the Continental organs) was laid off by his own company, so
he founded Jennings Electronic Instruments (JEI) and, over the next few years, manufactured a small range of organs in
direct competition with his previous employers. In large part, these were based on his Vox instruments, but with more
drawbars on the upper manual (identical, in fact, to the Extended Mode on Continental V) as well as some presets. The J70
was the dual–manual model, but the pick of the bunch was the rare, triple–manual J71, which added a bunch of confusing
mixtures, sustains and tone controls for its new upper (as opposed to the now middle) manual.
Given the speed of development in the late ’60s, it should come as no surprise to learn that there were some underlying
electronic differences between the Continentals and the ‘J’-series organs so, while it seems that Jennings was aiming at the GLOSSARY: technical terms
same sound as before, his new instruments had a slightly different character. To my surprise, Arturia have emulated this by explained
providing a knob that allows you to switch between two sound ‘engines’ — Vox and Jennings — either of which can be
invoked in standard and extended modes. So, while I still had the Continental II and Continental V set up next to each other, I Competitions!
brought in my J71 on this for further comparison.
Despite the differences between the two organs (for example, the choice of harmonics comprising the mixture IV drawbar Box (Americas Only)
on their lower manuals is different) there was more than enough that’s common between them to demonstrate that the Win Adam A7X monitors
Jennings tends to sound a little thinner and harsher than the Vox. For some reason, I had expected the Jennings engine in & Sub8 (UK, EU ROW
Only)
Continental V to be less accurate than the Vox engine but, tonally, it was definitely in the right ballpark, especially at low
amplitudes. Nonetheless, there are two glaring differences between the original and the emulation. Firstly, neither the standard
nor extended modes accurately reflect the voicing of the J–series, although the latter is correct for the upper manual. (Actually,
it’s correct for the middle manual of a J71, but let’s not be pedantic about such things.) Secondly, when I pull out lots of
drawbars on my Jennings it ‘crunches’, while Continental V remains clean and smooth at all levels. This may be in the nature
of the original, or it may be a fault that’s existed for so long that I’ve forgotten how it once sounded, so I’ll say no more about it.
There’s one other difference worth noting. When using the Vox engine, none of the tone drawbars (Flute, Reed and String)
need to be extended to hear the percussion; when using the Jennings model, one or more do, and the amplitude of the
percussion is proportional to the amount by which they’re extended. This means that you may not obtain the results that you
expect when switching between engines.
Given how much fun I was having playing the genuine Continental and J71 together, I next loaded two instances of
Continental V into Digital Performer to recreate the experience. Or, rather, I didn’t. Although my version of DP recognised the
plug–in and loaded both instances, it failed to display their GUIs. I then turned to Plague Bidule, which is a useful toolkit in
such situations, and loaded two instances of Continental V into this. This took just seconds, but I then lost an hour or two
playing with alternative tunings and different vibrato settings in each instance. This allowed me to create all manner of organic
and shimmering organs that have never existed in the physical world, and the results could be much classier than passing the
output of a single instance through a chorus unit.
The Effects
Ah yes... chorus units and other such things. Continental V offers a plethora of effects processors, although it’s not quite as
flexible in this area as I had expected. The first three are built into the organ itself. These are vibrato, tremolo, and an
emulation of three spring reverbs that Arturia call the Spring King, the RV–1 and the RV–2. Whatever their provenance, all
three work well, although they are more polite than the twangy slap of the genuine spring reverb in the J71.
Next, alongside the swell pedal within the GUI, there’s a single slot into which you can insert a ‘virtual stompbox’ of your
choosing: a flanger, a phaser, a stereo chorus, a delay with an integrated LFO for further modulation effects, an overdrive, and
a wah pedal. If these appear familiar, that’s not surprising — they’re a subset of the effects found in the Wurlitzer V software
instrument from the same company. But unlike Continental V, Wurlitzer V offers a ‘board’ with five slots, thus allowing you to
build the type of effects structures that you might have used with instruments such as combo organs or electro–mechanical
pianos in the 1970s. I have no idea why Arturia’s engineers removed this from Continental V.
At the end of the signal chain lie three additional options that allow you to direct the output, umm, well... directly to the
output, or through a rotary speaker effect, or through a guitar amp/cab model. I was relieved to see that, on the last of these,
all four microphone options and all four amp/cab options have been retained from Wurlitzer V. It was quite common for
Continentals to be amplified using cheap PA stacks or expensive guitar amps, and one played through a WEM EP40 and 2 x
12 sounds quite different from one played through a 100W Marshall stack.
If there’s a disappointment, it lies in the Leslie effect. This appears to be a dual horn/rotor emulation, but there’s just a single
rate control for both, and no way to control their accelerations. Consequently, its output sounds little different from one of the
baby, single–rotor Leslies. I hope that Arturia addresses this soon, because I made the same comment when I reviewed
Wurlitzer V in 2012 and, in my view, it’s even more important now that the effect is implemented within an organ emulation.
Thumbs Up?
If you’re an experienced Continental or J–series player you’ll find that, like the originals, Continental V has no trouble making
itself heard in a mix. Sure, it can be polite, but pull out the Reed and String drawbars and some of the mixtures, and it will cut
SOS Readers Ads through in classic Vox fashion. You’ll also be pleased to find that you’re no longer bumping your fingers against the drawbars.
GRAB A BARGAIN This was a flaw of many of Jennings’ designs, whereby the drawbars, when fully extended, protruded above the keys. If you
£516,050 knocked them while playing (and everyone did) the sound became quieter and changed tone, so that’s a significant bonus
of Second-User Gear for sale when using Continental V.
But let me summarise how I feel about Continental V as follows... During the course of this review, I liberated one rather
bulky and moderately heavy combo organ from storage, and then an even bulkier and ridiculously heavy combo organ from
storage, serviced the former (the Vox) because it had lost all of the Ds in its lower octaves, carried them up (and later down)
numerous flights of steps, and played them for a good few hours until all the notes and harmonics were triggering correctly.
Only then was I able to start reviewing and writing, so I was quite keen for Continental V to come up to scratch.
To be fair, it isn’t a perfect emulation of either of the organs against which I compared it, but I would have been gobsmacked
if it had been. Vintage (or, to put it another way, old and tired) electronics can interact in unexpected ways, and the differences
between the original organs and the software were smaller than I had expected. Sure, Continental V has a couple of bugs,
Arturia missed the fact that the Jennings’ mixture is voiced differently from the Vox’s, the Leslie effect doesn’t live up to the
standard of the rest of the package, and I was disappointed to find that it has lost Wurlitzer V’s five–slot pedalboard, but it
does the job it was designed to do and, although it doesn’t always have quite the bite of the original, I would be happy to
substitute it for the ‘real thing’ if there was ever a call to do so. All in all, I think that that’s a pretty significant ‘thumbs up’.
There’s just one remaining problem. Continentals look damn cool on stage, so who’s going to build a dedicated MIDI controller
for Continental V? .
Vox Continental Models

The Continental V manual mentions four Vox models: the original, single–manual Continental (sometimes called the
Continental I) first introduced in 1962, the dual–manual Super Continental, the Continental 300 (which added presets and
an internal spring reverb to the specification of the Super Continental and related models), and the Continental Baroque
with its integrated amp and speakers.
Nevertheless, there were others. Some, such as the single–manual Continental 301H made by the Thomas Organ
company in the USA, are easy to identify. (In this case because it used Thomas’s tone generators rather than Vox’s.)
Elsewhere, the names and dates of the models, as well as what was built where, can be less clear. In particular, the Super
Continental (which was the Italian version) boasted variants called the Continental II (the English version that I’ve used for
comparison in this review) and the Continental Super II, which seems to have been an English Continental II with
percussion. (So I guess that mine is actually a Super II, even through there’s no mention of it being Super on the organ
itself.) There were also two console models; the Home Club Continental for the domestic market, and the ‘split’ Continental
301 for gigging. And let’s not forget the Continental Futura, which was a hybrid of the Continental II and the (now super–
rare) Vox Riviera organ, and was built only to special order.
Extended Mode
When used in its standard mode, Continental V emulates the drawbars and footages of a Vox Continental II, Super II and
300. When used in its Extended mode, a more conventional set of footages is revealed, differing from the standard set of
Hammond B3/C3/A100 drawbars only by the omission of the 11/3’ footage. (See table.)
In addition to adding and reallocating drawbars, Extended mode adds the String waveform to all three manuals (upper,
lower and bass), adds a tremolo on/off switch to the control panel, and alters the percussion options.
Drawbar 1 2 3 4 5 6 7 8
Upper (Standard Mixture: 22/3’ + 2’

16’ 8’ 4’ Mixture: 51/3’ + 13/5’
mode) + 1’
Upper (Extended
16’ 8’ 51/3’ 4’ 22/3’ 2’ 13/5’ 1’
mode)
Lower (Standard Mixture: 22/3’ + 2’ +

16’ 8’ 4’
mode) 13/5’ + 1’
Lower (Extended
16’ 8’ 51/3’ 4’ 22/3’ 2’ 13/5’ 1’
mode)

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In this article:
Pass The Support
Audio Interfaces: The HDX Factor
Third Party Animal Spotlight Buy PDF
Star Of Avid Published in SOS October 2014

Reviews : Audio Interface
Links & Pricing
A
Avid HD I/O vid’s Pro Tools HDX, like its HD predecessor, has always been a fairly ‘closed’ system: it
Lynx Aurora 16 uses a proprietary plug–in format (AAX), requires an Avid PCIe or Thunderbolt card (either
Prism Sound Titan HDX or Native) to run, and is intended — by Avid, at least — to be used with the company’s own A–D/D–A converter
Apogee Symphony I/O interfaces. Most other DAWs will usually work with a variety of plug–in types (VST or AU being the commonest), while almost
Burl B80 all — including the non–HD version of Pro Tools — are quite happy to pipe audio through interfaces from almost any
SSL Delta Link MADI manufacturer, as long as ASIO or Core Audio drivers exist.
HD
iZ Technology ADA II This aspect of Pro Tools HDX has prompted some complaints, but it has the advantage of guaranteed performance and
DAD AX32 stability: a necessity for professional facilities, where a flaky system could spell disaster in terms of angry clients and lost
revenue. Because Avid only officially support their own interfaces (namely the HD Omni, HD I/O, HD MADI and Pre), they are
Mytek 8X192
able to focus all their testing and troubleshooting on just a few devices, unlike other DAWs, which have to be able to cope with
interface drivers that have been coded by different companies and with varying degrees of competence.
Pass The Support

The operative word there, though, is ‘officially’: there are now
many companies making converters and interfaces that, while
not explicitly sanctioned by Avid, either sport, or can be made to
sport (via expansion cards), an Avid DigiLink or Mini DigiLink
socket, for connection to an HDX or HD Native card. And
although such devices are officially ‘unsupported’ by Avid, third–
party interface manufacturers do perform rigorous testing
themselves, and almost always provide detailed information on
which systems and versions of Pro Tools they are known to work
well with.
The question remains, though: why would you opt for one of
these ‘unofficial’ interfaces when going down Avid’s preferred
route guarantees at least some recourse should tragedy befall
your HDX system? Although the DigiLink and DigiLink Mini connectors are a
proprietary Avid protocol, many third–party manufacturers
make interfaces that use them.
Third Party Animal
One reason is mentioned not once, but twice, in the name of this very magazine. People will almost always express a
preference for the sound of one converter over another, and while pretty much all HDX–compatible interfaces are of a high
standard — it is, after all, an expensive, ‘professional’ system — great import is often placed on the subtle differences in sonic
character imposed by different interfaces.
Given that they all perform well, though, perhaps a bigger reason lies in the differing feature sets. Some interfaces offer
more headphone outputs than others, say, while some offer a greater number of mic preamps (some, like those on the
Apogee Symphony I/O, are even remotely controllable).
Star Of Avid
One final issue to be aware of concerns latency. HDX systems are capable of running at extremely low latencies thanks to the
DSP cards they use, but an often–overlooked fact is that A–D and D–A converters impose their own latency, independently of
that caused by the combination of a computer and interface. Since Pro Tools HDX always assumes it is working with an Avid
converter, its automatic delay compensation is tuned to negate the specific amount of latency that is inherent to HD I/O, Omni
I/O and Pre interfaces. This could potentially cause problems when using a third–party interface, as the assumed delay
incurred by audio leaving through an analogue output and returning through an analogue input could well be different to the
reality (Lynx’s Aurora is the exception here, as it has been specifically designed to incur the same amount of latency as Avid’s
own interfaces). Don’t let that put you off too much, however: the very slight differences in latency between converters are only
ever likely to present a problem if you are wont to use complex routing of outboard hardware.
Interfaces For Avid Pro Tools HDXNeed some I/O for your Pro Tools HDX rig? Check out our round–up of current options.Avid
HD I/O
Along with the HD Omni and HD MADI, the HD I/O is one of the interfaces that comes as standard with a new Pro Tools HDX
setup. It’s available in three different I/O configurations, depending on what analogue and digital connectivity you need. The
8x8x8 version features eight line–level analogue I/O (on D–sub connectors), eight digital I/O via either AES3 or T/DIF, and a
further optical set of I/O that can accommodate either eight–channel ADAT signals or stereo S/PDIF.
Prism Sound Titan

Like the respected Orpheus from which this interface is derived, the Titan can connect to computers via USB or, using the
right MDIO expansion card, directly to Avid HDX DSP cards. The Titan’s front panel houses two independent headphone
outputs, two instrument inputs, a rotary encoder for I/O level control and comprehensive metering. The remaining I/O
comprises four mic preamps, four line inputs, eight line outs, TOSlink I/O for ADAT or S/PDIF signals and stereo digital AES3
I/O.
Lynx Aurora 16HD

The combination of a Lynx Aurora 16 and an LT HD expansion card provides up to 32 inputs and outputs in a 1U rack, making
it the most compact 32–channel HDX interface. Another unique selling point is that the Aurora 16 has been engineered to
incur exactly the same conversion latency as Avid HD I/O, which means that Pro Tools HDX’s automatic delay compensation
should work seamlessly, even when using outboard gear as software inserts.
SSL Delta Link MADI HD

As with all MADI products, the name of the game with SSL’s Delta Link is high channel counts — up to 64, in fact, from a
single unit, and all compatible with a large range of devices, from digital mixing consoles to broadcast multitrack recorders.
SSL claim that the Delta Link’s latency is only one sample for the inputs and two samples for the outputs, which equates to a
tiny 0.02 or 0.04 milliseconds. The two MADI ports can be set up to run in ‘mirrored’ mode, which is very useful for redundancy
or recording purposes.
Apogee Symphony I/O

Apogee’s highly configurable Symphony I/O offers direct connection to HDX cards, without requiring any expansion. Like
Avid’s own interfaces, the Symphony I/O is highly configurable, thanks to a range of I/O modules that provide various
combinations of analogue, ADAT and AES3 inputs and outputs. One of these modules includes eight microphone preamps,
SOS Readers Ads controllable via the Symphony software, that offer a substantial 85dB of gain.
GRAB A BARGAIN
£516,050
Burl B80
The Burl B80 ‘Mothership’ is a chunky modular rack into which can be loaded any combination of four–channel A–D (BAD4) or GLOSSARY: technical terms
eight–channel D–A (BDA8) converters, as well as the BCLK clocking module. Unlike most converters, the B80 aims not for explained
absolute fidelity, but tries to inject a bit of analogue-style warmth into digital recordings, thanks to its transformer–balanced
input modules.
Competitions!
iZ Technology ADA II Win Dangerous Music D-

Box (Americas Only)
The RADAR range of multi-track recorders has long been lauded for its ‘analogue-like’ sound, and the ADA II is iZ Win Adam A7X monitors
Technology’s way of offering the sound of RADAR’s converters to people who prefer recording to computers. It offers 24 & Sub8 (UK, EU ROW
analogue ins and outs, and comes as standard with a large, high-resolution touch-screen that provides access to metering, Only)
and configuration and routing settings.
DAD AX32
Able to handle signals up to a whopping 384kHz sample rate, the AX32 can be configured, in groups of eight channels, to
have either mic- or line-level analogue inputs, with up to 24 I/O in a single unit. When fitted with mic preamps, it can emulate
Avid’s Pre interface, meaning you can control the input gains directly from within the Pro Tools mixer.
Mytek 8X192
Designed for mastering applications, the 8X192 has eight analogue I/O and, unusually, a built-in summing mixer. Any of the
eight analogue outputs can be fed to it, either in mono or as stereo pairs. Other features include a high-current headphone
amp and an analogue monitor controller. The converter specs are also impressive: 120dB(A) for the A-D converter and
123dB(A) for the D-A.
Links & Pricing
Avid HD I/O
$2495 to $4995, depending on configuration.
Avid +1 650 731 6300
www.avid.com
Burl B80
B80 $1999, BAD4 $1499, BDA8 $1799.
Burl Audio +1 831 425 7501
www.burlaudio.com
Apogee Symphony I/O

Review: http://sosm.ag/sep11–apogee–symphony
$1995 to $4490, depending on configuration.

Apogee Electronics +1 310 584 9394
www.apogeedigital.com
iZ Technology ADA II
Review: http://sosm.ag/dec13-iztech
From $6580 to $12,050, depending on configuration. Pro Tools HD card $1995.

iZ Technology +1 800 776 1356
www.izcorp.com
DAD AX32
From $9507, depending on configuration.
Plus 24 +1 323 845 1171
www.plus24.net
www.digitalaudio.dk
Lynx Aurora 16
Review: http://sosm.ag/dec12–lynx
LT HD card $395, Aurora 16 (with card fitted) $3495.

Lynx Studio Technology +1 714 545 4700
www.lynxstudio.com
SSL Delta Link MADI HD

$1999.
www.solidstatelogic.com
Prism Sound Titan

Review: http://sosm.ag/sep14–prism
$4950 (HDX card yet to be released).

Prism Sound +1 973 983 9577
www.prismsound.com
Mytek 8X192
$3495. HDX DIO Card $495.
www.mytekdigital.com

Reviews Information
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In this article:
Drum & Drummer
Audio Spillage Drum Spillage 2
Algorithm Method Drum Synth Software For Mac OS Buy PDF
Spillage People Published in SOS October 2014

Reviews : Software: Instruments
Synthesis Algorithms
Audio Spillage Drum Spillage 2 lets you get down and dirty with
Drum Spillage 2 $129 synthesized drum sounds.
pros Jyoti Mishra
A
Immensely playable and
tweakable. udio Spillage’s Drum Spillage 2 is the (surprise!) second version of their Audio Units-format synthetic percussion plug-
Excellent sound quality. in for Mac OS X. Where it differs from the Stephen Slate Drums of this world is in its proud emphasis on synthesized
Very reasonably priced for drum sounds. If you’re after a vintage Linn, DMX or other sample-based beat machine, this is not the droid you’re
what you get. looking for. But if, like me, you miss Waldorf’s Attack and you like to roll your own sounds, Drum Spillage 2 is definitely of
cons interest. Remember, too, that a fair number of classic drum machines, like the TR808 and CR78, used totally synthesized
Preset kits come with non- sounds. Perhaps a drum synthesizer, albeit running as software rather than an unruly clump of wires, gets closer to the spirit
standard mapping. of these machines than static samples?
Potential CPU problems in
older Macs? Drum & Drummer
Could do with a few more
preset kits. The first thing that made me smile about Drum Spillage 2 is its aesthetic. Like Apple with iOS 7, Audio Spillage have
summary eschewed the dreaded skeuomorphic design that infects many plug-ins, and instead, present the user with a beautifully
Drum Spillage 2 is an spartan and utilitarian graphical interface. There are 16 pads arranged in a four-by-four grid. Hit some keys from C1 upwards
enormously capable and and waveforms pop up on the corresponding pad. It’s a wonderfully simple way of connecting real-world controller with virtual
flexible plug-in that doesn’t response, and it makes programming Drum Spillage 2 a doddle as you never get lost: you always see what is playing where.
sacrifice power for ease of Whilst this kind of interface has been done before, Audio Spillage’s implementation is leaner, cleaner and therefore more
use.
immediate than others.
information
The second smile was courtesy of the sound. It’s crisp, clean and punchy-sounding, very like something I’d spend hours
$129. patching up on my modular only to forget about and ruin the next day. I programmed some beats and was impressed by the
sales@audiospillage.com contrast between the traditional synthesized kicks and snares and the crazier gibberings of some of the pads. Cycling through
www.audiospillage.com the factory banks reveals all manner of zinging, whooping and metal-bashing.
Drum Spillage 2 arrives with 15 of these factory banks, which

are both eminently musical and experimental. ‘Electro’ is the
most ‘normal’ of the kits, coming closest to 808 territory.
However, even here, the programmers throw us some beautifully
mournful percussive tones that sound like goats lost in Scottish
fog. A lot of the kits seem to revel in wildly bizarre tones that leap
out of the speakers, but they are always part of an ensemble
which features usable bread-and-butter sounds too. Audio
Spillage are to be commended on their balance here: too wacky
and Drum Spillage 2 would be left languishing, never a go-to
plug-in; too staid and it would be out-competed by
aforementioned sample libraries.
Algorithm Method
The sounds themselves are produced by 12 different synthesis
algorithms, and are wonderfully tweakable via each pad’s
individual instrument editing screen. Again, Drum Spillage 2’s interface is paramount here: some of the algorithms are quite
complex, but at no time did I feel confused or swamped by parameters. Which is handy because even the simplest synth, the
cowbell, features four LFOs plus amp and filter envelopes, all of which can modulate each other and some of which are
sync’able to host tempo. Each model has a common bottom pane of LFOs, amp and filter section. At the top are individual
level and pan controls, along with Crush (sample rate) and Decimate (bit-reduction) controls with which you can turn even the
gentlest of sounds into outrageous noise. Lovely.
The more complex algorithms such as membrane, noise and FM offer even more possibilities. This makes Drum Spillage 2
tremendously powerful as a soft synth. I would say that even if you’re not in the market for a drum synth, if you’re a synthesis
nut you should buy this plug-in just for its sound-design possibilities. As an example, I took the boomy bass drum model,
extended the release, modulated it via all four LFOs (each modulated by the successive LFO), passed it through the
distortion, modulated the filter with the stepper, sent it through the resonator at really low frequencies and got instant Bebe
and Louis Barron. It was pure Krell evil. I headed back to the pad screen and clicked the little keyboard logo to the left of the
relevant pad and then the individual pad sound was mapped across my whole keyboard. No keygroups to wrangle with, no
squinting and cursing at tiny key numbers — one click and you’re good to go. Since Drum Spillage 2 lets you send 16
individual outs to your host, you can still stick your favourite reverbs or delays on anything you desire. This wonderful
tweakability and openness should be a model for all soft synths.
Any gripes I have with this plug-in are petty. Being grumpy and
old, I prefer my drum kits mapped in the traditional manner
beaten out by our ancestors around MIDI fires way back in the
dawn of time. If kick on C1 and snare on D1 was good enough
for Homo Erectus, why change it? And where are my goddam
hats? Sadly, I can’t moan on for too long because the way you
swap or copy sounds in Drum Spillage 2 is ridiculously easy: you
just drag and drop. And they give you an MPC pad mode. Dang
it. The CPU usage seemed a tad high occasionally, but even if I
mashed all the keys down at the same time, I couldn’t provoke
any audible glitches. Sometimes when I was trying to adjust GLOSSARY: technical terms
modulation via the little pink value strip, I’d click the bigger explained
parameter value next to it. Lastly, I would have liked more factory
kits, but 15 isn’t bad. Perhaps Audio Spillage can issue WIN Great Prizes in SOS
additional kits by celeb synth-botherers? I would love to hear Competitions!
what real patch fanatics could cook up. Win Dangerous Music D-
Each pad has its own synthesis screen, so you can tweak Box (Americas Only)
individual sounds to your heart’s content.
Spillage People Win Adam A7X monitors
& Sub8 (UK, EU ROW
In a fortnight or so of testing Drum Spillage 2, it hasn’t crashed once or misbehaved itself in any way. The best thing about it is Only)
how it actively seduces you away from its own presets — it’s so damn easy to fiddle with. It can deliver rich, evolving
complexities that would be impossible from sample-based drum plug-ins. This can be bad because you will inevitably end up
falling down a rabbit hole of increasingly insane synthesis, stacking up cross-modulated LFOs willy-nilly simply because you
can. The bottom line is this: if you want to escape the same old stock 909 snares and 808 bass drums that clog our airwaves,
if you want to actually explore percussive sound synthesis, this is definitely the plug-in for you. .
Synthesis Algorithms
SOS Readers Ads Bass Drum: The most 808ish, potentially boomy.
GRAB A BARGAIN Bass Drum 2: Thicker and more thowmping.
£516,050
Snare Drum: The most 808/606ish but very adjustable not to be.
now — don't miss out! Wood Drum: Quite boxy and cardboardy (in a good way).
Synth Clap: Very tweakable, I couldn’t quite get my fave 808 sound but very usable.
Cow Bell: Works as a cowbell and also a great synth for pseudo-gamelan.
Clave: Can be quite piercing.
Hi-hat: Can vary from smooth 808ness to grittier, lo-fi fare.
Cymbal: Great for any metal-bashing, not just cymbals (try with external gating).
Membrane: Reminds me of Logic’s Sculpture a lot. Can do convincing-if-quirky mallets.
Noise Drum: This actually features four noise sources in one synth and is immense.
FM Percussion: All the usual FM madness.

Reviews Information
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In this article:
Phoenix Rises Again
Crane Song & Dave Hill Plug-ins
Out Of The Ashes Tape, Vinyl & Distortion Plug-ins For Pro Tools Buy PDF
Peacock Published in SOS October 2014

Reviews : Software: Effects+Processors
A Colourful Display
RA When a top hardware expert turns his attention to plug-in
Logic Users Need Not design, the results are a bit special.
Apply
Jack Ruston
Cranes In Flight
D
Who Can Use These ave Hill is one of the most highly respected designers of professional audio recording equipment. His units sit
Plug-ins? comfortably in the world’s top studios, alongside the most revered outboard processors available (or not so available).
Crane Song & Dave Hill Whether they are marketed under the long-standing Crane Song name or the more recent Dave Hill Designs brand,
plug-ins $450/$499 the highest possible sonic standards are assured. Furthermore, these designs are not simply beautifully made clones, as
much high-end outboard can tend to be, but genuinely innovative pieces that often combine the best of analogue and digital
pros
technology. The HEDD two-channel converter, introduced in 1997, and the Spider eight-channel mixer and A-D converter
Unique without being
feature digital ‘colour’ circuits, providing a form of tape emulation. In combination with superb conversion, this ability to add
outlandish.
Exemplary sonics. harmonic colour has made the units indispensable to those who have access to them.
Encourage a musical rather
than technical approach. In 2001 Dave released Phoenix, a tape emulation plug-in for
the Pro Tools TDM platform, derived from the code in the HEDD.
cons
With Dave having designed the revered Aria electronics for
Pro Tools only.
Not cheap.
ATR’s tape machines, and with the HEDD algorithm sounding so
good, a lot was expected of Phoenix, and it did not disappoint.
summary
The plug-in has been heard on countless hit records since, and a
An outstanding set of tools
search through the archives of SOS’s Inside Track articles
for achieving analogue-style
colour within Pro Tools. reveals a long and distinguished list of advocates. One might be
Highly recommended. forgiven for asking “Well, back then there wasn’t much choice for
getting analogue-style colour in the box, and while it might have
information been great at the time, is it really still relevant 13 years on?”
Phoenix II and Peacock When you have mixers with effectively limitless budgets using it on current projects, the answer is an emphatic “yes”.
$450; RA $499.
Transaudio Group 702 While those original designs have maintained their reputation for sublime sonics, Dave has delivered new analogue
365 5155. designs, and has also ventured even further into the digital world. Phoenix has seen an update, becoming Phoenix II, the
www.transaudiogroup.com Dave Hill Designs RA plug-in offers a unique approach to second- and third-harmonic distortion, and the Crane Song Peacock
www.cranesong.com plug-in represents Dave’s take on the characteristics of vinyl. All three involve digital implementation of harmonic distortion,
and it’s worth mentioning that some of Dave’s current analogue designs, for example the Europa mic preamp, feature related
Test Spec (but not identical) processes operating purely in the analogue domain. Why is this important? Well, a common weakness in
Apple 2010 Mac Pro plug-ins that seek to create these sorts of distortions occurs when manufacturers are brilliant with code, but don’t fully
‘Westmere’ dual 2.4GHz 8- understand the analogue circuits they’re trying to emulate. What we tend to see is that those who have the deepest
core, with Avid HD Native understanding of analogue, and who have spent decades measuring analogue circuits on an oscilloscope, provide us with
card.
more analogue-like digital emulations — an obvious example being Universal Audio. There are few designers able to move as
Tested with Avid Pro Tools
11.
freely between these two worlds as Dave Hill, and it’s clear when we start to examine these plug-ins that they are the products
of not only a brilliant mind, but a great deal of experience.
Phoenix Rises Again

As the name suggests, Phoenix II is an updated version of the
original Phoenix plug-in. It now uses 32-bit floating-point maths
rather than the 48-bit fixed-point of the older Pro Tools TDM
system. It has a lower noise floor, an updated GUI, and some
functional enhancements in the form of an output trim and the
collation of the five different process types into a single plug-in —
the original Phoenix was actually a suite of five plug-ins.
So what exactly is Phoenix? Well, let’s start with what it isn’t. It isn’t one of those recreations of a specific tape machine or
tape formula. There are no perfectly modelled dancing VU meters, no comfortingly worn-looking Bakelite knobs, and no gently
spinning reels of tape to remind us what to hear. It was designed to give us the character of magnetic tape, to give us access
to the colour, but not the noise or generally unwanted side-effects. It provides the sort of saturation, frequency response and
compression that we normally associate with magnetic tape recording.
So if we don’t get variable wow, flutter and hiss, what do we get? Running from left to right there are knobs for Input Trim,
Process, Output Trim, Brightness and Type. Input Trim is fairly self-explanatory, allowing the user to boost or attenuate the
input signal by ±10dB. The Phoenix process is heavily dependent on the nature and level of the source, and this control allows
for both use and abuse in terms of how hard you hit the virtual tape, but most of the time it can just be left at unity unless the
source material is very hot. Next in line is the Process knob itself, which adjusts the amount of the distortion that is applied:
the distortion can be thought of as a side-chain process which is then summed with the main, untouched audio. Immediately
to the right of that is an Output Trim, offering ±6dB of adjustment to match the level of the processed audio with the original
source. This is followed by a three-way Brightness switch. The centre Gold position has an approximately flat frequency
response, with Opal being a little warmer and Sapphire a little brighter. Finally, and most importantly, the Type switch allows
selection of each of the five algorithms. These are described as follows:
“Luminescent is the most neutral-sounding process of the five.
Iridescent has a similar magnetic character, but with a fatter
bottom and mid-range. This [process] is the most similar to the
Tape knob on the HEDD-192. Radiant is characterised by a more
aggressive compression curve. Dark Essence is even more
aggressive — the effect is a colour with a wider frequency range.
When used on a vocal Dark Essence can reduce sibilance
problems by increasing the apparent loudness of the rest of the
signal. Luster starts more gently than the other four processes,
but becomes as aggressive as Dark Essence when the process is at full scale.”
The original Phoenix plug-in, while much lauded for its sound, was no looker, and Phoenix II is set to follow squarely in its
footsteps in both respects. It does look more refined than its predecessor, but since that had a distinctly ‘clip art’ vibe about it,
you should not expect the bleeding edge of graphic design. And I don’t care. I don’t care at all. Because for me, plug-in
graphics are like band names. If the name is bad, but the band is good, you somehow come to love the name because of
what it brings to mind. And this is much the same. All three of these plug-ins look frankly awful by modern standards, but when
I see them pop up on my display, I feel a little jolt of excitement and pleasure, because that’s what I get from listening to them.
I’m encouraged by the notion that more time and energy has gone into making them sound good, than making them look
good. Anyway, I’m giving the game away now
Out Of The Ashes GLOSSARY: technical terms

explained
In use, Phoenix II is the most straightforward of the three plug-ins. You wind in the Process control, switch between the
process types and tweak the Brightness switch until you find something you like. And it’s the process types that really make WIN Great Prizes in SOS
this plug-in what it is. I found myself going to Iridescent by default for most things, but when you experiment with Radiant, Competitions!
Luster and Dark Essence you can really change the sound a great deal. While Luminescent is subtle, and Iridescent quite
polite, the other three types begin to affect the character and colour of the mid-range and bottom end in increasingly obvious Box (Americas Only)
ways. You can really add guts in the low end with a good dose of Luster or Dark Essence. The frequency-response changes Win Adam A7X monitors
brought about by the Brightness control are beautifully judged. Opal just leans towards the low end and Sapphire the reverse, & Sub8 (UK, EU ROW
adding some sparkle and air without being too obvious. It’s the sort of thing that allows you to place sounds in a mix quickly Only)
and instinctively, darkening a synth or bass and lifting a snare or vocal. When you combine the changes in frequency
response that the different process types introduce, there are a lot of combinations to play with here.
You can make Phoenix II distort, but it’s not really designed to
do so in the way that, for example, SoundToys’ Decapitator is. If
you are after obvious distortion, the input trim can be cranked all
the way up to hit the processing as hard as possible, but be
aware that while the 6dB of available output trim is more than
enough to compensate for any gain introduced by the The tape emulation algorithms used in the Phoenix plug-in
derive originally from those developed for Crane Song’s
processing, it can’t compensate for all the gain that it’s possible HEDD A-D converter.
to add if you maximise the Input Trim. The range of the output
trim control may be adjusted in a future update to allow for that, but to be honest, smashing the input isn’t what this thing is all
about.
This is a plug-in that brings up detail, saturates, thickens and increases apparent size. Although the controls are fairly
simple, there are some different directions on offer here, and it’s easy to steer it where you want to go. I found myself able to
use combinations of the Sapphire position (the brightest frequency response) and the Iridescent process to increase the
presence and impact of vocals without introducing obvious distortion or darkening the sound in the way that many tape
emulations do. Phoenix II retains the clarity of the source and this, I think, is one of the secrets of its success. It provides the
life and size of tape, but it doesn’t veil or cloud the source in a way that may not suit some productions. In this respect it’s
different from some of its competitors, and even though its primary development took place some years ago, sonically it’s
arguably more modern by comparison. So should you take this to mean that it doesn’t really sound like tape? Well, there are
those that would say that no plug-in sounds like tape, but let’s deftly sidestep that debate and just say that it doesn’t sound like
other tape plug-ins, not being a ‘warts and all’ emulation of that medium. It’s capable of greater subtlety and could easily find
its way onto every channel of a mix.
Peacock
Peacock, the newest of the three plug-ins, is Dave Hill’s vinyl processor. And this is really quite an unusual one, because in
the same way that Phoenix steers clear of wow, flutter and hiss, Peacock has none of the crackles or pops that we normally
associate with vinyl emulations. So what does it do? Well, it replicates the RIAA equalisation curves, and provides frequency-
dependent harmonic distortion, which is subject to time modulation, thus replicating tracing and tracking distortion. It may be
helpful to briefly explain those concepts: the RIAA (Recording Industry Association of America) equalisation curve is a
specification for the recording and playback of phonograph records, introduced in the 1950s. A high-frequency boost or ‘pre-
emphasis’ is used when cutting to vinyl, and a corresponding cut or ‘de-emphasis’ is applied when playing it back, with the
opposite happening at the low end. The goal was to maximise running times, reduce high-frequency noise and prevent large
excursions of the cutter when vinyl masters were made. Since the audio as stored on a vinyl disc is so heavily equalised, the
non-linearities introduced by the medium are strongly frequency-dependent. There are then two further factors to consider:
tracing distortion, which occurs because the stylus and cutter have different shapes and thus sit differently within the groove,
and tracking distortion, which results from the inherent misalignment of the replay stylus angle (mounted on a remotely pivoted
arm) compared to that of the cutter, which moves tangentially. The replay stylus is only perfectly aligned to the groove at two
points on the record, and this inaccuracy results in varying amounts of tracking distortion throughout the duration of the disc.
Peacock emulates these effects as well as providing dither that’s described as being “the spectrum of record noise”.
Again, the controls are simple. From left to right, we have Harmonic, Dynamic, Output Trim (±6dB), Dither and Color. As
with Phoenix there are five colour settings which introduce progressively more colour. Silver is the least coloured and the
brightest, Gold the closest match to Dave’s test material (more on that in a moment), followed then by Rich, Fat and Deep. I’ll
SOS Readers Ads paraphrase Dave’s own descriptions of how these controls work:
GRAB A BARGAIN
£516,050 “The Harmonic and Dynamic controls interact and control the level of the harmonic distortion. The Color switch changes the
of Second-User Gear for sale character and interaction of the controls, and sets the maximum amount of mid-range and low-frequency color, and the high-
frequency compression characteristic. The Dynamic control time-modulates the distortion components which are very
frequency dependent due to the RIAA curve. The amount of time modulation with the Dynamic control is level-dependent and
has a maximum range. It is more of a matter of finding the control position that is optimum. The Dither control modulates some
of the internal functions and adds dither to the audio path at a level for 16-bit dithering.”
Of course, that all sounds fantastic and complicated in a reassuring sort of way. Better still, these complexities remain very
much ‘under the hood’ in actual use. This plug-in, and in fact all three plug-ins, are of the sort where you pretty much turn the
knobs until your ears tell you to stop. And that’s another Dave
Hill party trick: he has the ability to design the inside of his
products with an engineer’s hat on, and the outside with a
musician’s hat on. Perhaps the most obvious example of this
was when he put note pitches instead of frequencies on the Ibis
equaliser, and the principle of letting the music guide the settings
rules here too. There are no meters, and the controls are simply
scaled from 1 to 10. You can just keep your creative head on and
think about the music. By and large, it’s hard to get into hot
water, provided your sources start off at a sensible level.
It’s important to understand that because of the pre-emphasis

of the RIAA curve, and because the cutter is limited in how much
it can move at any given frequency, the vinyl medium can not
accurately reproduce high frequencies at high levels. Initially,
compression occurs, followed by distortion — and Peacock The distortion imposed by the vinyl medium is extremely
behaves in the same way. If you have a hot source with a lot of complex.
high-frequency content, like a sibilant vocal or a tambourine, and
you set the Harmonic and Dynamic controls too high and/or use one of the more coloured Color settings, you’ll start to get
some ugly distortion. You can back the controls off, or use the Silver setting, for example, which allows for more high-
frequency level before the onset of compression and distortion.
A Colourful Display
So what does it sound like? Well, it’s amazing. While it’s capable of adding significant colour and harmonic content, it seems
to me to be at its best at more subtle settings. It has the ability to make things sound more ‘real’. I know that’s infuriatingly
non-specific, but I suppose it’s something to do with making the recorded sound closer in character to the way our brains
process the sounds we hear in the real world. It just sounds better with it on. It can be almost inaudible, but it makes the
sources in the mix hang together in a more natural way. It’s very hard to explain, but there are some sound files on the Crane
Song web site which help to illustrate the effect. These files are of greater interest still because they include a comparison with
a vinyl record. Dave has taken a mix and had it cut to vinyl, and then recaptured to digital, and provided a comparison
between that and the original digital master processed with Peacock. He points out that the Gold colour setting is designed to
be as close a match as possible to the record, and that it’s optimised for use at 96kHz, although it works perfectly well at other
sample rates, both according to him and in my experience.
I’ve found this plug-in to be capable of doing quite extraordinary things both on individual sources, across buses and on the
mix bus. It’s possible to thicken the mid-range, to alter the guts of the bottom end, and to soften the highs. You can add
harmonics in such a way that you can change the perception of bass on smaller speakers, in a similar sort of way to Waves
MaxxBass. It’s difficult to say how accurate an emulation of vinyl it is, because I have no way of setting up my own
independent comparison. But whatever it’s doing, it’s very special, very subtle and works wonders in the context of a mix. I
can also see how it could be a significant asset in digital mastering, although obviously being Pro Tools-only it may not be an
easy bedfellow for mastering houses.
RA
The final plug-in of the three, Dave Hill Designs RA, is described as a non-linear plug-in that can be thought of as an amplifier
being overdriven, but with control over which part is being overdriven. It’s another unusual tool, with a unique set of controls.
From left to right, the Drive control is followed by a section labelled Even Harmonics, comprising Top Peak and Bottom Peak
rotary controls. Next comes Low Level, followed by a Peak Control section comprising Peak and Hardness controls, finishing
off with the Output Trim control. On the right-hand side of the GUI there’s a graphic display showing input on the X axis and
output on the Y. This provides a useful indicator of how the controls affect the audio path.
The three main processes — Low Level, Peak Control and Even Harmonics — affect the audio path in that order, rather
than in the order in which the controls appear on the GUI, so it makes more sense to look at them accordingly. Low Level is a
third-harmonic distortion process, affecting only the low-level portion of the signal, leaving the peaks untouched; it is, in effect,
a ‘detail’ control. The Peak section also applies third-harmonic distortion, but this time only to the signal peaks. It acts by using
distortion as a form of limiting, with the Peak control affecting the final output level, while the Hardness is rather like a
compression knee. The Even Harmonics section allows harmonic content to be increased in the top and bottom part of the
waveform shape independently. Dave explains:
“These controls are the most difficult to hear and describe, being that the second harmonic is an octave... but it does
change the feel of things and can have a thickening effect. The Top control rounds off the top of the wave shape and the
Bottom control rounds off the bottom of the wave shape. When used together it is possible to have the even-harmonic content
cancel and have third-harmonic distortion left. Try one of these in conjunction with the Low Level control on an acoustic guitar,
bass or kick drum.”
The Drive control can be thought of as an input trim affecting the level going into the plug-in. The process is very level-
dependent, and this allows for fine-tuning if the source is recorded either very hot or at a low level. The Output Trim provides
the usual ±6dB for balancing the processed signal with the original source.
I’ve owned RA for quite some time, and had a chance to use it on a good number of projects. The Low Level distortion is
extremely useful: you can just bring up detail but without the obvious ‘action’ of parallel compression. It’s more like parallel
distortion, but that tends to cause a build-up of mud which you don’t get here. The benefits are tangible on all sorts of sources,
especially vocals and acoustic instruments, and while you can create a little ‘hair’ when you maximise the control, it’s mostly a
subtle enhancement.
The Peak controls, meanwhile, are quite extraordinary. Somehow, you can apply limiting which is almost inaudible, to the
point of being quite unnerving. If you overdo it your source just seems to lose its dynamics but you can’t identify any obvious
compression artifacts. I have found it useful in high doses on bass and in smaller amounts across the drum bus. What, then,
of the Even Harmonic controls? Well, I’m not going to pretend that I can quite make sense of exactly why the distortion of the
top of the waveform sounds different to the distortion of the bottom, but this, again, is one of those situations in which you turn
the knobs and see how they make you feel. I’ve often found myself adding a good amount of one or other. When adjusted in
the context of the mix, rather than in solo, there’s often a sweet spot. This is another unusual and significant tool, capable of
special results that can’t easily be achieved in other ways.
Logic Users Need Not Apply

While I make no attempt to disguise my love of these plug-ins and my admiration for Dave Hill as a designer, there are a
couple of things that some potential users will not prefer. Firstly, there’s the issue of cost. In a world where our email inboxes
are filled to the brim with daily $99 deals, the price of admission is steep. I believe it’s worth every penny, and that quality
(usually) costs, but some will choose to vote with their wallets (or not, as the case may be). It might be nice to see a bundle
deal of some sort. It’s also bound to come as a disappointment to some that these plug-ins are Pro Tools-only. There have
never been VST or AU versions, and there are no current plans to release any products in those formats.
The distribution of these plug-ins is also somewhat antiquated. They’re sold exclusively through Crane Song dealerships,
and as such there is no direct online purchasing system. Demo licenses cannot be automatically obtained via the Crane Song
or Dave Hill Designs web sites. Demos are available, but you have to email the relevant company to obtain a licence. This
obviously takes longer than using an automated system and is inconvenient.
Cranes In Flight
Individually, each of these plug-ins would represent a significant enhancement to any mixer’s palette. Together, they’re
stunning. Where one may not quite hit the spot, another almost certainly will. The cumulative effect of this distillation of
analogue character, this sometimes slightly crazy but always musical and intuitive collection of colours, is profound. These are
tools that encourage you to mix with your ears and not with your eyes, the simplicity of the interfaces and unique nature of
some of the controls preventing ‘mixing by numbers’. In some ways, life before the digital studio was hard: equipment was
incredibly expensive to buy and maintain, and a great deal of skill and experience was required to use it to best effect. But in
other ways it was easy: analogue recordings have a certain character, an easiness on the ear which could be achieved simply
by competent use of the equipment. For those who now record and mix in the digital domain, that character must be actively
fought for, and intelligently so. While so much of their competition can tend to sound clumsy and a little overblown, these three
have the sort of subtle, understated quality that makes all the difference. I’m in. .
Who Can Use These Plug-ins?

Phoenix II is available in AAX 32- and 64-bit versions, Native and DSP, for Mac OS or Windows systems, and requires a
version 2 iLok. It is a mono or multi-mono plug-in, with a stereo version to follow. It runs 79 mono instances per DSP chip
at 48kHz. Natively, I found it extremely efficient, running hundreds of instances easily. Phoenix II introduces no latency.
Peacock is available in AAX 32- (on Pro Tools 10.3.5 or later) and 64-bit versions, Native and DSP (on the HDX card
only), for Mac OS or Windows systems and requires a version 2 iLok. It is a mono or multi-channel plug-in. It runs 11
mono instances per DSP chip at 48kHz. Natively, it was the most CPU-hungry of the three, but I was still able to run large
numbers; on a reasonably current system you could run it on every track of a decent-sized mix if required, although at
higher sample rates you’d need a high-end machine. Peacock introduces seven samples of latency at standard sample
rates, 13 at higher rates and 25 at 192kHz.
RA is available in AAX 32- and 64-bit versions, Native and DSP (on both HDX and TDM), for Mac OS or Windows
systems, including PPC G5 machines running Pro Tools 7 or later. It requires a version 2 iLok. It is a mono or multi-mono
plug-in, with a stereo version to follow. It runs 14 instances per DSP chip at 48kHz. Natively, I found it very efficient, and
while not quite able to match Phoenix II for instance count, I was able to run hundreds of instances with moderate increase
in CPU load. RA introduces no latency.

Reviews Information
SOS Site Analytics
In this article:
The Basics
Dangerous Music Compressor
Leftfield Ideas Dual-channel Compressor Buy PDF
Buttoned Up Published in SOS October 2014

Reviews : Processor
Tech Specs
Put To The Test The Dangerous world of dynamics processing promises
Doing The Dirty? plenty of pleasant surprises.
Cleaning Up
Hugh Robjohns
Alternatives
D
Dangerous Music angerous Music have become well known over the last 14 years for their range of analogue monitor controllers, source
switchers, and summing boxes, all of which are great–sounding and thoughtfully designed products. More recently,
Compressor $2799
though, they’ve begun to introduce signal processors into their range, starting with the BAX EQ and followed soon
pros
after with the Dangerous Compressor, which was announced at last year’s AES conference in New York.
In the default operating
mode, it performs flawlessly. The result of over two years of development and honing, the innocently named Compressor is designed to satisfy the
Smart Dyn mode handles quality expectations of the mastering fraternity, delivering a fundamentally transparent but musical character. However, as you
loud transients seamlessly,
might expect if you have any experience of Dangerous’s products, the design team introduced some unusual aspects to
almost negating the need for
a separate limiter. maximise the usability for project studios too. And I have to say up front that their ideas have resulted in the creation of a
Choosing soft knee provides fantastic compression tool that fully justifies its price.
an even gentler option.
Bass and sibilance filter The Basics
options included in the side–
chain. The Compressor is described by Dangerous as a dual–channel compressor with provision to link the individual channel side–
Balanced external side– chain control signals for stereo operation. However, many of the configuration options apply to both channels simultaneously,
chain with monitoring facility. so there are a few minor restrictions on true dual–channel operation. The Compressor is housed in a heavy–gauge 2U
cons rackmounting case made of black–painted steel. The whole assembly weighs a chunky 5.25kg and stretches 310mm behind
Operating modes applied to the rack ears.
both channels mean its use
as a dual–mono device is A pair of distinctive edge meters claims pride of place in the
slightly restricted. centre of the front panel to show the input/output levels or the
summary amount of gain reduction. Over on the right–hand side, 10 rotary
This compressor is very controls govern all of the usual compressor parameters, starting
easy to use and performs with a switch which offers eight ratio options (1, 1.4, 1.7, 2, 3, 4,
superbly, controlling the 6, and 20:1). A gain control allows attenuation as well as make–
dynamics in an extremely up gain over a ±10dB range, and is accompanied by threshold
transparent but effective (±20dB), attack (1–100ms) and release (10–500ms) controls.
manner. Impressive.
Leftfield Ideas
information
$2799 So far, so normal, then. But on the left–hand side are some less
www.dangerousmusic.com familiar facilities, controlled by 12 round push–buttons, each
housing a status LED. The first button, labelled Engage (red
LED), passes the signal through both channels of the
compressor circuitry. When not engaged, a hard–wire relay bypass is applied. Next, a group of four buttons with orange LEDs
configure the side–chain options (again, for both channels simultaneously). Normally of course, side–chain access is used to
allow an external equaliser to manipulate the compressor’s sensitivity to certain frequency regions. However, the Dangerous
Compressor features two basic EQ options in the side–chain signal path, in the form of separate bass–cut and HF–boost
equalisers. These can be switched separately to reduce the compressor’s sensitivity to kick drums and to firmly control the
sibilance region, respectively. The bass–cut option is a first–order (6dB/octave) high–pass filter reaching –3dB at 60Hz, while
the sibilance boost is achieved with a 2dB shelving equaliser above 5kHz.
In cases where more sophisticated EQ or other external

processing is required, the Compressor has fully balanced side–
chain sends and returns, with the sends always being active
provided the Compressor is ‘Engaged’. The return signals (for
both channels) can be selected instead of the internal signal at
the press of a button, and the side–chain signals can also be
monitored through the main outputs, if desired, making it much
easier to tune an external equaliser.
Three more buttons (green LEDs) make up the Contour

section, which modifies the threshold, ratio and attack/release
parameters. A button labelled ‘Soft Knee’ smooths the transition
into compression when the signal exceeds the threshold,
providing a gentler and more subtle or ‘transparent’ compression
effect. By default, both the attack and release times are set to
around 30ms, providing an automatic dynamic response which A shot of the interior. The VCA chips are the black rectangles
seems to work well with most transient–rich material. However, if to the left of the large black capacitors.
the attack and release times need to be adjusted manually,
pressing the ‘Manual Att/Rel’ button activates the front–panel rotary controls to allow just that, lighting a green LED between
the controls to warn when they’re in service.
The most unusual configuration mode is controlled with a
button labelled ‘Smart Dyn’, which effectively sets up the side–
chain circuitry with two independent threshold detection circuits,
one of which is only interested in the average level, and the other
only in brief transients. The idea is to allow a higher average
level to be maintained without occasional loud transients
‘punching holes’ in the sound. Dangerous Music recommend this
mode for most material, and it does seem to be extremely
effective and artifact–free. In essence, the system appears to
provide separate limiter and compressor side–chains, controlling
the same VCA at the same time — an approach I first noticed in
the Audio & Design F760 Compex, back in the 1980s.
Buttoned Up
The rear panel sports dual side–chain send/return loops on
Another trio of buttons (red LEDs again) configures the edge– XLR connectors, as well as meter trims.
GLOSSARY: technical terms
meter display, with the first switching between gain reduction (on) explained
and signal level. With the signal-level mode selected, the second button selects either output (on) or input level. The last
button attenuates the level meter signal by 6dB to accommodate hot signal levels, so 0VU is aligned to +4dBu as standard or WIN Great Prizes in SOS
+10dBu in the ‘hot’ mode. The meters have a typical VU ballistic, so when switched to indicate gain reduction they react fairly Competitions!
slowly, displaying the average gain reduction and omitting any brief transients. Consequently, two bright-green LEDs are Win Dangerous Music D-
provided as well, flashing when any level of gain reduction is being applied — these respond to even the briefest transients. Box (Americas Only)
Finally, a single button with a blue LED links the two channels together for stereo operation. When active, the control– & Sub8 (UK, EU ROW
voltage (CV) outputs (as opposed to the audio input signals) from the two independent side–chains are combined, with the Only)
largest one at any moment effectively taking control of both channels’ VCAs equally. This approach, which ensures a stable
stereo image, is a standard practice in dual–mono compressors configured for stereo operation, and is the only sensible
approach.
Dangerous Music say that some stereo–only compressors cut corners, using a single side–chain circuit working on a simple
mono sum of the two audio inputs. Such an approach is riddled with problems if there are out–of–phase elements between the
channels, or very strong central components, resulting in all sorts of unexpected and unhelpful gain-reduction effects.
Thankfully, there are few such designs — in all my years of reviewing compressors, it’s something I’ve only ever noticed in
one product.
In stereo mode, the upper channel’s gain and threshold rotary

controls affect both channels together, and the lower channel’s
corresponding controls are disabled. However, the ratio and
attack/release (if activated) controls remain completely
independent on both channels, and normally need to be set to
the same positions.
Moving briefly around the back, the electronically balanced line

input and output connections are presented on XLRs with 20kΩ
input and 50Ω output impedances. Four more XLRs provide
electronically balanced side–chain sends and returns for the two
channels. An IEC mains inlet incorporates the mains on–off
switch, fuse, and voltage selector. A pair of recessed multi–turn
trimmers allows the meter zero point to be adjusted when
switched to the gain-reduction mode, compensating for any Confirmation of the crosstalk specification at 1kHz
side–chain control–voltage drift that might occur over time.
Tech Specs
As I’d expected, the published technical specifications are exemplary, with a frequency response that’s flat within 0.25dB from
15Hz to 80kHz (not reaching –3dB until 6Hz). Given its maximum signal I/O level of +27dBu and a noise floor of –93dBu, the
Compressor’s potential dynamic range is an impressive 120dB, while distortion is below 0.005 (THD+N) and 0.007 percent
(IMD). Crosstalk between the two channels is also very impressive: it’s quoted as 115dB at 1kHz, but I measured it closer to
117dB, and better than 109dB at 10kHz. These figures confirm this product’s suitability for genuinely independent two–
channel processing, if you don’t mind the channel–linked configuration modes.
Taking a peek inside the case, there’s more empty space than
I was expecting. A linear power supply hugs the left–hand side,
featuring two compact toroidal transformers. One provides ±18V
DC audio power rails to maximise audio headroom, and the
other provides the DC control and LED rails. The one mildly
negative comment I’d level at the construction is a personal bug–
SOS Readers Ads
bear: the IEC mains inlet safety earth is routed via the PSU
GRAB A BARGAIN
£516,050 circuit board and connects to the case via a PCB track, PCB
of Second-User Gear for sale mounting screw, and metal stand–off post. From a safety
now — don't miss out! perspective, I’d prefer this to be connected directly to the
chassis. The Compressor has a two–year warranty period.
Confirmation of the frequency response
All of the audio electronics are hosted on a main circuit board
which abuts the rear panel and carries the XLR connections.
Surface-mount components dominate the board, all apparently being selected on the basis of their (lack of) influence on the
sound quality. The audio circuitry is DC-coupled, but a single Solen capacitor blocks DC to the VCA.
The main input and side–chain return stages employ THAT
1246 balanced line receivers, with Analog Devices AD8510
high–performance J–FET op amps providing most of the
buffering and gain–stage duties throughout the rest of the signal
path. THAT 1646 balanced output drivers are employed for the
main and side–chain send outputs.
Audio signal attenuation is courtesy of THAT 2181 Blackmer

VCAs, controlled from a combination of THAT 2252 RMS level
detector chips and some custom circuitry involving Analog
Devices AD8513 J–FET op amps. I counted 18 sealed relays on
the PCB, which configure the various operating modes, and 21
cermet trimmers in different circuit sections allow the circuitry to
be aligned and tuned, all carefully sealed in position after factory
Confirmation of the Intermodulation Distortion performance
calibration.
Put To The Test

Dangerous Music’s design goal with the Compressor was to
make it simple to use for the novice, but with the manual
controllability required by experienced users — and they’ve
certainly achieved that. In most cases, you simply decide on how
‘squashed’ the track needs to be, set the ratio control
accordingly, and then turn down the threshold control to achieve
the required amount of gain reduction. The automatic
attack/release times seem to work well with most material, as
does the ‘Smart Dyn’ mode. Switching on the soft-knee option
helps to maintain the inherently very transparent character. A plot of the different ratios in the hard knee setting
Whether working on a solo voice or a complete stereo mix, the

Compressor reigns in the dynamics effortlessly and naturally
and, in many cases, so well that I questioned whether it was
switched in! The same was true of percussive and transient
elements: no holes, no pumping, just less dynamic range. What
more can you ask? On vocals, the Compressor is sublime — it’s
one of the most natural and inaudible compressors I’ve ever
used, even with quite heavy-handed ratio and threshold settings.
The side–chain sibilance filter is pretty good too; using an
external equaliser may allow a more tailored and aggressive
approach, but the somewhat conservative generic setting seems
to minimise most typical sibilance effects quite effectively on its The same ratio slopes, but with the soft-knee option
own. I also found it useful in a mastering context, with a mix that engaged, showing how the transition into compressor starts
earlier and more gently.
was rather full–on through the high–mids.
As a bus compressor, the Compressor does a superb job of gluing all the elements together to form a cohesive sound, and
it does this without ever sounding controlled or constrained; it always seems to sound open and transparent, crisp and clean.
As I said before, I often found myself bypassing the compressor to convince myself that it really was wired in and doing what
the meters said it was doing. And when I bypassed the Compressor, all the quieter details I could hear fell away — so it was
was definitely doing something very useful, but in the most natural, unobtrusive way.
I can certainly see (and hear) the mastering influence. The

Compressor may not have expensive switched controls
everywhere, but it’s easily of mastering quality from a sound
point of view. The Smart Dyn mode is so effective at controlling
transients that I found my usual down–stream limiter often
became redundant, which is another big plus in terms of cost–
effectiveness.
Doing The Dirty?

