Sound On Sound June 2009 - In This Issue

June 2009 In This Issue

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Reviews People + Opinion
Ableton Live Suite 8 Origin Of The Species
Digital Audio Workstation [Mac/PC] Leader
The challenge was to add yet more features to Live without Paul White proffers more thought-engaging words.
compromising its legendary ease of use. Have Ableton
succeeded? Playback
Readers’ Music Reviewed
Sponsored by www.breed-media.co.uk
AEA A440
Active Ribbon Microphone Raphael Saadiq
Some microphones ooze quality to the extent that price almost Producing The Way I See It
becomes unimportant. Is AEA’s new active version of the Artist and producer Raphael Saadiq has channelled his love of classic
venerable RCA 44 design one of them? soul records to create something convincingly vintage, yet fresh-sounding
and alive.
Arturia Origin Secrets Of The Mix Engineers: Declan Gaffney
Rackmount Synthesizer Workstation U2 : 'No Line On The Horizon'
Arturia’s first true hardware synth aims to let you conveniently The sessions for U2’s No Line On The Horizon took the idea of
create your own modular system from modelled components of spontanaeity in the studio to new levels. Engineer Declan
some of the best-loved analogues of all time. How well does it Gaffney was the man charged with creating order from apparent
realise this ambitious goal? chaos...
Dan Dean Solo Strings Advanced Sounding Off
Kontakt 2 Player Mike Senior
Fractal Audio Axe-FX Ultra The end of the iPod era?
Guitar Amp & Effect Simulator Within Temptation: Producing Black Symphony
Fractal Audio’s Axe-FX is the guitar processor with ’80s looks Ronald Prent, Darcy Proper & Wouter Strobbe: Blu-Ray
and 21st-century power! Audio
Few artists so far have taken advantage of the Blu-Ray format’s potential
to deliver stunning audio quality. A concert film by Dutch metal act Within
Heil Sound PR40 Temptation shows what’s possible.
Cardioid Dynamic Microphone
Technique
Listening to the sound from this extraordinary dynamic mic, you
might be forgiven for thinking it was a ribbon or a condenser. Apple Notes
Educational Training in GarageBand
With the latest version of GarageBand, Apple have become the first
Lexicon Ionix FW810S company to integrate educational training into music-creation software. But
is this merely a simple gimmick, or a powerful new way of learning to play
Firewire Audio Interface
an instrument?
The Ionix FW810S comes with built-in Lexicon monitor reverb and onboard
Dbx dynamics and EQ. So does all this DSP add up to a genuine USP? Classic Tracks: John Lennon ‘Whatever Gets You
Line 6 Pod X3 Pro Thru The Night’
Guitar Modelling Processor Producer: John Lennon • Engineers: Roy Cicala, Shelly
In true Line 6 tradition, they’ve followed their Pod X3 and X3 Live Yakus
processors with a rackmount professional version, the X3 Pro, which has Engineer Roy Cicala worked on all of John Lennon’s albums
enhanced I/O and a few capabilities not provided on the standard ‘kidney’ from Imagine onwards, and in ‘Whatever Gets You Thru The
or X3 Live floor-unit versions. Night’, recorded the only solo number one hit single of
Lennon’s lifetime.
Native Instruments Maschine
Groove Workstation Creating Custom Touchscreen Interfaces In Logic
NI’s hybrid hardware and software beat machine promises the Logic Notes & Techniques
best of both worlds. Does it deliver? You'll be surprised at how cheap and easy it can be to enhance the
usability of your Apple Logic setup with a touchscreen.

Cubase: Funky Guitar Processing

Novation 61SL MkII Cubase Notes & Techniques
Controller Keyboard If you want something to spice up rhythm-guitar loops to use in your funk
The updated SL range rejoices in touch-sensitive knobs and and dance tracks, Cubase has just the Tonic
sliders, and in Novation’s ingenious Automap technology.
Cubase: Funky Guitar Processing | Audio Files
Hear For Yourself
Links to audio files that accompany the SOS JUNE 2009 Cubase article,
Philtre Labs Bollywood Elements for newsstand/bookstore readers who do not have eSub-access to the
WAV/Apple Loops main web article.
Plug-in Folder Effective Selecting & Locating In Digital

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Sound On Sound June 2009 - In This Issue

Reviews Performer
• D16 Silver Line
Digital Performer Notes & Techniques
• Sonic Charge Synplant
The ability to move around your project quickly and easily, making fast and
Pro Audio DSP DSM accurate selections for editing, can take much of the pain out of music
Multi-band Dynamics Plug-in For Mac OS production. We round up some of the best power-user shortcuts.
Pro Audio DSP’s first product is a novel dynamics plug-in, with applications Exploring Sonar 8’s Dimension Pro Synthesizer
ranging from de-essing to loudness maximising.
Sonar Notes & Techniques
Prodipe Pro Ribbon 8 Now that Dimension Pro is included with Sonar 8, it’s time to investigate
Active Monitors how to exploit it beyond the usual sample-playback functions.
Ribbon tweeters can yield a smooth sound, while still capably Getting Into Pro Tools
reproducing transient detail — and the Pro Ribbon range
Part 2: Installation & Session Basics
promises to do so for an attractive price.
Our short series on getting started with Pro Tools continues with some
advice on configuring the software and your first Sessions.
Sample Magic Minimal Techno
Multi-format Sample Library Mix Rescue: Adam Bevan
Sound Workshop
Sequis Motherload Elemental Beefing up bass parts played on guitar and importing GarageBand songs
Dummy Load & Speaker Emulator into Logic are among the challenges for Mix Rescue, as we add some
spice to Adam Bevan’s mix.
SPL Rackpack
Modular Preamp & Processor System Mix Rescue: Adam Bevan | Audio Files
Sound Performance Lab have developed an enviable reputation Hear For Yourself
for their hardware products, and now you can buy more for less These audio files are presented in both MP3 (for auditioning) and AIF, for
with their modular rack system. more serious critical listening and comparison in your DAW software.

PC Notes
Toontrack Drumtracker Time for an Upgrade?
Drum-replacement Software [Mac/PC] If you’ve built your own music PC in the last couple of years, you might
Need to use samples to rescue a dodgy drum recording? now be finding that aspects of it need upgrading. PC Notes offers some
Toontrack’s neat utility will generate MIDI hits and map them to advice.
the virtual instrument of your choice.
Studio SOS: John Clark
Home recordings
Waves GTR3 & GTR Ground
We lend some practical studio advice to a seasoned pro guitarist who is
Amp Modelling Software & Foot Controller
rather newer to the world of songwriting and home recording.
Waves have made a rare foray into the world of hardware with
the GTR Ground, designed to give hands-free control over their Music Business
Guitar Tool Rack software amp simulator.
Notes From The Deadline
TV Music From The Inside
Zero-G Vocal Foundry There’s learning, and there’s education. And when it comes to music
Multi-format Sample Library technology, the two aren’t always connected, believes Paul Farrer.
Zoom H4N The Remix Business: Part 1
Portable 24-bit Recorder The World Of The Remixer
The H4 remains one of the most flexible of the current crop of Remixing is a modern phenomenon that has turned into a viable way for
portable 24-bit recorders, but Zoom have found plenty of room hi-tech musicians to make money from their skills. But how do you get
for improvement — and, more importantly, they’ve used it. heard, how do you land a job, and how much should you charge to do it?
We get you started with the insider’s guide.
Competition
WIN a day's Mastering at Galaxy Studios, Belgium
Deadline: 2009-06-30

Sound Advice
Q. Does mono compatibility still matter?
Q. How can I improve my vocal recordings?
Q. What’s the best order for mixing?
Q. Which room should I record in?

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Ableton Live Suite 8

In this article: Ableton Live Suite 8

Eight Piece Suite Digital Audio Workstation [Mac/PC]
Subtle Changes Published in SOS June 2009
Track Grouping Reviews : Software: Sequencers+DAWs
Print article : Close window
Cross Clip
Cleverness
The challenge was to add yet more features to Live without
Instrumental
compromising its legendary ease of use. Have Ableton succeeded?
Developments
Looping The Loop Nick Rothwell

S
Bring On The ince its arrival on the audio software scene in 2001, Ableton
Vocoder Live has carved out a niche as a hugely popular music
Sample Libraries production and performance tool. The landscape eight
Conclusions years ago was dominated by well-established, complex linear
Other sequencing packages that were geared towards multitrack studio
Enhancements recording and production, but the newcomer quickly gained
popularity by aiming at a slightly different part of the market: live
Share And Share
performers and DJs. Live has always been something of a two-
Alike
headed beast — a loop-based performance instrument on the one
Maximum Potential
hand, and a linear recording and production platform on the other
Ableton Live 8 — and with the arrival of MIDI sequence support, opening the door to VST Instrument and effect hosting,
£600/£400 Live became a serious contender as a studio production platform, while still enjoying a position on practically
every performer’s laptop on the planet.
pros
Still sleekly designed, robust Live’s remarkable success comes down to some simple but well-considered design choices. Firstly, it
and beautifully ergonomic. knows about looping, and works very hard at cueing, synchronising and aligning looped material, and
Modest but well-engineered providing a sophisticated editing interface for loop-based music. Secondly, it implements a number of
improvements to the core powerful and versatile features in a clean and reliable manner: instruments and drums can be ‘racked’ and
environment. chained in infinite combinations, audio can be routed and mixed in ways an actual mixing desk can only
Suite 8 is a bumper pack of dream of, and the automation support is obsessively thorough and rock solid. Thirdly, Live’s interface is clear
versatile instruments and
and simple: one window, two views, and fixed panels for instruments and sample browsing.
high-quality sample sets,
offering a massive and Combining sophistication with ease of use is incredibly difficult, but Live manages to do so, with an
comprehensive production interface that promotes flexible working without clutter or distraction. But this elegance of design tends to fly
environment. in the face of the software industry, where an arms race pushes vendors to add more and more features to
cons their products to win on press release bullet points. Despite a major version release once a year, Live has
No clear mapping between already made it to version 7 without compromising its clean design and solid reliability. Does it still hold itself
Live Packs and the together in version 8?
instrument library.
The on-line sharing Eight Piece Suite
environment is rather
primitive and still in beta-test. I reviewed the full Live 8 package, called Ableton Suite 8. This is a four-DVD boxed set consisting of Live 8
The looping instrument is not itself, a core library of samples and presets, the full set of Ableton’s software instruments, a collection of
pitch-corrected. loops and samples from Cycling 74 and Zero-G, a library of Latin Percussion instruments and samples,
summary version 2 of the Essential Instruments Collection (EIC), and two DVDs dedicated to session drums. The
Ableton Live 8 is a very sample libraries are packaged as ‘Live Packs’ — compressed archives which Live itself unpacks and installs
impressive piece of — so you have some choice as to what you want to install.
performance-oriented, loop-
The full set of Live Packs occupies 48GB of disk space when unpacked, and installation on my MacBook
based audio/MIDI workstation
software, while the full Live Pro took several hours. All the instruments and sample sets occupy the same hierarchical navigation tree,
Suite 8 package adds 48GB and it’s not immediately clear which instruments are part of which Live Packs: for example, some presets
of sample libraries and a rich contain layers that mix samples with physically modelled synthesizers. Arguably this doesn’t matter much —
collection of modelled who cares what instruments are used so long as the result is good? — but it’s potentially tricky to share
synthesis instruments. projects with other users if you aren’t sure what instrument packs they own and which ones you are using.

information Subtle Changes

Live Suite 8 £599.99;
Live 8 £399.99; upgrades Thankfully, Live 8 looks and works almost identically to Live 7. I’m running both versions here, and found
from version 7 myself launching the wrong one by accident a few times and taking a minute or two to actually notice. It’s
£179.99/£159.99 tempting to feel a bit swindled — a major, paid-for upgrade and it all looks the same? — but the
respectively. Prices include improvements are non-trivial, and the lack of upheaval to one’s working process is actually a plus point.
VAT.
Focusrite +44 (0)1494 The audio file warping and grooving machinery has been given something of an overhaul in Live 8. Live’s
462246. ability to match up the beats and transients of recorded audio to the playback tempo of a sequence is one of

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Ableton Live Suite 8

Click here to email its most prestigious features. Live attempts some aspects of this itself, but others require guidance, either to
www.focusrite.com correct mistakes or to add some creativity into the editing and synchronisation process. The warp marker
www.ableton.com editing process is sufficiently different between Live 7 and Live 8, and sufficiently subtle, that it’s worth a
recap of how Live 7 supports warp editing, so that we can see what now happens in Live 8.
Photos too small? Click In Live 7, when a sample file is dragged into a Live Set as a new clip, the program attempts to determine
on photos, screenshots and the sample’s tempo and its length in bars and beats, so that it can be looped and played in sync in any
diagrams in articles to open desired tempo. One warp marker is placed at the beginning of the sample, and a second marker with the
a Larger View gallery. appropriate bar and beat number is placed at the end. If the rhythm of the sample is regular, then the bar and
beat divisions between the warp markers will fall on the sample’s beats. Additional warp markers can be
dropped onto bar and beat lines, and markers can be dragged back and forth along the waveform; bar and
beat lines between adjacent warp markers are scaled linearly.
In Live 8, things get a bit more complicated. Live attempts to identify the transients in a sample and marks
them on the waveform display as prospective warp markers. Transient markers can be added, deleted or
moved if Live has not been totally accurate in its analysis. A transient can be turned into a warp marker by
double-clicking, although warp markers can also be created on bar and beat lines just as in Live 7.
Dragging a warp marker in Live 7 shifts the marker relative to the waveform, and Live changes the audio’s
playback rate on each side of the marker, so that the selected part of the audio is synchronised to the beat.
Holding the Shift key while dragging does the same in Live 8, although in the new scheme the waveform
display moves while the timeline stays linear: same outcome, but rather disconcerting behaviour until you get
used to it. So what about simply dragging a warp marker? This moves the marker and its audio sync point to
a new quantised bar/beat division, stretching or compressing the audio on both sides. This is an editing
operation with no direct equivalent in Live 7, and appears to be the machinery Live uses when applying
groove templates, although you are free to employ it for your own creative ends.
Live 8’s groove features have also been significantly overhauled. Live 7’s groove functions were rather
elementary: it was only possible to select one of three swing resolutions for each clip and then set a global
swing amount for the entire Set. Live 8 comes with a library of groove templates, which can impose a variety
of new timing and volume settings on a clip. For MIDI clips, note events are shifted on the timeline and
velocities are modified, whereas for audio clips the audio content is warped and a clip envelope is created to
automate the gain levels. Applying a groove to a clip is non-destructive — a groove is just a timing and
volume template applied in real time as a clip plays — so grooves can be auditioned and hot-swapped
without risk of damage to the underlying material. If you want to make a groove’s influence permanent, you
can ‘commit’ it to its clip, causing the timing and volume changes to be applied as an edit.
Grooves have parameters which can be modified — active grooves are stored in a ‘groove pool’ in the Live
Set. It’s not possible to see the actual timing and volume parameters, but the time division, quantisation,
timing strength, randomness and velocity scaling can all be fine-tuned (or coarse-tuned, for radical effects).
I have to say I’m not a great fan of swing time in my sequenced electronica sets, but the new groove
machinery is sufficiently flexible that it can be used in all sorts of ways to subtly vary or randomise a lot of
otherwise inflexible material. Even with the swing timing disabled, it’s possible to slightly randomise the note
timing in MIDI percussion clips, to inject energy into otherwise lifeless arrangements.
Lastly, and most intriguingly, Live claims to be able to extract groove templates from existing MIDI and
audio clips. It’s hard to tell exactly what this involves, especially since it’s not possible to inspect the contents
of a groove, and my brief experiments with audio clips were rather inconclusive, but groove extraction from
MIDI clips seems to be a predictable way to create customised grooves from scratch or to match existing
material.

Track Grouping
For a while now, Live has supported composite devices
constructed by nesting and layering simpler ones: an Instrument
Rack enables several instruments (or chains of instruments and
effects) to be played and controlled in parallel, and a Drum Rack
maps each instrument in a rack to its own trigger and MIDI key.
Whenever a mixer channel contains a Rack, that channel can be
expanded in the mixer view so that each component of the rack
has its own sub-channel.
In Live 8, this kind of parallel nesting of channels can be done
directly in the mixer. Two or more tracks can be grouped together
under an enclosing group track. Each of these sub-tracks can
route its own output or send audio into the group track itself — this

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Ableton Live Suite 8

enclosing track has its own mixer controls and can host effects,
allowing it to act as a submixer.
The enclosed tracks are first-class citizens: unlike rack chains,
they have their own individual clips. In the Session view, the group
track shows clips as cross-hatched areas when several enclosed
tracks have clips in the same scene; the clips launch as a unit. In
the Arrangement view, the group track has a thumbnail view Three tracks in a group, with a
showing clips from the sub-tracks in the timeline. common submix.
Three tracks in a group, with a common
Cross Clip Cleverness submix.
Another improvement in the area of arranging is the addition of
programmable crossfades between adjacent audio clips on the same track. In the Arrangement view,
crossfade curves can be dragged and reshaped at the boundaries between clips, and, rather cleverly, the
waveform is actually redrawn to reflect the effect of the fade. Even for isolated clips, it’s now possible to edit
the in and out fade curves without having to mess around with volume automation.
The MIDI editing pane (in both the Session and Arrangement
views) has been improved. The pane now has an editing cursor, so
MIDI events can be pasted at any desired location, and there is
finally a step-record function: the cursor can be nudged back and
forth with the arrow keys, and if the track is record-enabled, any
notes held on a keyboard will be recorded into the clip at the
cursor location as the cursor moves. Crossfading between two clips in the
Arrangement.
In earlier versions of Live, a VST or AU plug-in’s parameters
Crossfading between two clips in the
would be made available in a panel in the track’s device chain, so
Arrangement.
that specific parameters could be attached to MIDI Controllers or
selected for automation. This worked well enough for simple plug-
ins, but many soft synths have hundreds of parameters, occupying a long strip of panel space and making it
difficult to find that very particular oscillator or filter setting. Live 8 still has a panel for parameters, but for
complex plug-ins the panel starts off empty, and there’s a configuration mode in which Live is taught which
parameters to present there — they are selected by pointing and clicking directly in the plug-in’s interface.
These parameters are also the only ones presented for automation in the Arrangement view, as well as for
clip envelopes, so a huge amount of irrelevant detail — all the parameters which are set once, purely in the
preset — are hidden from view. This is a simple but marvellous piece of streamlining, which focuses attention
on those things that change, by removing from view those that don’t.

Instrumental Developments
Live’s venerable four-operator FM synthesizer, Operator, has received an upgrade, although at first view its
on-screen appearance is little changed. There are actually slightly fewer built-in oscillator waveforms than
there were in the previous version, but Operator more than makes up for this by allowing you to draw your
own! A new oscillator pane presents a graph of editable waveform partials, while a thumbnail shows the
resulting waveform. The built-in waveforms are just presets in the partial editor. This doesn’t exactly
transform Operator into a full-on additive synthesizer, but it seriously beefs up its power without interfering
with its user interface. There are also additional filter types, and the filter response curve can now be viewed
and edited graphically. New control options have been added — MIDI Controllers and values can be routed
into the voice architecture via a small modulation matrix — and for added wackiness it’s possible to select
different FM algorithms at any time via automation or MIDI control, even while notes are playing.
Collision, meanwhile, is new for Live 8. It’s a physical modelling synthesizer, where mallet and noise

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Ableton Live Suite 8

oscillators feed into a pair of modelled resonators. Collision excels at modelling instruments like vibraphones
and glockenspiels, producing sounds that are clear, responsive and organic. There are also some
surprisingly good analogue synth bass patches, and the plucked guitars have a nice sense of life to them.
There are even some lovely piano presets, which I would have sworn were sample sets before examining
them more closely. The cost, though, is CPU load, which can be quite hefty with certain choices of resonator
algorithm.
My personal experience of modelling synthesis, starting with the Yamaha VL1 many years ago, is that the
process of sound programming is usually pretty opaque unless you’re the type of person who wears a white
coat and carries a slide rule, but Collision actually presents a fair chunk of its architecture in a clear and
accessible manner: the noise source has a conventional filter and envelope, and the resonators have
controls that make sense after a bit of consideration (and that have descriptive hints in Live’s Info View
pane). I found that I was able to constructively alter resonator settings without breaking the preset, wrecking
the tuning or or blowing up my speakers. I don’t think Collision will ever be my ‘go to’ instrument for bread-
and-butter synthesis, but it does present physical modelling in a demystified manner, which I applaud.
Bundled with Collision is an audio effect named Corpus, which is roughly equivalent to one of Collision’s
resonators with its own dedicated LFO. Corpus can be tuned by MIDI note number, and offers a variety of
rich effects treatments that can be applied to conventional instruments and samples in order to add a bit of
physically modelled ‘pixie dust’ to a sound.

Looping The Loop

Live 8 features a real-time looper, similar in intent to the footpedal
and rackmount loopers that are currently in vogue as performance
devices. The Looper ‘instrument’ is actually an audio effect, since it
The new Collision instrument feeding
generally needs an audio feed (which can, of course, be from an
the Looper effect.
audio clip, soft synth or effect), although you can drag existing
The new Collision instrument feeding
audio clips directly into Looper’s buffer, as well as exporting loops
the Looper effect.
as new clips. (Looper also saves its current buffer with the Live
Set.) Looper is designed to be operated by a MIDI footswitch, so it
has a big multi-mode button that toggles between record, play and overdub mode and provides single-level
undo and redo of recording passes, separately from Live’s global undo/redo stack. Looper’s play/record
mode can also be automated to allow it to be operated as part of an arrangement mix, although for some
reason the mode can’t be assigned to a rack’s macro controls. The maximum loop length is not documented,
but I dropped a two-hour clip into Looper’s buffer and it seemed to want to loop it after about 40 bars.
Looper can reverse its buffered audio and vary playback speed by up to three octaves up or down, but
that’s about the extent of its abilities. The playback speed control (which is also used for matching loop
tempo to the Live Set) does not feature audio warping, so if the speed varies, so does the pitch.
Compared to other looping systems, Looper is rather basic, but that rather misses the point: since Looper
is really just one of Live’s audio effects, there’s nothing to stop you using multiple Looper instances, and all
of the other machinery Live provides, to build arbitrarily large and sophisticated looping engines to do almost
anything you want. Looper is best viewed as a building block, rather than a complete instrument in its own
right.

Bring On The Vocoder

Perhaps not before time, Live 8 has its own vocoder effect. I’ve been a fan of vocoders since the days of the
Korg Wavestation, when I would spend far too much time modulating pad sounds with rhythmic wave
sequences. Thanks to Live 8’s Vocoder I can revisit a whole set of techniques I’ve not used for years.
Vocoder can run up to 40 bands in mono or stereo, with parameters for filter bank range and bandwidth, and
a formant control to shift the filter frequences as a unit. The filter bands can also be gated. A graphical
display allows the gain of each band to be individually set, and also shows the audio level in each band.
As an audio effect, Vocoder takes the modulator signal as input. The carrier can be tapped as an audio
signal from anywhere in Live’s mixer, or it can be a variable-density noise generator, or it can be the
modulator itself. Vocoder is clearly laid out and easy to use, running with a small CPU footprint, and does
exactly what it says on the tin.
Live also now sports a combination frequency shifter/ring modulator. One’s imagination immediately drifts
towards unearthly sound effects and early modular synthesis, and these kinds of application are indeed
catered for, but there are subtler applications of such a device: a slight frequency shift can produce a sweet
flanging effect, whereas a low-frequency ring-modulation induces a gentle tremolo. There’s also a place for
more extreme treatments, such as altering the tone or tuning of individual drum samples in a drum rack.

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Ableton Live Suite 8

Sample Libraries
The boxed version of Live 8 comes with a new release of the Essential Instrument Collection, EIC2 for short.
The collection contains acoustic and electric keyboards, strings, brass, woodwinds, plucked instruments
(harp, guitars, basses), mallets, choirs and drums: a good coverage of orchestral and band instruments. The
EIC2 presets are more than just collections of multisamples, as they have been assembled using Live’s
device-chaining machinery and feature multiple nested zones and layers, often with internal audio filters, and
always with easy-access macro controls allowing fast selection or editing of important aspects of the sound.
The EIC2 instruments are actually shipped in multiple versions, balancing fidelity against computer
resources, so that musical ideas can be sketched out using the low-resolution instruments and later
recorded with all the keyboard and velocity zones using 24-bit samples.
Suite 8 also ships with two whole DVDs of Session Drums, again constructed as sophisticated chains of
instruments and effects. The individual multi-miked drums are actually layered Live instruments carrying
distinct samples of the signal recorded from stereo overhead and room mics, and these signals are cleverly
routed back into local submixes, allowing for remixing from the original microphone inputs.
Finally, there is a library of Latin percussion kits, covering a range of exotica from agogo to wood block. I
was quite taken with the bells and chimes, and the timbale rolls are just a little bit too much fun.
With all of Ableton’s drum kits, the mappings from keyboard notes to drum parts is standard, so it’s easy to
assemble customised kits drawn from different presets without too many clashes across the keyboard. All the
drum libraries are packaged with a large number of Live Sets containing MIDI clips of drum loops using
various kits. The clips can be browsed and auditioned within Live and then dragged into a session as starting
points. I’m used to auditioning sample CDs, so had to remind myself that these clips are fully editable MIDI
sequences driving instrument racks which are themselves editable. The scope for creative exploration is
immense.
Ableton Suite 8 also ships with the existing set of instruments: the Sampler multi-zone sampler with filter,
Analog twin-oscillator modelled analogue synthesizer, Tension physical modelling string synthesizer, Electric
modelled electric piano, and 500MB of sampled drum machines.

Conclusions
Live 8 is the fourth major revision of Ableton Live that I’ve used, and, contrary to the usual software upgrade
practice, the environment still gives an overriding impression of stability. Each major revision delivers a
handful of important enhancements to the core package, but otherwise the process is one of minimal
upheaval and low disruption. It seems that Ableton are now looking more to adding value to the Live
environment, with a succession of new instrument releases and an ever-growing selection of sample sets
and loops. The full Ableton Suite is such a comprehensive package that it’s possible to imagine entire
production projects using it exclusively, without any additional instruments or effects.
The on-line sharing feature (see ‘Share And Share Alike’ box) has a lot of potential, but the amount of
mileage in that really depends on the effort that is put into the web experience and the support given to the
Ableton user community that exploits it. Finally, we have Max For Live just round the corner, and I can
absolutely guarantee that once the Live and Max systems — and user communities — intersect, there will be
a total rollercoaster ride ahead!
All in all, Ableton have succeeded in producing an upgrade which adds considerable value to the Live
environment and also points to an exciting future.

Other Enhancements
As well as the major changes described in the main text, Live 8
ships with a number of other additions and enhancements:
New effects include a multi-band dynamics processor (with
visual feedback), a brick-wall limiter, and an overdrive effect.
Live’s display can now be zoomed between 50 and 200
percent of its normal scale: the window size is unchanged but
all its contents are resized accordingly.
MIDI clips and audio files can be scrubbed and previewed in
the file browser using a miniature waveform display.
Mixer settings can be altered for multiple selected tracks at the
same time.
Tracks and scenes can now be assigned colours, as can
A collection of shared Live Sets and

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Ableton Live Suite 8

chains and macro controls in instrument racks.

the beginning of an upload.
A collection of shared Live Sets and the
beginning of an upload.

Share And Share Alike

Live 8 marks a push into the world of on-line musical collaboration and sharing of work and ideas. On-line
collaboration isn’t new, of course, and neither is uploading and downloading of music and data files, but
this is the first attempt by a music software company to integrate on-line sharing directly into a production
package. A Live Set can be made public, or shared with selected users.
All file sharing takes place via Ableton’s servers (in fact, the files are stored in Amazon’s S3 server
cloud) rather than being peer-to-peer, so collaborators need to set up on-line accounts with Ableton. The
release notes talk about “intelligent” file transfer — only changed files are transmitted each time, which is
obviously good news when working with large projects containing a lot of audio material — but the sharing
scheme is really a variation on traditional uploading and downloading, rather than something smart
enough to synchronise projects between multiple users (a process that is notoriously difficult to get right).
Uploading is initiated in Live, which launches a web browser pointing at the Set’s web page when the
upload is done. Downloading is initiated from a web page containing a special ‘ableton:’ URL which
transfers control to Live to do the actual downloading. At the moment the sharing service is free, but
Ableton have not ruled out charging for it in future.
Each upload of a Live Set results in a new copy, as does each download, so collaborators are
responsible for keeping track of the versions of Sets they transfer. Downloaded Sets are stored in a
temporary directory, so you are also responsible for saving the project somewhere permanent. In
practice, I found that Live tried to be rather too clever about locating audio files from Sets that had been
through a few upload/download cycles, so the default action of collecting and saving all referenced files
into the current project seems a wise approach.
Since there is no guarantee that all collaborators will have the same selection of MIDI sound sources
and effects — hardware or software — Live offers the option of freezing any tracks that use internal or
external instruments prior to upload. When a set is downloaded, tracks can be unfrozen if the recipient
has the appropriate software instruments, or else left frozen until the track is back in the hands of its
original creator.
The sharing process within Live is neatly integrated — a dedicated file-browser pane shows the
progress of file transfers — but the web experience is, at the time of writing, a little primitive. A user’s Live
Sets are listed in order but there is no exact note of the time of the upload, which is awkward when the
same Set can be uploaded several times under the same name. An uploaded Set cannot be renamed or,
for that matter, deleted, and the on-line experience is rather impoverished compared to other audio
sharing services such as Soundcloud. However, it’s early days yet — according to Ableton, the service is
still in beta — so I would expect it to improve over the coming months.
One wrinkle I discovered is that Live refuses to upload audio files in Sound Designer II format, such as
those created by MOTU’s AudioDesk — probably because such files have a Macintosh-specific internal
resource format. Since Live does not provide any means to convert files between different formats, Live
Sets containing such files cannot be shared.
It might also be worth perusing the small print of the licences of any sample libraries that you’re using,
since uploading and making available Live Sets containing copyrighted samples probably violates
numerous bits of licence agreement.

Maximum Potential
The most intriguing, and almost certainly the most powerful, enhancement for Live 8 is one that, alas, isn’t
shipping yet, despite having its own chapter in the manual. Max For Live combines (or will combine)
Ableton Live with Cycling 74’s Max/MSP audio and media construction kit, allowing Max/MSP programs
(‘patchers’ in Max parlance) to exist inside a Live Set. Max/MSP can already export VST plug-ins to run
inside most sequencers via a run-time system called Pluggo, but Max For Live is a whole new ball game.
Not only will fragments of Max inhabit Live’s window directly, looking virtually indistiguishable from Live’s
built-in instruments, but these embedded Max patchers will be editable: they will be MIDI and audio plug-
ins which can be taken apart, modified and rebuilt directly in the Live environment. All we know so far is
that Max For Live will be an additional product rather than a bundled component of Live, and that we can
expect to see it later this year.

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Ableton Live Suite 8

Published in SOS June 2009

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AEA A440

In this article: AEA A440

A Brief History Active Ribbon Microphone
Lesson
Published in SOS June 2009
The A440 Reviews : Microphone
Print article : Close window
Advantage
On Test
Some microphones ooze quality to the extent that price almost becomes
Verdict
unimportant. Is AEA’s new active version of the venerable RCA 44
AEA A440 £6348 design one of them?
pros Hugh Robjohns

T
Silky-smooth and fabulously
here is no doubt that ribbon mics are finally making a
warm character.
comeback in both the professional and project studio
Sensible output level.
Ribbon is fully protected from markets. More robust ribbon materials, stronger magnet
phantom power. structures, better noise performance in high-gain preamplifiers,
Class-leading low self-noise and lower-cost far-eastern production have all contributed to this
figure. resurgence — and I for one am very pleased to see it, because
Traditional R44 styling. ribbon mics generally have an inherent smoothness and natural
cons quality, particularly at the high end, which no other microphone
Mortgage-worthy expense. topology can match.
You’ll also need to budget for
Amongst the longer established high-end ribbon mic
an Atlas mic stand to hold it.
manufacturers, Wes Dooley — the man behind the US’s Audio
summary
Engineering Associates (AEA) — is one of the best known and
A new generation of
most vocal advocates of ribbons. Indeed, a large part of his
recording engineers has
professional life has involved refurbishing and maintaining vintage
discovered the sonic beauty
of the ribbon mic, and the ribbon designs, and the progression to developing an impressive
A440 represents the pinnacle line up of his own designs was inevitable. I’ve been very
of that particular mountain. impressed with AEA’s mics, to the point that I own (treasure might
All the sonic mastery of the be a better word) a couple of their R92 ribbons myself.
original RCA44, but with a far
The latest addition to the AEA fold is the A440, which is an
more practical output signal
level, whilst retaining a class- interesting revision of the huge vintage RCA 44 (and its current
leading low noise floor. AEA variants), with a pleasing name-play on a familiar musical term. It looks identical, in fact, but Wes and
his team have built a high performance head-amp into the mic case to provide a much stronger output
information signal.
£6348 including VAT.
Affinity Audio +44 A Brief History Lesson
(0)1923 265400.
The original, instantly recognisable RCA 44 ribbon mic was introduced in 1936, and has been revered ever
Click here to email
since, along with its junior stablemates like the 77 and KU3, many of which are still in daily use in film-
www.affinityaudio.com
scoring and classical music studios around the world. Towards the end of the last century, it was becoming
www.wesdooley.com
hard to get parts to maintain these classic mics, and that provided the impetus for AEA to re-start production
Photos too small? Click to the original specs, launching these new incarnations as the R44C and R44CX models. These recreations
on photos, screenshots and use the same 1.8-micron aluminium ribbon design as manufactured for RCA, and many of the parts are
diagrams in articles to open completely interchangeable with original RCA 44B and 44BX models. However, the output sensitivity of the
a Larger View gallery. standard model is a lowly 2.25mV/Pa (just like the original) and that places serious demands on the preamp,
particularly when the mic is being used for distant placement. Even the high-output CX model only manages
a feeble 5mV/Pa. No self-respecting capacitor mic would dare be seen with less than 12mV/Pa on the spec
sheet, and most are around the 20-25mV/Pa mark.

The A440 Advantage

Thankfully, the advent of phantom powering and miniaturised solid-state electronics have allowed this
common weakness of ribbon mics to be addressed, and there are already several ‘active ribbon’ mics on the
market taking the same approach for the same reason. The AEA A440 joins the fray by harnessing the
proven J-FET amplifier technology used in the company’s TRP preamps to the large-ribbon R44C
mechanics. The result is a very healthy sensitivity of around 30mV/pa — higher than most capacitor mics —
yet retaining a self-noise floor of just 6dBA, which equals the best of any current electrostatic design. The
internal head-amp requires standard phantom power, of course, and also ensures the ribbon is fully
protected from phantom power — something that’s a perennial concern for so many ribbon mic users.
Like its antecedents, the A440 maintains the instantly recognisable gigantic angular shape, with the huge,

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AEA A440

punched grille and slightly utilitarian-looking grey and silver finish. It is just as heavy as the R44C, at around
3.45kg, and just as big, at 31.25 x 11.7 x 84 cm (HxWxD). You’ll have to invest in a seriously heavy-duty mic
stand to support this monster safely. Like the other AEA variants of this design, the mic resides
(permanently) in an adjustable cradle frame with an integral stand adaptor-cum-shockmount, and the output
is via a captive cable terminated in an XLR. However, as the A440 is custom-built to order, any required
cable length can be accommodated, and the case can even be laser-engraved with a corporate logo.
Again, like the similar R44 models, the A440 is shipped in a bespoke and tough cordura case, lined with
protective foam, and complete with a zipped cotton bag to protect the mic once the original plastic packing
bags have been removed. It’s important to keep all ribbon mics covered when not in use, and packed away
vertically to prevent ribbon sag.

On Test
I reviewed the AEA R44C in the pages of SOS back in June 2002, and comparing my notes from back then
with my listening tests on the new A440, the family character remains clearly in place. As expected, the
A440 exhibited the perfect figure-of-eight polar response, with identical tonality front and back. The side
nulls are wonderfully deep and well defined, but across the wide frontal and rearward working angles the
sound quality remains remarkably consistent. The mic has been specified with a maximum SPL of 136dB (for
one percent THD), and although that is 30dB lower than the passive version it is still more than enough for
most applications (presumably, the internal preamp’s dynamic range is the limiting factor here).
The frequency response is very similar to the original model. In fact, it’s smoother, if anything, but with the
same characteristic 2dB/octave droop, falling about 10dB between 300Hz and 7kHz, above which the roll-off
steepens. The bottom extends powerfully to 20Hz and below, producing a rich, warm and smooth-sounding
mic, although one that still retains precise HF detail and clarity. The proximity effect is strong, of course,
becoming quite evident for sources closer than a couple of feet, but that can be used creatively when
appropriate.
Placing the A440 slightly to the side and a few feet from the bell of a trumpet captured a very vivid and
lifelike performance, and I needed almost 30dB less gain than for my own AEA R92 placed alongside
(feeding my AEA TRP preamp). On acoustic instruments like acoustic guitars the A440 revealed a fast,
detailed character, with a warm upper-bass response, and a rich, smooth treble, which is musically
complementary and easy on the ear. Twelve-string guitar was also captured without any harshness or signs
of intermodulation distortion, while stringed instruments — and orchestral strings in particular — sounded
fantastically smooth and natural.
But the different design philosophies of the A440 and R92 become very obvious when the mics are
compared side by side. The former is clearly designed for relatively distant placements, and sounds optimally
balanced when at least five or six feet away. The R92 gives its best when within about two feet. Given a
scoring stage or orchestral venue, the A440 can be used to advantage, but for close working in a project
studio, the R92 might prove the more practical choice.
The deep side-nulls of all ribbon mics can be used to enormous advantage when it comes to rejecting
unwanted spill, but the rear lobe pickup must always be remembered, especially when the rear lobe has the
same frequency response as the front lobe, as it does in the A440 (unlike the R92). Of course, this identical-
sounding rear pickup can also be very useful, for example for capturing a second performer on the same
channel (balancing levels by moving the mic between the two players). If the rear lobe isn’t required,
something like an SE Reflexion Filter placed behind the mic can be very effective in reducing the amount of
reverberation or spill captured.

Verdict
There can be no doubt that this is an impressive, imposing and fabulous sounding microphone, and the
addition of the built-in head-amp makes it far easier to match it with a suitable preamp. But it is really only
suitable for use in a large, good-sounding room, where distant placement is appropriate. And at the current
price, this is one seriously expensive mic that will be the province of only the few remaining high-end
professionals, or perhaps ‘resting’ city traders, indulging their musical hobbies while waiting for a stock
market revival!
As with the other RCA44 variants, though, it is enough for us cash-starved mortals to know that such
fabulous mics are still available, and easier to use now than ever before — and that they still sound utterly
wonderful. When I reviewed this microphone’s forebear, I suggested that it was like a vintage Bentley:
expensive and revered, but not easy to justify to the accountants. The same is true of this one, only it’s now
been fitted with a supercharger! Unfortunately, once you’ve heard this mic, you’ll always aspire to own it.

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AEA A440

Published in SOS June 2009

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Arturia Origin

In this article: Arturia Origin

Getting To Know Rackmount Synthesizer Workstation
You
Published in SOS June 2009
Building A Synth Reviews : Sound Module
Print article : Close window
More Than Just A
Synthesizer
Arturia’s first true hardware synth aims to let you conveniently create
Multi Mode
your own modular system from modelled components of some of the
In Use
best-loved analogues of all time. How well does it realise this ambitious
Conclusions
goal?
Vintage Templates
The Software Gordon Reid

I
Connection magine being able to take the oscillator and filter modules from
Physical Highs & synthesizers such as the Minimoog and ARP 2600 and
Lows converting them into self-contained software modules. Next,
Modules & Factory throw in a bunch of generic control modules and design a
Sounds framework into which these can be inserted and cross-connected,
and house the whole thing in a powerful signal processor with all
Arturia Origin £2238 the hardware needed to program and use the thing. Voilà, mes
pros amis — the Origin!
Generates a rich, involving
sound. Getting To Know You
Offers fabulous modulation
capabilities. The Origin is more than just a synthesizer. Like many modern
The effects are truly instruments, it provides a Program mode for building sounds, plus
multitimbral. Hurray! a Multi mode for combining them in splits and layers, and for multitimbral use. There are effects processors
cons accessible in both modes and, although there’s no multitrack digital sequencer on board, there is an
The external PSU would be analogue-style sequencer and arpeggiator. Consequently, the Origin is best described as an ‘analogue style’
hard to replace. workstation.
The manual is rather lacking Physically, it’s a substantial lump. With more than 50 rotary encoders, 12 of which can be used as push-
in places. button selectors, more than 80 buttons, a spin wheel (which is also a push-button) and a joystick, it needs to
It’s not cheap.
be. To be fair, its size and the number of physical controls is a blessing, because it reduces the amount of
summary farting around that would otherwise be necessary to create and manipulate sounds. Having said that, the
The Origin is a unique Origin still demands a considerable amount of farting around, especially to build and refine new synthesizer
instrument, combining architectures from scratch. Remember, this is a modular synth, so you have to configure new ‘instruments’
modular virtual analogue
before you can create sounds with them. To illustrate this, let’s build and program a simple synthesizer
synthesis derived from
imitative soft synths with architecture.
modern features such as
splits, layers and truly Building A Synth
multitimbral effects. It sounds
excellent and, if you can The Origin interface offers eight primary editing pages (most of
afford it, you should take a which have additional sub-pages), accessed by the eight buttons
close look, because there’s that run along the underside of the screen. These are the Home
nothing else that does quite page, followed by the Preset, Program, Edit, Multi,
the same thing. Sequencer/Arpeggio, Effects and Live pages. To create a new
synth from scratch, start by entering the Preset page and selecting
information
the Empty Program option. This then takes us to either the
£2238 including VAT. Edit/Patch or Edit/Rack sub-screen, depending upon which was
2twenty2 +44 (0)845 299
used most recently. We’ll start in the Rack screen, which offers a
4222.
representation of a modular synth with three rows of eight slots,
Click here to email
only two of which are populated: one with a keyboard control
www.2twenty2.com
module and the other with an output module.
www.arturia.com
We can now insert an oscillator by pressing Add, scrolling to the oscillator list, and selecting from five
Photos too small? Click options: Origin, ARP 2600, Yamaha CS80, Minimoog and Jupiter 8. There’s a sixth option — a Wavetable
on photos, screenshots and oscillator — but this, for reasons that are beyond me, is presented separately. For the sake of argument, let’s
diagrams in articles to open select a Jupiter 8 oscillator. No matter where you have placed the cursor within the 8x3 grid, the oscillator will
a Larger View gallery. now appear (sensibly) between the keyboard and the output modules.
If we now press a key on an attached MIDI controller keyboard, no sound ensues. This makes sense. Like
a true modular synth, the oscillator is in position, but it’s not connected to anything, so we need to patch the
keyboard controller to the oscillator’s FM input, and the oscillator’s audio out to the input of the output

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Arturia Origin

module.
If this were a true modular synth, there would now be a continuous drone, but there isn’t. Instead, if you
play the controller keyboard, the sound is produced with a square ‘organ’ envelope. This means that there is
an amplifier in the output module, controlled, in the absence of an envelope, by the keyboard’s Note On and
Note Off messages.
Let’s now add a second, slightly detuned oscillator by repeating
the previous operation and making the appropriate connections.
Well, we can’t; we have to add a mixer and patch the two
oscillators to its inputs, then direct the mixer’s output to the Output
All of the Origin’s I/O connections
module. This, again, makes perfect sense but, strangely, the mixer
are found on the recessed rear
does not locate itself between the oscillators and the output
panel.
module. This is not an impediment to programming, but it’s
All of the Origin’s I/O connections are
confusing. Happily, we can obtain a more pleasing arrangement by
found on the recessed rear panel.
jumping to the Edit/Patch screen, which offers a 6x5 matrix of
modules within which we can move the modules to any desired
position. (That’s weird — one representation offers 24 modules, the other offers 30!) Selecting the mixer in
this view, we can move it and connect it as desired. The patch now shows its interconnections on-screen,
with differently coloured lines to differentiate between different types of signal (audio, CVs and so on).
We can now continue to add modules, creating complex audio paths and cross-patching modulation
sources pretty much at will. What’s more, unlike most modular synths, the Origin’s modules allow us to direct
single audio signals and CVs to multiple destinations, and their inputs can receive from multiple sources.
This is equivalent to having a ‘multiple’ on every output and a mixer on every input, vastly increasing the
Origin’s patching capabilities.
Avoiding the temptation to go wild, I completed this simple architecture by adding a Jupiter 8 low-pass
filter, from the resonant high-pass, low-pass, band-pass and notch filter options, adding an envelope
generator and, finally, an LFO. I then removed the pressure sensitivity that was engaged by default, created
a suitable filter sweep and added a little pitch modulation. The result was warm and engaging, and on many
levels was no different from a large analogue synth. I was impressed.

More Than Just A Synthesizer

To the far right of the panel, you’ll find the Origin’s three effects
slots. Each can host the same five algorithms: chorus, delay,
reverb, distortion and a dual phaser, each with MIDI sync where
appropriate. Routing (accessed via the Program/Mixer page) is
sophisticated, with series, parallel and independent (ie. per output)
options and, with certain exceptions, you can place any three of
the effects in the slots you require. However, there’s nothing
sophisticated about the algorithms themselves; they are of the
stomp-box variety so, if you want exotic and power-hungry effects,
you should use the independent audio outputs provided by the Origin. These allow you to direct each sound
to the external processor(s) of choice.
Running along the whole of the bottom edge of the Origin, you’ll find the 16-step sequencer — except that
it isn’t. Although only one row of 16 knobs is provided, the sequencer offers three rows of sequencing, each
with up to 32 steps. There are lots of facilities on offer — quantisation, swing, accents and so on — and
you’re not limited to creating note sequences; you can direct the sequences anywhere within the patch. So if
you want to direct sequence 1 to the pitch inputs of two out of four oscillators as well as the pan and the rate
of LFO 1, connect sequence 2 to the filter FM, the LFO 2 rate and the envelope Attack rate, and sequence 3
to, oh, anything else you want, you can do so, just by adding connections.
There’s also a simple arpeggiator. This follows the Roland model, with up, down, up/down and random
modes across one, two, three, four or five octaves. If you’re after the fairy-dust sounds of arpeggiated Jupiter
synths, you can have it. Alternatively, crank a bit of distortion into a Minimoog patch and run the arpeggiator,
and ELP’s ‘Karn Evil 9’ is never far away.
Pressing the ‘Progr’ button takes you to a set of five pages, one
of which (the FX Mixer) I’ve already mentioned, and the other four
of which are devoted to advanced modulation duties. The first of
these is a five-stage, two-dimensional envelope generator.
Originally developed as the basis of Sequential’s Vector Synthesis,
this provides a phenomenal amount of control, and careful
programming can yield some remarkable results.

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Arturia Origin

The next page contains a version of Arturia’s Galaxy modulator.

When I first reviewed this on the Jupiter 8V I was a little sceptical
about it, but I soon found that it had a unique ability to generate all
The Home page allows you to
manner of cyclic and/or discontinuous modulations. Inevitably, the
browse through, select and preview
majority of effects you obtain from this are, well, just effects, but
patches. It is also the starting point
with a bit of planning it too can create some remarkable sounds.
from which — by pressing the
The other two pages contain global LFOs. These offer the System button — you enter the
standard waveforms, plus niceties such as waveshaping for the Origin’s extensive editing system.
triangle and square waves, sync, delay and fade-in time. There’s The Home page allows you to browse
lots of control over the parameters and, as always, the output can through, select and preview patches. It
be directed to multiple destinations. Simple? Yes, but not limited. is also the starting point from which —
Before moving on, I also have to give special mention to the by pressing the System button — you
enter the Origin’s extensive editing
Joystick Mixer. Similar in concept to the 2D envelope generator
system.
(and, again, descended from the Prophet VS) this allows you to
assign up to four audio inputs from anywhere in the synth, add two
modulation sources, and then direct the output to multiple destinations. Control over the sound is then
provided using the joystick, or by triggering another two-dimensional contour. Again, the amount of control
can be astounding.

Multi Mode
Having created an architecture and programmed some sounds,
Multi mode allows you to access up to four of them
simultaneously. Each program can be allocated its own MIDI
channel, transposed by up to +/- 24 semitones, and restricted to its
own range of MIDI Note numbers so, in addition to creating splits
and layers, this mode is perfectly suited to four-part multitimbral
duties.
It was while investigating this that I really started to appreciate
the Origin. Firstly, I discovered that edits made to Programs in
Multi mode could be saved when saving the Multi itself. (This might
The Edit/Patch screen allows you to
seem a subtle point, but it’s a huge benefit when compared with
insert and remove modules, to move
synths that demand that you edit and save the constituent
them around into sensible
Programs independently from the composite whole.) Then, quite by
configurations, and to see what’s
accident, I layered one of my arpeggiated Minimoog programs with
connected to what. This gives you
one of the factory presets, a slow pad called ‘Seqpadbell’. Jean
the best view of the architecture of
Michel Jarre would have sold you his granny for this combination,
your virtual modular synth.
and I suspect that, with a bit of haggling, you could have got
The Edit/Patch screen allows you to
Charlotte Rampling thrown in, too. insert and remove modules, to move
Next, I decided to make the Origin crash, so I added a third, them around into sensible
CPU-hungry polysynth pad to my Multi. This comprised a dual- configurations, and to see what’s
oscillator synth played through a heavy phaser sweep. The Origin connected to what. This gives you the
best view of the architecture of your
did not crash, but the mildly distorted arpeggio continued to be
virtual modular synth.
mildly distorted, the Seqpadbell patch continued to be delayed,
chorused and reverberated, and my new strings pad continued to
be phase-shifted. In other words, Multi mode is genuinely multitimbral, and the patches that you insert are
reproduced with all their effects intact. I can’t tell you how happy this makes me! Well done, Arturia.
Still trying to crash the Origin, I inserted an analogue drum sequence into the fourth and final slot. Don’t be
misled, the Origin does not have any analogue drum sounds other than those you program, and it doesn’t
have a drum sequencer. Nonetheless, you can do quite a lot with the existing sequencer, so I tried to see
whether I could do the whole Oxygene thing in a single Multi. Amazingly, I could, and still the Origin refused
to hiccup, let alone fall to the floor in a pile of over-stressed digits. The only thing I noticed was that I had hit
the CPU limit, because the polyphony — which has a maximum of 32 — had dropped to just four. I might

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Arturia Origin

have been able to lessen the CPU load by replacing the somewhat greedy Minimoog and Jupiter 8 modules I
had used with the more abstemious Origin modules, but no matter — by this point I was simply having fun.

In Use
Arturia make a big deal of their oscillators, claiming that their TAE
technology, “allows the production of totally aliasing-free oscillators
in all contexts.” For those new to the term, aliasing is a side-effect
that occurs when digital devices attempt to handle signals — input
from the outside world or generated internally — whose bandwidths
exceed that of the system. In almost all cases it results in the
generation of spurious frequency components that are unrelated to
the wanted audio, and can result in clangorous and unpleasant
sounds. Arturia’s words are an over-statement, but significant
aliasing is generated only in the Origin oscillator. Given the low
The Edit/Rack screen doesn’t show
CPU load of this module (ah, that’s where the processor ‘cost’ was
the interconnections between
saved) that’s not surprising, and quite acceptable.
modules, but it allows you to see
Nonetheless, you may wish to be aware that the facilities of the
what’s happening on the
imitative oscillators step well beyond their inspirations and do not
synthesizer’s virtual ‘control panel’
model them precisely. For example, the Minimoog oscillator offers
and to tweak the knobs and sliders
distinctly un-Minimoog-ish facilities such as PWM (which, not
much as you would on a physical
being a purist, I think is a good thing) and its square wave has a
modular synth.
duty cycle of 47 percent, which might emulate a Minimoog in
The Edit/Rack screen doesn’t show the
Grenoble, but is not the same as mine. Further anomalies are interconnections between modules, but
present in other modules, but whether these worry you or excite it allows you to see what’s happening
you will be determined by whether you choose to approach the on the synthesizer’s virtual ‘control
Origin as an emulator or as a new instrument in its own right. I panel’ and to tweak the knobs and
veer toward the latter. sliders much as you would on a physical
modular synth.
Moving on from the virtual analogue oscillators, I want to praise
the wavetable oscillator, a collection of 96 waveform snippets
modelled on the Prophet VS ROM. You can insert four of these into a single patch and control them using
the joystick mixer and the 2D envelope in a manner that’s very similar to the original. I very much liked the
sounds I obtained from this and, given that the VS is still one of my favourite synths, this is no small
accolade.
Let’s now talk about the Envelope module. Described by Arturia as an ADSR this is actually wrong, and the
manual is very poor at describing it. The contour generated is, in fact, an H1-A-D1-H2-D2-S-R envelope,
similar to that found on some (digital) Korg and Yamaha synths. In other words, it has a Hold stage before
the Attack kicks in, and two Decay stages with a definable hold duration between them. What’s more, as well
as being able to modulate the various A, D and R (Release) times, you can adjust their shapes from linear to
something akin to logarithmic, which means that you can imitate the various responses of many classic
synths, and even approximate the unusual envelopes of Yamaha’s CS-series. So while only one envelope
module is provided, it can emulate many of the subtleties of the four vintage synths on which the oscillators
and filters are based. Arturia should make much more of this in the Origin’s manual and marketing materials,
it’s good stuff.
Moving on to ease of use, the Origin is not the easiest of instruments to master, but it’s well thought-out
and straightforward once you get to grips with it. For on-stage performance, Arturia have even provided a
‘Live’ page that allows you to map the most important parameters to the eight programmable knobs on either
side of the screen, as well as to the joystick and the other control-panel knobs. Given the limited number of
suitable knobs in the oscillator, filter, LFO and envelope sections, this is not a luxury; it’s a necessity if you
want to tweak multiple elements of the sound as you are playing.
But what about the sound? Given that you can’t create an exact
copy of any of the synths from which the Origin draws its
inspiration, can it sound like any of them, or are its sounds just a
mish-mash of analogue timbres? The answer is a bit of both. You
can force the Origin to sound much like a Minimoog or a Jupiter 8,
but I preferred to create sounds that I couldn’t obtain from any
existing synths. An example? I placed three oscillators in a
Memorymoog type of arrangement, but instead of choosing a
Moog-style filter, I placed a couple of the CS80’s 12dB/oct low-
and hi-pass filters in the audio path. The result sounded neither

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Arturia Origin

The Minimoog template sits on top of

like a Moog nor like a CS80; it was a new synth with its own
an Origin configuration that is similar
character.
in form and sound to a genuine
To be honest, there’s much more that I would like to discuss MInimoog, and it encourages you to
here, but there’s no room to do so. Consequently, I must apologise create and play patches as you
for overlooking the Bode Frequency Shifter, the audio inputs that would on the original synth. Extra
allow you to treat external audio as if it were generated by an LFO and modulation capabilities are
oscillator within the Origin, the ‘Stereo Filter’ and ‘Crazy FX’ tucked away behind their respective
templates that take advantage of these inputs, the MIDI and buttons at the top of the screen.
System pages, the sound search facilities, the factory sounds, and
The Minimoog template sits on top of an
much more. I would also have a moan about the placement of the Origin configuration that is similar in
brightness and contrast controls, which are covered (!) by the form and sound to a genuine MInimoog,
wooden end-cheeks. However, there’s just enough room to report and it encourages you to create and
that, in the end, I managed to cause a problem (although not a play patches as you would on the
crash). I was removing modules from the Edit/Patch page while original synth. Extra LFO and
playing, creating some rather nasty, and understandable, clunks modulation capabilities are tucked away
behind their respective buttons at the
while doing so, and suddenly the Origin fell silent. I could still
top of the screen.
move around the pages and edit, but no matter what I did, no
sound was forthcoming and I had to switch it off and then on again
to restore things to normal. If that’s the only way to cause it to hiccup, the Origin is a remarkably stable piece
of kit.

