In This Issue

May 2005 In This Issue

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Reviews People
Allen & Heath Xone VF1 Business End
Analogue Filter Reader Tracks Evaluated
You probably know Allen & Heath as a studio mixer Listen online to the tracks whilst reading what some Music
manufacturer, but they also specialise in high-end DJ Producers Guild members think of the latest collection of
mixing and processing. Now they have combined their SOS reader recordings.
studio and DJ expertise into a multi-mode analogue filter
with sophisticated MIDI control. Crosstalk: readers' writes
Your letters, emails, faxes
Apple Mac Mini Replies from the editorial team to more of your valued
Desktop Computer feedback.
Apple have long been criticised for charging a premium for
their products, making the Mac platform a more costly Greg Ladanyi
choice than the alternatives. With the newly released Mac Jackson Browne, Don Henley & The SoCal Sound
Mini, Apple hope to change this perception, but is there Greg Ladanyi showed up at the right time in rock history to
enough Mac in the Mac Mini to make it useful for chair sessions for Jackson Browne, Don Henley, Warren
musicians? Zevon, Toto, Fleetwood Mac and the Jacksons — but while
50 percent of life may be simply showing up, the other half
ART TCS requires a lot of hard work.
Dual Compressor
Choose from optical and VCA compression, de-essing, Manny Marroquin
noise reduction, and valve voicing within a single Mixing For Kanye West & Alicia Keys
processor. You might not know his name, but you've definitely heard his
work: Manny Marroquin is the mix engineer of choice for
Audio Ease Rocket Science leading artists in both urban and rock music.
Effects Plug-ins (Mac OS)
A new plug-in bundle from Dutch developer Audio Ease Sounding Off
brings together three unique sound-shaping tools. Mr X
Soon all software could be FREE — like it or not...
Audix SCX25
Condenser Microphone Studio SOS
Large-diaphragm mics don't have to be bulky and East Norfolk Sixth Form College
unwieldy, as this compact new contender from Audix East Norfolk Sixth Form College needed help integrating their
demonstrates. hardware multitracker with their computer sequencing
system, so the SOS team travelled over to Great Yarmouth to
Bornemark Broomstick Bass lend a hand.
Virtual Bass Player Instrument (Mac/PC)
So how is your plug-in band coming on, then? With a The Dust Brothers
combination of Steinberg's Groove Agent and Virtual Sampling, Remixing & The Boat Studio
Guitarist plus Yamaha's Vocaloid, all that is required is a The Dust Brothers changed the course of record production
bass player. Enter, stage left, Bornemark's Broomstick with a new approach to sampling. In their first ever in-depth
Bass... technical interview, John King and Mike Simpson explain
their unique way of making records and open the doors of
CEDAR Audio Duo Auto Dehiss their remarkable LA studio, The Boat.
Broad-band Noise-reduction Processor
This new slimline rack processor makes CEDAR's latest The World At Your Fingertips
class-leading audio-restoration algorithms remarkably Paul White's Leader

file:///H|/SOS%2005-05/In%20This%20Issue.htm (1 of 4)9/27/2005 9:21:33 PM

 In This Issue

quick and easy to use. Last month's column aroused plenty of response so Paul
continues his thoughts on whether guitars or keyboards are
Edirol UA101 best-suited to being 'controllers'.
USB 2 Audio Interface (PC)
Edirol have pioneered USB 2.0 as a format for connecting Technique
audio interfaces, and their latest unit offers 10 inputs and Digital Performer: EQing Tracks
outputs at a price that compares well with Firewire
Digital Performer Notes
alternatives.
This month, how to tweak your tracks with MOTU's answer to
Fervent Software Studio To Go Sony's Oxford EQ and speed up your workflow with the
essential DP keyboard shortcut selection!
Bootable Linux Software Suite for PC
If you are attracted by the idea of Linux and open-source Pro Tools: latest news
music software, but put off by the thought of installing it on
Pro Tools Notes
your PC, there is another way: a bootable CD-ROM
Our new-look Pro Tools Notes column brings you all the
containing both the OS and all the software you need,
latest news from the Digidesign universe...
ready to go.
Reason: New Refills & Tips
JBL LSR6328 & LSR6312
Reason Notes
Active Monitors & Subwoofer
This month: new Refills and tweaking techniques for Reason
JBL's new monitors incorporate Room Mode Correction
v3, plus the usual haul of time-saving tips.
technology which claims to be able to reduce the bass
problems caused by standing waves at the listening Sonar: Studio or Producer?
position. But does it really work in practice?
Sonar Notes
Korg Kontrol 49 It's time to address the Studio versus Producer question, as
well as looking into a new Sonar remote-control option.
USB MIDI Controller Keyboard
Korg's Microkontrol was a highly versatile, yet compact Apple GarageBand 2, PowerBooks & iPods
MIDI controller — but perhaps, with its three octaves of
Apple Notes
miniature keys, it was too compact. With its four-octave,
Although intended as an entry-level application to introduce
full-size keyboard, the Kontrol 49 looks set to put that
new people to computer-based music making, Apple's
right...
GarageBand has received acclaim from beginners and
Mindprint Trio professionals alike. In this special extended Apple Notes we
look at version 2, which adds score editing, multitrack audio
Processor & Monitor Controller
recording, and more...
Mindprint cram mic and line channel strips, monitor
control, and talkback into a single desktop unit. Catch and Link modes in Logic
Native Instrument Battery 2 Logic Notes
Combining Logic's Catch and Link modes can greatly
Virtual Drum Module (Mac/PC)
increase the usefulness of multi-window Screensets, but it is
Two years on from its original release, Native's virtual
not always clear, especially to new users, how the various
drum module gets its first full upgrade. Is it all John
options work.
Bonham tom mayhem, or is it limper than a Kraftwerk
drum solo? CLASSIC TRACKS: The Who 'Who Are You?'
PMC TB2SA & DB1SA Producers: Jon Astley, Glyn Johns
The Who's final album with Keith Moon took almost a year to
Powered Monitors
record and pushed the band to the limit. Engineer and
Pioneering digital amplifiers are combined with PMC's
producer Jon Astley tells the remarkable story behind Who
proven transmission-line cabinet designs to deliver
Are You?'s title track.
spectacular monitoring performance at a project-studio
price. Composite Vocal Recording (Using Sonar 4)
Propellerhead Reason v3 Masterclass
The audio sequencing facilities we have at our disposal these
Virtual Electronic Studio (Mac OS X/PC)
days make it easier than ever before to produce world-class
Astonishingly, Reason is now over four years old! Version
vocal recordings by taking the best parts from a series of
3 adds performance-enhancing features and mastering
takes and producing a composite from them. Here's how to
facilities, losing only Mac OS 9 support on the way. We
do the job in Sonar 4.

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 In This Issue

bring you the first UK review of the full release version...

Demo Doctor
RME ADI2 & ADI4 DD Reader Recordings Analysed
A-D/D-A Converter & Digital Format Converter Listen to these tracks from SOS readers and see whether
Two new half-rack boxes offer considerable operational you agree with the good Doctor's prognosis...
flexibility and pristine sound in both the analogue and
digital domains. Maximum RAM
PC Notes
Roland Fantom Xa If you are tempted to approach the dizzy heights of 4GB of
Workstation Synth RAM in your PC, as supported by most motherboards, there
At £1099, the Xa is the most affordable keyboard in the are a few things to bear in mind.
Fantom range. But, inevitably, features have been
removed to make it such a bargain. Have Roland thrown Pop Shields: Why You Need Them
out the works from the workstation? Recording Tips
Pop shields are essential for most modern studio
Roland VC1 productions, but what are they and why are they so
D50 RAM Card for Roland V-Synth & VariOS important?
Roland's innovative V-Synth can now be reprogrammed
with a RAM card, effectively turning it into another Troubleshooting Your Pro Tools System
instrument. The VC1 card puts the clock back to 1987, Masterclass
perfectly recreating the S&S tones of the Roland D50. Whatever computer recording system you use, the chances
are it's going to fall over sooner or later, and Pro Tools is no
Sample Libraries: On Test exception. So before you call Digidesign's technical support
Sample Shop line, take a look at Sound On Sound's guide to diagnosing
Four new Sample Library collections get the aural and fixing the most common faults.
treatment from the SOS reviewers:
Using Folder Tracks in Cubase SX
Apple Jam Pack 4: Symphony Orchestra **** Cubase Notes
APPLE LOOPS Steinberg originally introduced the concept of Folder tracks in
Cubase VST, as a way of organising the track list in the
Sonic Boom Box **** Arrange window. This month we look at how this feature
APPLE LOOPS became even more powerful in Cubase SX, and how to make
the most of it.
Downbeat & Leftfield ****
MULTI-FORMAT Why & How To Partition Your Music PC
Platinum Essentials ***** Hard Drive
AUDIO+WAV PC Musician
Did you know that sensibly partitioning your hard drive or, if
Steinberg Groove Agent 2 it's already partitioned, simply swapping the positions of your
Virtual Drummer Instrument (Mac/PC) audio and sample partitions could result in a significant
From the same team that brought you Virtual Guitarist, improvement in PC audio performance? We explain the whys
Groove Agent has had an impressive upgrade, and is now and wherefores.
claimed to work better under non-Steinberg hosts. We put
it through a proper multi-platform test. Yamaha AW4416 User Tips
Masterclass: Part 1
Tascam FW1082 Combining the guts of an 02R mixer with a fully featured
Firewire Audio & MIDI Interface / Control Surface multitrack recorder, the AW4416 ended up being a
(Mac/PC) prodigiously complex beast. Our hands-on workshop shows
Joining the growing market for one-box devices combining you which of those snazzy features work best in practice, and
audio and MIDI interfacing with control-surface how to use them efficiently.
functionality, Tascam's latest Firewire unit might be all you
need for multitrack recording and mixing.

Terratec Phase 88 Rack FW

Firewire Audio Interface (Mac/PC)
Terratec's range of affordable audio interfaces now

file:///H|/SOS%2005-05/In%20This%20Issue.htm (3 of 4)9/27/2005 9:21:33 PM

 In This Issue

includes two attractive Firewire options: the eight-in/out

Phase 88 Rack FW and the two-in/four-out Phase 24 FW.

Ursa Major Space Station

Digital Reverb & Effects Processor
The original 3U monster that was the SST282 Space
Station has been reissued by Seven Woods Audio as this
small desktop unit. Not content merely to offer all the
classic sounds, though, the new incarnation has a number
of fresh tricks up its sleeve?

Yamaha Motif Rack ES

Workstation Synth Module
Yamaha's original Motif Rack was a fine-sounding, well-
specified synth module, but it suffered from MIDI timing
problems when reviewed in SOS. Two years on, we put
the follow-up Motif Rack ES to the test...

Competition
WIN: Digidesign 002 Digital Audio
Workstation
Sound Advice
Q Can I use a mono compressor for
stereo compression?
Q How do I convert a split-stereo file to
interleaved stereo?
Q How should I sync up my digital
inputs?
Q Is there any advantage to using two
subwoofers?
Q Should I buy a vintage analogue synth
or a modern modelling synth?
Q Will my PC run Garritan Personal
Orchestra?

file:///H|/SOS%2005-05/In%20This%20Issue.htm (4 of 4)9/27/2005 9:21:33 PM

 Allen & Heath Xone VF1

In this article:
Smooth Presentation
Allen & Heath Xone VF1
In The Xone: Studio Analogue Filter
Listening Tests Published in SOS May 2005
Comprehensive MIDI Print article : Close window
Functionality
Reviews : Effects
Optional Built-in RIAA
Equalisation
DJ Dream Machine

Allen & Heath Xone VF1

You probably know Allen & Heath as a studio mixer
£398
manufacturer, but they also specialise in high-end DJ
pros
mixing and processing. Now they have combined their
Stereo 12dB/octave or
mono 24dB/octave operation. studio and DJ expertise into a multi-mode analogue
MIDI control of filter type filter with sophisticated MIDI control.
and cutoff.
Balanced or unbalanced I/O.
Valve overdrive. Paul Nagle
cons
Mode changes require For over 30 years, Cornwall-based
powering down, and there's Allen & Heath have built a solid
no power switch. reputation for making quality mixers.
Switch for mono/stereo More recently they have diversified
mode mounted on the rear with the Xone range, tailored
panel.
specifically for the needs of DJs. A
MIDI Implementation could
logical extension to this range is the
be more extensive.
new Xone VF1, a compact 1U
summary analogue filter capable of 24dB/octave
A stereo analogue filter with mono or 12dB/octave stereo operation. Photos: Mike Cameron
valve overdrive, a built-in LFO
The VF1 is an enhanced design taken
and envelope follower, plus
MIDI control of important from existing mixer circuitry and offers low-pass, band-pass and high-pass
parameters. Other than a modes which can be selected in any combination. It also features a dual-triode
slightly awkward method of valve with two separate elements, an LFO, an envelope follower, and MIDI
changing MIDI modes, and control. With finishing touches that include balanced or unbalanced I/O, an
the rear-panel location of at
internal power supply, and stylish clickless switching of functions, you'll probably
least one useful switch, the
VF1 is remarkably agree that the market can cope with one more analogue filter.
straightforward and a delight
to use.
information
VF1, £398.36; VF1R, Smooth Presentation
£416.26. Prices include VAT.
Allen & Heath +44 (0) The matt-grey panel with its black text and Moog-style knobs conveys a tangible
1326 372070. air of finesse. Four large switches are provided, which activate the filters and
+44 (0)1326 377097.
select the three filter modes. These switches are backlit in orange (for filter

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Allen & Heath Xone VF1

Click here to email modes) and blue (for filter switching). With no audio clicks generated when you
www.allen-heath.com push them, they are exactly the kind of thing that inspires confidence.

Two large chunky knobs control cutoff and resonance. Smaller knobs set the rate
and depth of the onboard LFO, and also adjust envelope-follower and overdrive
amounts. Add switches for LFO waveform selection, envelope-follower decay,
and the routing of envelope follower to overdrive, and you have a simple yet
powerful set of options at your fingertips.

Where the front panel is a streamlined, ergonomic delight, the rear is as fully
populated as Jordan's T-shirt! There are balanced inputs and outputs on both
XLRs and TRS jacks, while unbalanced I/O is courtesy of phono sockets,
although the TRS jacks happily connect to balanced or unbalanced gear. With
the aid of a screwdriver, you can prod a tiny blue recessed button which sums
the inverted filter output with the raw input to create cancellation effects. This
inverted signal is then sent to the TRS jacks without affecting the signal routed to
the XLR outputs. Versatility is the name of the game; accordingly, you can use all
outputs at once.

When powered on, the orange glow of the valve is clearly visible through a grill
on the top of the unit. The manual advises against obstructing this ventilation
grill, so a little thought is needed in terms of placement in your rack. Thoughtful
placement becomes an even more important issue when you realise that the rear
panel has a couple of buttons on it. One of these, the Mono button, transforms
the VF1 into a single-channel 24dB/octave filter. In this mode, the right input is
ignored and the processed signal is sent to both outputs equally. Switching to a
steeper filter slope adds extra depth and resonance — handy qualities for bass
or solo instruments. Thus, positioning such a useful button on the back could
prove jolly inconvenient on a piece of gear designed to spend its days in a rack
or flightcase.

The other rear-mounted button is Local Off, which routes the front-panel controls
via the MIDI processor. Don't worry that MIDI control involves any sacrifices; you
can still use the large, friendly cutoff knob, for example, and the resulting filter
sweeps are smooth enough for Local Off to be activated permanently. MIDI In
and Out sockets are provided for transmission and reception of MIDI data.
Perhaps we can forgive Allen & Heath for the lack of a MIDI Thru because, on
such a densely-packed panel, you'd be hard pushed to find room for one! A
further audio output — the stereo Monitor Out jack — is provided as a means of
previewing the effect of the filter even when it's bypassed — ideal for live mixing
situations. The IEC power socket and fuse holder round off this well-provisioned
rear.

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Allen & Heath Xone VF1

In The Xone: Studio Listening Tests

Getting up and running is ludicrously easy, in part because the VF1 features no
input controls or level indicator. This is a conscious decision on the part of Allen
& Heath based on the philosophy that DJs shouldn't have access to level
controls! With a unity-gain structure and a healthy +22dB of headroom,
interfacing with most equipment should be a doddle. Indeed, I found the VF1
coped with a wide range of signals — keyboards, mixer sends/groups, and entire
mixes — without complaint.

After making the audio connections and enabling the VF1 with the Filter On
button, I set overdrive and modulation to zero, activated low-pass mode and
played a chord on one of my digital synths. The filter has a range from 20Hz up
to 20kHz, and a full rotation of the cutoff knob resulted in my jotting down terms
such as rich, smooth, and sweet as my first impressions. Several weeks of daily
use later, I'd classify the VF1 as ideal for enhancing a wide variety of material,
never dominating in the manner of, for example, the Sherman Filterbank. Not that
comparisons are really appropriate, but I believe the Xone VF1 would slot into a
mix far more readily, and thus see more general use, than a filter with more
'attitude'.

A built-in Automatic Resonance

Control automatically compresses the
resonant peaks and reins in the
sometimes speaker-shattering effects
of high resonance, giving a pleasing
sparkle without clipping, whistling, or
loss of bass. This should definitely be a
comfort when performing live. When
the mono switch is activated, the effect
of resonance becomes far more
This block diagram in the VF1 manual shows
pronounced and you can stray into
how the different processing elements
distortion. Here, the VF1 becomes interact with the controls and I/O.
more 'synthy', but remains one of the
most predictable filters I've
encountered. If this gives the impression of a lack of balls, any such suggestions
are dispelled when you reach for the overdrive knob.

Valve overdrive with an analogue filter is a marriage made in heaven — well, any
heaven that is famous for its pasties anyway! Low levels of overdrive introduce
soft clipping, adding a subtle warming effect. At higher levels, hard clipping
imposes a cutting distortion perfect for adding bite to solo instruments, basses,
percussion... you name it! A make-up gain circuit is used to maintain a relatively
constant level as you increase the overdrive amount.

For maximum tonal variety, any combination of the filter-mode buttons may be
pushed simultaneously. For example, activating low-pass, band-pass, and high-

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Allen & Heath Xone VF1

pass produces an 'all-pass' filter that adds a gentle fizziness to any source as
you crank up the resonance. low-pass and high-pass in combination create a
notch filter, whilst high-pass and band-pass together conjure up a gorgeous,
almost icy presence.

Having spent some time establishing a feel for the tone of the VF1, I increased
the LFO depth, adjusted the modulation speed, and enjoyed some traditional
filter warbles. Two LFO waveforms are available: triangle for smooth cutoff
changes and square for alternating stepped effects. Modulation rates vary
between 0.2Hz and 16Hz. The more experimentally-minded may wish for faster
speeds, but in this context I feel the range is judged correctly.

The filter can also be modulated using the envelope follower, its amount set
using a single knob. The envelope follower sets the response of the filter to input
signal level; the higher the depth, the more the input will drive the cutoff
frequency. A switch sets whether the envelope follower has a fast or slow decay;
for loops or full mixes, a fast decay can give unusual artificial 'pumping' effects,
but when used to filter synths, guitars, and so forth, a slow decay can sound
more natural. A second switch determines whether the envelope follower will
drive the output of the valve distortion. With dynamic sources this is very
effective, adding a controlled overdrive effect that is responsive to the source
transients.

Comprehensive MIDI Functionality

As shipped, the unit responds to MIDI channel 16. To change this, power on
whilst holding down the high-pass button and then set the channel using
combinations of all four buttons — a chart in the manual shows you how. The
VF1 responds to MIDI Continuous Controller (CC) numbers 80-83, which govern
the status of the filter mode and bypass buttons. Additionally, MIDI CC74 controls
the filter cutoff. By sending these MIDI controllers, you can remotely access or
automate key features via your master keyboard or sequencer. Helpfully, the
buttons and cutoff knob all transmit as well as receive their respective MIDI
controllers, but there is no way to control any other parameter, such as
resonance or LFO speed.

However, that's not the full story of MIDI control. Additional functionality is
available in the form of three additional MIDI input modes and, as when setting
the MIDI channel, these modes require powering down to change. To do this,
power up holding down the filter on/off switch and set the three available modes
using the three filter buttons. When complete, normal operation is resumed by
pressing the filter button again.

My only significant complaint about the VF1 is that it lacks a power switch, which
means that resetting the MIDI input mode requires you to pull out the mains plug
— not something I'd consider mid-performance, and I'm sure I'm not alone! The
additional modes are useful enough for this to be quite annoying, and Allen &

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Allen & Heath Xone VF1

Heath have promised to investigate

whether a future update could allow
changes to be performed via MIDI.

Of the three modes, the first is

designed to work in conjunction with
the Allen & Heath's own Xone:92 DJ
Mixer. I didn't have one of these, but
was able to reproduce the MIDI
controller messages it sends because
the manual helpfully records what they
are! In this mode, a single MIDI The three filter circuits in the VF1 can be
controller (CC13) is used to switch combined to create a variety of unusual filter
shapes, including an intriguing 'all-pass'
between filter types, although there are response if all the filters are activated
just five possible combinations simultaneously.
available: low-pass, low-pass plus
band-pass, band-pass, band-pass plus
high-pass, and high-pass. This means you can't set low-pass plus high-pass or
'all-pass' using this method, which is a pity. When in this mode, CC12 controls
cutoff and CC92 sets the filter on and off. I found the implementation of a single
continuous controller very handy for switching filters, even given the restriction
that not all combinations were accessible, but it was the other two MIDI input
modes that were far more exciting.