As a transparent compressor, then, this is a great tool, but
sometimes compression artifacts are intended as an element in
the mix. Impressively, the Compressor makes this possible too:
you just switch off the soft–knee and Smart Dyn modes, and turn
on the manual attack/release. Adjusting the release time to get Confirmation of the THD+N distortion performance.
the track bouncing along just right then becomes very easy.
The side–chain bass filter is even more conservative than the sibilance filter, but once more it works well, reducing any
tendency for the kick drum to modulate the rest of the mix. A more targeted result might be achieved with an external EQ, but
that doesn’t make it any less useful and the bass and sibilance filters are still available even when the external side–chain is
active.
Cleaning Up
The over–riding sound character of the Compressor is well, there isn’t one! It’s very open, clean and natural at the top end,
and tight, focussed and accurate at the bottom. Engaging the
stereo mode doesn’t narrow the image at all — not that it should,
but psycho–acoustically some compressors do seem to suffer in
that way. The astonishing sense of transparency and complete
absence of audible distortion, even when compressing hard, are
seriously impressive. For those who like compressors to
introduce grunge and dirt, the Dangerous Music Compressor
might not be ideal, but for anyone who wants to reign in the
dynamics and push up the average level subtly, musically and
effectively, this unit does a very impressive job. .
An oscilloscope plot showing the attack/release

characteristics in the default mode with a 4:1 ratio. The test
signal alternates between –20 and –10 dBFS. Note the initial
overshoot as the VCA takes time to react, and the under–
shoot as gain reduction is released. These are the default
attack/release settings of 30ms
Attack and release set to minimum manual times. Note

reduced attack and recovery times compared to default.
The attack response appears much faster in this mode.
Attack and release set to maximum manual times. Note

lethargic attack and recovery times compared to default.
Alternatives
There are a surprising number of stereo and dual–channel compressors available for around the same cost as the
Dangerous Compressor. Avalon’s AD2044 is a well–respected optical compressor, as is the Empirical Labs Distressor
EL8S (and XS) dual–channel combo, and then there’s the classic Universal Audio 2–1176 FET compressor too. Moving
into the less well-known alternatives, A–Designs’ HM2 Nail is a hybrid solid–state/valve design, with some very interesting
functions, including a blend control for parallel compression, while the Looptrotter Monster is really a sound–shaping FET
compressor with various tube-saturation modes. The Smart Research C2 also features a crush facility for really cranking
up the mix level. For truly transparent compression, Pendulum Audio’s OCL–2 is an optical compressor/limiter, Dramastic
Audio’s Obsidian is a stereo–only VCA–based unit, and the IGS Audio Tubecore Vari–Mu is a valve–based stereo
mastering compressor. I could go on...

EBS ValveDrive DI
Bass DI & Overdrive Pedal
Reviews : Effects
Could this thoughtfully designed bass front-end be the
answer to every bassist’s stage and studio prayers
Chris Korff
T
he ValveDrive DI is intended for bass guitar and is part valve DI box , part overdrive pedal. It has both a balanced, high-
impedance XLR output (suitable for mic inputs, at 20dBV) and an unbalanced, low- impedance, line -level ( 10dBV) jack
output. Its sole input, a 1MΩ quarter -inch jack, has an associated 10dB pad switch, which means it should work fine
with active or passive pickups. On the top panel is a large, polished -chrome chassis that houses an ECC83 (12AX7) valve.
EBS have thoughtfully contrived to power the pedal using a standard (and included) 9V DC power supply, while stepping this
up internally so that the valve and its attendant circuitry have a high-voltage supply. Below the chassis are the pedal’s self -
explanatory knobs (volume, treble, middle, bass and gain), and finally the bypass and Vintage footswitches.
The EQ controls work more or less as you’d expect, other than

that there’s quite a ‘scooped’ sound by default. To achieve a ‘flat’
response you have to turn the Bass and Treble controls down
but leave the Middle knob at around 12 o’clock. There’s enough
range on all the bands, though, to get pretty much any kind of
tonal balance you’re likely to need, especially once you take into
account the Vintage switch. This switch’s exact workings aren’t
made too clear by the manual (which describes it rather vaguely,
as “changing the tonality” and adding some compression), but to
my ears it seems to boost the mids a lot, which is useful if you’re
not after a modern, ‘zingy’ sound. The effect of the Vintage
switch varies quite dramatically depending on how much drive
you’ve dialled in. At lower gain settings, it seems to fill the sound
out a bit, negating some of that scooping, but when applying a
fair bit of distortion it adds quite a lot of extra bite.
There’s overdrive available in buckets — this thing really can

pile on as much of the stuff as you’d reasonably need — but it’s also capable of subtle saturation and everything in between.
With the gain turned just a notch above zero, and the volume turned up to compensate, you can coax some very authentic
soul and reggae tones out of this DI, especially with Vintage mode engaged and the treble backed off. At the other extreme,
you can achieve some delightfully nasty tones, along the lines of Aphex Twin’s brutally abrasive ‘Come To Daddy’.
So, this device is capable of some very useful sounds across a remarkably broad tonal range, then. But I did have one
gripe. The manual makes reference to the bypass switch being able to operate either as a ‘normal’ bypass or as a mute,
depending on the position of a switch on the back panel. Yet, this switch doesn’t exist! (When I contacted EBS, they told me
that the manual was out of date and that they found that there just wasn’t room on the back to accommodate what was an
early design feature). As it stands, the bypass switch works as a mute on the XLR output and as a bypass on the jack, which,
for me, rather limits the pedal’s usefulness. For example, if you wanted to use it as an occasional effect live, you’d need a
separate DI box to connect to the pedal’s jack output; if you tried to connect the XLR out straight to the FOH desk, you’d end
up muting yourself whenever you bypassed the pedal.
That criticism aside, the ValveDrive DI is a great -sounding and very versatile pedal. Whether you want to use it as a spot
distortion effect through a bass amp, or a valve ‘warmer’ on your way to front -of -house or a recorder. Chris Korff
$399.95.
www.musicaldistributors.com
www.ebssweden.com

In this article:
Personal Organiser
Electro-Harmonix B9 Organ Machine
Inside & Out Organ Emulation Effects Pedal Buy PDF
Getting Organised Published in SOS October 2014

Reviews : Effects
B9 Be Mine?
Alternatives Is EHX’s latest pedal perfect for guitarists who want to pull
Electro-Harmonix out all the stops?
B9 $220 Paul White
pros
I
found the demo videos of Electro-Harmonix’s new B9 pedal so compelling that I put one straight on the credit card, rather
Authentic organ–like tones.
than wait several weeks for a review model to arrive — and having now put it through its paces, I’m glad I did, as I will
No special playing
technique required. certainly be able to make good use of it on stage and in the studio.
Very affordable.
cons Personal Organiser
No modulation speed The concept is simple enough. You plug in a conventional electric guitar and the pedal generates a choice of nine different
switching.
organ sounds. There’s no hex pickup, no laptop full of samples and no MIDI — just a stompbox, which I have to say is very
summary reasonably priced. Depending on the preset chosen, you’re able to adjust either the amount of key click or the depth of
A surprisingly believable and modulation, and there are separate outputs for the dry guitar and organ sound, should you wish to feed separate amps.
versatile organ emulation,
considering that it’s If this seems longer than a typical pedal review, that’s because the B9 isn’t a typical pedal. I think it steers what we might
processing the guitar sound
think of as ‘guitar synthesis’ in the most promising direction yet — and I really think Roland should be pushing harder in this
directly. Very usable on
stage or in the studio.
direction, with their clever HRM processing technology. Although their VG devices already offer organ–like tones, they don’t
approach the realism on show here.
information
Let’s look at the technology first. This isn’t a pitch-tracking, sample–firing
$220.30.
guitar synth, with all the attendant headaches of such devices. And, as I say, it
www.ehx.com
doesn’t need a hex pickup. In fact, it’s based on the same type of pitch-shifting
and wave-shaping technology used in Electro-Harmonix’s HOG pedal, but
finessed to the point that it’s able to produce extremely credible organ tones.
I suspect some form of polyphonic pitch–tracking is used to control the

filters that shape the sound, but the sound itself comes from processing the
sound produced by the guitar string. The range doesn’t extend beyond the
23rd fret of the top E-string, but there are none of the familiar ‘guitar synth’
tracking issues to worry about, such as delay or wrong pitches being
produced. That said, the filters can be coaxed into the occasional tonal warble
with some chords, particularly major sevenths, with two notes in the chord
being just a semitone apart. If the organ is layered with the guitar this is
usually quite unnoticeable, but it’s something to be aware of when playing
more exposed parts.
The playing experience reminds me of working with the aforementioned

HRM pseudo–synth sounds generated in Roland’s current range of VG
processors, or even playing Roland’s ancient GR300 guitar synth, which also
worked by processing the string signal. Even so, clean and even playing is
always rewarded. Choosing suitable organ chord voicings, playing partial
chords or arpeggios sounds far better than just strumming, though it will follow strumming too if you really must. Similarly,
although the organ sounds will happily follow your pitch bending — or even slide playing, if you wish — that’s really not the
way we expect to hear an organ sound. Organs aren’t normally velocity sensitive, so players with wayward picking dynamics
might produce a more even and organ–like result by putting a compressor before the B9.
For the best–sounding results, the B9 needs to be placed either at the start of your pedal chain or directly after a
compressor. It doesn’t take too kindly to being fed ‘dirty’ signals, or ones with echo or reverb added — but you can always add
these effects afterwards. The manual also suggests that the bridge pickup of a typical guitar gives the best result, but in
practice I found that any pickup combination on my Strat sounded fine. You will, however, notice some changes in timbre if
you change pickups — remember that the sound is produced by processing the input signal, not by replacing it with something
else, so if you plan to use this a lot, some experimentation with pickup choices may be worthwhile.
Inside & Out

The pedal takes the form of a fairly unassuming die–cast box, but inside it’s full of surface-mount cleverness. The organ
output sends a mix of guitar and organ sound, depending on the Organ and Dry level settings, so when working with a single
amp you can balance the contribution of each. When you hit the bypass switch the guitar signal is output from the organ jack
at full volume. Note that the output is buffered (there’s no hard bypass). A power supply is included, but the pedal can also be
powered from a Boss–compatible external power source. There’s no battery-power option, probably because the 100mA
current draw would get through batteries pretty quickly!
Four knobs along the top relate to the Dry and Organ levels, modulation rate and key–click level, with the last also adjusting
the modulation depth on those sounds that don’t require a key click. Clicks are only generated on the first note of a chord or
phrase and don’t recur until the input signal has fallen below some internally determined threshold. A further knob operates a
nine–position voice–selection switch, with descriptions of the nine sounds printed on the front panel.
The nine voicing options are Fat and Full, Jazz, Gospel,
Classic Rock, Bottom End, Octaves, Cathedral, Continental and
Bell Organ. The first eight emulate popular organ sounds and
cover tone–wheel, electronic and cathedral organ tonalities. The
last setting merges the bell–like attack from an electric piano with
an organ pad sound, with the bell component being somewhat
touch–sensitive.
The modulation is essentially, in most cases, a chorus–like

effect, but it makes a surprisingly good stab at lending a rotary
character to the sound. The Continental voice has vibrato The user has separate control over guitar and organ output
modulation and the Cathedral has tremolo. I think the designers levels, as well as the degree of modulation and key click.
have missed a trick, here, as rotary effects always sound best
when you’re able to switch speeds during performance, ideally with the speed change lag built in. This could have been
achieved via either a second footswitch or an expression pedal input, and I’d have happily paid a little extra for this facility. As
it is, I’m tempted to buy a separate rotary emulation pedal to put after the B9 — but to be fair, a decent one of those costs
more than the B9 itself. GLOSSARY: technical terms
explained
Getting Organised
Having played with the pedal a great deal now, I’ve come to the conclusion that it sounds best when fed into a separate and Competitions!
fairly clean amp, especially if you need a dirty guitar tone. So, for my own live use, I plan to route the organ component to the Win Dangerous Music D-
PA mixer via a volume pedal or, for smaller gigs, to a separate guitar combo. You can always put other effects pedals, such as Box (Americas Only)
overdrive or delay, after the organ output if you need them. In the studio, of course, you can add whatever processing you like, Win Adam A7X monitors
including things like rotary speaker and amp modelling, via plug–ins after recording. & Sub8 (UK, EU ROW
Only)
How realistic the organ tones sound depends very much on how you play them, although when layered behind the guitar
sound with no change to playing style they always manage to convey the sense of a keyboard player in the background, as
the organ notes sustain much longer and at a more even level than typical guitar notes. For example, if you choose the
Gospel setting and ‘arpeggio’ your way through Dylan’s ‘Like A Rolling Stone’ with an equal mix of guitar and organ, the result
is uncannily realistic. The same goes for ‘House Of The Rising Sun’ using the Continental voice. Note, though, that you may
get fined for doing that in a music shop...
SOS Readers Ads

More exposed parts or solos require a bit more musical thought, but I’ve found that it is still possible to get very close to
GRAB A BARGAIN recreating the effect of a real organ performance in a band or duo context. The pedal also works with the higher notes of an
£516,050 electric bass guitar, though not quite so well below the instrument’s D-string.
now — don't miss out! The stars of the show for me are the Gospel organ, the Cathedral organ (very psychedelic) and the Continental, but each
voicing has applications, and all of them sound uncannily like the instrument they emulate. Bottom End is also useful for
adding a sub-octave with a bit more character than a typical octave divider or pitch-shifter would give you.
Of course, the physical playing experience is also vitally important. The B9 makes you feel as though you’re playing through
an effect — and I mean that in a positive way; you don’t get that detached feeling that comes from working with a pitch–
tracking guitar-synth type of device. The sounds are also touch-sensitive and change in character slightly depending on
picking intensity; if you strike a pinched harmonic, the organ sound follows it rather than going off into wailing histrionics. This
makes the playing feel immediate and natural.
B9 Be Mine?
To wrap up, then, if having the ability to recreate organ sounds either live or in the studio appeals to you and you don’t want to
fight with the quirks of conventional guitar synths or computers and MIDI, the B9’s cost of admission really will be money well
spent. .
Alternatives
Roland’s various HRM modelling devices come the closest, but they require a hex pickup to be fitted to your guitar. Other
than that, the only competition comes from other Electro-Harmonix devices, such as the POG and HOG, which are more
expensive but offer a wider tonal palette than organs alone.

In this article:
More Than Summing
Greiner Engineering Tools Sum.mation
Sum Difference Analogue Mixer With DAW Plug-in Control Buy PDF
Hardware Published in SOS October 2014

Reviews : Mixer
Soft Centre
The Sound If you use outboard gear with your DAW, this could be just
Hot Stuff! what the doctor ordered.
Sum.ming Up
Matt Houghton
Facts & Figures
B
Greiner Engineering Tools ack in SOS May 2014, Bob Thomas weighed up the merits of Solid State Logic’s Sigma, a 16 stereo/mono–channel
summing mixer, which features software remote control. You can find that review on the SOS web site at
Sum.mation $2500
http://sosm.ag/ssl-sigma, and if you haven’t done so already, I’d recommend reading it as a point of comparison for the
pros
subject of this review, Greiner Engineering Tools’ Sum.mation. Greiner’s product beat the Sigma to market by a nose, but
Very clean sound. made its way more recently to the SOS offices.
Switchable active/passive
summing bus. The Sum.mation and Sigma are broadly similar in concept, but there are some important distinctions. First, the Sum.mation
Simple to set up and use.
is always a 16–channel device: you can configure it to mix up to eight stereo or up to 16 mono signals. The Sigma’s 16
Adds basic level automation
to any analogue mixer. channels, on the other hand, can each be configured for mono or stereo use, giving you a possible 32 inputs in total — so
Seamless integration with when used in stereo it offers double the Sum.mation’s channel count. Second, the software side of things is less sophisticated
your DAW. in the Sum.mation. Finally, judged on the mono channel count alone, the Sum.mation is considerably more affordable.
cons
More Than Summing
Can’t assign/recall LCR
panning remotely.
It’s important to point out that, like the Sigma, the Sum.mation
Runs a bit hot.
offers much more than summing alone. As well as the summing
summary bus itself, there’s level automation for each of the 16 input
So much more than a channels (hence the ‘mation’ part of the name), courtesy of 16
summing mixer, and so
THAT Corporation 2180A VCAs (voltage controlled amplifiers).
much more convenient and
clean–sounding than stand- Each VCA’s control voltage is determined by a D–A converter
alone VCA automation chip, which in turn is controlled by your DAW software.
devices of yesteryear. It
lacks a few of the bells and Once attached to your computer via USB, you can set the Sum.mation up to receive control data via one of three MIDI
whistles of its main ports. Choosing the first port means that the Sum.mation is controlled by its dedicated plug-in, which supports AU, VST, RTAS
competitor, but it should still and AAX formats, in both 32- and 64-bit versions where the format allows, and both Apple OS X and Windows. You place the
hold plenty of appeal for plug-in on an audio channel in your DAW, select port 1 and select the desired channel, and the plug-in’s fader will govern the
anyone running a hybrid level of the corresponding channel in the Sum.mation. The plug–in includes a MIDI learn feature, which makes it easy to hook
analogue/digital studio.
up any MIDI control surface for real–time hands–on control. The remaining two MIDI ports cater for the HUI protocol, whereby
information
the faders of your DAW’s first 16 channels (eight per port) are used to control the Sum.mation’s channel levels.
$2500.
Greiner Engineering
Tools +1 510 931 7682
info@greinerengineering.com
greinerengineering.com
After the VCA gain stage, each channel’s signal is fed to the
summing bus, but also to its own dedicated analogue output. That
means that if you patch this thing into the inserts of a 16-channel
console you’ll effectively have added channel level automation and
recall to your desk. In a similar vein, you could patch it into a
console with eight stereo subgroups to allow for automated stem
mixing prior to summing.
Sum Difference
There’s plenty of debate in the various pro–audio forums about the
relative merits of passive and active summing. Greiner felt that the
user should be able to choose which they wished to use, and so
there are actually two summing ‘engines’. One is a conventional
active summing bus, whereas the other takes the form of a passive
resistor network. Despite each having a dedicated output, you can’t
use both engines simultaneously. The unit is set to active summing
by default, and to switch to passive summing you need to move
some internal jumpers. (I’d have loved to see an external switch for
that).
The plug–in allows for hassle-free remote control via
One of the main benefits of passive summing is that the almost any DAW software.
summing circuitry is about as ‘clean’ as is possible in the analogue
domain. There’s inherently some loss in level, but the user is left with a choice about the nature of the makeup gain used —
you could use a pair of very clean–sounding preamps, for example, or choose something with bags of tube or transformer
character if you prefer. Despite the passive summing, each channel still flows through the VCA and may thus still be
automated.
The rear panel caters for analogue I/O via DB25 D–sub
Hardware connectors, and the USB port for remote control. The active
stereo output is presented on balanced XLRs, while the
Outwardly, the hardware side of things couldn’t appear simpler. unbalanced TRS passive output and cascade input allow
units to share a single mix bus.
Each channel’s analogue signal is received via a balanced
connection on DB25 D–sub connectors (there are two, each
catering for eight channels), which conform to the ubiquitous Tascam wiring standard and are presented on the rear panel.
Another pair of D–subs provides the balanced analogue line–level direct output for each channel. The main (active) stereo
mix–bus signal is delivered to the outside world via a pair of balanced XLR connectors, while the passive output is unbalanced
and presented on a single TRS jack socket. An IEC power inlet with integrated on–off rocker switch and a USB port for class–
compliant connection (no drivers required) to a computer complete the I/O.
I had a single Sum.mation to test, but Greiner say two units may be connected so as to sum 32 inputs on a single stereo
bus. To do this, you set one unit to passive summing and connect its passive output to the unbalanced TRS cascade input of a
second unit. (The receiving unit may be set to active or passive.)
The simple, but attractively finished front panel is dominated by a series of numbers and associated switches, with a dual
10–segment LED meter for the mix bus indicating the level going into the bus, and thus how close the summed signal is to
clipping. The numbers range from 1–16, each corresponding to one channel. A row of push buttons above the numbers is
used to assign that channel’s signal to the left side of the stereo bus, and a row beneath to route the signal to the right. Each
button has an associated bright–red LED which becomes lit when the switch is engaged, and when both are engaged — FOR SALE: MIXERS in SOS
indicated by both LEDs being lit — the signal is routed to both the left and the right, placing it in the centre of the stereo field. Reader Ads
In other words, what we have here in essence is an old–school analogue LCR pan mixer, with visual indication of the pan GLOSSARY: technical terms
position via those LEDs. explained
Soft Centre WIN Great Prizes in SOS

Competitions!
Hooking up the Sum.mation to my Windows (7, 64–bit) and OS X (9.2) systems was simplicity itself. It’s a class-compliant
device, so you just plug in the USB cable, switch on and it is recognised by the OS. Once the plug–in is installed, you fire up
Box (Americas Only)
your DAW and it should appear in the list just like any other plug–in. I tried it in Pro Tools 10 HD (which is 32–bit only), Cubase
7.5.2 64–bit and Reaper 4.591 64–bit, and in each case the plug–in showed up as expected. The only exception was that I’d & Sub8 (UK, EU ROW
installed the 64–bit AAX plug–in, which Pro Tools 10 couldn’t recognise. Checking the manual, this is expected behaviour: Pro Only)
Tools versions prior to 11 can’t use 64–bit AAX plug–ins; all you need to do is click to get past the warning screen, and the
RTAS 32–bit version will appear and work as expected.
The plug–in interface is both simple and easy to understand: on the left-hand side you can set it up for external control,
while on the right you select the channel (and on which unit, should you have more than one connected to the computer) you
wish the plug–in instance to control. Play some material through the Sum.mation, waggle the virtual fader on the plug-in, and
the level on the channel will rise and fall, just as you’d hope.
The HUI mode is similarly easy to configure and use, though I personally
preferred using the plug-in. What you don’t have with either approach to software A row of THAT 2180A VCAs, each
control is access to the LCR pan setting: that can only be switched manually, controlled by its own dedicated D–A,
which allow for automated channel gain
using the buttons on the hardware. and recall.
As I mentioned earlier, this whole approach to software control is slightly less

sophisticated than that of SSL’s Sigma. The latter uses an HTML5 page that allows for control from any device, irrespective of
operating system and irrespective of physical location: if you can connect to the router, whether via computer, tablet,
smartphone or anything else that uses standard web-browser software, you can control it. The Sum.mation is less ambitious
than this, but while it doesn’t support mobile/tablet operating systems, such as Android and iOS, directly and while it’s
intended as a one–computer device (connecting, as it does, to a computer via USB), I don’t think this is a bad thing. SSL have
certainly created a very versatile system, and I can see that its approach extends the potential appeal for large–scale
production facilities. But Greiner seem deliberately to have focused on meeting the needs of the modern hybrid studio.
Everything’s quick to set up, uncomplicated and does almost everything you could wish for in a single–studio setup — and if
you really want to use your iPad? Well, if you can use an app to control your Mac or Windows DAW, then you can use it to
control the Sum.mation via that DAW.
The Sound
I must admit to being something of a cynic when it comes to the reported sonic benefits of pure analogue summing. I just don’t
see the point in coming out of your DAW via a D–A converter per channel, simply to sum signals back together, because 32–
and 64–bit DAW software is more than capable of doing that job very cleanly indeed. (And even if it weren’t, in most projects
you’d still be summing tracks down to stereo subgroups in order to run them through a 16–channel summing box — using
precisely the same algorithms!). Sure, there may be microscopic differences, the sound having passed though converters to
reach the summing mixer and, in active devices such as the Sum.mation, having passed through various ICs. That might
introduce the tiniest bit of distortion, but it is vanishingly small.
Having said that, I do feel that analogue summing has

potential advantages in some setups. First, when, as you The full circuitry of the Sum.mation
inevitably do with analogue summing, you bring every channel’s
signal out into the analogue domain, keeping them outside the box means you neatly side-step any risk of phase
misalignment when returning signals to your DAW. (My, admittedly unscientific, tests suggest that DAWs’ automatic plug-in
delay compensation systems aren’t always quite as ‘automatic’ or reliable as they should be, particularly when dealing with
high latencies or ‘external’ plug-ins). Second, if you want to come out of the box to use outboard on your subgroups, it avoids
some stages of A–D conversion. Third, it allows you to play with the analogue headroom, using your toys to ‘drive’ signals into
the analogue summing bus, to achieve further pleasing distortions.
SOS Readers Ads Sonically, the Sum.mation sounds very transparent, but if I ever felt that the sound was lacking in colour, then running the
GRAB A BARGAIN passive summing engine’s output into a pair of Neve 1073-style preamps compensated for that; as did running the mix into
£516,050 various bus compressors. The technical specs vary very slightly from the Sigma’s (see box), but I have no complaints here.
now — don't miss out! This means that the Sum.mation excels in this hybrid–studio role — the only thing that may be lacking here is a monitor or
headphone output, or any sort of monitor control facilities, as you’d find on a full console. I could also see the system
appealing to those who wished to add some degree of automation to an analogue console — patching one or more of these
into the channel or bus insert points, for example, might be just enough to persuade someone that it’s worth bringing a
console out of its metaphorical mothballs.
Hot Stuff!
Functionally and sonically, I don’t really have any major criticisms — only the minor point about the lack of pan-setting recall,
which I’ve already discussed. But there was one thing about the Sum.mation which made lead me raise an eyebrow: the
amount of heat being generated by the unit. On checking with Scott at Greiner, he assured me that they deliberately use the
chassis as a heatsink, and that the electronics inside the box cope with the amount of heat generated without complaint.
That’s all well and good, but I’d certainly suggest that you leave some sort of ventilation space around each unit when it’s
placed inside your rack, just to stop it cooking adjacent units if nothing else!
Sum.ming Up
There are more dedicated summing mixers on the market today than I’d care to count, but the Sum.mation offers something
very different and, for the right user, far more convenient. The standalone VCA level–automation device may not be a new
concept — not only did various mixing consoles feature VCA automation, whether factory or retro–fitted, but Mackie, CAD and
Behringer, amongst others, brought similar stand-alone products to market, in the form of the Ultramix, Megamix and
Cybermix, respectively. Yet, the sonic performance of the Sum.mation is far superior to what went before. Also, those devices
had no integration into the modern software DAW, which leaves SSL’s Sigma as the only direct competitor, and the Sigma and
Sum.mation will have different pros and cons for different users.
Whether analogue summing is for you is another matter entirely, but if the answer is yes and the idea of level automation
appeals, I’d urge you to evaluate the Sum.mation before deciding you need the slightly greater functionality of the SSL. .
Facts & Figures

The technical specifications of the Sum.mation are perfectly respectable. The manual quotes a frequency response that’s
within 0.25dB of ruler flat from 20Hz to 20kHz and tests confirmed this — I don’t have the facilities to test frequencies
higher than that, but the Sigma boasts the same response all the way to 40kHz. The quoted dynamic range of 110dB A–
weighted may be slightly lower than the Sigma’s >115dB figure, and well below the SPL MixDream’s 125dB — which is
very impressive — but in practice this just isn’t an issue in this application; 110dB is plenty. The maximum input level
exceeds +30dBu (higher than the Sigma), while the maximum output level is quoted as +27.5dBu, again higher then the
Sigma. The total harmonic distortion (THD) of the signal path from channel input to stereo bus output is given as 0.027
percent — fractionally higher than the SSL’s 0.025, though I doubt even golden ears would detect this by listening! The
input impedance is given as 48kΩ and the output impedance as 50Ω.

Reviews Information
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In this article:
Black Is Back
Headway EDM1 & EDB2
EDM1 Acoustic Instrument Preamps Buy PDF
EDB2 Published in SOS October 2014

Reviews : Preamp
In Use
Conclusion Headway’s portable preamps are designed to help you get
Alternatives the best possible sound from your acoustic instrument,
Headway whatever it may be.
EDM1 & EDB2 $199/$320 Bob Thomas
pros
F
inding the best pickup and/or microphone for any particular instrument isn’t especially difficult, since most
Superb sonic quality.
Range control covers a wide manufacturers, both large and small, have designed pickups for almost every acoustic instrument and application under
variety of acoustic the sun. Preamplifiers are a different matter entirely, since the difference in the resonant frequencies and overall
instruments. frequency responses between instruments of differing string lengths and sizes means that a preamp designed, voiced and
Switchable input impedance optimised for, say, an acoustic guitar is probably not the best choice for a violin.
handles all types of active or
passive pickup systems. Originally known for their pickups, Oxfordshire-based Headway Music Audio have significantly raised their game over the
Easy to use, especially the last few years, not only increasing the range of pickups they offer, but also recently updating their original stand-alone two-
EDM1. channel EDB blender/DI to EDB2 status, as well as introducing the EDM1, a more compact, single-channel preamplifier. Both
cons of these units are designed to cope with the preamp requirements of a variety of acoustic instruments and transducers.
The EDM1 can’t be
phantom-powered from a Black Is Back
mixer when its earth is lifted.
Other than that, there’s Designed in England and manufactured in Korea, these two preamps have an air of quality that inspires immediate
really none! confidence. The smaller EDM1 and the larger EDB2 share the same aesthetic: black metal casing with integral mic-stand
summary mount, white legending, black switches, grey, black and green knobs, and red, green and blue LEDs. Although the preamps
Two extremely useful can be phantom-powered, the primary power sources are either a 18V, 200mA power supply or 9V PP3 batteries (one for the
acoustic preamps, marked EDM1 and two for the EDB2). Despite the similarities in cosmetics and construction, there are significant differences in
by excellent sound quality, features and functionality, and it is these that will ultimately determine which of the two best fits your application.
great features and,
especially in the case of the
EDM1
EDM1, remarkable ease of
use. With its supplied belt clip, single–channel layout and small form factor, the EDM1 might appear to be aimed at the acoustic
multi-instrumentalist looking for a simple device that can be tailored to suit different instruments and pickups. Whilst that is
information
certainly true, the EDM1 is essentially a well thought-out, sophisticated and very capable preamp.
EDM1 $199, EDB2 $320
sales@headwaymusicaudio.com
The EDM1 carries a TRS jack input that can be switched to supply 9V DC phantom power on either the tip or ring, and this
www.headwaymusicaudio.com
can be used to supply power to an instrument’s onboard preamplifier or microphone system. This input also has a three-
position switch that selects between +Hi (20MΩ), Hi (5MΩ) and Active/Low (1MΩ) input impedances, thereby allowing you to
better match the EDM1 to the requirements of your particular pickup, preamplifier, microphone or line-level source.
The TRS output jack is a balanced DI output that also accepts 48V phantom power from either a mixing console or a
suitably equipped preamp or amplifier. To make life easier if you’re phantom powering, Headway supply a TRS-jack-to-male-
XLR adaptor. Personally, unless I was going to be using the EDM1 either on the floor or stand-mounted, I’d give the adaptor a
swerve and make up a TRS-to-male-XLR lead just to keep things simple. One small point to note is that, if you lift the ground
using the front-panel slide switch, this kills the phantom power from the mixer. This does seem to me to be a bit strange, if
only because I’ve got a very low-cost DI or two where the ground lift does not affect phantom powering.
Headway describe the EDM1 tone control setup as a “Baxandall interactive three-band EQ section”. In Peter J Baxandall’s
classic, 1952 two-band design, the controls do not interact, but in the EDM1 the three shelving bands (centred on 120Hz,
590Hz and 10kHz) are designed to overlap slightly. The bass and treble EQs have a ±12dB range, and the mid can go ±13dB.
The bass control additionally adds up to 16dB of cut or boost at 45Hz to help cut handling noise, low-end feedback, mains
hum and so on. A volume control plus a Mute/Live switch (with red and green indicator LEDs) is positioned just below the EQ
section alongside the anti-feedback Range control.
The Range function is actually a variable 12dB/octave high-

pass filter, which you can at set any frequency between 40Hz
and 300Hz. Although Headway suggest that this control can help
to eliminate feedback at frequencies below the chosen point, I
think of it more as a voicing control for the instruments specified.
To assist you in selecting a suitable frequency, the Range control
is labelled with suggestions for a range of common instruments,
from Bass at 40Hz to Rock Acoustic at 300Hz (see ‘Home On
The Range’ box). It is worth bearing in mind that the Range
control is in circuit before the EQ, so if, for example, you’re on
the Rock Acoustic setting, there won’t be anything for the bass
EQ band to act on.
EDB2
A thoughtful update of the original EDB1, Headway’s EDB2 is around twice the size and weight of its smaller stablemate.
Although a belt clip is supplied, I’d be more inclined to mount it on either a mic stand or the floor. The increase in size brings
with it an increase in facilities over the EDM1, with one possible exception that we’ll come to later.
The EDB2 is in essence a two-channel blender, allowing you to mix two sources — pickup, preamp, microphone or line-
level source — down to mono on both a balanced XLR DI and an unbalanced line-level jack. The channel 1 input is on a TRS
jack and, as on the EDM1, gives you the option of sending 9V phantom power to ring or tip. Channel 2 is also fed by a TRS
jack, but this allows you to access the channels separately: channel 1 on the tip and channel 2 on the ring (if you plug in a
stereo jack), or just channel 2 if you use a mono jack. Additionally, the second channel can come from a balanced XLR
microphone input, to which 18V of phantom power can be supplied, via a front-panel three-position switch that also enables
you to cut out the microphone input to reduce background noise. You also have the ability to reverse the polarity of either
channel relative to the other.
Both channels 1 and 2 have their own gain controls, plus individual input impedance selection (identical to that of the
EDM1). The EQ is the same interactive Baxandall type, but this time the EDM1’s three-band setup is joined by a high-mid
band at 900Hz and a ‘presence’ band at 2.8kHz. In addition, there’s a semi-parametric, fixed-depth (-12dB) notch filter with
variable width (five octaves to half an octave) and a range of 50Hz to 6kHz. The five-band EQ is always active and can be
switched into either or both channels, whilst the notch filter, which has its own in/out switch and indicator LED, affects either
channels 1 or 2, but not both together.
The Range switch on the EDB2 is about the only function inherited from the original EDB1 that hasn’t been upgraded or
improved in some way. In contrast to the continuously variable cutoff found on the EDM1, the EDB2 retains the EDB1’s three-
position switch, giving you the choice of voicings for violin, guitar or bass. Fortunately, because the Bass EQ band is centred
at 120Hz, setting up the EDB2 for instruments such as bouzouki, tenor guitar or viola isn’t really going to be much of an issue,
although duplicating the EDM1’s Rock Acoustic setting is going to require a bit of creative use of both the EQ and notch filter.
Finally, it’s worth mentioning two small tweaks that illustrate the thought that has gone into the EDB2. The first of these is an
‘iPod in’ stereo mini-jack that allows you to send a mono sum of the output of your MP3 player of choice directly to the EDB2’s
outputs. The second, and an on-stage saviour for me, is the standby position on the power switch. Being able to switch power Neve 1073LB &
to the EDB1 on or off without causing major thumps and bangs through the PA is a real convenience. 1073LB EQ
500-series Microphone
In Use Preamplifier & Equaliser
Neve’s venerable
I have a few instruments that have been fitted with active or passive pickup/mic or pickup/pickup combinations, sourced from 1073 preamp and
(among others) Fishman, K&K, Barcus-Berry, Mimesis, Pure Acoustic and Sunrise. In addition to tenor and five-string banjos equaliser are both
(I notice that for some reason the EDM1 doesn’t have a Range suggestion for these), there’s a mandolin, a mandola, a cittern, now available in
a tenor guitar and a bouzouki — plus bass, baritone, six- and 12-string acoustic guitars, an Appalachian dulcimer and the API’s popular ‘Lunchbox’
inevitable ukulele or two. Then there’s my wife’s fiddle, nyckelharpa and octave fiddle. Available amplification included my format. Were they worth
Adam studio monitors, JBL Eon G2 active loudspeakers and a Phil Jones Bass AG150 acoustic amplifier — so the EDM1 and waiting for?
EDB2 received a fairly comprehensive workout.
Cloud Microphones
Both units produce an exemplary sound quality and manage the trick of being clean and clear without sounding sterile. The Cloudlifters
In-line Microphone
three input-impedance settings come into their own when you’re having to deal with the very different requirements of
Preamplifiers
magnetic and piezo pickups, internal mics and active preamps. Switching between instruments and/or pickups on the EDM1
Do these in-line
proved the efficiency and ease of use of its variable Range control. I really liked the way in which I could start with the Range mic preamplifiers
set to the recommended position for a particular instrument, but then had the freedom to vary the high-pass cutoff point to get mean you can use
the best sound. On the EDB2, a combination of the Range settings and the notch filter (which is, in reality, parametric EQ with a passive ribbon
a fixed -12dB cut), together with a bit of bass EQ, got me more or less to the same point, but the EDM1 was definitely the mic with any preamp?
quicker on which to get a good basic sound.
Slate Pro Audio Fox
However, the EDB2 came into its own when dealing with more complex (from a sound-reinforcement point of view) sources. | Media
Most of my instruments have pickup/pickup or pickup/mic combinations (which Headway, incidentally, don’t recommend), and Dual-channel Microphone
Preamplifier
getting the best out of these setups does require a bit of work — which the EDB2 handled with ease. Having the five-band EQ
Test plots to
and the notch filter available on either channel (or both in the case of the EQ) makes life so much easier. Couple those with
accompany the
the three-position Range switch and you’re close to my ideal acoustic preamp. The phantom-powered XLR mic input available article.
was an extra bonus for me, as normally I’d have to also take out my TC Nova acoustic preamp to be able to use my DPA and
Accusound mics on stage.
Cloud Microphones
Conclusion Cloudlifter
Audio Examples
For simple acoustic pickup systems (active or passive) across a range of instruments, the EDM1 is, to me, possibly the best Audio files to accompany the
and simplest on-stage solution that I have come across to date. Other than the ground-lift/phantom-power issue, there isn’t article.
any reason that I can think of why you shouldn’t just go out and buy one — especially if you play a couple of different types of
instruments on stage. The EDM1 is so good and it is so easy to use. Slate Pro Audio Fox
Dual-channel Microphone
If you don’t have to cope a wide range of differing instruments, the EDB2 gets very close to being a no-brainer buy, although Preamplifier
you’ll have to balance its bulk and the little bit of effort involved in setting up a sound against the benefits of its additional With two channels
and four ‘flavours’
facilities over the EDM1. If you’ve got only one or two instruments to worry about on-stage, or if you have a dual-source pickup
on offer, Slate’s
system (especially if you have internal mics that need power), then owning an EDB2 makes an awful lot of sense. mic preamp
promises plenty of flexibility.
Headway have been enjoying a growing profile recently, and it is really great to see a British company succeeding in a very Does it also deliver on quality?
competitive market sector. Get your hands on either or both an EDM1 or EDB2 and you’ll find out two of the reasons for their
success. . Neve 4081
Four-channel Microphone
Alternatives Preamplifier
Neve believe that
Most acoustic preamps are designed specifically for a manufacturer’s own pickups and, as none that I know of have the there’s scope to
equivalent of a Range control, you’ll be unlikely to find anything quite like the Headway units. There’s a good case for bring classic
equivalence to be made for the Fishman Aura Sixteen and Spectrum DI, provided you use undersaddle pickups and load designs up to date
images appropriate to your instruments. You could also look at the D-TAR Mama Bear, but that particular unit is a serious — and that’s exactly what
investment for a guitar-only solution. they’ve done here, taking their
revered 1081 mic preamplifier
as the starting point.
Radial Tonebone PZ
Pre
SOS Readers Ads Acoustic Instrument Preamp
GRAB A BARGAIN James Dunkley is
£516,050 on the case of the
of Second-User Gear for sale Radial Tonebone
now — don't miss out! PZ Preamp.
Drawmer HQ
Preamplifier & D‑A
Converter
Can a preamp and D‑A
converter successfully
straddle the pro-audio and hi‑fi
In this article:
Screen One, Screen
How Effective Are Portable Vocal Booths?
Them All Screen Tests Buy PDF
Making A Noise Published in SOS October 2014

Reflect & Repeat Reviews : Accessory
Colour Me Bad Portable acoustic screens are designed to make it possible
Auralisations
to record good–sounding vocals almost anywhere. We put
What Else Can You
this claim to the test!
Do?
The Screens On Test Sam Inglis
W
The Test Protocols hen SE Electronics launched their Reflexion Filter Pro in 2006, they created an entirely new product category. The
Studying Acoustics At idea of a portable acoustic screen that can ‘dry up’ recorded sound by preventing unwanted room reflections from
Salford reaching the microphone has obvious appeal to the home recording market, where people often find themselves
Listening Tests working in untreated recording spaces. The Reflexion Filter was a great commercial success, and though certain aspects of its
multi–layered design are covered by an SE patent, the basic concept has been widely imitated by other manufacturers.
Eight years on, there are hundreds of thousands of these ‘portable vocal booths’ in homes and studios across the world. Yet
there is no reliable information in the public domain on their effectiveness, and, as far as we know, none of them have ever
been subjected to rigorous testing. Do they actually do what their manufacturers claim of them? And what, if any, are the
negative side–effects of using them?
To find out the answers to these questions, Sound On Sound teamed up with the
Acoustics Laboratory at the University of Salford. Thanks to their expertise and
world–class test facilities, we were able to compare the performance of 10 rival
products, including both the original Reflexion Filter Pro and SE’s latest RF Space.
The test protocols were devised by Trevor Cox, Professor of Acoustic Engineering
at Salford, and the tests carried out by doctoral student Nikhilesh Patil. Trevor then
oversaw the analysis of the results.
(See the accompanying media page for the audio

files) http://www.soundonsound.com/sos/oct14/articles/vocal-booths-media.htm
Screen One, Screen Them All

We tested 10 such screens, ranging in price from under $100 to almost $400. All
are claimed to reduce the audible influence of room reflections on a recording, and
many manufacturers make the additional claim that their screens can reduce the
extent to which external sounds such as traffic, computer fan noise or air–
conditioning hum are picked up.
The design is similar in almost all cases: a rigid outer shell is curved or angled to
enclose a microphone, and lined with acoustically absorbent inner material. Design
variations include the extent to which the rigid shell is perforated, and the choice of
lining materials: while the SE screens have a complex multi–layered structure, many others use simple acoustic foam, usually
shaped into wedges. There is also a surprising degree of consistency in the dimensions of these devices. With the exception
of the Kaotica Eyeball — which is a completely different design — and the much larger Real Traps booth, all are within 5cm or
so of each other.
The degree to which an absorber can attenuate sound is dependent on frequency. Most of these screens have an inner
lining that is around 2.5cm thick. This, in theory, should achieve complete absorption of sounds where the wavelength is 10cm
or shorter, which corresponds to frequencies of 3.4kHz and above. Below this,

absorption will be incomplete, but will still be useful down to wavelengths of 25cm,
which equates to a frequency of 1.36kHz. In principle, therefore, you would expect
these screens to be most effective at mid and high frequencies, though in designs
where the rigid shell is perforated, this should offer some bass absorption too.
The main way in which portable vocal booths reduce room reflections is by
absorbing the sound before it can escape into the room, though they also prevent
some returning reflections from reaching the microphone. However, as we have just
seen, these screens cannot be equally effective across all frequencies, and will
inevitably reflect some of the source sound back into the microphone, especially in
designs where the shell is not perforated. Because the screens are designed to be
placed in close proximity to the microphone, these reflections will arrive very shortly
after the direct sound, and will thus be audible as coloration of the wanted sound
rather than as reverberation or echo. The concave shape of most screens risks
amplifying this effect through acoustic focusing.
Making A Noise
The two tests devised by Professor Cox are described in detail in the ‘Test
Protocols’ box. The first, which took place in the small reverberation chamber, was Professor Trevor Cox in the anechoic
designed to evaluate the claim that portable vocal booths can help to block chamber at Salford.
unwanted sound such as traffic rumble and computer fan noise. The results, shown Photo: Chris Foster Photography
in Figure 1, are given as insertion losses, which is defined as the sound pressure
level without the screen present minus the sound pressure level with the screen present. Positive numbers indicate that the
screen is providing protection and attenuating the external noise; negative numbers indicate that it is actually increasing the
noise incident on the microphone.
Apart from the Kaotica Eyeball, all the screens display similar characteristics.
Above 2kHz or so, performance is largely independent of frequency, and between
2dB and 5dB of attenuation is achieved. This is brought about by acoustic
shadowing. The greatest level of attenuation — nearly 8dB in some designs —
occurs in the 800Hz–1kHz region, but further down the spectrum, all of these
screens actually amplify unwanted noise within the 200–400 Hz region!
Professor Cox thinks that both effects arise because sound waves diffract around
the edges of the screen — principally over the top and under the bottom, where the
path length is shortest. This gives rise to interference which is destructive in the
800Hz–1kHz region, but constructive in the lower frequency region. The uniformity
of the results reflects the very similar size and shape of the screens (the only one
with significantly different dimensions was the Real Traps Portable Vocal Booth, but
our couriers failed to deliver it in time for it to be included in this test).
SOS Readers Ads Figure 3 shows broadband figures for the attenuation of unwanted noise,
GRAB A BARGAIN calculated from the sum of intensity over the frequency spectrum from 100Hz to
£516,050 10kHz. As can be seen, even the most effective of these products achieves a
of Second-User Gear for sale reduction of only 1.5dB overall, while the least effective provide barely 0.6dB Salford postgraduate student Nikhilesh
now — don't miss out! attenuation of noise. Patil carried out the tests. GLOSSARY: technical terms
explained
Overall, then, these screens can offer a little protection from
external noise, but only in the mid and high frequencies. Where Competitions!
the external noise consists mainly of low and low–mid
frequencies, by contrast, using a ‘portable vocal booth’ might Win Dangerous Music D-
Box (Americas Only)
actually increase the amount of noise that is captured! This is a
real concern in relation to sources such as traffic noise, which
& Sub8 (UK, EU ROW
usually contain prominent low–frequency components. Only)
Reflect & Repeat

Ultimately, though, protecting the microphone from external noise
is a secondary function of these portable vocal booths. Their
main design goal is to minimise the capture of room reflections
Figure 1: The insertion loss achieved by each screen at
from the source being recorded. Professor Cox’s second test different frequencies. Apart from the Kaotica Eyeball, all
was set up in the listening room at the University of Salford, an produce fairly similar results. Note that below 400Hz or so,
environment which is deliberately constructed to be similar to a most of the screens actually amplify external noise slightly!
large domestic living room (albeit with rather more acoustic
treatment!). This test measured the effectiveness of each screen
at preventing room reflections reaching the microphone, and the
extent to which the screens themselves introduce unwanted
coloration.
Figure 4 illustrates the measured effect of one of these

‘portable vocal booths’. The black and red lines represent the
impulse response with and without the screen in place. The big
peak at the start of the impulse response is the sound travelling
directly from the loudspeaker to the microphone. This is the
same with and without the screen, so the (solid) black line is
exactly overwritten by the (dashed) red line. Towards the middle
of the graph, starting 6ms or so after the direct sound, the Figure 3: The broadband noise reduction achieved by each
response without the screen (red) shows room reflections. These screen in our reverberation–room test. Even the best figures
are relatively modest.
are clearly reduced by the presence of the screen.
To quantify how much protection was provided by each

screen, the RMS pressure for the late part of the impulse
responses (starting from 6ms after the direct sound) with and
without the screen were compared. This is expressed as an
attenuation value in dB, where the greater the attenuation value,
the greater the protection.
Figure 5 shows how this insertion loss (averaged in this case

from all the screens on test) varies with frequency. What
screening there is against room reflections is effective only from
500Hz upwards. This is to be expected given the size of the Figure 4: In this diagram, the impulse responses captured
screens, and the thickness of the foam: these devices are of the with and without a portable vocal booth present are
superimposed. ‘Late’ reflections from the room are reduced,
order of 30–40cm in size, and protection is seen from about half but the booth itself causes early reflections.
the wavelength upwards. This is what would be expected from
the simple physics of waves interacting with objects. At lower
frequencies, the screens are too small to significantly alter the
sound.
Figure 6 shows the results for each screen represented as a

single value for attenuation across the 100Hz–10kHz region. As
you can see, the amount of broadband attenuation achieved
ranges from approximately 2.5dB for the least effective, to almost
8dB for the most effective. The Real Traps Portable Vocal Booth
achieves better screening against room reflections than the
others mainly because of its size; conversely, although the
Kaotica Eyeball almost completely encloses the microphone, its
lack of mass means that it is only effective at high frequencies.
Colour Me Bad
No matter what materials are used in their construction, portable Figure 5: This diagram shows the way in which the
attenuation of room reflections varies with frequency,
vocal booths will inevitably reflect sound as well as absorbing it. averaged across all the screens in the test.
What’s more, the close proximity of the microphone to the
screen, and the curved or angled shape that is common to most
designs, make it likely that much of this reflected sound will be
directed towards the microphone.
The early reflections introduced by one of these screens are

clearly visible in Figure 4: the black line shows peaks at 2–3ms
after the direct sound, which are not present in the impulse
response captured without the screen present (red line). The
very short time difference between the direct and reflected sound
means that any effects will be audible as comb filtering, rather
than echo or reverberation. In other words, these reflections will Figure 6: A comparison of the broadband attenuation of room
introduce an unwanted change in the tonality of the source reflections achieved by each of the screens tested.
sound. To visualise the coloration introduced by each screen, the
impulse response taken with the screen present was converted into a spectrum using a Fourier Transform. This spectrum was
equalised to remove the frequency response of the loudspeaker, as measured in Salford’s anechoic chamber. The plots are
shown in Figure 7. Ideally, these spectra should be flat, but none
of the screens on test achieves this. All of the traces show
distinctive patterns of peaks due to comb filtering, though some
introduce more coloration than others. The ones with a distinct
regular pattern of peaks are likely to cause the most problematic
change to the timbre. Some also show a broader change in
frequency response: for instance, the Real Traps Portable Vocal
Booth’s relatively good performance at screening out room
reflections comes at the cost of a significant alteration to the
tonality of the sound being captured, with a 5dB boost centred
around 150Hz and a comparable cut at 1kHz. Figure 7: The coloration spectra for each of the screens
tested.
Unsurprisingly, then, all the screens on test introduce
measurable coloration into the wanted sound. But are these undesirable side–effects actually audible? And do they outweigh
the benefit of the wanted effects, namely a reduction in the amount of reverberation being captured at the microphone? This is
a subjective judgement which needs to be taken by the producer or engineer, and which will depend on the circumstances in
which a session is taking place. For the purposes of comparison, however, Professor Cox created ‘auralisations’ to compare
the audible effects of using and not using the screens as they were set up in our tests.
Auralisations
To create the auralisations, Senior Lecturer at Salford University, Dr Sabine von Hünerbein, was recorded singing ‘The Green
Dog’ by Herbert Kingsley in the anechoic chamber. An EQ was applied to the impulse responses with and without the booth to
equalise the frequency response of the loudspeaker. The EQ settings were based on the measurement of the Tannoy
loudspeaker in the anechoic chamber. Then the impulse responses were convolved with the anechoic singing.
In an ideal world, the use of a portable vocal booth would allow the anechoic recording to be recovered perfectly, but this is
never achieved in this type of auralisation, even if the screen achieved its goal perfectly: some artifacts caused by passing the
singer’s voice through a loudspeaker will remain even after equalisation. However, it is possible to compare the effects of each
screen on the anechoic singing, both with each other and with the impulses taken without the screens present. To make your
own comparisons, download the audio files associated with this article and listen to them yourself. It would be interesting to
see whether the coloration effects can be compensated for with EQ.
Listening through, the effects of the booths are only heard on certain notes within the music. Dr von H nerbein is a soprano,
and her voice is thus at the top of the human frequency range; the notes she sings in the audio example have fundamentals
between about 300 and 750 Hz. Even so, it is clear from Figure 5 that most of the attenuation achieved by these screens
takes place above these fundamental frequencies. Consequently, the screens mostly alter the strength of the harmonics
(overtones) within the reflected sound. The coloration graphs (Figure 7) likewise show the most distinctive effects for most
screens at 1kHz and above. Also, the audibility of the comb filtering will depend on whether the frequencies of the singing note
combine with the peaks and dips of the comb filtering to significantly alter the harmonic series of the note. This won’t happen
for every note. The lower notes, such as the opening few words of ‘If my dog was green’, seem to show differences most
clearly.
There are measurable and audible differences in performance between the screens that were tested. Interestingly, though,
these differences do not correlate in any obvious way with the cost of the screens! In terms of attenuation, for example, SE’s
new RF Space filter offers only a limited improvement over the original Reflexion Filter Pro, despite its being nearly twice as
expensive; and the Auralex Mudguard, one of the cheapest products on test, recorded better results than some of its much
pricier rivals. Another outlier is the Real Traps Portable Vocal Booth: its significantly larger dimensions mean that it is the most
effective at attenuating room reflections, but it also introduces the most coloration. The Kaotica Eyeball is a radically different
design which is effective only above 1kHz or so.
The differences are subtle, and sometimes it is hard to say whether the sound with the screen in place is better or worse
than without. In making this comparison it should be borne in mind that the Salford listening room is intended to provide a
relatively pleasant and controlled acoustic environment: the benefits of the screens may be felt more strongly to outweigh their
disadvantages in other rooms. What is clear is that none of the screens produce something that would be mistaken for
anechoic singing. If you’re forced to record in a bad acoustic environment, portable vocal booths may provide a worthwhile
reduction in the amount of unwanted room reverberation that is captured — but they won’t eliminate it. .
What Else Can You Do?