Conclusions
The Origin is not a hardware implementation of existing soft
synths, and it should not be approached as such. Yes, it draws
upon aspects of vintage synths, but the number and range of
modules provided is much less than a complete butchery of the
original instruments would produce, and it does not slavishly
emulate any of them, instead allowing you to create new synth
architectures and sounds using the modules as building blocks.
This synth can create fabulous sounds that are reminiscent of a
Minimoog, an ARP 2600, a Jupiter 8, a CS80 or a Prophet VS, but
I am more impressed by its ability to step beyond these.
The sequencer section offers three
Inevitably, the Origin is too expensive to appeal to everybody,
rows of virtual CVs plus an
but the development time and effort that went into it was
arpeggiator, each with a dedicated
enormous, and this is reflected in its cost. As for me, I’m looking
editing screen. The overview screen
forward to a number of promised upgrades (such as numeric
allows you to see all three
readouts and greater MIDI controller capabilities), as well as the
sequences simultaneously, helping
forthcoming keyboard version. This, in addition to a 61-note
you to understand how they are
velocity- and pressure-sensitive keyboard, will add a pitch-bend
interacting.
wheel, a mod wheel and a ribbon controller to the existing Origin,
The sequencer section offers three rows
and could be a live performer’s dream ‘analogue’ synth, obviating of virtual CVs plus an arpeggiator, each
the need to carry around a lot of old, heavy, delicate, and valuable with a dedicated editing screen. The
vintage gear. Hmm, now there’s a thought... overview screen allows you to see all
three sequences simultaneously,
helping you to understand how they are
interacting.

Vintage Templates
If you want to create sounds on something that looks like an existing synthesizer, the Origin features a
Minimoog template that offers the appropriate modules correctly configured to enable you to program and
play it like the original.
As you would expect, it includes three Minimoog oscillators, and the controls appear to emulate those of
the real synth. However, all is not as it seems. For example, there’s oscillator sync and, while the
envelopes appear to generate the Minimoog’s ADSD contours, the Release knob on the physical control
panel also controls the release of the sound (which, of course, is wrong). Furthermore, there’s a
dedicated LFO, a modulation matrix, and polyphony.
Strangely, given the quality of much of the Origin software, there are two obvious bugs in the template.
The manual describes the modulation matrix as eight-slot (it has only six slots) and the Unison mode

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Arturia Origin

described isn’t present. Notwithstanding these niggles, I’m rather glad of the additions to the template,
because they defuse the inevitable question, “ah, but does it sound identical to a real Minimoog?” I prefer
to view the template as a visual programming aid that enables players to recreate many Minimoog-esque
sounds (which, to answer the question, it does well) and to program advanced sounds that use the
Minimoog architecture merely as a starting point.
There are no factory template for the CS80, Jupiter 8, ARP 2600 or Prophet V at the moment. The
manual promises these for the future, but I thought that I would try to build a CS80 for myself. At first,
everything went well as I created the two separate audio paths needed, each with independent high-pass
and low-pass filters, dual envelopes and LFOs, plus global ring modulation. But by the time I had inserted
all the modules, the CPU meter was reading way over 100 percent and the Origin had muted itself. Not
for the first time (and not for the last) I realised that the Origin is not Arturia’s V-series soft synths in
hardware form.

The Software Connection

The Origin Connection software runs under OS X and Windows (XP and Vista) and provides two sets of
facilities: sound/sequence storage and firmware upgrading.
Connecting the Origin to the computer via USB 2, I expected the software and synth to synchronise
themselves almost immediately, but the delay was rather longer than I had expected. However, once up
and running, the backup/librarian functions seemed to work correctly, although I found details of the user
interface rather inconsistent. This doesn’t affect the sounds, but I think it could be tidied up a bit.
You can also use this software to cross-connect two Origins, to swap sounds between them. Obviously,
I was unable to test this, but I have no reason to believe that it wouldn’t work as promised.

Physical Highs & Lows

Weighing in at 8kg, the Origin is a chunky, 6U affair designed for tabletop use, but you can also remove
its front lip and wooden end-cheeks for rackmounting. The hardware is solid, and the high-resolution
colour LCD has a good range of viewing angles. (Wouldn’t it be nice if it were touch-sensitive, though?)
The only thing that I would criticise about the design is the choice of an external power supply, and an
odd one at that. Rated at 6.5V and providing a massive current of 3.85A, I have never seen its like
before. If it fails, you’re extremely unlikely to find a replacement in the local electrical store.
The Origin’s hardware connections are as follows:
Outputs
Two quarter-inch main audio outputs.
Eight quarter-inch auxiliary outputs.
One S/PDIF digital output.
One stereo headphone output.
Inputs
Two quarter-inch audio inputs.
One foot controller.
One footswitch controller.
MIDI
MIDI In, Out and Thru.
USB 2 connector.

Modules & Factory Sounds

There are currently 20 types of module from which you can build synthesizers within the Origin.
Notwithstanding the CPU limit, which can be exceeded more quickly using some modules than others,
there appears to be a hard limit of 20 modules that you can insert at any given time. There are also sub-
limits, such as nine oscillators, four filters and four mixers in any given program. The three effects slots,
FX mixer, 2D envelope, Galaxy module and global LFOs are always available.
Here’s a list of the Origin’s modules.
Oscillators

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Arturia Origin

Origin
ARP 2600
CS80
Minimoog
Jupiter 8
Wavetable
Filters
Origin
ARP 2600
CS80
Minimoog
Jupiter 8
Others
Keyboard Follower
Envelope
CV Modulator
LFO
MiniMixer
Ring Modulator
Bode Frequency Shifter
Joystick Mixer
Output
The Origin has space for 1000 Programs, of which 400 are preset factory sounds, and for 256 Multis, of
which 100 are preset. These range from relatively simple and often very usable emulations of vintage
analogue synthesizers to complex soundscapes comprising multiple layers of sequenced and arpeggiated
‘virtual analogue’ patches. I wasn’t a huge fan of the more complex factory programmes and preferred to
create my own (simpler ones) during the course of the review. Nonetheless, I have no doubt that other
players will be happy to use many of the sounds on offer.

Published in SOS June 2009

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Dan Dean Solo Strings Advanced

Dan Dean Solo Strings Advanced

Kontakt 2 Player Buy PDF
Published in SOS June 2009
Reviews : Sample/Sound/Song Library

D
Printer-friendly version

an Dean’s original Solo Strings library comprised violin, viola, and cello, beautifully recorded in 24-bit stereo and converted to 16-
bit. Now, in true Hollywood style, there’s a sequel, with lots more special effects: the 2.5GB Solo Strings Advanced (SSA) makes
good use of the Kontakt engine, with some very clever programming and convolution-based features.
Presets are presented in three groups. ‘All-In-One’ offers the same six articulations as previous
libraries (Arco vibrato, Spiccato, Pizzicato, Tremolo, Half and Whole Step trills), all available via
keyswitching, but the Spiccato and Pizzicato now feature auto-alternation of two ‘takes’ for
greater realism, and the new sustain-pedal-driven legato mode provides shorter attacks for
smoother real-time performance of fast passages (one of my few grumbles with the original).
With ‘Voice Control — Divisi’, each preset provides a single articulation, but the keyswitching (or
the mod wheel) controls the number of players. Unlike the huge gulf between the solo and
ensemble choices offered by most string libraries, SSA lets you change on a note-by-note basis
between solo player, three-instrument, and six-instrument ensemble in the case of cellos or
violas. For violins, you get an even more ambitious selection, courtesy of a new set of 2nd violin
samples, with solo, two first plus one second, three first, three of each, six first, or six of each.
This range offers unprecedented real-time control, and for even greater realism you can activate
the ‘Cloaking Device’, which performs subtle trickery with alternate samples and other note
characteristics, so that each note sounds slightly different.
In the case of the first two preset groups, you can also control real-time dynamics via the mod
wheel, or breath, or expression controllers, and release time from MIDI Continuous Controller 20.
The ‘Legacy’ presets are simpler, offering one articulation each, with no keyswitching. They revert
to the normal sustain-pedal function, but still offer attack and release tweaks: ideal if you plan to
ignore the manual!
All presets now benefit from 26 specially selected convolution reverb impulses, ranging from a
very small room to a large symphony hall, so your acoustic options are extremely flexible.
The Timbral Impulses are intriguing: they let you remove the resonant character of the original recorded strings and replace it with one of
24 different ‘sonic fingerprints’, captured from recordings ranging from live classical performances to famous pop songs. These offer great
tonal flexibility, ranging from smooth and dark to thin and wiry. Names are cryptic, but I’d bet ‘03 ER’ is Eleanor Rigby!
While it doesn’t offer velocity-switched layers, SSA more than compensates with its unique divisi and timbral impulse features, which
make it an inspiring and extremely versatile real-time ‘instrument’. SSA can only enhance the enviable reputation earned by the earlier
version of this library. Martin Walker
$199.
www.dandeanpro.com

Published in SOS June 2009

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Fractal Audio Axe-FX Ultra

In this article: Fractal Audio Axe-FX Ultra

Rapid Response Guitar Amp & Effect Simulator
On The Button Published in SOS June 2009
The Matrix Reviews : Processor
Print article : Close window
In Use
Summary
Fractal Audio’s Axe-FX is the guitar processor with ’80s looks and 21st-
At The Back century power!
Buying An Axe-FX
Paul White

T
Fractal Audio Axe-FX
his is the first product I’ve tried from Fractal Audio Systems,
Ultra 2029 Euros
and it’s one of a pair of guitar amp and effect simulators
pros built by the company. The Axe-FX Ultra and standard Axe-
Makes a convincing job of FX are similar but the former, reviewed here, has a faster
emulating most guitar amps. processor and more memory, which equates to the ability to run
Loads of effects, some quite more effect models at the same time. The Ultra also includes
unusual. additional effect algorithms, enabling it to create more abstract
Comprehensive routing
guitar sounds, including an arpeggiator, synth effects, vocoder,
options.
Includes a vast number of looper, multi-band compressor, ring modulator, quad chorus, diffuser, resonators, crossovers, and more
inspiring presets to tweak or sophisticated delay and pitch-based effects. My first impression is that if the designers of the Lexicon
edit as you see fit. PCM80, Peavey’s ReValver and the Line 6 Pod X3 got together with no budget restrictions, this is
cons somewhere close to what they’d come up with.
Expensive! Like the standard version, the Axe-FX Ultra is packaged in a 2U steel case and is powered directly from
No front-panel amp controls the mains. Its black front-panel design is somewhat utilitarian, but the rear view reveals a comprehensive
or headphone output. range of I/O options to meet both stage and studio requirements. There’s a reasonably large display window,
summary which shows both values and graphical control panels, and navigation is accomplished by the now familiar
The Axe-FX Ultra sounds matrix of dedicated buttons, cursor and page buttons and a data wheel. Power amp, microphone and cabinet
good and is an immensely simulations may be enabled or disabled on a per-patch basis, so the user can set up some patches for live
versatile device, though it performance and others for studio use.
works out quite expensive
and some players might find
Rapid Response
the user interface fiddly.
The Axe-FX’s cabinet emulations are based on Impulse Responses, or IRs, taken from real speaker and mic
information
setups, a technique that generally produces very accurate results, and Fractal Audio say that their tube amp
Axe-FX Ultra 2029 Euros; algorithms model aspects of the original circuit down to the component level, reproducing the dynamic way in
Axe-FX 1549 Euros. Prices
which the frequency response of the amp being modelled varies under playing conditions. Power-amp
include VAT and shipping.
damping, rectifier sag and transformer characteristics are all modelled and can be adjusted by the user.
G66 +49 461 1828 066.
They’ve gone into this level of detail for over 50 amplifier types teamed with 39 speakers, 10 microphone
Click here to email
types and a host of stomp-boxes and rack effects. However, the designers also point out that many of their
www.g66.eu
modelled effect devices are designed for optimum playability rather than as copies of specific devices —
www.fractalaudio.com
some users will view this as an improvement, while others may feel differently! There are also sophisticated
Photos too small? Click features such as intelligent harmony generation and the ability to swap tone stacks between amp models.
on photos, screenshots and Fractal Audio Systems are also keen to point out that they take the technical sound quality of the unit very
diagrams in articles to open seriously. They use 24-bit Cirrus Logic converters, and their analogue front-end and output-stage circuitry
a Larger View gallery. employs no electrolytic capacitors in the signal path. Analog Devices op-amps are used in the analogue
sections and a noise floor of better than -105 dB is quoted. The maximum analogue output level from the unit
is around +18 dBu. The Axe-FX Ultra’s dual-core floating-point processor gives the unit enough power to run
two different rigs at the same time, as some software plug-ins now also allow, and is claimed to have
enough horsepower to give some of today’s desktop computers a run for their money.

On The Button
The process of editing Axe-FX patches will be familiar to anyone who’s used a typical rackmount effects unit.
The value dial is used to adjust parameters, while Enter confirms an activity. Exit cancels the current step,
the four cursor buttons allow you to move around inside a page or screen, and the page buttons move
forwards or backwards through the available pages. Tabs at the top of the edit windows show you which
page you are on, and the Layout button is used to place processing blocks into the grid and to access the
routing menu.
The Control button accesses the menu for the internal controllers, including a tempo setting, two LFOs, two
ADSR envelopes and an envelope follower. Individual effects can be bypassed, and there are dedicated

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Fractal Audio Axe-FX Ultra

buttons to access the Global settings page, tuner display, I/O configuration and Utility menu.
The Recall button lets you select presets using the value dial (and they load almost immediately), while
Store and Bypass need no further explanation — though it should be noted that storing a patch overwrites
one of the factory patches, which can either be the current one or any other of your choice. A tempo setting
is saved with the preset, but a new one may be tapped in at any time using the Tempo button. Parameters
can also be assigned to MIDI controllers for real-time control, making live performance rather easier if you
add a suitable MIDI floor controller.
In addition to the buttons, there are also status LEDs below the display. The one labelled ‘Edited’ lights up
if you change a preset and haven’t yet stored the changes, ‘MIDI’ lights if MIDI data is being received, and
clip LEDS warn of excessive levels at the outputs. Internal clipping is unlikely because of the floating-point
architecture, but it is possible to overload the 24-bit output converters, in which case the internal mixer gain
or individual block gains should be adjusted downwards.

The Matrix
As with most such products, the user has the option of deciding how deeply to edit. Up to 12 simultaneous
effects can be set up in series or parallel, with a high level of editability, so you could simply create one
virtual pedalboard setup for live use, storing different control and bypass settings as presets.
That, however, would be to miss out on some neat routing options. The user interface’s routing section is
based on a 4 x 12 grid, where effect and amp blocks are drawn from an inventory of available items and then
set up in the order in which you’d expect their hardware counterparts to be used. Some items are available
in multiples, but once you’ve drawn all the available instances out of the ‘store cupboard’, that’s as many as
you can have. Once an effect has been slotted into the grid, it can be patched left-to-right or to any other
adjacent effect to create parallel signal paths. Each effects block has the same routing options, in that the
input sums up to four stereo inputs from the four rows and the output is in stereo. An output mixer then
combines the outputs from the four columns into a stereo output.
Blocks can be edited separately, a process aided by the easy-to-understand graphical interface, which
uses tabs where multiple pages are involved. Only four of the 12 columns are visible at any one time, so the
display can be scrolled across to ‘look at’ the section you’re working on, and unused locations can be linked
across. Real-time monitoring of controlled parameters is also supported, so if you have a gain control that
ramps up after being triggered by a guitar note, for example, you can see the on-screen knob turn as you
play. This is no gimmick, as it makes setting up dynamic effects much easier.
Choosing a new amp is done via the Type knob in the Amp edit page, which scrolls through the available
options once Amp 1 or Amp 2 has been inserted into the grid. When editing amps, you get the obvious gain,
EQ and volume controls on the first page, some slightly unusual ones on the second page (rectifier sag,
cabinet thump and other esoterica), then on the third page there are advanced functions that let you change
the tone stacks, transformer characteristics and so on. A fourth page addresses the mix parameters — level,
balance and bypass mode. A similar paradigm applies to effects, where some adjustments are graphical and
others in the forms of lists of parameters and values.

In Use
The first thing any self-respecting guitarist does when confronted with a product like this is to explore the
presets. There are over 350 here, and they show off the creative potential of the machine, both for emulating
recognisable amp types and for generating ethereal sounds that would normally take a rack of sophisticated
processors and a lot of patience. Many of the weirder sounds make use of slow-attack envelopes, pitch-
shifting and shimmery modulation. I’ve heard similar things from Native Instruments’ Guitar Rig and from my
own Roland VG99, but the Axe-FX Ultra does them really well, and with plenty of potential for variety.
The straight guitar amp sounds are pretty impressive: while most modelling boxes can manage filthy rock
sounds quite adequately, this one also delivers well on those cleaner sounds or slightly overdriven sounds
that clean up as you back off the volume control. There’s a definite sense of something solid behind the
sound, and those IR-generated cabinet responses really fill out the low end. Of course, you’re unlikely to find
a preset sound that just happens to work perfectly with your own guitar, so as with a real amp, you’ll
probably need to adjust the gain and EQ at the very least, and even if setting sounds up from scratch is too
daunting, modifying the ones that you’re given isn’t difficult.
At the time of writing, Fractal Labs are in the final stages of developing both a plug-in version of the Axe-
FX, called Axe-PC, and a fully fledged graphical editor that will allow you to tweak Axe-FX patches from an
attached PC. This will make a big difference to the user-friendliness of the system, as many guitar players
are reluctant to engage with anything that looks in any way technically daunting. Even though the on-screen
editing is fairly intuitive, the sheer number of variables and routing options makes the experience more like

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Fractal Audio Axe-FX Ultra

setting up a synth patch than using a guitar amp, especially as there are no familiar amp controls on the front
panel.

Summary
For some users, the Axe-FX Ultra might seem like overkill, especially if they don’t want to use the more
abstract sound capability of the unit, but for those who need as much creative freedom as possible, it’s
probably the only hardware solution that comes close to matching software for its sheer versatility.
Furthermore, unlike software, which has to be written to conserve CPU power and allow multiple plug-ins to
be run at the same time, the Axe-FX Ultra can have all its DSP for its own use, which is pretty much the
same as running one very sophisticated software plug-in that drains the power of the entire computer.
There’s little to dislike about the Axe-FX Ultra’s capabilities, so if anything is going to put people off, it’s that
the user interface feels rather dated, while the physical presentation doesn’t scream ‘Got to have it!’ either.
As I touched upon earlier, the control paradigm is straightforward enough, but because of the number of
options and adjustments, a well thought-out graphical editor with drag-and-drop capability will be a valuable
addition. I’d also have liked a headphone output and some front-panel amp controls for gain and tone
settings.
Sound is a subjective thing, of course, but I thought the Axe-FX Ultra captured the punch and dynamics of
a real amp being miked in a studio rather well. The array of effects is impressive, and most sound really
musical, though inevitably the pitch-shifter sounds a little bit grainy. Some of the sounds that can be conjured
up by, for example, feeding delay or reverb from a reversed version of the sound fed through a pitch-shifter
and/or a series of resonant filters rival synth sounds in their complexity, and this is an area where the Axe-
FX Ultra scores highly against the competition. However, if you simply want to wig out on metal overdrive
with a bit of delay, there are less costly options!

At The Back
Although there’s only a solitary instrument input on the right of
the front panel, the rear panel of the Axe-FX Ultra is a pretty busy
place. Input 1 is used when employing the unit as an effects
processor, and there is also a choice of a digital input and output
on both phono (S/PDIF) and XLR connectors. When you’re using the digital I/O, the input can accept 24-
bit/48kHz data and Input 1 is disabled. When the rear-panel analogue input is in use, the front-panel jack
is overridden. Input 2 and Output 2 can be used as an effects loop for externally connected devices, or as
an auxiliary output for monitoring, and may also be used as I/O for the effects loop ‘block’ when placed in
the grid with its (on-screen) input left unconnected.
There are two sets of outputs, one unbalanced and one balanced, with a ground-lift switch to lift the
screen of the cable from the chassis ground, helping to avoid ground loops. The IEC mains inlet is also
on the rear panel, though the power switch is sensibly located on the front panel. One obvious omission,
though, is a headphone output, which I would have found useful when editing patches.
To facilitate real-time parameter control, the unit also has MIDI In, Out and Thru sockets: the MIDI In is
on a seven-pin DIN rather than the usual five-pin version, enabling phantom powering, for those pedals
that support it, on pins 6 and 7. Two expression pedal inputs are provided, on jacks, and these may also
be used with footswitches.

Buying An Axe-FX
The Axe-FX and Axe-FX Ultra are currently available in Europe only by mail order from distributors G66.
Obviously, this makes it hard to try before you buy, so they offer a no-quibble, 15-day, money-back
guarantee.

Published in SOS June 2009

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Heil Sound PR40

In this article: Heil Sound PR40

Overview Cardioid Dynamic Microphone
In Use Published in SOS June 2009
All Hail Heil? Reviews : Microphone
Print article : Close window

Heil Sound PR40 £230

pros Listening to the sound from this extraordinary dynamic mic, you might
be forgiven for thinking it was a ribbon or a condenser.
A great all-rounder
Enough bass extension for Paul White

T
kick drums and bass
instruments here are days when the thought of one more ‘me too’ side-
Copes well with high- address cardioid condenser vocal mic doesn’t exactly thrill
frequency detail. me, so I was pleased to discover that, despite its familiar
cons appearance, the Heil Sound PR40 is not what it appears to be at
Only the price. first glance: although it looks like a large-capsule condenser mic,
summary behind that deceptive exterior lies an end-address dynamic mic
that’s designed for use with voices, kick drums, bass instruments,
The PR40 pushes the
boundaries of what can be guitar cabs and lots more.
achieved using dynamic mic
technology, which allows it to Overview
overlap into areas normally
dominated by capacitor Given its physical attributes and extended low-end response, the
models. PR40 will probably be compared with the Electrovoice RE20,
although I also see performance parallels with the Sennheiser
information MD421. Either way, you know that if Bob Heil is involved in the
£229.95 including VAT. design, you’ll get something a bit out of the ordinary. Readers of
Waters & Stanton +44 our sister publication, Performing Musician, will know that Bob Heil
(0)1702 204965. is both a ham radio enthusiast and a live-sound guru with an
Click here to email impressive provenance, first coming to public notice in the early
www.wsplc.com ’70s when he set up the now-legendary ‘wall of sound’ PA for the
www.heilsound.com Grateful Dead. He’s since been involved in the design of
communications microphones, so he has a lot of expertise when it
Photos too small? Click
comes to designing mics with precise pattern-control.
on photos, screenshots and
diagrams in articles to open The PR40, which is assembled and tested at Heil Sound’s facility
a Larger View gallery. in Illinois, USA, has a surprisingly wide frequency range for a
dynamic microphone, covering 28Hz to 18kHz (-3dB). It is
designed to withstand very high SPLs, but at the same time it
manages to sound more natural on voice and other instruments than most cardioid dynamic mics — many of
which have a noticeably coloured sound, due to the complex porting needed to create the cardioid polar
pattern. Although the frequency response is nominally flat between the upper and lower roll-off points, there’s
the gentlest hint of a presence bump from 3-5kHz. It only amounts to a couple of dB, but it gives a sense of
air at the top end.
Bob Heil’s approach to cardioid dynamic mics always seems to produce a tight polar pattern, with almost
perfect rear-rejection — a feat Bob attributes to “using the ideal combination of materials for the 1.125-inch,
low-mass diaphragm and a special mixture of neodymium, iron and boron that gives the PR40 the strongest
magnet structure available.” Aluminium is used for the voice coil and, as with the PR series hand-held
models, the large-diameter dynamic capsule is mounted in a Sorbothane shock absorber to decouple it from
the heavy steel body. An additional humbucking coil reduces the effect of interference from nearby electronic
devices or transformers, and proximity bass-boost has been minimised as far as is possible for a pressure
gradient microphone.
The basket screen comprises two wire-mesh screens of different diameters, augmented by what’s
described as an internal breath-blast filter (I couldn’t get into the microphone to see how this was arranged).
It does, however, keep popping to a minimum when the mic is used for vocal work, and also helps avoid
sibilance. A champagne-coloured satin plating is used over the steel body, and the signal exits on the usual
balanced XLR. A swivel stand-adaptor is provided with the mic, which comes in a compact, foam-lined
aluminium case, with a paper banner around the mic reminding the user that, despite appearances, this is an
end-fire mic.

In Use

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Heil Sound PR40

Bob Heil is particularly proud of this microphone’s performance on kick drum, although it’s also
recommended as a broadcaster’s voice mic. Given that most kick-drum mics have a massaged frequency
response, some heavy mid-cut EQ may be necessary to achieve a contemporary kick sound, but there’s no
lack of low-end extension, and you can get a great depth of sound. But this mic isn’t only good for kick drum,
by any means. I got some great djembe tones out of it, and on vocals it sounded impressively natural, but at
the same time full and solid — which would be ideal for radio DJs, as well as for some types of studio vocal.
However, you do need to be aware that the level changes hugely between what you get from singing right up
against the grille and what you get when working three inches or more away, so it would be a good idea to
use a pop shield, just to keep the singer back at a safe distance. The mic’s susceptibility to popping is
impressively low, given the extended bass response, but I’d still recommend using a low-cut filter and pop
shield for vocal work.
Because of its solid bass response, this mic works really well on bass guitar cabinets, and I loved it on
electric guitar too, where the results were rather less ‘honky’ than I’m used to from dynamic models. It has
an almost ribbon-like smoothness on highs, but without any loss of transient detail or any dullness to the
sound.
Another bonus is that, because the high end extends up to 18kHz, you can use the mic in many
applications where a capacitor model would normally used — and if you have a clean, quiet preamp, it
sounds fabulous on acoustic guitar, revealing plenty of detail and a dense mid-range, but without the glassy
grittiness that some budget capacitor mics seem to impart to the sound. In this respect, at least, its
performance might best be compared with the Sennheiser MD441.

All Hail Heil?

The Heil Sound PR40 isn’t a cheap mic, by any means, but then neither are the mics with which it compares.
In fact, some of them cost quite a bit more. This has to rate as one of the best dynamic mics I’ve ever come
across, both for tonality and versatility — and it’s probably a good thing for the rest of the industry that Bob
Heil hasn’t yet turned his attention to condenser mics!

Published in SOS June 2009

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Lexicon Ionix FW810S

In this article: Lexicon Ionix FW810S

Mixer & Routing Firewire Audio Interface
Dbx-powered Mixing Published in SOS June 2009
DSP-powered & Reviews : Recording System
Print article : Close window
Plug-in Reverbs
Verdict
The Ionix FW810S comes with built-in Lexicon monitor reverb and
Vital Statistics
onboard Dbx dynamics and EQ. So does all this DSP add up to a genuine
Dbx Dynamics & EQ
USP?
Lexicon Ionix FW810S Mike Senior

L
£753
exicon may be best-known for their reverb processing
pros know-how, but more recently they have been making
Low-latency DSP-based inroads into the computer-music market, with their latest
mixing with eight channels of foray being the Ionix range of control and audio-interfacing
high-quality Dbx mic
hardware. The FW810S audio interface is the only rackmount box in this range, and features eight-channel
preamplification, Type IV A-D
analogue and stereo digital I/O, with additional stereo analogue main and headphone outputs, as well as
conversion, gating,
compression and EQ. MIDI In and Out. In addition, there is DSP-based onboard low-latency mixing, incorporating eight channels
Mixer control software is very of fully-featured Dbx dynamics and EQ, and a Lexicon monitor reverb. Bundled with the unit are Steinberg’s
easy to use, despite a few Cubase LE4, Toontrack’s EZ Drummer Lite, and Lexicon’s own Pantheon II reverb plug-in. The details of the
operational niggles. hardware profile can be found in the ‘Vital Statistics’ box on the opposite page, but the real question, of
Excellent (and CPU-friendly) course, is how all these features actually perform in practice.
DSP-driven and plug-in
reverbs included.
Mixer & Routing
cons
The feature set of the DSP Beyond the handful of front-panel controls, everything on the
input processing doesn’t feel FW810S is configured via its software control window. Each of the
very well aimed at the needs 10 hardware and eight software inputs has its own channel
of computer-based tracking containing send level and pan controls for the five available
sessions, and you can’t
analogue output pairs, and each output pair has its own master
reassign the processors to
playback channels where level control. Odd/even pairs of channels can be linked for ganged
they would work better for stereo operation, and there’s a send per channel to the DSP-
mixing. powered monitor reverb, which can be returned to any of the five
No preamp high-pass filter or analogue output pairs.
polarity inversion functions.
The mixer is an absolute doddle to comprehend on account of its
Only one headphone output
traditional-style control layout and absence of tabbed pages,
and no talkback.
Only stereo S/PDIF digital I/O although you may need to scroll around a bit to find the channel
with no word clock you’re looking for. Only the dynamics and EQ controls are
connections. accessed via separate pop-up windows, but gain-reduction and
summary EQ-plot read-outs are nonetheless permanently on view. Visually,
The software mixer utility which
A meeting of the Dbx and I’d have liked the settings of the rotary controls to be much clearer,
controls the FW810S’s internal
Lexicon brands augurs well as the graphical style of the knobs makes it quite difficult to see
mixer. To the left is a scrolling pane
for any product, and sonically where they’re pointing — little sliders would have been more
for navigating through the available
the FW810S delivers on that functional, if less familiar to hardware junkies.
input and playback channels; in the
promise in both analogue
Little tooltip-style numeric read-outs pop up whenever you move middle are the control and return
and digital domains. If you’re
looking for maximum I/O a control, but not when you initially click on it or try to adjust it with parameters for the DSP-powered
count (particularly on the your mouse’s scroll wheel, which I found a bit annoying. Using the monitor reverb; and to the right are
digital side), however, the scroll wheel for adjustments also seemed to have a lower the master controls for the five stereo
FW810S loses out somewhat resolution than clicking and dragging, and no modifier key output feeds.
to the competition. increased this resolution either, so I’d stick to click and drag if I The software mixer utility which controls
were you. Clicking on any of the scraps of virtual masking tape the FW810S’s internal mixer. To the left
information
allows you to rename channels in the mixer utility window, and is a scrolling pane for navigating
£753.25 including VAT. while this is useful to some extent, it doesn’t change how the through the available input and playback
Sound Technology +44 channels; in the middle are the control
names appear to your sequencing software via the driver.
(0)1462 480000. and return parameters for the DSP-
Click here to email Metering is pretty good, although because it’s possible to add powered monitor reverb; and to the right
www.soundtech.co.uk gain to DAW playback streams within the FW810S mixer itself, you are the master controls for the five
do need to have the mixer utility open to be sure that you’re not stereo output feeds.
Photos too small? Click clipping the interface’s outputs. The meters have no peak-hold
on photos, screenshots and option, but each channel has its own clip LED which will stay lit until you click on it to reset. Snapshots of the
diagrams in articles to open entire setup of the mixer can be stored and recalled, allowing quick reconfiguration, but there’s no separate

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Lexicon Ionix FW810S

a Larger View gallery. facility for storing dynamics, EQ or reverb patches independently, or indeed any preset library of settings for
these.

Dbx-powered Mixing
The analogue mic/instrument preamps come courtesy of Dbx, and
offer up to 55dB gain, which is perfectly adequate for most close-
miking applications, but perhaps a bit on the low side for distant
miking or less sensitive mics such as ribbon models. The gain is
also slightly bunched towards the clockwise end of the control
range, so that last 6dB or so of gain is pretty difficult to set. In
terms of sound, this preamp is nice and clean, with a low noise
floor and little obvious coloration, which makes it ideal for general-
purpose use.
Analogue-to-digital conversion is via Dbx’s own Type IV process,
which emulates the soft-clipping characteristics of analogue
recording media, with a more gradual distortion onset than you’d The eight analogue input channels of
expect. I’ve been using Type IV conversion for years, and the the FW810S each have well-
system does work well, but distortion is still distortion, so I’d not specified DSP-based dynamics and
advise setting your recording headroom any differently. Better to EQ processing sections coded by the
think of the Type IV just as an extra safety net for worst-case boffins at Dbx.
scenarios. The eight analogue input channels of
the FW810S each have well-specified
Further Dbx input processing follows the A-D conversion, and
DSP-based dynamics and EQ
comprises (in order) gate/expander, compressor and limiter for
processing sections coded by the
each of the eight inputs. All of these processes are fully featured boffins at Dbx.
and well-behaved in practice. The gate/expander works very
effectively to reduce background noise, and the wide ratio range
lets you fairly easily adapt the effect to different sources. Spill reduction on multitrack drum recordings might
be a bit too challenging though, as there are no side-chain filtering options.
The compressor’s variable ‘Over Easy’ soft knee is as accomplished as I’ve come to expect at invisibly
reining in wide-ranging dynamics, and the automatic time constants make for quick setup, even with very
dynamic sources like vocals and DI bass. The hard-knee mode can, however, be used with the fully variable
time settings to create a good variety of more obvious compression effects if you prefer. Having a separate
limiter is nice too, allowing the compressor to concentrate on the gentle transparent soft-knee processing at
which it shines while ensuring that you catch any troublesome level peaks.
Unusually, both the compressor and limiter have a Hold parameter, which stops any gain-reduction from
beginning its reset phase for a specified duration after the signal level has dropped below threshold. I
imagine that many users will either simply leave this control well alone or set it to its minimum value, but it’s
worth playing with it because, especially on percussive material, if you extend the Hold time you can retain
more of a sense of transient definition when you’re using fast attack/release to emphasise sustain or
ambience. It can also reduce distortion of low frequencies in similar circumstances.
The EQ is commendably smooth, even when using peaking
boosts or adding general high-end, both of which applications tend
to provoke budget EQs into harshness. Again, you get enough
control resolution to do some pretty detailed tweaking if you fancy
it. All the dynamics blocks and the EQ have their own bypass
buttons, and the status of the dynamics blocks is reflected by
LEDs on the front panel. (Why not the EQ too? Beats me.) Bear in
mind, though, that these status LEDs only show you that the
dynamics block in question is switched on, not whether there’s any
gain-reduction going on.
Overall it’s hard not to like all this Dbx processing when you
consider it in isolation, but there is one general thing about it which
doesn’t quite seem to add up for me: it feels best suited to mixing, yet you can only use it for tracking on the
FW810S. Given that the quiet preamps and 24-bit conversion make it perfectly possible to leave gating,
compression and limiting to the mixdown stage without the risk of printed-in gate chatter or over-
compression artifacts, I’m not sure that many musicians are going to make much use out of the enormous
processing flexibility on offer. I’m certainly not going to risk tiring out my performer while I spend ages
tweaking dynamics parameters, and I’d rather leave the soft-knee 2:1 compression ratio set up on automatic,
tweak the threshold and make-up gain to taste, and leave the other 13 variable controls to gather dust.

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Lexicon Ionix FW810S

What’s more, despite this copious control set, Lexicon and Dbx have managed to miss out a number of
input-processing facilities which would have been much more useful during tracking than a compressor hold
control. For example, there’s no preamp high-pass filter, and you also can’t switch the EQ before the
dynamics, so there’s no way to stop subsonic thuds and plosive pops from sending the gain reduction
lurching. There’s also no polarity inversion, and although polarity can of course be flipped in most recording
software, that’s not going to help your DSP-driven cue mixes. To be honest, I’d rather have had a variable
high-pass filter, polarity inversion and a two-knob soft-knee compressor for each input, assigning the other
processing to the eight DAW playback channels for stem or aux-processing purposes at mixdown.

DSP-powered & Plug-in Reverbs

With the Ionix FW810S you get two different reverb processors,
one running on dedicated DSP as part of the internal mixer and
the other in the shape of the bundled Pantheon II reverb plug-in.
As I’ve already mentioned, the former is fed from a send control on
each internal mixer channel. What I didn’t say, though, is that this
send is pre-fader, which makes some sense given that you don’t
want fader level changes to alter the wet/dry balance in any
independent monitor mixes. However, you do have to remember
that reverb levels won’t track dry levels as they would with the kind
of post-fader send you’d use while mixing. While the DSP-
powered reverb is limited to a single instance, the Pantheon II
plug-in is shipped in both VST and AU formats, so you can open A big bonus feature of Lexicon’s
up as many instances as you like in most sequencers, and seeing FW810S is its bundled Pantheon II
how comparatively CPU friendly it is, it’ll even suit those AU/VST-format reverb plug-in, which
unenlightened souls who still set up all their reverbs as inserts! sounds very nice indeed despite
snacking only comparatively lightly
Both reverbs operate in a similar way: first you choose a reverb
on CPU cycles.
type (from various chamber, hall, room, plate and ambience
A big bonus feature of Lexicon’s
options) and then you refine the sound using a clutch of additional
FW810S is its bundled Pantheon II
controls — four for the DSP reverb and 16 for Pantheon II. As AU/VST-format reverb plug-in, which
you’d expect from Lexicon, these reverbs are very satisfying, sounds very nice indeed despite
adhering well to the dry sound in a way that really helps mixes to snacking only comparatively lightly on
blend. There’s little sign of any of the unpleasant metallic CPU cycles.
colorations that blight small-CPU-footprint algorithms, and I have
to say that I was extremely impressed with the sound on offer. In terms of usability within commercial
multitrack productions, the DSP-powered processor lost out a bit on account of its lack of low-frequency
contouring options, but even with the more parameter-rich Pantheon II, I was inclined to follow it with a plug-
in EQ. That by no means reflects badly on the realism here; it’s just that realistic reverb isn’t necessarily what
works best in a busy mix!

Verdict
If it’s just preamps and socketry you’re looking for, then the
FW810S faces pretty stiff competition from a bevy of other
manufacturers, so it seems to me that a decision about whether to
On the FW810S’s rear panel we find
purchase this smart rackmount box has to hinge on whether its
all of the I/O connections, with the
other selling points appeal to you. Dbx’s analogue and digital
exception of the single headphone
processing know-how is certainly a big selling point here, and the
socket and two of the combi inputs,
onboard DSP mixer gave me roughly 1ms of latency compared to
which are on the front on the unit.
the 4ms figure I was able to achieve with the lowest stable ASIO
On the FW810S’s rear panel we find all
buffer size on my machine (128 samples), which was good going.
of the I/O connections, with the
That said, I still feel that true zero-latency analogue monitoring is a exception of the single headphone
noticeable improvement for vocal work in particular, where even socket and two of the combi inputs,
1ms delay causes noticeable phase-cancellation between vocal which are on the front on the unit.
foldback and the singer’s direct spill.
Dbx’s excellent soft-knee compression is worthy of particular praise too, and is a boon when working with
singers, because it lets the singer change the timbre of their voice at different dynamics without the
distraction of huge changes in the voice/backing foldback balance; and it does this with the minimum of
undesirable compression side-effects. The EQ is also good enough that you might want to use it on the way
in, expecially at the top end, and Lexicon’s built-in monitor reverb is classy-sounding and straightforward to
set up. Add all this together and you end up with an attractive package which does away with much of the

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Lexicon Ionix FW810S

need for external mixing and processing hardware during tracking.

The reverbs are the other big selling point for me, as they sound great in isolation, mix beautifully and
exude a flattering richness that I always associate with the Lexicon marque. Even if you have access to
convolution reverb with good impulse responses, you may find it difficult to justify that kind of CPU munch in
most mix situations when the slimmer Pantheon II sounds so smooth.
So while you don’t have to look very hard to find aspects of the FW810S’s raw feature set which may lose
Lexicon some customers to their competition (things like the single headphone output, lack of talkback
functionality, and absence of multi-channel digital I/O or word clock), the quality of the preamps, Type IV
conversion, DSP processing, and bundled Pantheon II reverb constitute strong selling points in their own
right, which should certainly attract the interest of anyone familiar with reputations of Dbx and Lexicon.

Vital Statistics
Firewire Audio & MIDI interface, simultaneously capable of 10 audio inputs and 12 audio outputs.
24-bit digital recording and playback at sampling rates up to 96kHz.
Compatible with Mac OS 10.4.9 and above, Windows XP and Vista.
Balanced analogue inputs: two front-panel mic/line/instrument inputs on combi-jack/XLRs; six rear-
panel mic/line inputs; phantom power switchable to mic inputs in pairs with illuminated front-panel
buttons.
Balanced analogue outputs: eight line-level outputs on TRS jacks; two further line-level main outputs
with front-panel rotary level control.
Mixing functionality: DSP-powered mixer for low-latency monitoring; eight input channels include
separate gate, compressor, limiter and EQ, as well as independent sends to each of the line output
pairs; further channels mix the stereo S/PDIF input and eight mono software playback streams; global
monitor reverb processor; full mixer snapshot recall, including a ‘power-on’ snapshot for stand-alone
operation without the computer.
Metering: three-LED bar-graph meter per analogue input channel; status LEDs for all three hardware
monitor-mixer dynamics modules.
Headphone output: front-panel headphone jack output with accompanying level control.
Digital I/O: coaxial S/PDIF input and output.
Other I/O: MIDI In and Out; dual Firewire sockets.

Dbx Dynamics & EQ

Gate: ratio 1:1-100; time controls comprise attack (0.1-200ms), hold (0-500ms), and release (350-
5dB/s).
Compressor: ratio 1.5-100:1 with switchable soft knee; time controls as for gate, with additional
switchable fully automatic mode.
Limiter: time controls as for gate, with additional switchable fully automatic mode.
Other dynamics controls: +20 to -30dB post-limiter make-up gain; separate gate, compressor, and
limiter bypass buttons.
EQ: four-band parametric design with plus or minus 12dB gain per band; bands comprise low shelf (40-
320Hz), two mid-range peaks (160-1280Hz and 640-5120Hz), and high shelf (2.5-20kHz); global EQ
bypass switch.

Published in SOS June 2009

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Line 6 Pod X3 Pro

Line 6 Pod X3 Pro

Guitar Modelling Processor Buy PDF
Published in SOS June 2009
Reviews : Processor

I
Printer-friendly version

n true Line 6 tradition, they’ve followed their Pod X3 and X3 Live processors with a rackmount professional version, the X3 Pro ($979.99
MSRP), which has enhanced I/O and a few capabilities not provided on the standard ‘kidney’ or X3 Live floor-unit versions. It features
the same Dual Tone architecture as the X3, which means that you can set up two different effect/amp/speaker cab rigs at the same
time and combine them. Alternatively, you can process two completely separate guitars, basses or voices using one processor channel for
each — something not possible with the standard version. You also get the Pod Farm plug-in (see SOS Jan 09), which essentially provides
all the X3 facilities in the form of a Mac or Windows plug-in, or as a latency-free stand-alone ‘virtual processor’ without the need for a DAW.
The hardware acts as the ‘dongle’ for Pod Farm and makes the latency-free processing possible: as long as the Pod is connected to the
computer, the Pod Farm software will run as a plug-in. You can still use another interface if you prefer, but for direct USB recording from the
Pod or for latency-free processing using Pod Farm, the Pod (or other suitable Line 6 interface) must be selected as the DAW’s I/O device,
or be part of a composite audio driver. You can also use the Pod X3 Pro as a computer interface for a Line 6 Variax guitar, enabling you to
create new guitars using the free Variax Workbench software.
The Pod Pro X3 uses 24-bit A-D and D-A converters, and the internal processing employs 32-
bit floating-point arithmetic — which is the same as in many DAWs. Up to nine simultaneous
effects per signal chain can be used, with effects split between pre-the-amplifier (stomp) and post-
amp (send loop). DAW users will welcome the fact that the Pod X3 family’s USB 2.0 connectivity
supports multi-channel recording and stereo playback without the need for an additional interface,
and there’s also a digital Variax connection that allows a Variax guitar to be connected in such a
way that X3 Pro presets can store the Variax settings alongside those of the X3 — so a performer can call up the right guitar and pickup
combination with each Pod preset.
Output 1-2 is fed by whatever you’ve selected for the Digital/XLR Outs, 3-4 is Tone 1 separately in stereo, 5-6 is Tone 2 separately in
stereo, 7 is the sum of the inputs for Tone 1 and 8 is the sum of the inputs for Tone 2. This allows the simultaneous recording of both
processed and unprocessed sounds, where both Tone channel outputs may be recorded in stereo.
The main differences between a standard Pod X3 and this Pro version are the extra connectivity and the ability to use both channels
separately. The X3 Pro has two dual quarter-inch instrument inputs, and there are also two balanced-XLR mic inputs with phantom power,
gain-trim controls and switchable low-cut filters. Digital I/O is presented in S/PDIF coaxial and AES-EBU balanced formats, as well as the
Variax VDI Digital Interface, which uses a CAT5 cable. Analogue outs are on both unbalanced quarter-inch jacks (switchable to amp or line
level) and balanced Studio/Direct XLRs (switchable to mic or line level with a ground-lift option), so pretty much any live or studio scenario is
catered for.
A connector is included for a Line 6 FBV floor controller, there’s MIDI In and Out on standard five-pin DIN sockets and stereo insert points
on quarter-inch jacks. Another nice touch is the provision of separate, unprocessed DI output for each channel. The headphone jack on the
front panel is controlled by the Master Volume knob.
Physically, the Pod X3 Pro is presented as a mains-powered, 3U rack device but it has been styled rather differently from earlier Pod Pro
units. This time, the distinctive sculpted, red, anodised front panels have been confined to the mic/instrument inputs. These have been
arranged to look like plug-in modules and have the necessary XLR and jack connectors on the front, where they can easily be accessed.
Also located here are the gain adjustment knobs, plus silver buttons for low-cut filters and pads and LEDs indicating signal presence and
clipping. The rest of the unit draws on influences from both the Pod X3 and X3 Live: a reasonably large display assisted by a cursor
controller, a data knob and four context-sensitive knobs do most of the editing work. I thought the presentation very stylish.
The various sound-shaping sections, Tap Tempo and Dual Mode selection are accessed via six push-buttons, to the right of which is a
Tone Volume and Master Volume control. Below is a conventional set of amp controls comprising Drive, Bass, Mid, Treble, Presence and
Reverb. Anyone used to using a previous model of Pod should have little trouble in finding their way around.

Operation
As wih the standard X3, you get 78 guitar amp models, which can be fed through any one of 24 cabinet models. There are 98 stomp and
studio effects, 28 further bass amp models with 22 bass cab models and six vocal preamp models, the latter being rather more impressive
than you might imagine in recreating a vintage sound. Room reflections are modelled using Line 6’s AIR II technology, and there are four
virtual mic options. If you want the Pod Farm plug-in mentioned earlier, you’ll need to download it from the Line 6 web site.
The large front panel means that many controls have their own dedicated knobs, and here eight chromed-plastic knobs give direct
access to Drive, Bass, Mid, Treble, Presence, Reverb, Tone Volume and Master Volume. The Tap Tempo button can be held down to
access a guitar tuner and each of the five stompbox types has its own button: Amp, Stomp, Mod, Delay and Verb. The USB2 socket on the
rear allows computer recording in stereo of both wet and dry inputs simultaneously.
As touched upon earlier, the four controller knobs beneath the adequately large, back-lit display
access the functions displayed on screen, where you’ll also find a block diagram of the active
patch. The cursor control disk to the right of the screen steers the cursor, while the turn-and-press

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Line 6 Pod X3 Pro

knob to the left is used for choosing patches and for saving new settings. Flat, recessed buttons
above and below the cursor disk allow effects to be tuned on or off or to be accessed for editing
and also get you into the output setup options. Similar buttons above and below the left-hand
knob get you back to the home patch page if you get lost in the menu system, and also get you
into input select mode.
The quality of modelling amplifiers is very subjective, but I think Line 6 do a great job here. Their gently overdriven sounds are as good as
any I’ve heard from a modelling preamp: clean sounds and very dirty sounds are easier to get right, but even mildly dirty sounds here have a
real sense of power and energy. You can also do some neat tricks using the dual mode — for example, using a highly compressed, cleaner
arrangement in one channel, combined with a dirtier sound in the second to get great sustain without too much filth. The effects are simply
excellent, and although the factory presets include the usual range of classic guitar sounds, there are some intriguing experimental sounds
that really show off the range of the device, from floaty and ethereal to downright evil! The mic preamps are clean and very competent, and
the instrument inputs are impressively quiet. By using the vintage preamp models, the clean mic inputs can be given a convincing vintage
warmth.

Overall
For studio use, the extra flexibility of the Pod X3 Pro is well worth the extra cost, and the sound quality is excellent, with minimal noise.
Having two built-in mic amps is a real benefit, as is the comprehensive I/O and ability to record both dry and processed sounds. I like the
intuitive nature of the hardware operation, but the free software adds greatly to its usefulness in the studio — and the graphical interface of
Pod Farm makes creating new patches incredibly fast and straightforward. The sounds that can be achieved are no different to those you
can get with the Pod X3 or the ‘full-fat’ version of Pod Farm — so you’ll really have to base your decision on whether or not the added
hardware features would make life easier in your own studio. Me? Well I like it! Paul White

SUMMARY
The Pod X3 Pro takes the same approach as previous Pod Pro units in giving you all the features of the standard model plus enhanced
routing and connectivity options. As a combined audio interface and guitar/bass/vocal processor it represents very good value, especially
when the additional software is taken into account.
Line 6 UK +44 (0)1327 302 700.
www.line6.com

Published in SOS June 2009

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Native Instruments Maschine

In this article: Native Instruments Maschine

Plug & Play Groove Workstation
Lock & Load Published in SOS June 2009
Pattern Heading Reviews : Drum machine
Print article : Close window
Not Just A Drum
Maschine
NI’s hybrid hardware and software beat machine promises the best of
Sample & Slice
both worlds. Does it deliver?
Effects & Automation
Performance Simon Sherbourne

M
Conclusions aschine is a beat-production workstation with built-in
Maschine Control drum sequencing, sampling and loop slicing, and is the
Groove Box latest result of NI’s initiative to build hybrid hardware
and software instruments. The software element runs as a stand-
On-board Sounds
alone application or as an AU/VST/RTAS plug-in, while the
Native Instruments hardware is a controller for the Maschine software, and also
Maschine 599 Euros functions as a general-purpose MIDI controller.
pros With Maschine, NI wanted to take the tactile, free-flowing
Excellent hardware integration production style of hardware instruments like Akai’s MPCs and
with no setup time. add the benefits of the computer-based music studio, and they’ve largely succeeded. Although the
Fast, smooth workflow. hardware has no sound-generating capability on its own, the tight integration with the plug-in gives a similar
Nice pads. feel to using a stand-alone hardware device. The workflow and feature set have also been finely honed,
Solid library of electronic drum although with some notable exceptions that promise to be addressed later this year.
sounds.
Great for performance.
Plug & Play
Step sequencing option.
cons Installation is via a single DVD for the software and library, and a
Can’t write arrangements in USB cable for the controller. You can then launch the stand-
real-time. alone app or insert the plug-in into your DAW of choice. Nearly
No host automation or MIDI all operations from this point can be performed from the
CC support, so you can’t hardware. The controller is sturdy, constructed mostly from
capture performances.
plastic, with a matt-black metal front panel. The buttons and
It’s difficult to change kits and
patterns independently. pads are all made of translucent hard rubber, similar to those on
You can’t sequence external Korg’s Pad Kontrol, and can all light up. Above the main pads
MIDI gear/software — yet. are twin displays and a strip of eight knobs and buttons which The Maschine software, with Scene
No groove quantising. make up the main user interface. manager at the top, device controls
No REX support — yet. in the centre, and
The first thing I did was explore some of the NI factory projects
No real support for non-4/4 pattern/sequence/sample editor at
to get a feel for what Maschine can do. These examples are
time signatures.
complete productions constructed within Maschine. An instance the bottom.
summary The Maschine software, with Scene
of Maschine offers eight Groups, each of which contains up to 16
Maschine shows that fast, manager at the top, device controls in
sounds, so you can layer numerous parts, including kits,
hands-on, musical workflow is the centre, and
instruments, loops or single samples. In Maschine, each sound
not the preserve of stand-alone pattern/sequence/sample editor at the
hardware boxes. If some of its element is essentially a sampler, which can contain a single bottom.
early issues are addressed, sample or a multisample map. Drum kits each take up a whole
Maschine should become the Group of 16 sounds, while Instrument patches and loops only
last word in beat-production use a single sound slot.
workstations.
Lock & Load
information
599 Euros. The first step in most Maschine projects is to press the Browse
Native Instruments +49 30 button to load a kit, loop or sound from the library. When
61 10 35 1300. browsing, the left screen lets you filter the library by category,
Click here to email and choose which level of Maschine’s structure to load into
www.nativeinstruments.com (Master, Group or Sound). This system is powered by the
Library’s Kore-style tagging system. Maschine lists the potential
Photos too small? Click candidates on its other screen, and provides controls for
on photos, screenshots and selecting and loading your chosen kit. All this is mirrored in the An instrument sequence, with effects
diagrams in articles to open a software if you have the Browser column displayed. automation.
Larger View gallery. An instrument sequence, with effects
Once a kit is loaded, its 16 samples can be played from the
automation.
pads. I found the pads a pleasure to play, with a sensitive,

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Native Instruments Maschine

consistent response. The velocity sensitivity can be tweaked from

the plug-in’s preferences. An MPC-style Note Repeat button allows you to auto-trigger sounds at a chosen
rate, and this combines brilliantly with the pressure sensitivity of the pads: you can hold down a pad with
Note Repeat, then vary the pressure you exert to add dynamics. This is great fun, and if you are recording
a pattern, it all gets captured.
The library has a category for Patterns, although none are supplied. More seriously, you can’t change a
kit independently of the current patterns: if you load in a kit, it wipes all the patterns that existed in the
Group. The only thing you can do is save out each pattern, change the kit, then load them all back in again.
This is a significant oversight: you should be able to audition different kits with your recorded patterns.