When the second input mode is enabled, the filter cutoff tracks the pitch of
incoming MIDI notes, provided that they are on the VF1's MIDI channel. You can
therefore play a synth and process its output whilst using MIDI to track the pitch.
Another fun technique is to direct MIDI notes to 'play' the filter whilst you're
processing an entire track or mixer subgroup.

The last of the MIDI input modes is the 'keyboard mute mode'. This makes the
current filter active only as long as an incoming MIDI note is held. As soon as all
notes are released, the filter is deselected, resulting in no output. This is superb
for gating effects — particularly when you trigger the mute function from devices
such as sequencers or drum machines.

What's even better is that the MIDI input modes can be used in any combination
— for example you could combine keyboard muting with keyboard tracking. In
conjunction with your sequencer, this provides an endless source of freaky
chopped-up mixes, with filters changing modes faster than your fingers could
move. And all without glitches or annoying cracks and pops.

One final MIDI facility worth mentioning is that two VF1s can be linked together
via MIDI to implement stereo 24dB/octave operation, both units being set to
mono. Connect the MIDI Out of your first filter to the MIDI In of the second and
the filter switches and cutoff frequency can be controlled remotely from the first.
Nifty as this option may be, in most cases I actually found the 12dB/octave mode
was just fine for stereo mixes; the 24dB mode seemed best suited to processing
individual instruments.

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Allen & Heath Xone VF1

Optional Built-in RIAA Equalisation

The VF1 is available in two versions, the VF1 reviewed here and the VF1R. This
latter version is a little more expensive because it includes a preamp for
connecting turntables requiring RIAA equalisation. RIAA equalisation is a
specification for vinyl playback established by the Recording Industry Association
of America. It was designed to permit longer playback times and improve sound
quality, and has been an industry standard since the 1950s. Before then, each
record company applied its own equalisation — with over 100 combinations of
turnover and roll-off frequencies used, the main ones being AES, LP, NAB, and
FFRR.

DJ Dream Machine

From the moment I first unpacked the VF1, it screamed 'quality!' at me — and not
just in terms of looks. Every sound source I put through it, from virtual analogue
pads to drum loops and even full mixes, became more malleable and sonically
gratifying. The effects on offer range from subtle and uncoloured to hard and
overdriven. Low levels of overdrive add warmth, but at high levels a gritty
underbelly of distortion is exposed.

Yet even at its extremes the VF1 remains usable and controllable; careful design
wards off the harshness and squealing sometimes associated with analogue
filters. As the envelope shaper can optionally drive the overdrive effect too,
there's plenty of mileage in muckying up loops and samples. In fact, I'd say it's
more or less obligatory!

Using MIDI control of gating, filter selection, and filter cutoff, there's a wealth of
signal-chopping and swooshing activities to indulge in. Although not all
parameters are MIDI controllable, the options chosen work well. Indeed, the VF1
has relatively few shortcomings, but its inability to switch freely between MIDI
modes without pulling the plug certainly counts as one of them!

The keys to the VF1 are its simplicity and versatility — it just insists on being
tweaked! To this end, the smooth knobs and soft switches are a delight and
every feature seems to have a range of operation yielding only practical results. If
you're looking for a stand-alone stereo analogue filter, the Xone VF1 comes
highly recommended.

Published in SOS May 2005

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Allen & Heath Xone VF1

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The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
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 Apple Mac Mini

In this article:
What's In The Tin
Apple Mac Mini
First Impressions Desktop Computer
Time For An Upgrade Published in SOS May 2005
Performance Print article : Close window
Adding More Memory
Reviews : Computer
Build-To-Order Price List

Apple Mac Mini £399/

£339
pros Apple have long been criticised for charging a
The smallest, most premium for their products, making the Mac platform
affordable Mac ever.
Doesn't get too hot or
a more costly choice than the alternatives. With the
produce an unbearable newly released Mac Mini, Apple hope to change this
amount of fan noise. perception, but is there enough Mac in the Mac Mini to
Makes an interesting DSP
accelerator for Logic Pro
make it useful for musicians?
users, especially those with
Powerbooks.
Mark Wherry
cons
The cost of the Mac Mini
can quickly rise once you start As a computer company, Apple seem to be
adding the build-to-order obsessed with size. When they aren't
options. creating the biggest piece of equipment
Some users who aren't around, as seen with the recently introduced
skilled (or willing) to use a
putty knife may be reluctant to
30-inch Cinema Display, they're busy
perform their own upgrades to developing at the titchy end of the scale as
the Mac Mini. well. Having miniaturised the iPod into the
summary iPod Mini, Apple have now miniaturised the
While the Mac has long been
Mac into the Mac Mini, originally announced
priced above the price range at this year's Macworld show in San
many people consider Francisco back in January, ending what has
affordable, the Mac Mini is a been a 20-year wait for a low-cost Macintosh
perfectly priced model for computer.
those wishing to try out the
Mac platform for very little
money, and isn't so Photos: Mike Cameron
underpowered that musicians
What's In The Tin
With a DVI output and the option of
and audio engineers couldn't
Bluetooth, the Mac Mini is compatible
find an interesting use for the
with Apple's Cinema Displays and
device.
The Mac Mini is available in two wireless peripherals, though it comes
information with no display, keyboard or mouse
configurations: one with a 1.25GHz G4 as standard.
1.42GHz model £399;
processor and a 40GB Ultra ATA hard drive
1.25GHz model £339.
Prices include VAT. for £339 and the other with a 1.42GHz G4
www.apple.com/uk/ processor and an 80GB Ultra ATA drive for £399. Both models offer a 512k Level-
2 cache on the processor, a 167MHz system buss, 256MB of PC2700 SDRAM

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Apple Mac Mini

Test Spec upgradeable to 1GB, an ATI Radeon 9200 graphics processor with AGP 4X and
32MB of dedicated Double Data Rate (DDR) video memory, along with a slot-
Apple 1.42GHz Mac Mini
running Mac OS 10.3.7. loading Combo drive capable of reading DVDs at 8x and CD media at 24x speed,
and writing CD-Rs at 24x and CD-RWs at 16x speed.
Apple Logic Pro 7.0.1.
1GB memory chip from
In terms of connectivity, the Mac Mini offers one Firewire 400 port, two USB 2.0
Crucial (CT431640).
ports, a headphone output mini-jack, a modem, 10/100 Ethernet, and a DVI port.
Putty knife. That the Mac Mini has a DVI port is actually pretty neat, and while Apple supply a
DVI-to-VGA adaptor, the Mac Mini is capable of driving LCD monitors with a
resolution of up to 1920 x 1200; or, in other words, a 23-inch Apple Cinema
Display. For wireless networking, the Mac Mini is compatible with Apple's Airport
Extreme cards, and there's also a build-to-order Bluetooth option. It's worth
pointing out that another 'feature' that puts the Mini into Mac Mini is the lack of
keyboard and mouse. This is obviously a cost-cutting move as well, although
because the Mac Mini's target market is partly those Windows users who might
just want to add a Mac to their system, Apple justify it by reasoning that those
buyers will already have a spare keyboard and mouse sitting around.

For the money, the Mac Mini's

specifications aren't terrible: the G4
processors used are faster than in the
current iBook models, and almost the
equal of what you would have found in
the Powerbook range until recently.
The ATI Radeon 9200 graphics and
32MB video memory will give adequate The inclusion of a Firewire 400 port enables
the Mac Mini to be connected to Firewire
performance for Aqua (32MB is the audio interfaces, and also to other Macs as a
minimum required), and at least the Node in a distributed Logic system.
video memory isn't shared with the
main memory. And speaking of
memory, this is perhaps the biggest issue people will have the Mac Mini's
specifications: 256MB really isn't enough to get the most out of Mac OS X. Even
if you just use consumer-oriented software, all of the iLife applications (with the
possible exception of iTunes) will benefit from more memory. So while the small
amount of included memory was probably necessary to keep the cost down, you
will almost certainly need to order more memory with your Mac Mini or upgrade
soon after the purchase.

In addition to the base models described above and on Apple's web site, there
are many build-to-order options from the Apple Store, along with a slightly more
beefed-up model Apple sell to certain retailers, and this is the area where the
Mac Mini has come in for the most criticism. Once you start configuring your
dream Mac Mini, complete with 1GB memory, a Superdrive, Airport, Bluetooth
and so on (see the Build-To-Order Price List box), it's easy to go over the 1000
figure in any currency, which starts to make the Mac Mini look rather less of a
bargain, and probably too expensive for what you end up with. To Apple's credit
(and not in the financial sense), though, the company did respond to this issue a
few weeks after the Mac Mini's release and lowered the cost of many of the build-
to-order options. Arguably, if you want to get a really powerful Mac system, you

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Apple Mac Mini

probably shouldn't be looking at a Mac Mini in the first place, since the product is
clearly aimed at the lower end of the market.

First Impressions

Despite the consumer orientation of the Mac Mini, I couldn't resist buying one
and seeing if it actually could be of interest to those who use Macs for music and
audio purposes, although this was initially harder than expected. On the day of
the Mac Mini's release, I visited my local Apple Store in the US and already the
Mac Mini had sold out. The on-line Apple Store, while fulfilling pre-orders, was
clearly going to be a bit of wait, but in the end I found another web-based store
that was able to supply a basic 1.42GHz Mac Mini a few days after the product's
release — interestingly, they sold out completely a couple of days after this.

Since there's no keyboard or mouse,

the packaging for the Mac Mini is
incredibly compact, and once you get
the actual computer out of the box it's
hard not to be astounded by just how
small it really is. As a 6.5-inch square
that's just two inches tall, and weighing
in at 2.9 pounds (1.32kg), the Mac Mini
makes Apple's last attempt at a
compact desktop Mac, the G4 Cube,
look bulky. And perhaps this
comparison isn't unfair: Apple's
publicity shots of the Mac Mini with The Mac Mini is capable of running five eight-
keyboard, mouse and an Apple voice instances of Sculpture in Logic Pro
7.0.1, as shown here.
Cinema Display are strangely
reminiscent. But while the G4 Cube
failed because it didn't really meet a market demand, the Mac Mini seems not to
have that problem; although I always wanted a G4 Cube, I could never justify the
cost, but the affordable nature of the Mac Mini removes this issue.

One factor that helps keep the Mac Mini small is that the power supply is an
external unit, which looks much the same as those used in Apple's current range
of Cinema Displays. At first I thought the external power 'brick' might spoil the
small aesthetic of the Mac Mini, but the leads are plenty long enough to hide the
power supply away from the computer. That the power supply isn't in the
computer also helps with cooling, and although I worried that the Mac Mini might
suffer from getting rather hot in the same way as Powerbooks, I didn't notice this
being an issue in practice.

After switching on the Mac Mini and proceeding with the now familiar Mac OS X
registration process, the Mac Mini did initially feel a little more sluggish than the
more powerful Macs I'm used to using. Even using the Finder made it clear that
more memory was definitely going to be necessary.

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Apple Mac Mini

Time For An Upgrade

At the time of my Mac Mini purchase, it was impossible to order a unit with more
than 256MB memory and get it delivered the next day; ordering 512MB or 1GB
memory from Apple would have cost £50 or £220 respectively. Instead, I ordered
a 1GB memory chip from Crucial for $161.99 (the equivalent cost in the UK from
Crucial's UK web site would be £102.21), which obviously worked out a little
cheaper. However, there are two things to bear in mind: firstly, the Mac Mini only
has one memory slot, so upgrading actually means taking out the current
memory chip and putting a completely new one in. And secondly, if you do order
the memory chip to put in the Mac Mini yourself, it's not quite the easy upgrade
that Power Mac and Powerbook users are used to.

In order to perform any kind of upgrade

on your Mac Mini, whether you're
adding memory or an Airport Extreme
card, you have to take the lid off the
unit. This doesn't sound difficult, of
course, except for the fact it's
impossible to do it with just your hands.
But the great thing about the now
almost cult appeal of the Mac Mini is The Mac Mini could make a perfect Logic
that a simple Internet search will reveal Node when used with a Powerbook via
a vast quantity of information on how to Firewire TCP/IP networking. This screenshot
take the unit apart, from those who is taken from a Powerbook G4 running Logic
Pro 7.0.1, but with Audio Objects offloaded
have found and posted Apple's official
to a Mac Mini running Logic Node,
service manual for the Mac Mini, a connected via Firewire.
training video for engineers, those who
have put a Windows machine inside
the Mac Mini, and so much more.

It turns out that the way to take the Mac Mini apart (which you'll see was
achieved quite successfully in the photo on the last page of this article) is to use
a putty knife, although it's probably a good idea not to recycle the same tool used
to retile your bathroom! A very thin blade can be inserted between the point
where the upper case meets the base of the Mac Mini to unlatch the clips so you
can slide the top of the computer away from the base. It might take a little
courage the first time you try this, and it's still questionable whether doing it
yourself voids the warranty, but it's actually fairly simple and takes very little time.

Obviously, neither myself or Sound On Sound accept any liability if you try to
perform your own Mac Mini upgrades, but I can say that I did do this on my own
unit and ran into no problems.

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Apple Mac Mini

Performance

Given the specifications of the Mac Mini, it goes without saying that this isn't
going to be the most powerful Mac system on the planet for music and audio
purposes. However, given the attractive nature of the machine, both in terms of
the aesthetic and cost, just how much can you reasonably expect? To answer
this question, I carried the usual array of tests that I've written about in previous
Mac reviews and in Apple Notes, using Logic Pro 7.01 and some of the built-in
instruments and effects: Platinumverb, Space Designer, EXS24, and, just to
spice things up a little, Sculpture. To begin with, I carried out the tests using the
base 1.42GHz Mac Mini with just the stock 256MB memory installed.

Starting with Platinumverb, along with

a single ES P instrument as a test
signal, I was able to run 15 stereo
instances with 84 percent User CPU
indicated in the Activity Monitor utility;
adding a 16th instance made the
display a little sluggish, and going up to
21 instances made the display totally
unusable, with a 95 percent User CPU
reading. By way of comparison, a
1.8GHz G5 iMac managed 46 The Mac Mini's slot-loading Combo drive
instances, while a 1.5GHz G4 offers DVD reading and CD-R and RW
Powerbook managed 20. reading and writing functions.

Moving onto Space Designer, the 1.42GHz Mac Mini managed four instances
comfortably, while a fifth instance caused problems even though the User CPU
figure was only at 86 percent. The iMac was able to run 17 instances in this
same test, and the Powerbook performed the same as the Mac Mini, also
running four instances.

Next I tried Logic's EXS24 sampler. Since the 256MB memory of the Mac Mini
wasn't enough to load my favoured harp from the Vienna Symphonic Library, I
opted for the 30.6MB Stereo Grand instrument from Logic's stock library. With no
filter enabled and the standard 16-bit storage mode selected in EXS24's
preferences, I was able to play 150 voices across three instances with a 95
percent User CPU reading. With a filter, this value went down to 64 voices from
one EXS24 instance, using 85 percent User CPU. Enabling EXS24's 32-bit
storage mode, where the Stereo Grand instrument now required 61.3MB
memory, it was possible to play 256 voices across four instances with 98 percent
User CPU, although the system was admittedly very sluggish at this point. With
the filter turned on, the polyphony dropped to 86 voices and 94 percent User
CPU.

As a comparison, using the VSL harp with 16-bit storage, the 1.5GHz Powerbook
G4 played 276 voices with the filter disabled and 84 voices with the filter enabled;
with 32-bit floating-point storage used instead, the same Powerbook played 660

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Apple Mac Mini

voices with the filter disabled and 106 voices with the filter enabled.

Looking at a slightly different Logic instrument as well for a change, namely

Sculpture, the Mac Mini was able to cope with five, eight-voice instances with the
'Attack Flute' preset (Logic users will note how dramatically the performance of
the instrument can change depending on the patch used) at 90 percent User
CPU. The audio still worked if I added a sixth instance, but the user interface
ground to a halt.

Adding More Memory

Since the performance of Mac OS X benefits from having more memory installed,
I repeated the same tests on my Mac Mini after installing a 1GB chip. With
Platinumverb I could comfortably run 21 stereo instances with 95 percent User
CPU, but with Space Designer I was only able to run the same four instances,
this time with 91 percent User CPU. With EXS24 the results in terms of
polyphony were identical, and the only real difference was that with Logic taking
advantage of more physical memory, its general performance was less sluggish,
even with the high polyphony of the EX24 in 32-bit floating-point storage mode
with no filters enabled. And using Sculpture I was able to get six eight-voice
instances running, with the user interface slightly sluggish, but not completely
killing the system, with 96 percent User CPU; five instances used 90 percent.

In these tests, somewhat surprisingly,

the extra memory made little
difference, although it has to be borne
in mind that none of them really made
intensive use of the extra memory. The
only real benefit was that the operating
system could use more physical
memory, making the whole system far If you're handy with a putty knife, it's possible
more responsive. Another important to save money by upgrading your Mac Mini's
consideration with these memory memory yourself, rather than choosing one
comparisons, however, is that the test of Apple's buy-to-order options.
songs were almost empty except for
objects required for the test: a 'real' song would actually require more memory.

The fact that with a relatively empty song, these tests show little difference
between having 256MB or 1GB installed opens up an interesting application for
the basic Mac Mini, which is to use it as an alternative to something like TC
Electronic's Firewire Powercore for running Logic instruments and effects via
Logic Pro 7's distributed audio functionality. Indeed, as covered in last month's
Apple Notes, setting up a network via Firewire and getting the Mac Mini to boot
automatically into Logic Node is fairly trivial, and the cost of the Mac Mini makes
it quite an affordable option for Powerbook users, as an example, who want to
run a few more Space Designer or Sculpture instances for around £400 — see
screenshots for an example of this in action. And as a bonus, you're actually

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Apple Mac Mini

getting another Mac that might have uses beyond simply running Logic Node.

Overall, it's hard not to like the Mac Mini. Aesthetically, it's great. Financially, it's
great. And for those users just getting started who might want to play around with
Logic Express or an M Box, it really is a great product. Even for more demanding
users, the Mac Mini can be valuable, for instance as a Logic Node system.
Normally the highest compliment a reviewer can pay a product is to say they
bought the unit after review; in this case, I bought the unit before writing the
review. And while I have to confess to loving new toys, the Mac Mini is a great
and fairly affordable toy that has surprising usefulness in the studio.

Build-To-Order Price List

512MB memory: £50.01.
1GB memory: £220.01.
80GB hard drive for 1.25GHz model: £30.01.
4x speed Superdrive: £70.01.
Internal Bluetooth module: £35.
Airport Extreme card: £49.
Bluetooth and Airport Extreme: £69.99.
Wired keyboard and mouse: £38.
Wireless keyboard and mouse: £69.99.
Applecare Protection Plan: £129.
Prices include VAT.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Apple%20Mac%20Mini.htm (7 of 7)9/27/2005 9:22:00 PM

 ART TCS

In this article:
Technical Specifications
ART TCS
Preset Convenience With Dual Compressor
Manual Control Published in SOS May 2005
Variable Voicing Print article : Close window
In The Studio With The TCS
Reviews : Microphone
V3 Verdict

ART TCS £305

pros
Combines the ease of Choose from optical and VCA compression, de-essing,
presets with the flexibility of
manual adjustment.
noise reduction, and valve voicing within a single
Offers true optical and VCA processor.
compression.
Built-in noise reduction.
Valve stage for added Paul White
warmth.
Extremely affordable.
cons
The manual doesn't make it
entirely clear how the controls
interact with the presets.
Photos: Mike Cameron
No side-chain listen facility.
summary
This is a good compressor for ART's new TCS dual compressor offers the benefits of both application-specific
beginners, because the 'character' presets and full manual control (including the ability to adjust the
presets are so easy to use, presets) when necessary. Unlike other preset-based machines, the presets here
but it is also flexible enough to
satisfy the needs of the more are as much about character as about defining exactly how the compressor is set
experienced user. up, so you can use the default settings as a starting point and fine-tune from
information there. Alternatively, if you're a bit wary of compressors, you can select a suitable
preset, then adjust only the Threshold control to achieve the desired amount of
£305.49 including VAT.
gain reduction. Furthermore, because compressors are routinely used to add a
Sonic8 +44 (0)8701
musical flavour rather than simply to control levels, the ART TCS includes both
657456.
+44 (0)8701 657458.
optical and VCA (Voltage Controlled Amplifier) gain-control stages to offer the
Click here to email
best of both worlds. Traditionally, VCA compression tends to be the most
transparent, while optical types that rely on a lamp and a photosensitive device
www.sonic8.com
are best known for their vintage character. There's also switchable valve circuitry
www.artproaudio.com
to provide more choice over tonality by inserting a 12AX7 gain stage into the
signal path.