A ‘portable vocal booth’ is, at best, a partial solution to the problem of unappealing room acoustics. If you are forced to
record in less than ideal circumstances, there are other measures that should be taken too. The first is to ensure that you
are using a microphone with a cardioid pickup pattern. Compared with an omnidirectional microphone used at the same
distance, this will be more sensitive to the direct sound from the vocalist or other source, and less sensitive to reflected
sound arriving at the sides and rear of the microphone. A large–diaphragm cardioid capacitor microphone is the most
common choice for vocal recording, and this was what was used in our tests in Salford’s listening room.
The second measure that can be taken is to hang a broadband absorber behind the performer to trap reflections that
might otherwise bounce off the rear wall and into the sensitive front side of the microphone. This should be as large as
possible — in the absence of specialist acoustic treatment, a thick duvet often works well — and will be more effective if
hung away from the wall. The subjective effect of a well-positioned duvet is often more obvious than that of a portable
screen, and of course there is nothing to stop you using both.
Finally, it should be pointed out that the position of the mic in relation to the screen can make a difference. Our tests
positioned the mic at the centre of an imaginary line drawn across the edges of the screen, as recommended in several of
the product manuals including that of the SE Reflexion Filter Pro. However, James Ishmaev–Young of SE Electronics says
that moving the mic further into the screen should increase the attenuation of room reflections, and will make differences
in coloration between the better and less good screens more audible. We did not test this claim.
The Screens On Test

SE Electronics Reflexion Filter Pro ($249)
The original, iconic ‘portable vocal booth’ remains a best–selling product more than eight years after its launch, and our
tests show that it is still one of the most effective. It differs from most other designs in that multiple layers of different
materials are used in its construction.
SE Electronics RF Space ($399)
The latest iteration of SE’s Reflexion Filter design is the most highly
engineered of the screens on test. Its mounting hardware is improved over
that of the Reflexion Filter, but our tests suggest that it achieves similar levels
of attenuation of room sound. The coloration it introduces is also similar in
level, but shows a different spectral pattern.
www.seelectronics.com
RoXdon VB1
This design features a rigid frame with an internal lining of foam. Unlike many
of the others, its shape can be adjusted using hinged ‘wings’. It adds the least
coloration to the sound, but is also the least effective at protecting the
microphone from external noise and room reflections. This was one of the
more difficult screens to position, thanks to a mounting system which tended
to droop.
www.micsdirect.com
Auralex Mudguard ($109.99)
Another design that uses a hard plastic outer shell lined with acoustic foam,
the Mudguard provided probably the best price/performance ratio in our room
SE Electronics RF Space
reflection test, offering useful attenuation of reverberation without excessive
coloration of the wanted signal. Late arrival meant we were not able to test its
ability to attenuate noise.
www.auralex.com
Real Traps Portable Vocal Booth ($299.99)
Much larger than the other screens, the Real Traps design resembles a
conventional acoustic treatment panel which is hinged in the middle. Its size
means that it provides the most effective protection against room reflections,
but it also imposes the most obvious coloration on the wanted signal of all the
screens. In particular, the auralisation and graphs both show a noticeable bass
boost, though this could perhaps be corrected with EQ.
www.realtraps.com
Editors Keys Studio Series Portable Vocal Booth Pro ($299.99 including
stand)
This is a fairly generic design consisting of a curved metal shell lined with
foam. Unlike most, it is supplied with its own purpose–built stand, which
makes mounting easier. However, our tests suggested that it was not
particularly effective at attenuating room reflections, and adds significant
coloration to the wanted signal.
RoXdon VB1
www.editorskeys.com
Primacoustic Voxguard ($99.99)
This screen employs high–density acoustic foam within a shell made out of
hard plastic, with ports which are said to “prevent excess bass buildup and
resonance commonly associated with stand–mounted absorbers”. Its
attenuation of room reflections was roughly comparable to that of the SE
Reflexion Filter, but coloration of the wanted signal is more obvious.
www.primacoustic.com
Studiospares RED 100
This design employs a “complex sandwich construction of five differing
materials” within a perforated metal shell. Our tests suggested that it provides
reasonably good attenuation of room reflections, but also introduces some
coloration of the wanted signal.
www.studiospares.com
Vicoustic Flexi Screen Ultra
This screen looks a little different, thanks to its wooden shell. This, again, is
slotted, presumably with some acoustic goal in mind. It was one of the better
performers in our tests, offering worthwhile attenuation of reverberation with
relatively little coloration.
Auralex Mudguard
www.vicoustic.com
Kaotica Eyeball ($199.99)
A completely different design from the other screens in the test, the Kaotica Eyeball resembles an overgrown microphone
windshield. Unlike all of the other portable vocal booths we examined, the Eyeball is made entirely from foam, with no rigid
parts or shell.
Because the design almost completely encloses the microphone, the effectiveness of the Eyeball at high frequencies is
very good, both for protection from external noise and room reflections. However, as it is intrinsically lightweight, it
provides very little protection in the mid-range, and no protection at all at low frequencies.
www.kaoticaeyeball.com
Prices include VAT where applicable.
Real Traps Portable Vocal Booth
Editors Keys Studio Series Portable
Primacoustic Voxguard
Studiospares RED 100

Vicoustic Flexi Screen Ultra
Kaotica Eyeball
The Test Protocols

To evaluate the effectiveness of the SE Reflexion Filter Pro and other portable vocal booths, Professor Trevor Cox devised
two test protocols. The first is similar to those used for testing the isolation provided by acoustic enclosures placed around
noisy machines, and was intended to evaluate these screens’ effectiveness at attenuating external noise. This test was
carried out in the small reverberation chamber at the University of Salford.
A loudspeaker was placed in the corner of the reverberation chamber
emitting noise. This creates a diffuse field where sound is incident on a
microphone in the room from all possible angles. It is important to ensure the
sound is arriving from all directions, because this means weak spots are
tested. While someone using a screen in a real–world context will tend to
place the booth between the noise source and the microphone to shield the
mic, this isn’t going to be possible in all cases, for example when traffic noise
is coming through the structure of the building.
A half–inch Bruel & Kjaer omnidirectional measurement microphone was
placed in the body of the reverberation chamber at least 1m from any surface
and at least 2m from the loudspeaker. A B&K HATS (Head And Torso
Simulator) was placed in the singer’s location to simulate shielding and
reflections from the singer’s head and torso. The sound pressure level (in dB)
on the microphone was measured in third–octave bands using an acquisition
time of 30 seconds. The measurement of sound pressure level was then
repeated with the portable vocal booth in position around the microphone.
To confirm that these results were not unduly influenced by the position of
the screen and microphone within the room, the original SE Reflexion Filter
The ability of the ‘portable vocal booths’
was also tested in six different positions. While the fact that the soundfield in to screen external noise was tested in
the room is not completely diffuse means there is some variation in these the small reverberation chamber at
results (see Figure 2), the overall characteristics are very similar, suggesting Salford. A loudspeaker in the corner
emitted white noise, and a B&K ‘head
that the test protocol is fair. and torso simulator’ was used to mimic
Professor Cox’s second test was designed to measure how well the screens the presence of a singer.
achieve their main design goal: to prevent room reflections from the source
being picked up by the microphone. This test also allowed us to measure the amount of unwanted coloration each screen
introduced.
These tests took place in the listening room at the University of Salford. This is a noise–free environment with a
controlled reverberation time of about 0.3 seconds in the mid-range. To make the room configuration more like a typical
home studio, an area of acoustic treatment was removed, giving a clear specular reflection from the wall behind the
microphone. A Rode NT2A cardioid capacitor microphone was set up on a stand 2m in front of this wall, and a Tannoy
System 6 dual–concentric monitor speaker was positioned 22.5cm in front of the microphone — a typical distance for
close vocal recording. (A dual–concentric design was chosen to ensure that the high- and low-frequency components of
the sound arrived at the mic at the same time and from the same direction.)
The general principle is to measure the impulse response between the signal driving the loudspeaker and the electrical
signal picked up by the microphone. This was done with
and without the screen in place, allowing its effect to be
seen. For each test, the room configuration, the
microphone and loudspeaker positions, and settings on the
acquisition system were identical. For each product the
impulse response with and without the screen present was
measured. Where possible, the microphone and
loudspeaker were not moved between the two
measurements (this wasn’t possible for the Kaotica
Eyeball).
Most of the screens on test permit a range of possible
positions in relation to the microphone. Where a particular
position was recommended by the manufacturer, that Figure 2: Insertion losses achieved by a single screen at
different positions within the reverberation chamber. There
position was used in the tests; otherwise, the screen was are some differences, owing to the fact that diffusion of sound
positioned such that the microphone was in the middle of within the chamber is not complete, but the broad pattern is
an imaginary straight line drawn across the edges of the the same in each case.
screen’s arc. Positioning was complicated by the mounting
systems supplied with the screens, many of which did not work very well!
The impulse response was measured between the loudspeaker and the
microphone using a swept sine wave with sufficient duration to capture both
the early reflections and room reverberance. A one-second signal was used.
The sampling frequency was 48kHz. The loudspeaker was driven at a typical
listening volume level to minimise any distortion artifacts created by the
loudspeaker.
In addition to the above measurement, a calibration was undertaken in the
Acoustic Laboratory’s anechoic chamber, where the impulse response
between the loudspeaker and microphone was measured using the same
spacing as used in the listening room tests. These measurements were used
to equalise the frequency response of the loudspeaker, where necessary,
during analysis. In addition, a piece of vocal was recorded in the anechoic
chamber for auralisation.
To test the effectiveness of the screens

at cutting down room reflections, they
were set up in the listening room at
Salford, and a Tannoy monitor speaker
was positioned where a vocalist or other
source would usually be placed.
Some of the listening room’s acoustic

treatment was removed to create an
area of clear wall, similar to what might
be found in a domestic room. A Rode
NT2A in cardioid was used to capture
the impulse responses.
Studying Acoustics At Salford

The Acoustics Laboratories at the University of Salford include anechoic chambers, a listening room and reverberation
chambers, all of which were used in testing the portable vocal booths for this article. Acoustic research has been carried
out at the university for nearly 50 years. The first Acoustics Laboratories were established in 1965, and in 1975 the first
degree (Electroacoustics) was taught. Current courses include the BEng (Hons) Audio Acoustics and MSc Acoustics,
which are aimed at budding engineers who might go on to develop new audio technologies or scientists who might carry
out research in acoustics. Other courses, such as the BSc (Hons) Professional Sound & Video Technology and MSc Audio
Production, attract students looking to work as sound engineers producing high–quality audio content. The Acoustics
Research Centre is the primary partner for acoustics research in the BBC Audio Research Partnership.
Current major projects in audio include examining the future of spatial audio in the home, assessing the audio quality of
user–generated content, and new mathematical models for simulating room acoustics. In recent years, the establishment
of MediaCityUK at Salford Quays has led to an expansion of the University’s commercial work with the audio industries,
including testing many of the acoustic products used in the fit–out of the new studios at MediaCityUK.
The large reverberation room at Salford.
Listening Tests
To download and listen to Professor Cox’s ‘auralisations’ of each screen, surf to
http://www.soundonsound.com/sos/oct14/articles/vocal-booths-media.htm. In each case, you can hear the same piece of
anechoic singing convolved with an impulse response captured with, and then without, the screen present behind the
microphone. The files are in mono, 48kHz WAV format.

Reviews Information
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In this article:
The Old & The New
Korg SDD 3000
Overview Digital Delay Processor Buy PDF
Pedal Play Published in SOS October 2014

Reviews : Effects
Better Than The
Original? A modern stompbox take on a classic digital delay unit —
Alternatives
what’s not to like?
Korg Paul White
SDD–3000 $400
P
pros eople have lusted after ancient guitars and vintage analogue outboard gear for years, but in recent years people have
also started getting misty eyed about early digital equipment such as vintage Lexicon reverbs, as well as certain delay
Extremely musical sound.
Easy to use. units. One digital device to have triggered lustful thoughts is Korg’s rackmount SDD–3000 delay unit, which was
Versatile. launched way back in 1982. It hasn’t done its reputation any harm that U2’s The Edge is a regular user and has an enviable
cons stash of the original rackmount units — or that acclaimed producers, including William Orbit, have extolled its virtues.
The price is high (but not
unreasonable). The Old & The New
summary Given the SDD–3000’s enduring popularity, it’s no surprise that Korg have decided to bring it back to production, just as
An excellent and they’ve revisited the past in their analogue synth line. But the new SDD–3000 is not a slavish reincarnation: yes, you get
characterful delay pedal that everything you need to give you that classic SDD–3000 sound, including a modern take on its preamp stage (often used as a
does so much more than
clean boost with ‘character’), but you’re treated to a whole lot more besides. To achieve this, Korg have harnessed modern
recreate the original. The
pro price is easily justified DSP technology to allow them to add features and improve the specifications. They’ve also put this delay unit into a robust
by the quality. pedal format, which should make it much simpler than the original hardware to incorporate into a live rig, and for guitarists in
particular to control it.
information
$399.99. Despite the form factor, it’s almost an insult to call the SDD–3000 a ‘delay pedal’ because it is so much more than that
Korg US +1 631 390 description suggests. In addition to being able to emulate just about any kind of delay, modern or retro, it can also produce
6800 most of the familiar modulation effects and even incorporates pitch shifting and reverb into some of its party pieces.
www.korgusa.com
Overview
In addition to the original SDD–3000 voicing, today’s SDD–3000
offers a further seven delay types — Analogue (CCD), Tape,
Modern (clean DDL), Kosmic (pitched reverb), Reverse, Pitch
and Panning — and the delay time can be extended from 1ms to
4s. The SDD–3000, Analogue, Modern (DDL) and Tape modes
are pretty self explanatory, but the others are worth explaining in
more detail. Kosmic creates a shimmery ‘crystal’-type effect that
seems to combine one octave of upwards pitch shifting with
reverb, rather than delay. Reverse plays blocks of sound
backwards to fake a reverse tape effect, while Pitch places a
pitch shifter before the delay. Panning mode allows the creation
and adjustment of stereo effects, and there’s even a ducking
function that keeps the delay level low when you’re playing, but
allows it to swell during pauses.
There are five selectable modulation waveforms, with random

and envelope joining the expected waveshapes, and these morph from one shape to the next as you turn the knob. There’s
MIDI for patch changing and patch dumping, and the unit supports mono or stereo operation. Three–way switches for input
sensitivity and output attenuation allow for different I/O requirements, such as guitar, keyboard or effects loop connection, so
both studio and live users are catered for.
One novel feature is that, in addition to stereo delays that bounce from left to right, you can also use the LCR button to give
you delays that bounce left, right and centre. In this mode, independent feedback values can be set for the left, right and
centre delays. It can store more programs than you’re ever likely to need (80), it displays the exact delay time just like the
original, and you can even plug in an external expression pedal for more control. Separate three–position high– and low–cut
filter switches are available for shaping the delay tonality, but despite all this the user interface remains very musician friendly.
By adding modulation to shorter delay times, with the possibility of also polarity inverting the delays, it’s possible to create
flanging, chorus, vibrato, doubling, and pitch–modulation effects. For setting the delay time, you can dial it in directly (and
independently for the left and right channels) or use the leftmost footswitch as a tap tempo input. You’re also able to set 11
different note values, ranging from whole notes to 32nd notes (adjusted when holding down the R Time and Fine buttons) so
that you get the desired delay subdivision when tapping in a straight beat.
To keep the number of footswitches sensible, the memory is arranged as 40 banks of two patches, meaning two different
delay effects are directly accessible when you’re playing. If you need any more than two delay settings per song, you either
have to change banks using the first two footswitches or use a separate MIDI foot controller.
For bypass, you have the choice of active, which preserves any decay tails and also leaves the preamp in circuit, or
(passive) hard, in which case a relay shuts off the delays as soon as you hit the switch. Should you choose to use the optional
expression pedal, you’ll be able to adjust multiple effect parameters at the same time. The idea is that you set the rotary
controls with the pedal at one end of its travel, set them again for the other extreme, then save the patch. Now the pedal
morphs between one set of panel settings and the next so you can, for example, change delay and modulation parameters at
the same time.
The circuitry has a full 10Hz to 20kHz bandwidth, while the
dynamic range of 94dB means that background noise won’t spoil
the party. It’s also gratifying to note that the pedal runs from a
standard centre–negative 9V power supply (included), so it
should play nicely with other members of your pedalboard. On a
practical note, the raised bar running across the front panel is
very effective in keeping feet away from the controls.
Pedal Play The inclusion of an expression pedal input adds to this unit’s
versatility, as does the MIDI connectivity.
The vast majority of what the SDD–3000 does is pretty self–
evident from the front–panel controls, other than setting the note subdivisions, which I’ve already touched upon, and setting up
ducking delays, which requires holding down the MIDI/Sys button and then adjusting the Intensity knob to set the degree of
ducking. Having such an accurate readout of the delay time along with a Fine setting option means that you can set up very
precise timings for those styles that depend on it, and I also approve of the input level metering.
The factory patches do a great job of showing off the range of sounds that can be created, but it’s also trivially simple to set
up your own patches.
In terms of the sound itself, the SDD–3000 doesn’t appear to offer anything radically new, but it somehow manages to
explained
create a warm lushness that few other delay devices can challenge, and even with high feedback settings things don’t get too
messy or chaotic. The delays complement the dry sound rather than fighting with it, producing what might be described as a WIN Great Prizes in SOS
liquid and musically involving result. Competitions!
This decidedly mellifluous character (due to the analogue preamp circuitry?), which carries through to most of the other Box (Americas Only)
delay options too, is achieved without making the repeats sound soggy or dull (other than the mandatory softness of the Win Adam A7X monitors
analogue delay), though you can of course use the top and bottom cut filters to further shape the delay characteristics if you & Sub8 (UK, EU ROW
need to — adding low cut to a delay can be surprisingly effective in keeping things clean. Having separate left and right delay Only)
settings makes it possible to emulate a basic multi–tap delay, though there isn’t the option to emulate a true multi–head tape
delay, which might have as many as four playback heads spaced at different intervals.
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£516,050 I can’t quite put my finger on why the SDD–3000 setting
of Second-User Gear for sale sounds more musical than the modern, cleaner DDL setting —
now — don't miss out! but somehow it does. It has warmth and smoothness but retains
its clarity, and I find it easy to understand why so many people
are fans of the sound. The same goes for the preamp: it doesn’t
do anything obvious, but the sound just seems right.
Pitched delay can be adjusted anywhere from octaves to a

subtle detune. As with many pitch shifters, the sound quality gets
more ‘wavery’ the greater the pitch shift, and by the time you get
to an octave up, it sounds like I imagine it would if somebody
were to tickle an elf.
When it comes to analogue delay, the designers have

managed to nail the ‘best of analogue’ tonality without also
A clearer view of the front panel layout.
including the nasty aliasing and background noise that so many
of those things produced at longer delay times. Having a choice
of modulation waveforms makes it possible to stray beyond familiar territory where just a small amount of random modulation
is great for adding a tape–like wander to the delays.
Plugging in an expression pedal gives you a great deal more ‘in flight’ control if you need it, and those MIDI options might
come in handy if you have a larger rig that works under MIDI control.
During my tests, I fed the two outputs into two different guitar amps and didn’t experience any ground loop hum — though
your mileage may vary depending on the nature of your own stereo setup.
Better Than The Original?

Summing up then, yes, the SDD–3000 may be a bit on the pricey side for a delay pedal, but if your playing style places a lot of
reliance on delay, then I don’t think you’ll be disappointed: it sounds just fabulous. The sound quality, headroom and degree of
control makes it just as useful alongside a DAW or mixing console as with a guitar, and for live use its versatility can also save
a lot of board space, as you may not need those chorus, flanger and phaser pedals anymore. Recommended. .
Alternatives
There’s nothing directly comparable: if you want the sound of the original SDD–3000, then you’ll have to get this or the
original! Comparable in terms of quality are Eventide’s H9 and TimeFactor, which offer greater flexibility, TC Electronic’s
Flashback X4 and Alter Ego X4, and the digital delay models made by Strymon.

In this article:
MTP 840 DM
Lewitt MTP 840 DM & MTP 940 CM
MTP 940 CM Hand-held Vocal Microphones Buy PDF
Alternatives Published in SOS October 2014

Reviews : Microphone
Lewitt Printer-friendly version
MTP 840 DM & MTP 940 Lewitt have made some fine impressions in the studio
CM $249/$599 world with their innovative mic designs. Can they do the
pros same for the stage?
Very well made. Paul White
Flexible switching options.
L
Clean sound with plenty of ewitt have already carved a name for themselves in the studio mic market, but they’re also pushing ahead in the live–
headroom. sound world. All their mics are designed in Austria and then built in their own factory, located in China, which helps keep
cons the costs under control without compromising on build quality. Most of their models fall into the mid–price range, and in
These mics cost more than this review I’ll be looking at two hand–held live mics, one dynamic model and one capacitor model.
their passive dynamic
counterparts, many of which The dynamic MTP 840 DM features switchable sensitivity and a three–position high–pass filter, but most radical is the fact
do a perfectly adequate job that it has active, low–noise on–board electronics, which you can opt to use or not. When engaged, these are powered from
in a live environment. standard phantom power. In active mode the frequency curve is shaped for studio vocals.
summary
Both mics uphold the Lewitt Its super–cardioid capsule, which sits on an isolating suspension, is designed to have a good transient response, while the
reputation for quality and protective basket assembly doubles as a pop/wind shield. The designers claim that the capsule also maintains a more
innovation, and between the consistent directivity across the audio frequency range than most dynamic models, which helps mitigate the effects of tonal
two models (and their change when the mic is moved off–axis and also helps increase the feedback threshold. The frequency response extends
switchable options) there’s from 40Hz to 18kHz, which is pushing the limits for a moving–coil microphone, and the quoted sensitivity is 3.5mV/Pa (–
enough tonal variety to suit
49dBV) in passive mode or either 7 or 14 mV/Pa (–43dBV and –37dBV) in active mode.
just about any voice.
In its active mode, the mic is re-voiced for studio work and offers
information
a dynamic range of 121dB(A) with an EIN figure of 19dB(A). A
MTP 840 DM $249, MTP
signal–to–noise ratio of 75dB(A) is quoted, and the maximum SPL is
940 CM $599
140dB, depending on the gain setting. The amplifier gain can be set
TruNorth Music & Sound
+1 855 878 6668 to 0, 6 or 12 dB, while the three high–pass filter settings (off or
12dB/octave at 150Hz or 250Hz) help balance the proximity effect.
www.trunorthmusicandsound.com
www.lewitt–audio.com
Physically, the mic is fairly conventional in appearance and feel
with a substantial die–cast metal body and a hardened steel mesh
basket, both of which are finished in satin black. The XLR pins are
gold–plated to prevent corrosion. The mic is 51mm in diameter at
the widest point on the basket, its length is 183mm and its weight is
a comfortable 336g. As supplied, the mic comes with a standmount,
a foam wind shield and a storage bag.
All the user settings are made using four recessed slide switches
that are accessed by unscrewing the basket. You need some kind of
pointed device such as a small screwdriver to access these, but this
strategy offers absolute protection against someone changing the
settings during a performance. The switches are all labelled, though
you may need your reading glasses. Tiny LED indicators (which can
be turned off if not appropriate) make it easy to check you have the
correct setting when working on a dimly lit stage.
MTP 840 DM
In its passive mode the MTP 840 DM is actually quite bright sounding, which is no bad thing in a live environment as it helps
cut through the back line, though it can sound a touch sibilant with some voices. Of course you can always roll off a little top
on the desk if it sounds too bright. The key point in this mode is that there’s bags of clarity and the lows have been rolled back
to allow you to work very close to the mic without being overwhelmed by the proximity effect.
Switch to active mode and the tonal balance falls mid-way between a typical live dynamic mic and a studio mic, with warmer
lows and less in the way of added presence. Indeed you might be hard pushed to tell the mic from some capacitor models
when used this way. In either mode you may need to engage the low–cut filters to avoid popping. The passive mode would be
of particular benefit to singers who find that the more common choices of dynamic microphone leave them sounding a touch
muddy. This extra brightness could also help compensate when using less exotic PA speakers.
Though the active option is voiced for studio use, it makes a useful alternative for live use where a warmer and less cutting
sound is required. As you’d expect from the inclusion of active electronics, this mic costs a little more than most typical
dynamic models but it is still very reasonably priced for the quality on offer.
MTP 940 CM
Cosmetically identical and just a hair lighter at 332g, the MTP 940 CM uses a one–inch-diaphragm true capacitor capsule, so
it really can be considered as a studio mic in hand–held form. Here the response covers 20Hz to 20kHz. Again the mic has
been designed to minimise the effects of popping and acoustic feedback, which are always a challenge in live sound.
With a 135dB dynamic range and a maximum SPL handling of 144dB before engaging the pre–attenuation pads, the direct–
coupled circuitry produces a self noise or EIN of just 9dB — better than many dedicated studio mics. Sensitivity is 10mV/Pa (–
40dBV) in cardioid mode and standard phantom power is required for operation.
The MTP 940 offers three switchable widths of polar pattern: wide cardioid, cardioid and super–cardioid. As with the MTP
840 DM there are three low–cut filter switch settings (this time at 80Hz or 160Hz as well as flat) and three switchable pre–
attenuation pad settings. Those familiar Lewitt LEDs show the current settings, and the three slide switches are safely tucked
away inside the basket.
Lewitt say the mic is aimed at vocalists who want to recreate the sound of their studio mics on stage, but equally the MTP
940 CM can be used in the studio by those vocalists who only give of their best using a hand–held mic. This model comes
with the same accessories as the MTP 840 DM.
Compared with the MTP 840 DM, the MTP 940 CM is a very ‘big’-sounding mic, with robust lows balanced by a detailed yet
smooth high end. I tended to use the pattern switch in its standard cardioid position, but having wider and narrower pattern
choices certainly makes it more versatile. As suggested by the manufacturers, the MTP 940 CM does indeed sound like a
large–diaphragm capacitor studio mic, though even with the low–cut filter set at its highest position, I was still somewhat
concerned that the lows might get the better of it if used very close to the mouth, so some extra LF roll–off on the desk might
be required. In situations where it’s not used very close to the source, the tonal balance should be fine. This is a very classy–
sounding mic and is probably best suited to non–rock situations; for example, where a piano–playing singer needs to create a
warm, intimate atmosphere with a studio–like quality to the sound. When fighting it out with a loud band, the MTP 840 DM is
probably more appropriate.
As Lewitt suggest, there’s no reason not to use the MTP 940 CM as a studio mic, either hand–held or standmounted, so if
you need one mic that can meet both your live and studio vocal needs, the MTP 940 CM is certainly a contender. .
Alternatives Audio-Technica
While the dynamic MTP 840 DM in its passive mode has many alternatives, there are only a small number of active AT4047 MP
dynamic competitors, of which one is Blue’s Encore 200. The MTP 940 CM is similarly difficult to place as the only other Multi-pattern
two multi-pattern hand-held capacitor mics I know of are the Shure KSM9 and the Sennheiser e965. Condenser
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Audio-
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of the most
revered mics
on the planet, so when
they release a
commercial version of
the mic preamp they
use for testing, you have
to take it seriously...
Schoeps VSR5
Mic Preamp
Test Measurements
The
following
charts, made
using an
Audio Precision
Analyser, accompany
our review of the
Schoeps VSR5
microphone
In this article:
Special K
MeterPlugs Meter
Heads Up Metering Plug-in For Mac OS & Windows Buy PDF
MeterPlugs Published in SOS October 2014

Reviews : VST plug-ins
K–Meter $49 Printer-friendly version
pros An oft–overlooked advantage of analogue working
An elegant implementation methods is sensible metering. MeterPlugs’ K–Meter brings
of the K–Meter standard, the same approach to your DAW.
with some useful extras Hugh Robjohns
including an inter–sample
T
peak meter. he sample peak meters on most DAWs are derived from an obsolete design intended to address a specific limitation of
Very affordable.
the earliest 14–bit, undithered A–D converters. In these old systems, it was important to record as hot as possible in
cons order to overcome noise and distortion issues. Unfortunately, this concept still pervades the industry today, despite
Not a fully standards– those early technical shortcomings having long been resolved. Modern converters easily match or exceed the dynamic range,
compliant loudness meter.
noise and distortion performance of analogue systems, so there is no longer any technical requirement to record without
summary headroom. In fact, doing so is counter–productive and often reduces the end quality because it over–stresses analogue front
MeterPlugs’ K–Meter ends and monitor chains, as well as some plug–ins.
represents a practical and
very usable alternative to It makes much more sense today, both technically and operationally, to embrace the old–school analogue practices of
standard sample peak working with a nominal operating level and a safety headroom margin, a concept that was discussed in depth in Matt
meters.
Houghton’s Gain Staging article in SOS September 2013 (www.soundonsound.com/sos/sep13/articles/level-headed.htm).
However, to make practical this approach we require a different form of metering, and well–known audio guru Bob Katz came
information
up with an excellent solution almost 15 years ago with his K–Meter format. Importantly, the K–Meter format explicitly ties the
$49.
metering reference level to an acoustic sound pressure level generated by the monitors, and this fixed relationship between
www.meterplugs.com/kmeter
perceived volume and the meter display is a major aid to balancing consistently (see my article on establishing a calibrated
monitoring level in the May 2014 issue: www.soundonsound.com/sos/may14/articles/reference-monitoring.htm).
The spreading adoption of Loudness Normalisation and the use of BS.1770 loudness meters may well render all forms of
conventional digital metering obsolete over the next decade, but in the meantime, the K–Meter format has a lot going for it,
and I am an enthusiastic supporter. Several plug–in manufacturers offer the Katz meter format, but Ian Kerr’s MeterPlugs
version is particularly versatile and cost–effective, and it even includes some useful BS.1770 modes to help smooth the
transition into the loudness normalisation world!
Special K
The K–Meter installer contains AAX, Audio Unit and VST 2.4 plug–ins (32- and 64–bit), compatible with both Mac and PC
operating systems from OS 10.5.7 onwards or Windows XP onwards. Authorisation can be done in seconds with an Internet
connection, or by creating a simple authorisation file if the computer isn’t online.
The plug–in can be displayed in expanded or collapsed modes, with the former revealing the configuration controls (three
rotary switch knobs and three toggle switches) below two pairs of horizontal bar–graph meters. Each bar–graph pair indicates
an averaged level on the upper meter with the peak level below, and maximum values are displayed as numeric readouts on
the extreme right–hand side. The meter ballistics are easy on the eye, and a peak–hold option applies to both the bar–graph
and numerical displays.
The bar–graph scale varies with configuration, but always indicates signals down to 60dB below the selected reference
level. Helpful colour–coding shows green below the reference level, a 4dB yellow band above the reference level, and then
red up to the clipping point. A 4dB blue section above that allows inter–sample peaks to be displayed.
The K–Meter system is actually a family of three meters with differing headroom margins, intended to optimise mixing with
different dynamic range targets. The default mode is the K–20 scale which, as the name implies, provides a 20dB headroom
margin conforming to the SMPTE alignment. In other words, the reference level — the equivalent of 0VU in the analogue
world — is at –20dBFS. The K–14 and K–12 modes provide correspondingly smaller headroom margins and higher reference
levels (–14 and –12 dBFS), which are better suited to typical pop music and broadcast mixing applications, respectively. In
these genres dynamic compression is expected and so less headroom is required in the recording medium. In addition to
these three K–Meter modes, a fourth option provides a conventional ‘full scale’ meter option.
The average level bar–graphs can be configured for a standard RMS setting, conforming to the AES17 recommendations,
as well as two options derived from the BS.1770 loudness metering specification (with a fixed target loudness of –24LUFS).
The ITU–F mode has a fast (400ms) integration while the ITU–S mode has a slow (three–second) integration window. These
correspond with the ‘Momentary’ and ‘Sliding’ displays of standard loudness meters, and the slow mode gives a very useful
indication of perceived loudness while mixing. Although useful and serving as a gentle introduction into loudness metering, K–
Meter is not a fully compliant loudness meter, primarily because there is no indication of the ‘Integrated Loudness’ value, the
only number that matters in loudness normalisation. The gating function specified in the current BS.1770–3 specification is
also missing, which will tend to give slightly erroneous average values.
Three virtual toggle switches enable a peak–hold mode, and

activate separate noise sources for acoustic calibration, while the K–Meter can be viewed in a collapsed mode where it takes
last configuration knob sets the up–sampled internal processing up less screen space.
rate. The peak level algorithm conforms with the BS.1770 ‘True
Peak’ specification when running in the 4x mode, indicating inter–sample peaks with an error below 0.7dB. A key aspect of K–
metering is that the displayed reference level always corresponds to the same acoustic level generated by the monitor
speakers. In the professional world, that would be 83dBC, and the plug–in includes a switchable pink noise source at the
appropriate level to facilitate acoustic alignment. K–Meter’s installation notes outline the calibration process, but 83dBC is far
too loud for most project studios, and my May 2014 article provides a more detailed explanation of alignment taking room size
into account.
Heads Up
In summary, the excellent K–Meter format establishes sensible levels and headroom margins along with calibrated listening
levels, but MeterPlugs’ K–Meter builds on this foundation with an accurate true–peak display capable of revealing inter–
sample peaks, and options to view the RMS average or perceived loudness levels. It’s visually attractive, easy to install and
configure, and works extremely well. If your DAW meters make it difficult to work with sensible headroom, the MeterPlugs K–
Meter plug–in is a cost–effective and pragmatic solution, with some handy extras. Highly recommended. .

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Phoenix Audio Nicerizer Junior

Reviews : Mixer
The Nicerizer Junior aims to offer what analogue still does
best — subtle, yet pleasing warmth and distortion.
Neil Rogers
I
t’s been over a decade since the launch of Phoenix Audio’s ‘Nicerizer’, a high-quality 16-channel summing mixer which,
courtesy of its audio output transformers, can also be used to add a touch of pleasant tonal coloration. It’s also endowed
with a full monitoring section and stereo-width controls. Phoenix’s latest take on this design is the Nicerizer Junior, which
uses precisely the same summing and transformer-output circuitry as the ‘full-fat’ Nicerizer, but dispenses with the other
features — which are useful but not needed by all users — to push down the asking price.
The first thing you notice when you get hold of this unit is how heavy it is, and the reason for this is abundantly clear when
you open the case: it’s jam-packed full of analogue electronics. Eight PCBs each cater for a pair of input channels, with every
channel featuring its own Class A transformerless input stage. The 16 inputs are presented on the rear panel as two eight-
channel D-subs and can accept balanced or unbalanced signals (via a Class-A discrete buffer amp), with no difference in
level.
Each of the 16 channels is treated to its own rotary pan control on the front panel, but there’s no gain knob — the input level
is determined by the DAW, or whatever other device is delivering a signal to the Nicerizer. However, there is a switch to apply
an 8dB boost, should more level be required.
Once summed, the signals are routed in parallel to two independent stereo output stages — so there are four channels in
total, each of which passes through its own Class-A discrete output amp and custom-wound audio transformer. On the front
panel, each pair has a level control that operates on the signal before the transformers, which means it may be used to control
the degree of character imparted by the transformers.
Once I’d acquired the necessary D-sub cables and plumbed

the Nicerizer Junior into my system, it was easy to get started
and I really enjoyed the very straightforward way of working.
Whether keeping the unit’s pan controls in fixed stereo pairs or
mono positions, you can just break out an existing DAW mix over
the 16 available channels and decide precisely how hard you
want to run the mix sources through the Nicerizer Junior’s input
and output stages. The other option is to mix ‘through’ the
Nicerizer, performing all the panning elsewhere (eg. in the DAW),
and focus more closely on controlling the levels being sent into the Nicerizer’s inputs, and driving the output stages, according
to what you perceive this is all doing to the sound.
As with other Phoenix Audio products, I found that the Nicerizer Junior always lent a warm, musical character to the signal
which is very much to my taste, but it works better for some sources than others. I used two different tracks for my review
tests: one was a busy full-band track, featuring hard-hitting drums, bass, guitars, vocals, strings and so on; and the other a
more stripped-back acoustic track.
Listening to the busier track first, the element that I could initially hear being most affected was the snare drum; there was a
distinct thickening effect that increased as I drove the output transformers harder. There was — not just to my ears but those
of a couple of colleagues — a subtle but likeable improvement in the sense of depth and separation. I also noticed a tangible,
though once again very subtle, difference in the perceived width of the mix, which made the key centrally panned elements
(bass and lead vocals in this case) seem somehow clearer and more present.
Any positive effects were much less obvious on the acoustic track. Although I could alter the sound by getting things to
saturate a little, it became much more difficult to decide whether such changes were a clear improvement or simply ‘different’.
It’s so hard to put into words the subtle sort of effect I’m talking about, so I’d encourage you to listen for differences in the
audio examples I’ve included with the review (http://sosm.ag/phoenix-nicerizer-jr-media). I made three example clips for each
of the two tracks: one bounced straight from my DAW, one through the Nicerizer at a moderate level and another driven
harder at both the input and output stage.
Much of the warmth of this device can probably be attributed to the contribution made by the transformers. If you drive the
output stage more, you get a slightly saturated, almost valve-like sound, which is very effective if you like that kind of thing. I
liked the thickening effect it had on the drums, just as I’d expected I would — I find that the heavy transients of things such as
drums often benefit from the sonic effect caused by hitting good quality transformers hard.
But should you add a unit like this to your setup? The answer to that question needs to take account of the effect that using
it would have on your work flow, and balance that against the effect you believe it’s having on your mixes or productions. In my
studio, where I’m often switching between five or six mixes a day, bouncing off several mix revisions to email to clients, it
would perhaps be difficult to justify from a workflow and convenience point of view. If you tend to work more on one or two
tracks at a time, and have less need for switching between projects, then it becomes much more viable and more worthy of
consideration.
Whether you need the summing part of the mixer to achieve the warming effect is an interesting question, too. You could
probably achieve a similar sonic result by running the mix through a pair of nice transformer-balanced preamps, for example.
But if you have need of high-quality analogue summing, this is as good a place as any to have that character on tap — and it
is a nice character that’s on offer here. It’s also useful that you have dual outputs, because this means you’re able to create
SOS Readers Ads both a cleaner and a warmer-sounding mix at the same time, and decide later which one you prefer.
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£516,050 One thing that struck me was that the tracks I was using to test the device had been recorded with my own choice of high-
of Second-User Gear for sale quality mics, preamps and A-D converters. Thus, they’d already benefitted from a little warmth and smoothing, and so there
now — don't miss out! weren’t so many ‘harsh edges’ that required attention. For mixes where the source material is a little harsher-sounding, I
suspect the Nicerizer Junior could really come into its own, and I can think of several previous mix projects of mine that could
really have benefitted from what it has to offer.
Phoenix Audio pitch the Nicerizer Junior not only as a summing mixer, but specifically as a ‘mix sweetening tool’. In the right
scenario I would say that is a very fair description. Neil Rogers
$1999
www.phoenixaudio.net
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In this article:
Bodyswerve
Playback
As Time Stands Still Reader's music reviewed Buy PDF
Simple Hymns Published in SOS October 2014

Reviews : Miscellaneous
Get Reviewed Here!
Chris Korff
Dreamlake
Bodyswerve
Dreamlake vocalist Howard Pike has sent in for review his band’s album Bodyswerve, a “for the fun of it” collaboration
between himself and a keyboardist, a guitarist, a bassist and a programmer (there’s a “walk into a bar” joke in there
somewhere, but I’ll refrain).
It’s a pretty decent attempt at MOR/pop-rock, with some good, catchy songwriting and solid, if workmanlike, musicianship.
I’m also quite impressed by the drum programming — there’s no actual stick-work going on here, but you’d be hard pushed to
tell. The drum fills, in particular, sound extremely organic. The acoustic guitar parts are another technical highlight, sounding,
as they do, rather nicely recorded.
The album is let down a little by the mixing, however: there are
a few level issues, with elements occasionally either poking out
too much, or getting lost, and the hard-panned distorted guitars
can at times be quite distracting, I think because very similar
tones are used for the left and right parts.
The lead vocals deserve a mention here too. Singer Marcus

has a powerful voice and can more than hold a tune, but again
some iffy mixing takes the shine off. The whole album could be
massively improved, I reckon, if the lead vocals were pushed
more up front, and given a more authoritative, weighty tone. As
they currently stand, they sound somewhat raspy, and have a
tendency to be masked by the guitars in the more rocky tracks.
The vocal reverb is also maybe not the best choice, and would
probably be best replaced by something with a shorter tale and a
longer pre-delay. Chris Korff
Jimmy Brewer
As Time Stands Still

There are few swords more double-edged than a talent for pastiche. And Jimmy Brewer has such talent in spades.
Sometimes it’s directed in a knowing way, as on the ‘Blueberry Hill’-quoting ‘Watching Records Going Round And Round’;
elsewhere, there are songs that feel haunted by the ghost of Elliott Smith or John Grant.
Happily, though, Brewer’s talents go far beyond mimicry, and

for the most part, he has the self-restraint to prevent his
influences from overshadowing his own ideas. Unlike many
songwriters, he seems equally comfortable writing on guitar or at
the piano, and there’s no denying his way with a tune. Throw in
the very tasteful arrangements, good playing all round and some
nice production from Owain Fleetwood Jenkins, and As Time
Stands Still is several cuts above your average singer-songwriter
fare. Sam Inglis
jimmybrewer.co.uk
Phillip Sandifer
Simple Hymns
Back in the fourth century AD, music was becoming increasingly
important to Christian worship, and Augustine of Hippo wasn’t
sure this was a good idea. After all, music was a worldly
pleasure, and if singing was too beautiful and compelling, it risked
“inflaming the passions” rather than inspiring properly pious
devotion.
However, Augustine decided against banning it altogether, and through the succeeding centuries, the church accrued a vast
body of hymns and other sacred music. Which brings me to Phillip Sandifer’s album. The title Simple Hymns sums up its
contents very succinctly: it’s a collection of hymns, chosen and arranged with simplicity in mind. For the most part, that means
straightforward songs like the evergreen ‘Be Thou My Vision’, underpinned by strummed acoustic guitar and decorated with
hand percussion, mandolin and fiddle.
There’s nothing wrong with this as a concept, and I can

imagine that in the hands of a top-flight bluegrass band, for
instance, it could be really special. In this case, however, the
overriding musical virtue on display is not so much ecstatic
inspiration as basic competence, and the recording is likewise
workmanlike rather than spectacular. As a result, it all feels a bit
plodding and samey. It’s perfectly inoffensive, but at no point did
I feel my passions inflaming, or experience an overload of
sensual pleasure. I dare say Augustine would have approved.
Sam Inglis
www.phillipsandifer.com
.
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Recording Music On Location

Book Review
Recording on location presents many challenges: this book
will help you meet them. Printer-friendly version
Hugh Robjohns
I
t’s usually fairly easy to record soloists or a vocalist in a typical home studio, but trying to capture a whole band or larger
ensemble playing together can often be quite impractical, and finding an external location to serve as a makeshift studio is
often a better idea. However, choosing and rigging the appropriate equipment, and optimising the location for recording
can be confusing and daunting, but Bruce and Jenny Bartlett’s Recording On Location book addresses this specific subject
area in exhaustively comprehensive detail.
Bruce Bartlett has written many highly regarded books on microphone technology and techniques as well as recording
practices in general, and this latest edition (Second Edition, Focal Press, ISBN 978-1-138-02237-9) is a very readable guide
to making the best of the challenges faced when recording on location. The content is aimed primarily towards those with little
experience of working outside a self-contained studio, but there are also quite advanced techniques and ideas for the more
proficient too — even I found several useful nuggets of information!
Recording On Location was first published in 2007 and this new edition
has been thoroughly revised to include information about the latest
portable digital recorders and consoles, as well as formatting material for
the web and live streaming. The section on classical recording
(orchestral, chamber, quartets, organ, choir, for example) has been
greatly expanded, and there’s a separate section on recording popular
music (rock, country, jazz, R&B, gospel). Each section discusses the
different equipment and techniques appropriate to each genre,
highlighting the importance of careful planning, and contingencies to
cope with the unexpected.
Topics covered include the pros and cons of stereo or multitrack

recordings in different situations and the stand-alone and computer
equipment appropriate for each, equipment setup, power and mic
cabling, dealing with hums, setting recording levels, and monitoring
options. There’s also a section on editing, problem fixing and mixing. On
the classic side of things, the Bartletts go into considerable detail
discussing different stereo, surround and spot microphone techniques
before discussing mic placement and distance, and optimising the spatial
perspectives, acoustics and stereo imaging. To pull all the theory and
advice together, two in-depth real-world examples are also included of
recording a blues band and an orchestra. These sections illustrate the complete process from concept and planning, to setup
and troubleshooting, the actual recording, and through to mixing, mastering and delivery.
I found Recording On Location a very readable and interesting book, crammed full of down-to-earth advice and practical
information which will prove invaluable to anyone contemplating a recording away from the familiarity of their own studio.
Hugh Robjohns.
$39.95.
www.focalpress.com

In this article:
Green Machine
Roland System 1
We Come 1 Analogue Modelling Synthesizer Buy PDF
Thanks For The Published in SOS October 2014