Pattern Heading
To write patterns from the pads, you can either play them live, or
use the Step Sequence mode. Holding down the Pattern button
shows existing patterns on a grid on the right-hand screen,
where you can select them with the pads. Pattern length can be
changed quickly with a knob, and there’s also a handy Double
button, which doubles the pattern length and duplicates any
existing notes. Unfortunately, there’s no real support for non-4/4
time signatures at the moment: if you change the meter, the
loops in Maschine are no longer whole bar lengths.
The workflow for recording patterns is spot-on: while Maschine
is looping, simply hit the Record button and play. To rehearse an
overdub, disarm the Record button, then punch back in when Measuring 320 x 295mm, the
you’re ready. If you make a mistake, Maschine has its own Undo Maschine hardware control surface
function, or you can hold the Erase button at the same time as is about the size of a vinyl record
other pads, to remove notes in real time. Input Quantise is sleeve.
available, or you can quantise after a record pass with 50 or 100
Measuring 320 x 295mm, the Maschine
percent strength. The 50 percent option works cumulatively: hardware control surface is about the
multiple presses nudge the notes ever closer to the grid. size of a vinyl record sleeve.
The bottom section of the Maschine software interface shows
the current pattern. One slight irritation is that the software display doesn’t zoom automatically to reflect the
pattern length, so you have to grab the mouse and zoom out for patterns longer than one bar. Patterns can
be edited or created directly in the software, using a simple left-click to add notes, right-click to delete
scheme.
If playing pads is not your thing, the controller has a step-sequencing mode, which utilises the pad grid to
represent one bar of 16th notes in your loop (you can change this resolution). The right-hand screen allows
you to select which sound’s sequence is shown, and navigate through loops longer than 16 steps.
Touching a pad adds a trigger at that step. The pads light up in sequence to show the current position,
running from bottom to top.

Not Just A Drum Maschine

In addition to drum kits, Maschine has a library of sampled
instrument patches and loops. The quality is pretty good,
certainly better than you’ll find on hardware beatbox equivalents,
and each sound engine has enough sound-shaping elements,
like filters, envelopes, velocity assignments, effects and so on, to
bring these basic sampled sources alive.
Instrument sounds can be played from the pads if you’re out
and about with Maschine and a laptop, but in the studio you can
use any MIDI keyboard. Whichever sound slot is selected in
Maschine responds to incoming MIDI information, and as well as
playing instrument patches you can use this feature to play drum
sounds in a kit across different pitches. Each Sound in Maschine has its own Piano Roll display where
pitched sequences are recorded and edited.
With this exception, Maschine is a closed environment as far as MIDI is concerned; all sequencing must
be done internally. You can’t use Maschine simply as a sound source and sequence its kits and
instruments using your DAW. This might seem surprising, but when you stop and think about it, Maschine
is designed specifically not to work like this, in contrast to other drum sound plug-ins available, including

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Native Instruments Maschine

NI’s own Battery.

However, what is an issue is that Maschine can’t output MIDI data (other than beat clock) to external
devices, so you can’t currently use it as a stand-alone production hub like an MPC. NI say this is coming in
a future update (seems a good bet, as there’s a MIDI out port on the back!). Hopefully, it will also work in
the plug-in version, because it would be really cool to use Maschine’s sequencer and workflow to control
other software instruments, such as Ableton’s Drum Racks.

Sample & Slice

The final tricks up Maschine’s sleeve are sampling and beat slicing. The Maschine library has a
(disappointing) collection of loops, split into three categories: Guitar, Synth and Percussion. These are all
audio samples that have been beat-sliced, Recycle-style. Dropping any of these into a sound slot loads the
sample map, along with a MIDI pattern that plays back the loop in its original sequence. As with all beat-
slicing systems, this has the dual advantages of locking the loop’s tempo to the song, and allowing
individual hits to be played back via MIDI in new patterns.
Unfortunately Maschine doesn’t support REX files, but the word is that this is coming in the future. You
can, however, import standard audio loops, or record with the built-in sampling functionality. The sampling
is nicely done: simply route some audio to Maschine from external devices or other tracks in your DAW.
You can also resample Maschine’s own internal sources. The on-board sample editor module can record
on-the-fly or sync’ed to tempo. You can also specify a loop length if you want, then hit Start and sit back —
no topping and tailing.
Sampled or imported audio can be sliced at 16ths or on the transients. Maschine will automatically
generate a note sequence to play back the loop, and drop this into the first pattern slot. Alternately, you
can go old-school and re-pitch the sample to fit the tempo of your song. There’s no time-stretching option.

Effects & Automation

The Master section, Groups and individual sounds each have two insert effect slots. Groups and Sounds
also have two auxiliary outputs. Send effects can be set up internally, by using a spare sound slot as a
return channel, or you can route signals out to external buses for processing in your host DAW system.
There’s a good selection of standard and more unusual effects on offer, with patches also available in the
library.
One of my favourite features of Maschine is the automation system. All sound parameters, effects and
mixer controls can be modulated within patterns, and each pattern can have its own unique automation
data. All you have to do is hold down one of the function buttons while Maschine is playing back, and any
control movement will be recorded and looped. This automation is relative to the position of the control, so
you can still make adjustments later.

Performance
Maschine does an exemplary job of capturing patterns and sequences, and manipulating sounds and
effects. So how about creating a performance or arrangement with these building blocks? Pattern changes
can be triggered easily from the pads with various quantisation settings. The eight Groups buttons allow
you to move quickly between your layered sounds, and you can hold the Mute or Solo buttons while
pressing these, to drop out sections of your track. Individual pads can also be muted or solo’ed in the
same way. Effects can be loaded from the Browser on-the-fly, and automated. As live performance tools
go, it couldn’t get much more fun or immediate than this.
To facilitate both performance and song arrangement, Maschine uses a system of Scenes. A Scene is
simply a snapshot of which pattern is playing in each Group at a given time. So, for example, Scene 1
might be an introduction section, with pattern A1 playing in Group 1, pattern A4 playing in Group 2, and so
on. You might then set up another Scene appropriate to Verse 1, and so on. Scenes can be recalled in
real time in the same way as patterns, giving you control over the high-level structure of an arrangement.
OK, but what if you simply want to capture an improvised performance as an arrangement? (Sound of
needle scratching across record). Surprisingly, there’s no mechanism for this within Maschine, and very
little within a host. Pattern changes cannot be recorded on-the-fly, and the Maschine plug-in has no
support for MIDI modulation or automation from a DAW.
The only thing Maschine responds to from the outside world is Program Changes, which can be used to
trigger Scene changes. Therefore, to create an arrangement, you need to set up Scenes, then place
Program Changes in your sequencer at the appropriate transitions in the song. This isn’t so horrible in
Ableton Live, as you can create Clips which trigger Program Change messages. In other packages it’s not
brilliant. To record Scene changes in real time you need to download a MIDI control template for Maschine

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Native Instruments Maschine

that lets you record Program Changes in your host. This convoluted situation feels more like a workaround
than a well thought-out workflow.
As an alternative, you can set up a string of Scenes in Maschine and have them play back in step with
your song. However, the real issue is that you’ve fallen back to editing and thinking about arrangement on
a screen. And this still doesn’t allow you to capture mutes, solos, and linear control changes, or to drop in
a quick fill without triggering an entire Scene change. (Ideally you should be able to trigger and record
Pattern changes independently of Scenes, as you can in FXPansion’s Guru, for example).
Reassuringly, NI are aware that this system is not ideal, and are already looking to improve it. The
summer should bring an update to make recording Scene changes easier, and after that, hopefully, more
control of other aspects of Maschine may follow.

Conclusions
Maschine is the best example of a hybrid software-and-controller instrument I’ve seen. In most cases
hardware is developed to control existing software, which usually results in compromise and frustration.
Here, the hardware experience has been developed from the ground up, and the result is a convincing
illusion that you are using a stand-alone device. Playing the pads, recording patterns and creating
improvised performances is fast and fun, and, most importantly, allows you to stay in musician-mode
rather than technician-mode.
Maschine has a rich feature set: a great library of electronic drum sounds, strong effects, on-board
sampling and slicing, sampled instruments... Unfortunately, it also has weak song-arrangement facilities
and host integration. Although configuring and triggering Scenes works to a point, it doesn’t support the
fast creation of interesting rhythm tracks that a groove box like this should. Hopefully this, along with the
other niggles, will be addressed soon, and with any luck NI will have learned from the Kore 2 experience
and make sure early investors don’t face paying an upgrade fee to get these key features.
I’m cautiously enthusiastic about Maschine — it’s so close to being brilliant. What it does well, it does
really well, and it would be hard to go back to a generic pad controller after working with Maschine. If, like
me, you get frustrated programming in a traditional sequencer environment, you should give Maschine a
try. I can’t wait to see where it goes next.

Maschine Control
With a quick button press, the Maschine hardware turns into a general-purpose MIDI pad controller, and
a seriously good one at that. A deep programming and librarian utility ships in the box. You can switch
instantaneously between any of your templates from the front panel. In MIDI mode, the Group buttons
allow you to switch the pads between eight different sets of MIDI assignments, and the arrow buttons
step through any number of pages for the top strip of eight knobs and buttons. It’s good to see that this
aspect of Maschine is not just an afterthought.

Groove Box
Drum machines often need a bit of help in the ‘feel’ department, with a little added swing going a long
way. Maschine has a global Swing knob, which adds 8th-note swing to everything. You can also apply
individual swing settings to each Group, although this is in addition to the global swing. You have a little
more control here, being able to determine the swing resolution, although the only one that usually
sounds musically useful is 8ths again. You can also invert the swing. Unfortunately, that’s about the
extent of it at the moment. There’s no groove quantising system, or ability to create user groove
templates from loops or patterns. This is a shame, especially as many of the supplied loops have groove
in them. The only way to match these is to hard quantise them.

On-board Sounds
As well as many old favourites from Battery, there appears to be a good supply of new drum kits to get
your teeth into in Maschine’s 5GB library. All flavours of contemporary electronic and urban music are
well catered for, and there’s even a dozen or so decent acoustic kits with multiple velocity zones. There
are enough individual drum samples to keep most of us going indefinitely (900+ kick drums, 700+
snares). The 300 or so instrument patches cover a nice range of acoustic and synthetic sounds,
although, quite rightly, electronic, dance and urban sounds are favoured. The loop library is so limited
that it almost seems as though it’s just examples at this point. I predict that NI plan to go for a few add-
on Euros and push out optional sound packs some time soon.

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Native Instruments Maschine

Published in SOS June 2009

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Novation 61SL MkII

In this article: Novation 61SL MkII

Touch Of Class Controller Keyboard
In Use Published in SOS June 2009
Reviews : MIDI Controller
Novation 61SL MkII Print article : Close window
£500
pros The updated SL range rejoices in touch-sensitive knobs and sliders, and
Touch-sensitive knobs and in Novation’s ingenious Automap technology.
faders.
Mike Senior

F
LED encoder rings and
backlit buttons provide lots rom their origins as synth designers, Novation have, in more
more visual feedback. recent years, carved out an enviable niche for themselves
Streamlined LCD menu as manufacturers of hardware controllers for budget-
structure. conscious musicians. Under review here is their latest 61-note
cons knobular controller keyboard, the 61SL MkII. As is apparent from
One of the two previous LCD the name, this has evolved from the already popular first-
read-outs has gone walkies. generation Remote SL controllers and incorporates a number of
summary innovations (such as the clever Automap automatic controller
This new version of the SL, assignment utility) from other recent Novation products such as the
supercharged with touch- Nocturn. As with the old Remote SLs, there are four models to
sensitivity and twinkly lights choose from: 25-, 49- and 61-note keyboard versions, and a Zero
from the Nocturn, presents a version that offers the controls without a keyboard.
very appealing package for
anyone wanting hardware The front-panel controls are almost the same as those of the
control of their sequencer, previous generation, excepting, of course, that eight of the knobs
plug-ins and virtual have been replaced with continuous rotary encoders. The back
instruments. panel, however, is identical to the older SLs and features a 9V
power input, USB port, sockets for control and expression pedals,
information
and four MIDI ports: the usual In, Out and Thru, plus an extra Out.
61SL MkII £499.99, 49SL
MkII £399.99, 25Sl MkII
Touch Of Class
£329.99, ZeroSL MkII
£329.99. Prices include VAT. The most significant update is that all the knobs and sliders are
Novation +44 (0)1494
now touch-sensitive. You probably don’t need me to tell you that
462246.
this provides a big leap forward in speed of use straight away, and
Click here to email
Simon Sherbourne has already waxed lyrical about this aspect of
www.novationmusic.com
the Nocturn when he reviewed it in the August 2008 edition of
Photos too small? Click SOS. Backlit buttons and LED rings surrounding the endless rotary
on photos, screenshots and encoders transmit a tremendous amount of information to the user,
diagrams in articles to open which is likewise of enormous help. Although the other knobs and
a Larger View gallery. faders don’t provide this visual feedback and hence, like all
unmotorised controls, have the potential to be in a position that
doesn’t reflect the value of the software parameter that they’re
controlling, Novation have at least dealt with this situation fairly The SL MkII’s back panels offer the
elegantly: either the controls can ‘pick up’ the parameters the same connections as the previous
moment they’re moved (sensibly, not just when the control is SL models.
touched), or the parameter can wait until the physical control is The SL MkII’s back panels offer the
moved through its current value before latching on. same connections as the previous SL
models.
Against all this, though, has to be set the fact that the 61SL MkII
does away with one of the two backlit LCDs of its forebears,
leaving only the one above the rotary encoders. Novation valiantly attempted to spin this as offering “more
focused menu operation”, but I reckon I’d take my chances with menu operation blurring (whatever that is) in
return for a second display lined up above the right-hand bank of faders and buttons, because I found myself
forever grabbing the wrong control by mistake, and felt the need to pause for a moment before pressing
buttons, to be sure I had the right one. A cost-cut too far, I’d say, as it partially undermines the most
important characteristic of the unit: quick and easy usability.
Much as I might poke fun at the phrase “focused menu
operation”, Novation have actually overhauled the Remote SL
menu structure for the MkII versions, and while I couldn’t
personally compare it with the original Remote SL, I did find it very
easy to navigate — more so, in fact, than on any other LCD-based

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Novation 61SL MkII

hardware unit I’ve used. What’s particularly good is that each of

the eight parameter positions in the display can be controlled by
the rotary encoder for rapid value adjustment and the two buttons
above and below it for increment/decrement, depending on which
is most suitable for that particular parameter. You can also just Novation’s Automap application
touch the encoders to navigate through the menu layers, which is makes it easy to map the 61SL MkII
quick as the wind. What a shame this kind of hardware control to your software’s controls.
wasn’t economically viable 20 years ago when hardware samplers Novation’s Automap application makes
(remember them?) were in the ascendant... it easy to map the 61SL MkII to your
The additional Quick Menu is nifty too, giving you one-button software’s controls.
access to Bank Select and Program Change messages, keyboard
transpose settings, and the global tempo setting in your software. Dedicated buttons also allow quick access
to the Automap server application, its Learn mode, and its various Browser pages. The operation of the 61SL
MkII’s Automap functionality is very similar to that of the Nocturn, so rather than retreading old ground here,
I’ll refer you to Simon Sherbourne’s detailed Nocturn review for the full run-down on that — and for
descriptions of how the Speed Dial and generic MIDI controller facilities work as well.

In Use
My impressions of using the 61SL MkII with both Steinberg Cubase 5 and Cockos Reaper software were
extremely positive. The touch-sensitive controls, speedy controlled assignment, and masses of visual
feedback all added up to a very pleasant experience.
Some users may find that the closer-packed control spacing makes the touch sensitivity a slightly mixed
blessing, given that it responds to the lightest of touches — it’s easy to divert the LCD display by accident
while hitting the buttons, for example. However, this is a pretty minor niggle given the obvious benefits of the
touch-sensitivity overall and the availability of the more spaciously laid-out Nocturn. It also seems churlish to
have too much of a pop at the rather haphazard initial Automap assignments (which Simon also found when
working with the Nocturn), given that Novation are at the mercy of plug-in developers and Automap’s initial
assignments can so easily be changed and stored for future use. In fact, the lack of a second LCD screen is
the only gripe I can really muster with any enthusiasm.
What’s beyond doubt is that, overall, the 61SL MkII is a very worthwhile update to the Remote SL concept,
presenting a pretty mature vision of what a hardware controller for the masses should offer: affordable and
near-universal control that is both fast and simple to use. As such, it deserves to win Novation a big crowd
of new friends.

Published in SOS June 2009

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Philtre Labs Bollywood Elements

Philtre Labs Bollywood Elements

WAV/Apple Loops
Published in SOS June 2009
Reviews : Sample/Sound/Song Library

W
Printer-friendly version

ith Bollywood now a massive force in the world movie market, India-based Philtre Labs have followed up last year’s release of
Bollywood Grooves (SOS March 2008) with a new title, Bollywood Elements. The former title was based on performances from
various rhythm ensembles, but Bollywood Elements focuses on melodic instruments, so the two libraries form a nice
complimentary pair. All the loops are once again presented in 24-bit, 48kHz format (48kHz being the ‘norm’ in the film and TV world),
although they’re presented in mono rather than stereo. Over 1.3GB of sample data is provided, covering over 1000 loop and one-shot files.
A good selection of traditional instruments associated with Indian music have been sampled.
These include a Bulbul tarang (Indian banjo), Bansuri flutes, a Mandolin, the wonderfully
expressive and evocative Sarangi, the Sarod (similar in sound to a Sitar, but more mellow), Sitar,
Shehnai (a reed-based instrument, not unlike a raspy Oboe), Tumbi (a high-pitched single string
instrument), Ravan-hatta (a string instrument with percussive bells) and a Shankh (conch-shell).
The samples are organised into folders by instrument, and in most cases both loops and one-shot
phrases are provided. The latter include phrases that are not played strictly to tempo, and
therefore retain a more ‘human’ feel.
All of the instruments are well played, and there’s plenty of genuine Indian atmosphere. I
particularly liked the Tumbi samples: the almost scratchy, banjo-like tone and the short sustain
make it ideal for repeated phrases to sit under other instruments or a vocal. As with many of the
other instruments, the filenames give useful information on the original tempo and key. My other
favourite is the hauntingly beautiful Sarangi. There are some fabulous phrases and, whether
they’re solo or mixed with some subtle percussion, an instant mood is created.
If you prefer the construction-kit format, Bollywood Elements might not suit, as you’ll have to
work a little harder to blend the phrases together, and need some Indian percussion if you want to
create a complete performance (obviously the intention with the two titles). The only other quirk is
the mono file format, especially as many sounds contain ambience, but in practice I didn’t find it a
limitation. Whether you’re seeking a touch of genuine Indian influence to add melodic spice to a
pop or dance track, or want to create a full-blown Bollywood arrangement, this is a very good
place to start. John Walden
£69.95 including VAT.
Time + Space +44 (0)1837 55200.
www.timespace.com
www.philtrelabs.com

Published in SOS June 2009

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Plug-in Folder

Plug-in Folder
Reviews Buy PDF
Published in SOS June 2009
Reviews : Software: ALL
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D16 Silver Line

Formats: Windows VST, Mac VST & AU
D16 Group are a Polish development team whose plug-ins have an old-school flavour, but their
Silver Line effects and processors stand out from the crowd of vintage emulations. D16 appear to
be less interested in your classic Neves and Pultecs, and more in the design quirks and
technological limitations that gave older electronic instruments their individual qualities.
Reducing the bit depth and sample rate of a digitised signal is a fairly simple business, so it’s a
source of constant wonder that it can yield such individual and widely varied results. There are
still samplers and drum machines from the ’80s that command high prices because of the
particular brand of crunchiness they impart to the source, and of all the innumerable ‘bit crushing’
effects, no two ever seem to sound quite the same.
D16’s Decimort is the first such plug-in I’ve encountered that includes presets emulating the
sound of classic hardware like the Akai MPC60, Emu Emulator, Casio FZ1 and so forth. The
signal path begins with a preamp control, which imparts a crunchy, solid-state clipping distortion
when you ramp it up, before the left and right channels pass through independent but linkable bit-
depth and sample-rate reduction stages (I’m unsure why you’d ever want to unlink them,
personally). These are followed by an anti-aliasing filter and a conventional resonant filter that
offers all the standard filter responses.
Not being lucky enough to own an MPC60 or similar, I can’t vouch for Decimort’s accuracy in
emulating classic samplers, but I was very impressed by the sheer range of nasty noises that you
can make. All manner of fizzing, ringing, clanging and crunching effects can be applied to almost
anything; drums are the most obvious candidate, but if, like me, you retain a weird nostalgia for
the sound you got by plugging your first electric guitar into the mic socket on your parents’ music
system, the preamp distortion will get you surprisingly close. Ugly, but mostly in a good way.
Devastor likewise sounds like a sinister figure from a JK Rowling book, but is actually a
distortion effect. It’s billed as a multi-band plug-in, but isn’t, at least as I understand the term. There’s only one distortion element in the
signal path — a preamp stage similar to that found in Decimort, offering an emulation of analogue ‘diode clipping’ distortion, with Threshold
and Knee controls. The multiplicity resides in the addition of three separate filters, again similar to those found in Decimort. Any or all can
be placed before or after the clipper in the signal chain, and where two or three of them are in the same position, they operate in a parallel
configuration.
Again, many of the presets are designed to emulate the output stages of various pieces of vintage hardware, in which capacity they
generally do a fine job of adding a certain wonky character to the source. Ramp up the clipping and you can get some absolutely brutal
fuzztone effects, and although the presets called ‘Guitar Tube Amp’ ought to win some sort of award for sounding absolutely nothing like a
guitar tube amp, there is certainly something on offer here for adventurous guitarists, especially if you can pair Devastor with a cabinet
simulator to tame some of its innate fizz.
Redoptor, meanwhile, turns out not to be a small but vicious predatory dinosaur of the late Cretaceous era, but another distortion plug-in.
A preamp gain control with high- and low-cut filters precedes a valve distortion emulation with Tube Bias, Tone and Brightness controls,
which is followed by a four-band parametric EQ. Although some of the presets use this EQ to try to mimic the frequency response of a
guitar amp cabinet and speaker, the emphasis is, again, on other types of valve gear. Not all the controls do what you’d expect — Tone, for
example, starts off thin and trebly and gets progressively more mellow as you turn it up — but together they allow you to replicate anything
from barely noticeable warmth to almost pure white noise.
Last in the Silver Line collection at present is the Fazortan phaser, which, well, phases. It’s a warm-sounding, convincingly ‘vintage’ effect
that does pretty much what you’d want from a phaser. Many DAWs come with pretty good bread-and-butter effects these days, but if you’re
unhappy with the modulation effects in your own sequencer, Fazortan would be well worth a try.
Overall, I like the Silver Line collection a lot. The user interfaces are simple yet classy, with some of the nicest virtual 3D knobs I’ve yet
seen, and for a modest price, these plug-ins offer something that is authentically ‘vintage’ yet also rather different from the bulk of what’s
out there. Sam Inglis
119 Euros, or 35 Euros for individual plug-ins.
www.d16.pl

Sonic Charge Synplant

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Plug-in Folder

Formats: Mac/PC VST

Described as having a “biological approach to sound creation”, Synplant’s beautifully designed
graphic interface is a huge departure from the norm, featuring a spherical Bulb containing a
randomly-generated ‘Seed’. From this you can grow gently waving Branches, one in each of the
12 different directions of the outer Key Ring. Each branch corresponds to one note in the musical
octave, and the longer you drag a branch from the original seed with your mouse, the more the
sound parameters of that note in every octave alter from the seed sound.
The idea is to refine the branches you prefer, so if, during your sonic journey, you grow one with
a sound you particularly like, you can Clone it to all branches so that every note of the scale plays
the same sound, and then further refine each one. The mod wheel can grow or retract all
branches simultaneously, and you can at any time plant a new, randomly generated, seed to start
afresh.
There are various more traditional controls around the bulb and beneath it, but Atonality is
certainly new. In essence, it sets the balance between musical tones and weird effects, stretching
scale tuning and modulation depth, and generally introducing more mayhem to the proceedings
with larger settings.
Another collection of more traditional controls appears when you click on the Manipulate Genes
button, overlaying the main display with a DNA-like helix, each component of which acts as a
slider for a parameter. I counted 37 parameters in total, some with familiar names, others rather
more obscure, although I can tell you that this synth has two oscillators per voice with morphing waveform options, FM and sub-oscillator
options, two filters in parallel, a complex envelope generator, plus various LFO and sample & hold modulators, and is surprisingly versatile.
By now, some of you must be convinced that this is an April Fool concoction, but nothing could be further from the truth. Synplant’s
creator is Magnus Lidström (who created Reason 2’s Malström synth), and his mission with Synplant is to help those who want to develop
the perfect sound using their ears, rather than by blindly clicking through hundreds of presets.
So what type of sounds can it grow? Well, Synplant offers 550 factory presets covering basses, leads, pads, and loads of evolving sounds
and sound effects. However, by adopting this novel interface, many of the parameters can have complex relationships with each other, and
Synplant definitely has its own unique sound character, ranging from traditional synth sounds through soft organic creations to harder-edged
electronic ones. Particularly distinctive are its looped-attack envelopes, used to good effect with repeated attacks and complex pitch-bends,
its FM modulation and sample & hold effects, while having up to 12 different branches per seed allows unusual sounds where each note
you play has a slightly different timbre, or triggers an entirely different percussion sound.
There’s a rotation control that alters which note is allocated to which sound, and automating this lets you trigger ever-changing sounds
while playing the same tune. Automation of the mod wheel to grow and retract the branches in real time also produces some stunning
effects. You can also assign your choice of MIDI controller to any of the controls, for more hands-on horticulture. Some might find this GUI
too novel, since you never quite know what sound might emerge next, but if you have a boring patch that deserves some tender loving
care, Synplant will definitely grow on you. I’ve already harvested a crop of unique sounds ready for my next album! Martin Walker
£68 including VAT.
www.soniccharge.com

Published in SOS June 2009

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Pro Audio DSP DSM

In this article: Pro Audio DSP DSM

What It’s For Multi-band Dynamics Plug-in For Mac OS
How It Works Published in SOS June 2009
What It Sounds Like Reviews : Software: Effects+Processors
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Pro Audio DSP Dynamic

Spectrum Mapper £200 Pro Audio DSP’s first product is a novel dynamics plug-in, with
pros applications ranging from de-essing to loudness maximising.
Easy to operate. Paul White

T
Clear user interface.
Very versatile for mixing and he DSM, or Dynamic Spectrum Mapper, plug-in takes three
even mastering. processors with which many of us are already familiar —
cons the fingerprint equaliser, the dynamic equaliser and the
multi-band compressor — and combines them in an unusual way.
The process itself is not
immediately intuitive, so you In essence, DSM is a multi-band dynamics processor with the
need to think about what ability to calculate the threshold levels of the individual bands
you’re trying to achieve. according to the spectral response of a source or target audio file.
summary According to the manufacturers, DSM is “intended to provide multi-
DSM is an ingenious plug-in dimensional control over both the spectral response and dynamic
that can be used to match characteristics of audio programme, in order to bring a whole new
sounds (to a degree) or dimension of facility and artistic ability to the sound engineer”.
enhance them in a way not Sounds like no Time Lord should be without one! Currently, DSM
possible with other is Mac-only, and is available in RTAS and Audio Units formats,
processors. both protected via iLok. It was created by former Sony Oxford
plug-in designer Paul Frindle, and bears a noticeable graphical
information
resemblance to the existing Sonnox plug-ins.
£200.
www.proaudiodsp.com As with a conventional fingerprint EQ, the plug-in splits incoming
audio into multiple frequency bands (though we’re not told how
Photos too small? Click many) and is able to analyse and store the average frequency spectrum of a section of audio, be it a full mix,
on photos, screenshots and solo instrument or vocal, to act as a target response. A fingerprint EQ, such as Logic’s Match EQ or TC’s
diagrams in articles to open Assimilator, then compares the target spectrum with that of the audio actually being processed and adjusts
a Larger View gallery. the filter bank so that the average spectrum of the current audio matches that of the target. DSM operates
somewhat differently, as it uses the target analysis curve to set the thresholds for compressors in each band,
so that gain reduction occurs in each frequency band only when the current audio level exceeds the target
curve threshold in that band. If the compression ratio is set high enough to act as a limiter, then the
processed audio’s response will match that of the target’s, providing that the level is high enough to reach
the threshold. Audio below the threshold still retains its own spectral characteristics.
The threshold curve can be modified using three sets of familiar parametric EQ controls (these affect only
the threshold curve, not the actual audio) and also by a conventional single set of compressor controls.
Further control is available to change the response times of the compressors in a progressive manner across
the frequency spectrum, and the threshold curve can also be adjusted up or down using the compressor
Threshold control.

What It’s For

So why would you want to do this? Well, it turns out that there are lots of creative and corrective possibilities
for a plug-in of this kind, not least allowing the source sound spectrum to morph into that of the target
spectrum as it gets louder. It can also be used to emulate the softening effect of tape saturation, by lowering
the thresholds at higher frequencies and applying a suitable degree of compression. It can even handle de-
essing and de-popping with minimal audio side-effects, by using a target spectrum taken from a good part of
the vocal where no popping occurs. When a problematic section is reached, gain reduction will be applied to
those parts of the spectrum that are louder than normal — specifically, the unwanted sibilant or popping
frequencies.
Uses suggested by the manufacturer include: “Enhancing the perceived volume and presence of mixes,
compressing even the most difficult programme with increased accuracy and musicality, copying and
equalising sonic impressions between dissimilar sounding tracks.” The process is also said to be useful for
enhancing vocals to increase intelligibility or to match the sound of dialogue from different takes in music or
post production.

How It Works

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Pro Audio DSP DSM

The controls are split into four sections, and the graphical display window shows two curves: the Active
Spectrum of the input signal in red (or twin curves in blue, if the two channels are not linked for stereo
operation) and the threshold curve derived from the target audio file (which may be modified by the user) in
yellow. A Capture button is used to derive a spectrum from a section of the target file: this can be created
from a very short sample of audio, or the button can be held down to average the spectrum over a longer
time period. In Pro Tools, captured responses can be saved and applied to other projects. Freeze Gains
does as its name suggests and maintains the gain at the time the button was pressed, for situations where
static processing (more like a conventional fingerprint EQ) would be appropriate. A Limit button prevents
sample value overloads that might otherwise result in clipping, though it affects all bands, not just the ones
that are peaking, so some artifacts may be audible. A 16-bit dither option is available for mastering
applications.
The compressor section features the expected controls for Threshold, Ratio and Gain, plus Attack and
Release time constants. There’s also a Knee control that moves from hard-knee to soft-knee compression.
The attack and release time settings are common to all bands in the spectrum, but Timing Profile controls
have been added in order to vary these with frequency: LF Attack progressively increases the attack time for
lower frequencies, and HF Release progressively decreases the release time for high-frequency signals.
As touched upon earlier, there are parametric-style controls that are used to modify the reference (yellow)
curve, and any changes are reflected in the shape of the curve. An Options menu accesses preferences that
can change the way the controls and meters respond.

What It Sounds Like

The order of work is to first capture the reference or target spectrum, then play the audio to be processed,
adjusting the compressor controls (and the parametric controls, if required) to instigate the necessary amount
of compression when the audio exceeds the threshold curve. I found it simplest to paste a section of the
target audio at the end of the track I was processing so that I could capture it, then return to the song to
hear the result right away. Setting a lowish threshold and a very high ratio is the best way to hear what the
captured curve really sounds like before making the necessary adjustments.
I found the DSM controls easy to use, though I’d have liked part of the display to show the actual amount
of gain reduction taking place, perhaps by means of a shaded area under the current audio spectrum curve,
like the system used in Logic’s multi-band compressor. I also noticed that the curves created are far more
‘smoothed’ than the ones on typical fingerprint equalisers, so no really fine detail shows up.
On complete mixes, the results that can be achieved in sound-matching invariably depend on how similar
the styles of music are in the first place, but a benefit over conventional fingerprint EQs is that the spectrum
only changes when the audio level reaches the threshold — below that level it retains its own character. This
is useful, as you can create mixes that have lots of detail in the low-level sections but get smoother at higher
levels so as to prevent listener fatigue. I also worked with some vocal tracks and distorted guitar parts, again
with a fair amount of success, especially when I also made use of the three-band parametric section to
modify the curve to get the tonal result I was after. As a rule, you don’t get exactly the sound suggested by
the reference audio curve, but rather a flavour of it, and when used on mixes, the clarity and punch can be
enhanced quite significantly by using the parametric-style adjusters, without the end result sounding
obviously processed.
This is clearly one of those plug-ins that rewards experimentation, because it is far more capable that it
might appear from its description. It is very powerful as a dynamic enhancement tool and, of course, there
are those corrective uses, such as reducing sibilance and popping. The fact that the process is unfamiliar
and doesn’t have a single specific aim may make it a hard sell, but I think those users who embrace it will
find it a very useful tool.

Published in SOS June 2009

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Prodipe Pro Ribbon 8

In this article: Prodipe Pro Ribbon 8

Overview Active Monitors
Testing Published in SOS June 2009
Respectable Reviews : Monitors
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Ribbons?

Prodipe Ribbon Pro 8 Ribbon tweeters can yield a smooth sound, while still capably
£449 reproducing transient detail — and the Pro Ribbon range promises to do
pros so for an attractive price.
Good performance for a Paul White

W
sensible price.
Well-balanced, non- e’ve already looked at some of the microphones from
aggressive sound. French manufacturer Prodipe, but now the company
cons have expanded their range to include a new line of
active monitor speakers using ribbon tweeters (flat ribbons, not to
I have no complaints, bearing
in mind the cost of these be confused with the folded-ribbon type used by Adam).
speakers.
summary Overview
The Pro Ribbon 8s make Like Prodipe’s mics, their monitors are built in China, which helps
very effective project studio
to keep the cost down. The Pro Ribbon 5 and the Pro Ribbon 8
monitors at an attractive
price and without skimping are both active, two-way, front-ported designs, and, as the names
on bass response. They suggest, they have five-inch and eight-inch drivers respectively. They both employ the same rectangular
have a smooth, comfortable ribbon tweeter, and thus benefit from the usual ribbon characteristic of low mass — which equates to a good
sound yet you can still hear ability to follow high-frequency transients.
the high-frequency detail.
We had the larger Pro Ribbon 8 in for review, a model whose amplifier gives 140W of power to the two
drivers via a 24dB/octave active crossover operating at 2.4kHz. A frequency response of 45Hz to 30kHz is
information
specified, and a steep 35Hz low-cut filter reduces subsonic content that would otherwise eat up headroom
Pro Ribbon 5 £359; Pro
unnecessarily.
Ribbon 8 £449. Prices per
pair including VAT. The general look of these speakers reminds me of the Samson Rubicons (another budget monitor with a
Etcetera Distribution +44 visually similar ribbon HF unit), but I’ve no way of telling if these tweeters come from the same source.
(0)1706 285 650. Physically, the cabinets are quite conventional. They measure 381 x 265 x 316mm, and the tweeter is
Click here to email recessed in a rectangular waveguide. The port takes the form of a slot beneath the bass driver, which has a
www.etcetera.co.uk distinctive dull-gold colouring to its glass/Aramid cone, and the power LED is of the now-familiar bright-blue
www.prodipe.com variety. Cosmetically, the speakers look purposeful but not over-ornate. Moulded baffles around the drivers
and plain, satin-black cabinetwork with rounded corners help to break up their otherwise utilitarian lines. The
Photos too small? Click cabinet material is almost certainly MDF, and it seems very rigid, with no obvious resonances when tapped.
on photos, screenshots and Each speaker weighs 13kg, but there’s no figure given for maximum SPL — not that this model seems in any
diagrams in articles to open way shy in that department.
a Larger View gallery.
As is now standard for such speakers, the rear panel plays host to all the necessary connectors, and also
provides a mounting point for the active electronic circuitry within. The panel layout is fairly straightforward,
with a range of analogue connection options, as well as the expected power switch and IEC mains inlet. A
slide switch allows operation at 230V or 110V. Unbalanced inputs are catered for via an RCA phono socket,
with balanced inputs on both a TRS jack and a conventionally wired XLR. Of course, either of these may
also be used unbalanced, as long as the correct cable is used. A volume control offers a -30dB to +6dB
range, and to allow the user to customise the high-frequency end to their own room and listening
preferences, there’s a rotary ‘HF Level’ control that can be used to cut or boost the output of the amplifier
that feeds the tweeter. You can choose between settings of -2dB, -1dB, flat and +1dB. There are no low-end
tweaks to allow for varying speaker placement, as you find on some active speakers, and these may be
missed by some users — but this also means that there’s less for the inexperienced user to mess up.

Testing
The Pro Ribbon 8s’ current retail price should make them a very
attractive proposition — providing, of course, that they deliver in
the sound-quality stakes. I tested the Pro Ribbon 8s on a recording
and editing session, as well as using them to listen to some
commercial material that I know very well, and was generally
impressed, given the price bracket within which these speakers fall.
They were mounted on the upper shelf of my mixing desk, sitting

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Prodipe Pro Ribbon 8

on Primacoustic speaker pads to decouple speaker vibrations from

the desk itself.
The ribbon delivers a welcome degree of high-end detail without
sounding aggressive, while the eight-inch woofer provides plenty
of depth to the sound. The overall tonality has a warm and easy-
going character, and a solid low end, but with the all-important
transient details coming across clearly. There’s more than enough
level when working close to the speakers, as would normally be
the case, and although a more costly monitor might sound a touch
better focused in the mid-range, I enjoyed working with these
speakers and never felt that they were misleading me.
All speakers have their own tonal signature, and when I switched
The rear panel includes balanced
to my Mackie HR624 Mk2s or Adam A7s there was a noticeable
and unbalanced analogue audio
difference, particularly in the mid-range (which actually made my
connections, switchable-voltage
own speakers sound a touch hard), but they all get the job done
power inlet, and volume and HF
and I felt that I could work all day on the Pro Ribbon 8s without
controls.
fatigue. And the more I used them, the more I liked them!
The rear panel includes balanced and
unbalanced analogue audio
Respectable Ribbons? connections, switchable-voltage power
inlet, and volume and HF controls.
When I first unpacked these speakers, I didn’t know what to
expect, but after having worked with them for a few days, I have to
say that I’m pretty happy with their sound and tonal balance. They work well for tracking and mixing, and
they’re especially impressive for speakers in this lower-mid price category. I particularly appreciate the fact
that the sound isn’t over-hyped or aggressive, as so many monitors seem to be these days.
As always, you can buy better, but in this case you may find that you have to spend a fair bit more to do
so. If you’re in the market for new active monitors that can handle the full audio spectrum in a smooth and
capable manner, but you can’t afford — or don’t want — to spend more than you have to, then the Pro
Ribbon 8s should go on your ‘must-audition’ list.

Published in SOS June 2009

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Sample Magic Minimal Techno

Sample Magic Minimal Techno

Multi-format Sample Library
Published in SOS June 2009
Reviews : Sample/Sound/Song Library

N
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o-one can accuse Sample Magic of scrimping on the sub-bass in their Minimal Techno loop and one-shot library: it lumbers forth
from the provided bass and kick sounds in abundance. A good thing it is, too, given the importance of the dub influence to this style
— but I was pleased to see that the producers have taken care to keep the subterranean monsters firmly controlled, to maintain a
tight rhythmic backbone and clear texture. Still, only the foolhardy would attempt to layer raw materials like these without access to proper
full-range monitoring!
Once you’ve picked your mouse up off the floor, you’ll notice that there’s lots of other good stuff
going on in here as well. In a genre that combines pounding rhythmic repetition with a disregard
for ‘whistlability’, productions can stand or fall according to the imaginativeness of their synth and
effects design, so Sample Magic have rightly concentrated considerable effort in this department.
They reap the rewards in the form of apparently simple parts, whose inner details withstand
repeated listening. Featherweight snares and clicky percussion toy with your expectations without
losing the general plot; pitch modulations and real-time controller programming steer a selection
of skittering and frequently delay-garnished rhythm-synths well away from the realms of
complacency; and whole boxes of virtual patch-cords appear to have been applied in the pursuit
of sinuous filthiness in many of the bass-synth lines. In addition to the inventively textured
programming, the synth lines in general score well in terms of raw sonics. Somehow the
designers have managed to imbue these sounds with solidity and power, despite pseudo-random
glitchiness and sawtooth rasps aplenty.
Usability is pretty good, with plenty of supported formats and a sensible tempo-delineated folder
structure, although I’d have liked a bit more text description of the sounds and effects used in
each loop. This is particularly the case with the drum loops, where each loop comes in several
variations: it would be handy to know which is the ‘no kick’ version, for example.
Even if minimal techno isn’t your primary poison, there’s a lot that could be applied more widely across electronica styles, especially
where a down-the-line groove needs spicing up. The broad line-up of seismic basses and kick drums is another asset ripe for repurposing
into just about any track requiring both smart attack and capacious low-end. Overall, then, this is a good library, which not only provides well
for its chosen genre, but would also find applications beyond it, and with 850MB of sample material and very little filler, it also represents
good value. Mike Senior
£58.67 including VAT.
Time + Space +44(0)1837 55200.
www.timespace.com
www.samplemagic.com

Published in SOS June 2009

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Sequis Motherload Elemental

Sequis Motherload Elemental

Dummy Load & Speaker Emulator
Published in SOS June 2009
Reviews : Speaker Simulator

T
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he idea of a combined dummy load, speaker emulator and speaker attenuator to enable powerful guitar amps — especially those
with valve output stages — to be DI’d for recording or feeding into a PA is very appealing. The only problem is that in most cases
the resulting sound doesn’t really resemble what you’d actually get from miking your amp. One solution that comes extremely close
to getting it right is the Sequis Motherload, and now the company have launched a somewhat simpler and less costly version called the
Motherload Elemental, which retails at $799 in the US. This retains the same principles and concepts as the ‘full-fat’ model, but offers fewer
adjustable parameters. The Elemental is available in 4, 8 and 16? versions, and even a 2? version can be built to special order (but 8? is,
of course, the most common speaker impedance).
Supplied in a robust polyurethane carry case, the Motherload Elemental is built into a sturdy
steel casing, with four chicken-head pointer knobs and a switch on the front panel. On the rear
panel are an XLR socket, a couple of push-button switches and no fewer than nine jack sockets...
but in order to preserve some sense of mystery and anticipation, the purpose of (most of) these
will be revealed later in the review. There’s no power socket, because no power is needed — the
Motherload Elemental is basically a passive filter, with an input impedance of 8? and the ability to
dissipate the full power output from amplifiers rated at up to 100W. In fact, the load is very generously rated in this respect, as many tube
amps produce more power than their rating suggests, especially when used with heavy distortion. The Elemental is designed with a more
than adequate safety margin to accommodate this.
The speaker input for when the internal load is required should be connected via a speaker cable (not a screened guitar lead!). When
using this socket, the internal load takes the place of your speaker, so there’s no need for any speaker to be connected — although you
can, if you wish, connect up to two speakers to the rear-panel Speaker 1 and Speaker 2 jacks, as long as the total load doesn’t fall below 8?
. There’s a front-panel knob and a high/low switch that’s used to adjust the level going out to the speakers, so you can get a fully wound-up
amp sound at a much lower level (adjustable right down to zero) for smaller gigs or for studio use. These attenuated outputs are perfect for
monitoring or for enabling you to mic up a small, good-sounding cab — a useful arrangement in the studio — but if you want to use your
amp as normal on stage with its speaker cab and also take a DI feed to the house PA rather than using a microphone, then the Thru, In and
Out sockets are the way to go, as the ‘to speaker’ signal is not affected in any way. These jacks enable an amp of any output impedance to
be linked through to its speaker so that the Motherload Elemental can create a DI feed without using its dummy load. A pair of LEDs, one
green and one red, serve as signal present and ‘overcooked’ warnings.
The XLR and balanced jack outputs carry the speaker-emulated sound and are at a nominal
level for connecting directly to a mixer or studio preamplifier. The XLR output level can be
adjusted on the front panel using the XLR Out knob, while the jack output is always available at
full level (it is not affected by the front-panel level control). Separate ground-lift switches are
provided for the screens on each of these two outputs, and there’s also a chassis ground
terminal. Other rear-panel jacks provide a send and return for effects devices operating at a A peek at the rear panel reveals many
more outputs than you’d typically expect
nominal line level. A further socket labelled Remote Shifter is not mentioned in the user guide, but
on a dummy load, which means that
a phone call to the company revealed that it’s there to support an additional — and as yet you have a greater range of options.
undisclosed — accessory product due for release at a future date.
So far so good, but the way a speaker cabinet is designed can have a huge effect on the final
sound, so how does a box like this go about emulating them all? Of course it can’t, but it does provide the user with two very valuable
controls that allow the sound of the cabinet emulation to be adjusted over a usefully wide range. One knob controls the timbre of the
cabinet-emulation filter, so that when fully anti-clockwise it produces a 4x12 type of sound, whereas fully clockwise it is more akin to a
single vintage 12-inch speaker. Many useful settings are available between these limits, and there’s a further modifier knob labelled
Distortion. What this does is create the tonal changes that occur in real life when you position a mic either close to the centre of a speaker
or close to the edge. It doesn’t actually add distortion, but rather allows you to change the character of the sound in a way that’s most
noticeable when amp or pedal distortion is being used. By juggling these two controls you should be able to get close to what you need. At
the 4x12 cabinet end of the control range, the bass response of the speaker-emulation circuitry provides plenty of depth for bass guitar
recording and could well win over those who prefer the richer, more complete sound of a miked bass amp and cabinet to the cleaner, more
analytical sound of a DI. The Motherload Elemental’s speaker-emulated output combines particularly well with a conventional DI signal as
the two are inherently time-aligned — unlike a conventional, ‘real’ speaker and a DI.

Impressions
The secret of a good speaker emulator is for the load to behave like a true loudspeaker rather than as a pure resistance. In other words, it
needs to be inductive like the voice coil of a speaker, because the reactance of the voice coil and the back electro-motive force (EMF) it
produces affects the way that the output stage of an amplifier — especially a valve amplifier — behaves. Without it you lose the tactile
response of playing through a real amplifier, and it’s also difficult to replicate the low-end ‘thunk’ of a real cab without it. Fortunately,

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Sequis Motherload Elemental

reactive loads and filters are exactly how the Motherload Elemental works its magic, and I can confirm that the sound gets very close to that
of the amp to which it is connected. Many speaker emulators either dull the high end or leave it sounding ‘fizzy’, but this one sounds very
natural and lively. Equally importantly, it still feels right when you play, and the sound seems to retain its ‘weight’, which is rarely the case
with modelling devices.
As with moving a real mic, the changes between the centre and edge tone are well replicated, with the most direct, presence-laden sound
in the centre position. The cabinet type adjustment is also brightest and most aggressive in its counterclockwise 4x12 position, and morphs
to something warmer and darker as you move clockwise. Of course you can’t cover all cabinet eventualities using just two knobs — but
they’ll get you in the right ballpark, and you can always add further EQ on the desk if you need it.
I’ve used a lot of speaker emulators/loads, and at least in the case of my ancient Fender Class A Champ, the Motherload Elemental
comes as close as I’ve heard to the actual sound of the amp with a mic in front of it. It might not be as adjustable as the original Motherload,
but for many users that might be seen as a benefit, as it makes getting a good sound very straightforward. For the guitarist who wants to be
able to DI the sound of his or her own amp, whether on stage or in the studio, the Motherload Elemental does a great job with the minimum
of fuss. Paul White

SUMMARY
A simplified version of Sequis’ class-leading Motherload dummy load and speaker emulator. Suitable for stage and studio use, the
Elemental is more accessibly priced.
Sequis +44 (0)1206 823108
www.motherload.co.uk

Published in SOS June 2009

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SPL Rackpack

In this article: SPL Rackpack

Overview Modular Preamp & Processor System
Preference & Premium Published in SOS June 2009
Preamps Reviews : Preamp
Print article : Close window
Ranger EQs
Transient Designer
Sound Performance Lab have developed an enviable reputation for
Twin Tube
their hardware products, and now you can buy more for less with
Dynamaxx Compressor
their modular rack system.
Sound Performance?
Paul White

W
Summary
Pricing Information ith studio space at a premium, rack systems that
hold compact modules make a lot of sense. SPL’s
SPL Rackpack 3U Rackpack is designed to accommodate up to
pros eight of the company’s all-analogue solid-state or tube
Unique SPL approach to signal modules. Currently available for this system are the
processing. Preference Mic Preamp, the Premium Mic Preamp, the
Excellent sound quality. Transient Designer, the Full Ranger passive-coil graphic EQ,
Compact. the Bass and Vox Rangers (using the same filter design), the
Simple to operate. Twin Tube and the Dynamaxx compressor. Other than the
cons Preference Mic Preamp, all modules can optionally be
Fairly costly. equipped with I/O transformers from Lundahl, and all have dual outputs so that the signal can be split
Panel legending difficult to read to two different destinations. (Where a transformer version is applicable, the output transformer can
under normal studio lighting only go in the main output path). I’ll be looking at the features and performance of the individual
conditions.
modules a little later, but first...
summary
The Rackpack provides a compact Overview
and practical way of housing and
powering up to eight of SPL’s most Power to the various modules is distributed via a ribbon cable in the bottom of the case, and there’s a
useful preamps and processors. It dummy module at one end housing the power switch, leaving eight free slots (unused ones are
isn’t cheap — but then more covered by blanking plates). While there’s little remarkable about the rack itself, a lot of care and
attention has gone into the design
attention has gone into the external power supply (PSU), which is far removed from your average wall-
of the power supply than some
companies put into the entire wart, because it is around the size and weight of a house brick, and has been designed to deliver a
product! very generous amount of extremely clean power, so that the low-noise capability of the modules is not
compromised. This PSU connects to the main rack via a robust, locking connector and a heavy
information multicore cable.
See pricing information box. The review unit came loaded with the full range of modules, starting off with both of the mic
The Audio Professionals +44 preamplifiers. As with all the other modules, these have thick metal front panels finished in satin silver,
(0)1923 693770. with a recessed black centre panel for the controls, giving the whole system a very homogenous and
Click here to email stylish appearance, regardless of which modules are installed or in what order.
www.audiopros.eu
www.soundperformancelab.com Preference & Premium Preamps
Photos too small? Click on The Preference mic-preamp circuit is based around the
photos, screenshots and diagrams SSM2019 chip in a DC servo configuration, which combines
in articles to open a Larger View low noise and distortion with an impressively fast slew-rate,
gallery. while virtually eliminating DC offsets. This circuit produces up
to 72dB of gain, and its design places a minimum number of Each module has the familiar SPL
capacitors in the audio path, which in turn reduces phase styling, with easy-to-operate (and
shift and possible sources of distortion. Two additional op- surprisingly few!) controls.
amps function as a voltage differential sensor and a summing Each module has the familiar SPL
stage. Premium-quality components are used throughout all styling, with easy-to-operate (and
the modules, including MKP and Styroflex capacitors. surprisingly few!) controls.
The Preference Mic Pre’s retro-style, backlit moving-coil
meter has two switchable modes to display average or peak (PPM) levels, with a -10dB button to give
more meter headroom when working at hot signal levels. There are also two LEDs built into the meter:
one showing signal present; and one that comes on 3dB before clipping. (All the modules in this series
have these same warning LEDs). The preamp provides a very clean 48V phantom power supply, and
there are push switches for polarity inversion, a 20dB pad, and a low-cut filter with a gentle 6dB/octave
slope at 75Hz. Gain is adjusted via a chunky metal knob at the bottom of the module’s front panel.

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SPL Rackpack

As I touched upon earlier, there’s a single input and two balanced outputs, both on XLRs, and both
capable of driving very long cable runs — so each module also acts as an effective signal splitter. The
EIN is quoted as -129dB with a 0.047 percent THD + noise figure at 60dB of gain, and 0.0035 percent
at 30dB of gain. The frequency response extends from 10Hz to 200kHz (-3dB).
Outwardly, the Premium Mic Pre looks identical to the Preference model, but it uses a Lundahl input
transformer as standard, offering up to 80dB of gain. The transformer is where the extra voltage gain
comes from, and this is followed by both a discrete differential amplifier stage and an instrumentation
preamplifier stage.
Transformer-based audio circuits have a certain sound character that many engineers find more
‘musical’ than transformerless designs, but they also offer true electrical isolation, which can be
beneficial from both safety and ground-loop perspectives, especially in mobile live-sound setups. An
EIN figure of 128.3dB is quoted along with a THD + noise figure, at 60dB of gain, of 0.078 percent,
and better than 10 times less than that at 30dB of gain. The frequency response is inevitably less than
for the transformerless Preference, but still a more than adequate 10Hz to 68kHz (-3dB).

Ranger EQs
The three Ranger EQ models look like graphic equalisers
tipped on their sides, but they are in fact based around
passive inductor/capacitor filters. The Full Ranger has
frequency centres of 40Hz, 90Hz, 150Hz, 500Hz, 1.8kHz,
4.7kHz, 10kHz and 16kHz, and the cut and boost is
controlled by a set of miniature sliders. Unlike a traditional
graphic equaliser, where the filters are set one octave apart
with identical response curves, this SPL design tailors the
response of each filter band based on musical principles
(rather than mathematical ones). Essentially, the filter curves
get wider (lower Q-settings) the further you go up the audio
spectrum.
The Bass and Vox Rangers are conceptually identical, but
they have different filter frequencies and curve
characteristics, optimised for bass and vocal applications, as
the names suggest. The Bass Ranger has centre frequencies
of 30, 65, 95, 170, 230, 500, 800 and 2000Hz, while the Vox
Ranger is set at 220, 330, 420, 500, 800, 1600, 2800 and
4000Hz. The range varies slightly with the filter type, but is
roughly ±12dB. The input, as well as the output stages, may
be transformer balanced as an extra cost option, and each
module has an output level-fader and bypass switch, as well
as those simple but practical ‘signal present’ and overload
LEDs. All of the faders are centre-detented, with red LED
illumination in the fader knob. The frequency range is 10Hz
to 30kHz (-3dB) with an A-weighted noise figure of -85dB. The Twin Tube Processor enables
you to control tube saturation and
Transient Designer harmonic enhancement using only
Next in line is the Transient Designer, the standard two dials.
rackmount version of which we reviewed in SOS when it was The Twin Tube Processor enables you
first released. The Transient Designer’s clever trick is that it to control tube saturation and harmonic
enhancement using only two dials.
provides separate control over the attack and release
characteristics of a percussive sound using only two knobs,
as the circuitry automatically adapts its threshold to the level of the incoming audio, so that the user
doesn’t need to worry about levels. In the centre positions, the two knobs do nothing, but moving in
either direction allows the user to either enhance or suppress the attack and release envelope of the
source sound. Using the attack control, you can, for example, back off the control when processing a
kick drum, to get more of a ‘bouncing beach ball’ sound, or advance it to really bring out the initial slap.
Similarly, adjusting the release can either make the drum sound very tightly damped, or it can bring up
the decay for a really roomy sound. It works well on individual drum tracks or a complete kit mix, and it
can also be effective on instruments such as bass guitar.
A third knob sets the output level, and there’s also a bypass (On) button and a stereo link button.
Stereo linking requires two adjacent modules to be internally connected using a small ribbon cable,

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SPL Rackpack

and there are DIP switches on the circuit boards that also need to be set. This remains one of my
favourite processors, and it has rescued an indifferent-sounding drum part on more than one occasion!