Housed in a 1U steel rack case and powered directly from the mains via an IEC
connector, the ART TCS offers the user a choice of both balanced XLR and
unbalanced jack inputs and outputs operating at +4dBu and -10dBV respectively.
There's also a TRS insert jack that can be used to place other processors

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ART TCS

(usually equalisers) in series with the side-

chain. The front panel looks quite busy, but is in
fact very logically set out. First of all, a Level Technical Specifications
control adjusts the instrument input into the
Frequency Response: 10Hz-
compressor. Following this are the familiar 150kHz ±1dB
Threshold, Ratio (adjustable up to hard
limiting), Attack, and Release controls for THD: 0.01 percent at 1kHz
manual operation or preset tweaking. Note that Maximum Input Level:
there is no conventional make-up gain knob +20dBu
within the channel control area. Instead there Maximum Output Level:
are two knobs at the extreme right of the front +20dBu
panel that operate as stereo level and balance Input/Output Connections:
controls in stereo mode or as separate channel balanced XLR and balanced/
output-level controls in split mono mode. unbalanced quarter-inch TRS
jack
Side-chain Input: quarter-inch
TRS (ring send, tip receive)
Preset Convenience With XLR/Jack Input Impedance:
Manual Control 10k(omega)
Instrument Input Impedance:
The input Level control is linked to the high- 1M(omega)
impedance front-panel instrument-level input Output Impedance: 300
jack, and it has a 0-30dB gain range. Plugging (omega) (balanced)
anything into this instrument jack overrides the Dimensions 1.75 x 19.0 x 6.5
rear-panel input connectors. In addition to the inches (hwd)
compressor, each channel also features a Weight: 5.5lbs (2.5 kg)
noise-reduction section that can be either a
gate or an expander depending on which preset
mode is chosen. The only way to tell which is active is to observe the NR LED,
which will gradually dim when in expander mode and switch off more abruptly in
gate mode. The only user controls here are Threshold and Shelf, where the latter
controls a side-chain filter — when this is fully anticlockwise it makes the side-
chain most sensitive to lower frequencies, whereas the fully clockwise position
makes the side-chain sensitive predominantly to high frequencies. A central
setting gives a nominally flat response.

Gain reduction is monitored by an edgewise moving-coil meter, and though the

scale is unreadably small, it gives you a very good idea of what is going on.
Directly below this is a set of three buttons, one for bypassing the channel, one
for switching the valve circuitry in or out, and one for engaging or bypassing the
rear-panel side-chain insert points. This side-chain bypass feature is very useful,
as it allows the inserts to be permanently left connected to a patchbay or external
processor. To the right of the right-hand channel are two further buttons, one for
stereo linking the two channels and one for changing the two high-resolution LED
meters between reading the input level and reading the output level. These
meters have a peak-hold function and so are very useful for warning of potential
clipping problems. When stereo linking is selected, both channels are controlled
from the left-hand set of controls and the two side-chains are fed with the same
signal to avoid image shifts when compressing asymmetrical mixes.

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ART TCS

Variable Voicing

I've left the most important control until last, and that is the V3 voicing knob. This
operates a rotary switch to give different preset and manual processing modes
with either optical or VCA compression. Within the optical compression selection
there are settings for Bass, Mixes, Vocals, and Choral, as well as the fully
manual setting — the Vocals option includes a built-in de-esser for reducing
vocal sibilance, and different noise reduction (gate or expander) is used in each
case, as detailed in the manual. Switching to VCA mode reveals presets for
Vocals, Mixes, Percussion, and Bass, as well as the manual setting, and the
vocal preset includes de-essing again.

The remaining V3 knob settings use a

combination of optical and VCA gain
control. The first combination mode is
Stack, which combines both
compressor types in such a way that
the selected ratio value is effectively
doubled — you're really compressing
twice. The Sustain preset is designed
to do exactly that — add sustain to
guitars and other sources — while Nice
is a more extreme compressor with Here you can see the two gain-control
character. The Guitar knob position is elements used in each channel of the ART
a guitar-specific patch to add density TCS: a VCA (voltage-controlled amplifier) on
the left and an LED-based optical component
and to even out the sound.
on the right.

Finally three OPL presets use the

optical circuit for compression, teamed with the VCA circuit rigged as a limiter.
This is a nice combination, as you can dial in enough compression to allow the
sound to breath, but still catch those high peaks before they cause trouble. The
presets on offer are Leveling, Mix, and Vocals.

In The Studio With The TCS

The manual doesn't make it entirely clear exactly how the front-panel controls
interact with the presets, but looking at the gain-reduction meter, it's obvious that
they adjust whatever attack and release times are called up, which seem to differ
quite a lot between one preset and another. This is to be expected, as part of the
art of using a compressor is selecting the correct attack, release, and ratio
settings for specific instrument types. The tonality of the sound remains
reasonably consistent between presets, but the character of the compression
definitely changes, with the optical modes sounding smoother and more rounded
than the VCA alternatives.

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ART TCS

Having a vocal de-esser built in is also a nice touch, and though this isn't as
effective or transparent as a dedicated de-esser that attenuates only problem
frequencies, it is very welcome for the bonus feature that it is. The in-built noise
reduction also works smoothly enough and saves having to use a separate gate
or expander, though I couldn't find any means of monitoring the noise-reduction
side-chain filter or the effects of processors connected into the side-chain insert
point.

Switching in the valve circuit leaves the top end sounding much the same as
before, but there's definitely more weight and warmth at the lower end of the
spectrum. This function works equally well on guitar, bass, and vocals, and
seems to strike a good balance between being hyped and being too subtle to be
significant. On balance, the presets work pretty well for the sources for which
they're intended, and having the option to tweak them means you never have to
accept something that isn't quite right. I particularly liked the optical compressor
character, and the modes that combine this with a VCA limiter are particularly
useful, as you get the benefits of both processes in a single preset.

V3 Verdict

As stated at the outset, compressors are as much about adding character as

they are about controlling levels, and the ART TCS manages to do both jobs in
an efficient and straightforward way. As a gain-control device, it works much like
any other variable-ratio compressor, but the permutations of optical and VCA
voicings mean you can choose what type of character you need for each track or
mix you process. Having presets is a great way to get into compression without
making too many mistakes, yet you still have the opportunity to make manual
adjustments or even set up your compressor parameters from scratch if you feel
confident in doing that. No compressor will ever be entirely automatic, as the
Threshold level needs to be adjusted to take account of the input level and the
input signal dynamics, but with the system adopted here, it's very straightforward
to get musically consistent results.

Overall I like ART's approach to compression, and though there are more
esoteric compressors out there, few offer this amount of flexibility at this UK price
and few are easier to use thanks to the wide range of presets and the switchable
valve stage. There's even a side-chain filter on the noise gate/expander. You can
also use the ART TCS in stereo mode, as two completely independent mono
channels, or as an instrument DI with comprehensive compression, so there's no
shortage of applications. Given the very attractive price, I've no complaints.

Published in SOS May 2005

file:///H|/SOS%2005-05/ART%20TCS.htm (4 of 5)9/27/2005 9:22:02 PM

ART TCS

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/ART%20TCS.htm (5 of 5)9/27/2005 9:22:02 PM

 Audio Ease Rocket Science

In this article:
Roger That
Audio Ease Rocket Science
Into Orbit Effects Plug-ins (Mac OS)
Take Me To Your Leader Published in SOS May 2005
Fly Away Print article : Close window

Audio Ease Rocket Reviews : Software

Science £163
pros
Offers a range of interesting
and contemporary-sounding
effects that would be difficult
A new plug-in bundle from Dutch developers Audio
to achieve by other means. Ease brings together three unique sound-shaping
Available in a wide range of tools.
plug-in formats.
Easy to use, with great user
interfaces. Robin Bigwood
Excellent MIDI control
options.
Low processor hit.
There are some plug-ins you just have
Good manual and tech
to have: a neutral and a 'characterful'
support. EQ, a flexible reverb, a handful of
compressors, maybe a good delay.
cons
After that things get a bit more
For the money, it might be a
touch too quirky and
specialised in the murky world of filters,
specialised. tone processors and multi-band tools.
No parameter automation in Go further and you're at the outer
Logic except via MIDI. reaches, with granular synthesizers and
summary step sequencer-driven curiosities —
A classy suite of unique-
and it's here that Audio Ease's Rocket
sounding effects. What's not Science bundle belongs. This is a
to like? collection of three apparently dissimilar
information plug-ins which can do some interesting things to your tracks that would be
difficult to achieve in other ways. Just how useful they are might depend very
£163.33 including VAT.
much on the type of music you make and how willing you are to embrace the
Unity Audio +44 (0)1440
785843.
unfamiliar in your pursuit of new sonic horizons, but they're certainly an intriguing
+44 (0)1440 785845. bunch.
Click here to email
www.unityaudio.co.uk Rocket Science is available in HTDM, RTAS, VST, MAS and AU formats for OS
www.audioease.com X, and in MAS format for OS 9. Installation on my own system was a brief and
painless affair, and a decent PDF manual is included — not that you'll need to
Test Spec refer to it very much. Authorising the plug-ins is done via an on-line challenge/
response system, so you can do it immediately, and Audio Ease provide you with
Rocket Science v3.02. a couple of authorisations in case you need to run Rocket Science on your laptop
Dual 867MHz G4 Power Mac as well as your desktop Mac, for example.
with 1.25GB RAM, running
Mac OS 10.3.7.

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Audio Ease Rocket Science

Tested with MOTU Digital

Performer v4.52. Roger That

Roger is a 'Multiple Gender Vowel Bank'. It's not Roger himself who's of multiple
gender: the term is explained by the presence of his companions Cindy and
Patty. Together they provide filtering effects that make it sound as if your audio is
to some extent being 'pronounced' by voices of three different pitches.

Taken on their own, the individual

vowel sounds on offer just add what
sounds like a very specific EQ curve,
perhaps with some 'formant' resonance,
to whatever audio is passing through
the plug-in — a strong effect but not
terribly exciting. Roger comes to life,
though, through making transitions from
one vowel to another, as it's then that
the effect becomes really audible. You
can choose vowels manually, and if
your host sequencer supports it, record
the changes as automation data, while Audio Ease also provide Roger with a
step sequencer that switches between vowels automatically. As you'd expect,
this can be flexibly sync'ed to the host sequencer's tempo, and the resulting
rhythmic effects can be enhanced by tweaking the Beat Skips parameter. This
causes Roger to not always change vowel on every beat, instead introducing
some syncopation and cross-rhythms into the proceedings. The Portamento
parameter determines the speed at which one vowel 'morphs' into another.

The quality of the vowel sounds themselves is controlled by the Bandwidth

parameter. At its maximum value no filtering takes place at all, and when set to
its minimum, Roger breaks down into high-resonance ringing nonsense that
almost completely obliterates the input signal. Between these two extremes
there's plenty of usability. You have to hear the sound to really appreciate it, but
it's like a combination of a swept band-pass filter, a phaser and a vocoder, and
there's something undeniably charming about it that makes it into somewhat
more than purely a novelty effect, though at times it can be downright funny. It
works extremely well on pad and sustained sounds but can sound great on whole
mixes, on drums, and on vocals, too. Having said that, though, the full vowel
effect is only achieved at the expense of most of the high-frequency content of
the input signal — it's certainly a plug-in that draws attention to itself.

The final trick up Roger's sleeve is MIDI control. As long as your sequencer can
send MIDI to plug-ins, you can choose between all 30 vowel types (10 for each
character) by playing MIDI notes between C2 and B4, with the Bandwidth and
Portamento values being set by key velocity or cc1 (mod wheel). This type of
vowel selection, in particular, can make for some rhythmic effects which sound
absolutely superb.

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Audio Ease Rocket Science

Into Orbit

Having vowel-filtered your sound, what better than to hand it over to a

psychoacoustic flight-path simulator? Orbit is a processor that combines reverb-
like early-reflection generation with filtering and Doppler effects to 'localise' the
sound it's treating in a virtual room of anything from 8 x 8m to 80 x 80mm — it is,
in effect, a sort of 26th-century panner.

Your own position in the virtual room is set by dragging an icon showing two
speakers either side of what appears to be the planet Saturn (why not?), whilst
the audio you're treating is shown as a red ball. This always represents a mono
sound source, so if you place Orbit on a stereo track the two channels are
summed to mono in order for the location of the red ball to be meaningful. Having
set the distance your speakers (or indeed headphone drivers) are apart — which
can make quite a difference to the final sound — you can then start
experimenting with different room sizes and sound-source placements.

Orbit can calculate up to 60 early

reflections, which represents the dry
signal bouncing off the four virtual walls
up to a maximum of five times. To save
processor power you can choose to
make Orbit calculate fewer reflections,
or even none at all, but even with
comparatively few reflections the sense
of location can be uncanny, responding
as it does to your settings. The High
Damp parameter makes the walls more
or less absorbent, so you can choose a bathroom or airing-cupboard ambience,
or anything in between.

It is possible to just set your speakers and your sound source's positions and
leave it at that, but Orbit goes further, and lives up to its name, by offering some
movement paths for the sound source via a colourful pop-up menu. The sound
source can move smoothly from one position to another, as determined by your
mouse clicks and the Speed parameter, or continuously around the perimeter of
an adjustable oval (in either direction). It can also, as the Rocket Science manual
wryly observes, 'move randomly about like a drunk' within the oval. Where this all
gets interesting is when you put the 'listening position' inside the oval path of the
sound source, so the sound source seems to encircle you — that's where the
psychoacoustic filtering comes in. It's also possible to place the listening position
right on the sound source's path, which can make it sound like the sound is
passing right through your head. Additionally, with the right room size and
movement speed, Orbit will do a fire-engine-like Doppler effect.

Orbit is a very interesting alternative to a conventional panner for mono tracks:

even when set to produce no early reflections, it does seem to introduce a 2D

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Audio Ease Rocket Science

element in a way that a conventional panner can never achieve. Linked with a
really good reverb, perhaps Audio Ease's own Altiverb, I could see this as being
a viable tool for speech recording post-production, for example. It works nicely for
instruments, too, creating a tangible sense of separation. The 'moving' modes
are impressive but could clearly get tiresome if used for too long.

Take Me To Your Leader

Follo, an 'Energy Driven Band Booster', is another filter effect, but it's very
different to Roger. It's a resonant band-pass filter whose cutoff frequency is
determined by the input level to the plug-in and limited to a frequency range set
by a couple of sliders. The sensitivity of the cutoff frequency to input level is set
with the Analysis parameter, the overall strength of the effect with the Bandwidth
knob, and what can only be described as the 'squelchiness factor' by the Release
time. If this doesn't seem terribly complex that's because it's not — Follo is
brought to life by the audio it treats, and then it's simply a case of tweaking the
relatively few parameters to tease out the effect you want.

Audio Ease give two prime examples of Follo's possible applications — as a

guitar auto-wah, and as a 'bass generator', presumably for bass drum tracks.
There's no doubting this plug-in comes from the planet 'wah' but I did find it could
work on a surprisingly wide range of input material, from strings to synths, and it
can do some deliciously evil things to individual drums, even to a wide-band
drums submix. It's also good combined with other effects — Audio Ease's
suggestion of putting it after a delay works very well, especially as the gradual
decay causes Follo to respond in a slightly different way for each repeat of the
signal. I also tried putting Follo after a reverb which was set up in 'aux' rather
than 'insert' fashion — the effect was very nice, and very novel.

Fly Away

There's a lot to like about Rocket Science. To some extent the individual plug-ins
are one-trick ponies, in the way that some off-the-wall freeware efforts often are.
But that's where any comparison with freeware or shareware ends: Roger, Orbit
and Follo absolutely ooze quality, as much from their user interfaces as from the
often ravishing sounds they conjure up. Just as with Riverrun, a plug-in from
Audio Ease's Nautilus bundle which is without doubt one of my 'desert island'
effects, Rocket Science might not be what you reach for on every project, day-in,
day-out, but it has an uncanny ability to inspire and refresh, and also to create
some extremely novel and distinctive sounds. Also, the effects are not so limited
in scope that you'd hardly ever use them. I could imagine whipping out Follo and
Orbit fairly often, and Roger as often as I could! It helps, too, that the plug-ins run
very efficiently, even on my low-end dual G4, so you can experiment with them
without feeling they need to be handled with kid gloves.

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Audio Ease Rocket Science

Gripes? I don't have any, but it should be pointed out that AU versions of the
Rocket Science plug-ins don't support automation via manipulation of their
graphic interface controls. This could be a problem for some Logic users, but it's
possible to control virtually every parameter and setting with MIDI messages, and
the manual contains exhaustive information on this.

Perhaps a harder question to answer is whether Rocket Science is worth the

£160 asking price. For plug-ins that aren't of the bread-and-butter variety that's
quite a lot, and certainly lifts the bundle out of the 'whimware' category. But you
are getting the broad plug-in format compatibility as well as quality, and
additionally, Audio Ease's customer support is up there with the very best.

The bottom line is that Rocket Science won't appeal to everyone, and doesn't try
to, but for those creative types who are always on the lookout for something
unusual and distinctive it could fit the bill perfectly. It's always a pleasure to work
with software of this quality, and Rocket Science is certainly an extremely
welcome addition to my own plug-ins folder.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Audio%20Ease%20Rocket%20Science.htm (5 of 5)9/27/2005 9:22:05 PM

 Audix SCX25

In this article:
In-depth Studio Tests
Audix SCX25
Self-noise & Polar Pattern Condenser Microphone
Verdict Published in SOS May 2005
Print article : Close window
Audix SCX25 £595
pros Reviews : Microphone
Very compact size allows
easy placement, especially in
pianos.
Effective internal
shockmount. Large-diaphragm mics don't have to be bulky and
A slightly lean but natural unwieldy, as this compact new contender from Audix
sound that cuts through a mix. demonstrates.
cons
Subcardioid polar pattern.
Restricted SPL capability. Hugh Robjohns
No internal pad facility.
summary The Audix SCX25 is a large-diaphragm
A very compact large- studio condenser mic with a fixed
diaphragm capacitor mic with cardioid polar pattern, but housed in an
a slightly lean sound, boasting unusually small body. While there will
a very effective internal always be the need for physically large
shockmount. Minimal
microphones to help make the vocal
proximity effect coupled with a
limited bass extension allows talent feel cherished, large-diaphragm
close-miking, and a high mics in compact cases are a lot easier
presence peak provides a to place, and far more visually discreet.
sense of air and detail. Photo: Mark Ewing
The SCX25 represents the top of the
information company's capacitor mic range, and is produced entirely in America. It is shipped
SCX25, £595; SCX25 in a smart foam-lined wooden box within a cardboard outer sleeve, and is
matched pair, £1395; D-Clip accompanied with a sheet of generic specs and suggested applications.
mounting hardware, £8.49.
Prices include VAT.
Stirling Audio +44 (0)20
A side-address microphone, it displays more than a passing resemblance to a
8963 4790. lollipop — a 50mm disk supported from a 95mm stem. The physical design is
+44 (0)20 8963 4799. claimed to minimise acoustic reflections and diffractions, and enables the
Click here to email capsule head to be positioned with ease even in congested areas. The
www.stirlingsyco.com microphone is mounted to a stand using a simple clip-on bracket (complete with
3/8-inch thread adaptor) which slips onto the slim 20mm-diameter stem. The
www.audixusa.com
SCX25 is not supplied with a shockmount adaptor, but Audix claim that it doesn't
need one thanks to a patented capsule suspension within the housing disk.

The microphone body is made of brass, anodised matt black, with a highly
polished frame around the capsule and at the top of the stem body. The solid-
state, transformerless impedance-conversion electronics are contained within the
stem, which terminates in the usual three-pin male XLR connector. The front of

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Audix SCX25

the mic is indicated by the Audix name and model number etched into the brass
stem just below the capsule.

The capsule is hard to see through the dense but slightly flexible wire grilles on
either side of the disk, but it is apparently one inch in diameter with a five-micron,
gold-sputtered mylar diaphragm. This is a true externally polarised capacitor mic
that requires phantom power (quoted as 48-52V, but no figures are given for
supply current). The sensitivity is a generous 27mV/Pa, with self-noise of 14dBA,
and a maximum SPL of 135dB (for one percent distortion). The frequency
response is given on the supplied spec sheet as simply 20Hz-20kHz, but
elsewhere I found a rather wide tolerance range of +5/-2.5dB. What you see is
what you get with this mic — the polar pattern is fixed, and there are no high-
pass filter or pad switches. The mic weighs 170g, but much of that weight is in
the capsule, making it noticeably top-heavy.

In-depth Studio Tests

The obvious starting point in forming an opinion on the SCX25 seemed to be to

compare it with the Microtech Gefell M930, since they share similar physical
attributes, but I also put it up against my trusted standards: the Neumann
TLM103 and the small-diaphragm Sennheiser MKH40. All were hooked up to a
GML mic preamp as the front end, and then through a Yamaha DM1000 mixer
and on to PMC monitoring.

The first thing to test was the internal

shockmount. The M930 isn't supplied
with a shockmount as standard
(although a rubber 'doughnut' stand
adaptor is optionally available and
works very well), but my TLM103 sits
in a full cradle, and the MKH40s have
shockmounted clips as standard. I
mounted these mics on adjacent
stands and, having adjusted input
gains appropriately, I was impressed
with the isolation from mechanical
vibrations achieved by the Audix mic.
Far better than the M930, and very In comparison with many other large-
similar to the performance of the diaphragm mics, the SCX25 is remarkably
shockmounted Sennheiser and compact, and it's therefore much easier to
Neumann mics. position in cramped spaces, for example
under a piano lid.

Next up was a listening test to male

voice — mine, in fact! Working at a distance of about 30cm, the sound came over
as clean, slightly lean, and with a hint of added presence or air. It didn't appear to
have the bass extension of the M930 or the MKH40, and even the TLM103
sounded warmer. Neither did it sound quite as open at the extreme top end, and

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Audix SCX25

the overall impression was that it sounded slightly 'smooth' and light — no bad
thing in the right circumstances.

The microphone exhibits relatively little proximity effect and I found it possible to
work far closer with the SCX25 than with other mics before the sound became
excessively boomy, something that was particularly noticeable on acoustic guitar.
However, recording close vocals with the SCX25 demands the use of a proper
pop screen — it is simply not possible to work without one.