Memories Reviews : Keyboard
Mad Scatter The System 1 is a fully fledged synthesizer in its own right,
Effects
but it can also assume the identity of Roland classics using
Plug–out
its built–in ‘plug–out’ technology.
Conclusion
Paul Nagle
Alternatives
T
Roland railing slightly behind the forerunners of the Aira range, Roland’s eagerly anticipated System 1 is here at last. Its grand
System 1 $599 web site introduction refers back to the System 100, 100M and 700 modulars, although other than a broad remit to be
flexible and sound ‘like Roland’, the connection isn’t obvious. Though the System 1 might be modular under the covers,
pros
its base personality is as a four–voice virtual analogue synthesizer driven by the technology known as Analogue Circuit
Portable four–voice virtual
analogue synth with built–in
Behaviour.
effects.
Fabulous panel that sends
Unlike the TR8 and TB3, there’s been no attempt to replicate the sonic nuances of a specific favourite from the word go.
MIDI CCs. Instead, modelled classics are planned later down the line as ‘plug–outs’, with the SH101 available first as a free download for
Can host further ‘plug–out’ System 1 owners. Other than the option to reside in hardware, a plug–out acts just like a regular plug–in.
synths based on classic
Rolands. The SH101 is Since I had a few weeks with the System 1 before the software SH101 hit the streets, I’ll begin there. It’s a credible stand–
supplied to get you alone instrument in its own right, and one of the smallest, lightest and greenest four–voice synths ever.
started/hooked.
Can serve as 24–bit USB Green Machine
Audio/MIDI interface.
cons Powering up the System 1, you’re blasted with more green than the combined exhalations of Snoop Dogg, Willie Nelson and
The keyboard. Woody Harrelson. All the knobs, sliders and buttons are backlit and, as far as I can tell, there’s no way to turn off the eerie
The performance controls. glow if it gets overpowering. Thankfully, it is possible to deactivate the vomit–inducing ‘LED demo mode’ in which the synth
Only four voices and the shimmers like the Green Goblin on acid if not touched for a while.
analogue modelling isn’t 100
percent convincing. The controls are of good quality and, with no display or screen,
The 101 emulation isn’t the multi–functionality is almost entirely absent. The layout is in
hoped–for ‘complete traditional ‘signal flow’ order but fans of Roland’s other tradition
reproduction’. — the combined bender and mod lever — will be disappointed it
summary isn’t present. In its place is a sprung rotary control encircling an
A tiny polyphonic synth with alpha dial. This mechanism has a choice of two modes — pitch–
a creamy palette of bend or ‘Scatter’ — but it’s hard to see how anyone would rate it
analogue and virtual
as an improvement. Modulation suffers too, reduced to a single
analogue sounds. It’s
on/off button.
stocked with well–chosen
controls and supplied with
the modelled SH101 plug–
The System 1 is petite and wafer thin (472 x 283 x 70 mm)
out. Bundle in an audio and and its black plastic body tips the scales at 2.4kg. Matching the
MIDI interface and the other Airas’ tapering style, the green–framed panel slopes to a
System 1 is an ideal DAW slightly smaller base and the whole thing is so portable it’s a
companion that can operate shame batteries were not an option. In a frame as compact as
equally well stand–alone. this one, it always seems to be the keyboard that bears the brunt of the compromises. Here it’s just two octaves long. In its
favour, the keyboard is full–sized, but the keys are of the ‘minimal travel’ type seen on certain MIDI controllers. For me, this
information
action falls way short of even the lowliest monosynth of Roland’s glory years. To further limit its capacity to delight, the
$599.
keyboard generates neither velocity nor aftertouch. These are received over MIDI by the System 1 but both are interpreted as
Roland Corporation +1
vibrato, which is a bit uninspired.
323 890 3700.
www.rolandus.com The rear panel’s only surprise is its lack of an audio input. This is doubly unfortunate given that the USB port can serve, in
common with the other Airas, as a 24–bit audio (and MIDI) interface. There are conventional MIDI In and Outs too (the latter
capable of soft Thru if necessary) and if the brace of control inputs seem like expressive overkill, don’t get too excited. The
expression input is tied to controlling volume and the brief manual offers no hint of alternate options — or any clue about
swapping the hold pedal’s polarity. There’s an external power supply, which is small and innocuous. Last but not least, the
main output is stereo. This could be a choice that was more about the audio interface functionality because the synth lacks
voice panning or a stereo chorus and the delay is currently mono.
We Come 1
Roland grab a handful of brownie points from the word go for producing a panel so logical and welcoming. Die–hard synth
enthusiasts should be pleased by the carefully selected spread of knobs, sliders and buttons sitting comfortably together
despite the small footprint. Even though the System 1 has eight patch memories, it usually made sense to work directly with
the panel, which was a pleasant reminder of simpler times.
Before I mist up, let’s explore this, the most recent incarnation
of ACB and the first ‘full synth’ built using the technology. The
System 1 has two oscillators, a sub oscillator and a noise
generator; these are mixed then processed by a low– and high–
pass filter. The waveforms are the expected analogue regulars,
but in both single and doubled versions. Further modification The System 1’s rear panel features an input for the external
comes from the ‘Color’ knob, which adjusts the square wave’s power supply, an on/off switch, a USB B port, MIDI Out and
pulse width, the sawtooth’s phase and adds extra harmonics to In, and, all on quarter–inch jack sockets, inputs for pedal and
control, stereo audio out and a headphone port.
the triangle. When you select a doubled waveform, Color sets
modulation speed, plunging you straight into fuzzy ‘VA’
Supersaw, Supersquare or Supertriangle territory. The amount of Color can be modulated too, by sources that include the
three envelopes, the LFO and the sub oscillator. This latter is more radical in theory than reality, or at least I found it
unexpectedly subdued, but there’s no doubt Roland have dished up an effective system for squeezing the most tonal juice
from very few controls.
The pursuit of classic features continues with cross mod, ring mod and oscillator sync. Of these, cross mod is often a tough
test for modelling technology. Admittedly, it fails to convince here if pushed towards its maximum values, but when used
sparingly, the results are interestingly hard and cold. Cross mod is invaluable for synthesizing brittle, metallic tones, just don’t
expect to hear the analogue rawness of, for example, a Jupiter 6.
The ring modulator is a doorway to bell–like oddness, discordant celestas, chimes and so forth. To unleash the ring mod’s
wildest harmonics, you’ll need to find the hidden (and fortunately documented) coarse tune of oscillator 2. With no knob to turn
directly, it’s a two–handed operation involving holding the Ring and Sync buttons then setting the interval using the Scatter
dial. I’d also recommend trying ring mod and cross mod simultaneously, especially if you’re into spiky, fractured distortion and
digital weirdness. Strangely, there’s no master tune control, so it’s A=440Hz all the way.
When oscillator sync is active, the second oscillator is slaved to the first. At the same time, the two–stage pitch envelope
switches only to the slave. Increase the envelope depth and the resulting sync is creamy and very usable, if never quite
matching the lusty scream of a genuine analogue synth. However, those lush multiplied waveforms should already have been
a clue that the oscillators are not the result of obsessive analogue modelling. Diva this is not! It hardly matters because they
sound fine and in particular have a deep and full bass end. The digital wonkiness is only really noticeable at higher pitches (4’
and 2’), where a background distortion creeps in.
If the oscillators are more VA than ACB, the filter fares much better. It is switchable between 12 or 24 dB operation and has explained
a dedicated (and snappy) ADSR envelope. I was quickly won over by its wide sweeps, squelches and Virus–like smoothness.
It’s true that at high resonance you risk a broadside from a powerful whistle, but generally the resonance has a character Competitions!
reminiscent of Roland’s SH1 synth, which is no bad thing. In another similarity to the SH1, there’s a dedicated high–pass filter
with a single cutoff control. Bi–polar controls for envelope amount and keyboard tracking round off what could be Roland’s Win Dangerous Music D-
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The three envelopes all work as they should, with no frills

added or necessary. Ditto for the single LFO. Amongst its six
waveforms, you’ll find sample and hold and the smoother
‘sample and glide’ — it lacks only noise to have the full set. The
LFO’s fade–in time is programmable, which is particularly
important when you’ve only got a button for modulation. It’s your
choice whether the LFO is free–flowing or reset each time a key
is pressed. At its top speed, modulation creeps into the low audio
rates but isn’t quite fast enough for ripping excesses. Actually, I
was more disappointed that its speed wasn’t marked by a
flashing knob, as per the Tempo control. Three bi–polar knobs
set the modulation amount of the pitch, filter and amplitude.
Thanks For The Memories

The System 1 was shaping up as a powerful and full–featured
synth for its size, and with programming so inviting it should take
The plug–out SH101’s interface.
no time at all to populate the eight on–board patch memories.
While this number is far from generous, it’s better than none, and
whenever you need the panel to be live, the Manual button is your friend. Sequencer users will no doubt be glad of the ability
to send the current patch data as a burst of MIDI CCs.
If Roland are stingy when it comes to patch memories, they’re positively miserly when doling out polyphony. Four notes
doesn’t seem like a lot for a 21st Century modelled analogue synth in dedicated hardware, but maybe the short keyboard
implies monophonic operation is the main goal. For extra mono oomph, there’s a unison mode that stacks the four voices on
top of each other. I should also mention the legato button that, in mono and unison modes, ensures the envelopes aren’t
retriggered by legato playing. Similarly, the same button ensures portamento is engaged only when notes overlap.
Mad Scatter
No Aira family member has yet been spared a Scatter function and here its purpose is to perk up the otherwise plain vanilla
arpeggiator. Without Scatter you’d be left with just up, down, or up and down motions over one or two octaves. Bereft of more
wayward directions (such as random), Scatter’s preset patterns are invaluable for adding variety. Since the wheel imposes a
choice of either Scatter or pitch–bend, you can’t bend an active arpeggio as you can on other Roland synths (for example, the
SH101).
Having selected a clock division (from 1/4 up to 1/16 T), you choose a Scatter type by spinning the inner alpha dial. Taking
type 1 as an example, turning the outer wheel clockwise doubles the arpeggio speed. It also introduces modulation of both
low– and high–pass filters and adds cross modulation. The affected controls flash to indicate they’re being manipulated. In
contrast, turning the same wheel anti–clockwise drops the arpeggio to half–speed and introduces comparable tonal mangling.
Since the wheel is sprung, the hold button can be used to freeze any particular favourite setting en route, which is a thoughtful
touch.
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Effects
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After the simple tone control, the System 1’s output is finished off with ‘crusher’, reverb and delay. Crusher is a bit–rate
reducer that seemed rough and more than a little out of place, especially when a classic Roland chorus emulation or modelled
overdrive would have fitted so well. The reverb is better though; a reasonably expansive stereo hall that’s going to come in
handy. The delay offers two controls — amount and time — and when Tempo Sync is active, it helpfully locks to divisions of
tempo or incoming MIDI clock. Triplets are included but dotted delays are not. With no display of delay values, you must locate
each time division by ear, but this is hardly an issue. More limiting is the lack of a dedicated feedback control, although the
preset values are perfectly serviceable. The delay is mono and if further options lurk under the covers waiting to be exploited,
there’s no information yet. Incidentally, Tempo Sync applies to the times of both the LFO and the delay: you can’t choose one
or the other.
Plug–out
As mentioned already, a plug–out operates like a regular plug–in,
but with the twist that it can be transmitted to the System 1. From
that point onwards it can be accessed from hardware without
need of a computer connection. Alternatively, no hardware is
required to play the software synth in your DAW. Roland have
opted for VST3 and AU support so far, so if you rely on VST or
RTAS, plug–outs are currently off the menu. The pre–release
version I started with was only compatible with Logic Pro X, but
this was later extended to encompass 32–bit AU for older
versions. Hopefully this widening of scope will continue. The System 1’s front panel is desktop friendly, measuring just
472 x 283 mm.
Only one plug–out can be hosted in hardware at once. Taking
into account the System 1’s own synth engine, you’ll therefore always have two choices.
The only plug–out available at the time of writing is Roland’s SH101, an acid plastic classic and one of my personal ‘desert
island’ synths. It should also be a sporting challenge for ACB technology. After all, it’s an uncomplicated single–oscillator
design with mixable waveforms, noise and a sub oscillator. Yet thanks to a near–perfect interface, fast envelope, Curtis
oscillator and chirpy resonant filter, the 101 is still highly desirable today.
Having loaded the plug–out and fired it up in Logic, I performed the online activation and then took a few moments to
transfer the code to the System 1. At transfer time, the first eight patches are also sent, although these may be transmitted
and received later individually. Having the SH101 in software means you can maintain a large library of patches despite the
limited amount of storage in the System 1 itself.
Running the plug–out as a 64–bit Audio Unit, I quickly stacked up 10 instances without troubling my Mac Pro in the
slightest. I was glad to discover that several coloured skins were available. These are ideal for differentiating between multiple
copies. The System 1 worked seamlessly as a controller for every SH101 instance during the review period, but as all its
controls spurt MIDI it could serve equally well for other plug–ins.
It was while waiting for the SH101 plug–out to appear that I’d come across confident claims from Roland such as: ‘perfect
replica’, ‘complete reproduction’ and ‘right down to the fine details and odd quirks’. Imagine my surprise, therefore, on trying
the virtual SH101 for the first time, to find no sequencer was present. This is obviously some strange usage of the term
‘complete reproduction’ that I wasn’t previously aware of.
Actually, a number of other differences were soon obvious. I would hardly take issue with the provision of an extra envelope
even if it does slow down the programming of typical 101 patches. Nor do I have a problem with the extra LFO waveforms, or
that the System 1’s effects are retained — you can always ignore them. Looking closer, you realise that the modulation
amounts are like those of the System 1 — bi–polar — which isn’t particularly 101–like. Worse, this effectively halves the
knob’s resolution. Next I realised that the arpeggiator isn’t the same either. On the original, it only becomes active when you
hold more than one note — not so on the plug–out version. I also missed the sliders that are used to assign pitch and filter
cutoff modulation to the bender. However, these are small beer compared to the missing sequencer.
The SH101’s sequencer is wonderful. Requiring just four buttons to program and play it, it’s an accessible step sequencer
with a capacity of 256 notes. Introduce rests and slides, add transposition from the keyboard, and you’ve got the classic recipe
for enough SH101 patterns to last a lifetime. To imagine a perfect replica without the sequencer is nothing short of perverse.
Once my sulk had run its course, I got down to the serious business of audio comparisons, with the old and new synths
placed side by side. Having pressed the Plug–out button on the System 1, all the controls not related to the SH101 are
dimmed. Others take on new, unlabelled roles that you learn by trial and error. For example, the LFO Key Trig button performs
the same three functions as the SH101’s Envelope Trigger switch, with the ‘Gate & Trig’ option indicated by the button
flashing. Since there’s no second oscillator, the mixer is used instead to set the levels of the individual waveforms: square,
sawtooth, sub oscillator and noise. While not as nice as a properly labelled panel, you can cope with it in practice. More
limiting to A/B comparisons was the shortened keyboard — even losing half an octave hurts more than you might expect.
Generally the oscillator sounded authentic and is capable of fat, flappy sawtooth bass and typically hollow square waves.
There were odd giveaway warbles at the highest pitches, but far less than in the System 1 and in typical use I had no
complaints. Comparing filters was less clear cut. Either because it’s a knob, not a slider, or more likely because there’s a
noticeable ‘smoothing lag’, the virtual 101’s response didn’t match the real thing. The smoothing process ensures no stepping
is audible, from the hardware at least. (Adjust with a mouse and the increments are obvious). The greatest differences
between real and virtual 101 were heard at high resonance. Where you dream of the plummy, fuzzy tones of the 101, what
you get in the plug–out version is a harder and more edgy signal. If resonance is kept well away from maximum, the filter
sounds not bad at all.
Later, I compared envelope speed and modulation, finding the LFO’s noise waveform at the last position on the switch
(labelled RND). While at the greatest amounts the effect of noise modulation was slightly more abrasive than my 101, once
you know the score it’s easy to get decent results. The other waveforms handled as expected, except that sample and hold
didn’t lock to an incoming clock signal, as the 101’s does. Hopefully that’s something that can be added later because clocked
S&H is a brilliant addition to a 101 arpeggio.
Eventually, I couldn’t deny that the plug–out version can sound like an SH101 a lot of the time, but it doesn’t achieve this
feat without transitions that shake the illusion like an old Dr Who set. In contrast, a 101 always sounds like a 101. Apparently,
Roland are considering an optional ‘purist’ version closer to the simplicity of the original (ie. single envelope, no bi–polar
controls or effects). And perhaps to silence old campaigners like me, there’s a hint that the sequencer could appear in the
future too. If the extra buttons also materialise, it’ll be incontrovertible proof that ACB is magic.
Conclusion
Classic analogue synths are usually revered for two reasons: their sound and their interface. The System 1 is capable of a
generous range of analogue sounds and I suspect few will care whether there’s circuitry or code inside. Sure, the mask slips
when some parameters are pushed too far, but thanks to a wealth of knobs, buttons and sliders, this is a synth that’s highly
enjoyable to interact with regardless. Apart from its meagre patch memory and limited polyphony, the System 1 holds its own
against Roland’s previous top virtual, the JP8000.
On the other hand, I found the SH101 plug–out was best appreciated when it was spared the indignity of comparison. If it
weren’t for Roland’s bold claims (and the missing sequencer) I’d have enjoyed it far more because once you get past the 101
fixation, it’s a classy–sounding synth and a pleasant alternative voice to the System 1. It didn’t help that the keyboard and
performance controls chosen were no match for those of the original. However, I can see how the requirement for a very
minimal live rig could render them acceptable.
The plug–out concept is loaded with potential and while the SH101 is an exclusive treat for the owners of the System 1,
subsequent plug–outs should be available separately and individually. If the idea proves successful, perhaps it will pave the
way for more hardware in the form of a larger controller with standard keys.
A good engineer can tweak and modify ageing analogue synths many years after they are made but when it comes to
software–based instruments, there’s usually a ‘development window’ in which bug fixes and new features can be added.
When this window closes, remaining niggles have to be endured. Ongoing support is therefore going to be crucial to the
success of the Aira range.
Ultimately, the System 1 today is an affordable, giggable synth that has two takes on modelled analogue sound, both of
which sound good. It comes with Roland’s USB Audio/MIDI functionality, scope for a role as a control surface and the ability to
host future plug–outs. Despite a few misgivings, it’s a worthy addition to the Aira collection. .
Alternatives
If polyphony is important, most virtual analogues offer at least eight notes — and longer keyboards. However, many of the
System 1’s rivals are considerably larger or more expensive. There are now plenty of monophonic genuine analogues
around, and if you prefer a keyboard with velocity and aftertouch, plus a great synth engine and plenty of patch memories,
Novation’s Bass Station 2 is one of the best matches, but there’s also the Korg MS20 Mini and Arturia Minibrute, each with
their own unique selling points. Finally, if Total Integration scores higher than knobs and sliders, Access’s Virus Snow
might be worth a look, although it costs rather more.

Reviews Information
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In this article:
The Ways Of Love
Scuffham S-Gear 2.4
Rooms To Let Amp Simulator For Mac OS & Windows
Scuffham Published in SOS October 2014
Reviews : VST plug-ins
S–Gear $99 Printer-friendly version
pros Two contrasting new virtual amplifiers and a reverb add to
The latest version adds two the appeal of Scuffham Amps’ virtual amp simulator.
excellent new amp models Sam Inglis
and a versatile reverb.
S
Even better value for money cuffham Amps’ S–Gear appeared on the scene in 2012, and quickly established itself as offering a lot of amp simulator
than it already was! for a modest price. Although it offered only three amp models and two effects processors, these were cleverly designed
cons to cover a very broad range of tones, from clinical, studio-clean sounds, through blues growls and country twangs, to
A guitar tuner would be nice. full–on balls–out rock. Drawing on his experience as an amp designer at Marshall, developer Michael Scuffham took great
summary pains to model the ways in which the various stages of an amplifier circuit interact with each other, and when I reviewed the
Scuffham Amps’ S–Gear is version 2.0 in August 2012 (www.soundonsound.com/sos/aug12/articles/s-gear.htm) I felt it came closer than most amp
a deluxe amp simulator at a simulators do to capturing the elusive ‘feel’ of a valve amplifier.
Tesco Value price, and
recent improvements have Since that review was published, Michael has continued to work tirelessly on S–Gear, and many of his updates have been
made it even more tempting. free to existing users. Though the latest release still isn’t an ‘integer’ update, the sum of additions in version 2.4 more than
merits a second examination in these pages. A full list is available on the Scuffham site under ‘Release History’, and as ever,
information some are utilitarian while others extend the feature set. Most notable in the former category are support for the AAX 32– and
$99 64–bit plug–in protocols, allowing S–Gear to be used within Pro Tools, and a more flexible preset structure where each preset
www.scuffhamamps.com can store two amp setups.
The Ways Of Love

These practical improvements are very worthwhile, but of course it’s the new features that will turn heads. Chief among these
are not one but two new amp models. Introduced in version 2.2, the Wayfarer follows the lead of the existing Stealer, Duke
and Jackal in emulating a fairly broad family of related amplifier designs within a single model. In this case, the jumping–off
point is a clean sound which, depending on the setting of a Bright switch, recalls ‘blackface’ Fender Deluxe or Twin circuits,
and benefits from a convincingly sweet tremolo. There’s enough gain on tap to produce grit when you dig in, but if you want to
up the ante, you can switch in one of two lead channels. The first of these gives you the kind of ‘hot–rodded’ Fender sound
developed by the likes of Mesa Engineering, with a typically smooth, mid–focused lead sound, while the latter adds still more
gain, for a sound that Scuffham describe as ‘modern’ and I describe as ‘not my cup of tea, but very effective’.
The three existing S–Gear amp models were all highly

versatile, but Wayfarer takes this to another level. At the flick of a
switch or two, it can go from vintage, chimey Fender cleans to
thoroughly heavy palm–muted mayhem, with much else to
admire along the way. As with the other models, a hidden ‘expert’
panel extends the control set still further, adding the
aforementioned Bright switch plus a switchable three–band
graphic EQ and a three–position switch adjusting the shape of
the tremolo. What’s perhaps most impressive is the way in which
it’s usable throughout the entire range of almost every control.
Even the most bizarre EQ settings rarely sound noticeably
‘wrong’, and the Wayfarer seems to mate well with almost all the
different speaker impulse responses provided. Not only a jack of
all trades, it’s the master of many.
The other new amp is also Fender–inspired, but embodies a

rather different approach, based around the simpler ‘tweed’
circuits of that company’s ’50s designs. Custom ’57 thus
presents only a gain (Volume) control plus Bright and Bass tone
dials, but again, it’s more flexible than you might expect. This is partly because the Bright control is not a straightforward treble
cut/boost, but shapes the ‘contour’ of the sound; in practical terms, what this means is that rather than the sound becoming
too dark at one end of the scale and too piercing at the other, its character changes but it remains very usable throughout.
That gain control, meanwhile, encompasses the full range of ‘tweed’ possibilities. In the first third of its travel, the clean
sound is warm, punchy and direct, exactly as it should be. Turn it up to 12, and I’ve never before heard an amp simulator that
captures so well the wild, hairy, out–of–control output of a Fender Deluxe pushed to destruction. Perfect if you’re in a Neil
Young tribute band and tire of having someone constantly on hand with a fire extinguisher.
Rooms To Let
When I first reviewed S–Gear, one obvious omission was that none of the amp models featured reverb. Soon after that review
was published, however, Michael added the comprehensive Room Thing module, a virtual rackmount unit that complements
the existing Delay Thing and Mod Thing. Room Thing is much more than a simple spring reverb emulation: it is, in effect, a
fully featured reverb plug–in, offering plate, hall, stadium, booth and ‘delays’ algorithms too. User control includes pre–delay,
high- and low–frequency damping, room size and wet/dry mix, plus hidden parameters controlling the speed and depth of
modulation.
Considered purely as a substitute for the spring reverbs built

into guitar amps, Room Thing is perhaps a little too clean to be
wholly convincing, especially at longer decay times, but sounds
very nice, and I had a lot of vintage–themed fun using the pre–
delay control to turn a short spring reverb into a slap echo. The
other algorithms are also highly usable, whether you want some
realistic room sound around your virtual amp, or the spacey
sound of a plate with the modulation cranked right up. I did run Custom ’57 has the fewest controls of any of the S–Gear
into one glitch where the ‘wet’ part of the reverb seemed to mute amp models, yet is still capable of producing a good range of
sounds.
itself for no reason, but in general, this is a very well thought–out
and worthwhile addition to S–Gear. GLOSSARY: technical terms
explained
All in all, then, S–Gear is a product that goes from strength to strength. Despite the many improvements that have been
added since its launch, the asking price remains unchanged (except for exchange rate fluctuations!), and its affordability Competitions!
makes it easy to forgive the continuing absence of a guitar tuner, which I think would now be my number-one feature request.
In fact, S–Gear is so good, and such good value, that it’s well worth checking out even if you already have amp simulator Win Dangerous Music D-
Box (Americas Only)
software installed. It’s been a pleasure to review and a pleasure to use — what more can I say? .
& Sub8 (UK, EU ROW
Only)
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In this article:
Fully Featured
Sonic Farm Silkworm
On The Silk Road Microphone Preamplifier Buy PDF
Spinning Out Published in SOS October 2014

Reviews : Preamp
Technical
Specifications This 500–series preamp boasts huge tonal versatility.
Sonic Farm Bob Thomas
Silkworm $700
I
’ve been a late adopter of the 500–series concept, but now that I’ve finally got into the idea of having a bunch of
pros
interchangeable and relatively affordable hardware plug–ins in a ‘Lunchbox’ (or two), instead of a whole wall full of
Variable sound character,
with three switchable
expensive 19–inch rack units, it’s proving irresistible. Recently, I’ve been on the lookout for a couple of pairs of high-quality
voicings. microphone preamps, so Sonic Farm’s latest creation, the Silkworm 500–series solid–state mic/line/instrument preamplifier,
Cinemag input and output came to me for review at a most opportune moment.
transformers.
Output can be switched to Fully Featured
solid-state.
cons Sonic Farm describe their Silkworm as a “clean” preamp, by which they mean that its gain stage is designed for maximum
Once you’ve heard one, transparency, with a low distortion transformer at the microphone input and a FET buffer IC on the instrument input. But the
you’ll want two! word ‘clean’ tells only part of the story, as this preamp boasts plenty of tonal options. The signal passes to a discrete op amp
summary circuit that drives either a solid–state balanced line driver or a ‘100 percent iron’ output transformer. The range of available
The Silkworm is an
gain is controlled by both a three–position (High/Low/Medium) gain switch and the variable trim control, allowing for a
extremely versatile, great– maximum of 66dB on the mic input and 42dB on the instrument input, and a maximum output of +28.7dBu.
sounding microphone and
instrument preamplifier. The Besides the expected switches for phantom power, ‘phase’ (polarity) and a –20dB pad, the
Cinemag input transformer, Silkworm’s front panel also carries a somewhat mysterious three–position
switchable voicings and (Smooth/Present/Warped) Vibe voicing switch. The manual demystifies the switch a little,
switchable transformer and describing it as a “complex impedance manipulator”. In the Present position a mic ‘sees’ an
solid–state output stages input impedance of 8kΩ, and this produces the most natural–sounding frequency response.
mean that it can deliver a
The other two positions bring in capacitive/resistive networks that, in the Smooth setting,
sonic character ranging from
modern to vintage and from
reduce the higher frequencies somewhat, the result of which is a ‘warmer’ sound. The
clean to coloured. Warped setting, on the other hand, mildly boosts the high-mid and treble frequencies,
typically at around 8–10kHz, depending on the mic’s output impedance, and its reaction to
information the Silkworm’s network loads.
$700.
www.sonicfarm.com
Another design feature worth remarking on is that there are, by default, no coupling
capacitors in the main signal path; the only ones that appear are those which can be
switched in by the user, as part of the Vibe voicing circuit. This approach eliminates phase
shifts — that is, it allows for a ‘cleaner’ sound — while an active dual–stage servo loop
minimises any DC offset (the purpose for which coupling capacitors are normally used).
On The Silk Road

Once you’ve familiarised yourself with which legends apply to which switches, operating the
Silkworm is entirely intuitive. With the Vibe switch in the Present position and the output
signal flowing through the solid–state line driver, the Silkworm is the epitome of clean, with
the input transformer adding just a hint of character that manages to keep things from sounding too clinical.
Switching so that the signal is routed through the output transformer, the sound starts to warm up, and there are additional
variations to be found in upping the gain and pushing the Silkworm harder (for more characterful results, passive attenuation
may be required if your mixer can’t cope with +28.7dBu). At first, I tended to prefer the sound of the transformer output, but as
I grew accustomed to the Silkworm, I often found myself switching over to the solid–state option when I needed a little more
detail or clarity without disturbing the tonal balance.
The Vibe switch is the Silkworm’s killer feature, as it gives you two additional and very different sonic starting points to work
from or with. For the way that I approach recording, I found that the most effective way of working was to start ‘flat’ in the
Present mode, build a sound that worked using either the solid–state or transformer output, and then experiment with the
Smooth and Warped settings and the two types of output. Once familiar with how these Vibe settings interacted with my own
mics (moving-coil dynamics tended to be the most obviously affected), I found myself swapping my mics around as well.
The one aspect of the Silkworm that I liked above all else was that getting the best from every source and every mic always
resulted in a different voicing setup. If you do the maths, you’ll realise that the Silkworm produces six distinct sounds, and that
these are further multiplied by the number of different mics that you own — and that’s before you start experimenting with
pushing the gain up to ‘drive’ the output transformer.
The Silkworm is an extremely versatile, and great–sounding mic and instrument preamplifier, and it handled with ease every
microphone and sound source that I tried with it. The range of possible voicings (modern to vintage, clean to coloured) meant
that I could tailor the Silkworm’s response to capture sounds as I wanted to hear them. The Silkworm is also an excellent
preamp/active DI for guitar and bass and, with the output transformer in circuit, it makes a great device for re-amping.
Spinning Out
There are plenty of high-quality preamps available now, but I have to say that the sonic possibilities offered by Sonic Farm’s
Silkworm made me experiment more than I’ve tended to in recent years — and that alone makes it a must–buy in my book.
And if you’re already into experimenting, it should be equally desirable! In short, if you’re in the market for a preamp of this
quality, you really ought to audition a Silkworm. I think that you’ll be seriously impressed. .
Technical Specifications
Fully discrete gain stage.

Full DC path — no coupling capacitors.
Servo control for minimal DC offset.
Microphone input impedance: 8kΩ (for middle Vibe–switch setting only; more complex for the other two).
Instrument input impedance: 2.2MΩ.
Frequency response: 5Hz to 50kHz ±3dB.
Gain stage THD at 1kHz: <0.0005%.
Input mic transformer THD at 0dBu: 0.03% at 100Hz, 0.01% at 200Hz, 0.001 % at 2kHz, 0.0007% at 10kHz.
Output transformer THD 20Hz to 20kHz up to 10dBu: <0.15%; up to 20dBu <0.2%; up to 28dBu <1%.
Maximum gain 66dB (mic) and 42dB (instrument).
Maximum output level: 28.7dBu.
Cinemag input mic transformer.
Cinemag 100% Iron output transformer.
Switchable transformer or solid–state balanced output.
Three–way character (Vibe) switch.
Gain regulation: 3–position range switch and trim control.
Weight 1.1 kg.
Dimensions 24 x 20.5 x 11 cm.
Neve 1073LB &
1073LB EQ
500-series Microphone
Preamplifier & Equaliser
Neve’s venerable
1073 preamp and
equaliser are both
now available in
API’s popular ‘Lunchbox’
format. Were they worth
waiting for?
Cloud Microphones
Cloudlifters
In-line Microphone
Preamplifiers
Do these in-line
mic preamplifiers
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Test plots to
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Audio Examples
Audio files to accompany the
article.
Slate Pro Audio Fox

Preamplifier
With two channels
and four ‘flavours’
on offer, Slate’s
mic preamp
promises plenty of flexibility.
Does it also deliver on quality?
Neve 4081
Four-channel Microphone
Preamplifier
Neve believe that
there’s scope to
bring classic
designs up to date
— and that’s exactly what
they’ve done here, taking their
revered 1081 mic preamplifier
as the starting point.
Radial Tonebone PZ
Pre
Acoustic Instrument Preamp
James Dunkley is
on the case of the
Radial Tonebone
PZ Preamp.
Drawmer HQ
Preamplifier & D‑A
Converter
In this article:
Mural Volume 1
Spitfire Audio Mural Symphonic Strings
Short Hop Sample Library Buy PDF
Mural Volume 2 Published in SOS October 2014

Reviews : Sample / Sound / Song Library
Extra Legato
Mapping The Future Spitfire Audio enter deep symphonic waters with their
Brush With Destiny latest string library.
Alternatives
Dave Stewart
Microphone Positions
L
& Panning et’s face it, recording a symphonic string section is beyond most people’s means. Once you’ve factored in studio fees, a
U What? single session with a full–sized group would cost about as much as buying a small house in Bulgaria. Apart from
robbing a bank, what’s a cash–strapped media composer to do? Spitfire Audio have a solution: buy their Mural
Spitfire Audio
Symphonic Strings sample library and you can have 60 of the UK’s finest string players sitting on your hard drive, ready to do
Mural Symphonic Strings
your musical bidding 24/7 for the current price of a 32GB Apple iPad Air. If you’re worried about how this impacts on session
£479
players’ livelihoods, a consoling thought is that Spitfire pay their musicians a voluntary royalty from sales.
pros
Features 60 of London’s top At the time of writing, Mural is available in two separate volumes: the first contains essential articulations (including true
string players. legato samples), while the second adds more fundamental styles such as trills, effects and legato portamento slides. There’s
Section sizes are as large
no duplication of sample content across the two volumes, but the additional legato articulations in Volume 2 won’t work unless
as they come.
Recorded in Air Studios’
you already have Volume 1 installed. That apart, anyone considering using Mural as their main string library would be well
Lyndhurst Hall from five mic advised to start off with the first volume, since it alone contains the meat–and–potatoes styles necessary for general string
positions. writing. Unless you’re only interested in some of Volume 2’s more specialised articulations, purchasing it first would be like
Great performances, sound buying the wallpaper before you’ve built the wall.
and ambience.
cons Mural Volumes 1 and 2 are available only as downloads
Not cheap, but this is Savile (33.5GB and 30.6GB, respectively) from Spitfire Audio’s site.
Row quality, not Old Kent Both require the full version of Native Instruments Kontakt 4 or 5,
Road. and will not run on the free Kontakt Player or in any other
Mural Volume 1’s second sampler format. Since their recording conditions and
violins and violas lack presentation are identical, you can freely mix and match patches
certain articulations. from the two collections: this is explored in Volume 2 in the
summary shape of combination patches incorporating samples from both
Size is a factor in the libraries.
orchestral world, and the
large sections and spacious Upholding the Spitfire Audio tradition, the string ensembles
Air Lyndhurst Hall sound of were recorded from multiple microphone positions (details
Mural Symphonic Strings
below) via an analogue tape signal path in Air Studios’ Lyndhurst
are certainly scaled for the
big screen. Taken together,
Hall. Unlike the strings in Spitfire’s Albion series (which feature
its two volumes deliver a mixed–instrument octave and unison ensembles, Mural’s
great array of highly usable sections were recorded as separate, unison groups, allowing you to make your own decisions on orchestration and octave
performances (including true doubling. The section sizes equate to those used in a full symphonic orchestra: 16 first violins, 14 second violins (a completely
legatos), the recording different set of players), 12 violas, 10 cellos and eight basses, making a total of 60 players. There are no layered ‘ensemble’
quality is fabulous and patches, but with a little entry–level Kontakt programming it’s fairly easy to create your own. (You can read the SOS reviews of
multiple mic positions make
Spitfire Audio’s Albion at www.soundonsound.com/sos/oct11/articles/spitfire–audio–albion.htm and
surround mixing a doddle.
It’s not cheap, but if you take
www.soundonsound.com/sos/feb13/articles/albion–2–loegria.htm.)
the plunge with Volume 1
you’ll soon be thinking about Mural Volume 1
saving up for Volume 2!
The dozen or so articulations played by Volume 1’s string sections cover familiar territory, but one exception is the ‘sul
information ponticello’ (bowed on the bridge) style: this thin, icy timbre has a slightly menacing atmosphere, and when played by the
Mural Volume 1 &
double basses it takes on a kind of muted Arctic grandeur. A more conventional sense of threat is conveyed by the tremolos,
Volume 2 £478.80 each. particularly when you start quietly and slowly increasing their volume; unfortunately tremolos aren’t played by second violins
www.spitfireaudio.com and violas in this volume, but the other sections’ tremolo performances are excellent.
Looped, three–dynamic vibrato and non–vibrato sustains (‘longs’ in Spitfire’s parlance) are played with and without mutes,
and both options sound luxuriant and expansive: the muted ‘con sordino’ patches are particularly opulent, and being real,
separate performances rather than a filtered version of the unmuted samples, you can layer them with non–sordino patches to
excellent effect. I found that a three–way unison layering of con sordino first violins, second violins and violas produced an
amazingly sheer, lush symphonic texture.
In a similar vein, the ‘flautando’ sustains’ gorgeous breathy

tone is especially attractive in the low register, and I was also
taken by the soft, emotional quality of the basses’ harmonics.
This playing style is hard to control in real life, and so benefits
from being sampled and tamed! For some reason the cellos
harmonics patch (which sounds great played chordally) was
silent when first loaded. If you experience the same problem,
click on the ‘tree’ miking icon on the GUI to re–load the samples,
then re–save the patch.
Mural’s Overview panel, showing a large template which
Like all contemporary orchestral libraries, Mural features mod combines articulations from Volumes 1 and 2.
wheel–controlled dynamic cross–fading for its long notes. Spitfire
have totally nailed this technique: swells and fades sound smooth, natural and in tune, a great way of adding expressive,
subtle (or otherwise) dynamics to your parts. And that’s just as well, as Volume 1 contains no played crescendos or
diminuendos.
Interval–based legato sampling is a strong point: I very much enjoyed Volume 1’s ‘fingered legato’ patches, in which both
violin sections turn in positive, vivacious performances, the viola players throw caution to the wind and the low strings combine
strength, elegance and lyricism. My only complaint is that in the review copy, the violas’ legatos didn’t work properly on
upward major seventh and octave intervals in the G4 to Bb4 range. Spitfire’s Pest Control team promise this bug will be
exterminated in a free update.
A word on the low strings: superb. Layering cellos and basses

in octaves gives the classic, instantly recognisable orchestral low
strings sound, highly effective for sustains, legatos and
spiccatos. I’m sure that as I write, this great, sonorous noise is
being pressed into service in several UK TV music scores! If you
want to try it yourself, load the cellos and basses ‘Long’ patches,
select the basses patch, click on the little ‘spanner’ icon on the
left of the GUI in a yellow box and set Transpose to –12
semitones.
Short Hop Mural’s Kontakt GUI ‘Expert panel’, showing a ‘light’ palette of
four articulations, the five–way microphone mixer and various
On with the shorts (as we tennis enthusiasts say). As is often controllers. Select the articulation you want by pressing one
stated on these pages, media composers need tight, biting of the pink keyswitches.
short–note string articulations for their tense action scenes, and
Mural’s spiccatos are well up to the mark. As well as being just the ticket for accompanying movie mayhem, they also shine
playing a fast–moving JS Bach piece in one of Spitfire’s product demos, and Stravinsky himself would have admired their GLOSSARY: technical terms
explained
rhythmic incision and precision.
From time to time I earmark certain sounds which I know will inspire me to compose. Such a sonority is Mural’s low strings Competitions!
playing pizzicato in octaves: the combined effect of the weighty, elephantine string plucks and Lyndhurst Hall’s misty, wafting
ambience is awesome. The snappy ‘Bartok pizzicato’ style also gets a look–in, along with some nice, clacky ‘col legno’ bow Win Dangerous Music D-
Box (Americas Only)
hits, but neither style is played by the second violins and violas, you need Volume 2 for that.
The enticingly named ‘Time Machine Short notes’ patches use & Sub8 (UK, EU ROW
Only)
Kontakt’s time–stretching algorithm to artificially alter the length
of spiccatos, pizzicatos and col legno samples via a ‘Stretch’
fader on the GUI. This would be helpful if you need
exaggeratedly short spiccato notes or an ultra–brief pizzicato
stab. Mural’s interval–based legato patches have their own
dedicated controls.
My only complaint about Mural Volume 1 is the lack of certain
articulations for second violins and violas, which has resulted in the second violins ending up with only five playing styles.
Most of these omissions are put right in Volume 2, but if Volume 1’s contents really are ‘essential articulations’ (and most
would feel that tremolo is exactly that), then they should be provided for all five sections — especially at this price.
Mural Volume 2
Once you get hooked on Mural Volume 1’s ‘wall of sound’ (geddit?), you’ll probably want to branch out into more decorative
performance styles, and there a tasty selection in Mural Volume 2. As well as supplying tremolos for second violins and violas,
Volume 2 adds vibrant, two–dynamic trills for all sections. ‘Measured tremolos’ are fast, repeated 16th notes played in a
choice of 150 and 180 bpm fast tempos. The latter is pretty hectic, but by simply clicking on a button on the GUI, you can
make these patches sync to your host tempo. That deactivates the sample loops, revealing that in real life the players
galloped along for two bars of 4/4 before ending on a final short note. Use these samples for exciting rhythm passages
(obviously), but also for injecting rhythmic energy into a melody line.
I had a whale of a time playing the ‘FX’ patches. The effects samples fall into two distinct, drastically contrasting categories:
in the red corner, a set of wild, screeching upwards and downwards glissando slides, suitable for all manner of unhinged
musical madness and particularly scary when played by the basses. In the blue corner, some beautiful, gentle, evolving and
subtly swelling non–vibrato sustains with a hint of sul ponticello in the bowing. Spitfire call the latter samples ‘Tense’, but I
found them to be quite relaxing — a lovely strings texture to play over.
In the same vein, no–vibrato, muted sul ponticello samples have a dreamy, detached and pleasantly synthetic quality which
is very effective in the high register. The first violins’ ‘sul tasto’ (bowed over the fingerboard) style is also beautifully delicate,
producing a silvery, ethereal tone that the makers uncharitably describe as “timid, naive and heartbreaking”. An exasperated
record producer once described a well–known singer’s studio performance to me in similar terms, though I recall he used
rather more robust language.
Returning to the violence and mayhem, the ‘sul pont distorted’ articulation is a gutsy, rough–edged noise made by
vigorously ‘digging in’ the bow over the bridge: the first violins’ rendition of it sounds like the Devil’s Hoedown, while the
basses’ effort manages once again to sound frighteningly dramatic.
In addition, there are straight long and short notes designed to supplement those in Volume 1. ‘Molto
vibrato’ sustains feature the passionate, intense ‘espressivo’ style which is a staple of romantic string
writing; combination long–note patches fuse those samples with Volume 1’s medium–vibrato sustains, the
‘molto vib’ guys taking over at high dynamics. Cross–fading between the two produces a satisfyingly huge
volume surge. Shortish notes of 0.5 and one second duration can be used for medium–paced melody lines,
or shortened into more staccato deliveries via Kontakt’s Time Machine.
The
Extra Legato entertainingly
named ‘Close
Mural Volume 2 introduces two new legato styles for all sections except the basses (an omission that will Mic Pan
be put right in a free update). The ‘bowed legato’ articulation has a distinct bow attack on each note, giving Collapser’
allows you to
SOS Readers Ads melody lines more emphasis while retaining the legato smoothing effect between notes. I can imagine reduce the
GRAB A BARGAIN using this articulation in a soaring theme accompanying a movie’s opening credits. The ‘legato portamento’ stereo width of
articulation introduces a lovely pitch slide between notes: use it sparingly for occasional, impassioned the Close mic
£516,050
position down
of Second-User Gear for sale octave leaps, or liberally to unleash the wonderful ‘Bollywood Strings’ melodic style, a joyful and exuberant to mono prior
now — don't miss out! sound to melt the heart of the most grim–faced UKIP voter. As far as I can tell, the speed of the portamento to panning it.
slides can’t be altered by users, but I found it to be perfect the way it is.
The three legato options in the two volumes sound uniformly great, and the portamentos are among the best I’ve heard.
Playing ranges are generously wide, and the violas positively sing out in the upper register. Only one caveat: as mentioned
earlier, the Volume 2 legato patches won’t work without Volume 1 installed.
Space doesn’t permit more than a cursory glance at Mural’s innumerable technical features, but
dedicated tech–heads can download the manuals from Spitfire’s product pages and pore over the
fine details. In short, MIDI CCs can be used to control vibrato on/off, release duration, legato
interval speed and legato attack intensity. A new ‘Tightness’ control (CC#18) globally trims off the
initial, almost subliminal attack of notes, resulting in a more immediate rhythmic response. A new
Mixer Menu also adds some handy facilities, though its graphic controls are so tiny you might easily
overlook them! The Velocity Curve
control determines
While its great to have these features to hand, most people will be happy to play Mural’s patches instruments’ dynamic
as they stand and never bother to investigate the library’s user–customisable features. Nothing response to touch. Four
preset curves are
wrong with that, and happily, these patches do sound great straight out of the box. available.
Mapping The Future

Mural is part of Spitfire’s British Modular Library (BML) range, which includes Sable, a deeply sampled, four–volume chamber
strings group initially released in February 2013. The makers say they intend to expand Mural along similar lines to Sable, with
new articulations set to include fast legatos and run–making patches. All instruments in the BML range are homogenously
recorded with correct orchestral positioning in order that they can work together instantly as a unified group.
Also available in Spitfire’s BML series are individual flutes, trumpets, trombones, low brass and French horn section titles,
with further woodwind releases in the pipeline. Add Spitfire Percussion to the mix (see the SOS review at
www.soundonsound.com/sos/feb11/articles/spitfire-percussion.htm), and you have the makings of a complete orchestral
sample library recorded with British musicians at Air Studios, a prospect many orchestral sample users will view with relish.
Brush With Destiny

Spitfire have been here before: back in 2008, their bespoke ‘Symphonic Strings’ library also featured large string sections
recorded at Air Studios, but that collection was only made available to a limited number of invited licence–holders, which felt
like the musical equivalent of a gated community. Boasting larger low strings sections, a choice of mic positions and an
enhanced set of technical features, the all–new Mural Symphonic Strings opens up the gates and invites in the riff–raff —
sorry, I mean the general public.
Given the title of this excellent sample collection, I felt it my journalistic duty to work a few wall–related puns into this review,
but failed utterly when it came to finding a suitable ‘graffiti’ metaphor. That’s probably because if there was any writing on
Mural’s wall, it would be executed with an artist’s brush, rather than sprayed out of a can. This is quality stuff: an enjoyable,
sonically rich, beautifully performed celebration of British musical artistry, and another brick in the wall of Spitfire’s expanding
orchestral empire. .
Alternatives
Few orchestral strings libraries enlist such a large cast as the 60 performers featured in Spitfire Audio’s Mural. Two
notable exceptions are East West Hollywood Strings (312GB) and Vienna Symphonic Library Appassionata Strings (total
19.4GB), which feature 57 and 56 players respectively.
Other high-quality, all–in–one string ensemble collections include the 254GB Berlin Strings, Cinesamples Cinestrings
Core (50GB) and Cinematic Strings 2 (38GB). If the lack of second violins doesn’t bother you, 8Dio’s Adagio series
(available as separate violins, violas, cellos and basses volumes, these total approximately 160GB) and Audiobro’s LA
Scoring Strings (24GB) both offer flexible section sizes and also contain solo instruments.
Microphone Positions & Panning

At the time of writing, Mural Volumes 1 and 2 feature five microphone positions: Close, (Decca) Tree, Ambient, Outriggers
and Leader. New in this library, the last position is a mono spot mic trained on each section’s principal player: it carries
discernible spill from the rest of the players, but the bleed is considerably quieter in the low strings, which means that you
can use the Leader mic to do a reasonable facsimile of a solo cello or double bass. This miking can also be used to add
overall definition, and sounds particularly effective when mixed in with the outrigger and ambient signals.
Spitfire plan to release additional Stereo, Close Ribbon and Gallery mic perspectives for Mural: the first features a more
direct, immediate sound and the latter, very distant, miking shows off the hall ambience to full effect. Recording engineer
Jake Jackson has also created three custom mixes, two in the classic cinematic/symphonic mould and the third optimised
for pop.
Mural’s five string sections were recorded in situ, with first violins positioned to the left of the stereo image, second
violins and violas respectively slightly left and right of centre and the low strings appearing on the right. If you’d like to
reposition a section, Spitfire have provided a control for the Close miking which enables you to reduce its stereo width and
pan the narrowed signal anywhere you like in the stereo picture. It would be nice to see this feature extended to all the mic
positions, starting with the Decca Tree!
U What?
Spitfire Audio believe the keyswitch method of changing articulations used in most orchestral libraries has become
outdated, and should be replaced by a MIDI CC (continuous controller/control change) system. There are compelling
arguments for this: the MIDI notes used for keyswitches mess up the look of your score notation, and you have to roll
sequences back before the keyswitch occurs to select the correct articulation for a passage. Most DAW’s can chase CC
commands, which gets round the latter annoying problem.
In an attempt to unify articulation switching across Spitfire’s entire product range, the company has devised a system
called Universal Articulation Controller Channel (UACC, pronounced ‘you–ack’). UACC uses CC #32 as the default
controller. Like all MIDI CC numbers, it has 128 possible values, and Spitfire have suggested articulations for most of
them: CC32 #1 is ‘generic long note’, CC32 #11 is ‘tremolo/flutter’, CC32 #47 is ‘staccatissimo’, and so on. I also noticed
‘disco upwards (rips)’ and ‘disco downwards (falls)’ listed, a possible indication of a new musical direction!
The UACC standard is a work in progress, and Spitfire hope others will adopt and finalise it. If you’re tempted to use this
system for your orchestral template, you can see the whole specification at www.spitfireaudio.com/uacc-a-new-proposed-
standard.html, or download it from http://spitfire–webassets.s3.amazonaws.com/pdfs/UACCv2spec.pdf.

In this article:
Opened Up
Steinberg UR44
Installation USB Audio Interface Buy PDF
Bundled Bits Published in SOS October 2014

Reviews : Audio Interface
In Session
Verdict Could Steinberg’s mid–sized USB audio interface provide
Alternatives the perfect package for the home studio and music on the
Specifications move?
Steinberg Barry Watson
UR44 $299
T
he UR44 is the latest addition to Steinberg’s roster of audio interfaces, providing a definite step up in specification from
pros
the small and capable UR22 we looked at in June 2013 (see www.soundonsound.com/sos/jun13/articles/steinberg–
Great build quality.
ur22.htm). It features the same Yamaha D–Pre microphone preamps as the rest of the range (which are also found on
Clean mic preamps.
Easy operation and flexible Yamaha digital mixers) and a maximum sample rate of 192kHz. When compared to the UR22, the UR44 offers more inputs
DSP direct monitoring with and outputs (six ins and four outs) plus a flexible low-latency monitoring system with effects. The package includes the
dspMixFx. Yamaha DSP bundle, comprising the REV–X reverb, Sweet Spot Morphing Channel Strip compressor/EQ and the new Guitar
cons Amps Classics amp simulator, plus a basic version of Cubase. iOS compatibility is achieved through the use of an appropriate
Unused installed software Apple Connection Kit, making it an interesting option for music-making on the iPad.
steals lots of disk space!
Integrated direct monitoring Opened Up
within Cubase AI is not as
intuitive as using dspMixFx Just like the more diminutive UR22, the UR44 is packaged in an incredibly sturdy metal case, sporting the ever–popular
software. brushed aluminium and black finish. The build quality and paint job really do eclipse other interfaces at this price. The unit is
summary nicely weighted, whilst switches and pots feel positive and abundantly hardy in use. Chunky rubber feet beneath the case and
Despite minor shortcomings, handy labelling of connections on the top round things off nicely. The interface is powered by a wall–wart DC adaptor, which is
its decent audio and build included alongside a USB cable with a removable ferrite core. As with other larger USB audio interfaces, bus powering is not
quality, simple operation and an option, which might constitute a limitation for those interested in location recording.
capable low–latency
monitoring features make Starting with the front panel, four Neutrik Combo connectors
the UR44 a worthy give access to the D–Pre mic preamplifiers. On the jack front,
contender in the four–input
inputs 1 and 2 are set at instrument level for guitars and bass,
USB 2 audio interface
market.
whilst inputs 3 and 4 are line level (either unbalanced or
balanced). Above the Combo sockets is a series of LEDs: a peak
information indicator for each input as well as phantom power and mains
$299.99. power. It perhaps would have been nice to have multi–coloured
www.steinberg.net LEDs here to show input activity as well as input overload. To the
right of the four input connectors are the gain controls. Such a
layout is neat when the various cables are plugged in, and also
allows for more accurate gain matching between channels.
Phantom power switching is in pairs — a welcome feature if you
need to connect anything that doesn’t like it — and although
there is no pad switch, a high–pass filter and polarity switch are
accessible via software control. A monitor section follows, with
volume controls for two independent headphone mixes plus the master output.
Around the back, two further line inputs are provided and, usefully, these can be switched in sensitivity between +4dBu and
–10dBV. There are four line outputs here, and two additional main left and right line outputs that carry the same signals as
outputs 1–2. Such a specification is handy for users wanting to connect a second set of monitors, but as an engineer who
regularly dabbles with surround mixing, it would have been nice to have six independent outputs to make the interface even
more versatile. I suppose this decision makes sense when comparing the specification of the UR44 to the larger UR28M
(which does have six outputs), but I will continue to keep this on my wish list! Another omission that sets the UR44 apart from
the UR28M is the lack of digital I/O, which could prove problematic for those wishing to connect external equipment using
S/PDIF, for example. MIDI In and Out connections and a switch for class-compliant (iOS) operation complete the back panel
line–up.
Installation
As with other Steinberg products, getting started involves a number of steps and user registration. The interface is boxed with
a ‘tools’ disc containing the USB driver, the ‘dspMixFx’ software control application and the DSP effects. This software — plus
subsequent updates — is also available from the Steinberg web site, for those who don’t have an optical drive. The DSP
effects need to be activated before use, and this is achieved by first launching the installed ‘eLicenser’ application and
entering the serial number supplied on a licence card.
Cubase AI is downloaded directly from the Steinberg web site

using the code from a second licence card. User registration is
required, and the install is very sizeable at around 3.4GB.
Following investigation, it was revealed that such substantial size
is due to the standard installation including over 2.5GB of
content that isn’t even used by the free version of the software! I Two additional line inputs are located next to the four line
assume that this is related to the special offer upgrade to Cubase outputs on the rear panel. A further Main LR pair of line
outputs carry the same signals as outputs 1–2, so that two
Elements that is advertised when Cubase AI is first used, but it’s sets of monitors can be connected. MIDI input/output is
extremely presumptuous of Steinberg to assume that users present, as well as a class-compliant mode switch for use
would like to have the content there ready for a potential with iOS. Power is by means of a 12 volt external supply.
upgrade. Frustratingly, the content is fairly difficult to remove
manually afterwards, so users would benefit from clearer information on screen at the installation stage. To negate this issue,
‘Custom Install’ must be selected and all content options unchecked, ensuring that only data pertaining to AI only is copied to
the hard disk.
The firmware can be easily updated at any time via the Information window within the dspMixFx software . Clicking ‘Check
For Update’ here will search for firmware and software updates and then apply them. Those wishing to use the UR44 with an
iPad require firmware version 2.1 or later.
Bundled Bits
A colourful dspMixFx application is provided to set up foldback mixes and control other hardware features on the interface.
This software is to be used with audio applications other than Cubase, and is presented as a virtual mixer panel with a
reassuringly intuitive layout. On the left–hand side of the mixer panel, each input is represented as a channel with volume, pan
and mute/solo controls, plus a polarity switch and high–pass filter (inputs 1–4 only). DSP effects can be added to signals in the
foldback mix as inserts, but typically the blue send knob is used to effortlessly add a touch of reverb to incoming signals. The
master section on the right–hand side features master output faders and meters, and an effects return to specify the reverb
algorithm being used and control reverb time/return level. Two independent headphone mixes can be set up by toggling the
Mix buttons, and all settings can be simply stored for later recall. By pressing the Set–up button at the top right-hand side of
the window, users can access additional hardware settings, including sensitivity switching for line inputs 5–6 and to change
the high-pass filter frequency for the first four inputs.
An iPad app version of dspMixFx is also available and can be

downloaded once the firmware has been updated. The app
works alongside other iOS audio apps to provide the same GLOSSARY: technical terms
functionality as described above. Used creatively, the app can explained
alternatively act as a digital mixer for a live session or rehearsal,
for example with microphones running through the reverb, EQ Competitions!
and compression, and guitars passed through the amp simulator.
Cubase AI 7 features 32 audio tracks, up to 16 virtual Box (Americas Only)
instrument tracks and an updated graphical user interface. Win Adam A7X monitors
Notably, version 7 allows the use of sample rates above 96kHz, & Sub8 (UK, EU ROW
Only)
and so constitutes a more useful recording package than
previous versions (which were limited to 96kHz). A limited Halion
Sonic SE sound set is available, consisting of General MIDI
timbres plus a handful of pads and synths. The 28 stock audio The dspMixFx software (top left) working in conjunction with
Reaper. The Sweetspot Morphing Channel Strip and Guitar
effects are also basic, but are complemented nicely by using the Amp Classics plug–ins are also pictured, in this case being
VST3 versions of the bundled plug–ins on inserts and sends. used to provide low-latency effects on the foldback mix.
The REV–X plug–in is a parameter–rich digital reverb from
Yamaha, whilst Guitar Amp Classics offers a variety of usable tones and simple user interface. Likewise, the Sweet Spot
SOS Readers Ads Morphing Channel Strip offers a pleasant-sounding EQ plus more detailed control of compression parameters than the stock
GRAB A BARGAIN dynamics plug–in (for instance, higher compression ratios are available). Cubase AI integrates DSP monitoring controls for
£516,050 the UR44 into its native mixer panel, so the dspMixFx software is not required when using Cubase.
now — don't miss out! In Session
Using the UR44 with my DAW of choice (Reaper) was a delight and yielded excellent results. Simple hardware controls
combined with the intuitive layout of the dspMixFx made it incredibly simple to plug in microphones and manipulate the sound
coming from the headphones to my liking. Plenty of gain was available to the microphones (up to +52dB) and again, I found
the D–Pre preamps to be low in noise and transparent. Adding reverb on the foldback mix really allows the instruments being
recorded to work in context with the other tracks playing back, whilst panning the inputs helps to separate them in the
headphones, particularly in a self–operation situation where a number of inputs are being used by the performer/engineer. The
only down side to all this was that I was unable to access the VST3 versions of the bundled effects plug–ins for mixing, since
Reaper does not support this standard.
Meanwhile, in Cubase AI, a UR44 template provides a useful starting point for a recording project. Use of a template is key
to getting the device’s inputs and outputs active and direct monitoring switched on, but annoyingly, some fiddling with the VST
Connections panel was still required to get inputs 5–6 up and running. Seasoned Cubase users will not be fazed by this, but
the way in which inputs and outputs are accessed isn’t at all obvious for the beginner.
When using any of the Cubase variants, control of the foldback

mix and DSP effects is integrated into the software itself, so
there’s no need to use dspMixFX (which in fact will not function
whilst Cubase is running). Far from making the process more
streamlined, I found the somewhat cryptic graphics on the Input
Settings and Hardware Control panels within the Cubase mixer
to be less intuitive than the dspMixFx user interface.
Furthermore, it took me a while to get going, as the default
horizontal zoom setting on the mixer rendered many of the direct
monitoring controls invisible, a confusing situation that was only
rectified by undertaking an Internet search to find out the
keyboard shortcut for zooming out in the mixer (‘H’ on the
keyboard). However, once these hurdles were overcome,
Cubase AI did do everything I needed for recording, editing and
mixing some audio tracks, whilst also providing some passable The re–designed Cubase mixer panel displaying the UR44
string pads to layer up the mix. If you don’t own any other DAW direct monitoring controls in the top section for the first three
software, Cubase AI does represent a worthwhile option when mixer channels (which correspond to hardware inputs). The
REV–X digital reverb processor is also shown and offers a
starting out, if only to access the VST3 versions of REV–X and good range of adjustable parameters.
the other bundled plug–ins for mixing duties.
Verdict
Summing things up, the Steinberg UR44 is a high-quality and sturdy four–input interface with flexible direct monitoring
features, decent-quality bundled plug–ins, and a basic DAW to get you started. The dspMixFx software works incredibly well
when working alongside external software, but it’s frustrating that this cannot also be used when recording with Steinberg’s
own sequencers. This package will fit the bill for those who want to record more than two simultaneous inputs at high sample
rates using analogue inputs, or those wanting to switch between computer and an iPad rig with minimal headaches. The
UR44 is well worth a look! .
Alternatives
The Focusrite Scarlett 18i8 is a slightly more expensive option. It features a greater number of line inputs than the UR44,
plus SPDIF I/O and ADAT optical in. Software–controlled direct monitoring and two separate cue mixes are present, but
doesn’t include any DSP effects processing. In use, I’ve found the mic preamps to not be nearly as clean as the
Steinberg’s, and maximum sample rate is set at 96kHz.
The Akai EIE Pro is a fantastic budget option. It has broadly similar features to the UR44 with retro styling, VU meters
and bonus features including insert points and USB hub. The preamps don’t match the quality of those on the UR44, and
there is a simple direct monitoring system that doesn’t have software control or effects. Sister company M–Audio recently
launched a re–boxed version of this interface called the M–Track Quad.
Specifications
Four D–Pre mic preamplifiers with Neutrik Combo connectors.
Two instrument jack inputs on Combo connectors, two TRS line inputs on Combo connectors.
Two further TRS line inputs.
Four TRS line outputs, plus Main LR line outputs (which yield same signals as line outputs 1–2).
Two quarter–inch headphone sockets.
MIDI In and Out.
USB 2.0.
Requires Mac OS 10.7 or 10.8, or Windows 7 or 8 (32– or 64–bit).
Compatible with iOS (requires an Apple Connection Kit).
Comes with Cubase AI Elements 7 (via download), REV–X reverb, Sweet Spot Morphing Channel Strip & Guitar Amp
Classics amp simulator.
Supplied with DC power adaptor & USB cable.