Twin Tube
The Twin Tube processor’s purpose in life is to create
controlled tube saturation and harmonic enhancement, again
using very few controls — just one knob to adjust the
harmonic enhancement, and another to adjust the degree of
tube saturation. A pair of buttons used in combination allows
All of the I/O is hosted on the
the harmonic enhancement to operate on frequencies
modules themselves, so the rear of
centred at 10, 6, 3 or 2kHz, and there are separate ‘On’
the rack unit is very straightforward.
buttons for the Harmonics and Saturation sections. The
The channel numbers are usefully
actual frequencies are, in fact, 9.8kHz, 6.6kHz, 2.8kHz and
presented both ways up, so you can
1.9kHz, but the approximations have been printed to keep
see what you’re doing when peeping
the panel tidy. Each section has its own tube circuit and, as
over the top.
you might expect, the tube saturation effect is created by
All of the I/O is hosted on the modules
driving the tube into a non-linear region of its operation. themselves, so the rear of the rack unit
A more sophisticated circuit comes into play for is very straightforward. The channel
overtone/harmonic processing using an inductor/capacitor numbers are usefully presented both
filter to modify both the overtones and their phase ways up, so you can see what you’re
doing when peeping over the top.
relationships with the source sound. The designers claim that
the phase part of the circuit aligns the overtones, and works
not so much by generating harmonics, as a traditional exciter does, but rather by selectively equalising
the existing harmonic structure. It may be that this process relies on some of the principles pioneered in
the SPL Vitalizer, although this isn’t confirmed in the documentation. Using the harmonic enhancement
can make a sound seem closer and more present, without necessarily making it louder.
As this is a tube module, it takes some time to warm up, so the ‘On’ buttons flash for a few minutes
after powering up, to let you know when it is ready to use. The same signal and overload LEDs are
provided as for the previous modules. An overall frequency response of 10Hz to 80kHz is specified for
the Harmonics section, with an A-weighted noise figure of -87dB, while the saturation stage is quieter
still, at -96dBu A-weighted, and with a 10Hz to 77kHz bandwidth. Distortion figures are not really
relevant — because the module’s job is to introduce controlled distortion!

Dynamaxx Compressor
The final module fitted to our review system was the Dynamaxx compressor, which is, again, an
exercise in how much control the designer can put under a single knob, and is an update of the stand-
alone design specifically created for the Rackpack system. The compressor, which has a soft-knee
characteristic, has very sophisticated automatic attack and release circuitry, designed to dynamically
match the time constants to the type of material being processed. It’s also designed to prevent high
levels of compression compromising high-frequency detail. The gain-control element is based on a pair
of THAT2181 VCAs in a double VCA drive-mode configuration, to cancel distortion.
Increasing the Compression control setting increases the ratio, while simultaneously lowering the
threshold, and a gain-reduction LED ladder meter displays the gain change very clearly. The maximum
compression ratio is 3:1, and a make-up gain control may be used to restore any level lost through
compression, while a further button brings in a separate limiter side-chain — again with a soft knee —
that acts on the same VCA pair to provide a more progressive style of limiting than the usual hard-
knee type. Signal/overload LEDs are fitted at the top of the panel.
The ‘On’ button operates a hard bypass, and there’s also an ‘FX Com’ button that flips to a fixed
release time of 60ms, making it easier to create deliberate gain-pumping effects when required.
Unusually, there’s also a ‘De Com’ button. This converts the compressor into a type of subtle expander
and is designed to help undo the effects of over-compression on pre-recorded material. It’s worth
noting that the limiter also plays a negative role in this mode if active, and actually gives a boost to
transients. As with the Transient Designer, two units may be linked for stereo operation, providing that
the necessary cable and DIP settings are attended to when the modules are installed. This module has
a 10Hz to 200Hz frequency response (-3dB) and an A-weighted noise figure of -93dB when the make-
up gain is set to zero. The main difference between the Rackpack Dynamaxx and the rack module
that’s been around for years is that the rack module also has a gate section, which is omitted here.

Sound Performance?

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SPL Rackpack

Both mic preamps are very clean and quiet, although the Premium model has a slightly more assertive
sound that’s probably due to subtle non-linearities and phase-shifts in the input transformer. I
compared the Preference module on voice with my Universal Audio Solo 110 solid-stage preamp and
found that it came very close. By contrast, the Premium had a smoother, slightly coloured top end, and
it seemed to be adding a bit of weight to the lower vocal frequencies — which is pretty much what I’d
expected. Both are seriously good preamps, so it’s really down to picking the most appropriate for a
given recording situation.
The Ranger EQs are particularly effective and musical, but I have a couple of ergonomic reservations
regarding the panel markings. Unless you’re looking at them straight on, the sliders obscure the
frequency markings, which are, in any case, small and difficult to see in normal studio lighting, because
of the illuminated fader sliders. That aside, the EQs have a wonderfully analogue sound, and you can
add quite a lot of boost when needed without wrecking the sound. The Vox and Bass versions are
particularly useful in tackling the main frequency bands of those two sources, and save lots of time
hunting around with a parametric EQ. Even the top boost sounds warm, and there’s none of that
glassy, gritty edge that some solid-state active EQs impart when you add more than a dB or so of
boost. I also compared the low end of the EQ with some of my better plug-ins and found that the SPL
modules seemed somehow more solid and believable, with no tendency to make the low end messy or
boomy.
The Transient Designer works just like its rackmount counterparts, and is simply unbeatable for
polishing drum sounds, although it can also work wonders on other types of percussive or plucked
sounds. If your drums sound too dry, turning up the release will both lengthen the decay and bring up
the room ambience, but if virtual gaffa tape is what you need to damp down ringing toms, you just turn
it the other way. It’s the same with the Attack knob — go left to soften the attack or right to make it
more spiky and aggressive. Because of that auto threshold feature, it works regardless of the level of
the drum hits that you’re processing. I won’t get into this too deeply, as we’ve covered the Transient
Designer itself before in plenty of detail. There are just two knobs and it’s brilliant!
The Twin Tube is an interesting device because it really behaves like two processors rolled into one.
The harmonic enhancer adds brightness around and above the frequency of the filter, which is set
using the four permutations of the four frequency buttons. This works very effectively, and sounds more
natural than some harmonic-generating devices, but it still works differently from conventional
equalisers — at least subjectively. It’s really a matter of bringing out and enhancing what’s already
there, and I’m sure that the tube itself adds a touch of organic flavour to the proceedings.
The lower part of the front panel controls the tube saturation effect, and this is far less subtle unless
used very sparingly. At lower settings it adds warmth, but go much further and the distortion that it
introduces gets more obvious, becoming seriously crunchy at higher settings. Heavier distortion isn’t
only for electric guitar, however: it can work wonders on drums, some bass synth sounds and possibly
death metal vocals. (On ‘normal’ vocals you really have to use it sparingly.) Still, the ability to introduce
very obvious saturation extends the versatility of this module far beyond that of a vocal warmer!
Finally, the Dynamaxx compressor proved every bit as easy to use as the manual had promised. You
can really lay on the gain reduction, and unless you select the fixed release-time mode, it resists all
attempts to sound pumpy or dull. If you want dramatic pumping to use as a special effect, you just
switch in the FX Com button, hit the limiter button, and crank up the compression! The Dynamaxx
makes a great vocal leveller when used in its normal mode, and it adds a touch of warmth at the same
time, but if you want to use it as a de-compressor or expander, it’s probably best to err on the side of
caution — because otherwise loud sounds get really loud, especially if the limiter is switched in. You
also have to remember that all the controls seem to work in reverse in expander mode, even the
make-up gain. Use it with care, though, and it really can restore some useful dynamic range to an over-
compressed track.

Summary
Other than some issues concerning front-panel legend visibility, the Rackpack modules work brilliantly,
and where features have been lost to make them fit the space, this doesn’t really affect usability to any
significant degree. The sound quality is up to SPL’s usual high standard, too, while their innovative use
of technology puts some very complex processing behind a surprisingly small number of physical
controls. Also, as I completed this review, SPL announced a version of the Rackpack that can host the
ubiquitous API 500-series modules as well: a mouthwatering prospect indeed!
The Rackpack system isn’t cheap: the empty rack and PSU cost as much as a decent computer. But
there’s no denying the quality of what these guys build. SPL have a unique approach to product

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SPL Rackpack

design, due in no small part to the innovative thinking of chief designer Wolfgang Neumann — and
they deserve to do well with the RackPack, especially in smaller studios that still prefer analogue
mixing.

Pricing Information
Rackpack Frame £732
Preference Mic Preamp £260
Premium Mic Preamp £524
Dynamaxx £418
Twin Tube Processor £418
Transient Designer £281
Full Ranger £418
Bass Ranger £418
Vox Ranger £418
All prices include VAT.

Published in SOS June 2009

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Toontrack Drumtracker

In this article: Toontrack Drumtracker

Choose Your Drum-replacement Software [Mac/PC]
Weapons
Published in SOS June 2009
Not So Dumb Reviews : Software: Utilities
Print article : Close window
Tracker

Toontrack Drumtracker Need to use samples to rescue a dodgy drum recording? Toontrack’s
£75 neat utility will generate MIDI hits and map them to the virtual
pros instrument of your choice.
Very affordable.
Paul White

D
Straightforward in operation.
Includes MIDI templates for rumtracker runs on Mac OS and Windows platforms and
many of the popular drum- operates as a stand-alone drum replacer application that
sample instruments. can work with multiple drum tracks, generating MIDI data
cons from each track for triggering suitable drum samples. While
While the program is not aspects of the program are automatic, the user can adjust
intended to be able to threshold and filter settings to achieve better separation (less false
unravel a complete drum triggering) where spill is present, and it is even possible to work
mix, more sophisticated with a complete drum mix, though the user has to do quite a lot of
filtering might help separate
work to identify which hits belong to which drums!
sounds more reliably where
there’s excessive spill. As with most processors of this kind, the source signal is treated
summary with filtering to reduce the effects of spill from instruments at radically different pitches, and then with a
Drumtracker is designed to threshold detection system similar to a gate. Drumtracker can assign velocity values to the MIDI data based
turn close-miked drum tracks on the amplitude of the original hit. Each detected hit is shown by a vertical line on top of the waveform
into MIDI data, and includes display of the source audio, and the threshold points are shown as red lines that can be adjusted simply by
sufficient filtering and gating dragging. Where the drum part fluctuates in level, the threshold can be varied for different parts of the file.
to avoid false triggering in Once hits have been detected and converted to trigger points, the user can then delete, add or move them,
most normal situations. It
as well as adjusting the velocity of individual hits. Finally, the result can be exported as a MIDI file to use in
also allows manual editing of
the original DAW project.
the detected data, so you
can even get a good result
with difficult material if you’re Choose Your Weapons
prepared to put in the time.
Drumtracker’s window is a simple affair, with a waveform display at the top and an area below to determine
Given its price, it strikes a
good balance between ease how the software deals with each part. The first stage is to use the Add Input function to locate each desired
of use and ability. drum audio file, which may be mono or stereo, in WAV or AIFF format. The next stage is to create or choose
a MIDI Template: choices on offer include EZ Drummer, Superior 1, Superior 2, General MIDI, GM Extended
information Addictive Drums, BFD1 and BFD2. By default, this comes up with simple MIDI note numbers, so if you know
£75 including VAT. what note you need for a specific drum, you can select it directly. To extract more drum types from the
Time + Space selected input, you can select Add Instrument and enter the appropriate settings, though as I said at the
Distribution +44 (0)1837 outset, separating sounds from a mixed drum file isn’t easy, especially where several drums may be close to
55200. the same pitch, so some additional manual work is invariably necessary. If you want an easy life, it is far
Click here to email better to work from the individual close-mic tracks where these exist, as you then have little or no extra work
www.timespace.com to do.
www.toontrack.com
Once the desired drum files have been loaded in, they are displayed as separate strips in the Input section
Photos too small? Click of the window; the waveform panel above shows the currently selected part. The Part tool is used to select
on photos, screenshots and what section of the drum track to process: where you have more than one articulation in an input file, such
diagrams in articles to open as a snare track that includes a section of side-stick playing, you can identify the sections separately and
a Larger View gallery. enter the desired target MIDI parameters.
When you place the cursor before the desired region and hit Play, any detected events are shown as
vertical coloured lines, the height of the line being proportional to the MIDI velocity. The same results can be
achieved using the Render button, which processes the whole file as fast as possible but without the
possibility of listening to the result until you’re done. User adjustment consists of varying the threshold and
filter parameters, though the latter are quite basic, providing just high- or low-pass filters and high or low Q.
There seems to be no way to specify a filter frequency, nor to listen to the filter output during adjustment.
However, you can audition the trigger points that the software detects by assigning inbuilt drum sounds or
clicks to them, then using a balance fader to adjust the monitor mix between the MIDI-triggered sound and
the original drum file. You can also send MIDI to an external sampler, should you wish to, through for
verifying the process is working properly, the sounds provided are fine.
Along with the main filter, each part you select within an audio file can be given its own filter setting by

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Toontrack Drumtracker

clicking the ‘F’ at the bottom right of the waveform display, but these filters seem to have the same fairly
basic adjustment parameters as the main filter. My own preference would have been for variable high- and
low-pass filters for each part, with a means of auditioning the result.
Manual editing of the detected trigger points is done using modifier keys, and clicking on the trigger ‘lines’
allows the user to disable wrongly identified beats or insert new triggers where a hit was missed. Hits can
also be moved and their MIDI velocity adjusted, so the results that can be achieved come down to your
patience. When the trigger points have been tweaked to your satisfaction, you can output the results as MIDI
data to use in your original project. You need to set the required tempo for the MIDI file so that it matches
the original DAW project; where a tempo map was in use, you can choose to include the ‘conductor’ track in
the MIDI file so that all the MIDI hits sit correctly on the timeline.
By default, Drumtracker uses the MIDI Type-1 file format, where all the sounds are contained in a single
MIDI file. However, you can specify MIDI Type-0 format if you want to create a separate MIDI file for each
drum or cymbal. There’s also a ‘Split Instrument’ option that saves separate MIDI les for each articulation of
each specific instrument, such as open and closed hi-hats or snare centre and snare side-stick. The PDF
manual includes a list of keyboard shortcuts, and it may be worth printing these out and sticking them to your
desk, as they can save a lot of time.

Not So Dumb Tracker

Overall, the operation of the software is reasonably intuitive once you’ve read through the manual, though I
think the on-screen controls could perhaps be better arranged, in the order of workflow. I also had the
program quit on me a couple of times (on an Intel Mac running OS 10.5) for no obvious reason. As it is,
some functions are accessed within the Drumtracker window and others from the menus at the top of the
screen, which can be confusing at first. If the system has a weakness, though, it is that the filters aren’t really
flexible enough to separate out different hits of roughly the same level, so the software is only really effective
when dealing with close-miked sources — which, admittedly, is how it is designed to work. I also missed
having a filter ‘listen’ mode, as I like to know what the detection part of the system is ‘hearing’.
Providing you don’t expect miracles, such as flawless automatic separation of drum sounds from an
overhead or room mic, Drumtracker actually does a great job with minimal fuss, and the timing of the end
result is generally spot-on. Admittedly, there is room for improvement, but it’s still early days and the product
may well evolve over the next few revisions. Many band demos suffer from poor drum sounds, so even
replacing the close-miked ones and leaving the original overheads to carry the cymbals (ideally with some
low-cut EQ) can make a big difference. Drumtracker is a very useful tool and is very attractively priced, so if
you are struggling with drum sounds and have some good samples at your disposal, it could make a real
difference to your recorded results.

Published in SOS June 2009

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Waves GTR3 & GTR Ground

In this article: Waves GTR3 & GTR Ground

Tooled Up Amp Modelling Software & Foot Controller
What’s New? Published in SOS June 2009
On The Ground Reviews : Software: Effects+Processors
Print article : Close window
Not So Free
Expression
Waves have made a rare foray into the world of hardware with the GTR
Ground Control
Ground, designed to give hands-free control over their Guitar Tool Rack
Neil Citron Amps
software amp simulator.
Waves GTR3 & GTR Sam Inglis

W
Ground
aves were not the first software company to introduce
pros their own amp and effects modelling software for
GTR3 sounds good and guitarists, but they went the extra mile to ensure that
offers a broad range of amp their Guitar Tool Rack stuck out from the crowd. Perhaps the most
models.
notable feature was their collaboration with legendary luthier Paul
New Neil Citron amps are
Reed Smith on the design of the accompanying DI box, which was
great for modern guitar
sounds. said to be optimised to give the best possible DI’d electric guitar
Ground unit makes basic signal. Paul White reviewed version 1 of GTR back in SOS
effect and patch switching November 2005 (www.soundonsound.com/sos/nov05/articles/wavesgtr.htm), and since then, development
straightforward, with nice has moved on apace.
visual feedback.
The GTR software is now at version 3.5; the look has changed, a swathe of new amp models has been
cons
added, and the biggest news is the introduction of a floor controller optimised for use with GTR3, the GTR
GTR3 does not integrate with
Ground. GTR3, the Ground floorboard and the PRS Studio Guitar Interface are all available individually or in
floorboards as well as
various bundled combinations, and Waves are currently offering the cut-down GTR Solo free for one year.
competing products, offering
little control beyond the
basics. Tooled Up
Expression pedals do
In the original Guitar Tool Rack system, the individual components
nothing in the presets, and
don’t have associated of a typical guitar setup — effects pedalboard, amplifier and tuner
switches. — had to be loaded as separate plug-ins. Most of these individual
Ground unit can’t be bus- plug-ins are still included, but they’re now joined by a full-on Guitar
powered, and the power Tool Rack plug-in which incorporates slots for up to six ‘stomp’
supply doesn’t inspire effects and two amp/cab models, with a dedicated tuner page.
confidence.
This, as you’d expect, works in Pro Tools TDM and all native plug-
summary in formats, and is also included as a stand-alone program. You
GTR3 is a friendly amp can save global Guitar Tool Rack presets, but each stomp and amp slot also has its own presets, which can
simulator with plenty to offer be loaded into the individual plug-ins too.
sonically, especially if
modern rock is your bag, but Guitar Tool Rack is mostly very straightforward to use. New stomp effects are called up simply by clicking
integration with the Ground on the arrow beneath the appropriate slot and, once active, can be re-ordered simply by dragging and
controller is disappointingly dropping. Likewise, the position of the amp module in the chain can be altered, and there are buttons to
limited at present. switch both the pre- and post-amplifier effects chains into parallel and split modes. Doing so forces the
stomps in odd and even-numbered slots to form two independent effects chains, which are then fed one to
information each of the two channels (split mode) or equally to both (parallel mode). A View button superimposes
GTR3 software in TDM coloured patch cords in case you’re unsure.
and native versions, GTR
The only major source of confusion for me was the amp module
Ground controller and PRS
Guitar Interface all available itself. It seems to be actually impossible to set this up to contain
separately or in various only one amp: both must always be active, and the manual states
bundles: UK prices currently that one processes signal from the left input and the other from the
POA. right. I don’t know about you, but all my guitars are strictly mono
Sonic Distribution +44 affairs, so when I first called up GTR as a plug-in within Cubase, I
(0)1582 470260. used the ‘mono to stereo’ version. This, as you’d expect, routes
Click here to email the input signal equally to both amps, but doesn’t seem to do The View option allows you to
www.sonic- mono-to-stereo within Cubase: on a mono track, it returns the left visualise the effects routing you’ve
distribution.com
channel only. To get stereo output from a mono input in Cubase, chosen.
www.waves.com
you need to insert GTR on a stereo group or FX channel and route The View option allows you to visualise
Photos too small? Click your mono audio channel to that group. If you do want to use a the effects routing you’ve chosen.
on photos, screenshots and single amp and cabinet, you need to pan it centrally and turn the
diagrams in articles to open output volume all the way down on the second one. All in all, it’s an arrangement that does work once you’ve
got your head round it, but I can’t help thinking it would be easier to simply allow you to switch off one of the

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Waves GTR3 & GTR Ground

a Larger View gallery.

two amps.

What’s New?
The range of effects appears to be little changed since Paul
White’s original review, and the only two additions I spotted are a
pitch-shifter and a new dynamics stomp. Pitcher allows you to
manipulate the pitch of the incoming signal by up to an octave
either way, with a foot-controllable slider that operates between
limits you set with Min and Max dials. It works pretty well, except
that the lowest setting of the Min dial is very audibly not an octave
below the source! Axxpress, meanwhile, is a simple but very New effects include the Axxpress
musical compressor/limiter that offers only Attack, Press and dynamics unit and Pitcher pitch-
output level controls. I found myself using it a lot. shifter.
By contrast, amp and cabinet models have proliferated since New effects include the Axxpress
version 1, to the extent that there’s not space to go into them all in dynamics unit and Pitcher pitch-shifter.
detail here. Obvious highlights include the new Neil Citron models
(see ‘Neil Citron Amps’ box), while some of the models based on more obscure or boutique amps are very
welcome additions. The Gibson Skylark and Ampeg Gemini both offer novel (to me) and highly usable tones,
while others hail from Paul Reed Smith’s private collection and sound like it. There are few guitar sounds you
couldn’t get close to with this collection, and it’s perhaps at its strongest for modern rock. If you want that
hard-to-find brand of chunky riffing distortion that retains tightness and definition, you’ve come to the right
place. It took me longer to find a twangy country lead sound that I liked — some of the cabinet models have
a tendency to break up on note attacks in an unappealing way, just as real ones can — but in the end I was
very pleased with the results.
To my ears, GTR3 is one of the better-sounding amp simulators out there, and although its relatively
simple architecture means that it won’t be the most obvious choice for the experimentally minded guitarist, it
provides a huge range of very usable guitar sounds without a lot of fuss.

On The Ground
And so to the GTR Ground controller, which is an imposingly large beast — not quite as deep, front-to-back,
as Native Instruments’ Rig Kontrol 3, but about twice as broad — though it’s still easier to sling in the back of
the car than a Fender Twin. It seems solidly constructed, boasts 11 buttons and two expression pedals, and
connects to your Mac or PC via USB. Happily, it’s a truly plug-and-play device, with no need to install
drivers. Less happily, it can’t be bus-powered, and comes with a rather delicate-looking wall-wart. Although
the connectors are recessed, a slip of the foot could easily dislodge either the power or the USB cable.
Unlike Rig Kontrol 3 and some other competing products, GTR Ground is purely a controller, so you’ll still
need a separate audio interface to get signal into your computer. This means yet more cables and things to
go wrong on stage, but may be preferable for typical studio users who already have a multi-channel audio
interface and don’t want to complicate their systems unnecessarily. GTR Ground sends its control signals as
MIDI rather than embedded in the audio stream, as Native Instruments’ system does; this is slightly
cumbersome when you’re using it to control the plug-in version of GTR3, as you have to create a MIDI track
within your sequencer and route its output to the plug-in. In Cubase, I couldn’t find any way to return MIDI
from the plug-in to the floorboard, so although the controller worked, it didn’t give the same visual feedback
as it does with the stand-alone GTR3.
I’m guessing that the need for separate power is brought on by the GTR Ground’s displays, which are
impressively clear and bright. Each of the six buttons on the bottom row has its own three-character orange
LED display, as does the Preset button on the top row. This switches the unit between its two modes, which
are designed for preset selection and real-time control respectively. You quickly come to appreciate the
metal bar running along the middle of the unit, which firmly prevents you from accidentally hitting any of the
top-row buttons when all you want to do is turn a flanger on!
In its default Stomp mode, each of the six buttons on the bottom row switches on or off the stomp-box in
the corresponding GTR3 slot. If there’s no stomp-box in said slot, the display above the button is blank, but
where a stomp-box is loaded, you see a three-letter abbreviation of its name. If the stomp-box is active, the
letters glow brightly, while bypassed effects appear in more muted tones. This is all very clear and intuitive,
although the down side is that there’s no way to get these buttons to do anything else — you can’t, for
example, program one button to toggle two stomp-boxes at the same time.

Not So Free Expression

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Waves GTR3 & GTR Ground

The manual states that the two expression pedals are set up by
default so that the first one controls an appropriate effect such as
wah-wah or pitch-shift, while the second adjusts the output volume
of the GTR3 software. This turns out to be a mistake: in fact, the
pedals aren’t assigned to anything at all by default, so you have to
make assignments manually. This is easy to do — you simply
right-click on whatever parameter you want to control, select Learn
from the pop-up menu, and waggle the pedal you want to set up The Ground unit requires a
— but there are features of the system that limit its usefulness in dedicated 12V power supply.
practice. The Ground unit requires a dedicated
12V power supply.
Any variable parameter can be controlled using the expression
pedals, including stepped ones such as tremolo shape, and it is
possible to assign one pedal to more than one destination simultaneously. However, there are two major
limitations. First, neither of the pedals has a switch, so you can’t emulate a typical wah-wah, where you push
down fully to turn it on. Second, you can’t scale the action of the pedal: it always moves the target parameter
through the full range of its operation. This is not a problem for effects such as wah-wah, but is a serious
restriction if you want to do anything more experimental or ambitious. It also pretty much precludes the use
of the expression pedals to control GTR3’s Input or Output Level controls (even though the latter is
supposed to be the default destination), because turning either of them up beyond about halfway usually
causes clipping. Waves told me that they recommend using the Volume stomp effect instead of the Output
Level control for this purpose, but of course not all presets have a spare slot for an additional stomp.
Personally, I would also have liked the option to assign the pedal negatively as well as positively to some
parameters: this would allow you to have a tremolo that gets less deep as it gets faster, for example.
The currently selected preset number is usually displayed beneath the Preset button, and you can cycle
through the presets one at a time by hitting the up and down buttons to its left. As with all Waves plug-ins,
each GTR3 preset is actually two presets, which you can switch between using the A/B button; this goes
some way towards compensating for the inflexibility of the Stomp buttons. The Preset button itself turns the
six Stomp buttons into patch selectors, allowing you to choose any of six consecutively numbered presets,
with the up and down buttons now moving to the next or previous bank of six.
This is fine, although things can get a bit confusing when you start to create and edit your own patches, or
choose them from the drop-down Load menu. Any user presets that you create using the ‘Put Into Preset
Menu As...’ command aren’t numbered or easily accessed from the GTR Ground. The best option, especially
if you’re planning on using preset switching in a live context, is to create your own preset file and put all and
only the patches you intend to use in that. I can understand why Waves want to ensure that their preset
system is consistent across all their software, but it would be more friendly if there was a neat bank/preset
structure. Also, given that GTR3 includes a Preset viewing mode, it seems odd that hitting the Preset button
doesn’t switch to this mode.
The last button on the GTR Ground is Tap Tempo / Tuner.
Hitting this in a vaguely rhythmic pattern sets the master Tempo
value within GTR3, to which any time-based effect can be clocked
by enabling its Sync button. This works well enough, though there’s
no visual feedback of the tempo setting. Keep your foot on the Tap
Tempo / Tuner button for a few seconds, and it takes you to
GTR3’s Tuner page. Unlike most hardware tuners, this doesn’t
mute the output signal, which could be annoying in a live situation, Guitar Tool Rack’s dedicated tuner
and oddly, the switch only takes effect when you release the page. You can check in but you can
button, so it’s unclear exactly how long you have to hold it down. never leave — from the foot
More oddly still, there is no way to use the GTR Ground to leave controller, anyway...
the Tuner page again — you have to do so with the mouse. And Guitar Tool Rack’s dedicated tuner
although GTR3’s on-screen tuner is very nice, it’s a shame there’s page. You can check in but you can
no visual feedback on the Ground unit that would allow you to tune never leave — from the foot controller,
without looking at the screen. anyway...

Ground Control
I’ve reviewed a number of Waves products over the years, and in the case of GTR Ground, I’m experiencing
a feeling I’ve never had before: I’m a bit underwhelmed. It’s not that it’s a bad product, exactly, because it
does pretty much what it says on the tin. In conjunction with the GTR3 software, GTR Ground offers a very
acceptable substitute for a typical pedalboard or floor-based digital effects unit, and if all you need is the
ability to switch effects on and off, wobble a wah and step through presets, it does a fine job. My sense of

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Waves GTR3 & GTR Ground

disappointment is more down to high expectations, because I’m used to finding that Waves products go
further than the competition and do more than you’d expect, but in this case the competition is clearly ahead.
Specifically, I think that when it comes to integrating a floorboard and an amp simulator, Native
Instruments’ Guitar Rig 3/Rig Kontrol 3 combo wipes the floor with the GTR3/Ground setup. Both of them do
the basics well, but NI’s system really takes things to a new level, offering a degree of configurability and
controllability that simply isn’t there with GTR Ground. What’s more, it does so without being harder to use or
less intuitive in action, and although it has fewer switches and only one pedal, it does incorporate a high-
quality USB2 audio interface. NI’s system has clear advantages in the live arena too, thanks to its intuitive
on-screen Live View, the ability to switch instantly between multiple Snapshots, and the Rig Kontrol 3’s use
of bus powering. Likewise, I’ve no personal experience of IK Multimedia’s ambitious Stomp I/O, but it, too,
sounds very comprehensive, allegedly providing foot control over every parameter in their Amplitube
software, even down to editing, saving and loading presets.
It should be stressed, though, that few of the restrictions I’ve mentioned are inherent to the GTR Ground
unit itself. They are, rather, down to the way GTR3 handles external control. The Ground unit has the
potential to be a perfectly nice hardware controller — in fact, I tried setting it up to control Guitar Rig and it
did so very well — but this aspect of GTR feels under-developed at present. As it is, if you can live without
the displays, the level of control that is currently possible could be achieved from a cheaper, generic MIDI
floorboard. Let’s hope version 4 makes improvements here, because it diminishes what is otherwise a good
product.

Neil Citron Amps

Neil Citron is an American engineer who’s known for his distinctive electric guitar tones, and for GTR3, he
and Waves have teamed up to create ‘signature’ amp models. A total of six models have been created
from three amplifiers — a Carvin Legacy, an Ibanez Thermion and a 100 Watt Marshall Plexi — and the
key element is not so much the amplifiers themselves as Citron’s know-how in teaming them up with
different cabs and miking the results. There’s an element here of what you get with Waves’ Maserati
Collection, because multiple mics and processors are combined behind the scenes, yet all the user sees
is a simple choice of cabinet type. The results are excellent, with Citron’s expertise yielding a much more
three-dimensional sound than I’m used to from digital emulations. In particular, there’s a really weighty
cabinet ‘thump’ when you dig into the low ‘E’ string. You could happily record an entire rock album with
these models alone.

Published in SOS June 2009

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Zero-G Vocal Foundry

Zero-G Vocal Foundry

Multi-format Sample Library
Published in SOS June 2009
Reviews : Sample/Sound/Song Library

V
Printer-friendly version

ocal Foundry is Zero-G’s follow up to Vocal Forge, and it offers similar content: a series of vocal construction kits, each containing
all the elements for constructing a full, song-based lead vocal, including all the backing and harmony parts. A vocal ‘toolkit’, with a
series of additional vocal samples, is also provided, and includes scratches, vocal beatboxes, various spoken phrases, and a whole
bunch of processed vocal, all suitable for adding some ear candy to a track.
But Vocal Foundry differs from the earlier release in several ways. First, while the urban/dance
style still dominates, this collection is more diverse. Second, it’s not restricted to the Intakt-based
format, with Acid, EXS24, HALion, Kontakt and Reason’s NNXT all supported. Third, it has about
twice the sample data (2.5GB). Finally, it’s also considerably less expensive.
The wider range of styles is a welcome development. I could still imagine the clubland-friendly
vocal ‘Forever Loved’ going down a storm on the dance floor, but ‘Leaving’ (also a female lead)
has a nice mainstream R’n’B feel, while the male vocal ‘Lose Control’ ticks all the right hip-hop
boxes. Vocal Foundry also strays into other territory. For example, the rather beautiful sounding
male vocal on ‘Hurricane’ is more Simon and Garfunkel than Beyonce and Jay-Z; the female-
based ‘Sun Goes Down’ could be taken from a Dido session, and both ‘You Said’ and ‘When You
Come Home’ are straight, gentle pop. One or two kits are perhaps less successful (like the rather
odd lyrical content of the rapped ‘Bars On My Phone’) but there’s also the occasional genuine
comedy moment: ‘Say It With Honour’ includes the lyric ‘I have more flavours than Nesquik’
(classic!) but it’s actually a very nice female rapping workout.
As with Vocal Forge, the main issue with here is how many different songs you might build with
the 16 kits provided. Anyone who owns this sort of library is always going to be constrained by the
lyrical and melodic content. And while the ‘guide’ mixes are useful for auditioning purposes,
working with this type of material is more of a remix task than song-writing from scratch. Despite
this obvious limitation, though, Vocal Foundry is fun to work with, and if you liked Vocal Forge, this
more diverse take on the same approach should please you. John Walden
£78.25 including VAT.
Time + Space +44(0)1837 55200.
www.timespace.com
www.zero-g.co.uk

Published in SOS June 2009

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Zoom H4N

In this article: Zoom H4N

N-Powered Portable 24-bit Recorder
Four Play? Published in SOS June 2009
Bells & Whistles Reviews : Stereo Recorder
Print article : Close window
Sound Choice?
Conclusion
The H4 remains one of the most flexible of the current crop of portable
Zoom H4N £330 24-bit recorders, but Zoom have found plenty of room for improvement
— and, more importantly, they’ve used it.
pros
Matt Houghton

Z
Class-leading feature set.
Better preamps than the H4.
oom’s H4 digital stereo recorder remains hugely popular
XLR inputs and phantom
and, despite a few reservations, is one of my favourites.
power.
Capable of surround-sound With ‘combi’ jack/XLR inputs, 48V phantom power, a
recording. coincident stereo pair of mics onboard, and masses of extra
cons functionality (from four-track mixing to guitar effects and a USB
audio interface) it’s much more versatile than most recorders, and
None at this price.
the sound is good for the price. The H4’s preamps could be better,
summary
navigating the menu can be fiddly, the casing might be a bit more
At its heart, the Zoom H4N is
robust, and so on, but on balance it’s an excellent portable
a serious two-track recorder,
recorder.
where it is amongst the best
in its class. The multitrack
mixing and four-track N-Powered
recording functionality — as
well as the huge number of It should come as little surprise, then, that Zoom are continuing to
other ‘bells and whistles’ — sell the H4, while releasing an improved model that addresses
make it the most versatile, many criticisms made of the earlier model. The H4N comes in at a
too. A classy upgrade to an slightly higher price but, like its sibling, boasts a feature set that
already impressive product. makes other recorders look positively lightweight.

information The H4N’s body has a thick, rubberised coating, which tackles concerns about handling noise pretty
effectively — not to mention making the whole device feel more solid, better weighted, and altogether more
H4N £329.99; RC04
remote control £29.99. ‘professional’. If you do find handling noise an issue, there’s a camera-tripod mounting socket on the rear,
Prices include VAT. and a separate mic-stand adaptor for this. If noise is still a problem, there’s an optional remote control that
Zoom UK +44 (0)1462 plugs into the main unit via a mini-jack on the left-hand side panel.
791100. Other striking visual changes include a slightly clearer screen (the one on the H4 was already pretty good)
Click here to email and a new control layout. Gone are the rather fiddly controls of the H4: you now have dedicated transport,
input- and track-selection controls on the front panel; and the mystifying two wheel/button menu controls of
Photos too small? Click
the H4 have been replaced with a more intuitive system based around one menu button and a single scroll-
on photos, screenshots and
and-click selection wheel. The track-selection buttons double up to provide shortcut keys for some of the
diagrams in articles to open
more commonly used functions — folder and file selection, playback speed (yes, you can slow down tracks,
a Larger View gallery.
without pitch change, to learn your guitar licks), and recording format. The recording level and playback
volume settings are controlled by dedicated buttons on the right and left panels respectively.
At the top are more solid-feeling mics than appeared on the H4. These are, again, configured for coincident
stereo recording, but can be rotated between 90 and 120 degrees. Combi sockets on the bottom panel allow
the connection of external mics, or line/DI sources, and these sockets can deliver 24V or 48V phantom
power. The preamps have been upgraded; they’re not high-spec by studio standards, but are perfectly
adequate for this sort of device. You can also connect a stereo mic via a mini-jack socket (with plug-in
power), to use instead of the onboard mics.
As well as the line/headphone output, there’s now a small speaker on the rear. This isn’t for serious
monitoring, of course, but it’s a useful addition that brings the Zoom into line with more recent competition. It
means you don’t always need to use headphones or external speakers to check things are operating as they
should, or to locate a specific track or location within a track. Power comes in via the included adaptor, or a
pair of AA batteries.
Like the H4, you can record in different file formats, with a range of MP3 and WAV options, from the
cruddiest, media-efficient MP3 to 24-bit, 96kHz WAVs. Usefully, you can record broadcast WAVs, with
markers for use when navigating or editing audio files, or burning CDs. All this is recorded to SD card (a 1GB
card is included).

Four Play?

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Zoom H4N

The H4’s four-track mixer enabled you to do basic overdubbing,

and to apply effects to your recordings, but despite having two
mics and two external inputs, it wasn’t capable of simultaneous
four-track recording. With the H4N, you still get a multitrack mixer,
but the onboard mics and external inputs can be used The controls of the Zoom H4N —
simultaneously. Because the onboard mics are configured for particularly those on the side panels
stereo, your four-track options can seem limited, but you could — are much more intuitive to use
bypass them using the mini-jack input and a splitter cable for a than those of its predecessor, the
pair of battery-powered mono mics. Even without this, there are H4.
plenty of possibilities. A number of applications are suggested in The controls of the Zoom H4N —
the detailed manual, and there is plenty more potential. Capturing particularly those on the side panels —
jam sessions in stereo while, say, DI’ing keyboard and bass/guitar are much more intuitive to use than
parts, is easy; or you could similarly take a feed off an FOH desk those of its predecessor, the H4.
and DI a couple of instruments or set up a pair of ambient or
audience mics to give you more options in post production. It also
opens up the possibility of surround-sound recording, and it might
prove useful for applications such as impulse-response capture.
Putting the H4N into four-track mode is as simple as clicking the
menu button, scrolling to ‘Mode’ and clicking, then selecting ‘4CH’
and clicking again. A similar approach is used to set it back to stereo. There’s also an ‘MTR’ multi-tracker
mode, which allows you to record mono or stereo sources to each of your four tracks, apply effects to and
balance those tracks, and create a stereo bounce.

Bells & Whistles

There’s plenty more on offer too, from compressors and low-cut filters, to guitar amp and cabinet emulations
(not the most authentic out there, but certainly passable for demo purposes), and distortion and delay-based
effects.
As with so many of Zoom’s products, there’s literally too much to go into in detail in a review of this length,
and I’d recommend visiting Zoom’s web site if you’re genuinely curious about the range and quality on offer.
Suffice it to say that I’ve called these functions ‘bells & whistles’ as I suspect they won’t sway most people’s
decision to purchase. Even if you don’t plan to use these sort of facilities for mixing, they do provide a
convenient means for headphone-clad guitar and bass practising.
One really useful ‘whistle’ is the four-in, two-out 24-bit, 96kHz-capable audio interface capability, and you
even get a copy of Steinberg’s Cubase LE4 DAW software for Mac and PC thrown in — which makes the
package more than enough to get a would-be home recordist started. It wouldn’t be my first choice as a
hardware interface (this is, after all, designed with location recording applications in mind) but it’s certainly
very usable.

Sound Choice?
In terms of ergonomics and functionality, then, the H4 has undergone quite a transformation to acquire the
extra ‘N’ (which stands for ‘next’ — probably my only real gripe!). But what of the sound quality? Well, that’s
improved too. Tackling the handling-noise issues of the H4 and upgrading the external mic preamps makes
quite a difference. Using the preamps with a pair of AKG C451s, I was able to capture a perfectly usable
sound in the studio. The onboard mics are definitely an improvement, and comparable with those on other
portable recorders in this price range. The two stereo-width settings also give a useful option when trying to
achieve separation between two sources, or simply recording something like an acoustic guitar part in
stereo, while leaving a ‘hole’ in the middle for a vocal.

Conclusion
I’m a big fan of the H4N, and found very little to dislike about it. While there are more ‘professional’ portable
recorders out there, you’d be hard pushed to find something that’s better on sound quality alone. This
product is also aimed at a very different market, where it compares very favourably with the competition. The
H4N is justifiably a little pricier than its H-series predecessors, given the improvements that have been made,
but although there’s a lot of ‘bonus’ functionality I can’t imagine using, you’re not paying a premium for it. If
you want a good handheld recorder, this should definitely be on your shortlist; and if you want simultaneous
four-track recording thrown in, it will be a very short list indeed!

Published in SOS June 2009

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Q. Does mono compatibility still matter?

Q. Does mono compatibility still matter?

Published in SOS June 2009
Sound Advice

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’ve recently started working at a classical radio station in my area, and I was fully expecting to have to deal with mono issues and think
about miking live performance with those in mind. But everything is done in stereo and broadcast in stereo. Spaced omnis are common,
which is not very mono compatible. So when is mono compatibility a necessity, and is mono really ever used any more as a final
‘product’?
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: In a technical sense, mono compatibility is still
important. Whether a particular radio station chooses to bother about it is a decision for them, but
I would suggest it unwise to ignore it completely.
FM radio is transmitted essentially in a Mid/Side format, where the derived mono sum (Mid)
signal is transmitted on the main carrier and the ‘Side’ information is transmitted on a weaker
secondary carrier. A mono radio ignores the Side signal completely, whereas a stereo radio
applies M/S matrix processing to extract normal left and right signals.
However, there is potentially a noise penalty in this process, so in poor reception areas, and
often when on the move in a car, FM receivers are designed to revert to mono, to avoid
reproducing a very hissy stereo signal. As a result, a large amount of in-car listening will be in
mono (at least, here in the UK) because of signal fading and multi-path issues. In addition, a very Even popular modern DAB radios such
as this one from Pure are mono by
large proportion of radio listeners do their listening in the kitchen, bathroom or garden, using
default, and a large part of the potential
portable radios that are usually mono. So mono compatibility is still important to a very large
audience for radio and TV in the UK still
proportion of the potential FM radio audience. listens in mono — so mono compatibility
Amusingly, mono doesn’t even become less relevant in the digital radio market. The most is still a consideration for music
popular DAB digital radio receiver in the UK is currently the Pure Evoke, and although you can producers.
attach an optional second speaker to enjoy stereo from it, by default the stereo output from the
DAB receiver is combined to mono to feed the single internal speaker. So mono compatibility remains important in the digital radio market
too!
Considering TV for a moment, the primary sound on analogue (terrestrial) TV in the UK is in mono, transmitted by an FM carrier
associated with the vision carrier. Although a secondary stereo sound carrier was added in 1991, using a digital system called NICAM, there
are still a lot of small mono TVs on the market. Analogue TV will be switched off in the UK within the next three years, and digital TV (both
terrestrial and satellite) is broadcast entirely in stereo (or surround in some cases) — but even so, it is still possible to buy mono receivers.
So given that a significant proportion of the potential audience (for analogue and digital radio and TV) could well be listening in mono, I’d
suggest that checking and ensuring mono compatibility is still important. I know that some classical radio stations, in particular, argue that
only serious music enthusiasts listen to their output, and they would only do so on decent stereo hi-fi equipment. Perhaps that is the case,
but to my way of thinking, ensuring reasonable mono compatibility is still the safest approach, and needn’t restrict the way broadcast
material is produced in any way at all.
Using spaced omnis is a technique often favoured by classical engineers, largely because of the more natural sound and smoother bass
extension provided by pressure-operated mics. In some situations, particularly when using a single spaced pair, there can be mono
compatibility issues — but only rarely, and it is usually easily fixed. For example, if any additional accent or spot mics are used and panned
into the appropriate spatial positions, any phasing or comb filtering from the spaced omnis, when auditioned in mono, will be diluted and
usually ceases to be an issue. Even in cases where a single spaced pair is used, listening to the derived mono may sound different, but it is
rarely unacceptable.
To sum up, I would definitely recommend checking mono compatibility and trying to ensure that it is acceptable (even if not entirely
perfect). If the sound quality of spaced omnis is preferred, there’s no reason not to use them — even if the final output is mono — provided
suitable skill and care is used in their placement and balance. The BBC certainly use spaced pairs for Radio 3 transmissions in appropriate
situations.

Published in SOS June 2009

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Q. How can I improve my vocal recordings?

Q. How can I improve my vocal recordings?

Published in SOS June 2009
Sound Advice

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’m quite new to vocal recording, and am using either a Neumann TLM103 or AKG C414, depending on the material being recorded. I
have the mic set up with an SE Reflexion Filter in an enclosed corner of my room, with duvets hung around it, and I’m going direct into
my Digidesign Digi 003R audio interface. I get vocalists to sing six to 12 inches away from the mic, and use a pop screen. My problem
is that the vocal tracks are sounding rubbish! The current singer is a nightmare to work with: he sings really quietly one minute, right on the
mic, then shouts right into it, so I have to keep the input gain right down, resulting in a bad recording level. If I use a compressor to track
through, it sounds over-compressed and nasal even if I set the threshold really high. Do you have any tips? I’m a guitarist, not a recording
engineer, which is probably the problem.
Via SOS web site
SOS contributor Mike Senior replies: The first thing to say is that you needn’t really worry about
setting the recording level too low. The quality of the equipment you’re using (and much other
decent project-studio equipment) isn’t going to add appreciable noise to anything you’re close-
miking, and as long as you record at 24-bit resolution you can afford to leave enough headroom
to cope with the singer’s peaks. Just set the level to avoid clipping, which is much more of a
problem on vocal parts. By the same token, there’s now little to be gained by recording through a
compressor, as this could restrict your processing options at mixdown, and there’s not the need
for it now that most of us don’t record at 16-bit.
If your singer gets really loud, there’s some danger of him overloading the analogue circuitry on
the way to the A-D converters. The TLM103 should be able to handle 135dB before distorting, but
in the unlikely event that your singer manages to max that out, you can switch to the AKG and
Some singers need quite a bit of help to
use its pad switch to accommodate levels up to 158dB. He’d have to have eaten heavy artillery to
tame an over-dynamic technique. If you
generate that kind of volume! Even if you prefer the sound of the TLM103, you can record the have to apply a lot of gain reduction, but
padded C414 alongside to hedge your bets, if you’re worried, switching to the C414 for any fairly transparently, certain compressor
phrases or syllables that get scorched on the TLM103. As long as you don’t overload your mics, plug-ins can be invaluable.
and are recording straight into the Digi 003R, I can’t see you overloading those inputs, as they’re
designed to cater for line-level sources too.
However, it’s worth saying that most singers don’t sound as good when singing at their very
loudest. A lot of singers who sound like they’re absolutely caning their vocal cords aren’t, in fact,
that loud in the room — they’re just adept at squeezing a ‘loud’ character out of their voices at
more sensible levels. (This also means that they can do more takes before their voices are
exhausted.) Most voices I’ve encountered tend to thin out as they increase in volume, with a
couple of mid-range resonances dwarfing all the more characterful frequencies, so that the vocal
sound just disappears into the mix. On the other hand, quieter singing tends to let more of the
uniqueness of the voice come through, and tends to sound bigger and closer when faded up in
the mix. This is something that Terry Britten and John Hudson made use of for Tina Turner’s
comeback single, ‘What’s Love Got To Do With It’ (check out our interview with them back in SOS
May 2004), for example.
The headphone balance you set up is a valuable tool for manipulating the singer’s projection, so
be sure to give it enough thought. If you don’t put enough vocal into the cans, or you feed a
compressed vocal into the monitor mix while recording, it’s easy for the singer to feel that they
have to strain to be heard. On the other hand, if you turn their vocal way up in the headphones
and ditch the monitoring compression, they’ll be less inclined to belt things out and are likely to
effectively start ‘self-compressing’ to some degree. However, it’s worth sounding a note of
caution: if the singer tries to even out their levels by adjusting the distance between themselves
and the mic (a common live-performance technique), this can cause more problems than it
solves, because the change in position can dramatically alter the vocal tone, not least because of
the proximity effect on most vocal mics. Such changes can be really fiddly to correct for at the
mix.
You can do a lot to help the vocal timbre with mic technique, as well. Mic positions above the
level of the mouth might be worth investigating, as they’re normally smoother (and you get less
sibilance too). Also consider putting some acoustic foam on the ceiling if it’s of average height, as
that might be contributing a nasty coloration if it’s sending a strong reflection into the mic. Neither
the Reflexion Filter nor your duvets are stopping sound reaching the top of the mic.
If you can get a good, clean take into your system, you can tackle dynamic range issues with a
cool head and in your own time. The voice is one of the most dynamic sources you’ll encounter in

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Q. How can I improve my vocal recordings?

record production, so it’s normal for this task to take some careful work. If you’re having trouble
finding compressors that can cope, and you can run VST plug-ins, there are a few freeware ones
you might want to try: Jeroen Breebaart’s PC2 (with its Complex button switched in), Antress
Modern’s Painkiller, and the Tin Brooke Tales TLS3127 Leveling Amplifier. All of these are good
at delivering a great deal of gain-reduction fairly transparently. If you can’t load VST plug-ins, my
advice is to compress your vocal in stages, using more than one compressor. One good general-
purpose combination is a slower acting (and perhaps soft-knee) compressor, followed by a faster
peak limiter to catch rogue peaks. Mastering limiters can work surprisingly well in this application,
as their whole purpose is to squeeze levels with the minimum of side-effects.

Published in SOS June 2009

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Q. What’s the best order for mixing?

Q. What’s the best order for mixing?

Published in SOS June 2009
Sound Advice

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’ve been wondering what order people use when mixing. Mixing the instruments in order of priority? Mixing the rhythm section first?
Via SOS web site
SOS contributor Mike Senior replies: I’ve spent the last couple of years researching and
comparing the techniques of many of the world’s top engineers, and you might be surprised to
discover that they disagree considerably on the issue of the order in which to deal with the
different aspects of a mix. On this basis, it would be tempting to think that your mixing order isn’t
actually that important, but I think that this is a mistake, as in my experience it can have a
tremendous impact on how a mix turns out.
One reason for this is that each different track in your mix has the potential to obscure (or
‘mask’) certain frequency regions of any other track. The primary way to combat frequency
masking is to reduce the level of the specific problem frequency-range for the less important
instrument, letting the other one shine through better. So it makes a good deal of sense to start
your mix with the most important track and then add in successively less important tracks, simply
so that you can take a methodical approach to dealing with the masking problem. If any track you
introduce is obscuring important elements of a more important track that is already in the mix, you
Deciding on the right order for mixing
set about EQ’ing the problem frequencies out of the newly added track. If you don’t introduce
your tracks might well depend on the
important tracks until later, you’ll tend to find difficulty in getting them to sound clear enough in the
genre in which you’re working. The
mix, because there will now be umpteen less important tracks muddying the water. This is a approach could be very different on a
common problem for those who only introduce their lead vocal track right at the end of the mix, Rihanna mix than one of Dido’s, for
and can often lead to an over-processed and unmusical end result. example.
Another persuasive reason for addressing the most important tracks first is that in practice
almost everyone has mixing resources that are limited to some extent. If you’re mixing in the
analogue domain, you’ll already be well acquainted with the frustration of only having a few of
your favourite processors, but even in the digital domain there are only a certain number of CPU
cycles available in any given hardware configuration, so some compromise is usually necessary,
by which I mean using CPU-intensive processing only for a smaller number of tracks. In this
context, if you start your mix with the most important instruments, you’re not only less likely to
over-process them, but you’ll also be able to use your best processors on them — an improved
sonic outcome on two counts!
Taking another look at different engineers’ mixing-order preferences in the light of these issues,
the disparity in their opinions begins to make more sense if seen in the context of the music genre
they’re working in. In rock and dance music styles, for example, people often express a
preference for starting a mix with the rhythm section, while those working in poppier styles will
frequently favour starting with the vocals. As a couple of examples to typify how this tends to
affect the mix, try comparing Rihanna’s recent smash ‘Umbrella’ with something like Dido’s ‘White
Flag’. The first is built up around the drums, while the second has been constructed around the lead vocal, and you can clearly hear how
various subsidiary sounds have been heavily processed, where necessary, to keep them out of the way of the main feature in each
instance. In the case of ‘Umbrella’, check out the wafer-thin upper synths, whereas in ‘White Flag’ listen for the seriously fragile acoustic
guitars.

Published in SOS June 2009

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Q. Which room should I record in?

Q. Which room should I record in?

Published in SOS June 2009
Sound Advice

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am about to do a recording at a farm in the North York Moors. It will be done ‘live’, using two acoustic guitars and two voices, and we
will add bits of percussion, mandolin and accordion afterwards. The site has a choice of buildings to set up in (using my own
equipment). Should I choose the huge, high-ceilinged barn or the small, cosy stable?
Via SOS web site
SOS contributor Martin Walker replies: It’s really difficult to generalise about the acoustics of a
room without seeing it personally. The small, cosy stable might provide a more intimate acoustic
that works well with your small acoustic ensemble, especially if it includes lots of wooden stalls
that result in lots of pleasing diffused reflections. On the other hand, it might sound really nasty,
depending on its dimensions and whether it’s built of rough stone or breeze blocks. In general,
smaller rooms can tend to sound more ‘boxy’.
A larger space with good dimensions can exhibit a flatter response down to a lower frequency,
and hence better acoustics, and it may give you a richer and grander reverberation whose amount
With a choice of large recording spaces
you could alter by how close you place the mics to the performers. A larger space also provides
to work in, acoustic screens can be very
you with more opportunities to place several mics at different distances from the performers; close useful for tailoring the room
ones capturing the intimacy of the performance, and more distant ones capturing the ambience of characteristics to suit your players.
the space, recorded on to additional audio tracks that can be later mixed in to taste. However, this
doesn’t necessarily mean that your particular barn will sound good, especially if you end up with
several discrete reflections coming back off plain, unadorned walls and a concrete floor.
Ultimately, you really do have to use your ears. Don’t worry if you don’t have sufficient experience to judge room acoustics immediately
on entering the space. Just set up one of the performers (with an acoustic guitar, for instance), put on some high-quality, closed-back
headphones and move both performer and mic about while you monitor their performance. While this will be vital in helping you find the
optimum mic position to record each instrument and voice, it will also tell you a lot about the room acoustics (probably a lot more than
clapping a couple of times, as so many people do).
Acoustic guitars often benefit from a ‘live’ sound, so you may find it beneficial to place the performers near some reflective surfaces such
as doors, a hard floor, or those stable stalls. You may also find that using a couple of mics on each instrument works well, such as one
below the bridge and another near the neck of the guitars. You can find lots more useful advice on mic positions and distances for recording
acoustic guitar in SOS August 2001 (www.soundonsound.com/sos/aug01/articles/recacgtr0801.asp).
If the room sound proves to be poor, it’s handy to have a few movable acoustic screens (or even improvise some with clothes horses and
duvets) to add some extra absorption close to the performers. Such screens can also be used to increase acoustic separation between the
players. However, I suspect that your safest option is not to restrict yourself to the stable or barn. Since you’re recording at a farm, see if
other rooms in the farmhouse itself could be used. Many excellent acoustic ensemble recordings have emerged from such cosy
environments, which normally contain enough furniture to provide plenty of absorption and diffusion. Starting with a room that sounds good
is always easier than attempting to knock a poor one into shape.