On a 12-string acoustic I found the SCX25 tended to lose control of the complex
harmonics and delivered a slightly confused, almost harsh sound in comparison
to the reference mics, but as a percussion overhead it worked well to provide
clean, clear detail with crisp transients, without excessive splashiness from
cymbal crashes. However, on a couple of occasions there was a hint of transient
clipping in this application, which I'm fairly sure can be attributed to the mic rather
than the preamp, so this may not be the ideal overhead mic if you are working
with a powerful drummer.

Self-noise & Polar Pattern

The relatively high (by modern standards) self-noise was evident when compared
directly with the TLM103 and the M930, and it was also a brighter hiss. To be
fair, in musical applications I doubt the mic's self-noise will ever be an issue, but
the SCX25 might not be the best choice for spoken-word applications.

The polar pattern was, I found, subcardioid at low and mid-range frequencies,
and almost omni at the high end, albeit with a significant flattening of the sides at
extreme high frequencies. The latter is to be expected with a large-diaphragm
mic, but the poor cardioid shape across the entire range was something of a
disappointment, and made the mic much more prone to picking up unwanted spill
than my other references. When used for solo vocals, for example, the SCX25
captured noticeably more of the room sound than the M930 or TLM103. The rear
null was never particularly deep, reaching maybe 10dB around 1kHz and less
everywhere else. However, the off-axis response is quite smooth and reasonably
neutral, so at least the spill doesn't sound too coloured or phasey.

One of the specialist applications recommended for the SCX25 is that of miking
pianos, and in this role the mic's attributes work well in its favour. The slim design
allows it to sit above the strings of a grand piano very neatly (or within the body
of an upright piano), and with suitable miniature stands the lid can be supported
on the short stick or even closed completely. The insignificant proximity effect
and the slightly curtailed bass also help to prevent the sound from becoming
boomy, and the integral shockmount minimised mechanical damper noise. Of
course, close miking of this kind doesn't suit every musical genre, but where a
'pop sound' is appropriate, the Audix mic works very well indeed, producing a
detailed but clean and precise result. I used a small table stand sitting on the
soundboard to support the mic in a grand piano, but Audix can supply the

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Audix SCX25

optional D-Clip for the SCX25 which is designed to clamp onto the dividing bars
of a grand piano's frame.

Finally, I tried the SCX25 as a woodwind mic in a test recording of a school

orchestra. Its wide pattern and lean sound worked well, capturing a clean and
delicate recording with plenty of character and 'wood'. In fact, I could easily
become quite attached to the Audix in this role.

Verdict

In conclusion, the SCX25 is an interesting mic with an unusual blend of qualities.

Its small size makes placement easy, and the internal shockmount works
surprisingly well. It provides a high output level, ideal for preamps with limited
gain, and although it's noisier than the best it is quiet enough for all normal
musical applications.

The lack of proximity effect allows close placement, but I would place a question
mark over the maximum SPLs it can handle, and also over it's poor directionality.
The tonal quality tends towards the lightweight, with a noticeable, though smooth,
high presence peak, but this will often help instruments to cut through in a mix
without you having to resort to the EQ. The mic also tends to suit male vocalists
with 'deep brown' voices more than female and high male voices, and would
make a good alternative for those who find the TLM103 too thick or boomy.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Audix%20SCX25.htm (4 of 4)9/27/2005 9:22:07 PM

 Bornemark Broomstick Bass

In this article:
Broomstick Basics
Bornemark Broomstick Bass
Bass Box Virtual Bass Player Instrument (Mac/PC)
Touching All The Basses Published in SOS May 2005
Auto Pilot Print article : Close window
Manual Control
Reviews : Software
Big Bottom
Style File
Verdict

Bornemark Broomstick
So how is your plug-in band coming on, then? With a
Bass £169
combination of Steinberg's Groove Agent and Virtual
pros
Guitarist plus Yamaha's Vocaloid, all that is required
Very well designed and
easy to use. is a bass player. Enter, stage left, Bornemark's
While Auto mode is the key Broomstick Bass...
selling point, Manual mode is
also excellent.
Good value for money John Walden
considering the range and
quality of the sample library
alone. Given how well received products such
cons as Virtual Guitarist and Groove Agent
were, it is perhaps surprising that a
Relatively modest effects
section. 'virtual bass player' hasn't appeared
Could benefit from more hip-
before now. Just like buses, you wait
hop, R&B or other modern for ages for one and two arrive
styles. together. Off the mark slightly before
summary Steinberg's own Virtual Bassist is
Bornemark's Broomstick Bass Bornemark's Broomstick Bass. The
provides a solid and reliable Bornemark name might not immediately
virtual bass player. While the be familiar to many SOS readers, but
auto-accompaniment is Sven Bornemark led the production
perhaps the main selling teams for both Groove Agent and
point, the quality of the bass
sounds, plus their various Virtual Guitarist, so his pedigree is well
articulations, means that this established.
plug-in also makes an Broomstick Bass in Auto mode with its
excellent 'manual' sample- namesake instrument selected. The Memory
based bass instrument. Of course, musical auto- Tab at the top of the window allows 16
accompaniment is not a new idea, but snapshots of the plug-in settings to be
information saved for later recall.
the type of 'virtual band member'
£169 including VAT. provided by Groove Agent or Virtual
MI7 +44 (0)1446 754350. Guitarist has taken the process to a new level by including a high-quality sample
+46 40 6992509. collection. Whatever your own personal take on auto-accompaniment in the
Click here to email creative process, with virtual guitarists, drummers and singers already available,
www.mi7.com bass players were the next obvious target. So is Broomstick Bass a valuable
www.bornemark.se 'session musician in a box' or a bunch of 'cheesy keyboard-style' presets? Let's

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Bornemark Broomstick Bass

get the lowdown...

Test Spec
3.2GHz Pentium 4 PC with
2GB of RAM running Windows
XP Pro (Service Pack 2), with
Echo Mia 24, Egosys Wami
Broomstick Basics
Rack 24 and Yamaha
SW1000XG soundcards. Broomstick Bass is constructed from four elements. First, a large sample library
Steinberg Cubase SX v3.0.1. (just over 800MB) has been built from 21 different bass instruments and is
divided into four main types; acoustic, electric, keyboards and pedal basses.
Steinberg Groove Agent 2.
Incidentally, the plug-in takes its name from one of these instruments — a
Broomstick Bass version homemade 'bass' consisting of a metal bucket, a broomstick and a piece of string
reviewed: v1.0.0.
(see the screenshot on the left for an image of this). A full list of the instruments
is provided in the 'Touching All The Basses' box elsewhere in this article.

Second is the auto-accompaniment section. This consists of dozens of individual

styles, each grouped into a general musical genre (see the 'Style File' box over
the page for details). As described more fully in a moment, each individual style
actually consists of eight variations (bass riffs or patterns) and these are adjusted
automatically to fit the chords arriving at the MIDI input.

The third element is the DSP section. This provides a collection of the more
common processing options used with bass sounds and includes a three-band
EQ, a pitch-shifter, compressor, chorus and overdrive. Finally, a Manual mode is
included where Broomstick Bass can be used as a playable virtual instrument
using any of the sampled instruments from the library.

Bass Box

Bornemark recommend that Broomstick Bass is run on a minimum 800MHz

Pentium III PC under Windows XP, or a 600MHz G3 Mac under Mac OS v10.3,
both with at least 512MB of RAM and 850MB of disk space. Both platforms
require a DVD drive and a VST- or AU-compatible host sequencer (the latter on
the Mac only, obviously).

The plug-in itself is provided on a single DVD-ROM (not a CD-ROM) with a slim
printed manual. The latter is very well written and covers both the operation and
concept/background to the instrument. Installation is simple, and allows user
selection of the sample library location. Registration of Broomstick Bass can be
completed on-line using the supplied serial number, and this provides access to
updates and extras. These already include a 'gift pack' (a 14MB download)
containing samples for two additional instruments and a dozen additional styles.
Both the manual and the web site suggest that further add-ons will be made
available to registered users at no charge — an attractive feature of the product.
After installation, the plug-in was recognised without a problem by Cubase SX on
my test system.

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Bornemark Broomstick Bass

The main screen of Broomstick Bass is

divided into three areas. To the right, the
user can select the required instrument and
this displays an image of the selected
instrument. This section also includes the
master volume control, a switch for Poly/
Mono mode and smaller knobs for adjusting
Glide, Release and Release Noise. The
Release simply changes how long a sample
is sustained after a note is released, while
for added realism, the Release Noise
control attempts to add more (or less) of the
finger noise created as a note is released.

The bottom section of the window contains In Manual mode, Broomstick Bass
a virtual keyboard. This can be used for offers a selection of different
triggering patterns or individual notes if an performance articulations.
external MIDI keyboard is not available. The
shaded area of the keyboard is the 'Control Octave' and this is used to select
patterns in Auto mode or articulations in Manual mode. Both of these modes are
described more fully below.

The left side of the window is

somewhat busier. This includes the
Auto/Manual switch which flips Touching All The Basses
between the two main modes of
Aside from the homemade
operation. At the top of this section is
broomstick bass from which the plug-
the Style selector used in Auto mode. If in takes its name, Bornemark
the Link button is on, selecting a sampled 20 different bass
particular style also loads the instruments. These are as follows:
appropriate instrument. The Memory
tab brings up 16 user slots where you KEYBOARD BASSES
can save the current configuration of Fender Rhodes Piano Bass.
Broomstick Bass. You can automate Minimoog (four different sounds).
the process of switching between these
ARP 2600 (two different sounds).
with MIDI continuous controllers, so the
tab provides a simple way to move ARP Odyssey (two different sounds).
between different setups mid-song. ARP Omni (the bass preset).
This can include a change in the
Yamaha DX7 (preset Bass 1).
instrument used, but this may well
cause a slight glitch as it involves ELECTRIC BASSES
loading the sample data on the fly. Fender Precision (picked and fingered,
Running two instances of Broomstick both damped).
Bass provides a workaround to this
Fender Jazz (picked and fingered).
problem, as the plug-in only produced a
CPU load of three to five percent when Gibson Thunderbird (fingered, amped).
running under Cubase SX on my test Rickenbacker 4001 (picked, amped).
system. A further useful feature is the Hagström H8 (picked and damped).
Metronome button (with the drum kit
icon) and this provides a basic drum Music Man Sabre (picked and slapped).

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Bornemark Broomstick Bass

pattern (in a suitable style) to audition Chapman Stick (tapped).

the bass lines against. While it will not Ashbory Bass (fingered).
replace something like Groove Agent, it
Manne Acoustibass fretless (fingered).
is certainly helpful to have.
ACOUSTIC BASSES
Broomstick bass.

Auto Pilot Double bass (bowed and fingered).

Tacoma Thunderchief (picked).
Of course, the most obvious selling PEDAL BASSES
point of Broomstick Bass is its ability to Church organ bass pedals.
act as a virtual bass player. In use, the
basic operation of Auto mode is very Hammond B3 bass pedals.
straightforward. Broomstick Bass Moog Taurus (Taurus preset).
recognises basic chords fed to it via The first free 'Gift Pack', which can
MIDI and, on the basis of the chord be downloaded from the Bornemark
type, it will play a suitable bass line web site, includes two new
dictated by the selected style. Each instruments; a Hofner 500/1 bass
style contains eight variations and, in (made famous by Paul McCartney)
many of the styles, these are based and a suitably aggressive Elektron
around a similar riff or pattern, with SidStation patch.
higher variation numbers containing
more complex playing. You can switch between these eight variations via the on-
screen display (the green Reference section or the grey Control Octave can be
used). More generally, however, this would be done using the Control Octave
from an external MIDI keyboard.

The 'Stop' and 'Bar 1' options within the Control Octave provide further
possibilities. The 'Bar 1' switch forces any of the longer patterns (some run to
four bars) to play only bar 1, while holding down the 'A' key stops the auto bass
line until another key outside the Control Octave triggers it again. Usefully, while
holding down the 'Stop' key, you can play your own bass line (similar to working
in Manual mode). The modulation wheel on a MIDI keyboard also provides
access to this basic 'manual' mode; pushing the mod wheel towards the top of its
travel stops the auto playback and allows you to play the plug-in as a normal
bass instrument. Additionally, with the mod wheel set between 10 and 90 percent
of its travel, the pattern engine will just play repeated notes (usually the root note
of the current chord) and this can be very useful for linking between chords or
over rapid chord changes. Bringing the mod wheel back down to below 10
percent of its travel restarts Auto mode.

Other useful Auto mode features include the Latch option. With this on, a pattern
can be triggered by hitting a chord and, even if that chord is released, the pattern
will continue until the next chord is played. The Speed button allows patterns to
be played in half-time or double time, while the Shuffle knob can be used to add
swing to a straight bass riff (turn the knob clockwise) or to straighten out a bass
line that is already swinging (turn the knob anticlockwise). Just as Groove Agent
can output its rhythms over MIDI, Auto mode in Broomstick Bass includes the
ability to output the bass lines created as a MIDI track. I've always found this

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Bornemark Broomstick Bass

really useful in Groove Agent, as it

allows you to fine-tune the auto-
generated parts by, for example,
groove quantising. This MIDI output
process is well explained in the manual
and worked a treat in Cubase SX.

The whole operation of the Auto mode The MIDI output function allows the auto-
is very well thought-out and, once you generated bass line to be recorded (the
have a basic chord progression for lower track) from a sequence of chords (the
upper track). This worked flawlessly in
your song, creating a suitable bass line Cubase SX.
is a breeze — even if you do then
decide to edit this further. At first
glance, I was only really surprised by one feature of the Auto engine — the rather
limited range of chord types that are supported. Essentially, Broomstick Bass will
play patterns based on only major, minor and 7th chords and, while a much fuller
range of chord types are recognised by the software, they are essentially
truncated to one of these three basic types. Even using this fairly narrow chord
palette (which, incidentally, is the same as Virtual Guitarist Electric Edition with
the exception of sus2), Broomstick Bass is capable of producing solid, credible
bass lines. Aside from real jazz aficionados, most users are unlikely to find this a
significant restriction.

Manual Control

While Auto mode might be the main selling point of Broomstick Bass, the Manual
mode is also well featured. In this mode, Broomstick Bass operates as a normal
sample-based virtual instrument and, as well as velocity-sensitive sample layers,
each of the instruments also contains articulation layers, accessed via the
Control Octave. As well as the 'normal' sustained notes, Staccato, Slide Up,
Hammer On/Off, Slide Down and Legato articulations are provided. For the
acoustic and electric basses, further options of Fret Noise, Ghost Note (a heavily
damped note where the pitch is unclear) and Smack sounds can also be added
for additional realism. While this type of 'key-switching' system can add great
flexibility to a performance, it does require considerable practice to make it work
effectively.

When playing the same note in quick succession (for example, a simple bass line
playing eighth notes on the root of the chord) 'machine-gun' effects can often
befall sample-based sounds, as it soon becomes obvious that the same sample
is being played repeatedly. Broomstick Bass avoids this problem as the playback
engine detects repeated notes. For normal notes, re-pitched adjacent samples
will be thrown in to add variety, while for Staccato playing, a random selection of
four samples for each note are used. This end result does sound very natural,
with enough subtle variation to fool the ears.

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Bornemark Broomstick Bass

Big Bottom

So much for the technical details — what do the various instruments sound like?
The simple answer is very good indeed. With the exception of the broomstick
bass itself (for which I'm not sure I could find a genuine musical application!), all
the instruments are eminently usable and appear to have been well recorded and
edited. The recording methods obviously varied; while some have been recorded
via a mic/amp combination, others seem to have been recorded direct. The
manual includes an interesting section that details some of the methods used.

For a number of the instruments,

several different versions are provided.
For example, four versions of the
double bass are included, providing
fingered and bowed sounds, both with
or without vibrato. This variety is useful
but also necessary as, despite some
relatively minor sound-shaping
possibilities described below, there is
little that can be done to edit the
sounds themselves.

That point accepted, there are some

absolute gems here. For example, if
you want solid and reliable electric The Edit screen provides access to both
bass, the Fender Precision and Fender effects and some general settings as well.
Jazz basses are pretty much spot on.
In contrast, for something a little more
aggressive, the Gibson Thunderbird and Rickenbacker 4001 deliver the goods,
as both of these have been recorded via an amp. I also liked the very solid sound
of the Chapman Stick, while the 'slapped' version of the Music Man Sabre has
plenty of character. Of the keyboard and pedal instruments, perhaps my favourite
was the wonderful Hammond B3 bass pedals — instant 'Green Onions'. Given
how 'playable' the articulations make these instruments, the Broomstick Bass
sample set is probably worth the price of entry on its own.

The Edit screen provides access to various bass-orientated effects which are
both simple and effective in operation. The EQ section permits a reasonable level
of tonal control if some minor fine-tuning is required to make the bass sit within a
mix. The compressor and overdrive work well enough but don't overcook things,
while the chorus sounds really nice and can easily add a little movement and
character to any of the sounds. The pitch-shifter, while working well enough with
the keyboard-based instruments, did not produce very usable results with the
acoustic and electric basses. All these controls can be automated and the MIDI
controller numbers are listed in the manual. The Edit screen also includes
various settings that control how Broomstick Bass operates. For example, the
MIDI Output function can be enabled here. Depending upon the size of your
master keyboard, the other useful option provided is the ability to move the
position of the Control Octave to any one of four positions.

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Bornemark Broomstick Bass

Style File
There are several dozen individual styles included with Broomstick Bass. These
are grouped into a number of broad musical types including jazz, rockabilly,
boogie, blues, pop, rock, prog rock, reggae, funk, dance, and fusion. There are
also a few stranger headings, such as 'Classic Synth' and 'Odd Meters', and a
group named 'Streets Of....' which includes a small selection of bass styles from
around the world (eg. Rio, Dublin, Dakar).
The blues and rock styles are pretty safe and solid but right on the money, while
the funk styles contain some excellent material with plenty of slapping. The classic
synth and dance groups include great '70s and '80s moods, particularly the Disco
styles. While there is plenty of choice, I wonder whether Bornemark might add
some more styles aimed at hip-hop, R&B and other modern styles. These could,
of course, be easy targets for future 'Gift Pack' downloads.

Verdict

In Broomstick Bass, Bornemark have delivered the kind of characteristics that

most band members would think of as ideal in a real bass player; it is solid and
dependable, plays in time with the drummer, always uses an appropriate bass
sound and is happy to turn down (or off) when asked. I didn't experience a single
glitch during the whole of the review period, and this gives plenty of confidence in
the programming work done. The user interface is very straightforward, and the
large range of styles and the quality of the sounds are commendable.

As with any auto-accompaniment function, what you can't really expect is

virtuoso playing that, all of a sudden, is going to spring a musical surprise. In
Auto mode, what Broomstick Bass does provide is a solid and reliable bass line.
If more colour is required, then Manual mode includes enough creative
possibilities to get the job done. While I'm sure I will still be turning to my Fender
Jazz for many of my own needs, I'm also sure there will be times when
Broomstick Bass will do the job just as well and with a minimum of fuss.

By the time you read this, Steinberg's Virtual Bassist may also be available and it
will be very interesting to see exactly how Virtual Bassist and Broomstick Bass
compare — look out for a review of the former in a forthcoming SOS. However, if
you are keen to get a virtual bass player on board as soon as possible, then
Broomstick Bass is a pleasure to use and comes highly recommended. A 15MB
downloadable demo is available from the Bornemark web site for those that wish
to try before they buy.

Published in SOS May 2005

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Bornemark Broomstick Bass

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 CEDAR Audio Duo Auto Dehiss

In this article:
Duo Auto Dehiss
CEDAR Audio Duo Auto Dehiss
Setting Up Broad-band Noise-reduction Processor
Action Stations Published in SOS May 2005
Duo Auto Declickle Print article : Close window
Performance
Reviews : Processor
CEDAR Audio Duo Auto
Dehiss £4113
pros
Fast, accurate automatic This slimline new rack processor makes CEDAR's
configuration, with flexible fine-
tuning facilities latest class-leading audio-restoration algorithms
Options to process channels remarkably quick and easy to use.
separately or in Middle &
Sides format.
Phenomenally effective Hugh Robjohns
noise-reduction algorithm.
No audible side effects if
used sympathetically.
For anyone involved in the specialised crafts of audio restoration and 'forensic
audio', CEDAR Audio will be a familiar name. This Cambridge-based company
cons
have long been the standard bearer for the kind of highly specialised digital
This degree of signal-processing tools required in these professions. In recent years, the
sophistication doesn't come
cheap.
advances in computer processor speeds and general DSP functionality have
enabled many manufacturers to develop their own audio-restoration programs,
summary
and many are very cost effective, but to date no-one has quite managed to match
The CEDAR Audio Duo Auto the class-leading functionality and quality of the highly evolved CEDAR
Dehiss represents another
step forward in the technology algorithms.
of hiss-removal tools. Both
the hardware and the Most of CEDAR's restoration tools are available on a variety of software and
algorithm have benefited from
updating, enabling this new
hardware platforms. The company's flagship software platform is called the
unit to offer even more CEDAR Cambridge, a PC-based system which can be equipped with a wide
impressive performance range of the latest signal-processing 'modules' to suit a variety of restoration and
controlled through a forensic applications. Many of the same core modules are also available as plug-
remarkably simple but flexible ins for DAW platforms including AMS Neve workstations, Merging Technologies
user interface.
Pyramix, Studio Audio & Video SADiE, and Sydec Soundscape. However, this
information review is of CEDAR's latest generation of hardware, the new Duo series, which
£4112.50 including VAT. effectively supersedes some of the previous Series X rackmount units.
CEDAR Audio +44 (0)
1223 881771.
+44 (0)1223 881778.
Click here to email
www.cedaraudio.com
Photos: Mike Cameron

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CEDAR Audio Duo Auto Dehiss

Duo Auto Dehiss

The Series X Dehisser was called the DHX (reviewed in SOS July 2000). It
boasted a 50 megaflops DSP engine, with all-digital I/O and three simple rotary
controls to adjust the critical dehiss algorithm parameters (level, variance and
attenuation). By comparison, the new Duo Auto Dehiss reflects the advancing
technology of the last five years very clearly, both in terms of its looks and its
specifications.