Reviews Information
SOS Site Analytics
Ueberschall Brasil Nova Primeiro & Brasil Nova

Segundo
Sample Library
Reviews : Sample Shop
John Walden
T
wo reviews for the price of one here; Ueberschall’s Brasil Nova Primeiro and Brasil Nova
Segundo are two separate products but, as suggested by their titles, are closely related, covering the same music
genre and being presented in identical formats for their Elastik front-end. No prizes for guessing that we are dealing
with Brasilian-inspired Bossa Nova and Samba material here and, in each case, you get 10 complete construction kits, 2GB of
samples and around 1000 individual samples. All the construction kits feature multiple musical sections including both intros
and outros. Building complete musical arrangements is therefore very straightforward.
The key difference between the two libraries is tempo; the

Primeiro collection is focused on low- and mid-tempo (60 to 114
bpm) pieces while Segundo gives us mid- to high-tempos (107 to
145 bpm). The moods also reflect this; laid-back and mellow in
the Primeiro tunes and more high-energy in Segundo.
What’s really obvious is the very genuine Brasilian vibe. These

loops feel like authentic performances and, in particular, the
drum and percussion parts (all of which are supplied both pre-
mixed and as separate loops for each part of the kit) feel very
real. The instrumentation is also spot on with some rhythmic
piano playing, atmospheric nylon-strung acoustic guitars and
classic organ lines. In addition, most of the kits also include vocal
samples. These also sound very genuine and, throughout, you
get a very warm glow of ‘the real thing’, whether it is from the
mellow Primeiro or the more energetic Segundo.
For busy media composers looking for a genuine slice of

Bossa Nova or Samba to dip into, I’m sure both Brasil Nova
Primeiro and Brasil Nova Segundo would represent a very good
resource. In some ways, in terms of modern, mainstream,
musical production — electronic dance music, pop and rock —
both of these libraries are perhaps niche products. However,
Latin America’s musical influence (along with the use of Spanish)
has spread far and wide and, even if you just apply a few
rhythms or the occasional guitar part from these sample
collections into a more western musical context, in the right
context, it can add a real flavour of Latin America. Perhaps not
libraries for the masses, but both Brasil Nova Primeiro and Brasil
Nova Segundo catch a genuine sense of Bossa Nova or Samba
for those that need it in a ready-to-mix format. John Walden
$120 each.
www.ueberschall.com

Ukulele Free Agent: Oldies Uke, Tropical Uke, Folk

Uke & Kalakaua Ave Sessions
Sample Library
Tom Flint
Ukulele Free Agent founder Luca Tomassini is a multi-instrumentalist who has developed a passion
for the Ukulele, and now offers a selection of low-cost Ukulele construction kits and loops covering a variety of genres. The
products we are looking at here are Oldies Uke, Tropical Uke, Folk Uke and Kalakaua Ave Sessions, all of which come as 16-
bit, 44.1kHz WAV, REX 2 and Apple Loops files. Each kit has a Complete Pack option, but there is also the option of
purchasing individual folders, which is great for those who know exactly what they are after.
The most expensive kit is the Folk Uke collection, which comprises seven folders of loops and one of single chords. In each
folder the loops share the same rhythm and tempo so that they can be chained to make a song. Titles such as Lazy, Quiet and
Gentle Country leave one in little doubt as to how laid-back most of the material is, while the Plain Swing folder’s riffs are very
reminiscent of Labi Siffre’s intro to ‘It Must Be Love’.
Oldies Uke incorporates ragtime-, swing- and jazz-influenced material but focuses on quirky riffs rather than flashy finger-
picking.
Kalakaua Avenue is a main thoroughfare of the Waikiki neighbourhood of Honolulu, and Luca’s Kalakaua Ave Sessions is
an attempt to capture that Hawaiian feel. It includes loop folders titled Waikiki Blues, Relaxing On The Beach, Hawaiian Pop
and Under The Palm Trees, as well as 31 single-chord WAVs, which are very useful general-purpose samples. The kit’s loops
have been played with a great deal of precision and real gusto. In particular, the Hawaiian Pop loops feature a percussive
play-and-slap technique, which is executed extremely well, and the Under The Palm Trees’ loops display some seamless rag-
style bar finger slides.
Tropical Uke is not dissimilar to Kalakaua Ave Sessions in terms of feel. Overall, the playing is exuberant and exact, and
once again there are some handy single chords, lots of slap-and-play loops and some expertly executed slides.
From a technical point of view, it has to be said that not all of the Free Agent loops are totally clean. Some very low-level
background noise, possibly from a computer, can be heard as samples tail off, and there is the odd crackle of distortion and
rattle of Ukulele. Also, some loops end in a slight click where the waveform has not been cut at a zero-crossing point. On the
plus side, however, updates are free, so customers can re-download anything they previously bought whenever additions and
adjustments are made to the range, rather like a piece of software.
Luca’s sample-editing and recording techniques may not be perfect, but his ukulele playing is very good and always tight,
and his setup has brought out the warmth of the instrument without losing its characteristic toppy tone.
I can certainly imagine some of these samples being used in quirky advertisement tunes where something simple and up
front is needed to grab the viewers’ attention, and bearing in mind how little Luca is asking us to pay, jobbing composers
should have no reason not to add these ukulele packs to their library. Tom Flint
Prices from €2.50
www.ukulelefreeagent.com

In this article:
Culture Club
Universal Audio UAD v7.8 Plug-ins
The Mighty ’Mite Thermionic Culture Vulture, Valley People Dyna-mite, Tonelux Tilt & Neve Buy PDF
Tonelux Tilt EQ 1073

1073 And All That
Wrapping Up Reviews : VST plug-ins
Universal Audio Universal Audio’s catalogue of virtual vintage plug–ins
UAD v7.8 Plug–ins continues to grow.
pros Paul White
Authentic recreations of
W
hardware classic and hen Universal Audio launched the Apollo Twin interface a few months ago, they gave users a first opportunity to
modern. experience their Unison preamp emulation technology. This was described in detail in our June 2014 review of the
Unison technology makes Twin (www.soundonsound.com/sos/jun14/articles/apollo–twin.htm), and has since been rolled out on existing Apollo
for perhaps the most
units through a UAD software update. Unison is a combined software and hardware technology that allows various aspects of
realistic 1073 emulation yet.
the analogue preamp audio path within the Apollo unit to be configured to work alongside a plug–in, with the aim of emulating
cons the true performance of the original hardware. The interaction between the microphone and input stage can thus be modelled
None. in a way that is not possible in software alone — though UA’s preamp plug–ins are still very effective for processing tracks that
summary were already recorded through non–Unison preamps.
Universal Audio continue
their mission to bring us UA’s latest update to the UAD2/Apollo software includes four new plug–ins emulating items of high–end studio equipment:
software emulations of the Thermionic Culture Vulture, the Valley People Dyna–mite, the Tonelux Tilt and the Unison–assisted Neve 1073
classic studio hardware, to preamp/EQ. Once the software is updated you get two weeks to play with each of the new plug–ins after you first launch
great effect.
them, so you’re able to give them a thorough workout before deciding if you need them or not.
information
Culture Club
Thermionic Culture Designed by UK engineers Vic Keary and Nick Terry and reviewed back in SOS August 2003
Vulture & Neve 1073 $299 (www.soundonsound.com/sos/aug03/articles/culturevulture.htm), the Thermionic Culture Vulture must be one of the most
each; Valley People Dyna- exotic distortion devices ever created. Its high–gain valve circuitry offers a number of controllable distortion flavours that range
mite $199; Tonelux Tilt $99 from subtle warming to decidedly unsubtle grit. This unique device has now joined the phalanx of hardware modelled for the
UAD2 and Apollo platforms, and as with all UAD plug–ins, Apollo users can record through it in real time as well as apply it at
the mix stage.
A Dry/Wet mix control allows the plug–in to be used for parallel

processing, while the character and extent of the saturation are
controlled by a valve Bias control in combination with a Gain
control, Overdrive switch and three switchable distortion modes.
With Bias set anticlockwise, the distortion is fairly predictable, but
turn it clockwise and things get very ragged! Overdrive adds
extra gain, and again, is best saved for deliberate nastiness.
Low–pass filters switchable for 9 or 6 kHz are also available to
smooth the highs. The Mix control wasn’t present on the original,
but seems a sensible addition given today’s appetite for parallel processing, as is a stereo linking control to maintain a stable
image when processing stereo signals.
Of the three distortion modes, Triode is well suited to gentle warming, while Pentode One mode adds a bit of an edge and
Pentode Two mode dirties things up even more. While modelling such a non–linear device is extremely difficult, the UA team
seem to have got very close to capturing the character of the original. Used in moderation, the Culture Vulture plug–in can be
used to impart a tape–like fullness to parts such as bass, drums or vocals, while the Pentode modes can add edge to guitars
and snare drums. Vocals can be made to sound larger than life without the distortion becoming obvious, but there’s enough
gain on hand to get downright dirty if need be. I’m not normally a fan of excessive distortion unless it is tamed by a speaker
emulator, but it can be put to good use, for example, to add life to limp snare drum sounds, where several flavours of excess
are readily available. True filth is on offer in all its forms, though for me, the more subtle end of the scale works best.
The Mighty ’Mite

Designed in conjunction with Softube, the Valley People Dyna–mite emulation is a relatively straightforward to use, yet very
characterful, dynamics processor. It can be used as a gate, an expander, a limiter or a simple de–esser. Though it can be
employed as a standard peak limiter, the limiter section works best as an effect: when pushed hard it can add a real smack to
lifeless sounds, and the original is routinely used for this purpose on drums and bass.
Everyone loves acronyms, and here we have ARC or

Anticipatory Release Computation, an auto–release function that
works by looking at the input programme material and then
automatically adjusting the release time that’s been dialled in on
the Release control to minimise pumping artifacts. The side–
chain can be set to peak or RMS sensing; Peak provides a very
fast attack time, while RMS allows more of the original transient
to pass through. All the modes work as expected, but for me the
limiter is what this plug–in is really about, especially on drums,
as it really lifts the sound without beating it to death.
Tonelux Tilt EQ
The most straightforward of all the new plug–ins, the Tonelux Tilt
EQ is designed to speed up the process of getting a mix into
shape. Paul Wolff, the former owner of API and founder of
Tonelux, designed the original Tilt circuit, which has again been modelled in partnership with Softube. In its basic mode, the
Tilt control, when turned clockwise from centre, cuts lows and simultaneously boosts the highs; turning it the other way cuts
the highs and boosts the lows. A second mode, accessed via the Shape button, allows the control to boost or cut both the
highs and lows while leaving the mids alone, giving a ‘smile’ or ‘frown’ curve. Additional high– and low–pass filters allow you to
trim the extremes of the spectrum independently of the Tilt control, making this a very simple tool for quickly getting a mix
close to the tonal balance you need.
The main Tilt plug–in models all the nuances of the original’s
audio path, including its transformers, but UA also supply a
second version called Tilt Live, which leaves out the transformer
modelling; instead of a Gain control, there’s a Boost Ceiling
control, which prevents excess output level when extremes of tilt
are applied. This has a lower DSP load than the standard
version, allowing more instances to be used.
1073 And All That

Neve’s 1073 is a class–A, transformer-coupled, transistor
mic/line preamp with a classic Neve equaliser section, and UA’s
original emulation has proved enduringly popular with UAD
users. They have now released a new version that takes advantage of the Unison technology, which allows the Apollo’s input
impedance to be switched automatically to complement the relevant settings in the plug–in. Apollo users can track via the GLOSSARY: technical terms
explained
plug–in with negligible latency, as well as applying it to previously recorded material. UAD2 users forego the Unison features
and the ability to track through the 1073 at very low latency, but can use it at the mix to add ‘Neve flavour’ to existing tracks. WIN Great Prizes in SOS
According to UA, this is the only authorised and approved 1073 plug–in available. Competitions!
Box (Americas Only)
& Sub8 (UK, EU ROW
Only)
The controls are set out as on the hardware original, with the dual–stage ‘red knob’ preamp
above the three–band EQ section, which uses dual–concentric pots to adjust both gain and
SOS Readers Ads
frequency. There are 10 places in the circuitry where clipping can occur depending on how the
GRAB A BARGAIN controls are set, and all of these have been modelled in detail. That’s important as one of the main
£516,050 attributes of the 1073 is how it sounds when pushed towards the limits of its operating range, and
of Second-User Gear for sale this attention to detail should mean that the plug–in has the same sweet spots as the hardware. In
now — don't miss out! addition to its obvious use as a vocal preamp, the 1073 is popular for beefing up and adding
definition to drums, bass and other instruments. Depending on how hard you drive the plug–in, the
tonality goes from subtly warm to downright crunchy, so there’s lots of gain–related fun to be had.
Wrapping Up
Unison is clearly a big deal for the Apollo range, and UA’s commitment to it suggests that more
plug–ins, such as preamps and guitar amps, may be in the pipeline. This latest batch of plug–ins is
nothing if not eclectic, ranging from the elegant simplicity of the Tilt EQ to the extremes of the
Culture Vulture and the unrivalled heritage of the Neve 1073. They all do a great job at aping the
originals, and while detractors may claim that plug–ins will never replace hardware, these
Universal Audio emulations are so close that any perceived shortcomings are more than made up
for by the usual benefits of plug–ins — specifically the ability to use multiple instances and to recall
their settings as part of a project. .

Zero-G Extreme Environments

Kontakt Instrument
Paul Nagle
C
reated for Zero-G by producer and composer Si Begg, Extreme Environments is primarily a
Kontakt instrument for soundscapes, drones and atmospherics. It’s supplied with over 150 highly usable presets and
bundled with Reason and Logic sample instruments, plus loops in Apple and Acid formats, impulse responses, Space
Designer presets and Logic channel strip settings. The whole package is available as either a download (of approximately
1.65GB) or as a boxed DVD.
Starting with the Kontakt component, it’s split into two modes: Sampler and Time Machine. Sampler mode behaves like a
traditional sampler, while Time Machine employs Kontakt’s time-stretching, and is consequently far more CPU-greedy. Source
material is taken from 142 possible loops and there’s one small difference in layer handling. For a bit of variation, the second
layer’s loops play backwards.
The presets are divided into categories: Atmospheric Leads, Bad Places,
Beautiful Pads, Complex Bass, Complex Tuned and Hypereal (sic)
Environments. If you choose Time Machine mode, each sample can be time-
stretched between 25 percent and 400 percent, giving further scope for warped
variation. Each layer has an amplitude envelope and pan, and its sample can be
transposed by 36 semitones up or down. There’s also a (rather unsubtle)
overdrive, resonant low- and high-pass filters (complete with a filter envelope),
and a send level to the convolution reverb. Since this send is ‘pre-fade’, it’s
possible to drop the sample’s volume and hear only its effect on the reverb. With
such variety and complexity amongst the supplied convolution reverb patches,
this can be an awesome source of ambient drones with no further work required.
The base samples are taken from processed field recordings, real sounds and
old analogue synths, to name but a few. The result is an instrument that
combines playable tones with textures and atmospheres, often with an
industrial, alien or otherworldly theme. Composers scoring post-apocalyptic
games can take their pick of backdrops that include gurgling industrial pipes,
haunted music boxes, pulsing subs and seemingly deserted space stations.
When you want a break from the atmospherics, there are a selection of string,
bass and vocal pads, all of them odd and most drowning in strange reverb.
The interface is a breeze to get around. To make it better I’d have liked a
faster way of auditioning samples and reverb responses. If I were being greedy, I’d also like an assignable LFO to add even
more movement.
Leaving Kontakt aside for a moment, Logic users should find the EXS24 instruments, Space Designer presets and Apple
Loops worthwhile. They’re built from the same processed sample material but are much easier on the CPU. Similarly, the 86
impulse responses are suitable for any convolution reverb and although I didn’t get a chance to try the Reason instruments,
I’ve no reason (ho ho) to doubt their quality. However, the processor requirements might be off-putting for some. My six-year-
old Mac Pro suffered random crackles with some patches and as a workaround I often had to reduce the number of layers,
maximum polyphony or reverb time. Still, at just under £60, Extreme Environments represents great value for money and
anyone in need of a fresh batch of advanced, slightly off-kilter sound-design patches should check it out. Paul Nagle.
$79.99
www.zero-g.co.uk
In this article:
Aiming For Soul
Inside Track: Jack White
Eight Plus Eight Secrets Of The Mix Engineers: Jack White, Vance Powell & Joshua V Buy PDF
A Bunch Of Stuff Smith

Easter Everywhere
Important Pieces People + Opinion : Industry / Music Biz
Pieced Together Printer-friendly version
For Jack White, analogue recording is not about looking
Ultra Vinyl
Third Man Studios
back to the past, but choosing the ideal medium for his art.
Recording Neil Paul Tingen
“I
Young’s A Letter Home
think I was born in the wrong generation,” proclaims Jack White. “I am definitely somebody who is not supposed to be
in this time period. I probably should have been around in the 1800s, or 1930s. I am a lost soul in this time period,
with the Internet, with digital technology and so on. This is not my place to be. But I am finding my place in all of it and
am making it work for me. Third Man Records releases everything on vinyl as well as on digital. We spend 50 percent of our
energy on the Internet presenting what we put out in the digital format. I live in that world, whether I like it or not. The great
thing for me is that I get to craft things the way I want them to sound.”
Songwriter, artist, producer, guitarist and singer, White is also owner of

Third Man Records and Third Man Studios, and a board member of the
Library of Congress’s National Recording Preservation Foundation, an
organisation to which he last year donated $200,000. The titles of White’s
solo albums, Blunderbuss (2012) and Lazaretto (2014) exemplify White’s love
of bygone days — the former is an 18th–century firearm and the latter a
quarantine station for maritime travellers. But for a man who feels so out of
place in the modern era, Jack White is remarkably successful at “making it
work” for him. Lazaretto has topped the charts in many countries around the
world, selling a whopping 40,000 vinyl copies in one week in the US, and for
most of 2014 he’s been playing sell–out shows in major venues around the
planet — until recently he had two backing bands, one all–male and one all–
female. He also finds time to produce much of Third Man’s output, most
recently Neil Young’s album A Letter Home, which was recorded on a 1947
Voice–O–Graph vinyl recording booth and is surely the most lo–fi charting
release of the last 50 years.
Aiming For Soul

“Analogue is the medium of all the kinds of music that I am really fond of,”
comments White. “Form follows function. You have to ask yourself what you Photo: Mary Ellen Matthews
are trying to accomplish. What are you trying to make it sound like? When
you are recording and producing, you are aiming for something and if you
want vibe, warmth, soulfulness, things like that, you will always be drawn back to analogue. I was shocked by how many
people were upset by the Neil Young record, because it sounded scratchy and lo–fi. But they completely missed the point,
which was that we were obfuscating beauty on purpose to get to a different place, a different mood. European artists used to
talk in the cafés in France about the mediums that they were using, whether oil or watercolours or what type of canvas is the
best for what purpose. It’s normal as an artist to talk about craft and the beauty in how things are done, but a lot of musicians
don’t really seem to care about the recording medium. They don’t care about the advantages and disadvantages and different
vibes that you can achieve. They just think that if other people use Pro Tools, they should use it too, and that’s it.
“However, the actual sound of analogue is 10 times better than that of digital. I think the reason why many people say they
don’t like the way things sound on the radio or the television nowadays is because it’s all recorded digitally. Having said that,
it’s not really digital recording as such that’s the problem. A band playing live in a room recorded straight into Pro Tools doesn’t
sound bad. The problem is the multitude of plug–ins and clicks that are applied to the Pro Tools recording after that. In the
analogue world you just don’t do all that stuff. You don’t mess with the recording that much. Because it’s on tape, you tend to
leave it alone. But when it’s in Pro Tools, people keep clicking and editing and removing pops and buzzes and they place the
drums on a grid to get it in perfect timing. All those moves just suck the soul and life out of a song.
“On top, most of these digital plug–ins are emulations of the real thing. ‘Digital reverb’ is the most ironic phrase you can
come up with. It doesn’t make sense, because reverb is a natural, real thing. It needs springs, or a cave, to be there in real
life. A plug–in emulation of that may sound OK to many people, but when you pile several emulations on top of each other,
with nothing natural in it, you don’t get results that are very interesting. You can use these things to your advantage, of course.
The new Kanye West album is obviously recorded on Pro Tools but sounds unbelievable, because it is very simple and there
aren’t a lot of components going on, and this really allows the songs to shine. Plus he mixed using analogue components.
Bands like Daft Punk and Queens Of The Stone Age also know what sounds best for what they are trying to accomplish. I
really admire their production techniques. QOTSA probably record digitally, but they get a really good tone, using analogue
amplifiers and so on.”
Another essential aspect of White’s approach is the way he cherishes hard work and refuses to take the easy way. Asked
whether his previous career as an upholsterer plays a part in his adoration of the no–pain–no–gain principle, he replies,
“There’s no doubt that I can definitely relate to anyone who works in any kind of craft or trade where they get their hands dirty,
whether it is a carpenter or a plasterer. My dad always pointed out when people made something look easy, whereas in fact
what they were doing was really difficult. I put this on myself all the time: I need to make the process very difficult so that when
it sounds good at the end, I know I can be proud of it. I know it wasn’t auto–tuned, and the drums weren’t put on a grid and I
didn’t spend hours correcting ‘mistakes’. A lot of making music is dirty work, where you are getting in there, and in my case I
have to pull things from nowhere, and on top of that there’s the presentation of it, the packaging, how it is presented to people.
I have to think about those concepts as well. They become my challenges, and I create them for myself, and that is where I
start to feel inspired.”
Eight Plus Eight
However, White is not a Luddite, and he is not at all averse to using modern
technology when it suits him. He drives around in a state–of–the–art black
Tesla Model S electric sports car, which, as his engineer Joshua Smith
apparently once pointed out to him, is one of the most digital cars ever
invented. According to Smith, his boss replied that he’s never objected to using
the latest technology in principle. Moreover, White uses the car’s top–of–the–
range stereo system as one of his main monitoring references when judging
mixes, having them beamed to him via an FM radio transmitter. Similarly, while
the no–pain–no–gain principle is one of the reasons why most of Blunderbuss
and some parts of Lazaretto were recorded to eight–track analogue tape,
White’s pragmatism means that he had no issues using Pro Tools when
necessary. He simply notes: “I have never recorded to Pro Tools or mixed in
Pro Tools, but it is great for doing edits that you can’t do with analogue tape.”
All of Lazaretto was recorded to one or, usually, two of Third Man Studio’s
Studer A800 two–inch machines, which have John French’s JFR Magnetic
Science’s Ultimate Analogue eight–track headstacks with a proprietary ninth
timecode track for link–up, which allows the two eight–track machines to be
combined for 16–track recording. In addition, while Pro Tools was barely used
in the making of Blunderbuss, it saw extensive mileage during the making of
Lazaretto. Jack White’s Third Man Records organisation
encompasses a variety of business activities,
White went on record a few years ago stating that he couldn’t see himself including his own recording studio.
ever making a solo album, and both Blunderbuss and Lazaretto appear to have
come into being more or less by accident. Blunderbuss started as a by–product of sessions White had organised for an artist
who didn’t show, so he decided to record some of his own songs with the musicians who turned up at his studio. Lazaretto
came into being in a similarly sideways fashion
“When I did Blunderbuss we did a lot of songs that I never finished, and there were also some ideas that I put down that
maybe could have been used for the soundtrack for The Lone Ranger [a 2013 Disney action movie featuring Johnny Depp]. I
never wrote anything specific for that project anyway, but it did lead me to go in and record some ideas for some instrumentals
that I had. I also wanted to record both my female and male bands while we were on tour and really cooking. If we’d waited
until after the tour and with everyone having been home for six weeks or so there would have been a different vibe. So I wrote
many things specifically for those recording sessions during the Blunderbuss tour, to give us something to play. These
recordings usually were unfinished, so it was a matter of ‘OK, I’ll get back to this way later, maybe in six months or so when
we’ll be finished touring.’
“As a result I had all these songs sitting around that were half–finished, a quarter finished, three–quarters finished, often
without vocals. There were about 25 of them, which was a place I’d never been in before in my career. I’d never had that
luxury, or that problem, and I considered it a problem, because I’d never written vocals for music that had been sitting around
for that long. I’ve never had a problem writing music, I’ve never had writer’s block, but I did have a problem writing vocals for
my own music. That was a strange thing for me to do. I had never separated the two processes, of writing music and writing
vocals, over such a big distance, and I had to come up with some tricks to overcome that.”
One “trick” that White employed was reworking some short stories and plays that he’d written when he was 19. It was the
need to overdub his vocals, and often his guitar parts as well, that led to changes in song structures and arrangements which
prompted many of the Pro Tools edits. White explains: “A song like ‘Would You Fight For My Love?’, for example, had three
major edits, from takes of two different bands playing at three different times. We edited the tape to punch the male band over
the female band in the chorus, with all eight tracks, and this was a very dangerous move, but it turned out amazing. But many
other edits were so complicated that we had to put the material into Pro Tools, do the edits, and then we transported it back to
tape again after that.”
A Bunch Of Stuff
One person who has played a central role in helping White put Third Man Studios together and who has manned the controls
for many years is Vance Powell. The engineer, mixer and producer worked as chief engineer at Blackbird Studios in Nashville
from 2001–10, and in 2006 also set up his own facility, Sputnik Sound, together with Mitch Dane. Powell met White for the first
time in 2006, when he worked on Dangermouse and Daniel Luppi’s Rome album, on which White guested. The year following
White asked Powell to mix the Spanish–language version of the White Stripes song ‘Conquest’ and a regular collaboration
was born, with Powell since 2009 dividing his time between Third
Man Studios and Sputnik. Recently, however, Powell has
become increasingly busy at Sputnik: “I think I did 37 albums last
year!” Because White tends to only give a few days’, or even a
few hours’ notice when he wants to go into the studio, Powell’s
former assistant, Joshua V Smith, has increasingly taken over at
Third Man, and is credited with recording most of and mixing all
of Lazaretto.
However, Powell conducted the first sessions for Lazaretto, in

2012, with Smith still assisting. “We recorded a bunch of stuff for
Jack that we weren’t sure what it was for,” Powell recalls. “The
last sessions I recorded for Lazaretto were in December 2012:
three days with the guy band and three days with the girl band.
The basic tracks were cut live in the studio, and then there were Vance Powell at Sputnik Studios.
overdubs, like the harmonica solo in ‘Three Women’, and many
of Jack’s vocals, which were recorded by Josh. We recorded to tape, initially just eight–track, to RMG 900 at 7.5ips. Jack
loves that sound and we also are very happy with the quality. It has a hefty bass bump, almost an octave lower than 15ips,
and the top end is flat until 20k. Something happens at the top end, above 4k, with tape compression; the sound is easier
saturated and it meant that we never had to worry about things like de–essers. We did not hit the tape hard. In fact, we left
tons of headroom, because it retains transients better.
“The signal chains remained pretty much the same throughout the recordings. There isn’t a big console or a lot of gear at
Third Man, though Jack acquired this really cool, all–tube UA 1008 desk, which dates from a later date than the 610. It was
made for film and has nine mic inputs and three film inputs. The desk lives out on the floor close to the keyboards. I’d have a
Sennheiser MD421 or Shure SM57 or maybe a ribbon like an AEA R92 on the organ, a Neumann stereo SM2 on the piano,
and they went into the 1008, which came up on the main Neve desk as a pair of channels or just one channel. I probably had
a Fairchild on the keyboards, and maybe a Neve 2254 compressor on the desk. The studio’s 1960s Ludwig drum kit had an
AKG D12 on the kick, a Shure SM57 on the snare top and bottom, Sennheiser MD421 on the toms, and I think I had a
Neumann U67 for mono overheads. The drum mics all came in on the Neve desk, and I probably had an 1176 on the snare
and maybe also the kick, or a Fairchild 670. We normally record the drums to one track, plus we’ll have a kick drum track to
have more control over that. I don’t need stereo drums. For me, drums are right in front of me.
“On Blunderbuss I also used the Neve 33609 and RCA BA6A and an Ampex MX35 four–channel tube mixer to record the
drums, but these sessions happened so quickly that I did not have a lot of time to set things up. There was not a lot of upright
bass this time, but when there was one, I’d use an RCA 44 and something higher up like the RCA BK5A [cardioid ribbon mic].
There was an African drum on ‘Would You Fight For My Love?’, which had an AEA R92, electric bass would have been DI and
a Neumann U67 on the amp, with maybe some compression from the [Fairchild] 670. I recorded Jack’s acoustic guitar with an
RCA 77DX, and his electric almost always goes through his 1963 Fender Vibroverb in front of which I placed a U67, which
went into the Neve 1073 desk and then straight to tape. I did not record any of Jack’s vocals, other than on the song ‘Just One
Drink’ because that was done entirely live. I used a Shure SM57 or 58 on his vocals for that, and Josh recorded the backing
vocals.”
Easter Everywhere
Joshua V Smith, the man who gradually took over from Powell at Third Man Studio as the latter became increasingly busy
with other projects, is originally from Kernersville, a small town in North Carolina. During his high–school years he played in
bands and developed an interest in recording, and discovered to his complete surprise that there was a unique, top studio in
his home town: Fidelitorium Recordings, owned and run by Mitch Easter, who worked on REM’s early albums. Fidelitorium is
chock–a–block with vintage and analogue gear, and observing and assisting Easter for a number of years, on and off,
awakened a love of analogue gear and working methods in Smith. He moved to Nashville in 2006, where he became an
assistant at Sputnik, first under Mitch Dane and then Vance Powell. He eventually ended up moving to Third Man, where he
now works full–time. When White is on tour, Smith also acts as one of the singer’s stage techs.
Smith echoes Powell in explaining that the recording sessions

for Lazaretto happened very quickly and often unexpectedly, with
barely any time to set up. “There have been many situations
where I’ve sat in Third Man and had no idea where we were
going. But Jack always seems to know, so a lot of the time I
simply hold on and enjoy the ride! I was second engineer on
Blunderbuss, which was done fairly quickly, with most of the
songs recorded in one piece. This new album felt much more
pieced–together from different sessions. Jack will sometimes
bring in a demo recording and play it for the musicians, but more
often he simply starts playing on guitar or piano and the
musicians just jump in. There are people who like to rehearse a
song and really develop it, which technically could make for a
better performance, but Jack doesn’t do it like that. In my Joshua V Smith (left, with Jack White) is not only resident
engineer at Third Man, but also White’s stage tech.
observation he likes to capture the energy of people figuring it Photo: David James Swanson
out, even if it’s not going to be perfect. In that sense he almost
has a jazz mentality, in that he wants spontaneity and a kind of
innocence in the way people are playing, and in the people engineering.
“I can get very detailed about things and get carried away by that, but Jack is very quick and will not allow us to get stuck in
that kind of detail. If something isn’t working, sure, we try to troubleshoot it, but most of the time you have to, as Vance calls it,
‘spin to win’. You just have to push up the faders and hope that it’ll work; rarely do you get more than a few seconds to check
the tones and the levels of the instruments. There have been many times when Vance and I were recording at Third Man, and
we’d think it’s just a preliminary pass of a song, with the band just learning it, and we’re sure there’ll be another take, allowing
us to sort out some technical issues. And then suddenly Jack will say, ‘OK, that’s cool, let’s listen to that.’ And I’m going,
‘Aargh, what am I going to do about that floor tom that I flipped the phase on mid–take?’ It definitely made for tricky mixing
sometimes, when I had to figure out ways around these issues. They weren’t necessarily fixes, but more ways of covering
something up so it doesn’t sound like an issue any more.”
Important Pieces
Predictably, given the Spartan approach to gear at Third Man and the fact that Smith learned his trade under Powell, the
signal chains Smith used were very similar to Powell’s. There were a number of differences, though, as Smith explains. “Most
of the band tracking sessions were recorded to eight–track tape, and we tried to keep it on eight–track, but with almost all
songs there were overdubs and the amount of tracks got out of hand. Because we didn’t always want to bounce things, we
went to 16–track by sync’ing the second eight–track. The signal chains remained largely the same for all sessions, and certain
pieces of gear were important throughout. In addition to the D12 I’d also have a Neumann FET47 on the kick sometimes,
particularly when I had the D12 on the floor tom. It’s also not unusual for us to have the Coles 4038 or Sennheiser 421 on the
floor tom. I love the AEA R88 [stereo ribbon] mic, which we usually have above the kit, since it really helps capture the kit as a
whole, but for these sessions it was usually down at Third Man Records, for recording the high–school bands that come in. I
mainly used the U67 as a mono overhead for this record. The kit was always tracked in mono, but sometimes also with a
separate Fulltone Tube Tape Echo drum track. At times we used the Fulltone to create polyrhythms that inspired other parts of
the songs.
“I recorded Jack’s vocals mainly with a Shure SM57. Sometimes we used a Neumann U47, as well as an RCA 77D and a
Shure SM7, and I often pushed his vocals hard through an 1176, which can make it difficult to keep the sibilance under
control. But he likes that slammed vocal sound, and we used it quite a lot on this record. Jack also loves to amp his vocals.
We’ll split the mic signal, and have one line going clean to the console, and the other going to an old tube amp, then back to
the console. Sometimes we used amps with reverb, sometimes we didn’t, but overall we liked that biting mid–range sound
from the amp on his vocals. Very rarely I recorded the clean and amped vocals on separate tracks and would then balance
them later in the mix, but usually we just summed them to one track.
“I tend to like an RCA BK5B on Jack’s acoustic guitar, and occasionally an RCA 77, or an SM57 if everything else in the
room was pretty loud. He mostly played an old Gibson Army Navy or his custom Gretsch Round–Up ‘Claudette’. The Army
Navy doesn’t sound like a modern acoustic. It has a very punchy and boxy sound, without much resonance, so the things I’d
normally use on an acoustic don’t always work. The mic would go into the 1073, usually with a bit of 1176 compression. I
miked Jack’s electric guitar amp — often his 1964 Vibroverb — with a U67 and also a Shure SM57, and we sum them on the
desk to one track. Sometimes I’d only have the 57 on his amp.”
Pieced Together
As White explained above, Powell and Smith edited the recordings on both tape and in Pro Tools to get the desired final song
arrangements and structures. Smith highlights two songs on the album as being indicative of two different approaches: the
title track and the epic ‘Would You Fight For My Love?’ He recalls: “We have a small native Pro Tools system at Third Man,
with 16 inputs. We initially bought it for tape backups, just to have digital backups, but when we had some very complicated
edits to do, it became more of a creative tool. But it always ends up going back to tape. ‘Lazaretto’ and also ‘Three Women’
definitely were band tracks done in one take, and they also were done on the same day, so they have very similar sounds.
They were fairly straightforward to record and mix, and had no major edits. On the other hand, ‘Would You Fight For My
Love?’ was one of the most difficult tracks to do, because it consisted of three different sections edited together. The intro was
the guy band, the quieter section with the toms is the girls, and then it’s back to the guy band where the hi–hat comes in.
These edits were done by Vance on tape, and I did several additional edits in Pro Tools later on. That song was one of those
that were pieced together from bits that were not originally intended to be together, or so it seemed to me. We considered re–
recording the song, but Jack liked the tones and the vibe and did not want to redo it.”
Smith and White explain how they went about mixing the album together, with a particular focus on their mix of ‘Would You
Fight For My Love?’
“I have always mixed all my albums,” says White, “but I don’t like to turn the knobs. When several people turn knobs at the
same time, it’s not a good thing. So I basically direct the entire mix, everything about it: what compression to put on the kick
drum, what kind of reverb is going on the snare, the effects on the vocals, and so on. Every single mix choice is mine, but
again, I don’t touch the knobs. That feels more comfortable to me. At earlier occasions I sat down with the person and we
mixed together and that just doesn’t work. It’s like two people playing the guitar at the same time. But it does work when I
direct all components of the mix. I have always worked like that, with every engineer.”
“Yeah, Jack is very much part of the mix process,” agrees Smith. “I generally will set something up that I feel is a good start,
maybe in the evening, and he then will come in the next morning and do automation passes. He can be very hands–on.
Before we had Flying Faders on the desk, everybody had to push faders around, but that’s no longer necessary. I assumed
Vance was going to mix the album, but I think it didn’t fit with his schedule, and one day Jack came in and said, ‘OK, let’s mix
this song.’ We then mixed the entire album in one stretch, doing pretty much one song a day, or one every two days, and then
we went back and did several recalls. I have worked with Vance and Jack long enough to generally know what they like and
don’t like, but Jack would sometimes throw me curveballs or wanted to go in a totally different direction with a song than I had
imagined.
“A few of the songs were quite tough to mix, like ‘That Black Bat Liquorice’, ‘Just One Drink’ and ‘Would You Fight For My
Love?’ Songs that were kind of pieced together from different takes might have up to five instruments on one tape track, which
is where automation helped a lot. ‘Just One Drink’ is kind of a straightforward rock song. We did an early rough mix, that I ran
through a Waves MaxxBCL, which we use to ‘heat’ our mixes, and Jack fell in love with it. He didn’t really want it that heavy,
he wanted to mix and master with as little compression as possible, but with that song it was really hard to beat the rough mix
he was so used to. With ‘Black Bat Liquorice’ I had messed around in the computer with bit–crusher and stereo widener plug–
ins on the drums in certain breaks in the song, where the drums sound wide and distorted. It’s the only time I did an effect in
Pro Tools for the entire album. I did it for fun, but Jack liked it and asked me whether we could do it in analogue. I was like,
‘Not really,’ so I printed the effect to tape and we kept it.
“The mix of ‘Would You Fight For My Love?’ was challenging because the two bands that were stuck together meant that I
had to find a place in the middle that worked for the sounds of both bands. Luckily the recordings were similar enough, so I
didn’t have to EQ them completely separately; I just had to find treatments that worked for everything. The dynamics of the
song also were quite drastic. Trying to get all that to sit well was tricky. I had a Neve 33609 and also the GML 8200 EQ on the
mono drum track. I will generally do my initial EQ on the Neve desk, but the 8200 is unbelievable for notching out specific
frequencies — I also found the Inward Connections Brat EQ to be very handy in this department as well. I also like using the
Drawmer DS201 dual gate, to bring out a little bit more of the snare or the kick in the mix. We also had the second drums
echo track with the Fulltone in this song.
“I didn’t compress the bass any further, and maybe just used some Neve 1073 desk EQ. I used the Dbx 500 [Subharmonic
Synthesizer] in this song, as well as several others. Jack loves it, but without having a sub in the studio, it can be an easy way
to blow out your woofers! I didn’t have compression on the guitars either, again just some desk EQ. It was the same with the
keys. For Jack’s vocals I again used the 1176 to give it a more upfront and smashed sound. He did a vocal double that I ran
through a Neve 2254 compressor, plus there was an amped vocal track. You can hear this track after the bass break. I also
used some Master Room reverb on the drums, the vocals, and the fiddle, as well as some Moog 500 Series delay in places.
The only real delays we used on the album came from the Moog 500 Series, the Roland RE301 and the Fulltone Tape Echo.
Jack doesn’t like a lot of reverb, so I had to use it sparingly, just as a little bit of glue: something that you pick up
subconsciously, but don’t really hear most of the time.
“Finally, I ran the band, without the drums, through a separate desk bus for parallel compression from the API 2500. I did
this to bring out a bit more of the stereo image, because the drums are in mono and in the centre, and I wanted the band to
sound more cohesive. We normally have the API 2500 on our mix bus, but Jack wanted something different in this case,
which freed up the 2500. We instead used a couple of Neve 2254s on the mix bus for this song. “
Ultra Vinyl
White and Smith mixed to a Mike Spitz custom–built Ampex ATR102 one–inch tape recorder, at 15ips. White likes to use his
Tesla electric car as a mix reference, which, oddly, created some effects that won’t work as well in countries with right–hand–
drive cars. “The Tesla sound system is Jack’s mix reference now, it’s what he’s used to listening to,” explains Smith. “We
broadcast mixes to his car via an FM transmitter, and if we wanted higher fidelity we’d use a USB stick with 96k WAV files. We
have a set of walkie–talkies and if he heard something in his car that he wanted turned up or down, he’d communicate with
me via that. One of the interesting things is that Jack likes to mix from a driver perspective: he usually likes drums or vocal
delays on the right because it creates a cool effect when he’s listening to it in the car.”
Finally, and entirely unsurprisingly, White takes a dim view of the loudness wars. When mastering engineer Bob Ludwig, in
the process of mastering Blunderbuss, asked two years ago “Why don’t we just turn the gain up and not put any compression
on it?” White’s response was, “I have been asking that fucking question for 10 years and nobody ever said that we could do
that! So yes please!” Two years later White has taken a more nuanced approach. He’s pulled out all the stops for the Ultra–LP
vinyl version, which includes two hidden tracks behind the centre labels, one side that plays from the inside out, dual–groove
technology that creates two different intros for ‘Just One Drink’ depending on where the needle is dropped, and hand–etched
holograms of angels that only show when playing the album with a light source directly above it, and more.
“I was planning to do exactly the same as with Blunderbuss,” explained White, “and I talked with Bob about this. But then
the Ultra–LP project took shape and I wanted to make that its own beast, different from the digital version. So I decided on no
compression on the vinyl version. Bob just turned up the gain on the master for the vinyl edition. We did have some mastering
compression on the digital version, but without limiting or clipping.”
For someone who does not feel like a man of his time, Jack White III has a remarkable grasp of the things that work, and
don’t work, in this day and age. .
Third Man Studios

For many years, Jack White resisted the idea of setting up his own studio, preferring the time limitations associated with
traditional studio work. Many of the White Stripes’ studio albums were recorded very quickly: in the case of White Blood
Cells (2001), in as little as four days. But in 2009, after establishing a physical location for Third Man Records in Nashville,
White decided to also build a studio there. The physically small facility contains a 16–channel, 1073–based Neve console
originally from SABC in Johannesburg which also has four Neve 2054 compressor modules, with labelling still in
Afrikaans, plus the two above–mentioned Studer A800 tape recorders, an Ampex ATR102 one–inch recorder, and a
choice selection of mics and outboard. The latter includes Urei 1176, Teletronix LA2A, API 2500, Fairchild and Neve
33609 compressors, GML 8200 and API 550 EQs, and reverbs, delays and echos from Fulltone, Master Room, Moog,
Roland, Furman and EMT.
“10 years ago I did not care about equipment names and
numbers,” White elaborates. “I just liked the concepts. I
liked analogue compressors, ribbon mics, and so on, and I
did not care whether the ribbon mic was an RCA 77 or a
44. The first album by the Raconteurs, Broken Boy Soldiers
(2006) and the White Stripes’ Get Behind Me Satan album
(2005) were both recorded using nothing but a Coles [4038]
ribbon microphone. That was my concept: ‘I like this mic,
let’s use it for everything.’ The engineers said, ‘this is really
hard to do, I have to EQ the hell out of everything to get a
certain sound, why not use a 57 or something?’ but I said, ‘I
don’t care because it gets us to a place where we wouldn’t
have gotten if we’d used different microphones.’ We would
have gotten a totally different vibe and mood if we’d done
that. But when I decided to have my own studio I became
incredibly involved in choosing the equipment, and I’ve Jack White’s love of primary colours even extends to his
slowly been collecting things that I’ve used over the years studio track sheets (this one is taken from ‘Hypocritical Kiss’
and that I thought sounded great. The studio also is very from Blunderbuss).
small, so that the musicians are pushed close together.
Sometimes in a big studio, the drummer may be 40 feet away from the bass player, each with baffles around them, and
the result is that they’re not really playing together. I wanted everyone in everyone else’s face, working closely together.”
Recording Neil Young’s A Letter Home

“Some people get caught up in emulating albums or artists
from the past, and trying to make what they are doing sound
exactly the same. I have never done that. I have never tried
to make something sound like a record from 1962,” says Jack
White, and his music certainly is forward– as much as
backward–looking. There is, arguably, one exception in his
oeuvre, which is Neil Young’s recent album A Letter Home.
Recorded at Third Man, A Letter Home is marked down as
Young’s 34th ‘studio album’, but it barely deserves to be
called this, because it was not recorded in a studio but in a
Voice–O–Graph.
Also called the ‘Third Man Recording Booth’, the Voice–O–
Graph looks very much like a telephone booth from the
outside, and it was widely in use in the US from the 1940s to
the 1960s. The coin–operated machine allowed people to
record up to two minutes of extremely lo–fi audio, full of
crackles, scratches and wow–and–flutter, onto a six–inch
phonograph disc. White says that he’d been “looking for one
for 15 years” and he finally managed to acquire a Voice–O–
Graph a couple of years ago. “We spent a year and half
restoring it and getting it in tiptop shape. It sounds really
incredible now. People used it to send audio letters to each other. People in
the army, in wars, sending messages home, it was pretty amazing.”
The Third Man Recording Booth was officially opened on April 20th, 2013,
and the Third Man web site states that it is “the only machine of its kind in the
world that is both operational and open to the public”. Neil Young recorded 11
cover songs in it, and one spoken message to his deceased mother. Because
the message and the music sound like they’re coming straight from the 1940s,
they are strangely affecting and emotional — which, explains White, was
exactly the point. “If, say, his message to his mother had been recorded on
Pro Tools, it would have been like, ‘So what?’ There would have been no
beauty in it. Recording Neil in the booth gave us a certain vibe. It’s what Neil
would have sounded like in the 1940s, recording himself.”
The recordings, lo–fi and scratchy as they are, posed significant challenges
for the engineers, as Joshua V Smith explains. “Kevin Carrico, a wizard from
Detroit, and I worked on getting Jack’s Voice–O–Graph to function properly
between February to August 2013. It’s still a work in progress, but as long as
it’s working, we’re scared to touch it! Kevin and I thought of ourselves as
‘electro–mechanically’ engineering the Neil Young album. We had less than a
week to modify the booth to Jack’s instructions for Neil’s album project. Some Jack White’s Voice–O–Graph was
of the modifications included converting it from 45rpm to 33rpm to get more painstakingly restored by Joshua Smith
recording time, adding a Nixie tube countdown timer, installing fans in the and Kevin Carrico.
Photo: Jo McCaughey
ceiling, installing a camera and monitor inside of the booth so Neil could watch
the disc–cutting process and knew when to start and stop, and installing
controls on the back of the booth, so Kevin and I could remotely start and stop it without getting in the way.
“All the records we cut were made of polycarbonate. The original records were made of plastic or a cardboard
composite sprayed with acetate or covered in some kind of

plastic, and they had grooves cut into them with a steel
needle. The grooves in our discs were embossed in them
with a tungsten carbide needle. We tried cutting into the
polycarbonate but all we got was noise, so embossing was
the way to go. We also originally used steel needles and
actual acetate master discs cut down to six–inch, but they
were $10 per disc! The polycarbonate discs sounded better
for our applications and were way cheaper.
“After the discs were cut, we transferred them to one–
inch two–track with a 1953 Scully lathe at Third Man that
was previously used by Cincinnati’s legendary King
Records. We later found that the lathe had induced some
noise into the transfer, but by the time we realised this,
there was no time to do something about it. Because each Neil Young leaves the Voice–O–Graph after a take.
disc could only hold 2:27 of audio, I had to do some splices Photo: Jamie Goodsell
on tape. The tricky part was having to varispeed the two
takes before editing them so they were roughly in the same
pitch. I did this with discs on which the pitch jumped
dramatically in the middle of the take. I ended up having
two varispeed boxes, and Richard Ealey at Blackbird
Studios made me a switching unit to go between the two
boxes. For something that was done with one mic in a
phone booth, this project ended up being pretty
complicated!”