Published in SOS June 2009

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Origin Of The Species

Origin Of The Species

Leader
Published in SOS June 2009
People + Opinion : Miscellaneous

A
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coustic instruments have evolved in a strictly ‘Darwinian’ sense — the ones that work well together have survived, and the ones that
failed to find a place in the grander scheme of things faded into obscurity. The symphony orchestra is a perfect example: the timbres
and levels of the various groups of instruments complement each other with no requirement for EQ, compression or even gain
controls. If the flutes aren’t loud enough, you simply get more flutes! In more ‘compact’ forms of music, such as chamber orchestras, string
quartets or jazz bands, instruments tended, again, to be combined in naturally symbiotic groups, but all that changed with the modern ‘built
for volume’ drum kit and the electric guitar. While a drum kit played by a jazz musician can fit in with louder acoustic instruments, such as
the saxophone, there are few acoustic instruments that form a natural balance with a modern rock drum kit. Indeed, the electric guitar
evolved out of the necessity to be heard, but now the tables have been turned, because with today’s high-power amplification the electric
guitar can go loud enough to make your eardrums meet in the middle of your head, leaving the hapless drummer to shout “Unfair” and
demand microphones!
When you close-mic a drum kit, the sound is very different to that of the same acoustic kit played in a room, and once you amplify a
guitar, the sound can be processed in a number of ways, from twangy and clean to grungy and distorted. This flexibility is great for artistic
freedom, but it also means that the instruments no longer have a natural balance, so we have to create it ourselves using tools such as EQ
and compression. The development of the synthesizer posed similar challenges, as it can create sound in any or all parts of the musical
spectrum and at any desired level.
However, it would be wrong to place the entire burden of forcing these sounds to work together on the studio engineer. The tonal colours
of amplified instruments need to be worked on at source to get them to combine in a mutually supportive way, and that means not only
finding suitable guitar and synth sounds to mesh with the other instruments and voices, but also working on the musical arrangement so
that each instrument plays in the correct register and at the right time. If every instrument in the orchestra played solidly through every
composition, the result would sound very congested, as all parts of the audio spectrum would be filled all of the time, yet this is exactly what
happens when a single, heavily distorted rhythm guitar is strummed without a break from the count-in to the final cymbal crash. The less
thought that goes into the arrangement, the more the engineer has to fit a square peg into a round hole by EQ’ing off the corners. So if
you’ve ever wondered why the great pop records sound so good, the answer might just be that they were carefully arranged in the first
place.
Paul White Editor In Chief

Published in SOS June 2009

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Playback

Playback
Readers’ Music Reviewed Buy PDF
Published in SOS June 2009
People + Opinion : ReaderZone
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demo
Belle Tropez
On first acquaintance, Belle Tropez threaten to be one of the many Zero 7 soundalikes clogging
up the Playback files. They blend soft vocals, easy-listening grooves and tasteful synth beds in a
fashion that’s highly skilled but, on the face of it, hardly revolutionary.
As their music develops, however, it slowly becomes apparent that Belle Tropez have
something that’s missing in so many of their rivals. All too often, bands in this vein focus their
energy on the production, with no regard for the actual material they’re producing. The great thing
about Belle Tropez is thus not the excellent male and female vocals, nor the sumptuous brass
playing; nor is it their clever drum programming or neat integration of glitchy manipulated
samples.
No, what stands out above all is their musical sophistication, which stretches way beyond usual
dull three-chord tricks. The third track, ‘Pull It’, for instance, sounds so smooth and natural that it
took me an age to realise it’s actually in 7/4, while elsewhere, lengthy chord sequences develop
with a richness and confidence that’s so often missing in electronica. Frustrating, then, that
they’ve sent their demo in on a dodgy CD-R that will only play for half a track at a time without
skipping! Sam Inglis
www.myspace.com/belletropez

Compos Mentis
Captain Yange
Is that Yange as in ‘flange’, or Yange as in ‘mange’? I know not, but the good Captain’s war
record in the Stoke On Trent campaign of 2007 earns him a medal for bravery in attempting to
rescue falsetto vocals from the teeth of ridicule. He gets away with it, too, and it’s refreshing to
hear a band who have the songs to carry an entire album.
Captain Yange also have a definite production aesthetic, which they carry through with
impressive consistency, although I’m not sure that their take on modern rock will be for everyone.
The sound is uncomfortably claustrophobic, with a million distorted and fiercely compressed
sounds fighting for your attention. If your idea of rock heaven is being stuck in a lift with Muse,
Captain Yange are the band for you. Sam Inglis
www.myspace.com/captainyange

Hey Big Rock Star

Charlie Baxter
“Hey big rock star show me how to write a song,” go the lyrics of the opening track on Charlie
Baxter’s demo. Unfortunately, what’s missing from his plea, in my mind, is the word ‘good’. What
follows track one is an uninspiring collection of sort of nu-rave-flavoured tracks treated with a
dollop of pop-like production, for which Charlie should be commended. The tonal balance and
clarity of the instruments is pretty good, especially considering the sheer number of synth-based
sounds that have been employed.
However, the timing is awfully loose (if it’s deliberate, it’s a bit too lazy for my ears), and
Charlie’s voice is, well, rubbish. In conclusion, I can see some potential in Charlie Baxter, just not
behind the mic. Chris Mayes-Wright
www.myspace.com/charleswbaxter

The Conker Collection

Conker
Andy Harrison and Peter Harris are, it says here, “two musicians summoned to the shed of truth in
Wolverhampton”. I’d say they’ve been led up the garden path. In fact I’d say they’ve been led up
the garden path, round the compost heap and straight into the Gazebo of Rubbish Synth Presets.

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Playback

And I wouldn’t be entirely surprised to learn that they’d spent time in the tree-house of ugly guitar
sounds and the Nissen hut of cheesy drum patterns, too.
It’s a shame that Conker’s material had to be sacrificed on the altar of home-keyboard string
patches, because they have some nice songs, some of which recall XTC in their more pastoral
moments. I was going to add that their two voices harmonise astonishingly closely, but then I
noticed that Andy did all the vocals, so perhaps that’s hardly surprising. I will, however, say that in
“wizardry”, they have chosen too strong a word to describe the “Cubase engineering skills” on
display here. Sam Inglis
www.myspace.com/conkerband

ReDiscovery
Frisbee
On opening this package and discovering that Frisbee is actually the surname of Aaron Frisbee,
SOS reader and singer-songwriter, I was hoping for a compelling back story. You know the sort of
thing: inherits vast fortune from grandfather’s flying plastic disc empire, squanders fortune on
drink and drugs, finds redemption in vaguely Elliott Smith-esque AOR, emerges poorer but with a
cult following in Belgium.
No such luck, but fortunately Frisbee’s vaguely Elliott Smith-esque AOR is strong enough to
stand on its own. The whole album is beautifully recorded and mixed, and Aaron is possessed of
a fine voice, which is tastefully augmented with backing vocals and supported by thoughtful and
varied instrumentation. In fact, if I have a criticism, it’s that the whole thing occasionally feels too
smooth and polished, and might have more emotional impact if it were a little rougher round the
edges. But that, frankly, is nit-picking, because this is high-quality stuff, and there are plenty of
major-label releases that struggle to match it. Sam Inglis
www.frisbeemusic.com

Unspoilt By Daylight
Cellsonik
Graham Robinson, aka Cellsonik, has been producing electronic music since 1992. He admits
that he’s not the fastest songwriter (he only finished this CD in 2007), but, curiously, there are
elements here that I feel he should have spent a bit more time on.
Unspoilt By Daylight is a 15-track album covering an assortment of electronica bases —
including breakbeat, trance, drum & bass and house — to varying degrees of success. I have to
say I’m not too keen on some of the synth parts, which smack a little of lazy preset-tickling and, to
my ears at least, lack the kind of attention to detail that sets a really good dance tune apart from
the rest.
It’s not all bad though, fortunately. The drum programming, particularly on the handful of jungle
tracks (definitely Graham’s forté), sounds much more authentic and ‘pro’ — it’s just a shame that
the percussion is pushed so far back in most of the mixes. My advice to him would be to develop
that side of things and make the drums the main focus in his music, as this would allow him to be
a bit more sparing with the (none-too-convincing, frankly) synth sounds that pepper this album.
Chris Korff
www.cellsonik.com

Pirouette
Stickboy
Something about Stickboy’s music reminds me of Colin Vearncombe, who scored a series of hits
in the late ’80s under the name Black. In both cases, melancholy vocals and jaunty musical
backing are juxtaposed in a way that ought to jar, but actually works surprisingly well. However,
it’s a shame about the mastering on Stickboy’s effort, which sees an otherwise decent mix pushed
to the point where it becomes crunchy and uncomfortable to listen to. Sam Inglis
www.myspace.com/stickboyuk

Way Out South

Natascha Roth

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Playback

South African duo Natascha Roth and James Scholfield were lucky enough to borrow some
particularly tasty gear for the making of this album, and they’ve made full use of it. Their sound
layers arpeggiated guitars from Schofield with Emmylou Harris-style warbling from Roth, and
since the overall feel is very gentle and downbeat, there is precious little room for bad sounds to
hide. Happily, then, the playing, singing and recording are all excellent, and despite the slow
pacing, there’s enough instrumental variety to stop it getting too samey.
www.myspace.com/natascharoth

demo
We Happy Few
We Happy Few describe their music as ‘alternative blues’, and they have a garage-y, British take
on the genre, reminiscent of bands like the Bishops. It sounds as though it was recorded live,
which is definitely the way to go, but it seems to me that We Happy Few need a producer to crack
the whip over their rhythm section.
Upright bass is a notoriously difficult instrument to record well, and whether it’s the room,
pickup, mic placement or the instrument itself, some notes here boom uncontrollably, while others
are barely audible. Kick and snare, likewise, disappear into the mix, and the kit as a whole sounds
gutless and lacking in weight. The resulting sound could politely be described as ‘rough and
ready’, and seems to exaggerate small timing discrepancies between bass and drums, so that the
feel is often lumpy rather than locked-in. If We Happy Few can find a way of sorting this out, they’ll
be able to do full justice to their material. Sam Inglis
www.myspace.com/wehappyfewmusic

EP
Libelula
Now this is a bit special. Libelula are like the White Stripes of electronica, insomuch as they are a
male/female duo who share a surname but don’t discuss their marital status. Musically, they take
a well-trodden electronica-with-dreamy-female-vocal path, but do so with rare skill. Sarah
Villarous has a gorgeous voice, and deploys it to full effect over backing tracks that are by turns
lush, brutal, sinister and atmospheric. You could soundtrack a year’s worth of car commercials
from the four tracks included on this EP. And I mean that in a good way. Sam Inglis
www.libelula.co.uk
645102

Playback Is Sponsored By Breed Media

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of 1000 CDs to the lucky winner. This amazing prize will be professionally pressed, and includes glass mastering, print, cases and
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Playback

This month’s winners are Belle Tropez.

+44 (0)114 255 2460
Click here to email
www.breed-media.co.uk

Published in SOS June 2009

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Raphael Saadiq

Raphael Saadiq
Producing The Way I See It Buy PDF
Published in SOS June 2009
People + Opinion : Artists/Engineers/Producers/Programmers
Printer-friendly version
Artist and producer Raphael Saadiq has channelled his love of classic soul records to create
something convincingly vintage, yet fresh-sounding and alive.
Richard Buskin

I t’s one thing to recreate a legendary sound, quite another to authentically recapture a specific
feel, and to do so with completely new material. That’s what Raphael Saadiq has achieved on
his latest opus, The Way I See It. Released last year in the US to widespread praise and
recognition, including a trio of Grammy nominations and iTunes’ selection as the Best Album of
2008, it’s finally had an official release in the UK.
Born Charlie Ray Wiggins in Oakland, California, in 1966, Saadiq was a self-taught multi-
instrumentalist by the age of six, playing guitar, bass and drums, with the bass already his
instrument of choice. By the age of nine he was singing in a local gospel group, and it was under
the name Raphael Wiggins that he commenced his professional career, supporting Prince on his
1986 Parade tour before joining forces a couple of years later with brother Dwayne Wiggins and
cousin Timothy Christian in Tony! Toni! Toné!, who enjoyed mainstream success with their 1990
second album, The Revival.
Assuming the name of Raphael Saadiq during the mid-’90s, he began to expand his operations
into the production sphere, with projects material by fellow artists such as Macy Gray, TLC, the
Roots and D’Angelo, earning the last a 2000 Grammy Award for the song ‘Untitled’. Then, just
over two years later, he released his first solo album, Instant Vintage, on his own Pookie
Entertainment label. A collection of what he calls “gospedelic” tracks blending samples, soul,
gospel and R&B, it made him the first artist without a major label affiliation to garner five Grammy
nominations.
For The Way I See It, Saadiq signed to Columbia Records, but his unique retro-futuristic vision
remains intact. “While I was making the record, I watched videos by Gladys Knight & the Pips, Al Green and the Four Tops, and fused them
all together,” he says. “Once I got into this, I got almost stuck in character, the character of the old-school singers I listened to This album is
the culmination of a lifetime of experiences informed by the music I grew up on.”

An Inspirational Trip
In planning the album, Saadiq also drew on his experiences on a trip to Costa Rica and the
Bahamas. “I was surfing and ran into people from all kinds of places,” Saadiq explains, “and I
noticed everybody was listening to this classic soul music. When I came back home, the music for
this album flowed organically, naturally, and since I have my own studio I was able to perfect it
and take my time to make it right. I was able to live with it day after day, and I think that had a lot
to do with how the album turned out. It took about four months to put it all together.
“I’ve always wanted to be able to sing a two- or three-minute song and make people want to
hear it again. Stax did that, and so did Motown and the Beatles: artists who made real popular
songs that touch my soul. It’s the music that brings people together, the music that can even
Raphael Saadiq (left) and engineer
make animals stand there and listen. I always like to make music that will appeal to other
Charles Brungardt at Blakeslee Studios
musicans, as well as to people who listen to very commercial music; the cool cats of 40 to 50, the
during the sessions for The Way I See
cool kids of 15 to 16, the cool black rapper I want to bring all of those people together, because It.
that’s how music should be instead of how it is right now, which is really segregated.”
A case in point is the part-Spanish-language, doo-wop-flavoured “Callin’”, which Saadiq describes as “a jump back to the music of the
’50s. I wanted to make a track that would get the lowriders. People talk about the division between Latinos and blacks, but we all grew up
together loving the same music. This song is a reminder of how we do when we get together.
“Honestly, when I made this record I wasn’t paying attention to trying to recapture a particular sound. I was just being me. I wasn’t trying
to do a Temptations song or a Smokey song by the time I snapped my fingers and tapped my feet on the floor, I was there. I really lived it. I
wasn’t immersing myself in a role. It was like having a great dream and not wanting to wake up. That’s why when people ask, ‘What’s your
next album going to sound like?’ I respond, ‘What do you mean? This is me.’ That’s why the record is called The Way I See It. It’s me, more
than anything else I’ve ever done. In fact, making it has made me feel like I’ve never even done another record.
“I wrote all of the songs on the fly, most of the time with a guitar in my hand. I’d come up with some riffs, sing the song in my head, and I
basically did this on my own. I would love to bounce ideas off other people, do some writing with them, take the material to my band and
say, ‘OK, let’s cut it,’ with the orchestra already there. That’s my dream. I’d crank records out weekly if I had staff writers like they did at

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Raphael Saadiq

Stax and Motown, whereas right now, given the state of the industry, I had to sit in a room all by myself except for Chuck Brungardt, sing
each song to myself while playing the drums, and then play the guitar over the top of that, play the bass, play some basic piano, do the
vocal and record the strings later.”

About The Soul

“I find it easier to produce my own vocals with nobody in the room,” Saadiq asserts, “otherwise I’ll
be looking for answers from somebody who may not really know. I tend to record complete takes,
and if something isn’t quite right but it’s got a feel that I know I can never ever capture again, I’ll
leave it, even if it’s flat. I mean, there are flat parts on my record, because it’s not about
perfection, it’s about the soul.
“I’m pretty good at knowing when something is complete. It’s pretty easy for me to figure out
when it’s done. The way I live with it, I like to listen for hours late at night. That’s part of producing
it; listening to it and knowing when it moves me. I feel like it’s marinating, sitting it in the pot
overnight and letting it soak. When I play it, I know how it moves me and I feel like it’ll move
consumers in the same way. Having said that, I do work on multiple tracks at the same time, and
that’s really the only way I can work. I can work on a song for one or two days, but then I’ll jump
to another one. I might also get an idea for a song while I’m in the middle of working on another,
but ADD [attention deficit disorder] does play a part!
“I grew up playing in a quartet, and these days, even when I’m playing on my own, I’m just
grooving. I’m in it and you can’t take me out of it. If anyone watches me, they’ll think, ‘What is he doing?’ Once you start clapping along to
something, you’re locked into it, and that’s what I bank on.”

The Old School

Saadiq’s Blakeslee studio in North Hollywood combines Pro Tools, an SSL 9000 and other state-
of-the-art equipment with a plethora of vintage gear, including a kick-drum mic that was
purchased from Abbey Road. Indeed, as a means of preparing for the aformentioned sessions, he
and engineer Charles Brungardt familiarised themselves with old techniques by reading books
such as Mark Lewisohn’s The Complete Beatles Recording Sessions.
Accordingly, the recording setup was fairly basic in terms of both the miking and the outboard
gear, with a Neumann U47 providing character to the guitars, and a U47 and U67 alternating as
the overhead on Saadiq’s specially purchased ’60s Ludwig drum kit, while a combination of AKG
D12 and C414 were employed on the kick, and a variety of fairly standard mics were used to
The control room at Blakeslee houses
attain a solid crack on the snare. Some second-hand Ampex tape machines were also acquired to
an SSL 9000 desk.
add extra warmth to the kit; Brungardt followed the advice of the salesman in removing the
preamps and using them as a front end for Pro Tools.
For his vocals, Saadiq sought to embellish his characteristically clean delivery with a little edge, thickness and distortion by way of singing
into a Shure SM7 dynamic mic, while Brungardt added some compression before employing a Fairchild compressor and Massey Tape-
Head plug-in during the mix.

Cheating On The Girl

Just like many of the classic Motown recordings of the ’60s and early ’70s, The Way I See It
boasts an energy and an infectious quality that grabs the listener and keeps his or her attention
throughout the two-to-four-minute duration of each of the album’s 13 tracks. And this, in turn,
lends itself to some prime material for Saadiq’s ongoing concert performances, having already
toured Europe last summer, before spending November and December supporting John Legend.
“That’s what music always used to be about,” he remarks. “People were in love with buying
records because they were in love with the artists. These days, they’re not in love with them, and
that’s why they’re not in love with buying records. When you’re in love with a girl, you buy her
everything she wants, but the state of the industry right now is like we’ve been cheating on the
The main live area at Blakeslee.
girl. If you do that, she’s going to leave you. So we’ve got to start loving the girl again. We’ve got
to start making music to go perform it in front of people, and if they love it they won’t care what
format you put it in. They’ll buy it digital or on vinyl.”
Which is why The Way I See It has been issued both on CD and as a collector’s edition box of seven-inch, 45rpm singles. Regardless of
the medium, the content signals a return to the joyous and liberating feel of ’50s and ’60s popular music, and it is a feel that Raphael
Saadiq wishes to retain well into the future.
“I want to continue to make this type of popular music,” he says. “It’s the ride I want to ride on. Of course, it could change a little bit, but I
don’t want to do anything too slick. I love making people move in a bluesy way.”

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Raphael Saadiq

The Wonder Stuff

The Way I See It features a number of star turns from celebrity guests, most notably Stevie Wonder, who contributes his idiosyncratic,
instantly recognisable style of harmonica playing to the Motown/Marvin Gaye-flavoured ‘Never Give You Up’. A smooth, mid-tempo
number co-written with Charles ‘CJ’ Hilton, Jr, it sees the main artist interrupting his lead vocal to announce, “I’d like to invite Mr Stevie
Wonder to my album. Come on, Stevie!”
“It was one of those things where the stars kind of lined up,” Saadiq remarks. “CJ Hilton actually played the drums and keyboards on
that song — he was in the ‘B’ room — while I was on bass and guitar, and I said, ‘CJ, you sing your verse, I’ll sing my verse. Let’s get it
done.’ We’d both been procrastinating in terms of writing the lyrics, so we went in and got it done, and I did the whole Stevie rap before
Stevie was even on the record. That meant I was either going to have to erase that part or, if Stevie himself didn’t play it, sing the
harmonica solo and act like it was Stevie, jokingly saying, ‘Doesn’t it feel like he should be here?’
“I already knew Stevie, and whenever I’d called him in the past he’d never answered the phone, but this time he actually picked it up
and said, ‘What’s up, fool?’ Those were his exact words. I told him, ‘Man, I’ve got this record and I need you to play a harmonica solo.’
He said, ‘When do you need me?’ It was like 12.30, so I said, ‘In an hour.’ He said, ‘An hour, man?’ ‘An hour.’ ‘An hour?’ Stevie won’t go
nowhere for nobody in an hour. If you say an hour, he’ll show up in two weeks. However, he showed up in an hour and a half. He walked
in the room, played some songs on the keyboards for a minute, and then he said, ‘OK, let me hear the song.’ So I played it, and he did
the harmonica solo and said, ‘Is that what you need?’ I said, ‘That’s it.’ ‘That’s good for you?’ ‘That’s good.’ ‘Do you want me to do it
again?’ ‘No, it’s great.’ ‘Let me do it again, let me do it again.’ So, he played it again, and then he said, ‘Let me do it one more time.’
Which is what happened, and that’s what ended up on the record.”

Published in SOS June 2009

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Secrets Of The Mix Engineers: Declan Gaffney

Secrets Of The Mix Engineers: Declan Gaffney

U2 : 'No Line On The Horizon' Buy PDF
Published in SOS June 2009
People + Opinion : Artists/Engineers/Producers/Programmers
Printer-friendly version
The sessions for U2’s No Line On The Horizon took the idea of spontanaeity in the studio to
new levels. Engineer Declan Gaffney was the man charged with creating order from apparent
chaos...
Paul Tingen

J imi Hendrix, reportedly, was one of the first artists whose creative process involved having
the tape running all the time in the recording studio. It’s a sprawling way of working that
involves endless trawling through recorded material to mine the highs from the humdrum.
U2 are among the most famous present-day adherents of this working method, and, with
technology more complex and recording budgets much larger, the Irish band’s approach is vastly
more expansive and Byzantine than Hendrix could ever have dreamt of. In fact, it is so intricate
and seemingly endless that guitarist The Edge recently joked that U2 albums “don’t get finished,
they just get released”.
The band’s latest ‘unfinished’ release, No Line On The Horizon, was the result of sessions
lasting almost two years, beginning in June 2007 in Fez (Morocco), then moving on to The Edge’s
place in the South of France, to U2’s Hanover Quay studio in Dublin, Platinum Sound Studios,
New York, and finally, in late 2008 and early 2009, to Olympic Studios in London. It involved U2’s
customary process of writing, recording, editing and mixing, followed by ceaseless rewriting, re-
Declan Gaffney (right) with co-producers
recording, re-editing, and remixing, in any possible order and often simultaneously. At least 20 Steve Lillywhite (left) and Daniel Lanois
people were directly involved in assisting the band during the sessions, including a posse of (centre).
engineers, mixers, assistants, drum, guitar and studio techs, and so on, plus, of course producers
Daniel Lanois (see box) and Brian Eno, with help from Steve Lillywhite.
Central to the technical side of things was Dubliner Declan Gaffney, who engineered three songs, mixed ‘Get On Your Boots’ and ‘White
As Snow’, co-mixed five others, and has additional engineering credits for almost all the tracks. It’s fair to say that No Line On The Horizon
was a jump into the deep end for 27-year old Gaffney, whose previous CV constitutes mainly assistant credits. Gaffney spent three and a
half years working with Van Morrison, followed by a year at Windmill Lane Studios in Dublin, and another year at Metropolis in London, and
was asked in February 2008 whether he’d be willing to work in U2’s studio in Dublin as an assistant engineer, where his role quickly
became more important.

Keeping Track Of It All

One of Gaffney’s main tasks was to help keep track of what was happening. “My job when I first
started working with U2 was to tape-op the Radar [digital multitrack], and man the DAT recorder,
which runs all the time. The main thing about working with Radar is that you can very quickly
move from one song to the next. When U2 have put down an idea they’ll come up with new ideas
during the next playback and will just pick up a microphone and start singing or playing guitar, and
with Radar you can drop these things in on the fly, without stopping. I couldn’t imagine doing a
writing session with these guys using Pro Tools, because you simply don’t have time to create
three new tracks and put them in record. That new guitar or drum part could have vanished into
thin air by then.
“The only thing that’s tricky at the studio here is that we still manually patch everything; there’s
no such thing as routing. So when the sessions move at a million miles an hour, which they do, it U2 took up residence at Olympic
can be really frantic to re-patch things when they suddenly decide to work on another song. The Studios, London, during the late stages
outputs of the mixing desk are split to go to the Radar and the DAT, and I made notes of of the making of No Line On The
everything that was going on. Sometimes you get a directive, like ‘Mark so and so on the DAT,’ Horizon.
but for most of the time you need to use your own initiative. Literally months later someone will Photo: Kevin Davies
turn round and say ‘We did this Joy Division-like bass line on this song one day, can you find
that?’ You then need to go through your notes to see if there is a reference to a Joy Division-like bass part, and use your own memory. It
may sound like looking for a needle in a haystack, but funnily enough, the system works. The assistant engineer makes notes of absolutely
everything, no matter how insignificant it may seem. The band comes up with millions of ideas all the time, and everything gets a name.
After we moved the project to Pro Tools the comments fields were also always full with information, so that if three months later someone
wanted a hear the hook in a chorus of such and such a take, you could go back seven Playlists and find it.”
Gaffney must have impressed the other members of the company, because he soon graduated
from tape-op to acting as a fully fledged engineer and mixer. And so, when the band decided in
June 2008 to go to Edge’s house in the south of France for some more writing sessions, Gaffney

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Secrets Of The Mix Engineers: Declan Gaffney

was invited along. By this time, the project was moved over to Pro Tools. “We wanted to keep the
recording setup small, so we just had a few microphones, a Control 24, Pro Tools, and three
drives, so we could move easily and quickly between different songs. Other than Pro Tools, the
recording setups for most of the sessions were pretty similar. Edge’s old Neve desk was pulled
apart for on-location recording, and its 1091 and 1093 mic preamps were racked. Everything was Daniel Lanois at the riad in Fez where
going through these mic pres, or a small Neve sidecar, then through any required outboard, and No Line On The Horizon was begun.
monitoring was via two Mackie 24-channel desks. When we went to Platinum in New York we Photo: Anton Corbijn
had a much larger setup, with guitars and amplifiers everywhere. The idea had been to mix at
Platinum, but the band was still adding overdubs, or wanted sections added or taken out during
mixing. This will happen right up until the morning the album goes to mastering. The songs go through so many permutations, in some
cases you wouldn’t recognise the original versions.”
As Daniel Lanois explains, ‘Get On Your Boots’, the first single from No Line On The Horizon, was a case in point. “It came from The
Edge’s workshop. He had that riff all along, and he was very excited about it, and we served it the best we could. Almost all the other
material came from the sessions in Fez and France, and their beginnings were rhythmic. Brian came in with a lot of rhythmic computer
preparations, which he piped into Larry’s headphones, and Larry then improvised a beat running in tandem with these Eno beginnings. This
immediately brought us to a fantastic rhythmic place, and gave us the opportunity to approach our instruments in a way that we had never
done before.
“Bono’s singing is fantastic these days, so we were also afforded the luxury of some great vocal performances, live with the band. The
best emotional tracks are always recorded when you have the singer in the room with you, and that was the case on this record. We don’t
like living with a promise: we pursue something because we are excited about it. So a lot of the effects get printed along the way. We only
ever operate on excitement, and it’s not a good idea to try to recreate effects on another day. We also don’t want to wait for the mix. For
example, I’d do all kinds of things to The Edge’s guitar sound, putting it back through an amplifier and re-miking it, and so on. He also did
some nice slide guitar solos on the record that I processed through various outboard boxes to make them as exciting as possible.”

Early Beginnings
‘Get On Your Boots’ was recorded by the band in the first half of 2008 at Hanover Quay, and then
extensively reworked there. “Edge came in with a version of the song that had the riff. When the
band started a new version of the song in Dublin, we recorded Edge’s guitar, a loop by Eno,
drums, Adam’s bass, a guide vocal, and overdubbed the percussion and keyboards. Some
sections were slightly longer than in the released version, and there used to be this extended
guitar solo that Dan had taken sections of and turned them around, so you had alternating
forwards and backwards guitar. Then one evening Edge had the idea for half-time drums
underneath the guitar, and everyone was like ‘That’s fucking cool!’ One night Edge also
overdubbed a really cool guitar part, that was named ‘Spirit of punk rock’, which is a reference to
‘The spirit of jazz’, a character in the Mighty Boosh TV show. U2’s The Edge and Adam Clayton
thrash out an arrangement in Fez.
“Generally speaking, the last thing to be added to a U2 track is the lead vocal, so when we were
Photo: Anton Corbijn
in France, Bono worked extensively on the lyrics, the phrasing and getting the right vocal
approach. Of the other tracks I was involved in mixing, ‘Fez’ came out of Edge playing this cool
guitar sound in Fez, and Danny sampled it and chopped it up and remapped out the guitar part, and put some kind of rhythmic element
behind it. Then Brian treated it and added atmospherics — you can hear a Moroccan marketplace, for instance. Brian, Dave Emery and I
mixed the track in relay fashion; the two of them started the mix and I finished it towards the end of the project. Dan and I mixed ‘Cedars Of
Lebanon’ together, live, with everyone in the room, on a K-series SSL at Platinum. Dan insisted on doing this as a performance mix: we
redid each mix pass from scratch, rather than use automation and tweak previous mixes. Dan and I also mixed ‘Moment Of Surrender’ and
‘Unknown Caller’ at Platinum on the console, and we then used stems from these mixes to tweak them when we were at Olympic in
London. But the mixing work on ‘Boots’ in Olympia was done in the box. When we came to Platinum we had these huge analogue SSL
desks and we decided to run some of the in-the-box mixes via them, and compared that to the in-the-box versions, and we decided to stay
in the box where we could.”
Lanois has always advocated the idea of mixing as a performance, and unsurprisingly, his two
studios, in Los Angeles and Toronto, are each built around a console, a 38-channel Neve 8068
and a Midas 4000 respectively. So how does he retain that performance aspect while working in
the box? Lanois: “It’s difficult, but I found a way of working with it that still allows me to take
advantage of my instincts. I don’t sit behind the screen myself, but work with an engineer, like the
mighty Declan, and I get very specific, saying things like: ‘Go to this section, take this out, put this
in, make this louder, make that quieter, pan that hard to the left,’ and so on. So no tiny moves,
only quite broad strokes. And I work in sections. Once a section has radical moves, it usually
dictates what happens in the next section. The potential pitfall of working in the box is that the
changes are very tiny and take a long time. So I prefer to stick with a broad stroke philosophy.

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Secrets Of The Mix Engineers: Declan Gaffney

Declan Gaffney’s go-to processing for

That seems to work. And one of the advantages of mixing in the box is that you can go back to Adam Clayton’s bass involves
where you were at the push of a button.” Digidesign’s Digirack EQ, Cranesong’s
‘Get On Your Boots’ Phoenix Dark Essence (both top) and
the Massey L2007 limiter (below). The
Written by U2/Bono
Sansamp plug-in was used to dirty up
Produced by Brian Eno and Daniel Lanois. Additional production: Declan Gaffney the sound for four bars.
Declan Gaffney: “The mix of ‘Boots’ evolved over time. The original impulse happened one night
in Dublin when Edge and I were laying down the ‘Spirit of punk rock’ guitar, and we changed the
balance of the song, taking out the three tom mics completely. Edge asked me to put down the
mix, which I did on a CD. A couple of months later, just before we were due to go to France, he
called me and asked for that mix of ‘Boots’. I told him that it was done on a Mackie, which doesn’t
have recall, but that I would try to recreate it from listening to the original and from memory. I
spent a whole day in Dublin, using Pro Tools and an Icon and lots of plug-ins, to try to get that
balance and sound again. I remember thinking ‘This sounds OK, but I think I can make it better,’
and so I saved the mix as I had it, in case Edge wouldn’t like where I took it, and did some more
things to it. I sent that mix to Edge, and he liked it, and we tweaked some sections, and he
overdubbed another acoustic guitar, which you can hear in the chorus.
“After that, Bono put down his new vocal performances in France. We didn’t work on the song at
Platinum, though I would sometimes open up the file and listen to how it sounded in that room,
and tweak it accordingly. When we were at Olympic, we changed and added some sections, and
put the final hand on the mix. In May 2008 Steve Lillywhite had worked on another version of the
song, which was more rock & roll, and really good, but it wasn’t what the band was looking for.
For the finally released version, the ‘let me in the sound’ section was changed; the band wanted
me to make it a bit crazier, and one night when I was working on the song, when I thought
everyone had gone home, Bono happed to walk by and came in, and together we realised that
that section didn’t need fairy dust, but just to get the snare and kick f**king loud. We did some
work on the outro as well, and finished the mix. But the end result is 90 percent the mix that I did
on my own in Dublin. By the way, the work on the song in Olympia was done in the box. When we
came to Platinum we had these huge analogue SSL desks and we decided to run some of the in-
the-box mixes via them, and compared that to the in-the-box versions, and we decided to stay in
the box where we could.
“There are loads of tracks in the ‘Boots’ [Pro Tools] Sessions — a lot more than you can see,
because of the amount of ideas that people had during the recording process. The end result is
essentially what happened live, but they are always trying new ideas. Working with U2 is an
evolutionary process of constant writing, recording, and mixing all at the same time, and as soon
as a decision is made not to use a track, it’s pulled down to the bottom of the Session and hidden,
so it can’t play accidentally. But it will be labelled and have comments, so we can always revert
back to it. We’ll have track culls now and then, when the Sessions become too big and unwieldy.
We’ll do a ‘save’ marked as pre-clean-out, and then clear out the tracks and carry on. If you saw the Pro Tools Session folder for each
song, you’d see hundreds of different versions of that Session, so the band can revert to a favourite balance or mix at any time.”
“If you were to look at the ‘Boots’ Mix window picture with the bass plug-ins, you’d see the Eno
loop on the left, and all the main drum stuff to the left of that. To the right are two overheads and
a bongo part. ‘Taurus’ is a Moog Taurus bass pedal — Danny is always into getting the low end
right, and the pedal is doubling Adam’s live bass in the verses. ‘Edge Floor A’ and ‘Edge Floor B’
are the two mics with which his live take was recorded — ‘floor’ is a U2 term for a live take. You
can also see the ‘Nu Riff’ track marked as ‘spirit of punk rock’. ‘Backwards loops’ is Dan’s
backwards treatment of an Edge guitar solo, which you can hear in the outro. ‘Prime Time’ is Dan
creating textures and atmospherics with delays and things. The Sessions are as organised as they
can be, so you can understand what’s going on and quickly know where you are. I arrange and
clean up the Sessions when the band isn’t there.”
The Edge’s guitar sound is usually
Bass: Digidesign Digirack EQ, Cranesong Phoenix Dark Essence, Massey L2007, Bomb Factory shaped through pedals rather than at
Sansamp the mixing stage. An exception was the
‘James Bond guitar’ part, which was
“Adam’s bass is really cool, low-sounding, with no top in it. It’s the sound he gets. It’s pretty much
radically thinned with EQ.
the lowest bass you’ll ever hear. I have a Digirack seven-band EQ on it during the whole track,
notching out 108.3Hz and 193.3Hz, and cutting above 6kHz. It’s a weird EQ curve, but that’s how
it ended up. I don’t pay attention to what the curve looks like. I also had a Dark Essence on a
Sapphire setting. It appears to add a little bottom and top and makes it sound a bit sweeter, like a
compressor that isn’t working. It’s one of those plug-ins that can just improve the sound.”
“Using the Massey L2007 Mastering Limiter on bass is a trick I learned from Rick Rainey. It’s

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Secrets Of The Mix Engineers: Declan Gaffney

really good at getting an even bass sound. It flattens the bass, but it doesn’t make it sound
compressed. It just allows you to set the level, and bang, that’s it. Finally, the Sansamp is
automated to come in just for four bars immediately after the first chorus, where Edge plays an
overdub called ‘James Bond Guitar’. Many other instruments drop out at that point, leaving the
bass kind of exposed. The natural bass sound is so low that it didn’t really have any mids, so to
make it cut through more, I dirtied it up with the Sansamp, to give it more guitar-like frequencies
and to make sure that the James Bond guitar can sound quite small, which was the idea. The
Sansamp plug-in comes in again during the ‘let me in the sound’ section, where it’s just bass, drums, vocals and a textural guitar solo, and it
again fills the hole in that section.”
Guitars: Digidesign Digirack EQ, Massey CT4
“As I said earlier, Edge tends to find his own sound, so normally you don’t do much to his guitar tracks, other than EQ them a bit and
balance them. The ‘James Bond’ overdub is one exception. It was done a few months after the main tracks were laid down, and it happens
only once, immediately after the first chorus. Towards the end of the mix, Edge said to me ‘It needs to sound small,’ so I EQ’ed pretty much
out of it with the Digirack. Instead of this really great big guitar sound, which is what he usually goes for, it was this cool, small sound that’s
not in your face. The EQ does look quite extreme, doesn’t it? There’s also a Massey CT4 compressor on the James Bond guitar, which is
just touching it, to even things out a little bit for consistency.”
Lead vocal: Waves VEQ3 & Renaissance Vox, Bomb Factory Fairchild 660, Digidesign Trim, Digirack EQ & De-esser, Sound Toys Echo
Boy
“I used the Waves VEQ3 to get rid of some of the thickness in the voice that came from using the
58/Neve/LA2A signal path [see ‘Tracking ‘Get On Your Boots’’ box]. I’m cutting at 700Hz, just to
sweeten it a little. I like the VEQ3, because it’s great for broad brushstrokes. I’ll use the Digirack
for precise surgery; you can take specific frequencies out very quickly with it, and it won’t change
the sound too much. It’s a ‘go-to’ plug-in for me. The Fairchild 660 plug-in is doing quite a lot of
Most of Bono’s vocals were recorded
heavy compression. You can see that the input gain is up quite high, and that means the signal is through a Shure Beta 58 dynamic mic,
hitting the compressor hard. Bono’s voice sounds really good with this plug-in, or with the Bomb with some EQ (including the Waves
Factory 1176. If the one isn’t working, I’ll try the other. His voice also likes the SSL compressor, VEQ3, top) and heavy compression
and I use the RVox here to soften things a little bit, it’s not doing very much. The Digirack Trim is from the Fairchild 660 plug-in. No
used to turn the output of the Fairchild down, it’s not there for sonic purposes. reverb was used, but Sound Toys’ Echo
Boy provided delay.
“There’s also a seven-band Digirack EQ that notches out around 245Hz and adds some top
end above 10k for some sparkle. The Digirack De-esser and Echo Boy are on a separate track,
because I wanted to hit the De-esser first, so that any kind of sibilance didn’t hit the echo. The
echo is just a kind of warm ping-pong sound. There’s no reverb on the track, there’s not even any
feedback on the delay, it’s all about the dry sound with a little bit of space on the side, provided by
the delay. In fact, there’s no echo in the whole song. I recall Dan making a joke about how we
were trying not to use any echo on this record. It didn’t require it. I’m not a huge fan of reverb
anyway; it’s better to do the same thing with delays.”
Mix bus: SSL compressor, Massey L2007, Cranesong Phoenix Dark Essence, Waves VEQ4
“I always put on some mastering plug-ins, and we then listen and decide whether they work or
not. Edge preferred these plug-ins, so they remained.
“I printed the mix through an Aux track. The stereo mix hits the SSL compressor first, on a
setting that lets all the transients go through, and yet makes it sound very punchy. It’s hitting the
Massey L2007 mastering limiter after that, and the way the two act together works really well.
While the SSL takes care of making the sound snappier, the L2007 takes off some peaks. The
Massey has a very fast attack, which helps the vibe. I also tried the L1 limiter, but didn’t like it, and
switched it off. Cranesong do this box called the HEDD192, and the Dark Essence plug-in is more
or less a plug-in version of that. It simply makes the mix sound better. The Waves VEQ4 is a
Neve 1081 recreation, and I add a little at 100Hz, 270Hz and at 15K, again to improve the sound
and for some sweetness. It’s not surgery. The Neve is just there to make the whole track smile.”

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Secrets Of The Mix Engineers: Declan Gaffney

Declan Gaffney incorporated extensive

master processing into his mix. Dynamic
control was provided by the Waves SSL
Master Bus Compressor and Massey
L2007 limiter, while Waves’ VEQ4
added tonal changes.

Tracking ‘Get On Your Boots’

Declan Gaffney: “Most of ‘Get On Your Boots’ was recorded in Dublin by Richard Rainey, and the basic backing tracks were done live,
by the whole band together. There was an [Electro-Voice] RE20 inside the bass drum, with an SE Electronics Titan on the outside, a
[Shure] 57 underneath the snare, and Richard had his own Heil mic on top, which he alternated with a Beyer M201; the toms were
[Sennheiser] 421, overheads Coles 4038; ride cymbal was sometimes a 57, sometimes a [AKG] 451. Everything went through the Neve
1091 or 1093 mic pres. The microphone on the bass cabinet was a Shure SM7 going into a Neve preamp into an LA2A; the DI wasn’t
used. The SM7 was the only bass mic that was used on the whole record, it’s great for bass guitar!
“Edge’s thick guitar sound is entirely from the live band session, recorded with two Royer 121s, one on his Fender Deluxe and the

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Secrets Of The Mix Engineers: Declan Gaffney

other on his AC30, and the mics went through the Neve and then an LA2A, though it’s not doing anything, it was just there for the sound.
When recording Edge’s cabinets, it’s almost always a 121, or a Sennheiser 409, occasionally a 57. I record completely flat, because
Edge will have found a great guitar sound, and you just record it.
“When we were in France, we got this great vocal sound that Bono really liked, which was a [Shure] Beta 58, going through a 1091 and
then an LA2A, into Pro Tools. I even A/B’ed the different 58s and Neves, and found my favourite LA2A, to get the best ones. I’m very
proud of the vocal sound. I added a bit of compression while he was singing, and he got excited by that and adjusted his voice
accordingly. When we were at Olympic, the vocal chain changed a little. I normally have two or three 58s up in a room, and at Olympic
one of them would go through a Neve preamp and the LA2A, but the other would be Neve and then Distressor, and I actually preferred
that sound. The LA2A sounded a little too thick. The Distressor had a sort of hardness that balanced the thickness out better. Edge’s
vocals were also recorded with one of the 58s.”

Daniel Lanois: Producing No Line On The Horizon

Producer and artist Daniel Lanois’ association with U2 goes back to 1984, when Eno and he helped reinvent the band’s sound on The
Unforgettable Fire. Since then, Lanois and Eno have worked on classic U2 albums like The Joshua Tree (1987), Achtung Baby (1991)
and All That You Can’t Leave Behind (2000). For No Line On The Horizon, one of his contributions was to encourage the band to work
in Fez, Morocco, in May and June 2007.
“We went there because we wanted to be at a spiritual crossroads and we felt that Fez had that to offer, musically. We thought of it as
a Mecca of sorts. We wanted to be in a geographical location that was filled with plenty of music, and that didn’t have the usual fangs of
the music business and expectations. Was it to do with going to a Muslim country? We did not speak about that. We spoke about mutual
ground. I think that there’s something ancient in that location that gets in your bones. And I think that it’s a good inspiration for music to
draw upon historical sources. Yes, you can listen to music from the ’80s, the ’70s, and maybe the ’60s, and think that some kind of
mimicry will serve you. But why not look back at a thousand years and see what comes your way?
“We hired a riad, which is a large building with an open courtyard, rolled in an 18-wheeler truck full of U2’s Dublin gear, and set up in
the courtyard. The truck had about eight racks of gear for the Edge, one guitar pedal for me, a microphone for Bono, drums for Larry,
and a bunch of recording gear. We recorded to Radar. I highly recommend it, it’s a great machine. The riad had a sort of automatic roof
that could be peeled away, so we could play in the open sky, which is to be highly recommended. The courtyard works like an ancient
ventilation system; the warm air escapes and the walls provide you with shade. The co-writing invitation from the band was very kind and
very sweet. The fact is, we play well together, Eno, U2 and I, and those beginnings in Fez were very productive. Just a few weeks of
recording provided us with many fascinating beginnings, beautiful landscapes, isolated moments, and poetry. It was all there. It was like
a harvest.”
In a recent interview, the Edge explained that his Death By Audio Supersonic Fuzz Gun inspired his approach to the guitar on the new
album. Lanois: “You mean his guitar pedal? I’m sure he’s right about that. But he has so many pedals that I have lost track of them. The
Edge’s corner is like a minefield of pedals and you don’t want to go to his corner for fear of never coming out again. My one pedal?
That’s a little looping box called a Boomerang. There are lots of strange repeating sounds on the record and some of them come from
the Boomerang.”
“The mixing stage took place at Platinum Sound in New York. I forget all the reasons why we went to New York, but I think one of them
was that we could use three rooms at the same time in one building. Eno, two mixing engineers, and I, had a song brewing in each room
and we were able to waltz from one room to another. There’s something nice about having three songs burning at once. We did the
same at Olympic Studios in London, where we did the final mixing sessions, during which Steve Lillywhite helped out with the production
and mixing. I think that things worked out really well and that the album contains some of the band’s best work.”

Published in SOS June 2009

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Sounding Off

Sounding Off
Mike Senior
Published in SOS June 2009
People + Opinion : Sounding Off
Printer-friendly version
The end of the iPod era?
Mike Senior

I love iPods, and by the same token their frequently less visually appealing, and un-fruit-branded, MP3-playing cousins. I think it’s
brilliant that they’ve allowed musicians to drop musical examples easily into conversation and to share influences at the swap of an
earbud. It’s also great that music download sites make it so easy to explore the breadth of music history, whether fashionable or not,
and also provide retail exposure for less mainstream talent.
So why don’t I own an iPod, then? It’s certainly not the audio quality that holds me back — the sound’s fine for general browsing and
listening purposes, much as cassette used to be. The real reason is that I think the iPod, revolutionary as it has been, is actually just a
commercial stop-gap, because I don’t think most people actually care about owning music any more. They just want to listen to what they
want, when they want. In short, they value access, not ownership.
Now I actually own loads of CDs (remember them?), because I am one of the minority willing to pay for higher-quality audio. But I
acknowledge the fact that I’m rapidly becoming a dinosaur in this regard, missing out on the advantages currently enjoyed by those whose
digital lifestyles are already fully hubbed. Nonetheless, because the iPod still requires you to own the tracks you listen to, I still don’t want to
get involved, because of the pain of dealing with any music files I accumulate. Storing them. Deciding which ones to put on which size of
iPod. Migrating them to updated hardware. Trying to remember to back them up. Hoovering up splinters of the iMac that I’ve just kicked to
death because it’s corrupted the disk containing the library that I forgot to back up. You know the sort of thing.
What’s most recently woken me up to the demise of the ownership model is that I noticed that although I really like watching films, I own
no DVD collection. I’m perfectly happy to pay a monthly subscription to a web-based library that allows me to access anything I might ever
want to watch without having to buy, organise, house, insure or back up my own set of DVDs. And how much better if this service could
stream roughly-TV-quality films to a portable handset to play on demand when and where I fancied? Although bandwidth restrictions put the
kibosh on this daydream for films at the moment, I understand from friends of mine in the mobile phone industry that it’s already a much
more viable prospect for MP3-quality audio.
Humour me while I imagine a time when you might pay a subscription to receive a certain number of MP3 track ‘plays’ (or time-limited
downloads) selected from an iTunes-style database, via your existing mobile phone — the fee might even be bundled as a sweetener with
your phone contract, like free call minutes. I think that any portable music library, iTunes or otherwise, I’d actually built up before that point
would seem pretty much obsolete. (And after all that backing up, too...)
So I’m holding my breath for the subscription model of music consumption to come of age. And I have a feeling it may bring other benefits
with it. For example, I think it might hold the key to dealing with the music industry’s copyright theft problems. As Paul Sellars argued back in
SOS March 2009, DRM technology can currently make you feel as if actually owning music is less convenient than stealing it online. But if
your music access were near instant, offered practically unlimited choice, and were bundled as part of your mobile phone package, I think
that web-based file-sharing would seem a hell of a lot less convenient by comparison. And wouldn’t it be a lot easier to prosecute illegal
music-streaming services or phone-phreakers than to rely on largely ineffectual legal sabre-rattling against thousands upon thousands of
peer-to-peer MP3 rustlers? Finally, I think that ‘per-play’ royalty streams from establishment on-demand services might enable more
widespread support for niche music from the mass-market. My Internet DVD subscription certainly made me more experimental in my
choice of films, because I knew that it made no difference to the fee I’d already paid.
So, my slogan: Death to the iPod! It’s done great things, but the sooner we’re shot of it, the better.

About The Author

SOS contributor Mike Senior thinks that the access model would work for all sorts of other lifestyle areas. His wife has yet to be
convinced.

Published in SOS June 2009

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Within Temptation: Producing Black Symphony

Within Temptation: Producing Black Symphony

Ronald Prent, Darcy Proper & Wouter Strobbe: Blu-Ray Audio Buy PDF
Published in SOS June 2009
People + Opinion : Artists/Engineers/Producers/Programmers
Printer-friendly version
Few artists so far have taken advantage of the Blu-Ray format’s potential to deliver stunning
audio quality. A concert film by Dutch metal act Within Temptation shows what’s possible.
Sam Inglis

D espite being largely ignored by the mainstream music media, symphonic metal is a genre
that can boast legions of very loyal fans. Thanks to its fondness for spectacle and
ambition, it also pushes technology to the limit, both in live performance and recording.
These qualities are very much in evidence in Within Temptation’s Black Symphony, a live
recording of a special one-off performance at a sold-out Ahoy Arena in Rotterdam, with some
10,000 fans in attendance. For the occasion, the band worked up an elaborate stage show, and
for the first time performed live with the Metropole Orchestra. The resulting concert film was then
released not only as a DVD, but also in the high-definition Blu-Ray format, with unprecedented
audio quality: viewers whose equipment permits can listen in uncompressed 24-bit, 96kHz 5.1
surround. Wouter Strobbe (left), Darcy Proper and
Ronald Prent (right) at the API Vision
console in Galaxy Studios.
Meet The Team
The Black Symphony project was handled by a team working from Galaxy Studios in Belgium, one of relatively few studios that could offer
the necessary facilities. Overseeing the recording and mixing was resident engineer Ronald Prent; the audio was also mastered at the
same studio by Darcy Proper, while another Galaxy engineer, Wouter Strobbe, was responsible for managing the project and authoring the
master discs.
“We used Peter Brandt’s remote recording truck, which is fully analogue from the stage all the way to the truck,” relates Ronald Prent. “On
stage we had Neumann mic preamps and splitters, and from there it went direct into Pro Tools at 96kHz/24-bit, and we had 140 tracks. You
can get 96 tracks onto one Pro Tools at 96/24, so we had two Pro Tools setups, and four Tascam digital recorders as backup. It was the
best-quality recording we could make, with a full analogue front end, not splitting off the Digicos [front of house mixers] but splitting at the
microphone.”
As Prent explains, much thought also went into capturing the response of the crowd and the ambience of the arena. “We had an SPL
Atmos spider microphone at the front of house [position], then we had seven Schoeps mics in the roof as a 7.1 — because there was talk of
doing 7.1 — and we experimented with one of Peter Brandt’s ideas, which was Schoeps boundary microphones stuck on a big Plexiglass
plate positioned behind the stage. Those were actually the best for audience response. Then we had two B&K 4006s all the way in the rear
for slap and depth, and shotguns on the front — two on the left-hand side, two in the middle and two in the right. I used all of them. We
wanted the audience to be as loud as possible.”
Even though there were 140 signals to contend with, the demands of the staging and filming meant that visible microphones, stands and
cables were a no-no, so wireless units and lavalier or clip-on mics were the order of the day. “Everything was black, and when it wasn’t
black, it was blacked out with chalk!” laughs Prent. “We tried to get rid of as many microphones as we could without compromising the
recording. The amps were on the sides of the stage left and right, facing outwards so they didn’t have any crosstalk, miked with an SM57
and a Royer — that’s what they had for the PA and that suited me perfectly. The drummer likes Audix and I do too. They have these clip-
on sets you almost don’t see.”
Over the years, some orchestras have attracted the reputation of being Luddite. By contrast, the Metropole and its musicians are
thoroughly at ease with technology, to the extent of having their own studio and preferred choices of microphones. “Initially I suggested
some alternatives,” admits Prent, “but the orchestra was more comfortable with their own ‘tried and true’ approach, which, in the end,
worked just fine for me.”

Repair Shop
With the concert audio and video safely in the can, the next stage was what Ronald Prent
describes as ‘repair’. “For the orchestra, the only problem was the strings, because the huge
temperature differences on the stage with flamethrowers and big lights meant that after five
minutes, everybody was detuned. It wasn’t too bad during the original performance, but for the
record we re-recorded the strings. It’s something you have to do. The orchestra have their own
studio in Hilversum where they rehearse, and they also record there. So in one afternoon they
replayed the concert top to bottom and re-recorded only the strings. That was done with normal
miking for the orchestra. It’s pretty ambient where they record, so some ambient mics were used
so that we could blend it properly. That worked pretty well.”
Within Temptation and the Metropole
This done, it was left to the tech-savvy band and orchestra to tidy up their own performances
Orchestra rock the Ahoy Arena.

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Within Temptation: Producing Black Symphony

where necessary. “We split up the audio onto two hard drives,” says Prent. “One went to the
Photo: Laura Oldenbroek
band, and they went through their own performance and repaired or edited whatever they wanted
to, and then it came back to me. Then I just took my original recording and replaced what I had
gotten from them — a guitar lick here, bass note there, the usual. Once I had that, it became one Session that was the full length of the
concert; band only, including audience. From the orchestra I got back another Session with all their repairs in it, and I would put only the
repairs back into my second Session; the complete show, orchestral tracks only. I spent a couple of days — I think about four — preparing
the two Sessions in two different Pro Tools rigs and sync’ing them up. They were both running at 96kHz/24-bit, bringing it back to 128
outputs, 130 outputs sometimes.”