The first impression when unpacking the unit is that the steel case has a very
similar design to that of its forebear, with three horizontal engraved lines running
across the full width of the matt-black front panel and highlighted with a subtle
gloss-black CEDAR logo. The case extends a modest 200mm behind the rack
ears according to my ruler (although the handbook claims 240mm), but the new
model is almost twice as heavy as the older DHX, tipping the scales at 4kg.

As on the DHX, the rear panel carries only a few connectors: a pair of XLRs for
stereo AES digits in and out, a pair of phono sockets for S/PDIF in and out, and a
trio of MIDI sockets (In, Out, and Thru) for remote control. One new addition is a
USB socket (apparently for factory use only), and the IEC mains inlet feeds an
internal universal mains power supply accepting voltages from 85V to 260V at
50Hz or 60Hz.

Whereas the DHX could only operate at sample rates of 44.1kHz or 48kHz, the
Duo accepts any sample rate between 32kHz and 96kHz. The I/O supports 24-bit
word lengths (although the internal resolution is 32-bit), and the available DSP
power is an impressive eight times greater than on the DHX — a pair of Sharc
DSPs providing 400 megaflops of signal-processing power.

The front panel also reflects half a decade of technological advances. For a start,
the old-fashioned green power LED of the Series-X models has been upgraded
to the new millennium's omnipresent blue indicator. In fact, there are two blue
LEDs at the right-hand end of the front panel: one marked Power, and the other
Standby. As on the DHX, there is no mains isolation switch — the unit is
permanently powered as long as it is plugged in — but a front-panel rocker
switch enables a power-saving standby mode. Another sign of the times is that,
while the operating power consumption of the Duo is the same as that of the
DHX at 15W, the standby mode curtails the power consumption to just 1W
instead of the 10W of the older DHX.

Possibly a slightly less welcome change for the Duo is the replacement of the
DHX's three simple physical controls with a blue (what else?) LCD menu screen,
six soft keys, and a rotary encoder. The screen is very clear and the menu
operation simple and intuitive, thanks in part to the bright/dim illumination of the
backlit blue buttons to indicate which functions are available and which are not.

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CEDAR Audio Duo Auto Dehiss

Setting Up

After you switch the unit on, the LCD screen shows the CEDAR logo and the
product serial number for a couple of seconds while the software boots up. After
this, the main operating menu is displayed, with the three principal user
parameters on the left-hand side, and mode and menu options on the right. The
middle right soft key accesses the Menu mode, providing six new options
labelled Setup, Audio I/O, Process Mode, Memory, Close, and Status — and I
suspect the functions of most of these will be self-evident. The Setup menu
allows the screen contrast and MIDI channel number to be changed, as well as
displaying the software version and hardware serial numbers. There is also a
facility here to clear the user memories and restore the factory defaults.

The Audio I/O menu enables the output word length to be selected (16, 18, 20, or
24 bits, with TPDF dithering) and the output gain to be adjusted over ±10dB.
There is also a numerical signal-level display with a peak-hold function.
Strangely, there is no means of selecting a specific input — the unit apparently
selects whichever has a valid signal. So what happens if there is a valid signal on
both? The handbook recommends against this to avoid an incorrect automatic
selection being made! Both outputs are active at all times, and any status bits
present on the active input are mirrored to the outputs. The Status menu
indicates the input connection and clock lock status, along with the measured
input sample rate and the current state of the two DSP chips (indicated as either
OK or Error, the latter with numeric fault codes).

The Memory menu provides access to the 99 user memories, with the usual
Store, Recall, Rename, and Delete functions, while the final Process Mode menu
allows the unit to be configured for manual or automatic operation, and to
process stereo signals in either L-R (left-right) or M-S (Middle & Sides) formats.
This menu is, however, mostly redundant as these configuration modes can be
changed directly from the main operating screen.

Action Stations

In the default automatic mode, the screen shows the settings of three adjustable
parameters on the left, each accessed by pressing the adjacent soft key and then
using the wheel encoder to change the value. I'll return to these user parameters
in a moment. The right-hand side of the screen shows the processing mode at
the top (L-R or M-S) with underlines to indicate which channel is being
processed. Repeated pressing of the adjacent soft key cycles between left, right,

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CEDAR Audio Duo Auto Dehiss

and both (or Middle, Sides, and both), and holding the button down for three
seconds changes the mode between L-R and M-S.

The middle button always provides

access to the configuration menus, as
already described, while the bottom
button provides a bypass function to
switch the processing in and out to
check its subjective effect. Holding this
button down for three seconds will
change the operating mode from automatic to manual, and vice versa.

In automatic mode, the software itself identifies the hiss element of the signal and
optimises the various program parameters accordingly. The screen displays
three adjustable parameters, labelled Bias, LF Bias, and Atten. Normally, the last
is the only control that requires setting, as this determines the amount by which
the hiss is attenuated. This is a subjective decision and will vary with the material
and the desired effect.

The Bias and LF Bias controls allow the user to offset the program's automatic
hiss detection. Setting a positive Bias effectively instructs the program that there
is more hiss present than it has detected, and the output will therefore be
processed more heavily. A lower Bias setting does the reverse, which helps to
retain more ambience in the signal, although with a greater risk of noise artefacts
slipping through. The LF Bias control does exactly the same, but only influences
the system for frequencies below 5kHz — low and mid-range frequencies. This
allows the hiss reduction to be tailored to suit the spectral character of the
medium. An analogue tape, for example, where the hiss is most audible across
the higher frequency range, might benefit from less low and mid-range hiss
reduction, and the LF Bias control could be reduced accordingly.

Adjusting either of the Bias controls normally requires the Atten setting to be
tweaked as well, and there is an iterative process of optimising the bias settings
and fine-tuning the attenuation to get the best results. However, the effect of
each control is quite audible with a little experience, and in practice setting things
up is reasonably quick.

In manual mode, the menu screen looks much the same. The three right-hand
buttons perform the same functions as before: channel selection, menu access,
and bypass (with the same 'long press' options working as before). The three left-
hand parameters initially look the same too. The Atten and LF Bias controls are
exactly as before. However, in place of the Bias control the manual mode
provides a Level control. This is used to instruct the software of the absolute level
of hiss within the signal, and operates across the entire audio spectrum. In effect,
this is the parameter which is set automatically in automatic mode.

Clearly, this parameter is critical to the effective operation of the dehiss process,
although finding the optimum position is actually quite intuitive. If set too high,

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CEDAR Audio Duo Auto Dehiss

low-level ambience is treated as noise

and removed, resulting in an unnatural
and muffled sound quality. If set too
low, the device will not remove noise
effectively at all, and you may start
getting noise artefacts. Again, an
interactive process is required to fine-
CEDAR's previous generation of 1U
tune the Level, LF Bias, and Atten
rackmount audio restoration tools were the
parameters, but finding the optimal Series X, reviewed back in SOS July 2000.
settings doesn't take long. The march of progress has been swift,
though, and the Duo processors now provide
eight times the processing horsepower of the
The facility to process the signal in M-S Series X units, as well as a more flexible
mode is very useful. In many cases the LCD-based control interface.
strong central component of stereo
material may mask hiss quite
effectively, while at the edges of the image the hiss may be more exposed and
obvious. Processing in M-S mode allows the hiss reduction to be tailored
accordingly across the stereo image. Likewise, in L-R mode if there is more
noticeable hiss on one channel than the other, or if the two channels exhibit
different 'colours' of noise, the program can be optimised accordingly.

Duo Auto Declickle

The other unit in CEDAR's Duo
range is the Auto Declickle, which
gets its slightly odd name from the
fact that it combines declicking and
decrackling processes into a single
1U rack. Apparently, while
developing the flagship Cambridge
system, CEDAR's researchers found
that combining the two processes
gave better results. What makes Auto Declickle particularly clever is that it can
detect the remains of the original audio signal behind the click or crackle, so it can
repair the audio waveform much more sensitively than would a traditional
interpolation process. Stay tuned for a review in SOS soon...

Performance

The Duo Auto Dehiss is certainly an impressive tool, and the algorithmic
improvements over the previous generation are obvious, both when setting it up
and when listening to the processed output. This new version seems more
assured of what to class as noise within a signal, and the distinct 'twittering' and
'glugging' artefacts incurred while adjusting the Level parameter of the Series X
machine have been replaced with a more intuitive noise-pumping effect.

No dehiss program will ever be able to remove every trace of noise perfectly

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CEDAR Audio Duo Auto Dehiss

while leaving low-level ambience and reverb tails completely intact, but the Duo
Auto Dehiss comes very close to that ideal in a fuss-free way. I used the machine
to clean up a selection of classic jazz recordings on quarter-inch tape, some
dating back nearly forty years. The results were very good indeed, and optimising
the settings was always intuitive — although it certainly pays to fine-tune the
Bias, LF Bias, and Level controls in a reiterative loop a couple of times, because
adjusting one seems to have knock-on effects for the others. The more care and
attention paid during setting up, the better the end results, as you might expect.

While a lot of the material was mono, I found the M-S mode very useful with
some of the stereo tapes, and the ability to bypass the hiss removal processing
made it easy to assess the effect on reverb and ambience, as well as making it
easier to recognise any low-level artefacts. I suspect longer familiarity with the
product would result in very effective 'ear-training' which would allow the optimal
settings to be found even more quickly.

Like all CEDAR products, the new Duo Auto Dehiss is relatively costly — this
kind of R&D is expensive to fund — but it provides a real-time solution to the
problem of unwanted hiss. The automatic mode seems to optimise the settings
very well, and there are no audible artefacts from the processing if you use it
sympathetically. Overall, a very worthwhile improvement over its predecessor.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Edirol UA101

In this article:
In The Blue Corner
Edirol UA101
Edirol UA101 Brief USB 2 Audio Interface (PC)
Specifications Published in SOS May 2005
Installation Print article : Close window
Control Panel
Reviews : Computer Recording System
Testing Testing
Final Thoughts

Edirol UA101 £399

pros Edirol have pioneered USB 2.0 as a format for
Good audio quality in a connecting audio interfaces, and their latest unit
compact and rugged package.
Versatile monitor mixing
offers 10 inputs and outputs at a price that compares
options. well with Firewire alternatives.
Stereo and multi-channel
WDM drivers provided.
Built-in analogue limiter. Martin Walker
cons
No GSIF driver support. In 2003, Edirol launched the world's
Rear-panel DIP switches for first USB 2.0 audio interface, the
phantom power and +4/10 UA1000 (reviewed in SOS November
input switching are fiddly to 2003: www.soundonsound.com/sos/
operate.
nov03/articles/edirolua1000.htm).
summary Since then, however, few USB 2-
The UA101 finally provides a specific music products have been
cheaper alternative to the
UA1000 for anyone who
forthcoming — largely, I suspect, Photos: Mike Cameron
would prefer their audio because USB 2 products can't as yet
interface to plug into a USB be used by Mac owners. Firewire audio interfaces, on the other hand, can be
2.0 rather than a Firewire used by both PC and Mac owners, so when Edirol subsequently launched their
port, and its analogue limiter Firewire-based FA101, many people assumed that the UA1000 was a one-off
may sway those who don't
mind either way.
experiment.
information
The UA101 proves them wrong. As its name suggests, it's similar in many ways
£399 including VAT.
to the FA101 (reviewed in SOS September 2004: www.soundonsound.com/sos/
Edirol Europe +44 (0)20
8747 5949. sep04/articles/edirolfa101.htm), except that it connects via USB 2.0 rather than
+44 (0)20 8747 5948. Firewire. It's presented in the same half-rack format, and at a similar £399 price.
Click here to email Both devices offer 10 inputs and 10 outputs: of the eight analogue inputs, six
www.edirol.co.uk operate at line level only, whilst two also boast mic preamps, and one of these
can serve as a high-impedance instrument input. There are also eight analogue
Test Spec output channels with a versatile analogue monitoring system, plus S/PDIF optical
in/out and MIDI In and Out. Like the FA101, the UA101 supports audio formats
Edirol UA101 Windows XP
up to 24-bit/192kHz, albeit with a reduced number of channels at the highest
driver version 1.0.
sample rates.
Intel Pentium 4C 2.8GHz

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Edirol UA101

processor with You'd be forgiven for assuming that the FA101 and UA101 were otherwise
Hyperthreading, Asus P4P800
Deluxe motherboard with Intel
identical, barring a different colour scheme and the replacement of the pair of
865PE chip set running rear-panel Firewire sockets by a single USB 2.0 port. However, closer inspection
800MHz front side buss, 1GB shows that there are plenty of other differences between the two, the most
DDR400 RAM, running intriguing being a front-panel button labelled Limiter.
Windows XP with Service
Pack 2.
Tested with Cakewalk Sonar
4.0, NI Pro 53, Rightmark
Audio Analyser 5.4, Steinberg
In The Blue Corner
Cubase SX 3.0 and Wavelab
5.00a, Tascam Gigastudio 160 The two front-panel Neutrik inputs of the UA101 are fed to a pair of 'professional
v3.04. grade' mic preamps identical to those of the FA101, with up to 40dB gain
controlled via rotary sensitivity knobs, and optional +48 Volt phantom power. As
with the FA101, the button to switch this is on the rear panel — fine for mobile
and desktop use, but not very handy if you've bolted the unit into a rack. There's
the same optional high-impedance button for input 2 allowing you to use the
inner TRS jack socket to plug in an electric guitar.

There are also six line-level inputs on

the rear panel with balanced/
unbalanced connections on quarter-
inch jack sockets, but whereas the
FA101 offered adjustable sensitivity
between -10dBV and +4dBu only on Each pair of the UA101's inputs can be
inputs 7/8, here, each pair of line switched between -10 and +4 sensitivity
settings using a DIP switch on the rear panel.
inputs can be switched between -
10dBV and +4dBu operation via tiny
rear-panel DIP switches. This is not a job you'd want to do very often in the dark,
but is certainly a more versatile approach in the absence of software-switched
options. A further DIP switch lets you switch between USB 1.1 and USB 2.0
formats, so you can connect the UA101 to almost any USB-equipped computer,
although of course you won't be able to use the full quotient of I/O over USB 1.

The rest of the rear panel is straightforward, with 10 TRS-wired output sockets
instead of the FA101's eight (the extra two are dedicated stereo Monitor Outs,
hard-wired to the monitor mixer outputs along with the headphone output), plus
MIDI input and output sockets, a USB connector, and the input for the supplied
DC power supply.

The left-hand end of the front panel is completed by the aforementioned Limiter
switch, which places a limiter with a -4dBFS threshold in the signal path on inputs
1 and 2, plus the Toslink optical input and output sockets. The five-LED meter is
another improvement on the four LEDs of the FA101, and can display the peak
levels of any input or output. The five LEDs come on at -42, -30, -18, -12 and -6
dBFS, so they provide a useful graph from signal present through to imminent
clipping.

The black area at the right-hand end of the panel is devoted to digital and
monitoring options, looking almost identical to the same area of the FA101, and

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Edirol UA101

comprises sample-rate selection, Digital In and associated Sync LED for internal/
external clock selection, two buttons and one rotary control for monitoring, which
I'll come to in a minute, the rotary level control for the dedicated monitor and
phones outputs, the phones socket itself, and power and 'USB active' indicators.

Edirol UA101 Brief Specifications

Sample rates: 44.1, 48, 88.2, 96 and 192 kHz from internal clock.
Analogue inputs 1/2: Neutrik with outer balanced mic socket, optional global +48V
phantom power and analogue limiting, inner TRS quarter-inch jack, high-impedance guitar
input option on input 2, plus preamp with 40dB gain range.
Analogue inputs 3 to 8: +4/-10 switched line-level sensitivity on quarter-inch TRS
balanced/unbalanced jack sockets.
Analogue outputs: 10 balanced/unbalanced TRS quarter-inch jack at fixed +4dBu level
(two of which carry the dedicated stereo Monitor Mixer outputs), plus stereo headphone
output with dedicated level control.
Digital I/O: S/PDIF in and out on Toslink optical, two MIDI Ins and Outs.
Dynamic range: not stated.
Frequency response: 20Hz to 40kHz, +0/-2dB at 96kHz.

Installation

I received the final version of the hardware and Windows XP drivers, but a
preliminary Control Panel utility and no printed or PDF manual, so this was an
ideal opportunity to see just how easy the UA101 was to use. The driver CD-
ROM contained a Setup.exe file, so in keeping with most other USB and Firewire
audio interfaces, I ran this before I plugged in the interface, and was pleased to
find it provided on-screen instructions on the installation procedure, including
when to plug in and power up the UA101 — not everyone reads the manual after
all, even when it's supplied!

The UA101 can't be powered from the USB buss, so you need to plug in the
bundled 9 Volt DC line-lump PSU and switch on the UA101 via the switch
associated with the Phones/Monitor output rotary level control before it's
detected by Windows. For those interested in mobile recording who have both
USB 2.0 and Firewire ports available, this gives the FA101 an advantage over
the UA101, since it can be powered from the Firewire buss if your computer is
capable of doing so.

Like its USB 2.0 stablemate the UA1000, the UA101's drivers only currently
support Windows XP, but this is after all what most PC musicians are now using,
and I was up and running within a couple of minutes. I was pleased to see that
Edirol are bucking the trend by offering multiple stereo WDM drivers in addition to
a single multi-channel one, which is very handy for those whose applications only
support stereo pairs. The UA101 offers five stereo pairs of driver playback

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Edirol UA101

options (1/2, 3/4, 5/6, 7/8 and 9/10 digital) in addition to the multi-channnel one,
while the recording options add to this list the Monitor output from the UA101's
40-bit internal mixer. However, I was disappointed that Edirol still seem to be
ignoring Gigastudio owners, and there's no GSIF driver support at all.

Control Panel

I experienced a strong dose of déjà vu when I located the Edirol UA101 software
utility among my Windows Control Panel options and launched it: it was identical
to the one I last saw when reviewing the UA1000, even down to still having
'Edirol UA1000 Control Panel' displayed in its title bar. The UA101 will ship with a
control panel that looks rather more elegant, but its patchbay and monitor mixer
functions remain fairly similar, so let's remind ourselves briefly of what's on offer.

In essence, you get a patchbay that

lets you decide which signals get sent
to the five pairs of WAV inputs that
appear as recording options in your
audio application, chosen from the five
pairs of physical input signals, the five
pairs of WAV playback signals coming
from your audio application, or the
combined output of the control panel's
monitor mixer. The five pairs of
A helpful routing diagram is printed on the
physical outputs have similar options: UA101's top panel.
the five pairs of WAV playback signals,
the five input pairs, or the monitor
mixer output. In addition, a second set of output routings is available, and can be
selected using the front-panel Soft Ctrl button, so you can set up two different
monitoring presets.

The main control panel display is devoted to the monitor mixer, with on/off, solo,
pan, fader and link controls for each of the Wave Outs and physical inputs, plus a
signal 'blinky' for each of these channels and a pair of monitor master level
controls. Across the bottom there's a status display of various parameters
including current sample rate, internal/external clock, and the direct-monitor
Mono and Soft Ctrl buttons. The hardware front panel has the same unusual Mix
control as the FA101, with a centre detent marking the default mix, and clockwise
movements favouring the Input monitor mix, and anti-clockwise moves the
Output monitor mix. There's also a front-panel Mono button that affects the input
monitor mix. Overall, the monitoring facilities are comprehensive once you get
your head round them.

Testing Testing

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Edirol UA101

My double-blind listening tests against Emu's 1820M and Echo's Mia proved
interesting. Once again I picked out the Emu for its superior imaging and ability to
present greater front-to-back positioning during reverb tails, but I couldn't decide
which I preferred of the other two. The Mia was warmer and 'cosier', whereas the
UA101 gave a slightly more intimate sound, possibly because as I found later it
exhibits a slightly boosted playback response above 6kHz, rising to a tiny peak of
+0.3dB at 20kHz with a 44.1kHz sample rate. However, these are very subtle
differences, and given its similarities with the FA101, I'd agree with the
judgements Mike Bryant made in his review of that unit: it has a clean, open
sound and good stereo imaging.

While recording, the limiter option on inputs 1/2 proved extremely useful in
avoiding clipping, and let me achieve significantly higher input levels in a
transparent way without worrying about compromising audio quality — only by
applying input levels high enough to clip the analogue stages preceding the
limiters did I start to experience any distortion.

Rightmark's Audio Analyser proved

that the UA101 must have almost
identical analogue circuitry to the
FA101, returning exactly the same
dynamic range of 104dBA at 24-
bit/44.1kHz. Unusually, this stayed
almost the same at both 96kHz and
192kHz, even though its frequency
response was not capped at the higher
rates (the most common reason for The review unit was supplied with a
this). The -0.3dB points were at an preliminary Control Panel utility: the finished
version will look like this.
impressive 8Hz and 21kHz with a
44.1kHz sample rate, extending to
about 44kHz with a sample rate of 96kHz, and 57kHz at 192kHz.