Reviews Information
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Have Your Say! Sounding Off

If you would like to air your
views in this column, please Published in SOS October 2014
People + Opinion : Sounding Off
submit your ideas (900 Printer-friendly version
words max) by email or by Why music is better now than it ever has been.
post to the address at the
foot of this screen. We pay
Alex Marten
R
for contributions we print in
the magazine.
eceived wisdom states that music gets worse over time, that ever since Beethoven / Robert Johnson / Elvis / Kraftwerk
/ Miley Cyrus (depending on your point of view), music has entered into an inexorable decline, destined to reach its
SOS Forum nadir right about now in an unlistenable mush, at which point we’ll all just have to move onto something else not
Got a burning question to involving sound at all, like avant garde smell–art or something (sound is so last millennium — I’ve already trademarked the
ask or an opinion to share? phrase Smell On Smell for a magazine).
Join the community in the
ever-popular SOS Forum Speak to pretty much any muso and they will tell you the same: it’s just not what it used to be. Guitarists can’t play guitar as
well as the guys from the ‘60s and ‘70s, composers can’t compose as well as those 18th-century dudes, and producers are all
just grabbing at the coat–tails of the geniuses of yore like badly trained puppies.
I put it to you, though, that this is not the case. On the contrary, music now is, on average, better than it ever has been.
Judged on results alone, composers are better at composing, producers are better at producing, and guitarists are better at
guitaring.
Before you have me strung up with ukulele strings and forced

to listen to back–to–back badly recorded One Direction YouTube
clips, consider this: fifty years ago, around the time the Beatles
were first moistening undergarments, the barriers between
musical inspiration and the end product were huge.
First, you needed to buy an instrument; not easy considering

the huge relative cost and minuscule relative quality compared to
what is available now. In those days, a usable guitar would cost
several months’ salary; nowadays you can get an eminently
playable instrument for under £100.
Second, you had to learn to play the instrument. Good luck

finding a friendly local ‘rock school’ or masses of free online
tuition courses delivered by world–class musicians. You’d be
more likely to have a wooden plank shoved down your back, and
be forced to sit in a cold, damp room inscribing polyphonic
madrigals.
Third, you had to get yourself recorded. This could be the

subject of a whole other article, but suffice to say that the
technological firepower available in your phone is not that far off
what was only then possible in the largest, most expensive recording studios.
Fourth, you had to get that recording released. Even once you had negotiated Hunter S Thompson’s “cruel, shallow money
trench” to convince a nepotistic A&R bod to give you a chance, you probably still would have gotten fewer listens than a well–
promoted YouTube video nowadays. And that would only be in exchange for a contract that specified eternal ownership of
your soul and the souls of all your loved ones.
The technology we have now sweeps obstacles aside. Want a complex glitchy edit of your drum track? Whack a plug-in on
it and turn up the ‘complex glitch’ setting. Need a crunchy British valve tone on your amp? Turn the knob to ‘crunchy British
valve tone’. Can’t be bothered to learn how to play the keyboard? Choose your scale, turn swing quantise on, and headbutt
the keyboard while spasmodically flailing your arms about.
Every month, new technology is released that makes the music–making process quicker and easier. Ableton Live’s
incredibly intuitive UI not easy enough for you? Here’s a controller that makes it even easier. Can’t understand the principles
of gain structure? Let your interface do the job for you.
My point is that it’s now easier than ever to make and release well–mixed, well mastered, musically competent and
professional-sounding material. Every stage in the musical creation process has either been simplified, automated, or
completely removed.
The boundaries between inspiration and end product have fallen; 99 percent perspiration and 1 percent inspiration has
become 1 percent perspiration and 99 percent inspiration. We’re standing on the shoulders of giants, and we’ve got better
apps than them. Music has been democratised, so surely the end result must be better now than it ever has been. And if not,
why not? .
About The Author

Alex Marten runs London and Edinburgh–based musical equipment retailer Red Dog Music. He enjoys the unique torture
of having unlimited access to two shops worth of cutting-edge musical equipment, but never having the time to actually
use any of it.

Why Pro Studios Matter

Leader Buy PDF
Paul White Published in SOS October 2014
I
t is an unfortunate fact of life that the democratisation of music production, much of which is Printer-friendly version
down to the affordability of computers and music software, has had a serious impact on the
professional studio market, forcing down rates and causing many studios to close. These days it
costs more to hire a plumber than a decent studio, even though the studio represents a vastly greater capital investment in
equipment and premises. It seems ironic then that in recent years countless students have been taking music technology
courses in the hope of embarking on careers as studio engineers, but that’s a story for another day.
The thought process of the recording musician often seems to go along the lines of ‘If the professionals use Pro Tools or
whatever mainstream DAW, then if I use exactly the same thing at home I can get the same results, so I don’t need a
professional studio.’ Oddly enough, nobody seems to apply the same rationale to brain surgery, where it is widely appreciated
that a suitably clean operating room and some degree of skill is required in order to wield the necessary tools and achieve a
good outcome.
While it is true that the computer side of recording can now be considered almost as a consumer-priced commodity, a good
professional studio has two major advantages, and neither are really about gear — except possibly the monitoring. Firstly,
there will be acoustically sympathetic spaces both for recording and monitoring, and secondly, there will be engineering
expertise borne of experience. In my view these attributes far outweigh the importance of the actual recording system. A
capable engineer could take a Mac Mini loaded with Garage Band into such a studio and, armed only with a careful selection
of affordable microphones, produce a thoroughly professional recording.
What can we learn from this? Certainly it suggests that taking care of the acoustics in a home studio should be given a
much higher priority than many people seem to give it. It also tells us that we have to really hone our recording and mixing
skills if we want to get close to the results that a professional can achieve. Nobody said it was going to be easy.
Even then there are still occasions when using a serious professional studio can produce a significantly better end result,
even if you only use it, say, for tracking the drums or for mixing. Nobody says you have to do the whole project there, though if
you are both the artist and the engineer when working at home, working with another engineer will take the pressure off and
allow you to concentrate on your performance rather than having to swap ‘brain sides’ to deal with technical issues. And as a
bonus you might pick up a few useful tips that you can apply in your own studio, and there’s always the benefit that comes
from having a fresh pair of ears on the job. So, maybe next time you have a burst pipe, fix it yourself and use the money
you’ve saved to treat yourself to a few hours in a professional studio.
Paul White Editor In Chief .

In this article:
Why Mixing Bass Is
Mix Rescue
Buy PDF
Tricky Published in SOS October 2014
Technique : Mixing
Prep! Printer-friendly version
Splitting Up The Bass Patrizio Cavaliere: Our engineer sorts out the bottom end
A Test Of Character of a deep–house track.
Kick Drum Neil Rogers
Multiband Options
D
Bass–ic Sanity Checks J Patrizio Cavaliere is an accomplished producer of his own material, and he usually takes care of his own mixing in
Room For the comfort of his home studio. As with many home studio users, though, he often finds it tricky to judge the lower
Improvement? regions of the frequency spectrum, and thus to get the ‘bottom end’ of his mixes right. With that in mind, he got in
Louder Or Better? touch to ask if I could lend some fresh ears and perspective to his deep-house track ‘Dionysus’.
Audio Examples
Why Mixing Bass Is Tricky
The low end can be a tricky part of the mix to get right, both technically and subjectively, but home studios often make this
challenge even tougher. The limitations of budget monitor speakers or headphones don’t help, of course, but the biggest issue
is usually the room acoustics: low frequencies are often difficult to bring under control using acoustic treatment in typical small
domestic rooms, and without an even low–end frequency response, free from unwanted resonances and nulls, it’s pretty much
impossible to judge things reliably — you don’t even know whether you have any problematic issues, never mind what action
you should take to address them!
Decisions about the bass end of a mix are complicated further by the variety of
playback systems used by listeners. Some will be hearing your track via a laptop
or mobile phone, with their bass-light responses, while others may listen to
systems with over–hyped bass, whether in car stereo systems or popular
headphone designs. It’s not surprising that the novice engineer finds it difficult to
strike the right balance. In the case of a producer/DJ like Patrizio, a major concern
is also how the mix will translate to a club environment, with a PA often containing
large subs, which might have been brought along for a specific event and not have
been tuned to the room.
This month, then, I’ll focus on how I worked on the bass elements of Patrizio’s
track, and how sorting out the low–end instruments left me free to help shape
other aspects of the mix as well. I’ll also give a few suggestions about how you
might improve your listening environment and how your bass will fare when it’s
played back out there in the real world.
Prep!
Before starting on the bass elements themselves, I felt it important that I got whole
the mix organised and roughly in shape. There were about 30 parts to play with,
including individual drum hits, shakers, bongo samples, bass, pads, stabs and
Patrizio Cavaliere, a highly regarded,
other assorted synth parts, which came in and out at different parts in the track. well–travelled DJ, has been producing
Patrizio had also provided me with his mix and a reference track, which was both underground house and electronica for a
relevant to the style and a good example of a mix he felt translated well when number of years. He’s inspired by a blend
of eclectic rhythms, rare disco, Balearic
DJ’ing. beats, house and techno. He is the
founder of Also Ran Music and resident
I performed my usual grouping and organising of parts, and got a basic rough DJ at Cosmic Soup and this production
balance of the track using nothing but the faders and a little panning. I also used ‘Dionysus’ will be his third release on
German label Movida Records.
the time spent doing that to develop a broad idea of what the mix would require of
me. My first step in terms of processing was to apply some high–pass filtering on elements that didn’t seem to require any
frequencies around 50–100Hz. The effect of such filtering can be quite subtle — clearing out this unnecessary clutter doesn’t
seem to change the sound of the individual parts you’re filtering, but it does have a cumulative effect, eventually creating
much more space for the lower frequency elements in a mix.
Splitting Up The Bass

The bass elements themselves were quite straightforward, with
just one kick drum and one synth bass part that Patrizio had
created with his Mini Moog. Each needed its own place in the
mix, and obviously had to be full and generous in the bass
department, but I also needed to ensure they’d make a
worthwhile contribution when the track was played on smaller
speakers.
I decided to use an approach called ‘multing’, which simply

means creating a duplicate copy of a track, in order that the two
copies can be processed differently. Sometimes you’d do this to
apply different processes in different sections of the song, but
here my aim was to create a sort of crossover filter, so that I
could process a ‘low’ bass track and a ‘high’ bass track
independently.
To do this you just need to add a high–pass filter to one part

and a low–pass filter to the other. After a little experimentation I High–pass filtering out any unnecessary low information on
various mix elements helped make space that the bass
settled on a low–pass filter set to 335Hz, but with a very shallow tracks could occupy.
curve that meant it reached down to something like –20dB around the 1kHz mark. For the high–frequency track, a slightly
steeper high–pass filter centred on 85Hz reached 0dB at 1kHz, rolling off to about –12dB at 100Hz. I spent a while playing
around with the steepness of the filter curves because I find they can otherwise sometimes create a significant ‘null’ at the
point they cross over. Being careful not to overdo it — which would create a big and unwanted hole in your track! — you can
use this effect to scoop out some potentially ‘muddy’ frequencies around the 200–350Hz range, without resorting to further EQ
on the part as a whole.
I wanted to make sure that the low–bass part was controlled

enough that I could use it quite generously in the mix, and that
meant controlling it both in the frequency domain and in terms of
dynamic range. Thinking about the variety of playback systems
the track might encounter, I decided to remove the really
subsonic frequencies (below 25Hz), as they weren’t really adding
anything useful and removing them would leave me with more
mix headroom. I then applied a serious amount of compression
and limiting, to pin this low bass firmly in its place, removing
most of the dynamics and movement in level between the
different notes of the bass line. Using a compressor with a fairly
fast attack and release, I ended up with about 6–10dB of gain The bass part was multed, with each version filtered so that
reduction, and then paired with a brick–wall limiter, which tamed the low- and higher–frequency aspects of the sound could be
processed independently.
things by a further 5–8dB. When you start focusing on this DAW Tips from SOS
process, it’s quite interesting to observe just how much
100s of great articles!
difference in terms of gain reduction there can be between the
Cubase
different notes of a bass line — this can give you a good feel for
Digital Performer
just how much energy and space is often taken up by the low
Live
bass in a mix.
Logic
A Test Of Character Pro Tools
Reaper
With the low bass now providing the desired power and chest– Reason
moving qualities at high volumes, I could use the high bass Sonar
region to give the part a bit of character, whilst also ensuring it
would make a positive impression on smaller speaker systems.
In contrast to the approach I used for the low–bass track, I

didn’t set out to apply any compression but, instead, set out
solely in search of character. I experimented with a number of
different plug–ins before settling on what, when it comes to bass
processing, is an old favourite of mine — Sound Toys’ Devil–Loc
Deluxe, which is an emulation of the old Standard Audio Level–
Or.
Technically, the Devil–Loc was compressing the signal, but

that was more of a side-effect; I was using it more for distortion
and tonal sculpture than anything else. Used fairly sparingly, this
helped bring a nice bit of ‘growl’ to the bass line, which seemed
to work really well. I found myself automating the blend control,
to avoid the effect becoming too stale, with the amount
increasing towards the end of the track. Introducing this touch of The low bass was heavily compressed and limited to hold it
distortion, along with a whisker of gentle EQ boost at 1kHz, firmly in place.
meant that the bass now had a slightly pronounced mid-range —
perfect for helping it to be audible on smaller playback systems.
Kick Drum
With the kick drum, I took a similar approach to the bass synth
by ‘Multing’ or duplicating the track. This time, rather than
copying the part, I simply sent some of the signal to another
track (some DAWs let you route to other audio tracks, others will
require you to use a group bus or dedicated effects channel for
this). Although I quite liked the sample Patrizio had used for this
part, I felt it was a bit lacking in low bass; the sound was nice but
it didn’t quite feel powerful enough. So, after setting a low–pass
filter turning over at around 160Hz on one of the two channels, I
exaggerated the low end considerably with the Waves Puigtec High–bass processing: a touch of distortion, courtesy of
plug–in, a Pultec EQP1A emulation, at 100Hz. On its own this SoundToys Devil–Loc, and a small EQ boost to provide some
channel sounded quite over the top, but the idea was that it extra mid–range presence.
could be blended in to taste with the higher, more ‘normal’ kick
drum track. As with the low synth–bass track, I also compressed and limited this part heavily, so those low frequencies were
firmly held in place — and that meant that it could be pushed in level where required, without risking any nasty surprises.
The remaining higher-frequency copy of the kick drum — the

‘normal’ kick drum sound — required very little processing. I set
a high–pass filter at around 150Hz, to leave those frequencies
free for the low kick part, and spent a while adjusting the shape
of the curve to create a suitable crossover. The effect of the kick
drum and bass synth having nulls at different frequencies, as
created by my DIY crossovers, was that the two parts occupied
their own space — if they occupy the same space, they tend to
interfere with each other, robbing the track and its groove of
impact.
Often, when working on electronic or programmed material,

you can find yourself doing very little mix processing and The low frequencies of the kick were treated to parallel
worrying that you’re not doing enough, but remember that drum processing, with heavy compression and EQ greatly
samples or loops have probably already been compressed and exaggerating the low end.
EQ’d two or three times before you get your hands on them, so
always trust your ears! The only other treatment this particular kick track required here was a whisker of a boost around 3kHz
and 4.5kHz, which could be fine–tuned later if the kick needed a little help to cut through in the mid–range.
Multiband Options
Whilst I settled in this mix on splitting my bass and kick drum into just two parts, you could easily develop the technique by
adding a third band. This would enable you to go further with your split processing, perhaps by adding a stereo element to
your higher bass part, courtesy of a phaser, chorus or some other spatial effect: whatever you find works. When doing this,
you have to be careful to make sure those crossover points aren’t too wide, though. (Sometimes I’ll even draw a little sketch,
just to get an idea of how the low and high–pass filters are crossing over.) Also, you need to make sure that your DAW’s plug–
in delay compensation is up to scratch, because some CPU-heavy, or older, plug–ins can introduce a little latency even when
automatic delay compensation is switched on in some hosts, and if you’re processing different versions of the same track with
different plug–ins, that will mean they become slightly out of phase. The bottom line there once again is to use your ears.
The technique described in this article was effective here, but it doesn’t always work so well, and I certainly find it less
effective when working with live drums and bass — where each hit and note is slightly different, and the result seems to ‘move
around’ that much more. With dance or other programmed material, though, the nature of the music is more consistent, so you
get a more pronounced sense of the low, medium and high–frequency parts of the sound — which means it’s a lot easier to
home in on specific areas of those sounds.
It’s amazing how quickly things come together in a dance mix once the bass is working as it should. Patrizio had worked
quite hard on his arrangement so the rest of my job was very much about balancing levels and setting the panning to create
an interesting stereo field. Once I’d had some fun playing with delays and effects on some of the synth stab parts, my focus
was largely about making sure the mix had the right broad balance between the bass elements I’ve already described, the
synth and drum elements sitting in the mid–range, and the higher shakers and hand claps, which provided most of the top–
end information.
Bass–ic Sanity Checks

I’ve developed a few very non–scientific techniques for checking if a bass part is too prominent in a mix, and I use these in
conjunction with reference mixes to keep me on track. Although I don’t employ a dedicated subwoofer in my setup, I have a
set of Mackie HR428 speakers to complement the Adam A7’s I use as my main monitors, and I can, if required, pump a mix
out to a larger PA in a nearby rehearsal room. But checking bass isn’t all about large speakers — I find my ‘grot box’
particularly helpful in checking the bass. I certainly wouldn’t do any serious EQ work listening on it, but if I can get the volume
up quite high without the speakers waving the white flag, that seems to be a good indicator that the bass is nice and tightly
controlled! If I need further convincing, I’ll compare my mix to a couple of reference mixes on my headphones — removing the
room completely from the equation.
People often seem happy these days to bypass the professional mastering stage,
but if you never use a good mastering engineer you’re missing out on one of the
most important contributions they can make. Proper mastering engineers have
serious listening environments and ears that are tuned by experience to judge
appropriate levels of bass for real-world playback systems — so even if you only
get a track or two professionally mastered, that could provide you with a good sonic
reference for your other projects. Despite spending unhealthy amounts of time on
their own, most mastering engineers will also be happy to give you some often quite
specific feedback on your mixes — and that can be enormously helpful in improving
your own listening and mixing skills. .
The bass was checked on a few different

speakers, including a ‘grot box’, which
provided me with a good indication of
whether the bass was becoming over–
inflated.
Room For Improvement?

If your living situation prevents you from installing any serious acoustic treatment, there’s a lot to be said for getting to
know the limitations of your listening environment so that you can work within them. With careful use of reference material,
regular checking of your mixes on multiple playback systems, and the help of a decent set of headphones (which, of
course, takes the room sound out of the equation) and frequency analyser plug–ins, you can usually achieve decent
results, albeit with a little hard work. Working in that way, though, can be both frustrating and time–consuming, so if you’re
serious about improving this area of your mixes you really must do something to treat your listening environment.
Otherwise you just won’t be able to hear what’s going on with your low frequencies in particular.
Describing how to set up and tune a whole room is way beyond the scope of this article, but there are some simple,
quick and affordable ways to make significant improvements. My own mix room is about 3.5 x 4.2 m in size and all the
acoustic treatment I’ve used to target the lower frequencies is based around sheets of medium- to high–density Rockwool,
which I’ve positioned in the corners of my room. Unlike treating other areas of the frequency spectrum, where too much
treatment can make things sound too dry and skew the frequency response, it’s really very hard to overdo it with bass
treatment — and precise placement is less critical, too. Sort out the corners, and you should make a decent improvement.
If you’re struggling to fit things in the corners, remember that a cube–shaped room has 12 ‘corners’ — 12 junctions
between the walls/floor/ceiling — that may be treated, not just the four obvious ones. I’ve just installed traps covering the
top half of the two back walls in my room, to preserve some workable space.
Mineral wool is often sufficient, but it’s bulky, and a more efficient trap can be made using a ‘limp mass’ barrier mat in
front of the Rockwool, enabling you to build a more compact or better-performing trap. You can find more details about this
in our Studio SOS article in SOS March 2006.
sosm.ag/diy–basstrap
Louder Or Better?
It’s easy to be seduced by a plug–in or effect like the Devil–Loc I used here — ones that makes the signal louder as more
of the effect is applied — because our ears often naturally perceive louder as better. You need to be able to make
judgments based on the sound, not the level, so it’s a good idea when auditioning such plug–ins to follow them with a
simple gain plug–in, such as the freeware ‘Sonalkis FreeG’. By counteracting the gain with your own manual ‘auto–gain’
adjustment, you can get a more accurate idea of what your processing is actually doing. Of course, if you’re using a plug–
in which already features a constant gain facility, make sure you use it!
A gain plug–in can be placed

after other plug-ins to take
loudness out of the equation
when trying to evaluate the
results of any processing.
Audio Examples
We’ve placed a number of audio files on the SOS web site which enable you to hear what Neil did with this mix. Included
are examples of the kick and bass, as well as the final mix.
sosm.ag/oct14-mixrescue-media

Reviews Information
SOS Site Analytics
In this article:
EQ III
Package Deal
AIR Vintage Filter Pro Tools Tips & Techniques Buy PDF
Bomb Factory BF76 Published in SOS October 2014

Technique : Pro Tools Notes
Signal Generator
Pitch II The plug-ins that come with Pro Tools are more powerful
than you might think.
Mike Thornton
A
ll too often, people dismiss factory-supplied plug-ins in favour of third-party offerings — but a lot of the free plug-ins
supplied with Pro Tools are great, and you can do a huge amount without spending money adding to them. Even
though Avid arguably have a reputation for not being generous when it comes to giving things away, there are some
gems that ship with Pro Tools that you should not ignore.
EQ III
The one-band EQ III plug-in is surprisingly useful. It uses very few processor resources, so you can use instances across
every channel in your mix as versatile high-pass filters, for example, or to flip the polarity of channels where required — the
Pro Tools mixer does not have polarity switches of its own, so it’s necessary to use a plug-in for this. In post-production work,
or indeed on anything recorded on location, I will use it across every track to filter out all the low-end rumble.
The seven-band version is a great, neutral EQ plug-in with a friendly user interface that enables you to get good results very
quickly. For me, it takes a lot of beating, and has the added advantage of being installed on every Pro Tools system, so there
are never any problems when you’re collaborating with other users who might not have the same expensive third-party plug-
ins as you. It is easy to get a telephone effect for vocals by just using the low- and high-pass filters, on the 24dB-per-octave
setting. Adjust them until you are left only with the mid-range. Another thing you can do is to automate the frequency of the
low-pass filter to give you that sweep effect you often get on dance tracks.
One last tip is that if you hold down Ctrl and Shift and then adjust a parameter, the plug-in will temporarily operate in band-
pass mode as long as you hold down the modifier keys. As soon as you let go, the plug-in goes back to normal, but it’s a
really useful feature to help track down problem frequencies.
AIR Vintage Filter

You can use this plug-in on synths, guitars, bass, drums and anything
else where you want to add character. By automating its controls, you
can create familiar dance-music effects like filter sweeps, and to give
you some ideas, I am going to show you a way of creating noise rises
by combining the AIR Vintage Filter with two other factory plug-ins.
You can see in the screenshot that I’m using the Signal Generator
plug-in as a sound source. You will find this in the Other submenu in
the plug-ins list, but note that it is only available in mono and multi-
mono formats, not multi-channel. In this application, you want to
choose the White Noise option. Instantiate the AIR Vintage Filter plug-
in in the next insert slot, and enable automation for its Cutoff control.
Now draw a ‘sawtooth’ automation shape for this parameter like the
one you can see in my automation lane. Try it first with the Resonance
control off, then experiment with different settings for a more musical
sound. You could even automate that control, to get more variety and
character to the sound.
If you don’t like the way the sound cuts off sharply at the end of the
sawtooth, you can try adjusting the automation so it isn’t a vertical line,
but for me, the nicer solution is to use some reverb. In this example I In the setup shown in this screenshot, the Signal
have simply used the AIR Reverb plug-in with an RT time of over three Generator and AIR Vintage Filter plug-ins combine to
seconds and maximum room size, but of course you could use any create a filtered noise rise.
reverb plug-in you have at your disposal. Finally, try changing the
Signal Generator to output pink noise: pink noise has much more low-frequency content than white noise, so the sweep effect
will be a lot deeper and you will hear it earlier in the sweep.
Bomb Factory BF76

There are a lot of plug-in versions of the brilliant Urei 1176
compressor, which has been used on more recordings than I’ve
had hot dinners. You may think that the free one that ships with
Pro Tools is not worth considering, but in fact it’s a great little
plug-in which compares very well with some rivals costing
hundreds.
Like the original hardware, the BF76 works differently from

most normal compressors in that it has a fixed threshold control.
To determine how hard the compressor will work, you adjust the
input control: the more you turn it up, the more the signal BF76: beware the backwards controls!
exceeds the threshold, and the more compression is applied.
The other slightly unusual thing is that the attack and release controls work backwards: on most compressors, you would turn
the controls fully anti-clockwise for a very fast attack and release, but here it’s the other way around. Talking of very fast attack
and release times, the 1176 is famous for its speed: push things too far and you can get distortion, especially on low
frequencies. For a free plug-in, this is an excellent and very usable emulation of an 1176 hardware unit.
Signal Generator
Its primary application may be as a tool for creating test tones — after all, that is what it was designed for — but the Signal
Generator plug-in does have some musical uses, as we have already shown with the AIR Vintage Filter. Another application is
to couple it with a side-chained gate to give kick drums some real low-frequency energy. In the screenshot, I have a kick-drum
track and a second track with the Signal Generator inserted on it.
I have set the plug-in to Sine Wave and set the frequency to
65Hz (you can play around with the exact frequency, and even
set it to the root note of the key of the song if you want). Next
insert an expander or gate plug-in: I have used the Avid factory
plug-in, but you could use any gate as long as it has a side-chain
key input feature.
Back on the kick drum track, I have set up an aux send to

send out the kick drum audio to a bus, in this case bus 1. Now in
the Gate plug-in, I’ve switched on the key input and set it to bus
1, then enabled the key input into the side-chain. When you play
the track, the kick drum will open the gate and let a burst of the
Signal Generator’s output through, then close. You can adjust
Another musical application for the Signal Generator plug-in:
the threshold, attack, hold and release to get the best sound for adding low end to a kick drum.
the gated tone.
Pitch II
DAW Tips from SOS
A relative newcomer, Pitch II is a long-overdue replacement for
Avid’s lamented Pitch plug-in, and is already a favourite of mine 100s of great articles!
when mixing. You can use Pitch II to widen guitars, vocals and Cubase
synths, or to change the pitch of an instrument in real time, and Digital Performer
because it uses granular technology, it sounds remarkably good. Live
Logic
I am old enough to remember using the iconic AMS RMX16, Pro Tools
and one of the settings that we used back then was ‘99s’, in Reaper
which we took a track or an instrument and applied a stereo Reason
detuning effect, taking the left-hand pitch down to 99 percent and Sonar
the right channel up to 101 percent. This creates a great
thickening effect that works really well on all kinds of audio. I Avid’s Pitch II offers a variety of widening effects.
especially liked using it on brass, but it works on vocals, guitars
and synth pads too.
We can reproduce the ‘99s’ effect described above using Pitch II, either as a send and return effect or, as in this case, by
inserting it on a track. First, you need to unlink the pitch section, because we want to pitch down the left channel and pitch up
the right. I have got very close to the ‘99s’ effect by setting the left ratio to 98.9 percent and the right to 101.12 percent (or 19
cents in both directions). You can adjust the Mix control to add as much of the effect as you like.
You can also experiment with adding small amounts of delay — try around 15ms — and using the low-pass filter (LPF) to
take off some of the brightness from the effected signal. On guitars you can use much longer delays to get a slapback effect,
and you can often get away with more extreme detuning and filtering too. It’s also worth experimenting with delinking the
audio controls, so you get different amounts of delay, feedback and LPF on both sides. You could automate them to give you
some extra colour and movement, too. This way you can end up with something very dreamy, more like a tape-echo effect.
Next month we will take a look at some more under-used plug-ins that come bundled with Pro Tools, such as Sansamp,
Eleven Free and AIR Lo-Fi. Until then, have fun! .

Reviews Information
SOS Site Analytics
In this article:
Colouring In
Peak Performance
Meter Ranges Sonar Tips & Techniques Buy PDF
Showing Peaks Published in SOS October 2014

Technique : Sonar Notes
Mastering Meters
Reset & Forget Get the visual feedback you need with custom metering in
Sonar.
Craig Anderton
S
onar’s metering flexibility extends not just to the ‘LED’ meters themselves, but to specialised settings that can help
when tracking, mixing and mastering. Although Sonar’s documentation is quite complete in describing the metering,
we’ll take a more application–oriented approach in order to get the most out of that functionality.
You can change virtually every aspect of the meter’s ‘ballistics’ by choosing Edit / Preferences / Customization / Audio
Meter. We’ll start with the Refresh Rate. For the ‘snappiest’ meter response, set this to 25ms. Faster refresh rates stress out
your CPU a bit more than slower rates, but with today’s CPUs that shouldn’t be much of an issue (faster rates are visually a bit
more jittery, so you might want to slow down the rate anyway). If you have a zillion tracks running on a laptop and need every
last bit of power, you can lengthen the refresh rate up to 250ms.
The other settings are Peak Hold and Decay times. These become less important if you select Hold And Lock Peaks
(described later), which I prefer in some situations because I find it easier to scan peaks visually on the meters than look at
numeric peak values. This dialogue box is also where you can edit the RMS– and peak–metering rise and fall times. Although
these default to suitable values, if you set the Peak Fall time to 100ms (faster than the default), then the meters for tracks with
delay or reverb that continue after pressing stop will persist longer than for the other tracks. This makes it easy to tell which
tracks are the ones with a time–based effects ‘tail’.
There’s also the option to choose non–segmented meters (ie. a continuous

strip of colour instead of segmented meters) separately for the Track and
Console/Inspector views. Non–segmented mode is necessary to change the
meter colours...
Colouring In
Choose Edit / Preferences / Customization / Colours, then select Meters from
the Colour Category drop–down menu. You can now specify colours for the VU
Hi and Lo levels, as well as the VU meter tick marks/dB marking. In Track View,
where I’m more into diagnostics, my preference is light blue for the Lo level and
orange for the Hi level, with both set to maximum saturation and the default
brightness setting. VU Tick Marks are black. This draws my attention to when a
track is getting close to the red, and also makes it easier to see locked meter
peaks against the orange. For Console View, I usually retain the standard
segmented meters.
Meter Ranges
You can configure the Track, Console, and Inspector view meter ranges
differently, as well as have separate ranges for the Record, Playback, Bus and
The Track View meters have been set to
Mains meters. This isn’t just a “because we can do it” feature: it is extremely non–segmented mode with a light–blue look.
handy if, for example, you use the Track view mostly for tracking and the They are oriented horizontally for higher
Console view for mixing. These two views can also give different ‘windows’ into resolution, calibrated to a 42dB range, and
have both Peak Hold and Peak Lock
the mixing process, so tailoring the meters for each one is helpful. The available enabled so it’s easy to see the highest levels
meter ranges are 12, 24, 42, 60, 78, and 90 dB; here’s how to select what you attained by the tracks.
want:
Change range for an individual meter: Right–click on the meter and choose the desired range.
Change range for all Console view meters: From the Console view’s Options menu, choose Meters and then the Record,
Playback, Bus or Mains options.
Change range for all Track view meters: This works similarly to Console view, although of course Mains (which appears only
in Console view) is not an option.
Change range for Inspector meters: Click on the Display button below the Inspector, choose Module Options / Meters and
then the desired Record, Playback, Bus or Mains options.
Try setting the Recording meters to the 90dB range if you want to catch any low–level signals (like hiss) creeping into the
recording. For playback, 42dB is a good balance between focusing on what’s happening at higher levels but not missing
lower–level signals. The main bus setting depends on the music I’m mixing; for dance music or high–intensity rock, I’ll use
12dB to make sure there’s plenty of activity at higher levels. If I’m not doing the mastering, hopefully this will also discourage
squashing during the mastering process. For more subtle mixes, 24dB or 42dB works well.
Showing Peaks
In Console view and the Inspector, a numeric peak value attained by a track during recording or playback appears below the
meter. Track View can also show numeric peak values (this appears to the left of the track’s meter if enabled), but has two
other options. Show Track Peak Markers and Show Bus Peak Markers display a marker on the track or bus itself during
playback or recording to indicate the highest peak attained; this updates upon encountering higher peaks. If you right–click on
a track’s Numeric Peak Value and select Go To Peak, the Now time will locate automatically to the track’s highest peak.
However, there are two important considerations. The Peak
Marker reading takes the track fader’s setting into account, so if,
for example, the waveform reaches a peak of 0.0 but the fader is
set to –1.0, the peak will show as –1.0. Also note that the peak
marker will appear 40 milliseconds after the peak itself if the
MeterFrameSizeMS variable (in the ‘AUD.ini’ configuration file) is
set to the default. A lower value records peak info in smaller time
slices, but increases stress on the CPU and uses more memory.
If you keep the default, after you click Go To Peak or locate it
visually, look for the highest peak 40ms prior.
Mastering Meters
The peak markers let you see (among other things) where a
track or bus went above 0dB. What’s more, this is invaluable
when mastering if you need to create as loud a master as
possible to please the client, but don’t want to overdo the Peak values are shown above the meters in the track
dynamics processing. Here’s how... headers, but the locations of those peaks can also be
‘flagged’ in the tracks themselves.
Bring the file to be mastered into Sonar. I recommend turning
off any plug–ins for that track so you’re compensating solely for
level changes within the audio file as opposed to changes DAW Tips from SOS
caused by a plug–in. Normalise the track to –0dB, not because 100s of great articles!
that’s necessarily what you’re going to do when you assemble an Cubase
album, but because we need a reference level. Then, decide by Digital Performer
how much you want to increase the overall level — let’s say 2dB. Live
Logic
Play through the file to find the first peak that’s over –2.0dB (or Pro Tools
find it with Go To Peak). Select the half–cycle containing that Reaper
peak (it’s helpful to enable Snap to Nearest Zero Audio Reason
Crossings, found under Edit / Preferences / Customization / Sonar
Snap to Grid). Then use the Process / Normalise function to The Track Peak Marker lags 40ms after the highest peak,
normalise the peak to –2.0dB. Proceed through the file and which you can see here being selected and normalised to –
reduce all peaks greater than –2.0dB to –2.0dB, and now you 2.0dB. The peak to the left of it will need normalising as well.
can raise the level of the entire file by 2dB. Because you’re
processing such a short part of the signal and not creating any artifacts, these changes are essentially inaudible. If you do end
up using limiting or compression, you won’t need to add as much, and the effect won’t be as drastic. When assembling an
album, you can then raise or lower the track’s level to match the other tracks.
Reset & Forget

If you use Peak Hold and Lock, the meter peak values and held peaks stay visible after stopping the transport. Hitting Play
resets the peaks; you can also choose Reset Meters from any Options / Meters menu. However if you do that a lot, save time
by assigning a Key Binding to ‘Reset All Clips and Numeric Peak Indicators’.
Don’t overlook some of the other options, like choosing horizontal meters for track view if you want to see really high
resolutions, specifying pre– or post–fader for buses, and selecting among Peak, RMS, or RMS + Peak indicators. The ‘right’
choice of meter type is subjective; I often choose Peak + RMS for Track View so I can see at a glance the relationship
between peak and average signals (useful if there’s any dynamic processing in play). Some people prefer Peak for Track View
and RMS for playback with the Console, but with the buses set for Peak to see any ‘overs’.
Finally, note that meter settings chosen under Preferences persist from one project to the next, but those chosen within
projects via the meter options menus or right–clicking on meters are stored with the project. So if you end up with metering
settings you really like, make them part of your normal template and they’ll load just the way you like every time you open
Sonar. .

In this article:
Take It To The Max
Pushing The Envelope
Target Practice Ableton Tips & Techniques Buy PDF
When One Is Not Published in SOS October 2014

Enough Technique : Ableton Live Notes
We delve into the mysteries of envelope following in
Ableton Live.
Len Sasso
A
n envelope follower lurks under the hood of each of Live’s dynamics processors. It detects the audio level on which the
rest of the circuits base their compressing, limiting or gating action. This month we’ll look at a variety of applications for
envelope followers beyond controlling dynamics. I’ll start with a quick look at other Live effects that incorporate an
envelope follower and then discuss several ways to use the stand–alone envelope followers available as Max For Live
devices.
Four Live audio effects have a built–in envelope follower dedicated to modulating a single parameter: Auto Filter (Filter
Cutoff), Dynamic Tube (Bias), Flanger (Delay Time) and Phaser (Frequency). Each effect’s Envelope knob is a bi-directional
modulation–amount control, and the associated Attack and Release knobs affect the rate at which the modulation tracks the
increasing and decreasing level. Except when using Auto Filter’s side–chain input, these envelope followers track the level of
the signal being processed. Three of them (Dynamic Tube is the exception) have an LFO to modulate the same parameter as
the envelope follower. With the envelope follower the modulation follows dynamics, whereas with the LFO it follows a random
or waveform–based pattern. You can, of course, use the envelope follower and LFO at the same time. With a steady rise or
fall in level, for example, the envelope follower will move the centre of the LFO’s range up and down as the level rises and
falls.
Take It To The Max

If you have Max For Live, which is included in Live Session,
you’ll find two envelope followers in the Max For Live/Max Audio
Effect section of Live’s Category browser: ‘Envelope Follower’
and ‘Max Api CtrlEnvFol’. A third alternative, ’REF’ ( 5 from k-
devices.com), offers better graphics and advanced features
(most notably a threshold control). Although their feature sets Live’s Auto Filter, Dynamic Tube, Flanger and Phaser have
differ, each of these envelope followers works in the same way: built–in envelope followers (boxed in red) to control a single
you insert it as an audio effect device on the track playing the parameter. Auto Filter’s implementation is the most versatile
in that it allows for side–chain input to the envelope follower.
audio whose amplitude envelope you want to follow, and then Here Auto Filter is set up for a variant of ducking: kick–drum
you map the envelope follower’s output, which is a control signal, hits momentarily raise the cutoff of a high–pass filter applied
to the parameter you want to control. Although you insert it as an to the bass track to get it out of the way of the kick.
audio effect, the envelope follower has no effect on audio; the
track on which it is inserted plays normally. An option to keep in mind is that you can insert an envelope follower on a Return
track and thereby use it to follow the mix of the audio and instrument tracks sent to it (you’ll want to mute the Return track if
you use this strategy). Follow these steps for a graphic look at how Envelope Follower works:
Create two audio tracks, place a four–on–the–floor kick drum loop on one and a simple rhythm guitar loop on the other.
Strums on the quarter–notes are a good choice for the guitar.
Insert Live’s Simple Delay on a Return track and set its right and left delay times to two and three 16th–notes respectively.
Play both loops and increase the guitar track’s Send knob to audition the effect of the delay without Envelope Follower.
Insert Envelope Follower on the kick drum track, click its Map button and then click the guitar track’s Send knob.
Start both tracks playing and observe the Envelope Follower display. You’ll see it clearly tracking the volume of the kick
drum hits, and you’ll see the guitar track’s Send knob moving accordingly. You may or may not hear any of the Simple
Delay effect, depending on the level and shape of the kicks.
Note the two number fields adjacent to the Map button. They set the minimum and maximum values of the targeted control.
Change the minimum (left) to 100 and the maximum (right) to 0. You’ll now hear truncated delays because the Send level
traces the inverse of the kick envelope: it is maximum before the onset, drops quickly with the kick onset and then rises
back to maximum as the kick decays.
Return Envelope Follower to its default settings and create a modulation envelope for Envelope Follower’s Gain knob. Set
the Gain modulation to 0, 3, 6, and 9 dB respectively for the four kicks. You’ll hear the delays rise accordingly during each
loop of the kick.
Finally, delete the modulation envelope and increase the Gain to produce nearly full–level envelopes on each hit. Now play
with Envelope Follower’s Fall, Delay and Min settings. Their effect will be obvious.
Target Practice
As alternatives to modulating the Send level to Simple Delay you could modulate the Return track’s Track Volume slider, or
you could move Simple Delay to the Guitar track as an insert effect and modulate its Dry/Wet knob. With both alternatives,
Simple Delay will receive and process the full guitar loop rather than only the parts allowed through by the gyrations of the
Send knob. Envelope Follower’s action will then modulate Simple Delay’s output; the amount of delay that is heard. The
results are not as dramatic, but may be just what you want.
Many effects parameters other than level make good targets; reverb decay, filter resonance (Q), resonator gain and
distortion amount, to name just a few. Keep in mind that it can make a difference where you place the envelope follower
relative to the device it is modulating. If the envelope follower comes after the device and the modulation affects volume, that
will influence the envelope follower’s action (because it follows volume). This can be useful, but when you want to avoid it,
place the envelope follower before the device so that it follows the audio entering rather than exiting the device.
So far I’ve focused on effects processors, but instrument
controls make equally good targets. One such application is
fading instruments in and out, and if you have an Instrument rack
holding different instruments, targeting the rack’s Chain Selector
is one way to go about it. Another way is to use Live’s
Crossfader to crossfade between instruments on separate
tracks. To target the Crossfader, you’ll need to use Max Api
CtrlEnvFol because it’s the only envelope follower that lets you
select the target by menu, and that is the only way the
Crossfader is accessible.
When One Is Not Enough

The Max For Live Envelope Follower audio effect is inserted
One feature none of these envelope followers offers is the ability on the Drums track with its envelope output mapped to the
to control more than one parameter. There are two workarounds Send A knob on the Guitar track. The A Return track holds a
Simple Delay set for two– and three–16th–note delays. The
for this. The most obvious is to use a separate envelope follower waveform graphics at the top show the guitar loop before and
for each parameter you want to control. Because they pass after Simple Delay without using Envelope Follower. The
audio through unaffected, you can arrange the envelope bottom waveform graphics show the results with the
Envelope Follower settings shown above them.
followers in series in the path of the audio whose level you want
to follow, and that lets you see and edit them all at the same
time. You’ll save lots of time dialing in the envelope followers’ DAW Tips from SOS
settings if you enclose them all in a Group and then link their 100s of great articles!
common controls using Macro knobs. On the other hand, you’ll Cubase
want to avoid linking controls such as Min and Max for which Digital Performer
you’ll probably want individual settings. Live
Logic
When the devices whose parameters you want to control are Pro Tools
on the same track, you can enclose the devices in a Group and The Instrument rack on the left holds piano and pad Reaper
then map all the controls you want to target to a single Group instruments in separate chains, and its Chain Selector is Reason
used to fade in the pad. The REF envelope follower on the
Macro knob. You’ll be able to set the range of the Macro Knob right targets the Chain Selector (orange outline), and its Sonar
individually for each targeted control so that the effect of the Threshold is set so that no envelope occurs at lower volumes
envelope follower will differ for each of them. Setting this up is a (yellow outline). REF’s Attack is set to smooth the onset of
the pad (green outline).
bit tedious, and envelope followers are not CPU hogs, so I prefer
using multiple envelope followers whenever possible. .

Reviews Information
SOS Site Analytics
In this article:
Scratch & Sniff
Session Notes
Showtime Blue Rose Code: Recording A Live Show Buy PDF
Record In Haste, Mix Published in SOS October 2014

At Leisure Technique : Mixing
Spills, Thrills & We overcome unfriendly acoustics and troublesome spill to
Bellyaches
capture a memorable gig.
Double Trouble
Sam Inglis
Blue Rose Code
T
Audio Examples o weary passers–by and workers in London’s busy King’s Cross area, St Pancras Old Church and its grounds provide
Audio Examples welcome relief from the pressures of the world. And to musicians seeking an unusual location for a concert, it makes a
dramatic alternative to conventional music venues. Among the artists who have taken advantage of this striking space,
hidden away in a quiet churchyard behind St Pancras station, is singer–songwriter Ross Wilson, who records and performs
under the name Blue Rose Code. Ross hit on the Old Church as the perfect venue in which to launch the second Blue Rose
Code album, The Ballads Of Peckham Rye.
For his tour in support of the Arts Council–funded long player, Ross had assembled an excellent five–piece backing band,
and was keen to have the album launch recorded for posterity. Having recorded a couple of Blue Rose Code shows before, I
was happy to oblige — but when I started to think about the practicalities, it soon became clear that this was going to be quite
a challenge.
With no permanently installed PA, live sound in the church is

handled by Joel Cormack using a pretty basic setup centred
around a small Behringer Eurodesk mixer and a pair of Mackie
powered speakers. The interior of the building is almost entirely
faced with stone, creating a pretty hostile acoustic environment
for anything remotely loud or uptempo. And the band’s line–up
would involve no fewer than six vocal mics, not to mention a full
drum kit and potentially awkward instruments such as violin and
double bass.
Scratch & Sniff

Happily, I was able to contact Joel well before the event, and he
proved friendly, helpful and calm. As I don’t own a microphone
splitter, and there was no budget to hire one, my plan was to Blue Rose Code on stage. From left: Jessica Moncrieff (violin
‘sniff’ signals from the insert points on the Behringer mixer. This and backing vocals), MG Boulter (pedal steel and slide guitar,
barely visible!), Samantha Whates (backing vocals), Ross
requires special balanced cables with tip and ring wired together Wilson (guitar and vocals), Daniel Paton (drums) and John
at one end, which I already had, and of course is only an option if Parker (double bass)
the front–of–house engineer isn’t using the inserts, which Joel Photo: Zöe Barrie
wasn’t. This setup, on paper, would give me line–level feeds of
all the channels that were going through the PA, and by pairing a Creamware A16 Ultra A–D converter with an RME Fireface
802 interface, I cobbled together the necessary number of line inputs.
However, Joel explained that he doesn’t usually mic up drum kits or other loud instruments, so I knew I would need mics
and stands of my own. He also pointed out that his mix position is near the back of the church, so a reasonably lengthy snake
would be needed to connect these to my recording rig. It was about this point I abandoned the idea of getting to the gig on the
train...
By the time I’d made it through the London traffic, Joel’s setup was almost
complete, and as it turned out, he was using one drum mic: a Shure SM57 in front
of the kick drum. The drum in question was a small ‘cocktail’ specimen with a full
resonant head, so this sounded rather soft and lacking in impact; I thought about
adding a second mic on the beater side, but it proved physically impossible to
position one. I miked the snare with a Beyer M201, and used a not–very–widely
spaced pair of Neumann KM84s as overheads. I chose these because I knew that I
would be fighting significant bleed: KM84s have excellent off–axis response and
their small size makes them easy to position, so I could get them fairly low over the
kit.
The other vital source that I couldn’t rely on capturing via Joel’s FOH setup was
the audience. Though I realised I was already likely to be fighting an overload of
room ambience at the mix, I also wanted a true record of how the band sounded in
the hall, so I set up a Pearl ST8 variable–pattern stereo mic near the mix position to
capture both. As some audience members were sitting around and behind this mic,
theory suggested the use of a stereo configuration with a 360–degree acceptance
angle, such as M/S with an omni Mid mic. However, most of the action was coming
from the front, and as applause isn’t something that demands to be reproduced with
pinpoint stereo imaging, I compromised by using a subcardioid Mid and figure-of–
eight Sides setting. One of the great advantages of M/S is that as long as the Mid
mic is positioned appropriately, you’ll always have a decent mono signal to fall back on.
Showtime
All of my mics went through my Focusrite ISA828 preamp and into line inputs on the Fireface. This, in turn, was hooked up to
an Apple MacBook Air running Pro Tools 11. To keep things simple and avoid the need for USB hubs, I recorded directly to the
system drive. As well as the five mic signals of my own, I ended up with 14 active signals from Joel’s desk. Jessica Moncrieff’s
violin, John Parker’s double bass, and Ross’s acoustic guitar were all coming into the desk through DI boxes, along with
Cyrus Shahrad’s stage piano (in stereo) and a second acoustic guitar that featured on a few songs. MG Boulter also played
pedal steel through a Fender Deluxe amp; this was miked up, as were the six vocals and the aforementioned kick drum.
Finally, it turned out that Ross had a surprise up his sleeve in the shape of legendary double bass player Danny Thompson,
who would be taking over Parker’s instrument for part of the encore.
Thankfully, both support acts were solo artists, so most desk

channels could be left unmolested between soundcheck and
showtime. Recording the soundcheck allowed me to confirm that
all the signals were reaching my hard drive, but also threw up a
few red flags. Several of the sources I was sniffing from Joel’s
desk were clipping, so I had to ask him nicely to turn the gain
down at the preamps. More of a problem were the weedy kick-
drum sound and the Fender amp, which was surprisingly noisy.
My biggest concern, though, was the double bass. This was
amplified using a miniature microphone, presumably an omni,
mounted to the bridge — and if I hadn’t known better, I’d have
assumed that it was actually miking the drums! We were able to
improve the pedal-steel situation slightly by repositioning the mic, DAW Tips from SOS
but there was little I could do about the other problems except
cross my fingers.
Singing drummers are always fun to mix. Here you can see Cubase
Having recorded Ross before, I was also forewarned of a few the two Neumann KM84s I used as overheads, low over the Digital Performer
kit, and Beyer M201 on snare drum. Live
other likely issues. In the heat of performance, both he and his
female backing singers Samantha Whates and Jessica Moncrieff Logic
have a tendency to wander away from their microphones; had I had the budget and time, I’d have been tempted to attach a Pro Tools
lavalier radio mic to him. Once he gets going, his percussive guitar playing also puts out huge low–frequency thumps, and I Reaper
warned Joel to leave plenty of headroom. Reason
Sonar
Recording unrepeatable events to a laptop with no real–time

backup isn’t the most comfortable thing in the world. From
software crashes to cables becoming dislodged, there’s endless
potential for things to go horribly wrong. My nerves weren’t
eased when Pro Tools decided to stop recording and report a
non–existent CPU overload during one of the support slots, but
when it mattered, it managed to get through Blue Rose Code’s
ecstatically received 100–minute headline set without falling
over. The house lights were barely up before I was copying
20GB of audio files to a portable hard drive, and as ever,
breaking everything down was a hundred times easier than
setting it up.
Record In Haste, Mix At Leisure

Front–of–house engineer Joel Cormack (rear) and I set up
The next day, copying the files onto the Windows machine I use our equipment on a table at the back of the church.
for mixing confirmed that every source had been captured all the
way through, with no dropouts. However, it was far from being a pristine studio multitrack. I usually find that levels are all over
the place with signals that are ‘sniffed’ from the FOH desk, and so it proved here: some were healthy to the point of
occasionally clipping, while others were barely present at all. Low levels are not, in themselves, a problem in a 24–bit
recording, but for some reason the stage piano was both vanishingly quiet and audibly noisy. The vocal tone constantly
changed as Ross and his backing singers moved around, and, as I’d feared, my crisp drum sound very quickly went to the
dogs as soon as I raised the fader on the double bass.
Placing a room mic in the middle of the audience can be disastrous if people sitting nearby want to hold a conversation or
eat crisps, but the punters had been very well behaved, and the applause sounded good. Its audience perspective on the
sound of the band, however, was enlightening. Despite the presence of 200 human bodies, the ambience in the stone church
still sounded hard and mid–rangey.
Mixing a live recording often involves quite a bit of extra effort compared with working on well–recorded studio tracks, but
this labour is usually front–loaded: while many of the sources will require a lot of corrective EQ and other processing, you can
normally rely on being able to use similar settings across the entire set. In other words, once you’ve successfully mixed one
song, the rest will follow fairly easily. In this case, however, things didn’t really work out that way, because no two songs used
the same line–up or arrangements. The drum kit featured only on about half the songs, and even then, the drum sound varied
as drummer Daniel Paton switched between sticks, brushes and beaters. Thankfully, the buzzing piano was used only on a
couple of tracks, and then in a background role.
Anyone who’s mixed a live recording will be familiar with the

problem posed by guitars and other instruments DI’d via a piezo
pickup. No matter how much mid–range you cut, there always
seems to be too much; and no matter where in the mid–range
you cut it, some other part of the mid–range always sticks out.
Thankfully, the violin had been captured using a clip–on mic,
while MG Boulter’s slide guitar actually sounded pretty good; but
Ross’s guitars were typically spiky, requiring multi–band
compression as well as EQ to bring them under control. Ross, as
predicted, did his share of guitar–whacking, but played other
tunes fingerstyle, and also swapped between two guitars which
sounded somewhat different from each other. To add to the fun,
one of these had been dropped on a stone floor before the set,
and emergency repairs to a machine head hadn’t prevented it
‘Sniffing’ the insert points is a cheap but rather inflexible way
drifting out of tune. of recording the signals going through the FOH desk. It also
Meanwhile, reining in the variable vocal level and tone was not has the potential to generate some serious cable spaghetti!
easy, and my eventual solution involved an equaliser to remove
unnecessary mud, followed by a de–esser, and a multi–band compressor to keep the mids and low–mids in line. This then fed
a conventional compressor, with plenty of level automation also required. On top of all that, there were quite a few places
where I needed to apply a high–pass filter momentarily to address plosive pops. Similar treatment was required for several of
the backing vocals, too.
The question of vocal reverb posed a dilemma. The room mic

and spill from the vocal monitors onto other sources had ensured
that plenty of ‘reverb’ was captured, but unsurprisingly, it didn’t
sound very good. On the other hand, if I added enough artificial
reverb to overshadow this spill, the vocal ended up sounding too
wet, and the distinctiveness of the venue was eroded. In the end,
I decided that the best reverb plug–in in the world was never
going to make this sound like it had been tracked in Abbey Road,
so I erred on the side of using the room mic, albeit heavily
equalised. This was augmented with a short ambience and a
larger chamber from Acon Digital’s Verberate and Exponential
Audio’s Phoenixverb respectively.
Spills, Thrills & Bellyaches

Although all of these issues took a fair amount of trial and error to address, and some required solutions which compromised
the overall quality, none of them threatened to make the tracks unmixable. The biggest problem in this respect was the double
bass (see box). This can be a difficult instrument to record and mix at the best of times, and in this case, the level of spill from
other sources caused real problems. It’s heartbreaking when an otherwise decent drum or vocal sound suddenly turns to
mush as soon as you unmute the bass channel!
Ultimately, though, when you’re focused purely on technical considerations, it’s easy to worry too much about them. A live
recording is a ‘warts and all’ record of a moment in time, and there’s more to be gained by accentuating the ‘realness’ of the
performance than by tidying everything up to the point where it becomes sterile. When I began sending Ross rough mixes, it
was clear that he was happy with the sound quality. More than that, he was overjoyed to have a document of a one–off event
that meant a great deal to him, and is hoping to put the recordings out as a live album or EP for fans. The sound of a large
and appreciative audience really enjoying themselves does a lot to compensate for imperfections! .
Double Trouble
By far the biggest headache I encountered in mixing the Blue Rose Code show was John Parker’s double bass. The
problem had nothing to do with his playing, which was excellent, but with the sound that was coming from the bridge–
mounted miniature microphone. In its raw state, the signal sounded like a fairly trashy room mic, with as much drum and
vocal spill as bass; and what it had captured of the double bass did not sound very natural. The low frequencies and the
percussive ‘snap’ of the strings on the fingerboard were emphasised, while the mid–range ‘growl’ that is so characteristic
of the instrument (and so essential to making it cut through on small speakers) was largely missing. Any attempt to boost
the mid–range using EQ simply delivered more and nastier–sounding drum spill!
I tried all sorts of fancy solutions to the problem, including an attempt to
restore the missing ‘growl’ by sending a heavily filtered version of the source
signal to a distortion effect, and using Melodyne to generate a MIDI bass part
from the audio recording (which failed because of the amount of spill). In the
event, it proved impossible to make radical changes to the bass sound without
colouring the spill to a point where it ruined the entire mix, so I settled on a
chain of plug–ins configured to reduce spill as much as possible while
controlling the low end. Again, the heavy lifting here was done by FabFilter’s
excellent Pro–MB multi–band dynamics plug–in. One of the great features of
this plug–in is that you can set each frequency band’s key filter range
independently from the range the band covers. Using this trick I was able to
control drum spill to a certain extent by configuring a band’s key signal to
focus on the high frequencies, so it would reliably trigger only on spill, yet
having the band itself duck a much wider frequency range whenever the snare
or hi–hat hit. At the other end of the spectrum, Pro–MB also gave me control
over the rampant sub frequencies.
No matter how many times I returned to the bass with fresh ideas, though, it
remained the limiting factor as to how good the overall mixes could sound.
Without the mid–range ‘growl’ to help it cut through, the bass sound itself
always felt indistinct and lacking definition; and it was impossible to make the MG Boulter’s pedal steel went into his
bass audible in the mix without spill affecting both the vocal and drum sounds. Fender Deluxe amp. Perhaps because
There were, at least, a couple of songs which featured double bass but not he had the amp turned up loud in order
to control his level with a volume pedal, it
drums: here the bass was more exposed and spill less of a problem, so I could was surprisingly noisy.
push the mid–range a lot harder. Elsewhere, the result was inevitably a
compromise.
Folk–rock royalty in the house: Danny
Thompson guests on ‘This Is Not A Folk
Song’.
The presence of a capacity audience did something to tame

the reverberation within the church, but didn’t make an
enormous difference.
Blue Rose Code

Edinburgh–born singer–songwriter Ross Wilson, now based in London, has built up a big following for his music the old–
fashioned way, through touring and word of mouth. Despite much interest from record labels, he chose to set up his own
label to retain complete creative control, and 2013 debut North 10 brought him to wider attention, both in the UK and
further afield. New album The Ballads Of Peckham Rye features both the cream of Scotland’s jazz scene and double
bassist Danny Thompson, known to millions for his work with Pentangle and John Martyn.
Audio Examples
To hear songs from the show, and snippets that illustrate some of the problems encountered at the mixing stage, point
your browser at www.sosm.ag/bluerose.
Audio Examples
Recording and mixing a live show is always a challenge, and these audio files from Blue Rose Code’s set at St Pancras
Old Church illustrate a few of the issues that can arise!
First of all, here are two complete, mixed songs from the show. I’ve chosen ‘Edina’ and set closer ‘Julie’ because I think
they provide good examples of the way in which the energy of live performance and the presence of an enthusiastic
audience help to overshadow technical issues with the recording. Because there are no drums in ‘Edina’, the spill onto the
double-bass mic was manageable, but both the pedal steel and the piano were very noisy. To bring across the audience
response in ‘Julie’ I needed to keep the room mic fairly high in level, despite the dodgy acoustics; the bass sound also
suffers because of spill issues.
One of the problematic elements of this particular live recording was an inconsistent vocal sound. In the first of these
two files, you can clearly hear how Ross Wilson’s vocal changes radically in level and timbre as he moves around. The
second file is the same track as used in the mix, with multi-band compression to even out the timbre and extensive level
automation.
The most troublesome source in the mix was the double bass mic, which captured almost everything but double bass! In
the first of these examples, you can hear how the drum sound goes to the dogs when the bass track is unmuted halfway
through. The second and third represent respectively the raw bass recording, and my attempt to turn it into a usable
double-bass track by reducing spill.
This is what it actually sounded like in the room!
If you mix a lot of live recordings, you’ll have to find ways of making acoustic guitars recorded with piezo under-saddle
pickups fit into the track. As you can hear, the raw sound in this case is very aggressive, and I’ve tried to soften it using
multi-band compression and a short ambience reverb.