Managing The Mix

The mix itself was done on Galaxy’s custom-built API Vision analogue console, which was
designed to Ronald Prent’s specifications. “The API console allows you to do a true stereo and
surround mix at the same time. You have separate buses. The level automation remains the
same, but the pan-pot is designed in such a way that you can deal with it in stereo and surround
simultaneously, and the bussing of the console is the same way. It’s a discrete panner: left-centre-
right, and the next one is front-to-back, and then left/right back. The trick in there is that from the
front panner, the signal goes to two wipers. One feeds the stereo bus, the other feeds the
surround bus. So if you pan something to the rear, you still have your front pan-pot available on
the stereo bus. And because it’s a discrete panner it doesn’t have level compensation anywhere in
the image, so you can pan it anywhere without changing the balance.”
So how does one begin to approach such a vast mixing project? “I divided [the mix] up into band, audience and orchestra,” explains
Prent. “The API console is an all-discrete analogue console, so everything comes out of Pro Tools and becomes analogue and goes into
the console over 128 to 130 channels. You have your drums, bass, guitars, keyboards, audience, vocals groups, the strings, woodwinds,
and the percussion — just the percussion was 16 channels — and then you have the orchestral score. So I got the band going, got the
orchestra going, and then got a blend and a balance. When it got to the actual mixing, first I spent two days with the music director of the
orchestra, and we went with the score through the whole orchestra, and mixed each song for the orchestra, leaving the band where it was.
When that was done and the orchestra was happy, then the band came and we did the same with the band: they heard what the orchestra
was supposed to sound like, and I made adjustments to the band, and sometimes minor adjustments to the orchestra. And slowly it came
together. Then there were a couple of days where I went through it and did the fine details and watched the picture.
“The interesting thing about this project was that at the same time I was doing audio, the guys were doing picture editing upstairs, so
there could be a synergy between us — like they would say ‘We don’t have a shot for that, or we have a shot here, can you do something?’
That gives a different dynamic from just mixing audio to a rough cut. The film guys liked that I was ahead of them, so they got a rough mix
from me from whatever stage I was in, and then of course when the audience was especially exciting or the music was particularly dramatic
I would change balance, and they would accommodate that in the picture.”
The fact that, strings excepted, the band and orchestra had not had to re-play large sections of the performance made it possible to lean
heavily on the many ambient mics, rather than using artificial reverb. “I like to leave the audience open as loud as I can so it’s part of the
sound — which is only difficult when they start replacing notes that they never played live!” laughs Prent. “But in this case, they played very
well, so I could leave it loud. The hard work was getting the audience so loud that you actually have a feeling that you’re in the concert, but
still have the impact of a band and an orchestra. And there’s no audience sweetening, as they call it, on this concert. It’s the original
audience. We now have the advantage of workstations where we can calculate delay times, so when we record we can measure them, and
later when I mix, when there are some microphones that have too much of a delay, I can move them forward a little bit. That way I keep the
big space, but I don’t have all the reflections. It’s not exactly true to the original but it gives you the environment, instead of using an artificial
sound.”
A live recording such as this poses particular challenges when it comes to the lead vocal, but again, says Prent, “Nothing was completely
re-sung. Here and there a little was added, and some words are Auto-Tuned. We did some de-essing and a lot of levelling, but most of the
dynamic control is done by hand. Especially with live, the more you compress, the more shit you get from crosstalk, so a lot of the vocals
were done with fader rides or Pro Tools automation, whichever worked better.”

Mastering
Once complete, the 5.1 and stereo mixes migrated across the Galaxy building to the studio of
resident mastering engineer Darcy Proper. “Mastering, on a production like this, is probably the
simplest part of the process,” she says. “In general, if I get good mixes, then the mastering job is
easier — there’s less to be corrected and it’s simply polishing. In this production that was the
case, what I got was great-sounding stuff. In spite of that, I did the surround version twice. I did it
once, came back the next day and decided that in my enthusiasm for making it loud and powerful,
I’d pushed it too hard and wasn’t happy with it. I said ‘OK, that was a practice run, that was a day
for free,’ and then did it again and brought the level back a bit to allow it not to hit that brick wall at
the top, for keeping the dynamics.
A diagram showing the positions of the

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Within Temptation: Producing Black Symphony

“From a mastering standpoint, the biggest challenge with this project was to help it become various ambient mics used in the Ahoy
super-powerful, the kind of experience that makes your heart pound. To do that you need to keep Arena. Mics 1 to 7 were Schoeps MK2s
the dynamic, but you don’t necessarily want to have something start off with a whimper just to and 8 to 12 the five capsules of the
allow it to get bigger later on. So for the intro where it starts with the orchestra and the choir, the SPL Atmos 5.1 mic; 14 to 17 were
idea was to allow them to have this great size and impressive sound that Ronald had created with Sennheiser 418 stereo shotgun mics,
while 13 and 17 were Schoeps PLM2
the mix, but still leave room for the band to jump in on top of that and add that extra energy that
PZMs mounted on a Perspex sheet.
really ‘puts it over the top.’ But I would say technically it was all pretty easy. I took Ronald’s files,
96/24 from Pyramix, I put those into Pro Tools used with external converters — Pro Tools was just
a glorified playback machine for me — that then allowed me to record into my Pyramix. Rather than going out of Pyramix, through my
processing, and looping back in, I prefer to use one workstation for playback and the other for recording my mastered files. So from Pro
Tools to analogue, I worked through my mastering chain in analogue and then back to digital, running picture along with it for visual
reference.”
The chain in question consisted of “Basically EQ, compressing, limiting and some loudness maximising. For this, the compression is not
doing a whole lot. It’s long attack times, low ratios, just a level and feel thing, it’s not doing a lot of pumping. The thing to watch out for in
surround is that while you’re still looking for compression to do something desirable to the sound, you have to keep in mind that it’s not
necessarily all going to work the same way all the way around [the sound field]. You have to watch what you’re doing in the front, and make
sure that what’s happening in the back then doesn’t cause the image to teeter-totter or something strange to happen.
“The nice thing with surround, with regard to compression, is that you generally don’t need so much of it, because everything has its own
space to breathe. You don’t have to squash it all down into two speakers, so you can have all kinds of energy remaining. All you’re looking
for out of compression is to create a sort of tight and stable soundfield. You’re not necessarily bound to try to control the dynamics the
same way you do when you’re working in stereo, because you’re not carrying so much information in each channel. It’s part of what makes
surround more exciting — not just that you have the sound all around you but that you can leave a bit more life in all of your sound
sources.”

Balancing Bit Budgets

With the audio and video complete, it fell to Wouter Strobbe to assemble the Blu-Ray and DVD
discs themselves — a job complicated by the team’s desire to offer uncompressed 24-bit 96kHz
audio in both stereo and surround formats on the Blu-Ray version. “As far as we know, it’s the
first and only Blu-Ray disc that uses as much bandwidth for audio as for picture,” says Ronald
Prent.
“It’s 20.5 megabits per second for audio and 20 for picture,” concurs Strobbe. “The difficulty in
the consumer area at the moment is that if people buy a Blu-Ray player but they still connect their
old receiver, they’re not able to decode the high-definition audio. That’s the reason why we put
Dolby Digital 5.1 on it, the same format as is used in DVD-Video. We wanted to be sure that
everybody could decode the surround stream. The old receivers are mostly capable of decoding Galaxy Studios’ diagram shows the
complex digital clocking arrangement
DTS, so we put DTS 24/96 on it at 1.5Mbps, and then for the next-generation environment,
used during the mix of Black Symphony,
receivers which are capable of HDMI, we have uncompressed 24/96 PCM.”
to achieve sync between two Pro Tools
The challenge of squeezing uncompressed audio onto the disc created some difficulty in rigs, the V-Cube video playback system,
managing the video stream, which had to be high enough in quality to satisfy band and director. and the Pyramix system used as a
“You have three main video codecs for Blu-Ray. You have MPEG2, you have Windows Media master recorder.
and you have AVC. MPEG2 is based on a standard which was invented 15 years ago, so it’s old
technology. Windows Media is closer to the HD-DVD format, but if you look at 90 percent of all the Blu-Ray discs on the market, it’s all
video encoded in AVC, which is an MPEG4 variant. This is an advanced codec and the algorithm is very complex but the efficiency is very
high. With this project, I was really amazed that I could get the picture done with 20 Megabits per second. The picture was shot at 25 frames
for Europe and for broadcasting. We knew that this Blu-Ray would also be released in the US, so we had to bring the picture to the 60Hz
[ie. 30 frames] standard, because 60Hz is compatible with every player in the world. If you play a 50Hz Blu-Ray in the US you have a
problem. So we had to convert the picture to 60Hz, and that meant having to calculate five more pictures in a second, which of course
affects the encodes for Blu-Ray as well. You have more movement in the picture so the encoding parameters had to be adjusted
accordingly, which was really a challenge.”
Since its release last year, the Black Symphony DVD has been a conspicuous commercial success, reaching the top five in no fewer than
nine European countries, including the UK. As a showcase for the relatively new Blu-Ray format, meanwhile, Black Symphony has few
equals. In these days where audiophiles bemoan the trend towards ever louder CDs and lossy MP3 compression, is it too much to hope
that Blu-Ray can succeed where Super Audio CD and DVD-Audio failed, creating a market for truly high-fidelity music recordings?

The Rise Of The Authoring Engineer

The last few years have been depressing for anyone contemplating employment in the studio
engineering line. However, as consumer formats become ever more powerful and complex, it’s
possible we could see a new specialism: that of ‘authoring engineer’, one of Wouter Strobbe’s

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Within Temptation: Producing Black Symphony

roles in the production of the Black Symphony disc. Creating a master disc for CD duplication is
relatively straightforward, but the same is not true of DVD and even less so of Blu-Ray. Not only
must the audio and video be correctly encoded in multiple formats — an increasingly demanding
task — but there’s also the need to create menus and other interactive elements.
“I wish that people in audio and video could appreciate how important the authoring process is
to what the final result is,” says Darcy Proper. “Unlike CD production, mastering isn’t the end of The Blu-Ray authoring process requires
the audio chain in DVD and Blu-Ray. And good encoding isn’t as simple as just handing the stuff considerable expertise and specialist
to a guy who shoves it through the encoder using the same settings he uses for everything else. tools: a new career option in the music
There’s a lot of management that needs to happen and it makes a huge difference in end business?
quality. The authoring engineer understands the details of managing bit budgets and can give
you an idea of the bandwidth you’re looking at based on what you want to have for audio and what you want to have for picture. He may
have some advice on which format to use for shooting picture, for example, or which audio streams you should consider including —
ideas that are helpful to have from the very beginning of the project in order to get the result you want at the end. I suppose it’s possible
that an authoring engineer is a complete audio maniac and decides to squash the picture down to practically nothing — but because
most authoring people come to it from the picture side, that doesn’t typically happen and it’s more often the audio that suffers. For that
reason, those of us in audio should pay close attention to the authoring process.”
One of the most important aspects of authoring is understanding audio and video encoding processes. “Encoders are used widely in IT
infrastructures but also satellite uplinking, broadcast transmission, DVD production workflows and Blu-Ray,” explains Strobbe. “What I
see often is that an encoder is treated as though you just put something in and something else comes out by itself — but in an encoder
you have to tweak a lot of parameters to get the best output. If you put uncompressed audio in and you need to get compressed audio
out, there are a lot of decisions to be made in the box. In Dolby compression schemes, for example, you’re talking about dialogue
normalisation, or dynamic range compression, or the LFE handling. I have noticed a lot of discs in the market which are encoded using
only standard settings and, frankly, a lot of them could be much better, particularly for music-focused content.”
The Blu-Ray specification is, says Strobbe, much wider than that of DVD-Video. “There are more possibilities for interactivity, in
compression codecs for audio and video — a lot more possibilities and choices to make in the authoring process. The Blu-Ray
specification has two platforms. One platform is standard authoring and the other is advanced authoring, which includes even more
interactivity. Advanced interactivity means that you can combine Internet applications with audio and video content — you can link them
together — like adding interactive gaming to packaged media, for instance.
“The Within Temptation disc was built in HDMV, which is standard authoring. There are two tool sets for authoring HDMV — Sonic
Scenarist and Sony Blu-print. These are the spec-compliant tools that allow you to author to the full Blu-Ray specification. There are
also some other packages for authoring Blu-Ray, like Adobe Encore, but they don’t allow you to use the complete range of high-
definition audio or video codecs.”

Galactic Ambitions
The small town of Mol, in rural Belgium, is not perhaps where you would expect to find a world-class studio complex. Yet when brothers
Guy and Wilfried Van Baelen outgrew the original studio they had built in their parents’ barn, they decided it was as good a place as any
to locate the replacement. Their quest to build the quietest studio in the world led them to employ some radical construction techniques
— most notable of which are the enormous springs on which the entire building rests! The design is said to offer more than 100dB of
sound insulation, even in the main live area, which can accommodate a full orchestra.

Published in SOS June 2009

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Apple Notes

Apple Notes
Educational Training in GarageBand Buy PDF
Published in SOS June 2009
Technique : Apple Notes
Printer-friendly version
With the latest version of GarageBand, Apple have become the first company to integrate
educational training into music-creation software. But is this merely a simple gimmick, or a
powerful new way of learning to play an instrument?
Mark Wherry

S ince its introduction in 2004, GarageBand has become the entry-level music-creation
software de rigueur for beginners and seasoned professionals alike. With GarageBand,
Apple succeeded where so many other music software developers had previously failed,
creating an application with limited functionality that wasn’t just a cut-down version of something
more expensive. By designing a simple user interface, Apple encouraged beginners to make
music, but also caught the attention of professionals, many whom continue to rave about
GarageBand in the press to this day.
Each new release of GarageBand over the last five years has brought additional features to the
application, broadening its scope of use, but retaining the simplicity that appealed to users in the With GarageBand ‘09’s new Artist
first place. For example, version 2 added a notation display, while version 3 made it easy to create Lessons feature, you can learn to play a
podcasts or add soundtracks to movies made with iMovie. GarageBand ’08 finally implemented song by a popular artist, such as ‘Brick’
by Ben Folds, complete with video,
the ability to add tempo changes to a song, and also introduced a feature called Magic
synchronised notation, and an
GarageBand, providing backing tracks that could be used as the starting point for a song. instrument display that even shows the
Garageband has always catered for two different types of users: those with musical experience, appropriate fingering.
who can play and record music into the program, and those without, who can create a song by
employing various loops and other pre-made musical building blocks. But with the latest version of GarageBand, introduced as part of
Apple’s new iLife ‘09 bundle earlier this year, Apple is not only making it easier for beginners to make music, but also providing a teaching
platform to help users learn and develop musical skills.

Look & Learn

There have been many attempts over the years at creating music software that helps a user learn
to play an instrument. Who remembers the Miracle Piano Teaching System, for example,
developed in the early ‘90s? But rather than offering music teaching in a separate application,
Apple have built music lessons straight into GarageBand; and this is the first time, to my
knowledge, that such educational functionality has been integrated into music-creation software.
Apple supply 18 Basic Lessons with GarageBand ‘09, nine of which are designed to help you
learn the basics of playing the piano, while the remaining nine are for those wishing to study the
guitar. One of each of these lesson types is installed with iLife, and the remaining 16 can be
GarageBand ‘09’s new features include
downloaded free from the new Lesson Store within GarageBand.
a more refined appearance that makes
A lesson is presented in a new full-screen view, where you can watch a video of the teacher at the application clearer for beginners,
the top of the screen, while seeing a live, animated representation of the teacher’s keyboard or while nodding in the direction of the
fretboard at the bottom. One particularly nice touch is that you can also choose to see professional.
synchronised notation alongside (or instead of) the view of the instrument; and, in the case of the
guitar lessons, you can opt to display tablature alongside the notation, just the notation, or chord symbols.
Each lesson offers both instruction and a piece of music, which is introduced at the end of each lesson as an exemplar of the techniques
that have been discussed, performed with a suitable instrumental accompaniment, which the student can play along with. The piece of
music chosen at the end of Piano Lesson 8, for example, is the ‘Minuet in G’ from Bach’s Notebook For Anna Magdalena, and this is
provided with a suitably baroque-sounding string accompaniment.
As with a GarageBand Song, a lesson also features a timeline, complete with transport controls. And, like the regular GarageBand
timeline, the lesson timeline is completely interactive, making it easy to step back if you need to hear something again. To help you navigate
a lesson, the timeline is divided into different sections that represent the different parts of a lesson, and these function in exactly the same
way as sections one might create on GarageBand’s arrange track.
One very nifty feature available in the lesson’s transport controls is a speed slider, enabling a student to slow down the playback of the
lesson in order to better understand what the teacher is doing, or help them practise a particularly tricky bit. Fortunately, GarageBand mutes
the teacher’s voice if you’re not playing the lesson back at a normal speed, sadly avoiding any amusement that may have ensued; and
although the quality of the time-stretching is adequate for the purpose, it obviously gets a little grainy once you get to the lowest, half-speed
setting.
In addition to the Basic Lessons, GarageBand ‘09 also offers Artist Lessons, where a popular artist will teach you to play one of their

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Apple Notes

songs. So far, there are 10 of these lessons available, at least in the US, and the list of featured artists includes Ben Folds, Norah Jones
and Sting. Each Artist Lesson costs $4.99 and can be downloaded from the Lesson Store within GarageBand, while payment is handled via
Apple’s on-line store in a web browser window.
If the Artist Lessons prove popular, I’m sure Apple will create more content over time, and it certainly provides another way for the
company to ‘monetise’ additional content for GarageBand in much the same way as they’ve done with Jam Packs. It would be particularly
interesting if Apple allowed third-party content to be sold via the Lesson Store, although I’m not sure I can see this happening, since it would
require a large effort in terms of quality-controlling the results.
It would, however, be good to see Apple partner an educational establishment such as Berklee College of Music in the US, or an
examining body such as the Associated Board in England, as a way to provide further high-quality musical training for all levels of
technique, particularly for the latter’s graded exams syllabus.

Teaching Success
In that overview of GarageBand ‘09’s new lesson features, you might have noticed certain items
sounding rather familiar, such as the transport controls or synchronised notation. Similar to a
regular GarageBand song? Although I don’t have particular knowledge of how GarageBand is
developed, it seems reasonable to assume that, under the wonderfully-presented façade, a lesson
is essentially a GarageBand song. If you think about the lesson functionality in this way, suddenly
it seems obvious to build music lessons into a sequencer in the way Apple have done for
GarageBand, and it’s almost surprising that other companies like Steinberg, Cakewalk, and the
rest, haven’t tried this.
One advantage of integrating lessons into GarageBand is that it makes it relatively easy to
provide the ‘Open in GarageBand’ button that can be clicked on in many of the lessons. This
essentially opens the lesson as a GarageBand song, enabling it to be used as a starting point for
a student’s own musical explorations, adding tracks and loops as they see fit. But there must be
so many other facets of a sequencer that could be exploited for similar educational gain, such as
singing lessons based on Auto-Tune-like algorithms, to mention but the tip of the musical iceberg.
In the past, products for computer-based musical training have enjoyed perhaps limited
success. Going back to a product such as the Miracle system, its profitability was surely not
helped by the cost of producing the bundled piano keyboard hardware at the time, whereas today,
it’s possible for almost anyone to walk into an Apple Store (as, indeed, many have) and buy a
cheap keyboard for less than 100 units of the appropriate tender. With even an entry-level Mac
capable of delivering the high quality of lessons shown in GarageBand ‘09, the relatively low cost
of modern musical hardware, and the unlimited content that could potentially be available on the
Internet, computer-based music lessons might finally have their day.
GarageBand ‘09 requires Mac OS X Leopard (10.5.6) and is included with all new Macs.
Existing users can upgrade to the latest version of Apple’s iLife ‘09 bundle for $79, which also
includes the new versions of iPhoto, iMovie and iWeb.
GarageBand ‘09 offers a new Electric
Guitar track type, providing an on-
screen guitar rig with virtual stomp
boxes.

Repainting The Garage

In addition to music lessons, GarageBand ‘09 has also received a coat of paint, now appearing in that dark-is-professional shade of grey
that’s all the rage at the moment. I have nothing against dark-grey user interfaces, but I do wonder (as I did with Pro Tools 8) where the
fashion of dark-grey scroll bars on an imperceptibly lighter grey background came from.
Not wishing to dwell on the scroll bars for too long, the horizontal scroll bar makes up for its appearance with a very nice touch. When
you scroll the timeline with sufficient velocity, an indicator appears (in much the same way as when you adjust your Mac’s volume, for
example) to display the number of the bar currently on show at the far-left edge of the timeline. This is quite similar to the current page
display that appears when scrolling pages in Microsoft Word, and is one of those nifty-yet-obvious ideas you wish had always been
there in a sequencer.
One positive aesthetic change in GarageBand 09’s dark appearance, though, is that standard Aqua controls are no longer present in
the user interface. This results in a significantly cleaner and more consistent look, especially in the Editor views and the Track Info panel,
where the parameters for a Track are now given more space in a separate tabbed page from the browser.
Another nice touch worthy of comment is the way in which Regions are now represented in the Editor views. While the Region
boundaries in an Editor were pretty clear in previous versions, with appropriate colour-coding, they’re now really clear, featuring a larger
header that displays the name of the Region. But the icing on the cake is that each Region in the Editor now has its own play button,
which automatically puts GarageBand into cycle playback mode for the duration of the appropriate Region, and plays back only that

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Apple Notes

Region. Pretty neat. Like so many GarageBand touches, you just wish more of them would show up in Logic!

The Third Type

As well as the Software and Real Instrument track types offered by GarageBand for recording MIDI and audio respectively, GarageBand
‘09 introduces a third track type: Electric Guitar. This is similar to a Real Instrument track in that it records audio, but provides a
conveniently set-up track that makes it easy for electric guitarists to get up and running.
What makes the Electric Guitar track type special, though, is that rather than seeing the usual list of available effects and settings in
the Track Info Panel, with an Electric Guitar track selected you’ll see a virtual representation of a guitar rig, complete with a choice of
amps and a selection of drag-and-drop stomp boxes. Clicking on the amp or a stomp box brings up the settings of the appropriate object
for adjustment, and while the selection of effects isn’t completely new, it’s an elegantly simple user interface for newcomers.
Seasoned SOS readers who record electric guitars might scoff at such basic functionality, but in the same way that GarageBand has
proved to be a useful and simple bread-and-butter sound source for keyboard players when there’s nothing else to hand, guitarists might
now also find themselves glad every Mac comes pre-installed with GarageBand, for the same reason.
As a guitar-related footnote, now that GarageBand ‘09 caters for guitarists with the new features for electric guitars and the guitar
lessons, it seems a shame that Apple haven’t made it possible for the Score view to display tablature (and maybe chord symbols). This
would seem a logical follow-through, since the lessons do have the ability to show tablature, but maybe providing a way for the user to
edit such musical data was deemed too complex for the application’s scope.

Published in SOS June 2009

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Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’

Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’
Producer: John Lennon • Engineers: Roy Cicala, Shelly Yakus Buy PDF
Published in SOS June 2009
Technique : Classic Tracks
Printer-friendly version
Engineer Roy Cicala worked on all of John Lennon’s albums from Imagine onwards, and in
‘Whatever Gets You Thru The Night’, recorded the only solo number one hit single of
Lennon’s lifetime.
Richard Buskin

I n June 1974, John Lennon’s life was in disarray. Not only was he battling a deportation order
by the US government on the grounds of a 1968 UK drug bust, and embroiled in litigation
over the legal dissolution of the Beatles, but publisher Morris Levy was alleging copyright
infringement of the Chuck Berry song ‘You Can’t Catch Me’ on the Beatles’ ‘Come Together’. In
his personal life, meanwhile, he was separated from wife Yoko Ono — who had, bizarrely,
orchestrated his affair with her personal assistant, May Pang — and making tabloid headlines due
to booze-and-drugs-fuelled excesses with fellow revellers Keith Moon, Harry Nilsson and Ringo
Starr, which had disrupted sessions for a Phil Spector-produced album of rock & roll oldies.
Lennon was in the midst of what he’d later describe as his 15-month ‘Lost Weekend’.
Nevertheless, he was also about to turn things around.
The previous December, following three months of star-studded craziness at A&M and, after
they’d been kicked out of there, the Record Plant (West) — characterised by Spector once
showing up in a surgeon’s uniform and, on another occasion, firing a gun into the control room
ceiling — the maniacal producer had disappeared with the master tapes, which eventually had to
be retrieved at a cost of $90,000 to Capitol Records. Having renewed his friendship with Paul
McCartney and forged stronger ties with his estranged son, Julian, Lennon decided to get his
career back on track by placing the Rock ‘n’ Roll project on hold so that he could record a new
album of original material.
Eventually titled Walls & Bridges, this transported John Lennon to the top of the US charts,
courtesy of songs that variously documented how he missed Ono (‘Bless You’, ‘What You Got’, John Lennon on stage at Madison
‘Going Down On Love’), his love for Pang (‘Surprise, Surprise (Sweet Bird of Paradox)’), his Square Garden, November 1974, after
emnity towards former Beatles manager Allen Klein (‘Steel & Glass’), and his ongoing struggles losing a bet to Elton John about the
with insecurity and depression (‘Nobody Loves You (When You’re Down and Out)’, ‘Scared’). success of ‘Whatever Gets You Thru
The Night’.Photo: Steve
What’s more, it also provided him with a pair of hits in the form of ‘#9 Dream’ and ‘Whatever Gets
Morley/Redferns
You Thru The Night’, the latter featuring a harmony vocal and piano contribution by Elton John
that helped secure John Lennon his only chart-topping solo single during his lifetime.
Recorded at New York’s Record Plant (East) in June and July of 1974, Walls & Bridges dispensed with the many luminaries recruited by
Phil Spector for the Rock ‘n’ Roll sessions and instead utilised a core group of Lennon-enlisted session players. A rhythm section
comprising drummer Jim Keltner and bass player Klaus Voormann was joined by guitarists Jesse Ed Davis and Eddie Mottau, keyboard
player Nicky Hopkins, saxophonist Bobby Keys and percussionist Arthur Jenkins. The final piece of the puzzle was Ken Ascher who, as well
as playing electric piano, Clavinet and Mellotron on Walls & Bridges, arranged and conducted the string and brass musicians from what
Lennon listed in the liner notes as New York’s “Philharmonic Orchestrange”.
It was a stellar line-up, led by John Winston Ono Lennon’s own contributions as a vocalist, guitarist, pianist and percussionist — each role
attributed to an assortment of self-penned pseudonyms. Behind the console were engineers Roy ‘I only like singles’ Cicala and Shelly ‘I
can’t take the pressure’ Yakus, not to mention studio assistant Jim ‘What it is’ Iovine.

The Talented Mr Cicala

“He became like a brother and a friend to me,” says Cicala, who served as an engineer on all of
Lennon’s albums from Imagine in 1971 to Double Fantasy just before his death in 1980. Cicala
also produced and engineered many of the world’s biggest stars during two decades of running
the Record Plant in New York, including Jimi Hendrix, Aretha Franklin, Madonna, Elton John,
Sting, Frank Sinatra, Dire Straits, David Bowie, Harry Nilsson, Miles Davis, Queen, Aerosmith,
Bon Jovi, Liza Minnelli, Roberta Flack, the Who, Frank Zappa, Lou Reed, Prince, Santana,
Charlie Mingus... It’s a list that has garnered him 10 platinum discs over the course of a career
that commenced back in the early ’60s.
Born and raised in New Haven, Connecticut, Roy Joel Cicala initially helped install the church
organs that his father designed, before marrying singer-songwriter Lori Burton and running his Roy Cicala with John Lennon and Harry
Nilsson at the Record Plant, 1974.
own small demo studio. Then, having landed a job as an assistant maintenance man under the

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Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’

guidance of Tom Hidley at the four-track A&R Recording facility of Phil Ramone — whom Cicala
watched, on his very first session in 1963, record Stan Getz and Astrud Gilberto’s ‘The Girl From Ipanema’. There, he “learned how to make
phasing machines out of tape machines, keeping old-fashioned tape machines within half a second from reel to reel”. Soon, Cicala began
engineering there himself and, in late 1965 and early 1966, sat behind the board alongside producer Tom Dowd for the classic, eponymous
debut album by soul-rock outfit the Young Rascals.
“When we moved into 799 7th Avenue, that, to me, was one of the best rooms in the world,” Cicala recalls. “It was amazing, and I
recorded people there every weekend. I was a workaholic — or an experimentaholic. I tried everything, and I loved it. One time, I put a
Shure mic underwater on a record by the Four Seasons, and although it didn’t work, I at least tried. Besides, it was a cheap mic! Then
again, when I recorded the Rascals, I put the drum kit in the stairway.”
In 1969, Roy Cicala moved to the Record Plant. Chris Stone, a national sales rep for Revlon, had persuaded the cosmetics giant to fund
the studio at 321 West 44th Street which, as designed by Gary Kellgren, provided its clients with not only comfort but inspiration.
“He single-handedly was responsible for changing studios from what they were — fluorescent lights, white walls and hardwood floors — to
the living rooms that they are today,” Stone would later remark about Kellgren. “His feeling, more than anyone else’s, was that a studio
should be a comfortable place to record. He was the first one who thought of the diversions, like the jacuzzi he built... The day we opened,
we were booked for three months.”
Indeed, the very first album to be recorded at the Record Plant was Jimi Hendrix’s Electric Ladyland. “Other people have taken credit for
the record,” Chris Stone would assert a decade later, “but about 90 percent of it was done in Studio A in New York with Gary and Jimi.”
By 1972, there were three Record Plant facilities — in New York, Los Angeles and Sausalito, California — and Roy Cicala would end up
purchasing the Manhattan studio from Warner Communications.
“In the beginning we built our own boards,” he says, “and although they weren’t Neve quality, they were rock & roll quality. By the mid-
’70s, everything was API, and then we went to Neve, but right from the start we were one of the top 10 studios in the world, and that’s
where we remained throughout the entire decade.”

The Lennon Connection

In July 1971, having already done some of the recording on Yoko Ono’s avant garde double
album Fly — including the stereo sound of a toilet flushing, as captured with a pair of Neumann
U87s — Cicala, together with Shelly Yakus, recorded the string overdubs for John Lennon’s
Imagine LP at the Record Plant after the basic tracks had already been taken care of at Lennon’s
home studio in Ascot, England. Then, in late ’71 and early ’72, with the Ono-Lennons now living
in the Big Apple, Cicala engineered their joint opus, Some Time In New York City, as well as the
following year’s Mind Games, the first of Lennon’s post-Beatles albums not co-produced by Phil
Spector. Indeed, this record was released after the ex-Beatle had already split from his wife and
left for LA with her assistant.
All of which brings us to the Rock ‘n’ Roll sessions...
Roy Cicala and partner Apollo 9 in his
“I was on a bottle of Dewers a day,” Cicala quips as soon as we reach the part of the story
new SA Plant studio in Sao Paulo.
where Lennon, Starr, Nilsson and Moon were sharing a house while he was staying at the
Beverly Hills Hotel. “I really didn’t want to go, because I knew it was going to be chaos and I
might end up never going back home to New York. So, I told John that and he said, ‘Well, bring your family.’ I said, ‘I’m not going to charge
you for bringing my family,’ and he said, ‘No, I don’t pay for it. EMI does.’ That’s how it was with the biggest artists, and so EMI rented me a
bungalow at the Beverly Hills Hotel with four or five bedrooms.
“At A&M they gave us Studio B, which only had 18 musicians on the board, whereas we had 25, and so the keyboards went straight into
the Scully tape machine which had microphone inputs. As always, my attitude was ‘Let’s do it. Nothing’s impossible.’ I mean, when those
musicians cooked, man, it didn’t make any difference if the sound was Class A. It was all about the feel.”
The problem was, the musicians’ cooking in this case was flavoured with plenty of alcohol.
“The drinking started right away,” Cicala confirms, “or, at least, an hour into it. I myself took it a little bit easy sometimes, because I was
the designated driver. But then again, if you didn’t indulge you couldn’t get with it in that company. So, I would partake, too.”
Mayhem ensued, with Phil Spector at the helm of a listing ship and, as revealed on the session tapes that have either been officially
released or bootlegged, barking orders in true dictator fashion: “Cut... Cut!”
“He had a Napoleon complex, and pretty much everything you’ve read or heard about him is true,” Cicala asserts. “I remember, when we
were recording all of the violins for the Imagine album, he’d want to stop the tape and stop the musicians to make a change, even though
he didn’t even know what the change should be. So, as he’d go to push the talkback button with his right hand, he’d grab that wrist with his
left hand and pull it back!” — echoes of Dr Strangelove.
“Phil could often be the biggest challenge in the control room,” Cicala continues. “He’d just carry on talking while we were cutting tape,
from one take to another, so I’d ask him to leave and he’d do so. We had a mutual respect... Thank God!”

Rock ‘n’ Roll

At the same time, there could be no denying Phil Spector’s talent, which Roy Cicala first

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Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’

witnessed at Mirasound Studios in New York.

“The engineer was Brooks Arthur and they were working with four channels, a big orchestra,
room reverb and guitar springs,” Cicala recalls. “That was Phil’s sound. How do you do that when
you separate all of the channels? You don’t. Live recording was his thing, and he tried to do the
same with John.”
To hear a Beatle backed — or, rather, enveloped — by Spector’s Wall of Sound, listen to
George Harrison’s ‘What Is Life’ on his 1970 album All Things Must Pass. It can easily pass as
yet another of what the producer himself once described as “little symphonies for the kids”. In
Lennon’s case, this was never properly achieved, even though he gave Spector free rein to
produce him for the Rock ‘n’ Roll album — “I’ll be Ronnie,” Lennon told Spector beforehand, Roy Cicala with John and Yoko outside
referring to Spector’s former wife, the lead singer with the Ronettes. And while Lennon eventually the Record Plant, November 1980.
used only four of the Spector-produced tracks on the album that he himself completed at the
Record Plant (East), one recording that he considered unfit for release, a superb, sub-six-minute rendition of ‘Be My Baby’, actually came
closest to recreating the producer’s trademark sound. (Yoko Ono included it in the 1998 John Lennon Anthology box set, but in a remixed,
edited-down version that fails to capture the true feel.)
“John’s vocal reverb was on a separate track all the time, and we did that live,” says Cicala. “He used to sing to the repeat and make it
work, and I’d have a noise gate on it so it would shut off and not go into infinity. Well, they lost that when they did the remixes, and it’s a
shame. They tried to make it too perfect, and it wasn’t a perfect record. It wasn’t an audiophile-type record.”
Nevertheless, it was a record that, following its release in February 1975, provided John Lennon with a number six chart placing in both
the US and the UK, as well as a Stateside Top 20 single in the form of ‘Stand By Me’. Furthermore, Rock ‘n’ Roll’s Lennon-produced
completion at the Record Plant (East) between October 21st and 25th, 1974, took place just four weeks after the US release — and three
after the UK release — of Walls & Bridges, which had been recorded in that same facility’s Studios A and B; the former equipped with an
old 24-input Spectrasonics console that had been built by Auditronics, and the latter housing a 32-input API. Both had Tom Hidley-designed
Westlake monitoring systems comprising a horn, tweeter and two woofers on each side, as well as Ampex tape machines.
“In the beginning, we had MCI machines which were modified because the tape path was terrible, and then we went to Ampex because
we had two trucks and the Studers were too big,” Cicala explains. “The Ampex were more compact, like Otari, and although they could be a
problem, too, we kept them running because we had a great maintenance guy. To me, that’s as important as a mechanic is to a racing
driver.
“Back then, we did most of our basic recordings on 16-track and then sort of pre-mixed and bounced them over to 24. We liked that
format. Of course, there wasn’t a lot of outboard gear: we had four EMT 140 plates in each room along with three [Fairchild] 670
compressors, [Teletronix] LA2As, black Urei LA4s, Publisons, Eventides and Pultecs up the gazunga. My problem was that I got away from
using anything except the Pultecs, because I couldn’t get them. Whenever I worked, somebody else wanted them, and me being the boss
meant I had to hand them over. So, I got into modifying a thing called the CBS Limiter, and I used it from the Rock ‘n’ Roll album onward. I
still have it and I was going to donate it to the Hall of Fame, but I don’t know anyone there.”

Walls & Bridges

Two days were spent rehearsing Lennon’s new compositions and arranging the entire Walls &
Bridges album, before the recording process commenced on day three with him in the dual role of
producer and artist. Still, Lennon continued to develop and fine-tune some of the material.
“He knew what worked when he heard it,” says Cicala, who once co-wrote a song with Lennon,
‘Incantation’, that was recorded by an outfit named Dog Soldier. “You know, John wasn’t a Bach
musician, and when he played the piano his chords might be simple, but the feel? Forget about it!
He was incredible. What’s more, when he’d go in front of the microphone and start singing, all of a
sudden the melody might change. It was amazing. He was like a genius with the words and the
melody. Even after rehearsing with the musicians, he’d change everything in the studio. And if his
vocals were a little flat, I’d just say, ‘Let’s do another take,’ and he’d be fine with that. Still, I
remember one time when something he sang was out of tune and it was ready to be mixed — He
said, ‘I’m not going to do this vocal again. Let’s just make it a little louder.’ So, that’s what we did;
we just pushed it up and brought it back down. That’s where he was great.
“Another thing I really admired about John was how he loved, while he and the band were in
the mood of a particular song, to do everything necessary for it. Maybe not background vocals or horns, but everything to do with the
rhythm section and possibly his vocal if he wasn’t too tired. Only then would he move on to the next number. I used to love to do that even
if it left us exhausted — the only problem was that we’d have to come in the following day for the next session.”
TV shows, radio broadcasts, books, magazine articles, newspaper headlines — all were rich sources of lyrical inspiration for John Lennon
throughout his songwriting career, and among the works in progress on Walls & Bridges was a number whose title and theme were
presented to him by someone he saw on late-night television.
“At night, he loved to channel-surf,” May Pang told the Radio Times in December 2005, “and he would pick up phrases from all the

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Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’

shows. One time, he was watching Reverend Ike, a famous black evangelist, who was saying, ‘Let me tell you guys, it doesn’t matter, it’s
whatever gets you through the night.’ John loved it and said, ‘I’ve got to write it down or I’ll forget it.’ He always kept a pad and pen by the
bed. That was the beginning of ‘Whatever Gets You Thru The Night’.”

Enter Elton...
In New York while preparing to record his album Captain Fantastic & The Dirt Brown Cowboy, Elton John was added to the Walls & Bridges
line-up when he dropped into the Record Plant with his label manager, Tony King, to contribute backing vocals to ‘Whatever Gets You Thru
The Night’. As Lennon himself recalled a few months later, “I was fiddling about one night and Elton John walked in with Tony King of
Apple — you know, we’re all good friends — and the next minute Elton said, ‘Say, can I put a bit of piano on that?’ I said, ‘Sure, love it!’ He
zapped in. I was amazed at his ability: I knew him, but I’d never seen him play. A fine musician, great piano player. I was really pleasantly
surprised at the way he could get in on such a loose track and add to it and keep up with the rhythm changes — obviously, ’cause it doesn’t
keep the same rhythm... And then he sang with me. We had a great time.”
Elton, who also played organ on the track, later commented, “Me playing organ on someone’s record? I mean, really. That’s disgusting
because I’m the worst organist. But we put that on and it was over and done with in five minutes.”
Just as straightforward was the harmony vocal that he and Lennon performed around a single microphone, even though Elton’s manager,
John Reed, subsequently complained to Roy Cicala that, while Lennon’s voice was prominent in the mix, not enough could be heard of his
client’s piano.
“I said, ‘Well, who’s the main artist here?’” Cicala remembers. “He said, ‘John,’ and I said, ‘Right. I know Elton is the artist, too, but if we
bring that piano up, that’s all you’re going to hear on the radio.’ He wasn’t happy, but the rest is history. The record was a hit and you could
hear Elton very well on the radio.”
For his part, Elton John was less than ecstatic about his efforts to record a harmony vocal on a second track, ‘Surprise, Surprise (Sweet
Bird Of Paradox)’. Unable to match Lennon’s idiosyncratic phrasing, he struggled for over three hours to achieve the desired result, and a
few months later he recalled that “People were leaving the room. Razor blades were being passed out!”
In return for his efforts, Elton secured a promise from Lennon that, should ‘Whatever Gets You Thru The Night’ top the singles chart, the
ex-Beatle would join him on stage during his forthcoming US concert tour. Lennon was sceptical that this would ever happen, but after the
song did reach the top spot he was good to his word, performing three songs with Elton at Madison Square Garden on Thanksgiving night,
November 28th, 1974: the aforementioned hit number together with ‘Lucy In The Sky With Diamonds’ — Elton’s then-current single, on
which Lennon performed a harmony vocal — and the Beatles’ ‘I Saw Her Standing There’, which Lennon credited to “an old, estranged
fiancé of mine named Paul.”

Number Nine
Meanwhile, the follow-up single to ‘Whatever Gets You Thru The Night’ began life with the working title of ‘So Long Ago’ and took its initial
melody from the orchestral arrangement to Harry Nilsson’s version of ‘Many Rivers To Cross’, the opening track on his Pussy Cats album
which Lennon had produced earlier in the year. This was then embellished by words that came to John in a dream, involving a couple of
women echoing his name. Hence the eventual title, ‘#9 Dream’, which continued Lennon’s fascination with the number that followed him
from birth to the grave. Born on October 9th, 1940, his first home was at 9 Newcastle Road in Liverpool; Beatles manager Brian Epstein
first saw the group play on November 9th, 1961; John met Yoko on November 9th, 1966; in 1968, he constructed the sound collage
‘Revolution 9’ for the Beatles’ ‘White Album’; in New York, he and Yoko lived in the Dakota building on West 72nd Street (seven and two is
nine); in 1975, their son, Sean, was born on John’s birthday, October 9th; and when John was shot and killed just after 11pm on December
8th, 1980, it was December 9th back in England.
Of course, following its January 1975 release, ‘#9 Dream’ peaked at... yes, number nine on the US singles chart, although not before Roy
Cicala’s wife, Lori Burton — who also contributed backing vocals along with May Pang and Joey Dambra — tweaked the nonsense phrase
“Ah! bowakawa puss-ee, puss-ee” that had also been a part of Lennon’s dream.
“Al Coury, the promotion man for Capitol, said, ‘They’re not going to play this record,’” recalls Cicala, who yelled “take nine” with Lennon
every time tape was about to roll for the song. “When John asked Al, ‘Why?’ he was told, ‘Because you’re saying ‘pussy’ on it!’ So, Lori
changed it to ‘Ah! böwakawa poussé, poussé,’ kinda like French, and it worked. John listened to us. In fact, he listened to just about
everything. He never used to come to the mix sessions until we called him. After all, there was no automation, so why have a breakdown
over it? Just come in when you’re ready and then tweak it a little bit.
“One time, I gave John the tape after finishing a mix and he took it upstairs for Greg Calbi to master. Then John called me: ‘There’s been
a problem up here.’ ‘OK, what?’ ‘I don’t know. Come and help us.’ So, I went upstairs and I walked into the cutting room, and the mastering
machine had tape maybe 12 inches high spilled all over the place. ‘The machine went bananas,’ John said. I did an about-face, walked out,
and they caught me down on the first floor as I was leaving the building. ‘Ha-ha-ha-ha-ha!’ They were joking... We really had a lot of fun
doing those records.”

Artist: John Lennon

Track: ‘Whatever Gets You Thru The Night’
Label: Apple/EMI/Capitol

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Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’

Released: 1974
Producer: John Lennon
Engineers: Roy Cicala, Shelly Yakus
Studio: The Record Plant (East)

Microphones & Miking

For the Walls & Bridges sessions, Jim Keltner’s drums, positioned inside an open booth at the back of the live area, were miked with a
Shure SM57 suspended inside the kick with surgical tubing and both heads closed; an Altec 633A on the snare; another SM57 on the hi-
hat; Sennheiser 441s on the floor tom and two rack toms; and a pair of Neumann U87s as overheads.
“I never miked the snare from underneath,” says Cicala. “I still don’t. The ‘salt shaker’ [Altec]went through a Pultec, wide open at 10k,
because it had no top end but it did have a larger capsule that produced a better drum sound. As for the 57 inside the kick drum, I took
an aluminium insert from a reel of one-inch tape, drilled it three times in a triangle shape, threaded it, put eye screws in it, and then the
eye screws would attach the surgical tubing to the drum, with the wiring coming out of the little breather hole. Then again, if there was too
much attack I would take it apart, position the 57 facing the front of the drum — on the other side to the pedal — and then cover the front
with a blanket so there’d be more thump.
“I tried it with tom-toms, too, and I made my own dampers for them. You know, sometimes they ring out too much, and then, if you
dampen them with tape on the outside, it changes the characteristic of the drums. So, I would take a piece of the black sponge that
microphones are sometimes packed in, cut it into a one- or two-inch square, about half an inch thick, and use gaffer tape to attach it to
the head. As a result, every time you hit the drum, that would ring out and automatically close or go down against the head and dampen
it.
“A lot of times we had too many cymbals, and I guess everybody has that problem, especially with rock & roll, but I’d do something that
others, I guess, wouldn’t do — I would use de-essers that they use on vocals. Isn’t that logical? I actually used to make Keltner crazy. If
we wanted that real slap tom-tom sound, I would take his cymbals away. He’d say, ‘You can’t do that!’ I’d say, ‘Yes, we can. Here, have
a drink.’ It was very free-and-easy: ‘You do your thing, I’ll do my thing,’ and it was great. The musicians did anything we wanted and vice
versa.”
In front of Keltner, Klaus Voormann’s bass, going through an Ampeg B15N Portaflex amp, was both DI’d and recorded with an Electro-
Voice RE20, while Eddie Mottau’s rhythm guitar was miked with a Sony C37, Jesse Ed Davis’s lead guitar was recorded with a U87 —
with a combination of that and an SM57 employed for overdubs, making use of the room’s natural ambience when it wasn’t filled with
musicians — and three mics were employed on the acoustic piano: a pair of U87s near the holes on the Steinway, as well as another
SM57 down below, with 5dB of limiting. Percussionist Arthur Jenkins was also miked with 57s, while, in terms of overdubs, the three
trumpets were captured with a single 57 inside a pyramid formation, the tenor and baritone saxes were recorded with a C37, and the
backing vocals were alternately performed in front of either a U47 or U87.
“For John’s lead vocals we used a tube U47 and an SM57,” Cicala says. “I would treat the 57 and do whatever I had to do to make
him present, whereas the 47 was for quality. And although we had repeat on his voice, I never used it like everybody else — they all
used it in tempo, but I used it like a hiccup. It was more noticeable and it made the effect better. If you used it in sync, for me something
was wrong with it, it was too audiophile. Then again, John always wanted his vocals buried in the mix. He was insecure about his voice
and I’d have to talk to him about that — unless he felt it was a great performance.”

The Brazilian Job

In December 2005, Roy Cicala relocated to Sao Paulo, Brazil, and within a couple of years he had built a new five-room studio named
SA Plant (as in South American Plant) that is equipped with a 32-input Neve V3 in the main control area, Studer and Lyrec tape
machines, original Record Plant, Tom Hidley-designed JBL LSR 4328 monitors with TAD woofers, Pro Tools and Logic, and a host of
new and vintage outboard gear and microphones. The other rooms are used for overdubs, computer-based work and commercials.
“The room that I have is really New York-style, modelled on the Record Plant,” he says. “I did the acoustics and people are very, very
happy with it. The booth, which is about 18 x 8 feet, is just for vocals, and what I did was run high-level wires from this studio down to a
basement room and also to a studio that’s downstairs. So, we can see the drums up here, put the bass player and two guitar players in
the control room with the heads of their amps, and the speakers will be downstairs with mics on them. The only thing you can’t do is get
that feedback from playing a guitar in front of an amp.”
Apparently, it doesn’t matter. For, not only is the studio solidly booked with artists old and new who are familiar with Roy Cicala, but
also many from the US and Europe who are looking to save money.
“We can do things for a third of the price and they have a tax shelter,” Cicala explains. “If it’s recorded in Brazil, they don’t have to pay
American or European taxes. So, between the low outlay, the layout, the equipment, our great engineers and even my own ears, people
are finding plenty of reasons to come here.”
[email protected]

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Classic Tracks: John Lennon ‘Whatever Gets You Thru The Night’

Published in SOS June 2009

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Creating Custom Touchscreen Interfaces In Logic

Creating Custom Touchscreen Interfaces In Logic

Logic Notes & Techniques Buy PDF
Published in SOS June 2009
Technique : Logic Notes
Printer-friendly version
You’ll be surprised at how cheap and easy it can be to enhance the usability of your Logic
setup with a touchscreen.
Geoff Smith

I t will be little surprise to anyone that, thanks to Apple’s iPhone and iPod Touch products, the
profile of touchscreen interfaces has risen dramatically. However, what may be surprising to
you is just how easy and affordable it is to buy a touchscreen and incorporate it into your
music setup. Touchscreens regularly come up for sale second-hand, as they have been used in
shops, medical establishments and other outlets for many years now. Ebay is probably the best
place to find one; a quick search at the time of writing revealed that a used 15-inch touchscreen
can be bought for around $150. That size of screen is considerably bigger than the 12-inch
touchscreen on the Jazz Mutant Lemur controller (reviewed in SOS March 2007,
www.soundonsound.com/sos/mar07/articles/lemur.htm), which weighs in at around the $2000
mark, although it is worth pointing out that the Lemur’s capabilities are a lot more advanced.

How Do Touchscreens Work?

A touchscreen is a display that has the ability to detect the location of a point of contact within the display area. The point of contact can be
either from a finger or from some kind of stylus, depending on what type of touch the screen will recognise. The two most common types of
touchscreen interfaces are capacitive and resistive, the main difference between the two types being that capacitive touchscreens are
usually visually clearer than their resistive counterparts. (The iPhone has a capacitive touchscreen, while Palm’s Treo and Motorola’s ROKR
E6 use resistive screens. One benefit of a resistive screen is that it can be used with a stylus as well as your finger, but on the down side,
resistive screens are more easily damaged by sharp objects. Capacitive touchscreens can also be more sensitive than their resistive
counterparts, as they do not necessarily need to be pushed; some can detect your finger within 2mm of the screen, and therefore often
respond to a lighter touch.
A touchscreen interfaces with your computer via two connections: a VGA or DVI connection for video, and a USB or serial port connection
for the touch interface (Mac users need to make sure their screen has USB). It’s easiest to think of the touch interface in PC terms as being
similar to connecting a second mouse. The USB cable carries the X-Y coordinates of your touch
signal to the computer and requires a stand-alone driver, in the same way that some specialist
mice or tablets do, so when buying a touchscreen it’s important to ensure that the one you choose
has the drivers and software for the operating system you use. This is important because the
control interface needs to be set up as either a mirror of your first screen or a separate second
screen, and then calibrated to the area you are working in. The calibration software usually takes
the form of a simple program that generates a series of crosses on the screen, which you touch.
From that, the program calculates the working area, to track your touch accurately. Be aware that,
as with any device, there are good and bad examples. Watch out for screens that require firm
pressure to register your touch, as these will quickly become fatiguing and make adjusting
parameters via sliders hard work. These are the connectors on my own
touchscreen, the catchily named ELO
Because touchscreens are commonly found at tills in shops, and in other environments where a
ET1525L-8UWC-1, which I found on
traditional desk is not present, they are available with very flexible mounting arrangements and eBay for around $150. Left to right:
can be attached to a wide variety of stands. This mounting flexibility is a bonus for musicians: power, stereo audio input for built-in
manoeuvre the screen behind your keyboard for patch editing or place next to your computer to speakers, USB, VGA.
help with mixing.

The Touchscreen & Logic

To look at the potential of a touchscreen as a controller with all music applications is beyond the scope of this article, so I’ll be explaining
how to use one solely with one of the most popular DAWs around: Logic Pro, whose Environment also lends itself very well to creating
specialist interfaces.
A touchscreen can work just like a normal monitor and mouse combination, so straight off the bat you can interact with all aspects of
Logic’s interface. Changing levels and panning in the mix window are particularly easy. Editing plug-ins is also much more immediate than
using a mouse.
One of the things that has always set Logic Pro apart from other sequencers is the depth of what can be accomplished in the Environment
window. Because the Environment provides a configurable virtual view of your MIDI studio and mixing objects, you can design your own
Environments to suit a touchscreen interface. Some of the things you could build to take advantage of the touchscreen are a specialist

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Creating Custom Touchscreen Interfaces In Logic

touchscreen mixer, multiple X/Y Pads for synth control, and SysEx or continuous controller maps to give your hardware a touchscreen
interface.

Basic Mixing Screen

When mixing or editing on your main monitor, you can have your touchscreen showing a useful
Environment containing just the basics of what you want to adjust. On a 15-inch touchscreen I’ve
found that 32 channels of volume, pan, mute, and solo controls can be displayed at a size that
works well. You could try other variations, such as a screen containing 128 tracks of solo and
mute buttons.
To begin building the basic mixer, it’s best to create a new template from scratch:
Start a new song and add an audio track.
Call up the Environment window (Command-8), and create a monitor object
(New > Monitor).
Attach the Audio 1 object to the Monitor object. To accomplish this, go to The basic mixing screen offers 32
the small triangle at the top right of the Audio 1 channel object. This arrow channels of volume and pan controls,
plus solo and mute buttons.
represents the output or outlet for the control signals of that channel. Make
a connection between audio channel 1 and the monitor object, by click-
holding the triangle at the top of channel 1, moving the mouse to the centre
of the monitor object and releasing the mouse button. You should now see
the outlet of the Audio 1 object cabled to the inlet of the monitor object. The
click-hold method will be used throughout to connect objects in the
Environment.
Now it’s time to do a little investigation. Adjust the volume, pan, mute and solo of the track and
you’ll see that the monitor object displays the messages Logic uses for these functions. It’s worth
observing that all Logic’s mixing functions inside the Environment can be controlled by two
message types: Continuous Controller messages for volume and pan; and Fader commands for
controlling solo/mute and plug-in parameters. Continuous Controller messages start with a circular
symbol, then have three columns of numbers. An example message you might see in the monitor
box for the Audio 1 pan control would be ‘1 10 40’, where 1 denotes channel 1, 10 denotes CC 10
(pan), and 40 is the value of the control signal (in this case, pan position in the range 0-127).
Fader messages follow a similar format, but start with an ‘F’. For example, a mute button could
be turned on with the message ‘F 1 9 1’ . The ‘F’ means a Fader message; the first number
column is the object position, in this case, one (higher numbers denote the insert plug-in slot
number); the second column represents the parameter being controlled, in this case, the mute
channel; and the last number is the control signal value, in this case, one, meaning ‘on’. Call up a
plug-in on audio channel 1 and move some of the controls; you should see a specific Fader
message for each in the monitor object.
Now let’s get down to actually creating our own control objects! First, we’ll make a Fader object
to control channel volume.
Create a new Fader object (New > Fader > Vertical 1).
Cable the outlet of the Fader object into the inlet of the Audio 1 channel
object. In the Environment window, the Audio 1
Call up the inspector by pressing ‘I’ on the keypad. Click on the new Fader channel object is cabled to a Monitor
object to make sure the Inspector displays its parameters, and set them as object.
in the screen just overleaf. You will have recreated the message for audio
channel 1 that you saw in the monitor object when you moved the volume
slider, earlier.
We need an object to control the mixer’s Pan pot (‘New’ menu > Fader >
Knob 6). Cable the knob into the volume fader’s inlet and enter the values on
the right into the Inspector.
Test the two objects you have created, by moving them. If everything is
connected correctly, you should see audio channel 1’s volume slider and pan
knob mirroring these movements.
Next, we need to create two buttons to control the solo and mute functions.
Make one button (New > Fader > Button) and cable it to the inlet of the pan
object.
Create another Button and cable the outlet of that into the previous button’s

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Creating Custom Touchscreen Interfaces In Logic

inlet. You should now have a chain of objects comprising two buttons, a
knob, and a fader. The Inspector for the volume control
object.
The solo and mute buttons in Logic use Fader control messages, as
explained earlier. Set the two new buttons to control solo and mute
respectively, and copy the parameters from the Inspector screens below
right. Note that the output and input boxes are set to Fader.
Delete the monitor object from audio channel 1 and cable the outlet of the
channel to the inlet of the mute button, to create a loop. This loop will carry
the value from the mixer objects to your new objects, and vice versa (see
screen above).
You can now move and resize any of your new control objects, change the type of fader or
button, and even modify the colours. Once you have a single-channel interface you like, you can
add more channels by simply copying and pasting the channel and its control objects. Then
simply alter the audio object’s channel in the Inspector from audio 1 to audio 2 or audio 3 (and so
on). Note that you can change the audio channels to audio instrument channels and the control
objects will still work.