Like both the UA1000 and FA101, the UA101's sample rate can only be changed
from the front-panel switch, and the unit must be rebooted before this takes
effect. However, whereas the on/off switch for the FA101 is on the rear panel,
making it inaccessible if the unit is bolted into a rack, both UA units have the
practical advantage of front-panel on/off switches. The control panel utility
provides a handy readout of the unit's current sample rate, but it's up to users to
make sure that this is the same as the one set in their audio application — I
received no error messages when attempting to play back or record files through
the UA101 at one rate when it was set to another, but my RMAA results
displayed peculiar frequency responses when I did so.

As with the FA101, switching to 24-bit/192kHz reduces the I/O count from 10
channels to six, which is sufficient to play back DVD-Audio discs in 5.1 surround,
while anyone who needs to plug into a low-bandwidth USB 1.1 port instead of a
Hi-Speed USB 2.0 one will find themselves reduced to using analogue inputs and
outputs 1/2 at 44.1 or 48 kHz only.

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Edirol UA101

The slider to adjust the buffer size of the ASIO drivers is simply labelled from Min
to Max, rather than stating number of samples or latency in milliseconds, but the
default setting equates to an unusual value of 432 samples, resulting in 10ms
latency at 44.1kHz. I experienced no glitching with Cubase SX at this setting, and
was eventually able to drop the value down to the lowest setting on my system
with no problems, giving a 3.3ms latency. The WDM drivers managed an even
better 2.0ms with Sonar 4, and the Direct Sound and MME drivers proved to be
as good as any I've tried under Windows XP, achieving 30ms and 45ms Play
Ahead settings respectively with NI's Pro 53 soft synth.

Final Thoughts

Edirol have placed the UA101 into an already congested part of the market, and
at a price of around £400 you can also buy various Firewire audio interfaces such
as M Audio's Firewire 1814 and Guillemot's Hercules 1612FW. The Firewire
1814 provides the most potential I/O with its eight-in/four-out analogue plus eight-
channel ADAT support, and the highest dynamic range at a measured 109dBA. It
can be buss-powered, but its analogue inputs aren't balanced, and its outputs
don't offer the higher +4dBu levels compatible with more professional gear.

For those with more demanding recording requirements, the Hercules 1612FW is
another strong contender with its 12-in/eight-out analogue plus co-axial and
optical S/PDIF, word clock, and two MIDI Ins and Outs. It has very similar audio
performance to the FA/UA101 boxes, but it doesn't support 192kHz at all. Both
the 1814 and 1612FW also provide GSIF drivers, though, unlike the Edirol range.

However, the UA101's closest competitor is probably its own stablemate, the
FA101. If you don't have a Firewire port then you can buy one on a card fairly
cheaply, but nearly all modern PCs are already equipped with USB 2.0 ports, so
which is the best option? Well, a USB 2 interface might have the advantage
where high track counts are required from a Firewire hard drive, since it won't be
sharing the same bandwidth. On the other hand, some armchair experts say
USB 2 is unsuitable for professional audio, but I've now reviewed both the
UA1000 and the UA101, the only available multi-channel USB 2 interfaces, and
have had no practical problems with either. Since its launch, moreover, the
UA1000 has gone on to win an enviable reputation for its reliability.

The strengths of both the UA101 and FA101 are their robust and compact half-
rack cases, which make them the smallest of all these units, while their almost
identical eight-in/eight-out analogue audio quality is on a par with the Hercules
and only slightly behind the Firewire 1814. In its favour, the FA101 can be buss-
powered, which is handy for mobile recording sessions, but the UA101 provides
an additional stereo monitoring output, more flexible -10/+4 input options, and
slightly better metering. Its most significant advantage, however, is the stereo
limiter — the only other interface I can recall with this feature was the ill-fated
Lexicon Core 2, but it remains a powerful incentive for anyone wanting to record
anything cleanly when faced with unpredictable levels. For small live sessions,

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Edirol UA101

this is a big advantage, and should push the UA101 a long way up your shortlist.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Edirol%20UA101.htm (7 of 7)9/27/2005 9:22:20 PM

 Fervent Software Studio To Go

In this article:
On The Go
Fervent Software Studio To Go
System Requirements Bootable Linux Software Suite for PC
Trying It Out Published in SOS May 2005
ALSA Dancing Print article : Close window

Fervent Software Studio Reviews : Software

To Go £50
pros
Easy introduction to Linux
audio and music tools.
If you're attracted by the idea of Linux and open-
Hardly any configuration or
Linux skills required. source music software, but put off by the thought of
Cost-effective compared to installing it on your PC, there's another way: a
time spent getting a DIY Linux bootable CD-ROM containing both the OS and all the
music system working.
software you need, ready to go.
Official technical support,
rather than community
support.
Daniel James
cons
Only as good as the
hardware of the PC you run it It's certainly possible to produce and
on. record music on a Linux system, but
The suitability of any given setting up the machine could be a
PC for real-time audio can be challenge for many people without
unpredictable. experience of the platform. With nearly
Loading from CD-ROM can all of the source code available on
be slow on older machines.
various Internet sites, the DIY route is
No support for some audio
feasible if you have the time to figure
hardware, including all MOTU
and Firewire interfaces. out how the pieces fit together. While
that's undoubtedly the best way to
summary
learn about the technology, it could
A lot of software for under
£50, and far more complete
present a frustrating learning curve for
than any proprietary bundle. If those musicians who just want to get On the Studio To Go desktop, Sweep plays a
you're interested in using going. stereo track into the Jamin mastering tool via
Linux for music but fear the JACK.
amount of new skills required,
this could be the product forOne way of hiding the unfamiliarity of
you. novel software is to put it inside devices that resemble traditional equipment —
information the Hartmann Neuron resynthesizing keyboard and the Muse Receptor
rackmount VSTi host are both in this class, with Linux systems running inside.
£49.99 including VAT.
Another approach is to do what Apple have done with OS X: select a particular
Fervent Software +44 (0)
7905 932630.
combination of open-source software, add some user-friendly components on top
Click here to emailClick and market the result as a single product, along with technical support services.
here to email
www.ferventsoftware.
com

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Fervent Software Studio To Go

On The Go
Test Spec
Custom audio PC with AMD Fervent Software have done something similar to Apple with Studio To Go, which
Opteron 240 and 1GB RAM,
takes the form of a single, bootable CD-ROM. Linux is available in native
with M Audio Audiophile PCI
card. versions for just about all processors, including the IBM chips found in the Apple
G5 machines, but since most people have PCs, Studio To Go was compiled for
Dell Optiplex GX270, P4
the Intel 32-bit architecture, which includes the Pentiums and the Athlons.
2.6GHz, 512MB RAM, Intel
onboard sound chip set.
Acer Aspire, Athlon XP 2000 Because Linux-based systems are licensed in a very different way to proprietary
+, 256MB RAM, VIA onboard software products, it's quite legitimate to use the same software on any number
sound chip set. of machines you have access to. The principle behind these 'live' Linux
distributions aimed at musicians — and there are now at least half a dozen
available — is that you can have all your music-making tools in one portable
package, without needing to carry a laptop around. These distributions load the
Linux software into the available RAM and won't touch any software or data
already present on the hard disk, unless you want them to. A USB memory stick
is a popular complement to live distributions, as it allows you to take personal
data and settings around with you.

The bootable format has both advantages and disadvantages. On the plus side,
the software isn't tied down to any particular machine and there's little risk of
damaging the system other than by breaking the CD-ROM itself. On the other
hand, loading software from CD-ROM is slower and noisier than loading it from
hard disk, and will become tedious if the PC isn't a recent one with plenty of
RAM. Most live distributions offer the possibility of a permanent install to hard
disk if you're happy to wipe the system or data that's on there already.

Fervent Software are the first company to produce a live Linux distribution for
musicians in a retail package with a technical help service: buyers of the boxed
set can register for 30 days' support by email and in customer forums. The
company intend to market this product to both home-studio users and the
education sector, the idea being that students will be able to have access to a full
suite of music tools which they can take home and use without the fear of a raid
from the software police. To this end, Fervent Software intend to offer volume
pricing deals to educational institutions.

System Requirements
A PC compatible with 32-bit Intel software.
800MHz Pentium minimum; 1.2GHz or faster recommended.
256MB minimum RAM; 512MB or more recommended.
Any sound/MIDI chip set with ALSA driver available.
CD-ROM drive, the faster the better.
2GB or more hard disk space for optional hard disk install.

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Fervent Software Studio To Go

Trying It Out

I tried Studio To Go with a 64MB memory stick on generic Dell and Acer PCs,
and my own Opteron-based workstation, which was built specifically to run Linux
audio software. First, you have to make sure the PC is booting from CD rather
than hard disk. Once that's taken care of, there's a whir from the CD-ROM drive
and a boot prompt appears. At this stage you can press the Enter key for default
settings, or choose from several boot options if automatic configuration has failed
on a previous attempt. Assuming everything works on your particular PC, you
then see a loading screen followed by the KDE desktop, which should feel
familiar to anyone coming from a Windows or Mac background.

The Dell machine threw up an immediate problem: the screen resolution was a
pitiful 640 x 480 instead of the 1280 x 1024 I would have expected. The auto-
configuration of the display had not worked properly, and a Google search
demonstrated that this was a known problem on this specific model — the
amount of video memory allocated by the BIOS was only 1MB. Linux can
sometimes highlight cases where hardware is buggy, and PC manufacturers
bodge a machine to get it working with Windows XP and out of the door as
quickly as possible. A flash update with a newer BIOS, available on demand from
Dell technical support, is the proper solution in this case.

Other than the resolution problem, the Dell machine worked smoothly, with
program launching reasonably responsive and the JACK sound server
distributing audio between applications without buffer glitches, or xruns as JACK
calls them. The same couldn't be said of the bargain-basement Acer PC, which
required an increase in the number of periods per buffer to process audio
properly, and suffered from serious delays when launching applications. This isn't
really the fault of Studio To Go, since there are a lot of PCs out there which just
aren't up to the task of working with real-time audio, and JACK, by design, is
extremely unforgiving of inadequate machines. The lack of performance on the
Acer PC could have been due to low-quality hardware, or perhaps daft interrupt
assignment. Disabling ACPI in the BIOS was suggested by Fervent Software as
a possible fix; poor implementations of this standard also cause problems with
Windows XP on certain PCs.

ALSA Dancing

Drivers from the ALSA project are included in a single package, so Studio To Go
either will support your sound hardware or it won't — since a live distribution is
run from a read-only disc, you can't download drivers to it. To sum up the current
status of ALSA driver support: PCI cards and motherboard chip sets tend to
work, some PCMCIA interfaces have drivers, USB devices work if they are
standards-compliant, and Firewire devices can be assumed not to work. This is
not due to a lack of Firewire support under Linux, but because audio interface
manufacturers hardly ever stick to the proper Firewire specification, and they

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Fervent Software Studio To Go

won't release the details of their non-standard protocols.

This is a particular problem with Mark of the Unicorn

equipment, and as a result hardly any MOTU interfaces Studio To Go
provides
are supported under Linux. It's something to consider
access to
when choosing an interface, as a closed design means disk partitions
you will always be dependent on the manufacturer for on the host
continued driver support, regardless of whether you use system, as
Linux, Windows or OS X. The converse of this problem is well as a
memory stick
that some sound hardware is supported by ALSA even
if you have
though it has been discontinued and does not have one.
Windows XP drivers available — an example would be
the Midiman DMAN PCI card.

On my Opteron machine, which has a Matrox dual-head

video card, the display was configured perfectly on the
first boot without any tweaking required. The M Audio
Audiophile PCI card was correctly detected and configured, with JACK starting
automatically at a conservative buffer size. A row of icons on the left side of the
desktop provided access to disks on the host machine and the USB memory
stick, while folders contained tutorials, example files and sample VSTi plug-ins in
PC format. Although VST Instruments and effects are rarely open-source or
written specifically for Linux, they can be run on Linux systems as long as the
binary format matches that of the host CPU.

Fervent Software also support the Rosegarden project, so it's no suprise that this
MIDI + Audio sequencer, with classical notation support, is at the core of Studio
To Go. Audio recording and editing programs include Ardour, Audacity, Sweep
and Rezound, providing a choice of both destructive and non-destructive, disk-
based and RAM-based editing. A collection of soft synths, including the
Hydrogen drum machine and native DSSI plug-in instruments, means that there
is plenty of software to make music with, and VSTis can also be loaded from the
host machine's hard disk. Utilities such as software mixers for sound interfaces
are included, and there is also a collection of general-purpose tools including
Konqueror, KDE's combined file manager and web browser.

Ultimately, the portability advantages of Studio To Go will be constrained by the

quality and Linux-compatibility of the PC hardware that you have access to. The
price-to-storage ratio of USB memory sticks tends to make the combination of
Studio To Go and one of these more suitable for MIDI work than lengthy
multitrack audio sessions, and I suspect most people who want to use the
software seriously will take the step of installing it permanently to a hard disk.
However, running Studio To Go from the CD-ROM is a quick, easy and
reasonably priced way of taking a look at Linux music software, or testing the
suitability of your PC for running it.

Published in SOS May 2005

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Fervent Software Studio To Go

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Fervent%20Software%20Studio%20To%20Go.htm (5 of 5)9/27/2005 9:22:25 PM

 JBL LSR6328 & LSR6312

In this article:
LSR6328 Active Monitor
JBL LSR6328 & LSR6312
JBL Transducer Active Monitors & Subwoofer
Technologies Published in SOS May 2005
LSR6312 Powered Print article : Close window
Subwoofer
Reviews : Monitors
Room Mode Correction
Listening Tests
The Final Word

JBL LSR6328 &

JBL's new monitors incorporate Room Mode
LSR6312 £2680/£1545
Correction technology which claims to be able to
pros
reduce the bass problems caused by standing waves
Essentially neutral and
detailed character. at the listening position. But does it really work in
Powerful and dynamic. practice?
Solid construction and ease
of portability.
Matching optional Hugh Robjohns
subwoofer with bass
management.
Under review here are two members of
cons JBL's new LSR6300 series of monitors:
Room Mode Correction the LSR6328, a two-way midfield
facility is limited in what it can system, and the LSR6312 subwoofer.
achieve.
Completing the range are the compact
summary two-way nearfield LSR6325, and the
A powerful but competent three-way LSR6332, either of which
active monitor capable of can also be used in concert with the
resolving a lot of detail over a
wide bandwidth. It boasts a aforementioned subwoofer. The
wide and stable listening LSR6328 is of a size I associate with
area, with good imaging and midfield monitors, but JBL suggest a
excellent dynamics. The working range of one to three metres,
matching subwoofer extends the short end of this range conforming
the bandwidth downwards
and the headroom upwards, with nearfield conditions. Nevertheless,
while also providing simple I'd recommend placing these monitors Photos: Mark Ewing
bass-management facilities. on tall stands behind the desk rather
The Room Mode Correction than on the meterbridge.
feature works as advertised
and will be useful to many,
but cannot replace proper
acoustic treatment.
information LSR6328 Active Monitor
LSR6328 active
nearfield monitors, £2680 The LSR6328's two-way design has cabinet dimensions of 330 x 406 x 325mm
per pair; LSR6312 (hwd) and weighs 18kg. It is constructed from 19mm MDF, finished in a dark-
subwoofer, £1545 each. grey paint, and has convenient recessed lifting handles built into each side panel.
Prices include VAT.

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JBL LSR6328 & LSR6312

Harman Pro UK +44 (0) The integral amplifier's heat sink is on the rear panel, along with the connections,
1707 668222. setup controls and a rear-facing Linear Dynamics Aperture (LDA) port. (For more
+44 (0)1707 668010. information on LDA and other proprietary terms, take a look at the 'JBL
Click here to email Transducer Technologies' box elsewhere in this article.)
www.harmanprouk.com
www.jblpro.com The driver complement comprises an eight-inch bass unit with an unusually low 2
(omega) nominal impedance, and a 4(omega) one-inch titanium/composite
tweeter mounted in an Elliptical Oblate Spheroidal (EOS) waveguide. An LED
between the two drivers illuminates when the system is powered. The amplifier
chassis is a two-channel Class-AB design with a 120W monolithic (chip) amplifier
for the high-frequency driver and an all-discrete 250W circuit for the low-
frequency driver.

The rear control panel is surprisingly complex at first sight, but the handbook
provides clear setup instructions and a Room Mode Calibration Kit is supplied to
help configure the system (see 'Room Mode Correction' box for more details).
The audio input is hooked up via a combi jack/XLR connector with an associated
and recessed input-level trim control. Mains is provided via the usual IEC socket
with an adjacent power button. A set of eight DIP switches, three recessed rotary
switches, a bypass button, and remote control socket complete the configuration
facilities.

Three of the DIP switches configure the input, enabling the input level trimmer or
presetting the input level to +4dBu, +8dBu, or +12dBu sensitivity. The LSR6328's
input trimmer is aligned at the factory such that a -10dBV input provides
96dBSPL at one metre (in an anechoic environment). The fourth switch activates
low-frequency protection circuitry which replaces the default 24dB/octave high-
pass filter with a 36dB/octave filter. The next pair of switches adjust the high-
frequency level, introducing a subtle shelf to the response above 2kHz,
amounting to a very modest 1dB up or down. The final pair provide low-
frequency boundary compensation, with a shelf cut of -1.5dB, -3dB, or -4.5dB
below about 250Hz.

The specifications claim a nominal response of 50Hz-20kHz (+1/-1.5dB),

extending to 46Hz at -3dB and 36Hz at -10dB. For an SPL of 96dB at one metre,
distortion is claimed to be better than two percent below 120Hz and 0.6 percent
above. Maximum peak SPL is 111dB/1m and self noise is less than 10dBA.

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JBL LSR6328 & LSR6312

JBL Transducer Technologies

A number of innovative proprietary technologies have been used to improve the
performance of the LSR monitors. The first, Differential Drive, is an arrangement
of two separate voice coils on the bass driver, providing twice the surface area of
conventional speakers and allowing greater heat dissipation. This gives 3dB more
power handling with less power compression. The two coils are wound with
opposite phase to reduce their mutual inductance, which provides a more
consistent impedance relative to frequency and makes the driver an easier
amplifier load. Another advantage is that less iron is required in the magnetic path
than with most designs, making it lighter.
A third, shorted coil within the motor assembly does absolutely nothing most of
the time! However, during extreme cone excursions this central coil enters the
magnetic fields of the two driving coils and a current is induced within it. This sets
up an opposing magnetic force which acts against the motion of the cone, acting
as a brake. JBL call this Dynamic Braking and it is a clever protection device.
Apparently, the introduction of the third coil into the magnetic gap of the main coils
also helps to cancel the inherent distortion which builds as the main voice coils
reach the outer edges of their magnetic fields.
Reflex cabinet loudspeakers can suffer from turbulence within the port when
monitoring at high levels, simply because of the large volume of air moving
around. The LSRs use a Linear Dynamics Aperture (LDA) for their ports, which
smoothes the contour of the port exit to minimise this turbulence and thus improve
low-frequency accuracy at high listening levels.
Common to the tweeters of all the LSR range is the use of titanium and composite
materials, which are claimed to provide a better transient response with low
distortion, and thus reduced 'ear fatigue'. Furthermore, the tweeters a combined
with an Elliptical Oblate Spheroidal (EOS) waveguide, which helps to shape the
high-frequency dispersion, directing sound across a listening area of ±30 degrees
horizontally and ±15 degrees vertically, and maintaining an even frequency
response within 1.5dB across the entire zone.

LSR6312 Powered Subwoofer

The LSR6312 powered subwoofer measures 394 x 635 x 292mm (hwd) and
weighs 23kg. The cabinet features a front-facing port and contains a single 12-
inch driver powered by a 260W discrete Class-AB amplifier. The frequency
response is given as 28-80Hz (-6dB points), and maximum SPL is 112dB/1m.
The 6312 incorporates bass management for a three-channel satellite/subwoofer
system, as well as RMC facilities. Combi jack/XLR inputs and associated outputs
are provided for three channels (left, centre, and right) plus a fourth Sub Direct
input (with a 10dB gain-boost button). An eighth XLR socket provides a summed
bass output for additional subwoofers.

The three-channel input signals are summed together and then low-pass filtered
at 80Hz before being combined with the direct input (normally used for a
dedicated LFE signal. At the same time the input signals are individually high-
pass filtered at 80Hz and presented to the corresponding outputs to feed the

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JBL LSR6328 & LSR6312

satellite speakers. An LCR Bypass mode (remote-controlled with a footswitch)

disables the high-pass filtering to the outputs, providing a direct loop-through —
only the dedicated Sub Direct input feeds the subwoofer, so this essentially
bypasses the sub as far as the satellite monitors are concerned.

A recessed input trimmer and six DIP switches provide similar input-level options
as those on LSR6328 monitor, supplemented with a polarity inversion option, a
boundary compensation mode (a -4dB shelf below 50Hz), and an RMC bypass
mode. The RMC facilities are the same as before.