In this article:
First Among EQs
Side Effects
Bundles Of Joy Digital Performer Tips & Techniques Buy PDF
Useful Utilities Published in SOS October 2014

Technique : Digital Performer Notes
Forking Out
Easy Does It Enjoy the creative possibilities of taking Digital Performer’s
bundled plug–ins out of their comfort zone.
Robin Bigwood
M
ost of the time, when mixing in a DAW like DP, it’s sensible to use the ‘right’ plug–in for the job. Need a tone tweak?
Then reach for an EQ. Need a bit more scale and punch for a vocal? Load up good compressor.
However, it can be interesting and creatively liberating to take a more oblique approach to things. By deliberately opting for
a less obvious solution, you’ll often achieve unexpected results, and be creatively challenged, in a good way! And it’s all part
of really knowing and understanding the tools you have at your disposal, and how even the apparently most mundane can be
superbly powerful.
So here are various processing and effect types, along with the plug–ins and settings that might not usually spring to mind.
The fact that they all exist in stereo–to–stereo versions mean they’re quite suitable for stereo tracks or even whole mixes.
First Among EQs

Two of DP8’s guitar amp plug–ins can be amazingly effective for
strongly flavoured ‘character’ EQ. In this case, ‘character’ can
stretch to some mild distortion too — but in a mix, and especially
for rock–style bass, vocals and keyboards, that’s no bad thing.
ACE30 is based on some classic Vox guitar amps. It has two

basic tone characters, Normal and Top Boost, which you choose
between by clicking one of the virtual input sockets at far left.
While Normal offers a reasonably flat frequency response, Top
Boost gets a big, energetic treble lift, and brings into play the Some of DP’s plug–ins have useful roles beyond what you
might normally expect from them. Here ACE30 is used as a
Treble and Bass knobs (and their associated Modern/Vintage general tone/EQ treatment, the guitar pedal Dyna–Squash as
switch). To calm it all down, there’s also the Tone Cut knob in the a ‘character’ limiter, and Echo as a track retiming tool.
Master section, which smoothly brings in a low–pass filter.
What’s unexpected are the ‘H’ and ‘L’ options for each of the tone characters. These,
respectively, emulate high– and low–impedance circuits at the amp’s front end, and they
differ significantly in their character. Low impedance offers a more open sound, while High
interacts with the input, especially on broad–band material such as a full mix or mix group,
and can produce some pleasant pumping artifacts that might sound great for, say, a drum
bus.
There isn’t really a setting at which ACE30 is truly ‘clean’, but the cleanest is to use the
Normal L input and keep the Normal volume control low and the Master volume high. This
generates a full–sounding, creamier version of your input, which often sounds good for
vocals. A world away, in fact, from the Top Boost H option. That’s not much use on a mix, but
quite possibly stellar for snarling acoustic bass or really cutting piano, and you can massage
it with those tone knobs in the Top Boost section.
Another amp, Custom ’59, can pull off similar tricks, though it’s more complex and a bit
dirtier all round. Here the inputs are a touch confusing: ‘I’ is bright, ‘II’ is darker, ‘1’ is high–
impedance (which also pumps a bit), and ‘2’ is the cleaner low–impedance option. There are
selectable options for input, preamp, tone stage and power–amp characters, and all are
usable; distortion is generally least noticeable when Power Amp is set to ‘Preamp’. I’ve had
great results using Custom ’59 on a sampled Rhodes piano,
where it seems to provide some weight and presence that is
hard to achieve using conventional EQs.
Bundles Of Joy
There aren’t, sadly, any Fairchilds lurking undiscovered amongst
DP’s plug–ins. But the guitar–oriented Dyna–Squash does have
real potential for use with drum loops. Try whacking both Output
and Sensitivity up full, and leaving the stereo response ‘lock’
switch unlocked. You instantly get a nice surging/sucking effect
on material that has plenty of dynamic range.
For generating convincing artificial reverberation, eVerb

doesn’t compare well with more modern reverb plug–ins, but it
still has tricks up its sleeve. By pulling down the rightmost handle
next to its ‘graph’ display you essentially turn off the reverb tail,
leaving only the early reflections (which MOTU call ‘Initial
Reflections’). Used on a vocal, this can sound like a distinctive
kind of chorus, much like double–tracking, and it easily morphs
into a tight ambience treatment useful for all sorts of sources.
There are five Initial Reflection patterns, selected from the pop–up menu. Then the three handles nearest to the lower-left
corner of the graph control level, pre–delay and ‘size’, which is the degree to which the reflection spatter is stretched out over
time. The same can be set numerically in the Initial Reflections value fields, once the full control panel is opened with the
‘expand’ button.
DP has a dedicated Wah Pedal plug–in, but for auto–wah instantiate yourself a Multimode Filter. Choose the Low Pass
mode, turn up the resonance, range and mix controls, and set the cutoff frequency (unusually labelled ‘center’) somewhere
around the middle. The modulation switch should go to the env (envelope) position, and you should use the graph handles
here to create a pyramid shape. Also, switch on the L/R In Phase option.
What’s happening here is that the envelope is being triggered by the

level of audio on your track. That then modulates the cutoff frequency
in the filter section. You’ll want to play with the resonance and center
values to get the right kind of tone, and then control the depth and
range of response with both the envelope scale and the filter range
parameters. The mix control sets the wet/dry signal blend, just as with
many other plug–ins.
Useful Utilities
If you ever need to shift an audio track a little way ‘late’, instantiate an
Echo plug–in for it. Set it 100 percent wet, put tempo lock to Realtime,
and set the gain of the feedback path and all but one of its four delay Digital Performer Courses
taps to 0. For the remaining tap turn the gain all the way up to 1.00 and More info...
then specify the track delay in milliseconds (from 1–2000) with the
DAW Tips from SOS
knob or value field below.
For everyday stereo ‘utility’ tasks, you don’t need DP’s complex Cubase
Spatial Maximizer plug–in, just the humble Trim. Instantiated on a Digital Performer
stereo track, it offers separate but linkable pan pots for each channel, Live
so it’s really easy to narrow the stereo image, or completely flip it. Trim Logic
The right settings turn DP’s Multimode Filter into an
can boost or lower gain up to 40dB too, and can tame the output of auto–wah plug–in. Pro Tools
distortion plug–ins (for example) that might be constantly driving your Reaper
mix into the red. Reason
Sonar
If you ever need to apply vibrato at the mix stage, as a special effect for example, the Chorus plug–in can help. Set the Mix
knob to 100 percent wet. Then the maximum pitch range of the vibrato is determined (rather unintuitively) by the Delay knob,
though it can be tamed with the Depth knob. The vibrato speed is governed by the different options found in the tempo lock
section — set to Realtime you’ll get an additional rate knob appearing.
Vibrato applied this way can be surprisingly extreme, and at faster rates and
greater Delay settings almost sounds like vinyl scratching. It also shifts the track
audio later with respect to every other track in your mix, so you may need to
Freeze the track and manually drag the resulting audio back to an earlier time
position to restore correct start timing. However, even a 1ms delay time gives a
useful amount of vibrato for most purposes, and that usually won’t need
correcting.
Forking Out
If you’re ever on a session and need to generate a tuning reference note,
instantiate the Tuner plug–in somewhere in your mix. Then just click the tuning
fork icon towards the lower left of its interface. A nice harmonically rich ‘A’
suddenly rings out, at 440Hz by default, but adjustable well over a semitone up
and down with the frequency field to the right. .
With its tail removed, so to speak, eVerb

becomes an interesting ADT and ambience
processor.
Easy Does It
There was great and not–so–great news recently from DP–friendly plug–in developers Audio Ease. On the plus side,
Speakerphone 2.1 has finally been released, providing much–needed 64–bit compatibility and Retina display optimisation.
It’s a free upgrade for registered users of version 2, and still comes in native MAS format, though you’ll need a second–
generation iLok to store the licence. With DP an important player in the movie–scoring and mixing field, this is a great
development.
But while Speakerphone and Altiverb flourish, the quirky Nautilus and Rocket Science bundles, which included the
unique and brilliant RiverRun granular processor and one–of–a–kind EQ Periscope, have been discontinued. The guitar
tone processor Cabinet has gone too. None of these plug–ins had made it out of the 32–bit world, and while Audio Ease
say they’ll continue to provide basic support and generate authorisation codes, they’re no longer available for purchase,
and the web site user forums have disappeared. Ah well, nothing lasts forever

In this article:
Man & Wi–Fi
Studio SOS
Buy PDF
Desk Job Published in SOS October 2014
Technique : Recording / Mixing
Install Wizard Printer-friendly version
Reader Reaction The SOS team visit a community youth centre to help
them make the most of their new studio setup.
Paul White
L
ocated near Walsall in the West Midlands, the Electric Palace is a centre for young people — one of three run by local
charity Bloxwich Community Partnership — and one of the activities it caters for is music.
The Youth Music project was started nearly a year ago to provide opportunities for nine to 19-year-olds to get involved in
music recording, playing and live performing. The studio is open during youth–club sessions and in the school holidays,
enabling young people to develop their songwriting skills, to record songs and to come away with a CD or MP3. They can also
go on to perform at the Youth Open Mic session, which is staged once a month to allow young people to perform their songs
in front of an audience, which in turn helps them to boost their confidence. The project encourages the participants to learn
new skills and hopefully aspire to study music or media production at college.
As the centre was only an hour’s drive from where Hugh and I live, we chose to make two visits for this Studio SOS, the first
being to take a look around and to list what we’d need to complete the job. We then scheduled a second visit to get the job
done, but not before Hugh took a few pictures of the current state of the room. Unfortunately Hugh called in sick as I was
about to set off to do the actual installation, so I did this one on my own, but be assured that he was there in spirit.
Centre manager Dan Garbett initially asked for our help in improving their music studio as, in addition to the usual acoustic
considerations, he had a new Focusrite Scarlett 18i8 interface to install. Dan had also recently acquired an iMac, however he
was used to working with PCs and so wasn’t very familiar with OS X. He’d already tried running one of the light versions of
Pro Tools (the DAW with which he was most familiar) using a two–channel Mbox, but two channels wasn’t going to cut it for
recording bands. The Scarlett 18i8’s four mic inputs, four line inputs and optional expandability promised far fewer restrictions.
Unfortunately the version of Pro Tools that Dan had wouldn’t

work with third–party interfaces, so I suggested that for the type
of work being done, Apple’s GarageBand would meet their
needs without being unduly complicated. Though not the latest
version, this was already installed on the iMac along with the
Mountain Lion operating system. While the software updater
keeps trying to tempt the user with a free upgrade to Mavericks, I
cautioned against this as all the stories I’ve heard, confirmed by
my own experience, are that this upgrade is really only
successful if you do a full clean install. If you simply update your
existing OS, as I did on my laptop, there seems to be a lot of
wasted time when switching between applications.
The Focusrite Scarlett comes with a few very useful bits of

bundled software, including three plug–ins, some loops and a Before: the desk that was to be the centrepiece of the studio
was originally set up along one of the longer walls of the
light version of Ableton Live, which is ideal for anyone creating room.
loop–based dance music — which, from Dan’s description,
describes most of the people using the studio. To control virtual instruments, Dan had already purchased an M–Audio USB
MIDI Mini 32 keyboard, and for the main vocal mic he’d chosen an SE Electronics X1 bundle including a pop shield,
shockmount and Reflexion Filter. In the monitor department, he had a pair of small M–Audio AV40 powered speakers, so all
we had to do was put it all together, find the best place in the room to do it and to try to improve the acoustics.
Man & Wi–Fi

When I first spoke to Dan, he told me that though they had broadband Wi–Fi in the facility, it didn’t reach as far as the room
they’d been given to use as a studio. Given the number of updates and authorisations that take place to keep even the
smallest studio ticking over, getting usable Wi–Fi to the studio had to be close to the top of my priority list. I suggested that
Dan use some of his very limited budget to buy a Devolo Wi–Fi range extender kit. I use one of these at home so I know they
work OK and that they’re fast enough to keep up with most broadband systems. Rather than run new network cables
everywhere, the Devolo system sends the broadband signal along the existing mains wiring from a sender unit that plugs
directly into a power socket and that connects to the building’s Wi–Fi router via an included Ethernet cable. One or more
receiver units then pick up the signal and retransmit it as regular Wi–Fi, where each of the receivers has its own password
code. These repeaters show up in your Wi–Fi settings as Devolo devices, separate from your original Wi–Fi router. We only
needed one receiver from the two–receiver pack that Dan bought, but he thought the other might be useful in their staff room
as getting Wi–Fi at that end of the building had proven difficult.
After some head scratching while getting the receivers to pair with the main unit, it turned out that all you had to do was
leave them alone for a minute or two after pressing the pairing button and they’d automatically sort themselves out. You know
they’ve paired when the home button LED stops flashing and stays on. Note that these systems only work reliably on wiring
systems that all go back to the same fuse box or consumer unit, and they can also be thwarted if you connect them to a plug–
board with RF filtering built in as that attenuates the high–frequency broadband signal. Luckily we had no such problems.
Desk Job
Once we had working Wi–Fi in the studio we could start sorting out the space. Dan had his studio desk set up facing one of
the long walls of the rectangular room, which is never ideal in smaller rooms, even with small monitors. The bass end is
invariably inconsistent so we had to do some furniture moving to allow the speakers to face down the long axis of the room.
We cleared the end of the room nearest the door so that we could put the desk there — it would be offset from centre slightly
because of the door, but I wasn’t too worried about that.
I put a couple of square metres of acoustic foam on the wall behind the speakers,
with two further areas at the left and right mirror points, plus a smaller area on the
ceiling above the edge of the desk. These were all intended to minimise reflections
getting back to the listening position. The right–hand mirror point was partly
occupied by a window with slatted blinds fitted, so I fixed the foam directly adjacent
to the window. On the other side we had to obscure part of a notice board. For this
project, most of the foam panels were kindly supplied by Comfortex, a UK
manufacturer of acoustic foam based in Oldham. They make a wide range of panel
and trap types but we used one of their smaller tiles as it made it easier to fit into
odd spaces without the need for cutting. We also used two larger panels on the side
walls, these kindly supplied by Universal Acoustics.
The rear of the room had already been treated with acoustic foam, which made it
ideal for vocal recording. It also helped reduce the severity of reflections bouncing
back from the wall to the mixing position. Using a Reflexion Filter behind the mic
Recording/Mixing Books and with the singer standing with their back to a foam–covered wall such as this, the
result is adequately dry and much less tonally coloured than you’d get from a typical
Recording Techniques Join
in today's discussions:
small vocal booth, which is what Dan was thinking of building until I talked him out of
» advice needed for
it! In fact he had a second–hand UPVC front door and frame in the studio when I
DAW Tips from SOS
recording interviews on arrived that he’d earmarked for that very purpose, so we had to lug that out into the
location corridor. Some decoupling speaker stands were 100s of great articles!
crafted out of leftover acoustic foam, Cubase
» Vintage mic's... what ceramic tiles and some non–slip matting.
can they contribute I don’t like putting monitors directly onto the shelf section of studio furniture, as Digital Performer
musically? vibrations can be transmitted into the desk, blurring the sound. In this instance I Live
» S/PDIF connection of a fabricated a couple of impromptu speaker platforms using foam offcuts glued to the bottom of a couple of spare ceramic floor Logic
16 bit sampler to an tiles. A sheet of non–slip matting was then used between each speaker and tile to keep them stable. This setup left us with the Pro Tools
Audio Interface (24 bit) speakers at just the right height, and we angled them in to aim just behind the mixing seat. The Scarlett interface and USB Reaper
» Sennheiser MKH 110 MIDI keyboard both fitted on the desktop, leaving plenty of free space for mousing around. Once all the foam was in place the Reason
&110-1 Schematic sound from the monitors was clear and exhibited stable stereo imaging. You don’t get seriously deep low end from small Sonar
monitors like the M–Audio ones Dan had, but as long as any important mixes are double-checked on headphones that
SOS Mix Rescue articles
shouldn’t be a problem.
Install Wizard
After downloading the Focusrite support software and free
bundled goodies, which rattled down at a good speed thanks to
the Devolo Wi–Fi repeaters, I checked the setup using
GarageBand set to 24–bit/44.1kHz. I made some test audio
recordings via the SE microphone and also by recording some
MIDI parts using the available software instruments. No drivers
were needed for the USB keyboard — it really was a simple case
of plug and play. Everything worked first time, and the audio
quality from the Scarlett 18i8 on playback was perfectly clean,
just as I would have expected. Next I suggested that Dan avail
himself of any useful free plug–ins, as many manufacturers
make available simplified versions of their main products as a
taster and to expand their mailing lists. The FXpansion
compressor, for example, makes a really nice alternative to the Dan Garbett getting to grips with GarageBand in his newly
GarageBand compressor, so I suggested he start there. reconfigured studio.
As Dan wasn’t familiar with GarageBand, I showed him the basics, which was itself interesting as I have a later version than
he does, so not everything works in quite the same way! However, it did confirm that the program would meet the needs of the
majority of users, with the light version of Ableton’s Live being well suited to the more urban–inclined visitors.
After having a bit of a tidy up, I left Dan to explore his new kit before handing it over to the young musicians who would be
using it. .
Reader Reaction
Dan Garbett: “I am really grateful for all that Sound On Sound have put into this project, and for the resources given by the
companies who have supported our worthwhile cause. The Comfortex acoustic treatment has made a big difference and
now we have the Focusrite Scarlett installed we can record bands as well as individual musicians. The young people who
usethe studio should be really pleased with the results, and I look forward to seeing their reactions just from entering the
room! They will get even more excited now that the equipment has been set up in the right way. The Sound On Sound visit
has shown me that a few alterations and advice on equipment placement really can add up to an improved quality of
recording.”

In this article:
Beat Happy
The Beat Is On
I Like The Way You Logic Tips & Techniques Buy PDF
Groove Published in SOS October 2014

Technique : Logic Notes
Nine more ways to kick off your productions in Logic Pro X.
Dot Bustelo
H
ere’s the follow-up to last month’s feature on quick techniques for making beats in Logic. Let’s go!
Gating Effects with Slicing Flex Mode: Adjusting the parameters of the Slicing Flex Mode on a beat can quickly create
a new section of your track. This Flex Mode cuts the audio and moves the slices around without any time-stretching. Try this:
Throw two audio beats into a group to make adjustments together.

Enable ‘Flex’ and select the ‘Slicing’ mode in the track header.
Experiment with Slice Length in the Instrument Parameter box. This parameter shortens each slice by a percentage value,
which can eliminate pre-attack sounds from the following slice, or create a gated effect.
Try the Slice Length at 50 percent or 75 percent for a bouncy, pulsing beat.
Making an EXS instrument out of a beat: Turn any beat into an EXS instrument in
one step for drum programming. Take any audio file of a beat with a kick, snare
and hi-hat. In the Track menu, select ‘Convert Regions to New Sampler Track’
then ‘Create Zones from Transient Markers’ in the dialogue. Logic creates an EXS
instrument on the track below with the individual drum voices assigned to different
zones.
However, the resulting kit may be sloppy. Logic turns the areas between each
detected transient into EXS24 zones and there’s no musical intelligence to the
selection process. A tiny noise in the audio may get assigned to a zone, or maybe
the kick was hit twice in a row in the beat. In the latter case, both hits become
zones so the same drum sound is assigned to two consecutive notes.
Adjust Slice Length on Slicing Flex Mode
This is how to clean up the transient detection before making the EXS to create a gated effect on a beat.
instrument:
Select your audio file then open the File Editor.
Select Audio File / Detect Transients.
Click the +/- buttons to adjust the number of detected transients. The minus button thins out how many are detected, which
solves a lot of the problems. The button labelled ‘Transient Editing Mode’ enables the transient view.
To preview between transients, double-click the area then select the speaker icon labelled ‘Prelisten’, or change the
Command-click tool to ‘Solo’ in the File Editor’s tool menu and highlight between two transients while holding the
Command key.
Double-click to remove unwanted detected transients, drag directly on a transient to change its position, or use the Pencil
tool to draw new transients.
After tidying up, select ‘Convert Regions to New Sampler Track’ from the Track menu. You should now have a useful drum
kit for additional programming.
EXS instruments from regions and riffs: Here’s a great remix

technique with a longer sample, maybe even a whole song.
Place an MP3 or WAV of a song on an audio track.

Add the BPM Counter plug-in from the Metering subfolder to
find the tempo of the track. Adjust Logic’s tempo to it so the
audio file is on the beat grid when you enable the metronome.
For this exercise, cut the track at bar nine to create an eight-
bar region.
Move the playhead to bar two and select Option-Scissors to
make eight equal one-bar regions. Now you’re ready to create
the EXS instrument.
This time, in the Track menu after selecting ‘Convert Regions
to New Sampler Track’, choose ‘Create Zones from Regions’
in the dialogue. Clean up Logic’s transient detection before creating an EXS
instrument from a beat.
The EXS instrument is created on the track below. Hold down
different keys to trigger the riffs you’ve isolated.
Beat Happy
Varispeed for fast tempo changes: Quickly experiment with the tempo of your beat with Varispeed. You can’t bounce a project
with Varispeed and this feature is taxing on the CPU, but it’s a great way to try out different tempos.
Open a project containing audio and MIDI files.

In the control bar, enable the Varispeed button (with the plus/minus symbol). If you don’t see Varispeed in the control bar,
click-hold in the control bar to enable Customize Control Bar and Display. In the dialogue’s LCD column, switch the pop-up
menu to Custom and enable Varispeed.
Click-hold the field labelled ‘Percentage’ in the LCD display to see the desired tempo.
Change the Varispeed tempo and Logic’s tempo will update. You can adjust between 50 percent to 200 percent of the
original tempo.
Toggle Varispeed on and off to listen back at the original tempo.
Project Tempo change with Flexing: If you simply change Logic’s

tempo, the audio in your project will be out of time. Flex all your
audio tracks first, then easily experiment with the tempo of your
beat until it feels ‘in the pocket’. Adjusting Varispeed view to tempo rather than percentage.
In the Mixer, click hold and drag across all audio track faders to highlight them, then place them in a group by click-holding
in the black pane on any track’s channel strip.
In the Group settings menu, make sure ‘Editing’ is ticked.
Enable Flex Mode in the Toolbar.
Select any Flex Mode on an individual track, which will be applied to all tracks since they’re grouped.
Adjust Logic’s tempo and all tracks will play back in time and can be bounced or exported at whichever tempo works best
for the beat.
Making Groove Templates: The Groove Template is one of the main reasons early Logic users became such fans. You can
create a Groove Template using any audio or MIDI performance file, and then quantise other tracks to it for the same feel.
Create a project with some MIDI tracks such as drums, keys and bass. Play a short, groovy piano riff.
Highlight the MIDI region of the piano riff. In the Region Parameter box select ‘Make Groove Template’ at the bottom of the DAW Tips from SOS
Quantize menu.
You’ll see a quantise setting in the menu with the region name. You can then quantise any other MIDI or flexed audio tracks
Cubase
to its exact feel. Digital Performer
To change the name in the Quantize list, control-click on the region and scroll to ‘Name and Color / Rename Regions’. This Live
works even after you’ve created the groove template. The name will update. Logic
Pro Tools
You can also make Groove Templates from audio files if you Flex the audio first. Once you’ve done this, the same ‘Make
Reaper
Groove Template’ command is available in the Quantize menu. These templates live as MIDI files within a song. There is
Reason
unfortunately no folder on your hard drive where you can point to your Groove Templates. What you can do is save the song
Sonar
containing the Groove Templates as a Song Template (File menu / Save as Template). Pack the MIDI files into a folder and
mute and hide them so you don’t have a messy empty project. Any time you start a new track from that project template, you’ll
have those Groove Templates available to you.
I Like The Way You Groove

The Groove Track: The Groove Track allows tracks to follow the timing nuances of an entire performance. Even a Drummer
performance with all its fills can be used. This lends itself well to a live band, but also provides amazing groove possibilities for
electronic producers.
Control-click on any track header.

Click Configure Track Header and enable Groove Track.
On the left side of the track header of any track, hover your mouse over the Track Number.
Click the star that appears in a narrow rectangle. This will become your Groove Track.
Select which tracks in the project match the timing by ticking the ‘Match Groove Track’ tickbox, which is also in that narrow
rectangle now visible on the track header of the other tracks.
Give Logic a minute to analyse and apply the Groove Track to any matched tracks.
Keep in mind there can only be one Groove Track per project, and any other quantisation using the Region Parameter box or
Piano Roll Editor are unavailable for any track matched to the Groove Track.Logic04
Fatten a beat with drum doubling: This is good for live drum performances that have a great feel but need help sonically.
Highlight a kick drum audio track in the Main Window.

Select ‘Replace or Double Drum Track’ from the Track menu.
Logic creates an EXS24 sampler instrument beneath, detects the transients of the drum hits and creates a MIDI
performance on the EXS track.
The Library opens, defaulting to Single Drums, specifically a bunch of kicks. In the dialogue, you can switch to another type
of ‘single’ drum or select Replace, rather than Doubling.
Preview the different kick sounds then make a selection.
Experiment with the Threshold to control how sensitive Logic is when determining which hits to create note triggers for. The
setting depends on the relative level of the recorded drum and how well it is isolated from other signal sources.
Advanced quantisation: Logic’s Region Parameter box is one of the best collections of quantise tools for subtle fine-tuning of
your groove. Here are a few favourites:
Swing letters A-F are Logic’s classic swing options taken from the Akai MPC60 drum machine. The Delay parameter (under
the heading, ‘More’) is like telling someone in your band, “Love what you’re doing, but can you play a little behind the beat and
a little sloppier?” This delay can be as subtle as a tick. Also, try doubling a keyboard part and delaying the copy by a larger
value like a 1/16th or 1/8th note.
Q-Strength softens the quantising so it’s less machine-like. At the default of 100 percent, which is displayed as a blank box,
notes are moved to exactly the correct timing. With a 50 percent setting, recorded events are moved half way to the correct
position. Try a Q-strength of 90 percent for a subtle, laid-back feel. .

In this article:
New Tricks
The Eightest & Greatest
Hi–fi Lo–end Reason Tips & Techniques Buy PDF

Technique : Reason Notes
We look at what’s new in the eighth instalment of Reason.
Robin Bigwood
I
n a somewhat unexpected move, Propellerhead have announced the forthcoming release, on September 30th, of Reason
8. Following the pattern of the last few years both the full version and its cut–down Essentials offshoot are being updated,
and anyone who bought Reason (or an upgrade) after July 1st 2014 will enjoy a grace–period upgrade to this latest
version.
New Tricks
So what’s new in version eight? What seasoned Reasoners will notice straight away is the new user interface design. There’s
a whole new browser pane that opens on the left side of the main window, fulfilling the duties of the present Patch Browser
that has previously always appeared as a dialogue box. This kind of software design (seen in Ableton Live, PreSonus Studio
One and elsewhere) allows for extensive drag-and-drop functionality, and that’s exactly what Propellerhead are promising. It
seems it’ll be possible to drag patches, samples, instruments and effects into the sequencer, the rack, and on to individual
devices. If any devices needs to be instantiated as a result Reason 8 will do all the donkey work.
The Reason 8 browser, sequencer and transport panel all take

design cues from operating systems like Apple’s forthcoming
Yosemite, featuring a more subdued ‘flat’ look. Buttons become
unashamedly 2D, all cool grey and white, and generally the
photorealistic light reflections and relief shadows are being done
away with. But that’s in stark contrast to the mixer and rack,
which remain the same as in Reason 6 and 7, retaining all their
wood–grained, shadow–strewn, 3D–rendered skeuomorphism. A
bit of an odd mix, but no more so than in some other DAWs.
Next, there are two new bundled rack devices from Swedish
developers Softube, offering amp and speaker modelling for
guitar and bass. They look promising, but the slightly
disconcerting flip side is that Propellerhead are also giving notice Reason 8, coming this September, gets an updated interface
on the existing Line 6 Guitar Amp and Bass Amp devices. They’ll and new amp simulators from Softube.
“no longer be available in the Reason rack by October 2016”,
and users are encouraged to Bounce tracks in any songs that utilise them before that time. You’d no longer be able to tweak
your tones, but at least you wouldn’t lose them forever.
Some users are questioning whether design tweaks and two guitar–oriented devices (which will be of most interest to a
subset of the user–base) are really worthy of a whole–point version number change and a $129 upgrade fee. Certainly some
previous releases have included a much longer list of new goodies and enhancements. But there could be additional features
afoot that we don’t know about yet, so if you’re itching to be at the cutting edge you’ll want to keep your eye on
www.propellerheads.se. And rest assured we’ll cover everything in detail here in SOS as soon as we can!
Hi–fi Lo–end
Some great new third–party Rack Extensions have recently come into the Propellerhead shop which explore vintage,
simplistic and lo–fi sound from a few different angles.
Jiggery–Pokery Sound have released two electronic organ instruments simultaneously. If you like vibey transistor tones of
the ’60s and ’70s — and who doesn’t? — they’ll both be mandatory purchases.
Combo Compact Organ ($35) is an emulation of a Farfisa

Combo Compact, used by Pink Floyd and many others. It’s
actually several organs in one; you choose whether to use the
named stops (like ‘Flute’ and ‘Strings’) or, by turning Multi Boost
on, the generic flute stops labelled with footages (’16’, ‘8’, etc.).
There’s also a bass section, triggered from pitches C0 to B2,
amplifier drive, and a fantastically evocative mono spring reverb. Jiggery–Pokery’s new transistor organ REs are updated,
more flexible and much easier to use versions of their
The reverb also makes it into the Combo Continental ($35), existing ReFill instruments.
Jiggery–Pokery’s Vox Continental RE, even though the original
didn’t have one. But in other ways this is a faithful and great-sounding digital emulation that immediately recalls the Doors, the
Monkees and other bands of decades past. The proper drawbar behaviour has been copied, so that no sound emerges until
you draw either the sine (~) or reed (M) ‘Foundations’. There are upper harmonic combinations too, labelled with Roman
numerals, alongside the more familiar numerical footages.
If you like your vintage more ’80s than ’60s though, try Chip 64
Synth ($39) from Ochen K. This little gem generates the tones of
five ‘80s and ’90s computer and games console sound chips: the
TIA from the Atari 2600, the SID from the Commodore 64, the
TA0x series found in Nintendo’s NES and Gameboy, the Vic–I
from the Commodore Vic–20, and the snappily named SN76489
used in ColecoVision and Sega consoles. Each gets its own
complement of waveforms, and there’s a simple synth
architecture with multi-mode filter to shape those
characteristically raw sounds from the oscillator. It’s a sound source unlike any other in Reason, full of character and interest,
and should be useful in many musical genres.
The same goes for another Ochen K creation, the EDS06s

drum synth module ($25). One instance generates up to six
different drum sounds, all of which are synth–based rather than
having anything to do with samples. The six identical drum
‘channels’ can respond to any MIDI note you choose (so it’s easy
to layer several to respond to the same note), offer the same 49
waveforms and streamlined synth architecture, plus there’s
distortion, reverb and delay, and you can choose the order in
Ochen K does nothing to harm his reputation as one of the
which the post–oscillator elements are connected using the most exciting and inspiring RE developers with these unique DAW Tips from SOS
Routing parameter. What’s more, each channel has a rear-panel and highly usable REs.
Aux Send/Return loop, so it’s easy to patch other effects devices
Cubase
into the signal flow. You could probably tease out some vintage drum machine tones from EDS06s, but I don’t think this is the
Digital Performer
real point of it. It’s the drum synth you choose when you want something different, experimental or novel, not just another
Live
TR808. Bear in mind, too, that it’s really easy to layer individual channels on top of existing Reason drum sound sources. Just
Logic
connect one of the Gate Out connections on ReDrum or Kong to an EDS06s Gate In. Then playing that corresponding
Pro Tools
channel on one of the Propellerhead devices will trigger the EDS06s.
Reaper
To round off this little survey of old–school goodness, here’s a Reason
quick plug of another Jiggery–Pokery product, this time a Sonar
freeware ReFill you can download from www.jiggery-
pokery.com/free.html — 8–bit Magic: ZX Spectrum provides yet
another option for naüve but funky home computer sounds from
decades past. Despite the name, there are tones here from both
the 1980s Sinclair ZX Spectrum and the Commodore 64. The
charmingly simplistic sounds are faithfully captured, and
presented as patches for ReDrum, NNXT and Combinator. And,
if anything, the notable lack of multisampling in the NNXT patches adds yet more character, suggesting a weird but interesting
mash–up between those pioneering early computers and early memory–starved hardware samplers. .

Reviews Information
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In this article:
Kiesza
The Mix Review
Rita Ora Commercial Productions Analysed Buy PDF
Duke Dumont Published in SOS October 2014

Technique : Mixing
Kaiser Chiefs
Mr Probz We examine the production of some recent hits to help you
Classic Mix Santana brush up on your listening skills.
featuring Rob
Mike Senior
Thomas‘Smooth’ (1999)
Kiesza
‘Hideaway’
It’s not that common to hear a true soprano in chart music, as a lot of divas prefer to stick to alto registers to achieve more
soulful timbres. In this case, though, Kiesza’s performance, ranging all the way up to top ‘G’, really works for me, making
something of a production hook out of the resultant ‘hollow texture’ — you get lots of bass and treble notes, but very little
happening in the middle registers. And by virtue of this textural hole you can easily hear some great examples of filtered noise
being used as a transition effect. Notice in particular how extended each of the filter sweeps is: the rise at 0:53 lasts at least
eight seconds, while the falls at 1:02 and 1:18 are almost twice that length.
If you’ve not experimented with this kind of thing before, it’s

ridiculously easy to do with almost any modern soft synth. Just
use a single oscillator with white noise selected as its waveform,
route that through a filter set to band–pass mode, play a MIDI
note to trigger the noise, and then sweep the filter’s cutoff
frequency. Depending on the slope and resonance of the filter
you use, you’ll get a different noise tone — to emulate the flavour
used in this song I’d suggest starting with a 24dB/octave type,
and only a touch of resonance to keep the emphasis peak from
‘whistling’ too ostentatiously. Mike Senior
Rita Ora
‘I Will Never Let You Down’

The odd distortion sound on the guitar intro of this production
sounds to me like aliasing, so if you like the sound of it then
some kind of ‘lo–fi’ or ‘bit–crusher’ kind of plug–in would be the
way to go. In this case I think it’s very effective at indicating from
the outset that we’re hearing some kind of electro style, despite
the distinctly Red Hot Chili Peppers guitar line/tone.
The quirky little rhythmic hook in the chorus backing also

caught my ear: notice how the eighth–note off–beats are absent
from the first half of the second bar of each four–bar unit (you
can hear this for the first time at 1:02). It does seem a bit of a
shame, though, that the electric guitar part seems to push the
beat there, as this unsettles the groove and weakens the musical
effect of those moments for me. Mike Senior
Duke Dumont
‘I Got U’
There are lots of nice little touches in this production — the steel
drums first heard at 0:16, the little party–vibe whoops on the left–
hand side of the stereo image that first make their appearance at
0:32, and what sounds like a reverse–envelope reverb on the
lead vocal at 1:04, just to mention a few of them. My favourite
element of this production, though, is the piano that enters at
0:48. Tinny piano sounds are something of a house cliché, of
course, but I was surprised that its entry had something of a
warming effect, despite what sounds like the usual astringent
high–pass filtering. Closer listening (particularly to the Sides
signal) suggests that the warmth is actually coming from some kind of gently percussive keyboard pad that’s layered stealthily
behind the LF–shy piano timbre. I wonder whether we hear the layer on its own later at 1:04–1:35, in fact. Because our ears
tend to identify instruments primarily according to their attack characteristics and the balance of their upper–spectrum
harmonics, layering a dull–sounding synth underneath something as percussive and bright as a house–style piano pretty
much guarantees that the former will remain hidden by the latter
in the listener’s consciousness. (The same principle underlies
the common rock–music trick of underpinning a bass guitar with
synthetic sub–bass — as long as the synth is dull, the character
of the live instrument’s timbre subsumes it, so that all you hear is
a single bass guitar with better low–end oomph.) Mike Senior
Kaiser Chiefs
‘Coming Home’
This mix features a textbook example of slapback echo on the
lead vocals during the verses (0:21 and 1:35) and middle section
(3:01). To my ears it sounds like a single–tap echo with a delay
time of approximately 60ms. As I understand it, this mix comes
from Michael Brauer, a big fan of classic tape echo units, so I
wouldn’t be at all surprised if that’s what we’re hearing in this
case — certainly there’s that kind of smoothness to the delay tap
I’d expect of an analogue unit.
The drum sound is also tremendously compressed, but at the

same time the hi–hat isn’t at all overbearing, as it normally DAW Tips from SOS
becomes if you’re working with completely live drum recordings, no matter how busy you get gating/editing spill. As a result, 100s of great articles!
I’d wager there’s at least a snare–drum sample supplementing the live tracks here, because I find it hard to believe that you Cubase
could get such a compressed snare so upfront in the mix otherwise. There could also be a kick–drum sample in there too, but Digital Performer
it’s always trickier to tell there, because it’s usually fairly straightforward to isolate the kick–drum sound of any live kit recording Live
during mixdown by using a combination of gating on the kick channel and high–pass filtering on all the other mics — Logic
especially if an ‘inside–the–kick’ close–mic placement has been used. (It’s also interesting to note that there don’t seem to be Pro Tools
any toms used in this track at all, and the heavy snare timbre makes sense to me within this context, because it helps retain Reaper
more power in the end–of–section fills.) Reason
Sonar
Finally, here’s a little poser for you: what’s the purpose of the
level hike that appears to be applied to the whole mix at 1:27?
It’s a boost of around 0.7dB, I think, and while that might seem
rather small, it’s nonetheless clearly audible. While the idea of
riding the overall level of the mix with your master fader is by no
means new or unusual (Andy Wallace mentioned it in his recent
interview in SOS July 2014, for instance), such tactics are
typically reserved for subliminally supporting section changes —
the hoariest old chestnut being to boost the level a decibel or two
whenever a chorus hits. But what purpose does it serve in this
case, where it comes in such an odd place musically, namely the
sixth eighth-note of the first bar following the chorus? I suppose
you could argue that its very unusualness helps maintain the
attention of the listener. But it’s probably just a goof... Mike
Senior
Mr Probz
‘Waves’
The lead vocal sound here is rather striking, with a kind of
rasping digital edge that constitutes a good contrast against the
mellowness of the rest of the mix. While it’s impossible to reverse engineer exactly what effects were used to achieve this
without direct information from the artist, here’s what I’d suggest for anyone who wants to recreate that kind of timbre in their
own work: try ring–modulation. In my own experiments with Melda Productions’ freeware MRingModulator I got best results
using comparatively low modulation frequencies (around 50 to 70 Hz) and modulation depths in the 20 to 30 percent range.
Setting up the ring–modulator as a send effect also helped a great deal, because I could use fairly severe high–pass filtering
(24dB/octave around 700Hz) to scotch unwanted low–frequency side–band pitches and also tame the rather extreme HF
aggression of the effect with low–pass filtering.
While we’re on the subject of vocal effects, there’s a nice

example of the ‘outside the speakers’ widening effect on the
“drifting away” backing vocals first heard at 0:41. In other words,
the stereo backing vocals submix has been processed such that
the Sides component of the signal is considerably stronger than
the Middle. This gives the illusion that the voices are located
slightly outside the normal boundaries of the stereo panorama,
something beyond the capabilities of traditional hard–panning.
The effect is a nice one in stereo, but the downside is that it
inevitably leads to mono–compatibility problems for any such
widened parts, as you can hear if you toggle your DAW’s or
monitor controller’s Mono button — the apparent level of that line
in the mix drops dramatically in mono. Fortunately, Mr Probz has
sensibly chosen to restrict its use to a fairly incidental part, so no
probs there. Sorry, couldn’t resist... Mike Senior .
Classic Mix Santana featuring Rob Thomas ‘Smooth’ (1999)

There are a lot of thought–provoking mix decisions in this track, first and foremost Carlos Santana’s lead guitar sound,
which trades heavily on its 3 to 500 Hz region to anchor itself in the mix balance. This is a little unusual in my experience,
because that spectral region is usually fought over tooth and nail in most mixes — the subjective warmth of many
instruments resides in that zone, and mix clutter also normally builds up most quickly there. The nature of this guitar sound
therefore means that the brass and piano have had to be pretty severely thinned out to prevent overall muddiness
developing in the mix as whole, and the bass also relies on its fundamental and an equally strong first harmonic so as to
leave space for the guitar above 300Hz.
The additional difficulty here, though, is that the lead guitar doesn’t play all the time, so how does the mix compensate
for this loss of low mid-range? Well, the lead vocal helps in this respect during the choruses in particular, where its
reasonably full tone also helps make it stand out nicely in the balance. In the verses, though, the telephone–style distortion
effect thins out the singer’s timbre, and it seems to me that the piano is the main part to step into the breach — you can
hear it being ridden up in the balance considerably at those points, as well as being ridden down again whenever the
guitar makes an occasional interjection. The brass arrangement also seems to help a little too, by shifting to a lower
register there.
The second verse has a rare treat for headphone listeners: a clearly distinguishable whispered-vocal double–track in the
right channel only, something that’s much more dramatic when you hear it over cans. What makes it even better, though,
is that it’s disconcertingly sporadic, only appearing on some lines, so it keeps taking you by surprise. And then, just when
you’re beginning to get used to it, it suddenly swaps sides to emphasise the line “you hear my rhythm on your radio” —
rather fitting really, given the filtered vocal timbre. And the next swap, this time a left–right–left sequence, is even more
apt, appearing as it does for the lyric “turning you round and round”. Mike Senior

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In this article:
Event FX
Unfolding Events
Multiple Events Studio One Tips & Techniques Buy PDF
Keeping It Real Published in SOS October 2014

Technique : Studio One Notes
Workarounds &
Leftovers Studio One lets you apply effects directly to audio Events,
In Any Event
opening up innumerable creative possibilities.
Larry the O
T
here are many situations in which it is useful to process audio on a clip–by–clip — or, in Studio One–speak, Event-by-
Event — basis. For example, punching in a vocal or instrument on a different day from the original recording can result
in tonal differences between the original and the overdub, even if everything is done carefully. Should different
equipment be used on the two days, tonal differences are even greater. Fortunately, Studio One offers a particularly strong set
of Event–based tools that can help in this situation, and many others.
Event FX
Let’s just go straight for the good stuff. It is common for DAWs to have a way to render a
plug–in on a single clip, but Studio One takes things to another level altogether with Event
FX. This feature allows you to put an entire effects chain on a single Event, play through it
in real time, and edit as needed before committing to rendering the audio — which, at any
time, can be reverted back to the live, real–time effects with a single click. Pretty darn cool,
say I.
We all know how addictive plug–ins can be, and how easy it is to take a powerful
computer to its limits by throwing them around like candy. Now think about how much
faster you can bring your machine to its knees instantiating effects on a per–Event basis!
Sarcasm aside, everything has its limits, and you’ll hit them pretty quickly if you start using
a lot of real–time Event FX on a number of Events.
The Event FX area, populated
The Event FX feature is really easy to use; but before you can use it, you have to see it, with a few plug–ins.
which means making the Inspector panel visible. Select any Event, and the Event
inspector shows up in the Inspector panel, with the Event FX section immediately below the Event name. Click the disclosure
triangle at the left, and an insert effects rack drops down. This rack works just like mixer channel insert effects racks: you can
add a new effect by dragging it from the Browser and dropping it, or click the plus sign at the top and choose an effect from
the drop–down menu. Drag effects to reorder them, drag a complete FX Chain from the Browser, all the same stuff as usual.
When your beautiful effects chain is tweaked to perfection, simply click the Render button to generate a new file with the
effects. The insert rack disappears and the Render button now says ‘Restore’. As you’ve likely surmised, clicking the Restore
button is how you make the effects live again.
There is also a Tail field, so that additional time can be added on the end of the render to
accommodate delay and reverb tails. And, as with insert racks in mixer channels, the
whole chain can be dragged to the Browser to create a new FX Chain to use on other
Events or tracks. Best of all, once effects are rendered, they are unloaded, freeing up the Once Event FX have been
rendered, the effects rack
power and resources they consume. Parameters of Event FX can be manipulated in real disappears and the Render
time by a controller through Control Link, just as can channel effects. Obviously, this only button becomes the Restore
works when Event FX are running live; once they’ve been rendered, Studio One is simply button.
playing back a file with effects baked in.
Multiple Events
Event FX have numerous applications, both creative and corrective. The first thing that
leaps to mind is putting effects on a single word or phrase for emphasis. Maybe you need
a specific EQ setting to match the sound of an overdubbed phrase to the rest of the take,
as previously suggested, or a reverb to match the ambience on a few touch–up fixes to a The Tail field adds time to the
Event, to capture reverb and
live recording. delay tails when you render
Event FX. Note the Process
First, create an effects setting that accomplishes what you want and save it as a preset Volume after the Event FX button
or FX Chain. Now select each overdubbed Event in turn, drop the same effects in the to the left of the Tail legend.
Event FX area, load the preset you made, and render. In this way, Event FX can provide
an alternative to automating effects. You save work in your mix and reduce the number of plug–ins needed for it.
As a final consideration, I pose this question: if you specify an Event volume, does it get computed before or after the
effects chain? The results often can be quite different between the two. In most cases, the best results are achieved by
computing volume after effects, but Studio One gives you a choice, with a button bearing the clever name Process Volume
after Event FX, visible to the left of the Tail legend.
Keeping It Real
Event FX is clearly a superstar feature, but it is not the only Event–level mixing function in Studio One. There are a number of
other interesting features, mostly providing real–time functions.
Speedup is a useful little box I mentioned a few months back when I wrote about time–stretching in Studio One. Speedup
is a means of providing time compression or expansion for an Event without affecting either the source file or any other
Events made from it. Consequently, it enables you to stretch or compress a single Event taken from a file, where time–
stretching applied to follow tempo changes affects all Events based on a file. Rendered time–stretching is also available,
but Speedup being real–time means you can, for instance, play with the length of the
last word of a line until you find a length you like. That’s a painful trial–and–error
process otherwise. Speedup is best thought of as a percentage, though it is displayed
like a multiplier (the default value is 1.00, instead of 100 percent). It has a pretty wide
range: from one–quarter speed up to four-times speed.
Transpose and Tune are quite straightforward and, again, can be used for corrective or
creative applications. They offer an alternative to using Melodyne to tweak the pitch of
an Event.
Normalize is a tick box that is particularly handy when working with very low levels in a
file. The Normalize box only does peak normalisation to 0dBFS; it is hard to do RMS
normalisation in real time because it requires computing an average value and so must
wait until enough data is accumulated to calculate an average, introducing delay. It
would be nice to be able to set the normalisation level, but it’s very useful as it stands.
Gain is really straightforward, simply adjusting the overall gain of the Event.
Fade–In and Fade–Out are equally straightforward, but even more powerful, since they
can work in a batch fashion. After selecting some Events, entering a Fade–In or Fade– The Event Inspector contains a
Out value adds the specified fades to all selected Events. Top and tail, anyone? number of real–time Event–level
processes.
Workarounds & Leftovers
Naturally, Event FX have their limitations. Even though you can control live Event FX using Control Link, it is not possible to
automate Event FX. However, like a few other DAWs, Studio One can carry automation along when an Event is moved, DAW Tips from SOS
allowing you to automate channel effects and have the automation be attached to the Event. The ‘Automation follows events’ 100s of great articles!
tick box in Preferences / Options / Advanced /Automation enables this behaviour. Cubase
Digital Performer
However, in an effects–automation scenario, Event FX is a way to audition effects on an Event without having to do bypass
Live
automation in the track, as well as to audition control using Control Link that you intend to automate. Here’s a workaround for
Logic
automating Event FX:
Pro Tools
Create the effect you want. Once you have it, save it as a preset or FX Chain. Reaper
Disable or remove the Event FX. Reason
Instantiate the same effects as channel insert effects and recall the preset you stored, or drop the FX Chain on the Sonar
channel’s insert section.
Automate as desired. As long as the automation occurs within the Event bounds, you should be able to move the Event
around on the track and have the effects stay with it.
I’ll admit, it’s a little clunky, but there you are. One last note: in spite of appearing in the Event inspector, Audio Bend markers
are applied to an entire file, not just an Event. Strip Silence, on the other hand, can be applied at the Event level.
In Any Event
Event FX is a wonderful feature that can often obviate the need to automate effects. But Studio One offers a full palette of
Event–level processing features beyond Event FX, and those should not be forgotten. Event–level processing gives you the
same fine control over processing that Event fades and volume give you over gain at the Event level. You may quickly find it
addictive! .