Want Four Kaoss Pads? The Inspector for the pan control object.

The Kaoss Pad is an effects processor from Korg that has a touchpad interface for altering, in real
time, the parameters of whatever effect is selected. For example, if a Delay effect were selected,
the X-axis of the touchpad might control delay time and the Y-axis the amount of feedback.
The Vector object inside Logic allows you to recreate Korg’s touchpad interface on a touchscreen.
The advantage of using a touchscreen and Logic is that you can create as many Vector objects
as you like, and connect them to any instrument or plug-in parameters. You could use the X-axis
of a Vector object to change oscillator type in the ES2 synth and the Y-axis to add distortion and
pitch modulation and open the filter, for example. Another Vector object could then control the
parameters of any insert effects you put after the ES2.
Create a new project and add a single audio instrument track.
Go to the Environment page and load an ES2 on audio instrument channel 1
and attach a monitor object to the outlet of that channel.
Select the audio instrument 1 channel and press copy (Command-C). Now
you’re going to create a new environment layer to put your Vector objects
on. The Inspector for the Mute and Solo
Call up the inspector, by pressing ‘I’. At the top of the Inspector is a menu objects.
that allows you to choose which layer of the Environment you want to view.
From that menu, choose Create New Layer. To rename the layer, click on the
layer name in the Inspector, type ‘Kaoss Pad’ and press Enter.
Paste audio instrument 1 on to the empty page (Command-V).
Add a Vector object (New > Fader > Vector). Cable its outlet to the inlet of
audio instrument channel 1.
Resize the Vector object to a quarter of the size of your touchscreen: click
on the Vector object to select it, then go to bottom right and click-drag to
resize.
Hold down the Alt key and click-drag to create three more Vector objects.
Name them Vector 1, Vector 2, and so on.
Now that you have your Vector objects, it’s time to do something fun with them.
Go back to the Environment ‘Mixer’ layer that has your monitor object on it.
Call up an ES2 synth and select the preset Synth Leads > Big Trance Now.
Move the ES2 window so that you can still see your monitor object, and
adjust the Cutoff and Resonance controls of Filter 2. You should see the
following messages for Filter 2: Cutoff F 2 35 [value] (where value is the
setting between 0 and 127 of the Filter 2 Cutoff), and Resonance F 2 36
[value] (again, between 0 and 127).
Return to your Kaoss Pad Environment layer and enter the two Filter 2
messages as the control destinations for Vector 1. Click on the Vector 1
object so that it’s selected. In the Inspector, you should see that you can set
vertical and horizontal messages. Set the vertical message to change Filter

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Creating Custom Touchscreen Interfaces In Logic

2 Cutoff and the horizontal to change Filter 2 Resonance.

You should now have a fully working Vector controller for Filter 2’s Cutoff and Resonance. Play a
bass line on your keyboard and move the Vector object around to hear it working. You could then
use your second Vector object to control some effects processing, as follows:
Go to the ‘Mixer’ layer of the Environment and call up Logic’s PlatinumVerb
plug-in on insert 1 of audio instrument channel 1.
Move the plug-in window out of the way so that you can see the monitor
object you created earlier, and adjust the Wet output level control and the
Reverb Time control. You should see the Fader commands F 3 19 [value] and
F 3 8 [value] in the monitor object.
Return to your Kaoss Pad layer and click on the Vector 2 object to select it,
then go to the inspector and enter those messages into the horizontal and A complete signal loop is created by
vertical outputs. You should now have one vector object controlling Filter 2 cabling the outlet of Audio Channel 1 to
and a second controlling Reverb. the inlet of the Mute button.

Now set up the other two Vector objects to control other ES2 or plug-in
parameters of your choice.

Things To Make & Do

It’s worth mentioning that you can use the outputs of the Vector objects in much more complex
and flexible ways. For example, using the Transformer object you could enable the X-axis of a
Vector object to control many plug-in parameters at one time in specific ranges. Vector objects
can also be set up to feed the user-definable controllers in ES2’s MIDI section, which can then be
used as ‘sources’ inside the modulation matrix, allowing that source to be a controller for any
destination within ES2. Because you can build System Exclusive and controller maps for your
hardware synth inside Logic, you could also build your hardware instruments a touchscreen Fancy owning four Kaoss Pads?
interface. I’ve created a series of Environments for my Akai MPC4000 that improve the user
interface dramatically. The first significant gain is that I can now control any sample’s parameters
all on one page of my touchscreen, laid out exactly as I want them. The difference this makes is
nothing short of amazing. Another interesting benefit is that because Logic allows long strings of
SysEx data to be sent using just a button, you can effectively create presets for any groups of
SysEx commands you like. It’s easy, for example, to set up presets for all the synth and effect
parameters within the MPC: one SysEx change could set the selected sample’s pitch down a
fourth, filter to low-pass, filter cutoff to 50 and amplifier envelope decay to 34, and add reverb —
all at the touch of a button.

Conclusion
The touchscreen interface I made for my
There are so many ways to use Logic’s Environment with a touchscreen to improve your working Akai MPC4000.
methods that I think the case for owning and using one is very persuasive. I wouldn’t recommend
throwing out your mouse and keyboard yet, but as an addition, a touchscreen makes a lot of
sense. Basic tasks such as soloing and muting different tracks while editing and mixing become
so much easier. So before you spend a hundred pounds on a controller with a few plastic knobs
and sliders, bear in mind that buying a touchscreen gets you a second monitor and a touchscreen
interface for your DAW and hardware into the bargain!

Published in SOS June 2009

A set of buttons I made to use as

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Creating Custom Touchscreen Interfaces In Logic

‘presets’for the MPC. They control many

different settings at once.

A useful Environment mixing template,

offering (top to bottom): 0dB button,
phase invert button, gain fader, high-
pass on/off and cutoff frequency
controls, solo, mute and pan controls.

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Cubase: Funky Guitar Processing

Cubase: Funky Guitar Processing

Cubase Notes & Techniques Buy PDF
Published in SOS June 2009
Technique : Cubase Notes
Printer-friendly version
If you want something to spice up rhythm-guitar loops to use in your funk and dance tracks,
Cubase has just the Tonic
John Walden

N obody who’s listened to the late ’70s and early ’80s funk/disco music of Chic or Sister Sledge can have failed to recognise just how
danceable rhythm guitar can make a track. In more recent years, that same rhythmic sensibility has become an important part of a range
of contemporary dance styles, particularly house. I was recently fortunate enough to try out Rob Papen’s excellent RG. This plug-in
combines a programmable rhythm guitar engine with filter and effects processing and, for dance styles such as house, it’s a great tool for creating
a range of rhythmic guitar styles. The down side of RG is that you can’t use the filter or effects elements to process your own guitar parts or loops
— which got me thinking about how to create similar results using Cubase. So, armed with my trusty Strat and a few pre-recorded guitar loops, I
set to work.

A Sound Source
Whatever processing we might subsequently decide to apply, a well recorded original sound makes the best starting point. I started experimenting
with two different sources: first, I played some parts in myself, recording them through a Line 6 Pod X3 using a clean sound with a touch of
compression; and second, I created a couple of patterns in Rob Papen’s RG, but without using that instrument’s own filter, EQ or effects, so that
they contained just the unprocessed guitar sound. If you want to work through the ideas in this column on your own system, then a small selection
of these clips are available for download on the SOS web site at www.soundonsound.com/sos/jun09/articles/cubasetechmedia.htm.

Step On
With our guitar parts in place, we need to look at what processing options in Cubase
might help us replicate those in RG. Probably the most influential (and fun) feature of RG is its filter, and Cubase includes an equivalent plug-in,
Tonic. This was introduced way back in SX2.2 (see SOS November 2004) and, while it isn’t as versatile as some other filter plug-ins (or the filter
section of RG), it is capable of some great sounds, as well as being pretty easy to use.
The first screen (above) shows a good starting point if all you want to achieve is a gentle filter sweep
with Tonic. Here, the settings in the Env Mod section are basically set to ‘off’, meaning that the filter is
not subjected to envelope modulation. Any changes in the filter with time are therefore controlled by
either the LFO Mod settings or, under user control, via the X/Y matrix pad (bottom left). The Cosine
preset from the Step window creates a smooth, cyclical change in the filter, while the Depth setting of
about 50 percent means that the tonal changes are not too extreme. The Rate control is set to modest
two beats per step, so as to give a fairly slow sweep of the filter. The result is a slow change in the
sound: not overly exciting, but enough to add subtle movement for a rhythm guitar part without attracting
too much attention. Tonic: capable of gentle filter sweeps
So far, so vanilla — so let’s try something more radically processed. Probably the most fun is to be (as shown here) through to more
had with the controls in the Env Mod section, but experimentation with the LFO Mod and Step windows extreme sound mangling.
is well worthwhile. For example, selecting the Square preset for the Step Window and dialling in a rate of
four steps per beat causes the filter to open and close (to an amount controlled by the Depth control)
once every bar. If the timing doesn’t quite match your playing (for example, if the strumming is slightly
ahead or behind the beat), you can simply edit the step pattern to match. Using this same approach, you
can fully customise the positions of the steps to match the strumming pattern so that the filter changes
provide an additional emphasis to the natural rhythm of the part.
Bringing in the Env Mod controls adds some further possibilities by applying envelope modulation to
the filter cutoff. Three modes of operation are offered (Follow, Trigger and MIDI). Follow responds to the
dynamics of the input signal, whereas Trigger uses the input signal to trigger the envelope to run
through a complete cycle. MIDI uses MIDI notes to trigger the envelope (from a keyboard or a recorded
MIDI track — you simply set the MIDI output of the track to Tonic in the Inspector), with the filter cutoff
tracking the note number. All of these can be useful, but as Follow responds to any volume changes in
the guitar part, it is a good starting point. The screenshot below shows an example that produces a really
nice wah-style effect that syncs with the project tempo. Try it out with the 90bpm loops that accompany
this article on the SOS web site.
Tonic can be pushed much further, as demonstrated by a number of the presets. For example, try the
‘ReGroove 3’ or ‘Beat Restructure 2’ presets. The first adds an extra pulse-like element to the sound: try
adjusting the Depth controls in both the Env Mod and LFO Mod sections to vary the effect. Very funky.
In contrast, the second turns the guitar part into something almost percussive in nature — and adjusting The step window within Tonic can be
the Cutoff and Resonance controls can change the tonal character of the sound to generate a turntable- used to provide step-like opening and
closing of the filter that synchronises to
style effect (try this on the 90bpm loops). It certainly doesn’t sound like a guitar!
the strumming pattern of the loop.
Get In Shape

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Cubase: Funky Guitar Processing

Another key element in RG’s arsenal is the Amp section. This is a synth-like amplitude envelope, rather
than a virtual guitar amp (although the latter is found among RG’s FX options), and it can be used to
create some interesting changes to the guitar sound.
Cubase’s Envelope Shaper plug-in can be used to achieve similar effects by, for example, increasing
the Attack but giving it a short Length setting. This accents the initial portion of each strum, resulting in a
much stronger rhythmic feel. Alternatively, decreasing the attack and increasing its length can make the
sound less guitar-like, although it also moves the rhythmic emphasis. As an example, set up an instance
of Envelope Shaper after Tonic (any gentle filter sweep will do) and apply it to one to the RG-based Adding in the Env Mod section allows
guitar loops I’ve created for download. The removal of the initial attack makes it less obvious that we’re for more creative possibilities — try
hearing a guitar, but the performance still has a guitar-like rhythmic quality. these settings with the 90bpm example
loops for a lazy wah-like sound.
Delay Tactics
RG includes a couple of nice delay settings in its FX section, and it almost goes without saying that the
addition of some suitable tempo-matched delay from one of Cubase’s delay plug-ins can add further
interest and rhythmic movement to your processed guitar part. This does, however, need to be done
with some care. If the playing is fairly busy, don’t add too much feedback — or mix it too high — because
the additional repeats will begin to sound messy. Where the original part is a little sparser, or at a lower
tempo, you can probably get away with laying the delay on a little more thickly.
Any of the Cubase delay plug-ins can be put to good use here, but if you want to create something
more unusual, ModMachine provides both delay modulation and filter options, allowing the tonal
character of the repeats to change over time. In combination with the filter effects of Tonic, this can give
you some truly inspiring combinations. It’s well worth engaging the Sync option for the Delay setting, but
beyond that, experimentation is very much the order of the day. That said, there are a number of
ModMachine presets for ‘funky guitar’, which provide obvious starting points.

Guitar No More
If desired, a combination of the Tonic, Envelope Shaper and ModMachine plug-ins can easily turn a
simple guitar part into something quite unlike a guitar. However, if you want to go further down that road
and create some synth-like sounds, then modulation-based plug-ins such as Ring Modulator,
Tranceformer or Metalizer can also be pressed into service.
For example, try a combination of the Tonic settings shown in the initial screen shot (just a gentle filter The Envelope Shaper plug-in can either
emphasise the percussive attack of the
sweep) followed by the ‘Boomerang’ Tranceformer preset and apply it to one of the downloadable
strumming or, as here with a low attack
example loops. The end result is rather ambiguous in terms of harmonic content, but takes on the setting, make the guitar less guitar-like.
character of a synth-based percussion part. If you replace Tranceformer with the ‘Cobalt’ preset from
Metalizer, a little more of the harmonic character is retained and a soft, rhythmic synth part is generated.
This can sound great when a more lightly processed version of the guitar loop is panned to the opposite
side of the stereo image, because the two parts obviously play together very tightly.

Order, Order!
As with all such creative processing experiments, there are no set rules as to the order in which the
various plug-ins are combined. It’s certainly worth trying Envelope Shaper both pre- and post-Tonic,
because the order of the processing can produce very different results, particularly when using higher
Attack settings in Envelope Shaper. Similarly, placing ModMachine before Tonic gives Tonic’s filter
something different to work with, and therefore produces a more complex output. As with a number of
the ideas outlined here, I’ve created a series of channel presets to illustrate this. Along with the example
audio loops, these are also available for download from the SOS web site at the URL given earlier.
If you fancy turning a simple funky guitar part into something that might be more suitable for synth-
based dance or electronica styles, I’ve no doubt that a dedicated and well-equipped tool such as Rob
Papen’s RG makes it easier to get instant results. But if RG and its ilk are luxuries you can’t afford, the
plug-ins provided as standard with Cubase include all the necessary processing and effects, and
together are certainly capable of some very similar effects. Time for me to get the funk outta here!

The ModMachine plug-in can add some

excellent delay-based effects.

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Cubase: Funky Guitar Processing

Above: For something more synth-like,

the Ring Modulator, Metalizer or
Tranceformer plug-ins do a nice turn.

Right: Don’t be afraid to experiment with

the order of the effects processing
made easier by the drag ‘n’ drop
function allowing easy re-ordering of
Insert effects within the Mixer channels.

Tip: Wave Goodbye To Slow Boot Times

If you run Cubase or Nuendo on a PC, and find that your Waves plug-ins result in a very slow boot-up time for your DAW, there’s a free
solution at hand at www.xlutop.com/buzz/zip/shell2vst.zip.
Download and unzip the file to your Waveshells folder, drag your existing Waves DLL files onto the shell2vst.exe file in Windows Explorer,
then simply copy the resulting Waves folder to your VST plug-in directory, and remove the existing Waves DLLs (that is, back them up in
another location) and you’re ready to go. The first boot-up of Cubase will take a while — because all the plug-ins need initialising again — but
thereafter you should find that your load time is dramatically reduced. What’s more, you should now be able to organise your Waves plug-ins in
folders, making them much more manageable in the Cubase or Nuendo GUI than they are with the default Waves installation. I don’t know
whether users have been experiencing similar problems on Mac-based setups — I don’t have my Waves bundle installed on the Mac — but if
you have, feel free to drop us a line. Matt Houghton

Knob Fiddling
As much as I like Tonic, it does have one real irritation: accurate setting of any of the rotary knobs is really fiddly using a standard two-button
mouse. However, if you have a mouse with a scroll-wheel, thankfully, things are a lot easier. With the mouse cursor placed over the control you
wish to adjust, the mouse wheel can be used to rotate Tonic’s knobs more accurately and, as usual, the setting is displayed in the panel
immediately beneath the Env Mod section. This also works with other Cubase plug-ins including the Monologue, Embracer and Mystic synths
[Not to mention the Channel Faders! — Ed]. If you also hold down the Shift (or Alt, depending upon the plug-in) key, this can give even finer
control via the mouse wheel.

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Cubase: Funky Guitar Processing

Published in SOS June 2009

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Cubase: Funky Guitar Processing | Audio Files

Cubase: Funky Guitar Processing | Audio Files

Hear For Yourself
Published in SOS June 2009
Technique : Cubase Notes
Printer-friendly version
If you want something to spice up rhythm-guitar loops to use in your funk and dance tracks,
Cubase has just the Tonic
This ZIP file includes the audio and VMX files that accompany John Walden's column about funky guitar processing using Tonic and other effects in
Cubase.

Published in SOS June 2009

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Effective Selecting & Locating In Digital Performer

Effective Selecting & Locating In Digital Performer

Digital Performer Notes & Techniques Buy PDF
Published in SOS June 2009
Technique : Digital Performer Notes
Printer-friendly version
The ability to move around your project quickly and easily, making fast and accurate selections for
editing, can take much of the pain out of music production. We round up some of the best power-
user shortcuts.
Robin Bigwood

M IDI programming, audio editing, comping, mixing, bouncing to disk: what these and dozens of other routine DP tasks have in common is
the need to frequently locate playback position and make accurate and appropriate editing window selections prior to carrying out edits.
Location and selection are such a fundamental part of using a DAW like DP that it pays to get quick at them, so you can potentially work
much faster, focusing on your production more and wrestling with the application less.

Location, Location, Location

DP puts only one true location command ‘up front’, and that’s Rewind, otherwise known as Return To Zero. Click this, or use the keypad ‘1’ key
command, and normally the playback wiper jumps to the start of the sequence. If you have Memory Cycle turned on, though, and the wiper is
somewhere within or to the right of the cycle region, Rewind locates the wiper to the Memory Start location: click Rewind or hit the keypad 1 button
again to get back to the sequence start.
What if you want to jump somewhere else in your sequence? For this, a single powerful location feature stands out: the dot trick, which I’ve
mentioned before in DP workshops:
Assuming your main time format is bars and beats, hit the dot key on your keypad, then type in the bar number you
want to locate to.
Hit the dot again and type in the beat number. For real precision, hit it again and type the tick number.
Press the Enter key to get DP to locate to the point you just specified.
You don’t have to define bar, beat and tick: hit the enter key after the bar number if you want. It works just as well with hours, minutes and seconds,
if your time format is SMPTE or Real Time. If you have a laptop or similar keyboard without a keypad, you can either use its ‘embedded’ keypad (if
there is one) and the return key, or hit Command-T, then start typing in your location, using the tab key to move to the next time subdivision, then
hit Return. The dot trick works whether DP is in playback or stopped.
Now for some other location tips. There’s a Marker menu near the top left of the Tracks Window, and at
bottom right of the Control Panel time display in DP 6.02 or later. These give you single-click access to
The keypad ‘dot trick’ or Command-T
various points in your sequence — always the beginning and end, plus the Memory and Auto-Record
keystroke allows you to quickly locate to
start points, when these features are enabled, and any markers. This last fact is handy: take 10 seconds
any point in your sequence, expressed
to drop markers for verses, choruses and other sections in a song and you have single-click access to in the main time format.
them from then on. If keystrokes are more your bag, you can jump one marker left or right at a time using
Photo credit
the Control-Shift-[period] and Control-Shift-[comma] commands.

Selection Process
Selecting something in your sequence is the precursor for many possible operations. You can select
individual data events (such as MIDI notes) or a time range, which could be a fraction of a second or
literally hours long, you can have a selection on one track or on several simultaneously, and there are
many ways of making these selections.
Recent versions of DP6 reinstate this
All Or Nothing: Select All and Deselect All are amongst the most basic selection commands. Select All very useful location tool, the Marker
(Command-A) is useful prior to a Bounce To Disk operation, for example, and I use Deselect All menu.
(Command-D) constantly when editing. For example, if you use the scissors tool on a selected soundbite,
all the slices will become selected. Need to drag just one slice? Hit Command-D to clear the multiple selection, and then drag.
Cherry Picking: Particularly when you’re editing MIDI, soundbites, automation data and so on in the Sequencer or MIDI editors, you’ll need to make
a lot of ‘data’ selections. You’ll know this because when you click the note or soundbite to select it, you’ll get an arrow-style mouse pointer. (Make
‘non-contiguous’ data selections, such as a few MIDI notes not necessarily next to each other, by clicking the first and then shift-clicking the others.)
It’d still be a data selection, too, if you were to click and drag over a group of notes with the cross-hair cursor, or select a single MIDI ‘phrase’ in the
Tracks window. So what’s the big deal about data selections? You won’t notice anything unusual for most familiar editing actions but they do make
a difference to a few things. For example, you’ll find that the Edit menu’s Repeat command, and anything related to it, like Paste Repeat or Merge
Repeat, is greyed out, and so are Snip and Trim, as they all rely on having a defined region: a time-range selection. If you want to use one of these
greyed-out commands on a data selection, there’s a quick solution. Open the Selection Information panel (Studio menu) and in the ‘Set To...’ pop-
up menu choose Set To Selection Bounds. This converts the data selection into a time-range selection and those extra edit actions become
available.
Home On The Range: Time-range selections come into play when you’re making large-scale changes to
one or several tracks, editing audio, selecting everything within a region, getting ready to bounce to disk,
and so on. They can also be trickier to apply, as they often span large regions and dozens of tracks, yet
still require great precision. It can feel particularly awkward when you’re well zoomed-in on a track but
then need to quickly make a large-scale selection — so knowing a range of selection techniques is
handy.

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Effective Selecting & Locating In Digital Performer

For quick, shortish selections on audio tracks, move your mouse pointer to the bottom third of the
Sequencer editor track lane and drag with the cross-hair pointer. Alternatively — for any track type —
drag a selection in a lane holding down the ‘I’ key. This temporarily selects the I-Beam tool, ideal for time-
range selections. With a selection made in one track, Command-click extra track names to include those
tracks in the selection. Drag in the time ruler, with or without I-Beam, to make a time-range selection in all
tracks (except ones hidden in the Track Selector). Again, Command-click track names to include them in
or exclude them from the selection.
Really long selections, especially when you’re zoomed in, need more planning. If you just start to drag
out a selection, it quickly extends past the end of the edit window and you sit there like a lemon as it
leisurely scrolls its way along. So what are better ways of making long selections? It sounds obvious, but
one way is to simply zoom out first. Use Command and the arrow keys to control horizontal and vertical DP6’s Selection Information window lets
you make precise selections
zoom level. Go out as far as you need to, make the selection and, if you like, jump back to your previous
numerically, or, as here, convert data
zoom level by hitting Command-[left square bracket] once or twice. This steps back through the zoom
selections to time-range selections.
‘history’. Another way, good for single tracks, is to locate to where you want the selection to begin and
click (not drag) with the cross-hair cursor or I-Beam tool in the track lane to place the insertion point.
Then locate to where you want your selection to end, and shift-click. This selects everything in between.
In the Selection Information window, you can set selection start and end points (or duration)
The I-Beam tool is the key to a range of
numerically. Just click in the fields and type a value, using the dot or tab keys to switch fields before
powerful selection techniques.
hitting Return or Enter. The arrows to the right of the time fields are actually clickable buttons: click them
to ‘load’ a start or end time, even during playback.
Time Travel: There are times when it’s handy to keep coming back to a particular selection. Maybe you want to make three bounces to disk of a
song, each the same length but requiring different audio and MIDI edits. Editing between bounces blitzes the selection you’ve made for the bounce,
but there’s a way around that. With the selection in place, hit Control-R. This saves it, and to recall it after your edits you just choose Set to
Remembered Times from the Selection Information window’s pop-up menu. The same remembered time can be used to set Memory Cycle or Auto
Record boundaries, using commands from their pop-up menus in the Control Panel.
Miracle Grow: With a selection already made, or the insertion point placed, try these commands, otherwise known as ‘select everything to the left or
right’. For some kinds of editing, they’ll become your best friends:
Shift-Return: Grow selection to start
Option-Shift-Return: Grow selection to end.

The Small Print

Various DP settings have a bearing on making selections.
Snap To Grid status determines whether selections can be made completely freely or, um, snap to grid.
Turn it on or off by clicking the tick box in the Snap Information window or the Edit Grid info bar, or by
holding down the Command key as you click or drag.
To exclude tracks from large-scale selections (such as Select All) made in the Sequence editor, hide
them first in the Track Selector list.
Check the ‘Edit Tracks In Closed Folders’ item in the Tracks window’s mini-menu to control whether
tracks in closed folders can still be selected by ‘large scale’ techniques like Select All, a time-ruler
selection drag or a marker/Memory Cycle click. You might never need this, but now you’ll know what to
do when you do!

DP News In Brief
BPM: MOTU’s newest software instrument, BPM, was announced at the NAMM show, but by now is widely available and has already received
an important update. In case you’re wondering, BPM stands for ‘Beat Production Machine’, and MOTU dub it an “advanced urban rhythm
production instrument”. That, no doubt, relates to its Akai MPC-inspired look and feel.
Just like an MPC, BPM allows samples to be loaded into its 16 virtual pads, and has pattern sequencing and song-construction features, as
well as built-in sampling and re-sampling. But, as you’d expect from software, its capabilities go well beyond what could realistically be offered in
hardware. The pads can have unlimited layers of samples and/or synthesized drum sounds, complicated velocity splits can be set up, and a
wide range of effects can be applied at any level in the sound production architecture. There are also two Racks to host an unlimited number of
REX and other beat-sliced loops, phrases, single hits, and even multisampled instruments from MOTU’s or UVI’s other sound libraries.
Consequently, BPM is far more than just a drum machine: it could conceivably look after entire backing-track production duties.
Aside from all those features and a bundled 15Gb library, what interests me about BPM is its polished, functional and friendly graphical
interface. Also intriguing is how the BPM plug-in uses a small additional application and plug-in to assist with sourcing signals for the sampling
feature. This is, I think, the first time I’ve seen these kinds of audio ‘senders’ used for inter-application and inter-channel routing, and they seem

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Effective Selecting & Locating In Digital Performer

to provide an elegant solution to what has sometimes depended on third-party utilities like Cycling 74’s Soundflower.
Interface Evolution: A large part of MOTU’s Firewire audio interface range — specifically the 828, Traveler and Ultralite — is now in Mk3 guise. If
you’ve got one of these you can already take advantage of their near-zero-latency effects processing, but there’s even more on offer now,
courtesy of an updated CueMix FX application. The EQ can now have an FFT frequency-content display superimposed on the ‘curve’, just like in
the MasterWorks EQ MAS plug-in, and there’s a new Spectrogram — a rolling waterfall display showing frequency content across the spectrum
as differing colour intensity and brightness. There’s even an oscilloscope, perfect for synth nerds trying to track down that perfect Moog
sawtooth. The updated CueMix FX is a free download at www.motu.com.
Volta: Maybe even more ‘out there’ is Volta, the MOTU Audio Unit plug-in that runs in DP and other DAWs and utilises certain electrical
characteristics of many MOTU audio interfaces. The idea is that you patch your MIDI-less synth (or any other gear that talks CV/Gate) into your
interface, then Volta does the MIDI-to-CV conversion required to communicate with it. So you can sequence your modular monster from a MIDI
track, with the same timing accuracy as a virtual instrument, and bring its audio output back into the DAW mixer if you wish. Volta’s calibration
features can cope with less common synth electrical standards as well as more mainstream ones, and can build accurate pitch-tracking maps to
keep synths in tune even if their CV response isn’t quite linear. It should be available very soon, so warm up your oscillators, ladies and
gentlemen.

Published in SOS June 2009

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Exploring Sonar 8’s Dimension Pro Synthesizer

Exploring Sonar 8’s Dimension Pro Synthesizer

Sonar Notes & Techniques Buy PDF
Published in SOS June 2009
Technique : Sonar Notes
Printer-friendly version
Now that Dimension Pro is included with Sonar 8, it’s time to investigate how to exploit it
beyond the usual sample-playback functions.
Craig Anderton

D imension Pro was Cakewalk’s first instrument made for the Mac as well as Windows, and
the first instrument to be available separately, and not just bundled with other Cakewalk
products. Now the full version comes as standard with Sonar 8, which not only adds value
to the Sonar package, but will surely introduce a new group of users to this very useful
instrument.
Dimension Pro is basically a sample-playback engine with a decent amount of synthesis
flexibility. It has four ‘Elements’, which can be split and layered, and although it’s multitimbral, it’s
only four-part multitimbral, so it’s not exactly a 16-channel ‘workstation’ like IK’s SampleTank or You can point Dimension Pro to a
NI’s Kontakt. On the other hand, its CPU efficiency is excellent, so opening up multiple instances different Multisamples folder using the
is generally not a problem. registry. The numbers superimposed on
the screen shot correspond to the steps
Using The Internal Audio Input listed below.

But, you may be saying, Dimension Pro has no audio input — and you’re quite right, which is too bad, because the synth’s LoFi, Filter, and
Drive effects are just begging to abuse some unsuspecting waveform. However, there is a workaround. Dimension Pro’s oscillator can
accept long WAV or AIFF files, even entire tracks, and the files can be mono or stereo. I’ve yet to find a limit to how big a file it can accept.
To process a Sonar track with Dimension Pro’s facilities, proceed as follows:
1. If the audio track you want to process consists of multiple clips, first bounce them together into a single track. If the first clip doesn’t start
at the beginning of the project, extend its beginning (click the clip’s left edge, and drag to the left). Doing this makes an easily-defined start
point. Select all clips, then go Edit > Bounce to Clip(s). All the clips are now one long track.
2. Choose the Dimension Pro Element you want to load the track into, then right-click on its Element-select button. Choose Reset Element
to start off with a clean slate.
3. Drag the clip you created in step one from the track view into the synth’s Load Multisample window.
4. Enter a C5 note in the MIDI track driving Dimension Pro, and extend it for the length of the track.
Now, when you start playback, Dimension Pro will play back the track (make sure the original track is muted), and you can go nuts with the
processing. However, there is an extra point to bear in mind: to use the DSP processing (anything that’s not an effect), you need to enable
any processing you want to use before you begin playback. If you want to turn on, say, LoFi or Drive in the middle of playback, you have to
re-trigger the note first. As long as the desired processors are ‘on’, though, you can automate them and any control manipulation will play
back properly as automation data. The effect options (chorus, delay, and so on) can be turned on at any time.
As a variation on this kind of processing, here’s how to get a nice, organic flanging sound. The principle is the same as the previous
example. First, reset two Dimension Pro Elements, and drag the same track into each Element. In one of the Elements, click on the Pitch
button and set the LFO status to On. Start with an LFO frequency of around 0.15 and an LFO depth of around 20, then adjust for the
desired flanging effect. Try a triangle or sine wave as the LFO waveform. If you want to make the flanging effect permanent, bounce the
Dimension Pro output to an audio track.

Samples, Wavetables & Grooves

The oscillator in a Dimension Pro Element has two main playback modes. In my previous example of using a Sonar track as the sound
source, Dimension Pro’s oscillator serves as a standard sample-playback engine for a WAV or AIFF file. However, you can also load an
SFZ file into an Element. The open-standard SFZ file format (which is a simple text file that ‘points’ to samples) defines how multisamples
should be played with respect to split points, layers, tuning, level, and so on. The Sonar workshop in the April 2008 issue of SOS gives a
full description of SFZ files and how to create them.
A Dimension Pro oscillator can also play back wavetables, single-cycle audio files that are shorter than 4000 samples. When you load this
kind of short sample, Dimension Pro creates all the transpositions necessary to play that single cycle back over the entire keyboard range,
and does so with minimal aliasing. Wavetables are ideal for emulating analogue synthesis, and early digital synthesizers such as the PPG
Wave and Ensoniq SQ80.
The two types of oscillators mentioned above create Sound programs, but Dimension Pro also comes with Groove programs, which are
loops that can be time-stretched and transposed. They work in the same way as REX files, which we’ll cover next.

More REX Education

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Exploring Sonar 8’s Dimension Pro Synthesizer

Sonar 8 isn’t short of ways to play back REX files, but Dimension Pro is particularly good at the job. To play a REX file, first reset an
Element, then drag the REX file into the Load Multisample window. A Note symbol (an eighth note tied with a 16th note) appears toward the
right of the window. You can drag this into a MIDI track driving Dimension Pro to trigger slices of the REX file, as well as turning it into a
MIDI Groove Clip so you can ‘roll out’ as many iterations as you want. On your keyboard, C3 plays back the entire REX file. If you lift your
finger off the key the file stops playing, while if you hold your finger on the key, the file will play through to the end (it doesn’t keep looping).
You can transpose the file pitch, with the tempo staying consistent, over the range of C2 to B3. Starting upward from C4, the keyboard keys
play individual REX slices. You can also transpose the REX file’s ‘root’ note using Dimension Pro’s Transpose parameter.
Any DSP processing (LoFi, Filter, Drive, EQ, and so on) and effects will affect the entire file, but envelopes affect each individual slice,
which is pretty useful. For example, you can use the Amplitude envelope to set a very short, percussive decay for each slice. A very cool
aspect of using Dimension Pro as a REX file player is that you can take advantage of its multitimbral mode to assign Elements 1-4 to MIDI
channels 1-4. To do this, click on the Options button and select Set Program as Multitimbral.
Now you can load separate REX files into each Element, drag each Element’s Note symbol icon
to its own MIDI track, and run four REX files at once — an entire rhythm section, for example,
with drums, percussion, bass and a guitar riff. And, of course, you can automate volume, panning,
and so on with Dimension Pro, so that you can do a ‘remix’ of the REX files. It’s also worth noting
that you can edit the MIDI files to jumble up the slices in various ways, and that moving the mod
wheel reverses the individual slices, for ‘backwards audio’ effects.
Dimension Pro is in multitimbral mode,
Multisampling and each Element contains a REX file.
In Sonar, four MIDI tracks containing
As mentioned earlier, Dimension Pro’s native method for handling multisamples is the SFZ file the MIDI components of the REX files
format. However, for some people the process of creating an SFZ file is intimidating. As an drive Dimension Pro channels 1-4. And
alternative, you can use Dimension Pro’s four elements to create up to a four-way multisample. I yes, I did load a Dr: REX track icon into
should add here that the synth’s audio engine is something special. I find I can often stretch a the MIDI tracks!
single sample over the full keyboard range and subjectively it sounds fine, though maybe not as
accurate as a multisampled instrument, where the high notes have different timbres than the low ones.
As an example, I sampled a Chapman Stick at four pitches (D3, E4, F#5, and F#6), then looped them (in Sound Forge). Here’s how you
would use those samples to create a multisample within Dimension Pro.
1. Drag each sample into its own Element.
2. Edit each Element’s Shift parameter so that each sample plays back at the correct pitch when you play a keyboard key. For example, the
shift amounts for D3, E4, F#5, and F#6 are 22, 8, -6, and 18 respectively. Once shifted, if you play a key, you should hear all Elements
playing at the same pitch.
3. Now let’s map them. The object is to set split points using each Element’s Lo/Hi Key parameter, so that each sample covers a particular
range of the keyboard.
Root Note Lo Key Hi Key
D3 0 44
E4 45 59
F#5 60 72
F#6 73 127
The above is the ‘classic’ way to do mapping, but another option is to double samples over particular ranges. This does two things: thickens
the sound (a bit of detuning adds some nice chorusing) and evens out timbral variations. Here’s the mapping you’d use to create this effect.
Root Note Lo Key Hi Key
D3 0 44
E4 0 59
F#5 45 127
F#6 60 127
Once the samples are mapped, you can make tweaks, such as fine-tuning if the samples are
slightly out of tune, using keyboard tracking on the filter to alter high-frequency response (for
example, less treble as you play higher up on the keyboard), and so on. Another nice trick that
you can use with multisampling is adding separate amounts of reverb to the different samples —
light reverb on the lower samples to avoid muddiness, for example, and more reverb on the
higher samples. This shows mapping in a graphical
You can also use either of the above mapping methods with the SFZ format, and load that into format. The top mapping corresponds to
the classic mapping in step three of my
a single Element. However, you then lose the flexibility to process each sample individually.
explanation, left, while the lower
mapping corresponds to the ‘doubled’
Moving The Library mapping explained afterwards.
Dimension Pro has a good-sized library folder of Multisamples, amounting to around 9GB. This

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Exploring Sonar 8’s Dimension Pro Synthesizer

folder location defaults to C:/Program Files/Cakewalk/Dimension Pro, but I prefer to keep large sample libraries off my root drive. I installed
a TeraByte eSATA drive inside my computer to dedicate solely to samples, and I wanted to put the Dimension Pro library there as well.
Dimension Pro has a registry entry that tells it where to look for the Multisamples folder. If you’re worried by the idea of modifying the
registry, don’t proceed — or at least back it up before doing it, because if you screw it up your computer could be in trouble. But bear in
mind that the registry isn’t really that scary a place if you take your time and double-check your work. Here’s how to modify it.
1. First, Go Start > Run and type Regedit in the Run field.
2. In the left pane, unfold the HKEY_Local_Machine key.
3. Unfold the Software key under HKEY_Local_Machine.
4. Unfold the Cakewalk Music Software key under ‘Software’.
5. Click on the Dimension Pro folder and several data types appear in the right pane.
6. Right-click on the Multisamples Folder and select Modify.
7. Enter the new file path for the Multisamples folder, then click on OK.
8. Close the Registry.
To make sure all went smoothly, open Dimension Pro to verify that you can load its programs.

Keyboard Shortcuts
If you click on the ‘Give All Keystrokes to Plug-In’ button in Dimension Pro’s upper right corner, then click on any Element button, you
can select the other Elements with the QWERTY keyboard number keys (1 = Element 1, 2 = Element 2, and so on). Also, if you click on
one of the Modulation buttons, the number keys can be used to switch between the various modulation options (1 = Pitch, 2 = Cutoff, 3 =
Resonance, 4 = Pan, 5 = Amp). Note that the numeric keypad number keys do not work for this.

Published in SOS June 2009

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Getting Into Pro Tools

Getting Into Pro Tools

Part 2: Installation & Session Basics Buy PDF
Published in SOS June 2009
Technique : Pro Tools Notes
Printer-friendly version
Our short series on getting started with Pro Tools continues with some advice on configuring
the software and your first Sessions.
Mike Thornton

L ast month, we looked at the hardware choices that are available when buying a Pro Tools system. Having guided you through that
particular maze, this month we offer advice on getting the software side of the system up and running smoothly.

Configuring Your Computer

Digidesign make some fairly detailed recommendations about how to set up your Mac or PC to work well with Pro Tools: the
Troubleshooting guides on the Digidesign User Conference (http://duc.digidesign.com/showthread.php?t=240754) give detailed guidance
about what doesn’t play well with Pro Tools on both platforms.
A few of the more obvious recommendations include turning off automatic OS updates. This is because Pro Tools isn’t immediately
compatible with the latest version of the Mac OS, so don’t upgrade your operating system until Digidesign have approved it, if you want a
quiet life! On Mac OS X, the default keyboard shortcuts conflict with Pro Tools shortcuts, so go to System Preferences and turn off all the
shortcuts in both Dashboard and Exposé. Likewise, in Spotlight you need to turn off the two shortcuts at the bottom of the window, as they
conflict with the Record shortcuts in Pro Tools. While in System Preferences, turn off Energy Saver too — Pro Tools really doesn’t like the
hard drive to spin down whilst it’s running!

Authorising Pro Tools

The first time you run Pro Tools LE, you will asked to enter your serial number. You will find this on a piece of Digidesign paperwork, usually
inside the front cover of the ‘What’s new’ documentation, and it will start with ‘DIGI0800 xxxxxxx xxxxxxx’ where the x’s are a series of
numbers and lower-case letters.
The Pro Tools LE software won’t run without a Digidesign interface plugged into your computer, making the interface effectively a dongle.
Pro Tools M-Powered and HD systems, by contrast, are authorised using an iLok key. This is a
USB device that holds licences for some Pro Tools software and plug-ins. It is a multi-platform
device and can hold up to 100 licences. You can have software installed on a number of
machines but it will only run on the system you have the iLok plugged into.
Most plug-in manufacturers use iLok to protect their software from piracy, so although you might
not need an iLok to start with, it won’t be very long before you do. Once you have any software
that uses iLok, you will need to register and open an account with iLok.com. This account keeps
track of your licences (‘assets’); when you buy plug-ins on-line, the suppliers will often deposit
your assets into your iLok account and then you transfer them onto your iLok.
Two final tips on iLoks: if you own any Waves plug-ins, it is best to keep their iLok licences on a
separate iLok from other manufacturers, as they work differently when it comes to resolving
problems with a broken or lost iLok. Second, make sure your iLok is covered by your insurance,
as if it is lost or stolen, you will need to claim on your insurance to replace the lost software. For
more details on iLoks and their use, and what to do when things go wrong, take a look at the Pro
Tools workshops in the August 2008 (www.soundonsound.com/sos/aug08/articles/pt_0808.htm)
and November 2005 (www.soundonsound.com/sos/nov05/articles/ptworkshop.htm) issues of
Sound On Sound.

Creating A New Session

Before you can make any music in Pro Tools, you’ll need to create a project — or a Session, as
they’re known. Pro Tools 8 gives you the option to do this at start-up, or you can choose New
Session from the File menu.
Pro Tools M-Powered and HD systems
When creating a new Session, especially in version 7 and earlier, it is easy to just give it a name
require an iLok key, as do most third-
and click the Save button without filling in the Session Parameters. Since these include the party Pro Tools plug-ins.
location of the Session on disk, this often leads to the cry “I can’t find my Session!” If these
settings aren’t correct, you won’t notice until much further down the road, when fixing it could take
quite a while, so take my advice and spend a little time here. Happily, in Pro Tools 8 the Session Parameters are harder to miss, as once
you have selected New Session from the File menu you are presented with this window. Let’s take a look at the various options that are
available here.

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Getting Into Pro Tools

The Audio File Type determines the file type for all audio files created by Pro
Tools for this Session. These days I can see little if any justification for using
any file type other than BWF (Wave files).
Moving on to the Sample Rate drop-down menu, the hardware you have will
determine what options you are offered here. The screen shows what you get
on an HD rig with a 192 I/O attached. Some LE systems will offer a range of
sample rates from 44.1 to 96 kHz, while others are limited to 44.1 or 48 kHz.
Don’t automatically choose the highest number here! Depending on the rest
of your signal chain, you may get higher quality at higher sample rates, but
you will produce larger audio files and put more load on the computer. It’s
also worth bearing in mind the final format you’re working towards: for music
or radio project, 44.1kHz or one of its multiples is usually best, but if you are
working on a TV project, choose 48kHz or one of its multiples. If you don’t pay attention to your
Session Parameters, you could be
Bit Depth, like sample rate is a quality versus performance trade-off; 24-bit
storing up trouble.
files can represent a greater dynamic range, but will be 50 percent larger
than their 16-bit counterparts. Again, you pays your money and takes your
choice.
Pro Tools will offer a range of default I/O settings, and you can create your own from the I/O Setup window by
selecting ‘I/O...’ from the Setup menu. For much more detail on configuring your own I/O setups, see the Pro
Tools workshop article in the August 2007 issue of SOS
(www.soundonsound.com/sos/aug07/articles/ptworkshop_0807.htm).
Once you have made your choices, in Pro Tools 8 you can click OK and the system will offer you the usual navigation box to select where
on your system you want to save your project. Remember that although you can save Pro Tools Sessions on your ‘C’ drive or Macintosh
HD drive, it isn’t recommended: an external audio drive is the best location for your Session.

The Edit & Mix Windows

Now you have created a new Session, you will get two mostly blank panes, which are the Edit and Mix windows. These are the two main
windows in Pro Tools, but before you can do anything with them, you need to create at least one
track; hit Shift+Command+N (Windows: Shift+Ctrl+N) or select New from the Track menu. From
the window that appears, you can choose your track types and how many of each you want. This
isn’t a ‘once in a lifetime’ choice; you can go back into this option at any point and add more
tracks as you need them. Now you will see that the Edit and Mix windows look a little more
populated. There are many options as to what is displayed or hidden in each of these windows —
visit the View menu to explore the possibilities.
An Edit window with two newly created
Across the top of the Edit window is the main toolbar. From left, this contains the Shuffle, Slip,
audio tracks. Above these is the
Spot and Grid buttons, which determine how you can move material (‘Regions’) around in the Edit Toolbar.
window. Next are the Zoom settings, then the main Tool selection — more on what each of these
does in a minute. After this is the Counter section, which also will display the start, end and
duration of any selected Region(s). Then come the Grid and Nudge settings. When Grid edit
mode is selected, regions will ‘snap’ to grid lines, and the grid itself can be set up so that the lines
fall on the beat for music, or on the frame boundaries for video post production. The Nudge setting
enables you to move Regions forward or back by the set amount displayed in one of the full
variety of timeline (ruler) options. Finally comes the Transport section, with the usual transport
function buttons, most of which duplicate what’s available in the separate Transport window.
Before you actually record anything, you’ll need to select the correct input for your track, and
click the ‘R’ button to record-enable it. It’s important to name the tracks (double-click on the track
name to do so), as Pro Tools derives its default file names from the track name, so if you don’t
name them, you’ll end up with hundreds of files called things like ‘Audio 1_01-152’. If you do want
to rename a file, select the Grabber tool and double-click the Region either in the track or in the
Region List (the fold-out panel at the right of the Edit window). Always choose to rename both the
region and disk file, so that the file in the Audio Files folder gets renamed as well. This will help
you when you want to borrow files from another Session.

Be Smart
I promised to talk more about the Tools pane. You can find details of what each tool does in the
Pro Tools documentation, and I recommend that you start to use the Smart Tool straight away.
You will find routine editing and manipulating in Pro Tools much easier and quicker.

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Getting Into Pro Tools

The Smart Tool is a composite tool that changes its function depending on whereabouts you
hover the mouse. To activate the Smart Tool, click on the linking bar above the three main tool
buttons in the toolbar, or hit F6 and F7 simultaneously. The Smart Tool becomes the Selector (I-
beam) tool when you hover it above the centre line of the track you are working on. When the
cursor is below the centre line of your track, you will see it change to a ‘hand’ — the Grabber tool.
Place the cursor near a Region boundary near the middle of the track and you will see that it The same two tracks as they appear in
changes to the ‘sideways staple’ Trim tool, allowing you to drag the Region to make it longer or the Pro Tools 8 Mix window.
shorter.
However, we are not done with the Smart Tool yet! If you place the cursor near the beginning or
end of a Region and near the top of the track, you will see that it changes to a little box with a
diagonal line. Click, hold and drag the mouse, and you will create a fade on the Region. And there
is still more: if you place the cursor near a Region boundary between two Regions near the
bottom of the track, Pro Tools will offer you the option to create a crossfade: again, click, hold and
drag the mouse out to create a crossfade of the desired duration. The shape of all the fades is set
by the default fades settings in the Editing tab of the Preferences, which you can get to from either
the Setup or the Pro Tools menus in Pro Tools. (In general, the default Preferences settings will By default, files that you record are
enable you to get going with Pro Tools, but sooner or later you’ll want to take a detailed look at named after the track they’re recorded
your Preferences settings, as described in our March 2007 Pro Tools workshop: on to.
www.soundonsound.com/sos/mar07/articles/ptworkshop_0307.htm).

Conclusion
Next month, we will continue this series with some more detailed trips into different parts of Pro
Tools that you will need to come to terms with as you use Pro Tools more and more. In the
meantime, there is loads of advice and a huge amount of support from Digidesign and users like
you and me on the Digidesign User Conference (DUC) at http://duc.digidesign.com. There are
‘Stickies’ at the top of each section of the DUC, which are a mine of information. If you want a
quiet, no-hassle life with Pro Tools, I recommend you read and follow the advice.

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Getting Into Pro Tools

Fail to name your tracks, and your

Region List could look like this!

The Smart Tool morphs between

several different functions depending on
where the mouse is in relation to your
target Region.

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Getting Into Pro Tools

Get Your Levels Right

When recording audio into Pro Tools, it is worth taking a little time to make sure you are recording at the optimum level. With digital
recording, you need to make sure you leave enough headroom to cover the unexpected peaks, without the overall level being too low.
After all, you don’t want to be in the horrible position of telling your client that the superb take they have done is in the bin because it is
distorted!
This is something that needs to be set on the recording hardware, as Pro Tools software doesn’t have any record level controls — the
fader only acts as a monitor level control, even though it turns red in record mode. On an HD rig, you will need to adjust the level coming
from your mixer or mic preamps, while most LE hardware has record level controls on the interface that you can adjust. You are, ideally,
looking to have the Pro Tools meters in the lower part of the yellow section but leaving some headroom, so that the occasional
unforeseen peak does not light the Peak lights and distort.

Published in SOS June 2009

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Mix Rescue: Adam Bevan

Mix Rescue: Adam Bevan

Sound Workshop
Published in SOS June 2009
Technique : Recording/Mixing
Printer-friendly version
Beefing up bass parts played on guitar and importing GarageBand songs into Logic are among the
challenges for Mix Rescue, as we add some spice to Adam Bevan’s mix.

Paul White

A dam Bevan is a university student (no, not music technology!) who records his own songs at home, using a
very modest system based around GarageBand and running on a Mac laptop. His song ‘Deserts’ comprises
mainly ‘real’ instruments and voices (all played and sung by Adam), but the rhythm comes courtesy of Apple
Loops drum samples.

The Raw Material

The version of the song Adam gave me was imaginatively structured, and included a processed but clean-sounding
electric guitar solo over a rhythm groove during the intro, some fairly gritty rhythm guitars during the main sections,
and a very effective chorus — where everything drops out other than the drums and an acoustic guitar, allowing the
three-part vocals to come through strongly. Adam hadn’t bought a bass guitar at the time he made this recording, so
he used his electric guitar and then tried to process it to make it sound more like bass, using GarageBand’s Guitar
Amp plug-in and some EQ. Although tonally successful, the resulting sound inevitably lacked the low-frequency
punch of a real bass.
The song also had the potential for strong dynamics, with extra rhythm layers and additional guitar parts coming in
towards the end, but the mix itself was starting to get muddy. Adam had used distortion plug-ins on some of his vocal
parts to create an obviously effected sound, but this also detracted from their clarity, and covered up the fact that he
really has a very good voice. He had spread his vocal parts over several tracks so that he could apply different
effects to the different parts, and he’d also tried the same trick with the guitars, to introduce some tonal variety
throughout the song. Part of his difficulty in creating a strong but uncluttered mix was that many of the sounds
This month’s Mix Rescuee, Adam
occupied the same part of the audio spectrum, so that the fuzzy guitars, treated ‘fake’ bass and fuzzy vocals tended Bevan.
to blur into one.

Import-ant
Before working on the song, I imported it into Logic Pro 8. It’s not something I’ve needed to do before, but the process turned out to be pretty straightforward.
Still, I did find that there are a couple of things that you need to watch out for...
Adam first brought me the mixes on a Flash drive, but he hadn’t checked the ‘Save Assets’ option when saving the GarageBand project, which meant that if I
didn’t have the same Apple Loops as he did, they wouldn’t play back. Furthermore, for some reason neither of us could fathom, every mix that we imported had
all the channels routed to bus 11, which was then routed to the stereo mix. They played without problem, of course, but it was only a matter of moments to
redirect the channels to the main stereo mix output.
Prior to starting the ‘real’ work, it was necessary to trim the various audio regions, especially the distorted guitar parts, to ensure that there was no hum or
buzz preceding them, and I also used Logic’s fade tool to fade out the ends, ensuring a smooth transition to silence.
Adam had used aux sends correctly, and had set one up to feed an instance of Platinumverb, and another to feed a delay effect. Other than swapping
Platinumverb for a short, bright EMT plate (courtesy of Univeral Audio’s Plate 140 UAD plug-in), I left these as they were, adjusting only the levels.
Curiously, any plug-ins used within GarageBand show up as a plain panel of controls in Logic, rather than the more friendly Edit view, so all these had to be
switched over manually. Another curiosity was that when switching on the low-pass filter in the channel EQ, the frequency always defaulted to 20Hz — rather
than the more logical 20kHz — so that had to be reset manually as well.
These were all very minor irritations, but rather more serious was that the audio files imported from GarageBand didn’t seem to allow the region-enlarging
function to be used in Logic’s Arrange Page. To get around this I found that I had to do an unnecessary edit, such as adding a bit of silence to the end of the
audio taken from elsewhere, then using the glue tool to create a new file. The new Logic-format audio file that was created as a result could then be extended
as normal, where required.

Arrangement Tweaks
Before fine-tuning the sounds, I looked at the song structure and identified a repetitive instrumental section that felt as though it went on a bit too long without
developing. Some of the breaks also sounded a bit unfinished, partly because Adam had simply used level automation to stop the drum loops from playing
when he needed a break, but there were no ending or starting drum fills to create the illusion of a real performance. Often, all that was needed was an extra
pasted beat with a cymbal crash layered on top, but in other areas more sophisticated solutions had to be found.
I shortened the rather repetitive middle instrumental section by four bars, and also trimmed the end of the song, where I felt the solo guitar outro went on a bit
too long before fading out. Adam had sung a long, sustained note over the first half of the instrumental section, and I stretched the tail end of it quite
considerably, to extend over the entire length of the section, gradually fading it out and also putting in some automated panning.
I also took the liberty of reshaping the two-bar break after the solo, using chopped-up pieces of Adam’s original guitar parts, including two heavily stretched
guitar sounds that now sound more like a car skid and crash! There’s a limit to how far Logic allows you to stretch a sound in one go, so I repeated the
process two or three times to move the sound as far away from reality as possible. One short chord became a drawn-out screech — but, crucially, it still
retained its pitch. Those stretched guitar sounds were dropped in again towards the end of the song, just as the solo’ed rotary guitar outro takes over, giving
more of a sense of finality.
During the first half of the break I also alternated two very short sections of guitar (less than one beat long and
pasted four to the bar), copied from Adam’s previous parts, then panned them left and right, fading their levels over

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Mix Rescue: Adam Bevan

the course of two bars. An automated low-pass filter was also used on these same chordal stabs, to progressively
dull down the sound as it repeated. To add interest and impact, I brought the song back in after the break using a trio
of very short guitar clips (again, copied from earlier guitar tracks) but edited to sound like stabs or stutters, followed
by a simple snare-drum fill to get back into the rhythm loop that was about to start up. Adam’s solo bass part during
the break also didn’t seem to fit as well as it might, so I opted to take just half a bar of his bass part and loop it four
times.
Adam also hadn’t used any panning, because he was unsure what approach to use, so I decided to try to get the In this screen you can see how Paul
mix sounding good in mono first, before using panning to separate the layered guitar parts and layered vocals. Now it reworked one of the break sections,
was time to work on the sounds themselves. copying a short guitar part and ‘multing’
it to different tracks, panned left and
Guitar Sounds right to generate more interest.