Room Mode Correction

The LSR-series Room Mode Correction (RMC) is a grand name for what is, at the
end of the day, a simple single-band parametric equaliser. This can be used to
tame a single low-frequency response peak, typically formed by the interaction of
standing waves within an untreated room. It is no substitute for properly designed
acoustic treatment with effective bass trapping, but it can help damp the effect of
a boomy resonance in a room. If your mixes often come out bass-light (because a
boomy room mode fools you into thinking the bass is more prominent than it really
is) then this could be a simple, if partial, solution.
The RMC controls incorporated into the LSR6328 monitors and the LSR6312
subwoofer all provide a single parametric equaliser, with adjustable frequency,
bandwidth, and attenuation. The control ranges are 26-96Hz, Q of 0.5-0.05, and
up to 14dB of cut. The difficult thing is to adjust each of these controls
appropriately, but that's where the supplied RMC Calibration Kit comes in. A
simple sound level meter and dedicated calibration CD, both provided, are used to
chart the low-frequency response of your room, and a series of charts in the
manual can then be used to reposition speakers as necessary and adjust their
rear-panel RMC controls. The system is simple and elegant, if a little time-
consuming — the whole setup process for all three speakers took me about 40
minutes.

Listening Tests

The LSR6328 is a powerful nearfield monitor with excellent dynamics and

transient handling. Not only can it play very loud indeed, it can do so effortlessly
and with 'rifle shot' attacks when called for! The mid-range is very clean and
neutral, and reproduced the critical vocal range with commendable clarity and
precision. I found it easy to listen into a complex mix, and heavy bass
instruments didn't mask the mid-range detail at sensible listening levels. Stereo
imaging was portrayed equally well, with a wide and stable sound stage over a
relatively large sweet spot.

With all the configuration controls in their default positions I found the tonal
balance to be a tad on the hard side, but in a more heavily damped control room
I suspect it would sound pretty neutral from the off. In my listening room I found
setting the high-frequency shelf to the -1dB position was all that was needed to

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JBL LSR6328 & LSR6312

tame the top end — something which I

find typical of most monitors tested
here. Recording flaws and imperfect
mixes are revealed clearly, but, while
many monitors with this ability quickly
become fatiguing, I found I was happy
to listen for extended periods without
any problems.

The bottom end of the monitor is very

well controlled and has a reasonable
extension for the size of cabinet. In a modest control room the LSR6328s do not
demand the use of a subwoofer on grounds of either extension or headroom —
although both obviously improve with the dedicated subwoofer in service.
Hooking up the LSR6312 subwoofer is as simple as re-plugging a couple of
XLRs. By default the input signals are high-pass filtered at 80Hz for the satellites,
and the subwoofer handles everything below 80Hz. The integration with the
satellites was excellent, and the greater extension was immediately obvious on
suitable material, with the bass becoming more powerful and solid. On bass-
heavy material the system really came alive, and if you are looking for a system
that can really move some air, this combination would be a good place to start
your auditioning.

The Final Word

As straightforward monitors, with or without the dedicated subwoofer, this new

LSR6300-series system performs impressively. When the music calls for it, these
monitors will sound smooth, delicate, and very revealing. However, they can also
deliver a really solid, weighty, and dynamic wall of sound when asked, with huge
sound pressure levels that can be sustained all day if your ears can take it.
Spoken voice and classical music are portrayed with neutrality and fidelity, and
clearly this speaker is a good all-rounder, equally capable regardless of the
source material's genre.

The trumpeted Room Mode Correction will be useful in some situations, but I
can't help feeling that messing up the speaker's low-frequency response in an
effort to correct a physical standing-wave issue is not the right way to go about
things! Having said that, the calibration tools are well thought out and easy to
use, and when applied carefully they can at least help to tame response peaks at
the listening position.

Published in SOS May 2005

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JBL LSR6328 & LSR6312

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/JBL%20LSR6328%20&%20LSR6312.htm (6 of 6)9/27/2005 9:22:30 PM

 Korg Kontrol 49

In this article:
The 49er
Korg Kontrol 49
USB & Kontrol 49 USB MIDI Controller Keyboard
Summing Up Published in SOS May 2005
Print article : Close window
Korg Kontrol 49 £319
pros Reviews : Keyboard
Full-sized keys, separate
pitch-bend and mod controls,
and the Microkontrol's vector
joystick.
Programming easy via the Korg's Microkontrol was a highly versatile, yet
software interface, with loads compact MIDI controller — but perhaps, with its three
of ready-to-use templates.
octaves of miniature keys, it was too compact. With
Powered via USB.
MIDI interface capability.
its four-octave, full-size keyboard, the Kontrol 49
looks set to put that right...
cons
Functionality over USB
limited to users of Mac OS
Nicholas Rowland
10.2 and higher, and
Windows XP.
Similarly, the editing These days we are blessed with a wide
software and presets cannot choice of brilliant, affordable music
be used with any other
software whose power and versatility
operating system on the host
computer. often puts far more creativity at our
fingertips than the hardware which
summary
inspired it, and so, not surprisingly,
A competitively priced, well
most of us are becoming more
thought-through package
that's also easy-to-use software-centric in our music-making.
(especially if you can run the Which is why we're seeing a growth
bundled editor/librarian industry in hardware control surfaces, Photos: Mike Cameron
software). and particularly those that are easily
information configurable to work with many different programs, often simultaneously.
£319 including VAT.
Korg UK Brochure Line Korg's new Kontrol 49 is intended as just such a 'universal' device and one
+44 (0)1908 857150. particularly suited to the computer-based musician whose wallet and studio
+44 (0)1908 857199. space are both relatively modest. Its grand title is 'USB/MIDI Studio Controller': in
Click here to email a nutshell, it's a controller keyboard equipped with a fair amount of knobularity
www.korg.co.uk plus a great deal of programmable MIDI intelligence enabling it to interface at the
deepest level with any music software (or hardware) you can shake a MIDI
parameter at.

If you think those sleek silvery looks are Korgishly familiar, then you're right. Look
into my eyes, look into my eyes... and cast your mind back to the Korg
Microkontrol reviewed by Paul White in SOS March 2004. Those with long
memories (or with browsers pointing to www.soundonsound.com/sos/mar04/

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Korg Kontrol 49

articles/korgmicrokontrol.htm) might recall the deal here: a neat velocity-sensitive

keyboard married to eight sliders, eight rotary controls and a master control, 16
velocity-sensitive drum machine-like trigger pads (also doubling as data-entry
buttons), plus a programmable vector-control joystick to handle such
performance features as modulation and pitch-bend. There was also an array of
colour-coded, eight-character LCD 'scribble strips' above the faders and rotary
controls to display parameter names and also give instant feedback on changes
to parameter values.

In terms of its flexibility and intuitive handling, the Microkontrol generally earned
itself a big thumbs-up from Paul. However, it was precisely in the big thumbs
department that it wouldn't score so highly, being equipped with mini keys, and
only 37 of them at that. So Korg have administered a portion of Alice's 'Eat Me'
cake, and created the more grown-up version you see here, sporting 49 or four
octaves of full-size keys (which incidentally, can be transposed up or down to
give you a range of C1 to C9).

The 49er

The keyboard itself is of the non-weighted variety, and is velocity sensitive with
eight programmable velocity curves to suit different styles. However, a true
player will bemoan the fact that there's no aftertouch. As on a lot of keyboards/
synths, pressure control has to be applied through either the programmable
modulation or pitch-bend wheels — features which were missing on the
Microkontrol. A Vector joystick is also present with separate control messages
assignable to the X (up/down) and Y (left/right) axes. Compared to the rest of the
package, which feels solid and well-built, this control does seem rather flimsy and
lightweight to the touch.

If my maths is correct (and counting

the keyboard as only one) the Kontrol
49 offers a total of around 40 different
controllers, each of which is
extensively programmable. The
shorthand for a complete set of
controller assignments is the word
Scene, and like the Microkontrol, the
Kontrol 49 can store up to 12 Scenes
on board with the trigger pads The editor/librarian enables you to speedily
providing the means to switch between assemble sets of Scenes and gives you
visual access to all the Kontrol 49's various
them. As it comes out of the box, it's control elements. Double-clicking on any
preloaded with a variety of sample item brings up a smaller menu (shown top
Scenes primarily designed to show off right) where you can quickly input values and
its muscles controlling the various name names. When you're finished, you can
demo packages which are included on upload the Scenes via USB to the keyboard.
Names of parameters appear on the LED
the installation CD-ROM. However, scribble strips to give you vital feedback on
these are just the tip of the iceberg: the which slider/rotary control is assigned to

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Korg Kontrol 49

CD-ROM has an extensive collection what, though the eight-character limit makes
of templates covering all mainstream some descriptions rather cryptic.
packages from the major software
houses, including Ableton Live, Propellerhead Reason, IK's Sampletank 2, most
NI instruments, and the major sequencers.

Korg also include a brilliant scene-management software package (shown

below), which in terms of functionality goes well beyond the original librarian
software that was included with the Microkontrol. This software offers complete
on-screen programming of the all Kontrol 49's assignment and functions,
enabling you to very easily modify the presets or create custom Scenes of your
own, assemble them into Scene sets and then upload them to the keyboard.

Using the editor I was quickly able to assemble the included templates into a
custom Scene set, giving me control over my Cubase mixer and EQ screens,
Steinberg's A1 software synth, Native Instruments' Battery, which I use as a VST
instrument within Cubase, and also a Korg Electribe.

Of course, you can also use the Kontrol 49's front panel to edit patches or create
your own from scratch, but to be perfectly honest, it really is a lot less hassle and
kinder on the tips of your fingers just to fire up the editor and do it all via the big
screen. Unfortunately, like the USB drivers, the editor software is only for users
of Windows XP and Mac OS 10.2 and above. What's more, 12 scene memories
is not that many, and if you need to control lots of different packages (or aspects
of packages) then you will probably find yourself swapping between different
Scene sets quite regularly.

USB & Kontrol 49

Like the Microkontrol, the Kontrol 49 is designed to interface with your computer
through a USB connection, which also provides the necessary power to the
keyboard. There's an included 9V wall-wart adaptor if you've got too many
devices on your USB connection for this to work.
USB drivers are provided for Windows XP and Mac OS X (10.2 or later) and
installation on to my shiny new iMac G5 was simple and straightforward. But my
laptop runs Windows Professional, to which the Kontrol 49 is only any good as a
MIDI keyboard connecting through its five-pin sockets.
One thing to get your head around when setting up the Kontrol 49 with your
software is that although it is equipped with two physical MIDI Outs and one
physical MIDI in, what the computer actually sees are five ports. Two of these are
virtual routes through which the Kontrol 49 sends note and controller data to your
software, and another is the virtual route by which computer transmits info
(typically program dumps) back to the Kontrol 49. This means that you can use
the unit both as a USB controller for your software and as a one-in/one-out MIDI
interface between your computer and external hardware.

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Korg Kontrol 49

Summing Up

In both the Microkontrol and the Kontrol 49, Korg have succeeded in producing a
cost-effective way of providing flexible and relatively intuitive control over a
typical collection of studio software. If you're the kind that knows your NRPNs
from your RPNs and devours MIDI implementation charts in the way others do
romantic novels, then the level of programmability you can achieve with the
Kontrol 49 is impressive. But the good news is that if you're the sort who doesn't
particularly want to get down and dirty with arcane MIDI parameters, then the
included templates do seem to work straight out of the box.

Personally, I think it's actually the combination of the hardware with the editor/
librarian that really makes this an appealing package, which makes the cheese
very hard indeed if you're not running the right operating system.

The Microkontrol deserved its positive review, and with the benefit of its refined
software and its bigger complement of full-sized keys, the Kontrol 49 is a
welcome extension (if you'll pardon the pun) of the original offering.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Korg%20Kontrol%A049.htm (4 of 4)9/27/2005 9:22:35 PM

 Mindprint Trio

In this article:
Front End For DAW Studios
Mindprint Trio
Input Conditioning Processor & Monitor Controller
Connectivity Published in SOS May 2005
Working With The Trio Print article : Close window

Mindprint Trio £329 Reviews : Processor

pros
Cost effective.
Combines a very capable
front end with flexible
monitoring.
Mindprint cram mic and line channel strips, monitor
control, and talkback into a single desktop unit.
cons
DAW connections on
unbalanced phonos only.
Paul White
Talkback switch noisy.
summary
Mindprint's Trio is designed to meet the
As an alternative to a
separate front end and
specific needs of the desktop audio
monitor controller, the Triomarket and comprises a very practical
provides a solution that is combination of a channel strip and a
monitor-control section, complete with
both cost effective and tidy. I
was also impressed by the headphone amp and talkback facilities.
musicality of the compressor.
Although it doesn't include a computer
information audio interface as such, it does have
£329 including VAT. optical S/PDIF inputs and outputs, so it
Mindprint +49 6851 9050. could be used with the new Apple Mac
+49 6851 905 200. G5 computers without the need to buy
Click here to email a separate audio interface. The voice-
www.mindprint.com channel part of the package is based Photos: Mark Ewing
around a very respectable Class-A mic/
instrument preamp with switchable 48V phantom power, compressor, and
equaliser, but there's also a separate stereo line input with its own EQ that can
be mixed with the mic input, as well as an auxiliary monitor input that could be
used as a two-track return. Mindprint have taken part of the Trio's compressor
design from their rather more costly DTC dual recording channel, and it provides
programme-dependent adjustment of the time constants so that only a single
knob is needed to adjust the processing. Furthermore, the mic channel's EQ is
tailored specifically to vocals so you get the job done with fewer distracting
controls.

Front End For DAW Studios

Clearly Mindprint have tried to provide all the essential front-end and monitoring

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Mindprint Trio

features required for use with a typical desktop recording setup, though they've
steered clear of including a USB or Firewire computer interface, as different
users will have different requirements in this area. All the analogue inputs may be
used at the same time, and there are also two separate headphone amps, each
with its own volume control. Latency-free monitoring is catered for by allowing the
input signal to be routed directly to the monitoring section during recording,
though this obviously means switching off software monitoring in the computer to
avoid hearing both the direct and delayed signals.

Also included in the monitoring section

is talkback functionality; speaker level
control with Mono, Dim (20dB
attenuation), and Mute buttons; and
provision to switch between three pairs
of speakers. On a practical level, the
unit is compact without being cramped, and its cast-metal case is finished in a
pleasant rubbery coating with a very nicely styled red front panel. There are
rubber feet on the base, and the unit is surprisingly heavy, so it shouldn't creep
around on your desk. All the connections, including the two headphone jacks, are
on the rear panel, and the integral talkback mic is just below the centre of the
front panel.

All the channel-strip controls are located on the left of the unit, and all the monitor
control facilities are on the right. Those controls relating to the zero-latency
monitoring have their own section at the bottom centre of the panel, where you'll
also find the auxiliary input's level control. All the pots have red caps with clear
white marker lines, and most of the buttons are chunky rubber affairs with in-built
status LEDs. Power comes from the obligatory power adaptor, and there's a
ground terminal (of the type used to earth record decks) on the rear panel — this
is a useful addition for systems powered entirely from mains adaptors, as you
may get a hum if you have no grounds at all. Also on the rear panel are three DIP
switches for setting the optical S/PDIF to internal or external sync and for
selecting 44.1kHz, 48kHz, 88.2kHz, or 96kHz sample rates.

Input Conditioning

The Trio has two input channels that are mixed before being fed to your DAW,
the main one being the instrument/mic channel. This features a gain control with
switchable 80Hz low-cut filter, a two-band EQ tuned for vocals, and a Fat knob
which brings in the compression. Unlike the more general-purpose EQ in the line
channel (which features shelving at 120Hz and 9kHz), the mic-channel EQ's
shelving frequencies are set to 100Hz and 7.5kHz and have a Chebyshev curve,
which simply means there's a bit of a dip before the boost comes in to give it a
vintage EQ character. All the EQ bands have a ±12dB range. A round button
activates the phantom power, while a further knob sets the output level being
mixed with the line channel, the latter accommodating mono or stereo line
signals.

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Mindprint Trio

Mindprint have designed the mic channel's

Fat compressor to automatically adjust its
gain depending on how much compression is
applied, so the level should remain
reasonably constant during adjustment.
Although there's no gain-reduction meter in
the traditional sense, a multi-coloured LED
shows green when there's no compression,
orange during moderate compression, and
red when the signal is really being hammered!
This compressor has a soft-knee
characteristic and so behaves fairly benignly.
Each channel has its own Mute button. The
line input incorporates a basic two-band
shelving EQ and a level control. A 10-
segment meter at the centre of the panel can Unusually for a desktop unit, the
underside of the Trio carries
be switched to read the input or the output important controls. A set of DIP
level but not the compressor gain reduction. switches assigns the mic and line
When in input mode, the left meter shows the input channels to the recording and
mic-channel level while the right meter shows monitoring outputs. At the top of the
the summed line-input level. picture you can also see the pair of
trimmer pots which are used to set
suitable levels for the rear-panel
Bottom centre on the front panel are three DAW Interface connectors.
knobs, two to control the level of mic and line
channels to be fed directly to the monitor section for zero-latency monitoring, and
there's also a control for the stereo auxiliary input. In the output section are feeds
for three sets of monitors, where the first (A) is on jacks and the other two (B &
C) are on phonos. A single large level control adjusts the speaker output levels.
There is a screwdriver level-trim control on the bottom panel for monitor output B,
so active monitors fed from output C need to have their levels matched using
their own input trim controls — passive monitors can be adjusted by using their
own amplifier gain controls. Two further trims set the main DAW analogue input
and output levels. Right next to the speaker outlets are the two independently
controllable headphone outputs, leaving just the AC input and power switch at
the end of the panel, next to the ground terminal.

Connectivity

The mic input is on a balanced XLR, while the mono instrument input and stereo
line inputs are on quarter-inch jacks. A pair of quarter-inch jacks provide an insert
point that may be used for connecting additional processors between the preamp
and the compressor. As separate send and return jacks are provided, the send is
always active and can be used as a further output without upsetting the signal
flow. The stereo mix of all the possible input sources is fed to the stereo DAW
Interface Out which, in this case, is on unbalanced phono connectors, as is the
corresponding DAW Interface In.

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Mindprint Trio

There is, however, a cunning set of DIP switches on the bottom panel which
serves two functions: firstly, it allows the routing of the mic and line channels to
the DAW Interface Output jacks to be configured; and, secondly, it lets you
determine how the channels are monitored, so an overdub can be monitored in
the left ear while the original track plays in the right, for example. The default
setting is to have all four switches on, which sends both the mic and line signals
to both sets of DAW inputs as well as to both channels of the monitoring system.

The DAW output, also on phonos, is fed via an internal 24-bit converter to the
optical S/PDIF connector. Both optical S/PDIF ports are located at the lower
edge of the rear panel along with the DIP switches for setting the sample rate
and sync status. Two further Aux In phonos feed the monitor section directly. As
mentioned earlier, these could be used as a two-track return, but you could also
use them to play a CD, keyboard, or guitar preamp directly into the monitors
without the computer being on.

The integral talkback mic routes into the headphone outputs. The switch is non-
latching and any other audio being monitored is dimmed when the talkback is
active. A Monitor On switch allows the input channels to be routed directly to the
monitor section‚ again useful for trying things out without your computer running.
Similarly, a DAW On button provides a fast way to feed the outputs from the
computer's audio interface to the monitor section. Separate latching buttons are
used to switch on the monitors, so all three pairs can be active at once when
needed — the Mono and Dim buttons apply to whichever monitors are selected.

Working With The Trio

The Trio proved to be very easy to set up and work with, and with very few
exceptions it worked impeccably. The mic preamp sounds clean and solid,
though a little hiss is evident when you turn up the HF EQ controls, even when
using a sensitive capacitor mic close up. This probably wouldn't be noticeable
under normal vocal-recording conditions, but could become audible when
recording more distant sources or particularly quiet vocalists with less sensitive
microphones. The EQ frequencies and curves seem well chosen from a musical
standpoint, while the compressor is particularly impressive in thickening up
vocals without making them sound excessively processed.

Although the general monitoring facilities are comprehensive and the phones
feed is both loud and clear, there is quite a loud 'clonk' when you activate the
talkback. This is simply the mic picking up the switch operation, but if you have
your phones turned up loud, it's rather like having your head in the bucket and
somebody tapping on it with a spanner. Looking on the bright side, it certainly
gets the attention of your listeners on the other end of the cable!

Overall I really like the concept of the Trio, and it delivers a lot of quality and
flexibility for the UK price. Its mic amp is at least the equal of what you'd expect
to find in a mid-price mixer, and I think the compressor is exceptional for

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Mindprint Trio

something you can drive with only one

knob. I love the packaging — it's just
the right size and it stays put on your
desktop, though I'd still have preferred
a choice of two-track monitor sources
for playing back CDs and so on. You
can use the Aux input for this, of
course, but on the Mackie Big Knob
that I reviewed recently there are three
two-track inputs, and I often used
these to connect my Roland V-Drums
and Line 6 Pod guitar preamp for
simply playing through the studio A further trim pot on the left-hand side of the
monitors. Having optical S/PDIF I/O is Trio's underside allows level adjustment of
a undeniable bonus, though, and while one of the three pairs of speaker outputs. All
many pieces of audio equipment use the trimmers are recessed to avoid
the coaxial rather than optical variant accidental misalignment, and are slotted to
allow easy adjustment using a flat-blade
of S/PDIF, it does provide a simple screwdriver.
way for users to connect to those
computers that have optical S/PDIF I/
O, such as the Apple G5 range.