Reviews Information
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In this article:
Outside Assistance
VCA OK
Basic Setup Cubase Tips & Techniques Buy PDF
Overcoming Obstacles Published in SOS October 2014

Technique : Cubase Notes
Matt Houghton
I
’ve long wished Steinberg would implement true VCA–style automation in Cubase — it is, for
me, a critical omission, and it’s one that’s nearly led me to jump ship a couple of times. Sure, Cubase’s grouping, routing
and track–linking facilities are better than they ever have been, but it just doesn’t offer anything quite like automatable VCA
master faders. What do VCA masters offer that standard automation doesn’t? For one, you can ‘group’ any channels together,
even if they’re routed to different outputs. For another, each channel can be a member of multiple VCA groups at the same
time which, again, isn’t possible, at least not without implausible aux–send setups to multiple groups.
Outside Assistance
Help is at hand, though, in the form of a freeware plug–in by Blue Cat called BC Gain. The plug–in comes in three versions,
bundled together in their Gain Suite (www.bluecataudio.com/Products/Product_GainSuite), and it enables you, with a little
lateral thinking, to add VCA groups and master faders to pretty much any DAW that lacks them. Unlike most ‘gainer’ plug–ins,
different instances of BC Gain can be assigned to one (or none) of nine groups, and each instance can be controlled via MIDI.
When you change the gain on one instance of the plug–in in a group, all other instances move accordingly. It doesn’t take a
great leap of the imagination to realise that you could place instances in any insert slot you wish, assign them to a group, and
then create a dedicated MIDI track whose fader may be used to control the group level.
Basic Setup
Although simple in concept, my experiments suggested that there are good ways and less good ways to implement this
system in Cubase. Let me take you, step by step, through the way I find works best.
1. Open a new Cubase project and create a handful of audio tracks. In the example, I’ve made four stereo and four
mono tracks.
2. Create one more track. It can be an audio, Group or FX channel. It doesn’t really matter which, but you may find
that it’s more easily identifiable in the mixer if you make it a Group or FX Channel, as these will appear to the right of
your mixer by default. Call this channel ‘VCA Master’.
3. Now, create a MIDI channel and label it ‘VCA Group A’. This channel’s fader will become your VCA Group A
master fader.
We now have all the channels we need to test this out, so let’s put some sound through that first group of audio
channels. You could import or record some audio, of course, but you can alternatively use Cubase’s Test Generator
plug–in, just to get the level meters showing something. Now to wire all this mess together so it works as a VCA
system...
1. Insert an instance of the BC Gain plug–in in the first insert slot of your VCA Master channel. Whether the channel
you created is mono or stereo doesn’t matter, but be sure to choose the stereo version of BC Gain (there are also
mono and dual–mono versions), as I was unable to link mono and stereo plug–ins in a single group successfully.
2. Click both the cog (settings) and chain (link) icons at the top left of the plug–in to reveal some hidden features.
Click the pointer that just appeared under the gain knob and select ‘settings’ to bring up the MIDI control dialogue.
Make sure MIDI Enabled is ticked, and click the MIDI Learn button. In the Group section of the plug–in select ‘A’. Use
a
3. On your MIDI channel, VCA Group A, set the output destination to the gain plug–in on your VCA Master channel. ‘spare’
audio,
It should be easy to identify in the drop–down list. Waggle the fader on your VCA Group A MIDI channel and the FX or
plug–in’s gain control should respond. If the Learn function isn’t working properly, you might need to set the plug–in Group
to receive on the correct MIDI channel, which is listed on the MIDI channel in the Routing section of the Cubase channel
to
Mixer. host
your
4. Insert another BC Gain stereo plug–in instance on your next audio channel. Click the link button and assign the ‘VCA
plug–in to Group A. There’s no need to do any MIDI assigning this time. Just waggle your MIDI fader again, and you Master’
instances
should see the control of both instances reacting to your fader movements. of
BC
Congratulations: you just used a VCA Master channel to control the level of two separate channels in Cubase! But Gain
we can take things a little further... —
but
be
1. In the MixConsole, Alt–drag and drop the second plug–in instance to copy it to any other channels you want to be sure
part of the same VCA group. All of the plug-in’s settings will be copied, including the Group assignment. In this way to
you can quickly assign several channels to a single VCA group. choose
the
stereo
2. Now create a second VCA Master fader in exactly the same way —ie. create a new MIDI channel and call it VCA version
Group B. of
the
3. Put another instance of BC Gain in the VCA Master channel’s second insert slot, set it up to respond to your plug–
in.
second MIDI channel fader, and assign the plug–in to Group B.
4. Then insert yet another instance of one of your audio channels — it can even be in another slot on the same channel as
before. Change the routing to group B and Alt–drag to create new instances on other channels. All these instances will now
respond to your second MIDI Fader.
It’s much less hassle the second time around! Note that by placing multiple instances on a channel, that channel’s level can
be influenced by more than one VCA Master Fader at a time. So
far, so good, then — it works. But there are a few issues we
need to overcome.
Overcoming Obstacles
First, although a single channel can be assigned to multiple VCA
groups (as just described), you’re limited by the number of insert
slots on each channel, which is fixed at six pre–fader and two
post–fader. There’s no elegant way around this: if you need more
slots, you’ll either have to route the channel in question to
another channel and use the inserts on that one (you can always
hide the second channel if you want), or use a third–party plug–
in chainer such as DDMF Metaplugin.
Second, you might have noticed that the MIDI protocol’s 0–127
settings are causing a couple of issues: by default, the ‘zero’
More Cubase Notes point (ie. no gain, no attenuation) lies somewhere in between
values 63 and 64 — your plug–in’s gain control can be set to
+0.47 or –0.47 dB, but not unity. There’s no getting around the
resolution of the MIDI fader itself, but you can tweak how the BC
Gain plug–in responds. I found that restricting the maximum gain DAW Tips from SOS
to 20.2dB (leaving the minimum at –60) gave a more useful 100s of great articles!
control range that placed an almost no boost/attenuation position Link the VCA Master instances of BC Gain to a MIDI Cubase
(–0.01dB) of the MIDI channel fader at about the same place channel’s fader, either using MIDI Learn, or programming the Digital Performer
(MIDI value 95) as unity gain on the regular audio channel values by hand. Setting the plug–in’s Control Max value to Live
+20.2dB results in a more useful gain range and near–unity
faders, leaving you with around 20dB of gain and 60dB of position. Logic
attenuation available. What’s more, if you make sure you create Pro Tools
and set up all of your VCA Master instances of the BC Gain plug–in first, and only add new instances as you need them for Reaper
each channel, the grouping will be relative — so as it loads at unity by default, you’ll always be able to return to that value by Reason
placing your MIDI fader at 95. Neat! Sonar
With all these extra channels and instances, your project can
soon start to feel very cluttered. You could, if you wished,
abandon the MIDI faders and use your VCA Master channel’s
Quick Controls to govern the VCA group master levels.
Personally, I prefer to have the option of having the VCA faders
in the MixConsole. You can take advantage of Cubase’s multiple
mixers, though. Go to Devices/MixConsole 2 (you can assign a
keyboard shortcut to this). Use the mixer’s Visibility function (on
the mixer’s left, or revealed using the mixer’s Setup Window
Despite the slight (–0.01dB) offset on the Master instance,
Layout icon, top left, if hidden) to hide all channels except those the knob on the channel instance can be set precisely to
you wish to see. This doesn’t affect what you see on the main 0.00dB gain/attenuation. Assigning the channel instance on
MixConsole, but on MixConsole 2 you can see only your the same group as the master means it, and any other
instances in the group, will be controlled by your MIDI Fader.
dedicated VCA controls. In the Project page, you can hide all
these MIDI channels inside a collapsed Folder track if you wish.
And there we have it. While it may not be the most elegant
VCA system ever implemented — you can’t merge the VCA
automation with the track automation, as in Pro Tools, for
example — it’s free, it solves a genuine problem many users
have with Cubase and Nuendo, it’s both easy to configure and
reliable in use, and you can save the setup as a project template,
in which assigning a channel to a VCA group is as easy as
inserting a plug–in and clicking a letter. I’ve created a template
project with all the VCA Master side of things set up, so I just
have to insert an instance on a channel when required. As luck
would have it the ‘V’ key isn’t assigned to a shortcut by default,
so I was able to assign that to MixConsole 2, which I’d set up to
show only the VCA faders — you can find it along with a short
demonstration video on the SOS web site at
You can use the Visibility controls of Cubase’s second (or
http://sosm.ag/oct14-cubase-media, though you may need to set third) Mixer to create a dedicated VCA mixer, which you can
up your own keyboard shortcut for the mixer, as these are stored assign its own shortcut key (‘V’ for VCA, perhaps?).
in your own Cubase preferences. .

In this article:
Villains Of The Piece
Can The App Save The Album?
App, App & Away Power Apps Buy PDF
Behind The Glass Published in SOS October 2014

Music Business : Online Music
App Research
Making Tracks Smartphones and tablets are giving artists new ways to
Uncharted Waters explore multimedia. Does the ‘album app’ have what it
Appwardly Mobile takes to rekindle music sales?
Planet Of The Apps
David Greeves
I
t’s official: the album is dead. It’s a refrain we’ve all heard many times before. The death of the album has been loudly and
regularly trumpeted for more than a decade. If the album really is dying, it’s being damned slow and noisy about it.
The fact is that most record companies still plan their schedules around tent–pole album releases. More importantly, bands
still want to make albums. For the majority of artists, the long player is still the definitive statement: a group of tracks linked by
theme, time and place, presented in a set order and adding up to more than the sum of their parts. And from a practical and
financial point of view, if you’re going to gather a group of musicians together and book a studio, you might as well record
more than one song.
There’s no denying that the latest figures look grim. At the

beginning of 2014, Nielsen SoundScan announced that US
weekly album sales had hit a new low. In the week ending
January 12th, 4.49 million albums were sold, the lowest weekly
total since records began in 1991. That’s less than a tenth of the
all–time high of 45.4 million albums sold, in the week ending
December 24th, 2000. The mid–year report was no more
encouraging, with album sales down 15 percent on the previous
year — when sales were already down 8.4 percent on the year
before that. It seems that any ‘new low’ you choose to mark will
be surpassed by a new ‘new low’ before very long.
Villains Of The Piece

The suspects wanted in connection with the album’s decline will
need to form an orderly queue. Illegal file sharing was the original culprit, but the panic began to recede after iTunes had
established a credible model for paid digital downloads. But the 99p–per–song model that iTunes introduced turned out to be
even more damaging for the album, effectively turning every long–player into a singles collection to be picked apart.
Worryingly, digital downloads themselves are now in decline. Digital song sales fell 5.7 precent in 2013 — the first time a
decline has been recorded since the iTunes store opened for business — and they were down again a whopping 13 percent in
the first half of 2014. Step forward the latest threat: streaming. It seems that services like Spotify and YouTube may be
harming not just album sales but music sales full stop.
The album’s condition looks terminal, and there are those who believe we should all stop moping around the intensive care
unit and get on with our lives. People want to stream playlists, not listen to albums, they say, so let’s give them what they
want. But sales are not the only thing that the album has lost to streaming and digital downloads. Much of the wider
experience that made a great album by a great artist such an enjoyable artifact has gone, too.
Consider the good old vinyl LP. Despite decades of progress in digital technology, vinyl is still, for many, the definitive album
format. It offers high–quality, uncompressed audio, large–format artwork, credits and sleeve notes for those who have to know
who played on and produced each track, and the sense of physically possessing the music in a pleasingly tactile form — and
these things are entirely absent from our interaction with MP3s and streamed music. It’s interesting to note that vinyl is the
one area where album sales are actually growing. The numbers are small compared to the overall market, but is it possible
that, by retaining the old–fashioned virtues of the ‘album experience’, vinyl is being rediscovered by an audience not currently
being satisfied by what’s on offer?
App, App & Away

That’s where the concept of the ‘album app’ comes in. The idea is that an album presented in app format, bringing audio
together with images, text and multimedia content in a handy, non–pirate–able package, can provide a 21st Century take on
the immersive experience of old.
Several artists have already explored the idea. Though it subsequently appeared on CD and vinyl, Björk’s album Biophilia
was originally released solely as an app. More an interactive work of art than a conventional album, it invites the user to
explore the tracks in any order through graphic visualisations and game–like elements that alter the music itself. Brian Eno,
Peter Gabriel and Philip Glass have also produced intriguing music apps that encourage the user to rearrange and play with
their work. Meanwhile, Lady Gaga’s Artpop app was a typically idiosyncratic companion to the album of the same name,
among other things inviting users to create animated GIFs to share on social media. But these creations are really more app
than album. What can the format do if your priorities are the other way around?
A great example is the Islander app created for Norwegian soul singer–songwriter Bernhoft by British developers HD360.
Designed for iPhone and iPad and billed as “the first true HD music app”, it features the full album in 24–bit/48kHz, with a
built–in music player. It’s a beautifully designed thing, inviting users to swipe across a series of different elements, including
extensive background information on each song and hand–written lyric sheets from the recording sessions, complete with
crossings–out and rewrites. With the core checklist of vinyl plus–points — high–quality audio, sleeve art, liner notes, a tactile,
physical experience — well and truly ticked, the Islander app goes much, much further.
There are 10 videos to watch, including interviews and exclusive live performances, and a unique 360–degree video of a
live studio performance. The user controls the camera and can
rotate their viewpoint around the studio while the band plays,
taking in individual performances and scrutinising equipment and
mic placement. Icons appear, giving supplemental information on
the members of the band. Less exciting perhaps, but absolutely
essential for the modern–day pop star, are the merch store and
social media section, featuring embedded feeds from all of
Bernhoft’s social media channels.
Things get even more interactive in the Music Studio and Loop
Station sections. The former is an eight–channel mixer that gives
you direct access to the song’s stems, adjusting levels, pan and
reverb, while the lyrics scroll karaoke–style across the top.
There’s even an additional channel where you can record your
own parts over the top, via your device’s built–in microphone or
audio input. The Loop Station lets the user get creative in a Bernhoft’s Islander app is a luxury product, featuring very
different way. Many of Bernhoft’s songs are built up using a high production values and a price to match.
Visit SOS Music Business
forum
looper pedal and here you can do the same by switching on and
off the various song elements, layering them how you please. A
October 2014 limited number of songs are available to play with in each of
On sale now at main these sections, but the suggestion is that more may be added in
newsagents and bookstores future updates. That’s another great strength of the album app:
(or buy direct from the the content needn’t be static. The artist can keep adding new
SOS Web Shop) music and video, or sell additional material via in–app
purchases.
“I’ve come from the world of CDs, to file sharing, to streaming

and then back to a vinyl focus,” states Bernhoft in the official
press release. “What we wanted to do with the app was to bring
all this together and get the feeling of a gatefold sleeve into the
digital world. It’s full of amazing content but the music is the heart
of the app and it is in HD — exactly the way I heard it when we
were creating the music in the studio for the first time.”
The Islander app not only includes a conventional high–
Behind The Glass quality music player, but an eight–channel mixer allowing you
to rebalance the stems of some of the songs.
The creation of the app was steered by Philip Tennant, a former
Click image for October 2014 producer and engineer whose credits include Depeche Mode,
Other recent issues: the Cure and Dexy’s Midnight Runners. Like Bernhoft, he’s
September 2014 passionate about the medium and its potential to bring the
August 2014 listener right into the studio.
July 2014
June 2014 “All the stuff that happens in the studio that I love — that’s
May 2014 what we’re trying to provide people,” he explains. “That
>> Access All Issues excitement of seeing a song being created, seeing how the lyrics
change and the arrangements come together You don’t just get
the same audio quality that the artist is hearing in the studio
when they’re mixing. You also get all the liner notes, the lyrics
sheets, the photos — these are the kind of things we’d take for
ourselves during the making of a record. Then with the Loop
Station and the mixing console, you can really see how the
tracks were created. You can listen to the individual elements of
the record and remix them yourself, so you can see what the
Yet another way of interacting with Bernhoft’s music is via the
drummer’s doing, how the guitar part works with the bass.” Loop Station.
While the Islander app achieves its objectives brilliantly, some

significant questions still remain. Priced at £13.99, the app is a premium product and, at almost 2GB in size, it’s not small
given the relatively modest storage space on most phones and tablets. For both of these reasons, it’s hard to imagine anyone
buying all of their music this way. So is this a format just for hardcore fans, or will the album app become a standard mode of
release alongside existing digital and physical formats? How will an audience used to skipping from artist to artist on a
streaming service, or browsing their whole music library using a media player like iTunes, feel about a format that deliberately
places an album as a separate entity packaged in its own discrete wrapper?
“We feel that people are not going to fill their iPads with hundreds of these things. We don’t have hundreds of favourite
albums, we have a few,” Tennant says. “The whole point of the app is to pull people away from streaming and from buying one
MP3 here and there. I understand streaming and MP3s as a way of discovering new music, but once you’ve heard something
you like, you want to immerse yourself in it, and that’s where I see the app coming in.
“It’s meant to be heard as an album where you listen from start to finish. You immerse yourself in it because you care about
the artist and the music, and you concentrate on it the way we used to do when we listened to vinyl albums. Letting an album
play from start to finish wasn’t exclusive to vinyl — we did it with CD and cassette too — but it has kind of stopped happening
with streaming and playlists. This is trying to pull people back and say, ‘Here’s a complete artistic statement. You start here
and you end there, and if you want to know more about how we created this you go to other aspects of the app and find out
how it was all put together.’”
The successful Kickstarter campaign to launch the Pono high–resolution music player also taps into an evident desire for
better audio quality. Does Tennant see this as part of the same picture as the album app? “Pono is very interesting,” he says.
“It’s trying to find a similar audience, in that it’s high–end audio for people who really want to listen to music. But I think it’s
flawed in that you have to buy new hardware, and it doesn’t have any visual content. I think a lot of people nowadays watch
music as much as they listen to music, and I think we’ve got to accommodate that. Pono has its attributes but you’re still
missing sleeve notes, you’re missing album artwork You’re missing a huge aspect of what an artist is really about, even
though you are getting better sound quality.”
The album app, he says, allows the artist to keep all these aspects together and present a complete picture of their work to
his or her audience.
App Research
The Islander app is an impressive package and one that delivers a rich, immersive experience. Though not every album or
artist perhaps merits this depth and breadth of content, it’s easy to imagine a future for album apps as the medium for deluxe
editions of A1 new releases and expanded reissues of classic albums from the past. The idea of sitting on a sofa, iPad in
hands, and getting lost in a great record is an attractive one. The album app also arrives in good time to exploit the move
towards interconnected home entertainment systems. Perhaps you’ll soon be connecting your tablet to your wireless speaker
system to play around with the stems of yet another remastered reissue of Dark Side Of The Moon, then flicking to some
archive concert footage, streamed direct to your smart TV. It’s the album reimagined as a multimedia entertainment centre.
However, before that can happen, there are still a few things that need to be sorted out. How will licensing and royalty
payments work, given that this is a new format that combines a range of content — from audio and video to lyrics and sheet
music — all covered by different agreements? When an album app is sold, will it register on the official charts like a standard
download? What will album apps cost to make, and what price is the public willing to pay for them?
These are the kind of questions that another album app hopes to answer. A project supported by the UK’s Digital R&D Fund
for the Arts has been specifically looking at the viability of the app as an album format. As a test case, the project team —
music agency Script, app developers AgencyMobile and researchers at the Cultures of the Digital Economy Research Institute
(CoDE), part of Anglia Ruskin University in Cambridge — have produced an album app for Domino Recording Company artist
François and the Atlas Mountains.
“There has been a lot of buzz around the experimental album apps released so far by a
handful of artists such as Björk and Lady Gaga, but until now no one has properly looked
at the costs and benefits for consumers, artists and record companies,” explains Dr. Rob
Touson, Director of CoDE. “We wanted to find out how receptive the audience is to
consuming music in this way, and we wanted to see what artists could do with the format
when given the opportunity. Most importantly, we wanted to understand the obstacles —
from technical issues to the cost of development to a lack of awareness — that may need
to be overcome if this format is to succeed. Although the album app has many potential
strengths, questions surrounding the handling of royalties, chart eligibility and distribution
remain, and these must be answered if the format is to be embraced more widely.”
Making Tracks
In designing the iOS app version of François and the Atlas Mountains’ album Piano
Ombre, the development team consciously stopped short of some of the more
experimental interactive features seen in other album apps, sticking instead to what might
constitute the features of a ‘standard’ album app release. The app’s music player offers
up the full album in the iTunes Plus (AAC) format — still compressed but, at 256kbps,
higher quality than most MP3s — and the user also has the option to download the full
The Piano Ombre app was
album onto their device in MP3 and uncompressed WAV formats. This seems a sensible created for François and the Atlas
step to placate those who don’t like the idea of only being able to listen to the album Mountains, in part to test the
through the app. While you are listening to the album on the app, however, you can swipe waters regarding the commercial
viability of the format.
left and right to bring up extensive notes on the background to each song, along with the
lyrics in both French and English, which scroll in time with the song. There’s also the
option of turning on guitar chords above the lyrics so you can play along.
Exploring the rest of the app, you’ll find a list of upcoming gigs, with the option to buy tickets directly through the app, plus
band photos and a tour diary, all of which are pulled from the Internet and are continually updated. There’s some background
on the band, a full list of album credits, a link to the Domino web shop to buy more music by the band, links to social media,
and a selection of video playlists including a nice compilation of the tracks and artists that directly influenced the making of the
album.
The most unusual feature is something called the Sun Tracker, apparently suggested by frontman François Marry. Tapping
into the album’s overarching theme of choosing light over darkness, the Sun Tracker uses the iPhone and iPad’s built–in
gyroscope and compass to show you where the sun is, whether it’s night or day and you’re inside or outside. You simply move
the device around to scan your surroundings with the camera until a large red circle (as seen on the album cover) appears on
screen. You can then take and share a picture. Doing so unlocks one of 24 different bonus tracks included in the app — one
for each hour of the day, so to listen to them all you’ll need to do a lot of sun–tracking in the middle of the night!
It’s a great–looking app and enjoyable to use, again giving a sense of depth and
substance missing from digital downloads. Featuring hand–drawn animation created for
the app by François himself, there’s a definite sense that the personality of the artist
comes through, making it more quirky and alternative than Bernhoft’s app, with its
comparatively polished and glamourous feel. Even the Sun Tracker, as eccentric as it is,
is a reminder that this is a creative work, not just a functional music delivery system.
The Piano Ombre app costs £7.49 in the Apple App Store — less than a straight digital
download from the iTunes Music Store — and since the app’s price includes a full,
lossless digital download as well, it seems very reasonable.
“The feedback we’ve had from both consumers and the music industry has been very
positive,” says Dr Rob Toulson. “When we’ve put the app in front of consumers and
demonstrated what it can do, the majority have been really receptive. I think the challenge
will be establishing that awareness of what album apps are and the kind of experience
they offer, and that may require the backing of major labels and artists, the technology
companies who make the tablets and smartphones, and the big digital distributors of
music like Apple, Google and Amazon.”
Uncharted Waters
The Sun Tracker is a neat way of
The Piano Ombre app has scored a minor success in that it’s the first album app to be drip–feeding additional content to
fans.
recognised by the UK’s Official Chart Company, albeit via a slightly circuitous route. When
users download the album tracks onto their device, a sale is registered on the chart via
the record company. There’s no reason why album app sales shouldn’t count towards the charts directly, but it will require the
collective will of the key players to make it happen. So, is the music industry going to get behind this new format and give it
the momentum it needs?
“We’ve been speaking to labels, music publishers and people working in music licensing and there has been a lot of interest
in the format,” Toulson says. “I think everyone feels like this is going to happen — has to happen — it’s just the ‘when’ and
‘how’ that are unclear. Record labels are interested in releasing albums as apps and
using apps as a pre–release promotional tool, publishers see the potential to offer sheet
music and music tuition videos within the apps, and artists are obviously excited about a
digital format that lets them do more and get more creative than they can when their
songs are just being dumped onto iTunes.
“However, with no agreements in place for who gets what in terms of fees and royalties,
licensing has to be negotiated on an app–by–app basis. If the industry can come together
and agree on a standard licensing template for album apps — ideally one that sees the
artist and rights holders properly rewarded while leaving enough money for the record
company to recoup the cost of developing the app after the digital distributor has taken
their cut — the decision to go ahead and create app versions of new albums will become
a much easier one to make. Because this is a new format, the cost of producing an album
app is currently high, but it will come down. I think we may see the bigger record
companies each developing a standard format of their own that can be customised for
each of their artists and albums.”
Appwardly Mobile
There’s no doubt that we’ll be seeing more music released in app format in the near
future. What the longer term holds for the album app, and just how widespread the format
will become, is less certain. What we can definitely say is that the days when artists and A great benefit of the app format is
producers could just worry about making music are over, if they ever existed. Photos, the potential for cross–promotion
videos, blogs, social media — these are all things that most media–savvy music makers with other activities such as live
shows.
are doing already, but in a world of album apps, they’re part of the product rather than
promotional activities supplemental to it. But an app can also serve to draw these strands
together while charging hard cash for the artist’s output.
The album app offers some very significant advantages to musicians and producers. The importance of album credits is
often overlooked, but in the digital sphere there’s often simply no record of the musicians and engineers who work hard to
create the music. On top of this, the app allows the artist control how their work is presented, from the quality of the audio to
the running order. Above all, it’s a satisfying product with lots of added value that will hopefully encourage more people to not
just pay for the music they listen to but to buy the whole album.
But will this be the format for consuming albums in the future? It seems unlikely that any format will be ‘the format’. Other
forms of media, from TV and film to books and magazines, are already moving towards a however–you–want–it model, where
the technology serving up the content is web–based and the consumer can choose how they access it — through a web site
or an app, and via their PC, smartphone, tablet or TV. So it may be with music. The vital role that album apps can play in the
short term is to provide a focal point for the work that artists are producing, pulling together the diffuse strands of their output
and giving them more control. If albums apps can prove there is a public appetite for a richer kind of music experience that,
crucially, people are willing to pay for, the industry as a whole must surely take notice. .
Planet Of The Apps

If you want to explore the ways in which musicians and artists
are already taking advantage of the app format, here are five
of the most interesting ‘album apps’ yet created:
Björk: Biophilia
This astonishing, mind–bending app invites users to explore
a three-dimensional galaxy where each constellation is a
different track from the album. Each one features abstract
animations that let you interact with the music in a different
way. The app will output MIDI to control external instruments
and has been widely used in music education.
Philip Glass: REWORK_
The daddy of minimalist music is still down with the kids! The
app version of Glass’s Rework_ remix album features
interactive visualisations of all 11 songs, plus the ‘Glass
Machine’, which lets you play with polyrhythmic counterpoint Two classic creative ‘album apps’: Björk’s Biophilia and Philip
Glass’s REWORK_.
by interacting with two rotating discs.
Peter Gabriel: Music Tiles
Released to coincide with the 25th anniversary of Gabriel’s
So album, the Music Tiles app features stems taken from the
original session tracks, represented by colour–coded tiles.
When the tiles are placed on a grid, the stems that play are
dictated by the position of and connections between the tiles,
creating new arrangements.
Jorge Drexler: n
This app from Spanish composer Drexler features three
tracks — one orchestral, one choral, one vocal — written
specifically as interactive pieces. The user can arrange parts
on the fly and decide which elements are to be used in the
composition before sharing their work with others via the
Internet.
Paul McCartney: Band On The Run, McCartney, McCartney
II, RAM & Wings Over America
Notable more for the artist’s profile than the interactive content, five of Macca’s solo albums have been re–released in app
format. Featuring remastered audio, videos ranging from rehearsal footage to documentaries and interviews, rare photos
and album and single artwork, the apps cost £5.49 each — significantly less than the remastered albums themselves. A
sign of things to come?

In this article:
Safe Bets
Notes From The Deadline
Music For Music’s TV Music From The Inside Buy PDF
Sake Published in SOS October 2014

Credits can be a vital resource to the media composer, but
they can also be a ball and chain. Printer-friendly version
Paul Farrer
I
n 2012, the Nobel Committee awarded its highest honour to the entire European Union in recognition of its contribution to
the advancement of peace, democracy and human rights. Which means that not only can I claim credit for the same
number of Tour de France victories as Lance Armstrong and the same number of Academy Awards as Alfred Hitchcock,
George Gershwin and Leonardo DiCaprio combined, but I also have the same number of Grammys as Björk and Tupac and I
am the proud (although admittedly joint) recipient of a Nobel Peace Prize.
The concept of credit is a funny thing. When you buy a

hamburger, it doesn’t come with a list on the side of the box
naming all the people (and creatures) who contributed to the
finished product. Your car doesn’t have a little plaque on the
back telling you the names of everyone who helped to design
and build it. Yet in the world of media and show business, credit
is one of our most vital commodities. At the very start of your
career, an on–screen credit marks your all–important entrance to
the party. It announces that you play at a certain level and, of
course, for future work, allows potential clients to assess what
kind of work you’ve done in the past. The dark side of credit is
that it also represents a potent negotiating carrot for getting
competitive deals from people. Do you think sausage makers
might be able to reduce labour costs if all meat industry staff had
the option of an on–package name check? “I always tattoo my name next to the scar. I get a lot of repeat
business that way.”
Safe Bets
The global financial credit crunch was caused largely by not heeding the familiar warning that should probably also be applied
to on–screen credit, namely that “past performance is not an indicator of future success”. Established composers, like
everyone else whose job titles are listed on screen, always run the risk of being creatively associated only with the types of
show they have been credited for. If you are lucky enough to be part of a project that gets noticed, that risk becomes a near–
certainty. We are all guilty of pigeonholing each other in all walks of life, and producers looking for the best composer for a
project (or, to put it more accurately, the composer least likely to muck it all up) are no exception.
The most useful way to view TV and film projects is like betting on football teams. People stump up large amounts of money
in order to make them happen, and the longer (and more project–appropriate) your list of credits, the more likely you are to get
picked for the team. You might think you’d be really good at scoring a dark, twisted serial killer drama. But perhaps you’ll never
know, because no–one has ever asked you to score one. And why don’t they ever ask you? Because no–one has asked you
yet. Clever, isn’t it?
Music For Music’s Sake

These days, end credits seem to be a diminishing return, often reduced to a small scrolling box in the corner of the screen
which can only be read by those watching in high definition and not distracted by the trailer for the next scheduled show taking
up 90 percent of the rest of the viewable area. Perhaps this is a good thing. As someone with an apparently exotic and
untypeable surname, which is beamed to the nation incorrectly spelt on at least one project each year, I say roll on the day
when we no longer seek public glory for our work and instead focus solely on getting paid properly. As someone cleverer than
me once remarked, “Graffiti scrawled anonymously on a toilet wall exists for neither creative praise nor financial reward and
as such is the highest form of art there is.” .

In this article:
Real Estate Plays
Off The Record
Did Grohl Get It Music & Recording Industry News
Wrong? Published in SOS October 2014
Historic studios might hold emotional value for us, but they
are, after all, still only buildings. Printer-friendly version
Dan Daley
B
en Folds, the eclectic artist, composer and producer, has been ensconced in RCA Studio A in Nashville for the last
dozen years. A robust relic, Studio A was built by producers Owen Bradley and Chet Atkins in 1957, as a kind of
bunker that would keep country music production firmly planted in Nashville, at a time when the country divisions of
major labels often had their strings pulled by headquarters in New York.
Folds has rented Studio A since 2003, at first as his own place to work and since 2008 also as a for–hire studio facility. It
must have always seemed a tentative arrangement, based on what are now dozens of three–month leases with the landlord
estates of Bradley and Atkins (which still own the building). A lot of great music has come out of Studio A, with artists like Don
Gibson and Jerry Reed, through to Kacey Musgraves and Amanda Palmer recording there, but its future has become even
more unsure in recent months. As Nashville’s residential real–estate market has heated up, the building housing it became the
target of a condo developer. As pressure mounted on the Bradley and Atkins estates to sell, Folds posted an appeal to save
the historic studio on his web site, on what would have been Chet Atkins’ 90th birthday.
It worked. The developer agreed to make an effort to keep the studio intact within whatever new residential structure he
would erect, and to pull the plug on the entire project if that couldn’t be done. But Studio A’s saga may be ending sooner than
Folds would care to acknowledge. This raises a fundamental question that music production will face more frequently in
coming years: where does heritage end and history begin?
Real Estate Plays

New York’s RCA Studios shuttered in 1993, long before anyone had ever heard of Napster. What felled it, along with several
other major studios in the 1980s and ‘90s, were real–estate plays. As an island, Manhattan is especially sensitive to the
scarcity of land, and even during the go–go decades of the late 20th century, when lucrative major–label deals were
ubiquitous, music was no match for finance, medicine, academia and other big–money tenants. London experienced its own
version of this syndrome, on Denmark Street, where music publishers and recording studios mushroomed along what became
known as ‘London’s Tin Pan Alley,’ among them Regent Sound and the eponymous Tin Pan Alley Studios, London’s oldest.
Media Sound and The Hit Factory are now condo buildings; the old RCA Studios became offices, and even today the real–
estate flu is still taking its toll: Masterdisk’s multi–room facility downsized and moved to smaller quarters this year. There’s little
left on Denmark Street to suggest what went on there, save a round, blue plaque. Now, as the economy shakes off the Great
Recession, and demand for housing rises, market forces are drawing a bead on Music Row, and RCA Studio A with it. In fact,
migration away from the Row has already begun: Sony Music’s Nashville headquarters were just purchased by nearby
Vanderbilt University, and the label has decamped to the Gulch, one of a number of previously dingy neighbourhoods that in
recent years have sprouted glistening condo towers and upmarket shops. Music Row, meanwhile, despite a small but ardent
groundswell of support for some kind of historical overlay to preserve it, is becoming to Nashville what Bourbon Street has
become to New Orleans — a tired tourist attraction.
Did Grohl Get It Wrong?

There are scores of studios around the world that can assert historical significance. Do all of them need to be preserved, as
either museums or working facilities? Harold Bradley — the brother of Owen and himself regarded as the most recorded
session guitarist ever — sent his own letter the Nashville city council. In it, he cited the famous Quonset Hut (built by he and
Owen in 1954) as deserving of legacy protection. Bradley, now 88, also contended that nostalgia shouldn’t be a part of this:
the entire building housing Studio A, which also was RCA Records’ Nashville HQ for decades, was always intended as a
financial hedge for the owners’ families, he said, built way before Garth Brooks or Taylor Swift money hit the Row, and a place
they’ve been trying to sell for a quarter of a century. “This was business,” Bradley wrote. Owen said, “One day we might not
have anything, but if we buy this property and build this office building, we can at least have something to sell.’”
In fact, Bradley went on to say something that might rattle Sound City documentarian Dave Grohl: “The architecture of the
Nashville sound was never of brick and mortar,” Bradley opined. “Certainly, there are old studio spaces that, in our
imaginations, ring with sonic magic; but in truth, it’s not the room; it’s the music... Music City isn’t about making a perfect room,
or hanging just the right baffling. Turns out, the architecture of Nashville’s evolving sound is a synergy of creative energy.
That’s still here, and it has nothing to do with this building.”
For the sellers, the property’s value is the reward for 60 years’ worth of foresight and risk taking. For Folds and others, it’s
an affordable acoustical oasis at a time when visceral economic realities are putting the space needed to roll out a 20Hz wave
in jeopardy. But the Atkins estate, and the Bradleys, are no mere opportunists — Music Row and the Nashville music business
as we know it arguably wouldn’t have existed at all without them. Meanwhile, music gets successfully made in a wider variety
of environments than ever before. The scene won’t miss one more conventional space. And it begs a larger question: does
music production need another museum, working or otherwise? For better or for worse, apparently, it does need more
condominiums. .

In this article:
Part–time Past
Studio File
Dec To Dex Teldex Studios Berlin Buy PDF
Hannes Bieger Published in SOS October 2014
O
nly a very few studios in the world are capable of accommodating the largest recording Printer-friendly version
gigs: full orchestra sessions, with up to 80 musicians playing in the live room all at the
same time. The majority of these large recording venues are located near the centres of
the international film industry, like the Sony and Fox scoring stages in Los Angeles. Other important studios for orchestral
recordings include the venerable Abbey Road and AIR Lyndhurst in London — and Teldex in Berlin. Located in the rather
quiet and upscale borough of Lichterfelde in the South West of the city, the studio has a rich history which began in the late
19th century.
Part–time Past
Well hidden in a green backyard, the building housing the studio is located a mere two kilometres away from the Berlin Wall,
close to the southern edge of West Berlin. Built in 1890, the large brick complex served as a multi–functional cultural venue
and was originally called the Lichtenberger Fests le (Lichtenberg ballrooms). It was home to a wedding hall, a restaurant and
a beer garden outside, and it staged a wide range of cultural events, from fun fairs to concerts of the Berlin Philharmonic.
Shortly after World War II, the venue was restored to use, and
around 1950 Telefunken began to rent the building, making it
their recording headquarters. For the first few years, recording
took place only during weekdays. A system of rails was installed,
and the equipment was carted into the hall on trolleys, to be
stowed in the basement over the weekend, when the dance
bands would enter the hall again. But this was not sustainable in
the long term, and from 1954, the building was used exclusively
as a recording studio. It was operated by a joint venture between
Telefunken and the British Decca company. Named Teldec, it
served as one of the main recording venues for the record
company of the same name. In this era, Hildgard Knef and
Caterina Valente used to record at Teldec, and in the ’70s,
German rockers Udo Lindenberg and Peter Maffay, amongst
The live area at Teldex is almost exactly the same size and
others, began to craft their productions here. Telefunken also set shape as Abbey Road Studio One, and is likewise capable of
up a facility for equipment production in the front building: Direct accommodating a full orchestra.
Metal Mastering technology, where the music is cut straight into
copper plates instead of the conventional acetate lacquer, was
developed on site in the ’80s in conjunction with Neumann.
Dec To Dex
Teldec was acquired by Warner Bros in the early ’90s, and in the
following years, the studio was mostly used for mixing and post–
production, not so much for recording sessions. At one point
there were no more than two recording gigs per month, but the
highly skilled team toured the world with a mobile kit to record on
location, only returning to the studio for mixing. Around the turn
of the millennium Warner merged with AOL, and in early 2001,
all worldwide recording facilities previously owned by Warner
were closed practically at the same time. This was the big
Teldex’s control room houses a Studer 980 desk with an Avid
chance for three former Teldec employees, Friedemann 5–MC controller.
Engelbrecht, Tobias Lehmann and Martin Sauer, who acquired
the equipment, took over the lease and reopened the studio as Teldex in 2003. It is
the largest independent recording facility in Germany, and the 455–square–metre
live room, which has almost the same size and proportions as Abbey Road’s Studio
One, remains in its original shape.
The control room, by contrast, was completely remodelled prior to the reopening.
It was home to a 72–input SSL 9000 for many years, but, according to Tobias
Lehmann, maintenance eventually became too difficult, so this was replaced with a
48–input Studer 980. Configured in a quite modern layout with an Avid 5–MC
controller as the centrepiece of the console, the Studer proves an ideal solution for
the Teldex staff. It is accompanied by two eight–channel Millennia HV308 preamps,
a couple of vintage valve–based Telefunken preamp modules and some other
hand–picked outboard units, which offer plenty of high–quality input channels for
the classical and film soundtrack recordings which form the lion’s share of the
recording business at Teldex these days. Besides the main hall, Teldex today also
operates a number of smaller studios and editing suites.
The studio still operates one of its two original echo chambers. Both were wired
as mono chambers when they were built, and together they served as a luxurious
dual mono/stereo chamber at one point, but the chamber that remains in use today
is equipped with a mono Klein+Hummel loudspeaker and a pair of Neumann KM84
Teldex’s mic locker includes one of the
microphones. Speaking of which, the Teldex team secured the entire microphone largest collections of vintage Neumann
collection of the original studio, which counts all kinds of classic Neumann mics in the world. Here an M49 is being
microphones literally by the dozen, including numerous small–diaphragm capacitor employed as a close mic for a ’cello.
mics such as the KM84 as well as several of each of the valve
classics like the U47, M49 and M50. This is clearly one of the
largest collections of vintage Neumann mics in the world! Decca
engineers often modified their microphones, and a number of
Teldex’s M50s were equipped with XLR connectors and FET–
based output stages in the Teldec era. According to Tobias
Lehmann, the sonic difference is negligible when being used as
room mics, and they are much easier to set up than the valve
originals.
Their work in the field of classical music has earned the Teldex
team a number of Grammy awards. They have worked with the
Berlin, Vienna and New York Philharmonics and with conductors
such as Daniel Barenboim, Nikolaus Harnoncourt and Sir Simon
Rattle, to mention just a few names. Many celebrities in the field Most preamp duties are handled by the input section of the
Studio 980 console.
of pop and rock music have also found their way into the studio,
including artists such as Nina Hagen, Roger Waters, Rammstein,
Visit SOS Music Business
forum
Alicia Keys, Celine Dion and even Britney Spears. The musical
life of Teldex today is almost as diverse as that of the
October 2014 Lichtenberger Fests le 100 years ago! .
On sale now at main
newsagents and bookstores
(or buy direct from the
SOS Web Shop)
The outboard collection includes vintage units such as the

ubiquitous Fairchild 670 and this EMT 156 compressor.
Click image for October 2014

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One of the original echo chambers

remains in operation today. It is
equipped with a pair of Neumann KM84
microphones and a vintage
Klein+Hummel OY speaker.
The impressive microphone collection counts classic

Neumanns literally by the dozen!
Q Are impedance mismatches causing my vocals to

sound harsh and aggressive?
Sound Advice : Recording
Hugh Robjohns
I work in a facility where AES3 110Ω cables have been used to connect some splitter outputs to mic
preamps. What kind of influence, if any, can this ‘impedance mismatching’ have on the signal quality? I find the sound from the
vocal booth generally quite harsh and aggressive, with excessive sibilance. To add extra complexity to the picture, the
preamps are Focusrite Isa 430s with variable input impedance! Thanks for your help.
Vincent Ladriere, via email.
SOS Technical Editor Hugh Robjohns replies: In short, none at

all. There is no impedance mismatching here because analogue
microphones and preamplifiers do not operate in an impedance–
matched environment. They operate within a strict voltage–
matched interface, where the load impedance is roughly 10
times higher than the source impedance. The ability to change
the preamp’s input impedance affects the way the microphone is
loaded, which in turn alters the tonality with some dynamic and
transformer–coupled mics — but it still has nothing to do with a
matched-impedance operating environment!
All cables have a ‘characteristic impedance’ and most

balanced mic cables are around 100Ω or so. However, this
nominal cable impedance only becomes significant and relevant Not all balanced XLR cables have the same ‘characteristic
impedance’ — but is this something the home recordist really
when used to pass signals within an impedance–matched needs to concern themselves with?
interface arrangement, and even then only for signal frequencies
over 1MHz or so (ie. digital audio signals), or in cable runs over 10 of kilometres in length (as with telephone lines, for
instance). None of these conditions apply in a simple studio mic-to-preamp connection, which is only a few 10 of metres long
at most, carries no significant audio above 20kHz or so, and operates in a voltage–matched environment.
It’s worth noting that the AES decided upon the use of a nominal 110Ω cable impedance for the AES3 digital interface
specification precisely because that is what most standard studio mic cables exhibit. The original idea was for people to be
able to run AES3 digital audio using standard studio cables to make it a ‘familiar and friendly’ interface. Unfortunately, that was
a flawed concept, mainly because standard studio mic cables have way too high a capacitance for conveying digital signals in
the MHz region, and that’s why very low–capacitance balanced cable is available for AES3 applications. However, the low
capacitance is not a bad thing for analogue audio either, and so it is very common to standardise on AES3–compatible cabling
when wiring new installations.
I can guarantee that the source of the harsh, aggressive, and sibilant sound you’re complaining about has nothing to do with
the cabling. I would suggest that it’s actually due to a combination of the vocal booth’s acoustics, the microphone choice, and
the mic placement, and these are the areas in which you need to experiment to find a solution. .

Q How do I emulate a real drum sound using

EZdrummer?
Mike Senior
I’m currently trying to emulate the drum sound on the recent Lower Than Atlantis song ‘Here We
Go’ using EZdrummer, and I have each drum component going to its own channel to be mixed, before sending the kit as a
whole to a bus where I then apply reverb. Each component of the kit will have some degree of compression applied to it, so
would it then be overkill to add more compression to the kit as a whole on the bus channel? I just find this then helps bring the
kit ‘together’, but I’m concerned that it might be a bit fatiguing on the ear or detrimental in some other way. I use Stillwell
Audio’s Rocket compressor, because I like my drums to sound like they’re having the crap beaten out of them, and that
compressor gives EZdrummer that power and punch.
SOS forum post
SOS contributor Mike Senior replies: That song is pretty

extravagantly compressed, I grant you, and in a general sense
any compression setting is fair game as long as you like the
sound of it; there’s no need to second–guess yourself on that
count. However, if you’re already working with a MIDI–triggered
drum instrument which effectively gives you complete control
over the simulated performance, then one of the fundamental
roles of compression is rendered redundant, namely its ability to
make the performance more even. As such, I’d concentrate more
on what compression has to offer in terms of shaping per–hit
amplitude envelopes and adding appealing tonal
coloration/distortion. In this context, I’d expect to be using more
compression on ‘mixed’ channels such as the overheads or room
signals (and indeed the whole drum mix) than on individual
‘virtual mic’ channels, because it’s easy to stamp the life out of
each drum’s attack if you’re not careful — and you need all the Given the control you have over the performance of MIDI–
life you can get when you’re working with virtual drums as it is! triggered drum instruments such as EZdrummer 2 (pictured),
you may find when it comes to compression that you can
focus more on group channels such as room (or ‘ambience’)
I’d also suggest expanding your repertoire of compressor mics and overheads, and less on the close-miked kit pieces.
plug–ins. It’s not that I don’t like Rocket — I use it regularly —
but it’s very fast, which means that if you rely on it for everything you may lose too much transient definition across the board.
It’s also a compressor with a particular tonal character and it may not suit some sounds as well as it does others. If you like
the Stillwell plug–ins, then try out Bombardier too, which works especially well for mixed bus signals because of its automatic
attack/release time, RMS window, and side–chain contouring options.
Another thing to notice about this Lower Than Atlantis mix, though, is that the amount of drum room sound and/or reverb
being used varies a great deal during the song. With that in mind, be careful how you compare your own work–in–progress
with this production. If you reference against the verses (where it’s easier to hear the details of the sound) you’ll almost
certainly mix too dry, and the temptation may then be to over-compress your own sound to achieve greater sustain during your
choruses. Automation is the way to avoid problems here — consider riding up the room–channel and reverb–return faders for
the more high–energy moments of the arrangement. .

Q How does knowing the key of a song help when

mixing?
Sound Advice : Mixing
Mike Senior
Some audio forums are really keen on pitch/frequency charts as an aid to mixing. I kind of get it:
if a song is in A, you’ll have lots of 440Hz, 220Hz, 110Hz, 880Hz, and so on — but so what? What are you supposed to do
with that information? And what about all the harmonics? And all the other notes in the key of A, and all their harmonics? So
does knowing the key of a song actually help when mixing? Personally, I don’t think so, but I’d be happy to be corrected!
Jonathan Kaspar, via email
SOS contributor Mike Senior replies: Broadly speaking I agree

with you. On the face if it, knowing relationships between in–key
pitches and specific Hertz figures might seem potentially useful if
you wanted to pick out individual note fundamentals or
harmonics for surgical EQ corrections, but those kinds of EQ
moves aren’t that common in the normal run of things, in my
experience. Even when you do need to find individual harmonics,
I don’t think knowing the key makes it substantially easier —
personally, I find it quicker to use a high–resolution spectrum
analyser plug–in, such as Voxengo’s cross–platform freeware
SPAN, because the relevant frequency peaks will usually be
abundantly obvious to see (and measure) in the display.
Alternatively, where I’m looking to boost a specific harmonic, it’s
a piece of cake to just sweep a high–resonance peaking filter
around in my EQ to find the frequency I’m looking for. Basically, I
can’t see that knowing the key will really save you time in this
If you need to do pitch–correction work while mixing, it can be
regard. handy to know which key you’re working in to reduce
mistracking by the note–detection algorithm.
However, when you need to apply pitch–correction to any part
at mixdown, it’s often a lot quicker to set up the required correction scale if you already know the key of the music, and the
same might also apply if you found yourself using any kind of intelligent harmony generator — to surreptitiously thicken
backing vocals, for instance.
Another specific situation where it’s useful to know the exact relationship between musical pitches and frequency values in
Hertz is if you’re trying to set up a short, high–resonance delay line as a pitched resonator — not exactly an effect you reach
for every day, but it can be a surprisingly useful effect on occasion. In this case, you can work out the delay time you need by
dividing 1000ms by the pitch’s frequency value in Hertz. So, for example, a 440Hz ‘A’ would require a delay time of around
2.27 milliseconds. Although it might be possible to sweep your delay time control around to find this value, the resolution
available via your plug–in’s GUI may not make this very easy at such small delay times. .

Q Why has Paloma Faith's latest track been recorded

'out of tune'?
Mike Senior
I realised yesterday looking at the official YouTube video for Paloma Faith’s new song ‘Only Love
Can Hurt Like This’ that it has been recorded out of tune by about 50 cents. This means that unless you have a keyboard that
can be detuned you cannot play along! I found this out when teaching it to a singing pupil and I’ve never come across it
before. What would be the reason for doing this?
SOS forum post
SOS contributor Mike Senior replies: Having imported this track into my own DAW, it sounds like it’s about 35 cents sharp, in
fact, and as you say it’s enough to put a spanner in the works if you want to play along — so I’d expect any producer
operating in the mainstream pop arena to have second thoughts about doing that deliberately.
From a technical perspective, one reason you might want to

do this is because you’re building the track around some crucial
element that isn’t at concert pitch. Live ensembles will often
retune to match an unavoidably off–pitch piano or organ, for
example, and in this specific production I wonder whether the
brass riff you hear at the outset might be a phrase sample that
was slowed down to fit a new tempo, thereby dropping its pitch.
Paloma might very well have then recorded a load of vocals over
the programming without anyone noticing the non–standard pitch
reference, at which point any subsequent change would have
risked unwanted pitch–processing artifacts on the vocal Is the eccentric tuning on Paloma Faith’s ‘Only Love Can
recordings — potentially a reasonable justification for sticking Hurt Like This’ due to design or mistake?
with the sharper pitch. That said, I don’t give that explanation
much credence myself, because I reckon there’s pitch processing all over this record already, and I’m not convinced 35 cents
more shift would have made a tremendous sonic difference with any self–respecting pitch–correction algorithm.
Producers frequently change the pitch of a production so that it better fits the singer’s vocal range, especially in cases like
this where the line happens to cover such a wide range — almost two octaves. But I don’t think this would be a good reason
here, as a shift of 35 cents would make very little difference to the singability, so there would have been no impetus to stray
from simple concert–pitch transpositions on those grounds in my opinion.
My best guess is that a repitching decision was made more on aesthetic grounds, and that the track’s playback rate was
increased late in the production process (possibly even during mastering), to remedy a perception that the initial tempo choice
was too slow — a common pressing–room quick–fix back in the analogue days wherever the master tape machine boasted a
varispeed control. Although software can now change the tempo of a complete mix without changing its pitch, many people
prefer a bit of pitch–change to the warbling side–effects that can arise with even the most sophisticated digital time–stretching
algorithms.
Mind you, another forum poster suggested an intriguing alternative thought: that the music might have been deliberately
recorded in the RA Natural tuning system, which uses a 424Hz reference pitch, thereby shifting the musical ‘grid’ of semitones
about 64 cents flat — or 36 cents sharp (depending on how you look at it), which would tally pretty accurately with what’s
going on in this song. You can check out the full philosophy behind RA Natural tuning at www.ramusic.com, but basically it
revolves around the two mathematical values Pi and Phi, manifestations of which are abundant in Mother Nature. Proponents
claim it helps music to communicate more directly and instinctively. However, there are no other pitch–offset tracks on Faith’s
album, A Perfect Contradiction, and I couldn’t find any amongst other work from the single’s two producers either, so I’m pretty
sceptical that esoteric mathematical theory was a hot topic in that particular control room.
But given that this song has been the artist’s biggest UK hit, might the RA Natural effect have been a contributing factor?
Come to think of it, the song is in tune with UK mains hum too, which also happens to be 35 cents sharp of concert pitch —
maybe that was responsible! Clearly, Diane Warren’s songwriting credit had nothing to do with it. (And, before you ask: no,
Warren’s very much a 440Hz kind of lady!) .

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