I noticed that although Adam had inserted compressor presets on various guitar tracks, he hadn’t adjusted the
threshold, which meant that on some of them there was no gain-reduction taking place at all. Heavily distorted guitars tend not to need compression anyway,
so I simply removed any unnecessary compressors.
Adam wanted an aggressive, Slipknot-style guitar tone, but had DI’d the guitar and was not actually playing in a very aggressive way, so getting the
necessary angst in there proved to be a bit of a challenge. For his main guitar part, I took out the Guitar Amp Pro plug-in and instead tried Line 6’s Pod Farm,
running a Bomber XTC model with some EQ lift at 1.2 kHz and 100Hz. I also fell back on my old trick of using a
tremolo plug-in, adjusted to work as a chopper (sixteenths), to add a sense of rhythm to one of the guitar parts, but
then adjusted the modulation depth so that the effect was a little more subtle than usual. As Adam wanted the guitar
to be reasonably loud, I also configured a noise-gate plug-in to work as a ducker triggered from the main vocal, so
that the level would drop by 2dB when Adam was singing. This wasn’t obtrusive in the finished mix but it did give the
vocals a bit more room to breathe.
For another heavy guitar part I tried a different Pod Farm amp, based on a well-known British stack, in conjunction
with a very mild dose of Logic’s Phase Distortion plug-in patched before it, which helped to get a ‘nastier’, more
aggressive sound, with some sense of ‘death metal’ intermodulation and grind.
In general, Adam had been tempted to add quite high levels of distortion, but this can actually detract from the
impact of the sound if you go too far, and all those extra harmonics spread right across the audio spectrum made it
hard to create any separation between the guitars. The distorted guitar tracks responded well to some ‘bracketing’
EQ using 18 or 24dB/octave high- and low-cut filters to remove the fizzy high-end and boomy low-end. In Logic’s
Channel EQ, the low-pass filter also has a resonance control, which was useful in peaking up the upper mid-range of
the guitar before the cut came in as, in conjunction with some gentle EQ boost in the 3kHz region, it helped enhance
the ‘bite’ part of the guitar spectrum.
The acoustic guitar part was treated using the Waves Maserati Acoustic Guitar plug-in, followed by some extra
compression to thin out the sound and create some ambience, while the chorus electric guitar that brings the song in,
and again reappears at the end, was changed to a slow rotary speaker effect.

Vox & Variety

One thing I learned from Adam’s mix was that he likes variety, and he doesn’t like the voices or guitars to sound the
same all the way through the song. He’d tried various tricks, such as changing the amount of reverb or delay, but
he’d also experimented again with vocal distortion. As well as passing the guitar sound
Simple distortion plug-ins tend to make vocal parts sound fuzzy and rather messy, so I tried SPL’s Twin Tube plug- through some better amp-modelling
in on the main vocal, and added just a bit of thickness to it that way. This gave a tube-like warmth and a bit of growly plug-ins, and using a bracket of high
distortion — but without the fizz. Prior to the Twin Tube I used a Universal Audio LA2A compressor to level the vocal and low-pass filters, Paul used Logic’s
inbuilt Tremolo plug-in (above), set to
and give it a bit of punch, and a Waves Maserati Vocal plug-in (which combines compression, EQ and delay effects)
act as a chopper, to give a much
gave the part more of an up-front sound. Because this plug-in can add both delay and reverb, no additional stronger rhythmic emphasis to some of
ambience effects were required. the parts.
As Adam wanted a double-tracked kind of sound, I first dusted off my old ADT trick, using Logic’s Platinumverb,
and following it with an instance of the Antares AVOX doubler to further thicken the ADT effect. Platinumverb has a
balance control for the reverb tail and the early reflections. If you set this to 100 percent early reflections and then
put in a pre-delay of 90 to 100ms, you get the makings of a good doubling effect. Setting the reverb parameters for
bright and short, with low diffusion and density settings, makes the doubled sound more distinct, and in this case I
used only 14 percent wet, 76 percent dry to get the required balance. (On other platforms, any synthetic reverb plug-
in that lets you adjust the ER and tail balance should be able to create a similar effect.) This came close to the
doubled sound Adam was looking for. One benefit of the Antares plug-in is that the two voices it generates have
slight and random pitch and timing variations and can be independently panned. Panning these hard left and right
gave the main vocal a more believable doubled sound and added some dimensional interest to the mix.
The next vocal layer was treated with just a small amount of Logic’s Overdrive plug-in after being compressed,
then Echo was added as an effect, with the delays sync’ed to sixteenths, and the effect level at around 25 percent.
This chain was set up by Adam, so other than trading his distortion plug-in for Logic’s Overdrive and tweaking the
settings, it ended up pretty much as he originally had it, but — as with other parts — less obviously distorted. Adam
had laid down a further vocal part, using the Vocal Transformer in GarageBand to simulate a female backing line
during the choruses. While not entirely convincing in isolation, this actually worked very well in context, so I kept it
and just adjusted the level balance so that it sat under the other two vocal parts.

Guitar Bass Or Bass Guitar?

As I explained earlier, the bass line was played on a regular electric guitar, and then treated using the GarageBand
Bass Amp plug-in and EQ. I didn’t change the basic processing chain much, but added a send to feed a Waves pitch-
shift plug-in to add some sub-octave content, and also patched in a Waves Maserati Bass plug-in (set to Synth
mode) to shape the basic sound, making it more solid, and adding gentle modulation, to create what felt almost like a More appropriate vocal distortion was
‘synth bass with pulse-width modulation’ effect: it would be impossible to replicate the exact sound of a bass guitar, added using SPL’s TwinTube plug-in.
so I thought we might as well make a feature of the bass sound. It was also necessary to roll off the high end from
the sub signal above 150Hz or so, to add depth without it sounding like an obvious added octave effect, and the sub

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Mix Rescue: Adam Bevan

level had to be adjusted carefully, so that it just filled in that missing lower octave without being allowed to dominate.

Drums & Stereo Width

For drums, Adam had mainly used Apple loops, augmented by the occasional sampled cymbal. One of the loops
used towards the end of the song has something of a hip-hop sound to it, and I enhanced this by using SPL’s
Transient Designer to shorten the decay time, imparting more of a beatbox effect. Apple Loops usually sound pretty
good as they come, and this was no exception, so no other processing was necessary. All the drum parts were based
on stereo loops that could be panned to the centre, as could the bass and the main vocal. Where two or more guitar
parts were playing together, panning was used to separate them, while maintaining reasonable symmetry in the mix,
and the same thing was done with the backing vocals. In combination with the rather more obvious stereo effects in
the break section, this was enough to give the track all the width it needed.
Paul used a combination of Antares’
Fairy Dust & Polish Doubler and Logic’s Platinumverb to
create an ADT effect on Adam’s vocal.
As Adam was compiling his own album of songs, to include ‘Deserts’, I used Logic’s Multipressor on the final mix,
followed by Logic’s Limiter, to create a sense of loudness and energy. Very low compression ratios were used in the
four bands (less than 1.2:1), and the thresholds were set at -20dB so that most of the song’s dynamic range was
gently squeezed, rather than just the top few dBs being stamped on. The relative levels of the middle two bands were
also dropped slightly, to create a hint of a smile curve, which gave the mix a bit more punch and sparkle. The limiter
was set to catch only the occasional 2-3dB spike, so the mix didn’t sound over-processed.
I went back over the song with these processors in place and fine-tuned the levels, adding a little level automation
on some of the vocal and guitar parts where necessary. Any added reverb was kept fairly short and bright to prevent
it from ‘washing out’ the sound, and then I checked the overall balance from the next room, made some final tweaks
and hit Bounce.
With a song like this one, you could continue making changes and refinements almost indefinitely, and some
engineers might spend several hours fine-tuning the vocals, but I felt I’d reached the point where further changes
‘wouldn’t affect sales’, and on the whole I was quite happy with the way the song had tightened up, with much stronger sounding vocals, less clutter and a
generally lively vibe. Of course I didn’t know at this point what Adam would make of my meddling!

Remix Reaction
Adam: “The first thing I noticed about Paul’s mix was the introduction of the new ‘evil’ bass guitar sound, which immediately gave the track a heavier and
darker feel — which is what I was after to begin with. It also meant that the main guitar could be lowered in level to allow the vocals and drums more space.
Paul added a tempo-locked tremolo to the guitar part, which gave a nice feel, and I was very pleased with that.
“I also noticed a dramatic improvement in the vocal sound, and the guitar-amp hiss that had been obvious on the original mixes had gone. I felt there was
a significant improvement in the overall sound, as in my mix the sounds had tended to blur into each other. Now the mix had more punch and definition.
“My original mixes lacked any sense of stereo width, as I wasn’t sure about how best to use panning, and Paul has remedied this by panning my
additional guitar and vocal parts. There isn’t really anything I dislike about the new mix, and I think the new break section works really well, although I might
have left more distortion on the voice where I sustain a long sung note over the instrumental section. However, the trick Paul did, stretching it so that it
covered the entire section, was excellent — I loved it. In all, I absolutely loved the new mix of ‘Deserts’ and am grateful to Paul for explaining his mixing and
mastering techniques to me.”
www.myspace.com/devilswithcleanfaces

Hear The Changes On-line!

We’ve placed ‘before and after’ audio files on the SOS web site so that you can download them, audition them in your DAW, and hear for yourself the
changes Paul made to Adam’s mix. Go to www.soundonsound.com/sos/jun09/articles/mixrescueaudio.htm for more details.

Published in SOS June 2009

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Mix Rescue: Adam Bevan | Audio Files

Mix Rescue: Adam Bevan | Audio Files

Hear For Yourself
Published in SOS June 2009
Technique : Recording/Mixing
Printer-friendly version
These audio files are presented in both MP3 (for auditioning) and AIF, for more serious
critical listening and comparison in your DAW software.
Full Mix Original MP3 AIF

Adam Bevan's original mix of 'Deserts'.

Full Mix After MP3 AIF

Paul White's remix of 'Deserts'

Acoustic Guitar Original MP3 AIF

This files is the original acoustic guitar part as supplied by Adam.

Acoustic Guitar Processed MP3 AIF

This is the acoustic guitar part as processed in Paul White's mix.

Grunge Guitar Original MP3 AIF

The original grunge guitar part as supplied by Adam Bevan.

Grunge Guitar Processed MP3 AIF

The re-processed grunge guitar part as used in Paul White's mix.

Guitar Bass Original MP3 AIF

The bass part played on guitar, as used in Adam's original mix.

Guitar Bass Processed MP3 AIF

The bass-line guitar part as processed by Paul White to get a more obviously synthetic sound.
Remodelled Break MP3 AIF

The slightly re-worked break section in Paul White's mix.

Tweaked Ending MP3 AIF

The re-worked ending of Paul White's mix.

Vocal Original Dry MP3 AIF

The original unprocessed vocal part.

Vocal Processed MP3 AIF

The vocal part as treated in Paul White's remix.

Published in SOS June 2009

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PC Notes

PC Notes
Time for an Upgrade? Buy PDF
Published in SOS June 2009
Technique : PC Notes
Printer-friendly version
If you’ve built your own music PC in the last couple of years, you might now be finding that
aspects of it need upgrading. PC Notes offers some advice.
Martin Walker

B ack in October 2008, a Microsoft official was quoted as saying that Mac buyers are paying an ‘Apple Tax’, and as I write this in early
2009, Microsoft’s swipes at Apple pricing are getting even more aggressive. In the US, TV ads have just been aired that follow a
woman with a $1000 budget trying to buy a laptop with a 17-inch screen. She’s filmed walking into an Apple store that has just one
laptop that she can afford, which only has a 13-inch screen, and ends up buying a Hewlett Packard PC laptop with 17-inch screen for just
$699 in another store.
Microsoft CEO Steve Ballmer also commented that “The economy is helpful. Paying an extra $500 for a computer in this environment —
same piece of hardware — paying $500 more to get a logo on it? I think that’s a more challenging proposition for the average person than it
used to be.”
PC pricing is certainly getting more cut-throat, and this also has worrying implications for DAW builders. Contrary to popular belief, most
buy their components not in bulk and at discount prices, like mainstream PC builders, but in smaller quantities from the same on-line
companies that DIY builders use, and at the same price. They offer an almost ‘bespoke’ service, generally building music PCs that match
individual customer requirements, provide guaranteed performance with audio hardware and software, at low acoustic noise levels, and
also offer generally excellent engineer to customer tech support. Yet they are often accused of ‘rip-off pricing’, when in fact most now get a
relatively small return for the many hours they spend building, testing and supporting each machine. Let’s hope the majority manage to stay
afloat, providing their services to musicians who just want to make music on a PC without worrying about hardware problems, and who
don’t want to negotiate the added learning time and possible frustrations of building their own music PC. Talking of which...

Upgrading The SOS DIY PC

The SOS PC Music forum is still awash with people asking for advice on building their own music
PCs, but recommendations have to change more and more rapidly nowadays, as product life-
cycles get shorter and new products appear ever more frequently.
Anyone who used the recommendations I gave in SOS February 2007 when building my own
dual-core PC will have an Intel DP965LT motherboard with 965 chip set, Intel E6600 dual-core
processor running at 2.4GHz and 2GB of Corsair XMS2 DDR6400 RAM. I personally still haven’t
outgrown this system — it provides me with more than three times the power of my previous
Pentium 4 2.8GHz machine — but you may be finding you need more RAM or processing power.
RAM is easy enough to buy, and a lot cheaper now. My 2GB cost me £190 in the UK when I built
the PC in December 2006, but nowadays you can upgrade by a further 2GB for a minuscule £23. Building an Intel Quad-core music PC
and need plenty of PCI slots?
Although Intel haven’t updated the DP965LT’s BIOS for some time to recognise newer Gigabyte’s EP43-DS3 motherboard
processor types, there are nevertheless a couple of possible CPU upgrades. As I write this, you could be the perfect candidate for the
can double your processing power by upgrading to an Intel Q6600 quad-core model running at job.
the same 2.4GHz clock speed — but since the Q6700 quad-core at 2.66GHz is selling at an
almost identical price, of around £140, you’d be foolish not to take advantage of its 11 percent faster clock speed.
Hopefully, those who built a system following my more recent advice in PC Notes January 2008 are still happy with the performance of
their Intel DP35DP motherboard with P35 chip set and Q6600 processor. However, if anyone is now searching for more processing power,
that system could be upgraded to a Q9650 model running at 3GHz, although this will cost about £270. That seems, to me, a little expensive
considering that you’ll only gain 25 percent extra performance.

DIY Budget PC Recommendations

If you’re considering a budget DIY build right now, I’d recommend a Q8200 running at 2.33GHz, which retails for the same price as the
Q6600 but should run significantly cooler, as well as having an 8MB instead of a 4MB L2 cache size, and support for the faster 1333MHz
FSB (Front Side Bus). If you want to bring your green credentials to the fore, the slightly more expensive Q8200S provides exactly the
same performance but consumes a maximum of 65 Watts compared with the 95 Watts of the Q8200. It does cost about 50 percent more,
though. I suspect that most musicians with a little more cash will plump for the Q8300 (£160), given its higher clock speed of 2.5GHz.
Sadly, these more recent Q8000-series processors aren’t an option for the DP965LT motherboard.
For a new budget PC I’d also recommend a different motherboard to Intel’s DP35DP, since this is now getting rather long in the tooth.
Gigabyte motherboards seem to be getting almost universal praise from DAW builders, and they mostly use the Texas Instrument Firewire
controller chips that are so widely recommended for their compatibility with Firewire audio interfaces.
Two suitable models are the GA-EP43-DS3 and EP45-DS3. They have many features in common, including up to three IEEE1394 ports

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PC Notes

and up to 12 USB ports, but the latter model supports RAM up to the slightly higher DDR1333 spec (although this probably won’t result in a
significant performance hike with audio applications). The biggest difference between the two is the expansion slot complement. The GA-
EP43-DS3 provides more PCI expansion slots (four, along with one each of PCIe x1, x4 and x16), while the EP45-DS3 offers only two PCI
slots, but more of the PCIe variety (three x1, plus x8 and x16).

Core Blimey
If you need more processing power than you can get from a 3GHz quad-core processor (and that’s already a huge amount!), a more
expensive Core i7-based system is probably the best alternative. Intel’s 920 model running at 2.66GHz is by far the most affordable option
(£240). Gigabyte motherboards are, again, widely recommended for Core i7 systems, with the GA-EX58-UD4P and GA-EX58-UD4 models
providing two different combinations of PCI and PCIe expansion slots. The UD4 has one extra PCI slot, while the UD4P model instead
offers an extra x8 PCIe slot, plus support for Dolby Home Theater.
The amount of RAM is up to you, and both boards support up to 24GB, for those running Vista 64-bit. However, three 1GB sticks should
still be sufficient for many musicians running Windows XP or Vista 32-bit, while three 2GB sticks will make your new machine slightly more
future-proof, especially if you anticipate installing Vista 64-bit or the forthcoming Windows 7 in its 64-bit incarnation.

Husserl Sapphire Edition

Once in a while, I stumble across a niche product that deserves to be introduced to a wider
audience, and Husserl Sapphire Edition definitely falls into this category. This impressive
product is the latest release in a long line of ‘metamusic’ products from Heavens On Earth
(www.heavensonearth.com), and is currently available only as an Ensemble for NI Reaktor
owners.
Essentially, it’s a polyphonic step sequencer with 16 separate channels that can modulate
each other, although this simple description doesn’t do justice to the wonderful results. It has five
panels: Husserl Sapphire itself, two analogue synths and one drum synth that it drives by default
(although you can divert its output to any software or hardware synths of your choice), and an
18,000-word, on-line Help panel that’s supplemented by a well-written, 68-page PDF manual.
There’s a lot to take in at first, but it’s a tribute to the panel design that I found myself creating
memorable results within a few minutes of first launching HSE. At the heart of each channel is If you’re a Reaktor user and fancy a
the Pattern, with between one and 16 steps, but even here Husserl goes beyond the typical step polyphonic step sequencer capable of
generating complex, evolving melodies
sequencer by offering note durations long enough to overlap and create polyphonic effects, or
that may not repeat themselves for
up to three-note chords played by each step, while the Map section forces sequences to fit your
hours, you should investigate Husserl
chosen scale, and the Bar panel lets you modify your sequence each time it plays, offsetting Sapphire Edition.
parameters such as pitch, velocity, duration and tempo.
HSE can be triggered from its own internal clock for generating stand-alone music, or clocked from external MIDI notes so you can
sync it with other tracks in your sequencer, as well as tracking incoming MIDI pitch and velocity information. Another fruitful area for
exploration is the various trigger modes: MIDI Layers, where each note you play generates its own sequence; and MIDI Fugue, where
each incoming note retriggers a sequence, overlapping if the previous one hasn’t finished.
However, HSE moves way beyond the competition with the intermodulation possibilities of its Matrix panel, where one sequence can
change the melody of another in a variety of ways, and your mind begins to boggle at the possibilities! With all the modulations available,
using just two or three channels can result in complex, evolving melodies that may not repeat themselves for hours, and there’s even a
Jitter panel where you can apply random variations to each note, to restore the human touch.
The 107 snapshots (presets) have been carefully written to demonstrate all the main features, while the various demo songs show how
Husserl can create believable music ranging from church organ improvisations and fugues through to Gamelan orchestra performances.
However, for me, its biggest strength is in creating slowly evolving ambient soundscapes. Considering that several thousand hours of
development time have gone into Husserl Sapphire Edition, it seems extremely good value at just $79.99.

PC Piracy News
Prosoniq (www.prosoniq.com) have announced that, with effect from June 1st 2009, they will discontinue development of Windows
products, largely due to piracy, and will in future be exclusively working on the Mac platform. This is indeed sad news; obviously,
software developers are feeling the pinch like everyone else at the moment. However, I think blaming the demise of a PC range on PC
piracy is rather misleading. I’ve long been a fan of Prosoniq’s Orange Vocoder, including it in ‘Exploring PC Spectral Multi-band Plug-ins’
in SOS April 2007, but the most recent PC version is still dated 2002, and I could find no Prosoniq press release relating to any Windows
products more recent than 2004. Prosoniq also admit that 90 percent of their customers are Mac users anyway. So, unless I’ve missed
something, this announcement is really the confirmation of a decision made five years ago, rather than the result of continued PC piracy
eroding profits.
Meanwhile, Microsoft (www.microsoft.com) have announced a new strategy to prevent pirate copies of their ‘Games For Windows’

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PC Notes

being leaked onto the Internet before their official release date. Such is the anticipation for some products that even otherwise honest
gamers will apparently download a cracked version if it appears before shops are in a position to stock it, resulting in decimated sales.
Unscrupulous shops have also been known to start selling before the official release date to gain an advantage over their competitors.
Microsoft’s solution is an unlocking system that makes games unplayable until a set date. Termed ‘zero day piracy protection’, it
involves encrypting the game such that you can only play it once you’ve gone on-line and received the decrypt code that only becomes
available on the release date. Let’s hope this doesn’t catch on with audio developers: I can imagine the outcry if your sequencer needs
reinstalling on a Sunday and the developer’s web site is down for maintenance!

Published in SOS June 2009

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Studio SOS: John Clark

Studio SOS: John Clark

Home recordings Buy PDF
Published in SOS June 2009
Technique : Recording/Mixing
Printer-friendly version
We lend some practical studio advice to a seasoned pro guitarist who’s rather newer to the
world of songwriting and home recording.
Paul White & Hugh Robjohns

J ohn Clark’s Wiltshire home provides him with a fairly small but quite workable room in
which to record, and he has bought himself a large-screen iMac with Logic Pro 8, a MOTU
828 MkII audio interface and a pair of Behringer Truth B2030A active monitors. By the time
he called us, he’d already made some recordings of himself singing and playing guitar into both
single and dual-microphone setups, and while these had a really nice ‘early recordings’ vibe to
them, he was keen to explore ways of getting a more contemporary sound. He also needed
advice on buying microphones, as his AKG C1000S and Electro-Voice RE16 were hardly state of
the art. He also said that he’d appreciate some tips on using Logic Pro 8, because he was fairly
new to the software.
John Clark had a very basic setup with
Studio Diagnosis no acoustic treatment, but it took very
little work to give him a usable
John’s studio gear was set up in an upstairs study measuring about 2 x 2.5m. The computer, monitoring space, with just a couple of
pieces of foam and a pair of Auralex
speakers and interface were all arranged on a flat table, facing across the width of the room, and
MoPads needed to tighten things up.
the doorway extended into the room by half a metre or so, cutting off the corner of the room. Our
first comment was that the room would probably work better from an acoustics perspective with
the speakers set up along the length of the room, which would mean putting the desk in front of the window (and radiator). John didn’t want
to do that unless absolutely necessary, so we arranged to do some listening tests and acoustics experiments first. Fortunately, the room is
entirely dry-lined with plasterboard on battens, and three of the walls are lightweight stud walls (which allow a lot of low-frequency energy to
pass right through or be absorbed). Because the room hadn’t been treated, John’s recordings had picked up a significant amount of small-
room ambience, but as he wasn’t playing any bass instruments, no low-frequency problems were evident.
Playing some test material through his Behringer Truth monitors showed these to be rather more neutral sounding than I remember the
very first incarnation of the Truths being, but the reflective walls were diluting the stereo image and, as expected in a small room, there was
a dip in the perceived low end when listening from the exact centre of the room. John’s mixing position was slightly forward of the dead
zone, so we concluded that any compromises caused by setting up across the room would be more than compensated for by John’s being
more comfortable working that way.

The Treatment
The most obvious room reflections could be treated using straightforward acoustic foam, which
absorbs in the mid- and high-frequency ranges, but does nothing for the low end. The
plasterboard skins of the room acted as impromptu bass traps, as did the window and door, and
we set about gluing some thin plywood slats to the rear top edge of some foam panels, so that we
could hang them on the walls, ‘picture-style’, using a single wood screw: this would avoid making
any irreversible changes to the room. We put the wood screws directly into the plasterboard, and
they were more than adequately secure to hold up a lightweight foam sheet. The arrangement we
settled on was to treat the two corners behind the mixing seat, as well as the two walls directly to
either side of the monitors. Ideally, we’d have had an absorber directly behind John’s chair, too,
but he had a rather splendid print dominating that wall, so we worked around it. In the corners, we Paul tries out mics and placements for
hung one vertical 4-foot x 2-foot sheet of foam on the side wall and one on the rear wall, so that recording John’s guitar and vocals. The
foamed corner and Reflexion Filter dried
both pieces met in the corner. Not only would this dry up the overall room sound to a useful extent
up the room ambience on John’s
but it would also allow John to sit with his back to the foam when singing, to reduce the amount of recordings — but perhaps a little too
reflected sound getting into the front of the mic. much?
We used Auralex MoPads to isolate the monitors from the desktop, and arranged the foam
wedges to give the maximum eight-degree tilt-up, angling the monitors so that the tweeters aimed towards John’s head when he was in his
usual mixing position. Repeating our listening tests showed an improvement in stereo imaging and a noticeable improvement in the
perceived dryness of the room, and the amount of vibrational energy getting through to the surface of the desk was also much reduced.

Better Recordings
One effective way to dry up the vocal sound further is to put an absorber behind the mic as well as behind the singer, and Sonic Distribution
had kindly provided us with an SE Reflexion Filter, which is ideal for this purpose. It absorbs sound that would otherwise get into the rear

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Studio SOS: John Clark

and sides of the mic, and also reduces the amount of the singer’s voice getting out into the room to cause reflections in the first place.
John was interested in buying an AKG C414 for vocals, so Hugh brought one of his own along (a previous generation C414 B-ULS) just
to make sure that it would suit John’s voice. We set this up on a stand with the Reflexion Filter (after assembling the hardware in our own
way to achieve a better mechanical balance), and used another of Hugh’s mics (a Neumann KM184) to mic the acoustic guitar. This was
actually a very inexpensive instrument, with flatwound strings fitted, but it gave exactly the vintage tonality John was looking for.
We made a test recording, miking the guitar from below rather than above, in order to improve the separation between the voice and
guitar. This was a major consideration, because John likes to play with the guitar in a fairly high position, and when miking the guitar from
the front he’d noticed a phasey quality to the recordings he’d made with dual mics — due to the amount of vocal also being picked up in the
guitar mic. Our arrangement produced a nice dry vocal, with much less spill on both guitar and vocal mics, and gave a natural guitar
tonality, with none of the unpleasant phasiness.
John prefers not to use headphones while playing his solo pieces, though, and he found that the lack of room sound caused by singing
directly into the Reflexion Filter made his voice sound rather dead as he sang, which didn’t help his performance. We tried the same mic
setup without the Reflexion Filter and found that the foam we’d put up behind the singing position was performing well enough in drying up
the sound, so he could get away with relying on that alone if necessary — the amount of room sound increased, of course, but the results
were still very usable, and much better than before the treatment had been installed.
Previously, John had been recording and playing from his mixing position, and he asked if he could try that again, so we moved the mics
and made another test recording. The room coloration was less than on his original recordings, due to the treatment we’d put up, but was
still audible, so we felt that recording close to the dead corner in front of the foam wall-panels would work best. This made John wonder
how he’d control the sequencer from his playing position, but a simple solution would be to use a USB extender cable, and put the Mac
keyboard on the window ledge to his right while tracking. There are also numerous compact remote-control hardware options, including
those from Frontier Designs and Presonus.
Moving back to the corner, we repeated the miking tests using an omnidirectional Neumann KM183 and then a Rode NT55 fitted with an
omni capsule. Using an omni pattern gives a slightly more open and natural sound, and seems to capture the percussive edge of an
acoustic guitar with more realism, but of course room reflections are picked up equally well from all directions, so there’ll be more room
sound than when using a cardioid in the same position. To overcome this, we screened the rear of the mic during these tests by holding up
a jacket behind it. If this was successful, John could use the Reflexion Filter behind the omni guitar mic to achieve a controlled pickup
pattern. All the options we tried produced acceptable results, but my own preference in this instance was for the result we got using the
NT55, as it seemed slightly warmer than the KM183. We recorded all the results so that John could decide for himself which mic setup
worked best for him. We also showed the importance of using headphones while changing the guitar mic position to find the best-sounding
spot, as this changes from instrument to instrument and also depends on both the mic and the room.

Logic Lesson
Our visit included a long session on Logic, (which saw Hugh slowly slipping into a coma!). John’s
pretty new to Logic, so we showed him where it saves its files and how to do a track bounce,
which places the mixed stereo file in the Bounces sub-folder of the main Project folder. I also
tweaked some of the preferences, so that 24-bit recording was selected by default and the
‘Independent Monitor Levels’ option was ticked. The latter function is very useful, as it allows the
user to set different fader positions for recording and playback, and Logic will remember them. In
John’s case, the monitor level during recording can be set to zero to avoid any of his performance
getting back to the mics from the monitor speakers.
John doesn’t record to a click track, so the usual comping techniques aren’t suitable: he’ll need
to record the same song two or three times, then paste together the best sections from each.
There’s always a risk that the guitar and vocal tracks can be slipped out of time, and to prevent John Clark after the Studio SOS —
much more confident in using his Mac
this, the vocal and guitar tracks can be assigned to a Group and the Group parameters all
and Logic setup than before!
switched off, other than the one that relates to selecting (the very first tick box). What this means
is that whenever either track is selected for editing or moving, the other is selected automatically,
so they always move as a pair.
John had created a default song, with a number of audio tracks and commonly used software instruments, but we noticed that he’d
inserted a separate instance of the Space Designer reverb on each audio track. Convolution reverbs such as Space Designer use quite a
lot of CPU overhead, so we went through the process of setting up aux send buses to feed a single Aux channel with just one instance of
Space Designer — exactly as you’d do on a hardware mixer. A quick way of creating sends for multiple tracks is to select them all by click-
dragging over the coloured boxes at the bottom of the mixer channel strips, then selecting a bus on one of the channels. The same bus
send will be added to all the selected channels: a big time saver!
John was experiencing some trouble with levels, because although his channel levels weren’t clipping, the master output level did
sometimes hit the end stops. The ideal recording level in Logic is with the channel meters peaking around halfway up, which still leaves
some 6dB or more of headroom. If the levels are still too high, you can use the ‘select all channels and drag’ feature to move the channel
faders down by a few decibels (when linked like this, they’ll all move by the same number of dB, which maintains the relative levels), but if
any of the channels include level automation they’ll just sneak back up to their original level. Some DAWs allow you to reduce the input gain

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Studio SOS: John Clark

on the master bus, but in Logic you need to do a simple workaround: insert a Gainer plug-in into any automated channel (usually after any
existing plug-ins) and then lower the gain in Gainer by the same number of decibels as you just reduced the fader levels. If you have any
automated channels with EQ or compressor plug-ins running, you can also use the level controls in these plug-ins as overall level trims.
We spent a while optimising a startup song and then saved it as a Logic Template, which could be selected from the ‘New’ menu. This
included screensets that use the number keys to switch between different useful window views, a default reverb and delay on two sends,
and some commonly used software instruments, such as piano and drums. John made copious notes as we went along and seemed very
keen to put this new information to use.

Bidding Adieu
Our last job was to move our tools back into the car and tidy up, in preparation for taking a few photos of the new arrangement. We’d like to
thank John for providing us with lunch and for all the Jaffa cakes (apparently they don’t count as one of your five a day!), and we look
forward to hearing how he progresses with his recordings.

Sending Out An SOS

John Clark has enjoyed a long career as a professional guitarist, first gaining serious recognition when he successfully replaced the
departing Alan Holdsworth in ‘Bruford’. Following Bruford he became part of Cliff Richards’ backing band, a job he’s held down from
1981 to the present. John: “Although I’ve been a guitar twanger for a long time, electric has always been my thing: acoustic never really
‘flicked my switch’ somehow. That is, until I had this rather late-in-the-day creative binge. In the past, when I’ve done sessions on
acoustic, I’m ashamed to say I never really took a blind bit of notice of mic selection or placement. Now I wish I had! Touring with Cliff
there’s always acoustic to be played, but on stage it’s always DI’d, and mics don’t come into it. Also, up to now I’ve never been a singer,
so all in all I need a proper sorting out!”

Reader Reaction
John: “Paul and Hugh’s expertise and X-Ray ears have certainly pointed me in the right direction. Recording in the corner of the room
with my back to the foam panels dried up the recordings nicely (and the Reflexion Filter dried it up even more) but although this was
great for recording, I prefer to play without wearing headphones, and the sound in the room was now so dry as to be a bit of a vibe-killer.
“Since Paul and Hugh’s visit I’ve tried moving back to the centre of the room and recording nearer the mouse, keyboard and screen,
but this time using the Reflexion Filter behind the mic. This still seems to damp down the ‘room tone’ pretty well, and as I find it much
easier to work nearer the screen, this seems like a good compromise.
“I agree with Paul that the Rode NT55 mic seemed to work best with my old archtop guitar. The fact that I wear the guitar stupidly high
really doesn’t help with the ‘separation’ issue, I know, but I’ll just have to work around that.
“The Logic techniques that Paul showed me are brilliant time-savers and speed up the workflow — which is always good for the
creative bit. Anyone who has any mic tips for me or who’s interested in the tunes, do contact me at [email protected]”

Published in SOS June 2009

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Notes From The Deadline

Notes From The Deadline

TV Music From The Inside Buy PDF
Published in SOS June 2009
Music Business
Printer-friendly version
There’s learning, and there’s education. And when it comes to music technology, the two
aren’t always connected.
Paul Farrer

‘W e don’t need no education,’ or so the song says. I’m inclined to agree.

A kid emailed me recently asking for advice on how to make it as a film and TV composer. He was 19 years old, in the last
year of an A-level in Music Technology, and yet he’d never heard of Sound On Sound. Didn’t know what it was. Never seen the web site.
He didn’t know what a BAFTA was and had never heard of Ennio Morricone. But he said he had Reason on his PC. So I suppose that’s all
right, then.

Hard Choices
Imagine the scenario. You’re 17 and your mum gives you an ultimatum: either get up at 6am
every morning and go and be an apprentice plumber with your uncle John, or get some
qualifications. What’s that you say? Local college offers music tech courses? I can hang out for
three years, join a band, then hand in some course work involving bongos and an effects pedal?
These crossroads in life are always so tough on the young people, aren’t they?
I truly believe there are loads of excellent music tech learning institutions crammed with
dedicated people who set their hearts on working in music and will stop at nothing to achieve
their dreams. Good luck to them. I also know how education can be the silver bullet with regards
to unlocking potential in people. However, I also know that a picture of a mixing desk looks
fabulous on even the dullest university prospectus and, as far as the deans are concerned, pupils
equals pounds — so why would they care if you can’t get a decent plumber these days for love
nor money?

More Music, Less Drugs “Students at Giggleswick University

(formerly the North Yorkshire College of
The UCAS web site (the organisation that deals with placing people in further and higher Apiary) are given every opportunity to
education) lists, across the UK, 286 courses relating to ‘music technology’. To put that into present their work in the public arena.”
perspective, it lists just 134 in pharmacology, 114 in zoology, eight in cancer care and, curiously,
none in plumbing.
My point is that the impetus for any kind of success comes from you. If you plot your own focused, unwavering and realistic course
through the business, work incredibly hard and never give up, chances are you’re already destined for good things — educated or not.
Or, to quote Henry Ford; whether you think you can or you think you can’t — you’re probably right.

Spurned!
As an experiment, I recently contacted most of the educational establishments that advertise in the back of the UK edition of this magazine
and offered to visit their media students and give a talk or have a question and answer session. I sent them a brief CV and links to my
showreel. Told them of my 20 years of service in the music business, the shows I’d scored and the awards I’d won, and mentioned the
number of emails that I get from eager (and often desperate), budding composers wanting to break into TV. I’d travel anywhere in the UK
and would give my time free, for the purposes of trying to paint as accurate a picture as possible of how music and the media business
works, to as many of the students as were interested. Want to know how many of them took me up on the offer? Not a single one.
I’d probably just depress everyone anyway.

Published in SOS June 2009

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The Remix Business: Part 1

The Remix Business: Part 1

The World Of The Remixer Buy PDF
Published in SOS June 2009
Music Business
Printer-friendly version
Remixing is a modern phenomenon that has turned into a viable way for hi-tech musicians to
make money from their skills. But how do you get heard, how do you land a job, and how
much should you charge to do it? We get you started with the insider’s guide.
Simon Langford

A ccording to Wikipedia, “A remix is an alternative version of a song, different from the

original version. A remixer uses audio mixing to compose an alternate master recording of
a song, adding or subtracting elements, or simply changing the equalization, dynamics,
pitch, tempo, playing time or almost any other aspect of the various musical components”. Sounds
simple enough, right? Well in theory, yes, it is, but like any musical art form, there’s more to it than
meets the ear.
In this article, I’ll be telling you what it is like to work in the modern remix industry, and how best
to go about breaking into it. In next month’s SOS, I’ll follow this up with some down-to-earth,
practical advice on the creative side of remixing. I won’t lie to you and tell you that this is an easy
industry to get into — it isn’t. But it is a very rewarding one. In many ways you can write and
produce a track exactly as you would if it were your own, but with the benefit of having the song
already written, the vocals already recorded, and a record label already committed to the song.
You can also get a lot of publicity from the popularity of the artist that you’re remixing. Over the
years, I’ve remixed artists including Rihanna, Robbie Williams, Sugababes, INXS and many more. By remixing tracks by artists such as
these, it’s quite possible that the fans of the artist will keep an eye out for your remixes in the future, simply because you remixed their
favourite artist. That kind of PR is invaluable — in fact, it can often help to establish you as a producer in your own right, further down the
line.

Remixing: A Potted History

To understand the industry, you have to know where it comes from, so before I move on to today’s scene, I’ll run through some of the more
important historical aspects of remixing. People have been ‘remixing’ music for almost as long as recorded music has existed, especially
since the advent of convenient recording media such as magnetic tape in the late ’40s. Although technically not ‘remixes’, music forms such
as musique concrète used manipulation of existing recordings to create new musical pieces, although in this instance they were often
largely unrelated to the original recording — and, for the most part, unrecognisable. It wasn’t until the late ’60s and early ’70s that the remix
as we know it began to develop. This period saw the development of the dance hall culture in the Caribbean (and in Jamaica in particular),
something that was driven by advances in multitrack recording technology. Producers such as King Tubby and Lee ‘Scratch’ Perry would
take the multitrack recordings and, with some heavy editing, create their own interpretations of songs. These were often just extended
versions of the originals, created by manipulating the arrangement and dynamics of the original musical parts, but this arguably paved the
way for the whole remix culture that has become so prevalent today.
At more or less the same time, the exploding disco scene drove a similar evolution of remixing techniques. Frustrated by the need to keep
changing records, and motivated by a desire to get people on the dance floor, DJs would create their own ‘extended’ versions of the popular
tracks of the day.
To many, the godfather of the modern remix is Tom Moulton, whose career started out by making ‘mix tapes’ for a Fire Island nightclub in
the late ’60s. Eventually, he progressed to being an adviser on the nightclub-oriented recordings of the time: his skills were called upon to
make sure that the records were club-friendly prior to release. Finally, he began to specialise in actually doing remixes, specifically for the
nightclubs. He is said to be the inventor of the ‘breakdown’ and the 12-inch single format.
However, to many others, the true pioneer of the modern remix was Shep Pettibone. With credits as spanning The Bee Gees to Betty
Boo, Erasure to Elton John, and Madonna and Metallica, Shep was truly a revolutionary, and it was his work that made people realise the
value of specific remixes crafted for nightclubs. Alongside this, he also helped to establish ‘house’ music as part of the mainstream dance
culture. In fact, without his work, I probably wouldn’t be doing what I do today. And for that, Shep, I thank you...
Technological developments in musical instruments and production tools have continued to create opportunities. In the ’80s, the
proliferation of relatively affordable synthesizers heralded a new dawn in the age of the remix: prior to this, remixes had been largely based
on the original recordings, but remixers now had the opportunity to create something entirely new, without the need for huge recording
studios. As dance music (as we now know it) arrived in the late ’80s, this approach developed even further — to the extent that often only
the vocals and perhaps a few of the original elements were kept — and further still with the unstoppable rise of computer-based recording in
the ’90s, which has given birth to a vast array of versatile production tools.
In the ’90s and ’00s, remixes ranged from subtle reinterpretations of recordings to completely new and — ironically enough — often
sometimes unrecognisable versions of songs. Another new arrival on the scene was the ‘mashup’, the process of taking two recordings and

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The Remix Business: Part 1

placing them on top of each other (‘mashing’ them together). Normally this takes the form of an instrumental track and an acapella. Legally,
this is something of a grey area, especially if the ‘remixer’ intends to sell them, but it is still something that’s gone from strength to strength.
In fact, there are artists, such as The Cut Up Boys, whose career relies largely on their ability to pick out songs that will work together
without sounding like a train wreck.
This pretty much brings us up to date. In the world of the superstar DJ and superstar remixer, the fees commanded by the remixer can
often exceed the advance paid by the record company to sign the record in the first place! And with an ever-increasing diversity of musical
genres to deal with, the need for remixes becomes greater than ever. And yet it is, perhaps surprisingly, a very competitive industry, and
one which it takes a great deal of effort, and determination — not to mention luck — to get into.

So You Wanna Be A Remixer?

OK... so you love the idea of remixing and have decided that this is an industry that you want to
get into: what’s the first thing you do? In my opinion, you really must learn how to produce your
own songs first. Remixing a record today is a very similar process to producing a record of your
own; it’s a multi-faceted job, which requires you to have skills as a musician, programmer,
arranger, producer, engineer, publicist and diplomat. Yes, you read right: diplomat. One of the first
things you learn when you start to do commercial remixes is that the remix is only partly about
what you want. In essence, the point of a remix is to bring a record to an audience which it might
not otherwise have reached. This is more complex than it sounds, because to get to that point
you’re going to have to please a lot of people. Working from the target audience backwards, you
have:
1. The ‘Punters’. That is, the punters in the nightclub. If they don’t dance to it, forget it, you’ve
A quick trawl through the iTunes store
failed your mission. will soon show you just how much
2. The DJ. If a DJ doesn’t like it, they won’t play it — and the punters will never get a chance to competition you’re up against if you
hear it. want to get into commercial remixing...
and these are the ones who’ve already
3. The Promoter. Because of the often huge amounts of records that some promotions companies
‘made it’!
work with every year, and because of the feedback that they get from the DJs, a good promotions
company will often know in advance what is — and what isn’t — going to work on a dancefloor.
4. The Artist. While many artists understand the need for remixes, some are still very protective of their work, don’t really like ‘dance’ music,
and may feel that a remix affects their artistic integrity. To convince them that your interpretation of their beautiful work of art isn’t outright
blasphemy is sometimes quite a task.
5. The Record Label. As they’re the ones in posession of the cheque book (or Internet banking password) they have to be happy with your
work.
6. The Remixer. Yes, that’s right, you’re down here at the bottom of the list.
The reason why the remixer comes last on this list is not because they are the least important (far from it!). Of course you, as the remix
artist, have to be happy with what’s going to be released, but ultimately there are a lot of people after you that have to be happy with it too.
When you’re more established and have some profile of your own, you can be more rigid about what you do: people will hire you to remix a
record because they want your name on it, and as such they’ll be more accommodating. So in the beginning, you’ll need to be prepared to
be a little flexible... but not too much.

Carve Out A Niche

With modern recording technology, it is more than possible to create commercial release-quality
recordings with a small investment of a few thousand pounds, or perhaps even less. This is great
news in some respects, but it also means that the music-production marketplace — and that for
electronic music production in particular — is becoming increasingly saturated. This, in turn,
means that one of the toughest challenges for budding remixers is to establish a niche for
themselves.
What do you do to stand out from the crowd? That’s really a hard question to answer, but if,
financially speaking, you can afford to do it, I’d advise you to stick to your guns and do what you
want to do. But at the same time, you need to be prepared to take constructive criticism — and
Even if you can get clearance for a
always remember the list above.
copyrighted sample, it’s unlikely that
you’ll be able to do so in time to meet
Management? your remix deadline. Some commercial
sample libraries can have restrictive
Once you have your studio, you have your ‘sound’, and you know what you’re doing, what do you
terms attached, so remember to check
do next? This question is similar to the perennially asked “How do I get my band signed?” and, as
the license agreement.
with that question, there’s no simple answer. I was fortunate enough to have a manager to help
promote me, and that made all the difference in my case, but there are certainly other ways.
Credit Crunch, global financial crisis, recession... call it what you will. We’re all acutely aware of it, and like most other industries the

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The Remix Business: Part 1

music industry is suffering. Illegal downloads remain a very large thorn in the record labels’ sides, too, and record sales are down. Way
down: labels are having to seriously tighten their belts, and that includes what they pay out for remixes. Nonetheless, there are labels
looking for remixes, and there’s still a good chance that if you approach one that is, they’ll consider you. They’ll probably want to hear a
showreel of your previous work (a Catch 22 situation, if ever there was one), but if you can provide them with examples of your own tracks
and productions, and if they like what they hear, you might get the gig.
Unfortunately, your work as a remixer will almost certainly be ‘on spec’ — meaning that the label will only pay you if they like what you do.
That’s par for the course these days, and it’s only the real A-list remixers that aren’t working in this way. So you’ll need to be prepared to
put in the effort and face the possibility that you might do all the work for nothing. The silver lining in this situation is that even if they don’t
decide to use the remix commercially, you can still include it in your showreel. What showreel? Don’t worry: I’ll come on to this later!
It might take a few attempts to get your first remix accepted, and even then, it might not earn you a lot of money. But it is a foot in the
door. After that, and with a liberal does of hyperbole, you’re on your way to your second remix. And then your third one... and before you
know it, you are on your 171st!

Technicalities
There are other things to consider along the way, including some legal issues that you need to be aware of. In almost all cases, although
you may make recordings during the remix, you do not own the sound recording; that is covered by part of the fee that you are paid for
doing the remix. It is actually surprising how few of the remixes that I’ve done I actually have contracts for. More often than not, the ‘majors’
will give you contracts, but only really in order to protect their interests rather than yours. Most independents simply can’t justify the expense
of issuing remix agreements (which are still contracts) given the expected revenue of the track. In the absence of a remix contract, it is fair
to assume the following three points as standard:
1. Payment & Royalties: Payment for the remix is ‘full and final’. In other words, don’t expect anything else out of the record label in the
future. On some rare occasions, and if your remix becomes the ‘lead mix’ (that is, the version that is considered the main version), you
could perhaps be entitled to a very small royalty on the record. But this is increasingly rare.
2. Copyright: Ownership of the sound recording lies fully with the record label. This is pretty much standard and to be expected. There are a
few (and by that I mean you can probably count them on one hand) record labels that allow the artists they sign to retain ownership of the
masters. In this instance, it’s more a case of simply licensing the track for release — but in my experience this is almost impossible to find. If
the label owns the sound-recording rights to the original recording, there’s no chance of them allowing a remixer to own the rights to the
sound recording of the remix.
3. Sample Clearance: Any samples used will be declared by the remixer (not including the original parts of the recording being remixed,
obviously) and, normally, will have been cleared by the remixer prior to submitting the remix. Sample clearance itself is a very complex and
often time-consuming issue, so it’s advisable to avoid using any samples if you can help it: there usually just isn’t time to clear them within
the remix deadlines, even if you want to. Of course, sample libraries don’t tend to fall into this area — because when you buy the library you
are usually given the right to use the samples contained within for commercial purposes. But it’s a good plan to check the licensing
agreement carefully, because some of them do have restrictions on usage that you might not expect.

Great Expectations
Many remix contracts will stipulate a certain number of alternate mixes (such as Instrumental,
Radio Edit, ‘Dub’ mix, ‘TV Edit’ and so on), but in the absence of a legal agreement, it’s often wise
to find out from the label exactly what they expect prior to commencing work on the remix. For
me, an average ‘remix’ job will comprise an extended Club Mix, a Dub Mix (often the same as the
Club Mix, but with less use of the vocals), an Instrumental and a Radio Edit. Sometimes the label
will also ask for a ‘PA’ Mix or ‘TV Edit’, which are basically different versions of the Radio Edit
where the lead vocal is absent, or, at the very least, reduced substantially in volume, to allow the
artist to perform the track ‘live’ — but this is far from being a standard requirement. As I said
above, if you have any doubt at all, it is probably best to ask. While these alternate versions do
not require the same amount of time to complete as the main mix that you do, it’s still a factor in Contract, what contract? Although the
balancing the fee that you’ll get against the amount of time it will take to complete the remix as a major labels will usually issue a formal
contract, many smaller labels don’t.
whole.

Building Your Showreel

To persuade a record label to part company with their treasured masters, you need to be able to show them what you are capable of. When
you get to a point where you can name-drop some of the people you’ve remixed, and when you can provide evidence that your remixes are
popular (in the form of Club Chart positions, quotes from key DJs and suchlike) it all becomes a lot easier. But in the beginning you have to
persuade them to give you your lucky break. This is why it is important for you to know how to produce your own tracks — because, in the
beginning, and in the absence of any other remixes, they need to get some idea of what you can do, both in terms of your style and your
quality of production. So if you have a few of your own tracks available that you can send to the label — either physically or via email or
download links — then, even if the tracks haven’t been signed or commercially released, it will at least give them some idea of what you do.
As I’ve already suggested, a manager will be able to help you if he or she has good connections with record labels — who may take you

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on based on the manager’s recommendation — but, of course, you’re in almost the same situation when it comes to getting a manager,
because you need to persuade them to take you on. And they’ll want to hear what you are capable of. So you need a showreel... Can you
see a pattern emerging here?
One of the most crucial factors when it comes to building your showreel is quality control: you need to be one hundred percent certain
that the work you put on your showreel is absolutely your best work. This is your one chance to make a good impression with the label
manager or A&R person. Dance music may have a higher degree of anonymity than most other forms of music, and a producer today can
have many pseudonyms under which they work, but if you make a bad first impression with your showreel it is still unlikely that you’ll get
another chance with that person. So make it count.
Even in these informal days of text, email and downloads, presentation is still important, so if you’re sending a CD, make sure it is clearly
labelled with your name, the names of the tracks that you’re submitting and — most importantly of all — clear and legible contact
information. As for providing information about yourself with the CD, I used to always send a printed biography, but these days I am less
sure that this is relevant. Often a link to a MySpace page or a personal web site (you do have one or both of those, right?) will be just as
relevant and informative, as most people spend more time on-line than off-line. But again, if you are providing details of a web site or
MySpace page, make the effort with that as well: not everybody is a whizzkid HMTL coder, and no A&R would expect a ‘newbie’ remixer to
have a web site that looks like thousands of pounds have spent on it, but do your best. If you can, beg or call in a favour from somebody
you know to help give it that little bit of extra gloss. And while you might wonder what relevance your web site or MySpace page could
possibly have to your production skills, don’t underestimate the importance of the former. It all adds up to say “I am professional. I care
about the impression I give, and I will put just as much effort and attention to detail into my remixing work as I did into this”.
It also helps to do some research to find out, where possible, the name of the correct person to send your showreel to. Sometimes this
information can be found on-line, while at other times simply phoning the label and asking for the name and telephone number or email
address of the person can be all you need. Some labels seem far less willing to give out that information than others, though... I won’t name
names here, but I was once chasing up an overdue invoice with one of the majors. I called their main switchboard and asked to be put
through to the accounts department and they actually refused to put me through without a specific name of a person that I wanted to speak
to! It was resolved in the end, but not without a lot of effort on my end, and the patience of a saint.
Believe it or not, these record labels do talk to each other, so if you simply do a mass mailing (physical or email) to every A&R department
in the country, you might soon acquire a reputation for being desperate. You see, the funny thing about the music business — and, from my
experience at least, especially the remixing industry — is that if you’re available to do remixes, the labels will want you less, whereas if
you’re too busy they’ll want you even more! So another thing to consider is being selective about who you send your showreel to: carefully
target labels that you think might be receptive to your style of work and send them your tracks, but choose only one or two labels at a time,
and be patient. Wait to hear back from those few that you sent out initially, and if you don’t get any interest from them, move on to the next
few. Any time that you spend waiting, you can consider a golden opportunity for refining your production skills.
As a final word on this subject, I’d recommend that you don’t expect too much initially. It can be incredibly difficult to manage your
expectations when you truly believe that you have got what it takes, but don’t go into record labels demanding exorbitant fees from the very
beginning.

Value Your Work

Some remixers — and The Freemasons spring to mind — can command fees of £10,000 (in the UK) or more for a single remix, but that’s
far from being the average. When I started out remixing, my first remixes paid in the region of £250. That might not seem a lot of money for
something that could take three or four days, but it is a start. Once you have those first few remixes under your belt (assuming that they’re
well received!), then you have more ammunition to approach record labels with. I did my first Soul Seekerz remix back in October 2004
and, as I said, I earned £250 for it. But a couple of years later I did three remixes for Rihanna in just over a week and earned £3600 for the
three — after I’d split the fees with my partner and paid my manager his commission.
With the technology so accessible these days, there are hundreds, if not thousands, of people out there looking to remix just for the profile
it might give them — and you have to compete with these people. That doesn’t mean to say that you should put a value of zero on your
work: while it might sometimes be tempting to do things for free, just to get a foot in, remember that nobody else in the process of releasing
that record will have worked for free. So why should you? Just remember to be reasonable: be patient, and start slowly, but never sell
yourself short.

From Little Acorns...

“Poca favilla...gran fiamma seconda”. This is a quotation from one of my favourite books, La Divina Commedia by Dante Alighieri. It means
“a great flame follows a tiny spark”. Let me go back to the example I gave earlier of Shep Pettibone: not only was he one of the most prolific
remixers of the ’80s (perhaps the most prolific), but he ended up co-writing and producing songs for Madonna. So while remixing may
appear to be a case of reinventing the wheel; while it might not always have the appeal of having one of your own songs lauded and
complimented; while it’s far from an easy thing to get into; and while it can involve (mainly artistic) compromises along the way, it really can
lead to other things.
It requires you to have quite a high level of skill and competency from the outset, but it’s a good way to get a level of exposure that you
might not otherwise get. You may be able to work with great people, and this all looks very good on your CV. There are less immediately
obvious bonuses too: sometimes, for example, you’ll get sent the whole multitrack for the original song, which means that you get to listen

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to the individual parts and learn how these songs were originally constructed.
Personally, I find remixing satisfying. I have had a couple of the songs that I have written and produced in the UK Top 40, and I’ve even
appeared on Top Of The Pops (RIP). But I still derive great personal satisfaction from doing remixes, simply because of the level of surprise
that you sometimes get from people when they hear that you have completely reinvented a song and given it a whole new interpretation.
Sometimes you can even change the context and meaning of a song (to some extent) if you are clever with what you do. But that’s a whole
other story...

About The Author

Simon Langford is a professional songwriter, producer and remixer who, as part of Soul Seekerz, has worked for some of the biggest
names in pop music, including Robbie Williams, Rihanna, Sugababes, The Ting Tings and many more.

Published in SOS June 2009

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