Of course you can't provide all the features to satisfy everyone without hiking up
the cost to the point where people aren't prepared to pay for it, so a line has to be
drawn somewhere. I would have liked balanced I/O for connecting to the DAW,
rather than phonos, though this won't be a significant shortcoming in most studio
setups. My feeling is that Mindprint have come up with a great product. For those
musicians who record only one or two audio parts at a time, it combines all that is
essential in a stereo monitor controller with a simple yet very smooth-sounding
front end, and it therefore does away with the need for a separate channel strip
or mixer. For the desktop studio user who works mainly alone or with one
musician at a time, Mindprint's Trio is a very appealing all-in-one product for
recording and monitoring.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Mindprint%20Trio.htm (5 of 5)9/27/2005 9:22:41 PM

 Native Instrument Battery 2

In this article:
The Master Section
Native Instrument Battery 2
Sample Matrix Virtual Drum Module (Mac/PC)
Edit Pane Published in SOS May 2005
Conclusions Print article : Close window

NI Battery 2 £150 Reviews : Software

pros
Compatible with a wide
range of sample formats.
Powerful modulation options.
Two years on from its original release, Native's virtual
Ability to swap sets of Cells
across kits. drum module gets its first full upgrade. Is it all John
Inspiring range of dynamic Bonham tom mayhem, or is it limper than a Kraftwerk
filtering options. drum solo?
Large number of audio
outputs.
cons Paul Ward
Supplied library lacking in
variety. I've been an ardent fan of Native
summary Instruments' original Battery sample-
If you're messing around with based software drum module since it
a full-blown hardware or was unveiled back in 2001. It was
software sampler just to great for slinging together a 'kit' of
handle your one-shot drum
sounds, Battery will be like a drum samples and for coming up with
breath of fresh air — it's so rhythms quickly, and much faster than
quick and easy to use. It using my old Akai hardware samplers
handles just about any to do the same thing. But there were a
sample material you can get few quirks in its operation, and I found
your hands on and the simple
editing options allow you to myself hoping that an upgrade would
tailor the sounds quickly for take care of them someday. Now, at Battery 2 has the same quasi-military look as
use. In short, I can't imagine last, it's here — does it smooth out the the original version, as you can see in this
anyone working with wrinkles of the previous version? shot of the program running under Cubase
percussion samples that SX on the review PC. The Master section is
wouldn't benefit from having the top strip of this window, above the
Battery 2 on their computer. Battery 2 will run as a stand-alone Sample Matrix of grey Cells in the middle,
information application, or as a plug-in under and the Edit Pane is the section at the
VSTi-, DXi-, RTAS-, or Audio Units- bottom. The left-hand part of the Edit Pane
£149.99 including VAT. changes according to the selections made
Arbiter Music
compatible hosts. Native Instruments on the tabs at the top left of the pane, as
Technology +44 (0)20 8970 claim that it will run in 256MB of RAM shown in the other screenshots in this article,
1909. and up to 4GB of disk space under while the display on the right shows the
+44 (0)20 8202 7076. 400MHz Athlon or Pentium 3-based waveform of the currently selected Cell for
PCs, or a 500MHz G3 Mac, but as editing.
Click here to email
www.arbitermt.co.uk usual, they suggest using beefier
www.native-instruments. machines than these for optimum performance; a 1.2GHz Pentium 3/4 or Athlon
com PC, or a 1GHz G4 Mac, with 512MB of RAM, is recommended. I ran the review

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Native Instrument Battery 2

copy on a 3.4GHz Pentium 4 in 1GB of RAM with no problems, and it demanded

Test Spec little more than 10 percent of my total CPU power in typical use (ie. when using it
3.4GHz Pentium 4 PC with to handle main rhythm, percussion and tom-fill duties) with the largest of Battery
1GB of RAM running Windows 2's new factory kits loaded.
XP (Service Pack 2).
Native Instruments Battery 2. At first I thought that NI had forgotten to include the library disks with Battery 2.
Steinberg Cubase SX v3.01. The disk insert states that it contains the 'Installation CD' — but it's actually a
DVD. Anyone lacking a DVD reader on their PC or Macintosh, take note...! In
addition to Battery 2 itself, the DVD contains over 3GB of sampled drums
including acoustic, electronic and percussion kits. The kits from Battery 1 are
also provided, which is handy, as you can use version 2 in your old songs without
having to re-import the v1 sounds. Whatever minor gripes I may have had about
the original Battery's user interface (of which more in a moment) I can vouch for
the quality and effectiveness of the Battery 1 kits, having used them time and
time again. I'm glad to report that the new material is generally of an equally high
standard. There's a power and liveliness to the samples that sometimes makes
you feel as though there's a real drummer sitting next door surrounded by
expensive microphones! I would have preferred rather more variety in the factory
sets, however — particularly a few more acoustic kit variants. Some of the
acoustic kick drums sounded a little flabby, and I would have liked a few more to
choose from.

The user interface is similar to that of the original version of Battery, with its rows
of 'Cells' containing your samples, but NI have tidied up the main panels and
arranged them more logically. The Battery window is conceptually and physically
divided into three sections, comprising the Master Section, the Sample Matrix
and the Edit Pane (see the large screenshot opposite). The Master section is
where drum kits are managed and where the overall volume of Battery is
controlled. The Sample Matrix, which dominates the window, maps out the Cells
containing the samples into columns and rows, much like a spreadsheet, and
gives you an overall view of the current kit. The Edit Pane, in the bottom third of
the window, is where you get down to modifying the behaviour of Cells and the
samples within those Cells.

The Master Section

This section, at the top of the Battery window, contains drop-down menus for
File, Edit and View functions and a 'quick-select' drop-down menu for selecting
kits stored in the hard drive location specified by Battery 2's sample 'path'. Useful
information displayed here includes polyphony/used polyphony, memory
requirements for the currently selected kit and the master volume.

As with some of their other recent sample-based releases (such as Elektrik

Piano), Native Instruments have endowed Battery 2 with 'Direct From Disk'
facilities, making it possible to use sample files that would otherwise take up
more space than is available in RAM. You do have the option to turn this feature
on or off, since the downside of DFD is that it increases disk and processor

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Native Instrument Battery 2

activity.

I felt the original file- and sample-

handling system in Battery was
somewhat clumsy, and I was pleased
to see that this had been streamlined
in the new version — I've lost count of
the times I've picked the wrong option
from Battery's File menu, ending up in
the 'Add Cell' dialogue when I really
wanted to load a new set of Cells.
Mapping samples to a Cell at different
Battery 2 treats its files intelligently; so velocities.
if you choose a kit, the program now
realises that you want to load a kit and
performs that function, whereas if you choose a WAV or AIFF file, it loads that
sample into a Cell. There's also a list of recently used kits, which is a nice time-
saver. The influence of Native Instruments' Kontakt is discernible in the wealth of
supported file formats: Kontakt Instruments, Battery v1 kits, Battery 2 kits and/or
Cells, Sound Designer II files, WAVs, AIFFs, Soundfonts, and files from the
following hardware and software samplers: HALion, Samplecell, LM4,
Gigasampler, Recycle, the Akai S1000 and S3000 and MPC-series sequencer/
samplers — and all from eight- to 32-bit resolution. It's an extremely impressive
list.

A new 'Import' function brings all of the supported file formats together into a
browser window, similar to that used in Kontakt. I've never been totally convinced
by this browser window, even in Kontakt; I find it a bit pokey and I'd also prefer to
have access to some simple file-management functions from here, such as the
ability to rename, copy and move samples — the browser format merely hints at
such possibilities. I would also like to see the double-pane approach adopted,
like the standard Windows Explorer panel, to make the navigation area less
restrictive.

Choices for saving information have been given some attention too. There's now
the option to save pointers to your samples, rather than have them replicated by
Battery in its own folders — although you can still choose that method for
maximum flexibility. You can save selections of Cells, making it easier to build
kits from these collections, rather than having to load them one at a time, or start
from another full kit. It's simple to save a Cell collection of your favourite
percussion rack and then pop it into any kit. Brilliant! This has the potential to
avoid an enormous amount of repetitive work.

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Native Instrument Battery 2

Sample Matrix
The spreadsheet-like Sample Matrix displays your currently selected kit. Each Cell
represents a sample, or collection of samples assigned to a specific MIDI note
number, or range of notes. Up to 128 samples can be held in a Cell and these are
layered, or split across velocity ranges (with or without crossfades) as required.
The number of Cells is no longer fixed, as it was in the previous version. You can
add or delete rows up to a maximum of 72 Cells and view them in rows of six or
12 Cells. Individual Cells may be soloed or muted, or you can select non-
contiguous Cells with combinations of modifier keys. Selecting, muting, or soloing
rows and columns of Cells is a one-click task, and you can similarly combine the
selection of both rows and columns. Personally, I like to separate kicks, snares,
cymbals, toms, percussion and loops onto separate rows for selective muting/
soloing. A pair of indicators at Cell, row and column level show whether they are
muted or soloed.
Just to the right of the mute/solo buttons in each Cell is a field that can display
one parameter, such as volume, pan position, filter cutoff, or one of many other
values. This parameter can be fixed, or can change depending on the parameter
you are currently editing. One feature that I sorely miss from the earlier version of
Battery is the ability to see a Cell's full key range in this display field. You can now
see either the low or high key value in there, but not both simultaneously. This is
no doubt down to screen space restrictions, but it's certainly going to slow me
down, and I'll wager I'm not alone. However, it is possible to move, copy and swap
Cells, with or without their associated key range. This is the real strength of
Battery's ability to organise drum kits quickly and easily.

Edit Pane

The Edit Pane is where detailed sample editing takes place. Here, Cells can be
treated to an array of editing options and individual samples can be tweaked into
shape. And when I say 'tweaked' you can read 'mangled beyond all recognition'!
Firstly, setting up sample layers is much easier than it was with the original
Battery, and is helped enormously by the user-friendly, Kontakt-like assignment
display (shown on page 209). I'd still balk at the thought of mapping 128 velocity
layers in a Cell, but if you've got the time, Battery 2 will allow you to do it.

The basic editing features are familiar from the original version of Battery,
including all the typical features you'd expect, such as volume, pitch, and also
amplitude and pitch envelopes. A bit-reduction control has been added, in case
you long for the days of the Akai S900, and there's a sample-frequency reduction
control if you really want to get back to the days of the Sinclair Spectrum! Gone is
the 'Shape' control of Battery 1, but the 'Saturation' knob does a similar job,
adding distorted higher harmonics, as its name suggests, and is always worth a
turn if you feel that your drums need a little more energy.

If there's one feature I always felt was missing from Battery's armoury it was a
dynamic filter. I have quite a few old samples that have a great attack phase, but

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Native Instrument Battery 2

become very noisy once the signal

level starts to fall away. In the past I
used the old Battery's volume envelope
to deal with this, but this inevitably
leads to some 'choking' of the sound,
and this is particularly unwelcome
when the after-ring or ambience forms
an important part of the sample.
Fortunately, Battery 2 arrives with a
freely assignable modulation envelope.
I can simply apply an envelope to the
filter cutoff frequency and with a little The new dynamic filter.
tweaking, that nasty noise is removed
while retaining the character of the sample's decay. The filter's pretty flexible, too
— there are 15 types, including low-pass, high-pass, band-pass, three-band EQ,
phase and vowel variants.

Battery 2 also adds a built-in compressor, and this is welcome, removing the
need for external processors in many cases. It doesn't have all the control and
flexibility of a dedicated dynamics processor, but I believe it would get the job
done in all but the most critical cases. More importantly, it can be applied on a
Cell-by-Cell basis, rather than across the output of the entire instrument, so it
may allow you to avoid having to tie up outputs simply to add compression to one
sound. Controls are provided for Threshold, Ratio, Gain, Attack and Decay.

If you found Battery a little lacking in

modulation features, then Battery 2
should be more to your taste. Up to
eight modulation paths can be created,
with controls to determine the
modulation source, destination and
amount in positive or negative values.
The list of modulation sources is
impressive, including velocity, the mod
wheel, mono/poly aftertouch, key
position, LFO or any of Battery 2's
envelopes (there's even a dedicated The built-in compressor can be applied to
AHDSR modulation envelope). You individual Cells with different settings.
can also define up to eight global MIDI
controllers to act as modulation sources across any of Battery 2's Cells. The
modulation destinations are no less impressive, taking in such staples as volume,
pan, filter cutoff and resonance, but there are also more esoteric options, such as
bit depth, saturation, loop start/length and pitch envelope. Battery 2's LFO is also
surprisingly flexible, with controls for waveform (a choice of sine, sawtooth and
square), fade-in and pulse width (on the square wave only). The LFO can be
made to re-trigger when the Cell is played, and you can also make the LFO
speed sync to the host sequencer's tempo.

Battery has never been a loop player in the way of tools such as Phatmatik or

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Native Instrument Battery 2

Intakt, and that hasn't really changed (although there's absolutely no reason why
you can't use Battery to trigger loops), but NI have added some simple looping
tools. Up to four loops can be created within a sample, with crossfades and
variable tuning. Each loop can repeat a fixed number of times, or infinitely. While
we're on the subject, I often use Battery to split a single-hit drum sound from a
loop by setting the sample start position and using the volume envelope to isolate
a particular sound. Using this method means you can keep the loop as a single
sample, but use variants across a range of Cells to pick out other drum hits you
want to use. Battery 2 makes this very easy. Finally, each Cell now has the
choice of being sent to one of up to 16 mono and eight stereo outputs. This
number of outputs is a vast improvement over the previous version and wins
another big smile from this reviewer.

Conclusions

All in all, I'm a big fan of Battery 2 and its many enhancements. I'm still not keen
on the Browser window in its current form, and I am going to miss the
simultaneous high/low note display in the matrix. If I were to compile a wish list I
would suggest the ability to use one or two outputs as aux/effects sends. I'd also
like to be able to choose the colour of Cells to assist my brain in finding its way
around — no matter how I arrange the matrix, I can't seem to find things quickly
enough. A mixer page with faders would assist when balancing kits, perhaps with
the ability to choose the parameter (other than simply volume) that the faders are
currently controlling — balancing tom tunings, for example, can be a bit long-
winded on a Cell-by-Cell basis. The library material, although high on quality, is
somewhat lacking in breadth. I would really have liked some Simmons drums,
classic analogue drum machines or some more 'produced' acoustic kits. I've
often used Battery's simple interface to play bass lines and vocal samples, so
some more examples of this nature would be useful to show off Battery's
potential.

I liked the original Battery so much that I had it plugged into my default song, so
that it fired up with Cubase. The best praise I can give to Battery 2 is that it is so
good that it makes Battery seem cumbersome and inflexible! Battery 2 is now
loading with my default song, and I doubt I'll ever look back. In my opinion, there
is simply no easier way to get your drum samples into a host sequencer than by
using Battery 2.

Published in SOS May 2005

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Native Instrument Battery 2

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Native%20Instrument%20Battery%202.htm (7 of 7)9/27/2005 9:22:47 PM

 PMC TB2SA & DB1SA

In this article:
The Mystery Of Model
PMC TB2SA & DB1SA
Names Powered Monitors
Technical Specifications Published in SOS May 2005
Flying Mole Amplifier Print article : Close window
Listening Tests
Reviews : Monitors
PMC TB2SA & DB1SA
£1616/£1169
pros
Tweeter and crossover Pioneering digital amplifiers are combined with PMC's
upgrades have raised the
standards yet again. proven transmission-line cabinet designs to deliver
Uncanny bass extension spectacular monitoring performance at a project-
and power handling from such studio price.
small boxes.
Superb resolution and
stereo imaging.
Hugh Robjohns
The Flying Mole
amplification approaches that
of the Bryston amps, but The Professional Monitor Company —
without the weight, heat, or now known more simply as just PMC
cost. — have built up a phenomenal
The tweeter upgrade and reputation for their entire range of
Flying Mole amps are
available for older systems.
monitors, from the massive BB5/XBD
system down to the tiny DB1. These
cons products can be found in mastering
You'll have to take them houses, film dubbing suites, recording Photos: Mark Ewing
home once you've auditioned The TB2SA and DB1SA monitors (outside
them!
and broadcast studios, and outside
broadcast trucks, and are increasingly and inside, respectively), complete with their
summary making inroads into the homes of well-
Flying Mole amplifier packs.
PMC's two smallest passive heeled hi-fi buffs.
monitors are now available
with an integrated Class-D
amplifier to form self-powered Many of these products have already been reviewed in the pages of Sound On
Activated monitors that deliver Sound, including the DB1, TB2, FB1, and AML1, and all employ highly
class-leading performance in
a convenient and cost-
sophisticated transmission-line loading principles unique to PMC. The majority of
effective package. The PMC's products are passive, meaning that they employ high-level crossovers
remarkable quality of the constructed from passive components — albeit highly specified components laid
Flying Mole amplifiers, out with enormous care on huge and complex circuit boards behind the terminal
coupled with recent upgrades panels. Only the largest systems — the BB5 and MB2 speakers — and the
to the DB1 and TB2 monitors,
produces stunning sound
compact AML1 are available in fully active configurations.
quality that has to be heard to
be believed. PMC have long argued that while the active approach has some significant
information advantages over a passive design, it is an expensive solution requiring high-
TB2SA, £1616 per pair; quality active components — filters and amplifiers. The large active BB5 and
DB1SA, £1169 per pair. MB2 systems, for example, employ specially modified Bryston active crossovers

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PMC TB2SA & DB1SA

Prices include VAT. and Bryston amplifiers throughout. Even the AML1 employs Bryston circuitry built
PMC +44 (0)870 444 under licence.
1044.
+44 (0)870 444 1045.
Click here to email
Making active speakers to suit the budgets of the home-studio market while still
meeting the quality threshold demanded by Peter Thomas, Managing Director
www.pmc
and Chief Designer of PMC, has so far proved impossible — not, apparently,
loudspeaker.com
through lack of trying! To date, Peter has simply not found a way of incorporating
www.flyingmole.co.jp
active crossovers and amplifiers of sufficient quality at a low enough price to
make this dream a reality, and the company has long recommended instead
partnering their passive speakers with a top-quality amplifier to achieve the
product's performance potential.

To that end, they have long offered the neat solution of the Bryston Powerpac
amplifiers — single-channel self-contained units, available in a range of powers
from 60W to 240W, which can be bolted to the rear panels of the speakers to
produce a convenient 'powered speaker' — an approach PMC like to call
Activated. While this solution lacks the 'tweakability' featured in many active
designs, and it is relatively expensive, it works very well indeed and the
resolution and quality surpasses many similarly priced active designs.

Regular Sound On Sound readers will already know that I have used a pair of
TB1s, Activated by Bryston Powerpac 120s, for many years with no complaints
whatever. The sound quality and resolution are superb given the system's size,
although the combination is relatively heavy and quite expensive. However,
these criticisms have now been addressed with the launch of the TB2SA and its
diminutive brother the DB1SA. Essentially these are standard TB2 and DB1
speakers with re-engineered cabinets to incorporate Flying Mole monobloc digital
amplifiers in a very neat package. The amplifier is incredibly lightweight and
highly cost-effective, while giving remarkably little away in sound quality terms to
the heavy and expensive Bryston. It may sound too good to be true, but the proof
is in the listening!

The Mystery Of Model Names

First off, it might be useful to explain the logic behind the model names and some
of the recent developments, which have left some potential purchasers a little
confused. The TB2 is named as the second generation of the 'tiny box'
loudspeaker — it's older and discontinued sibling being the TB1. The newer
model introduced improvements to the transmission line and crossover, as well
as rounded baffle edges to enhance stereo imaging. The 'S' in the model name
refers to the Studio version — the most cost-effective model in the range, with a
painted black cabinet rather than the more expensive range of real wood veneers
favoured by up-market studios and hi-fi types. Finally, the 'A' suffix points to the
Activated status of the model, with the incorporated Flying Mole amplifier.

Interested parties may become confused by another suffix — the '+' model.
Recently, PMC decided that economies of scale would allow the original metal-

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PMC TB2SA & DB1SA

dome tweeters fitted to several models

— including the DB1 and TB2 — to be
upgraded to the soft-dome tweeters
used across the board in the more
expensive models. This upgrade was
indicated by the '+' suffix, and the
current production models of the
passive speakers are marketed as the
DB1S+ and TB2S+ (older metal-dome
systems can be upgraded — contact
your dealer for details). However, in The Flying Mole's simple controls are
the interests of simplicity, the Activated provided on the top panel of the amplifier
models, which all employ the same pack for ease of access — this picture was
taken from directly above the monitor.
soft-dome tweeter, don't include the '+'
suffix.

Last but not least, is the 'M' version, which means that the monitor is
Magnetically Corrected — this is only available as an option, rather than being
built in as standard. Given the almost universal use of LCD computer monitors,
and the growing use of LCD and Plasma TV screens, magnetic shielding is far
less important than it once was, and hence many will enjoy the reduction in cost
that derives from making the shielding an optional extra.

Phew! Hope you were paying attention, because I'll be testing you later...

Technical Specifications

Supplied for review were pairs of both DB1SA and TB2SA monitors. The TB2S
and DB1S have both been reviewed before (in SOS November 2001 and
January 2003 respectively), so rather than go over exactly the same ground
here, I'll concentrate mainly on the differences. First, though, it might be worth
giving the outline specs of each model as a reference point. The TB2 measures
400 x 200 x 350mm (hwd) and weighs just over 9kg. The amplifier is mounted
vertically in a cabinet extension from the upper half of the rear of the speaker,
and the connection between amp and speaker is courtesy of a right-angled
Speakon connector that protrudes from the foam of the transmission-line port at
the bottom of the rear panel. It is a very neat solution, and the integration of amp
and speaker is elegant and practical. The new rear panel is equipped to accept
an Omnimount bracket for wall mounting, if required. PMC claim the usable
frequency range for the TB2 to be 40Hz-25kHz, with a peak SPL (at one metre)
of 111dB. The bass driver employs a doped-paper cone in a 170mm cast-alloy
chassis, coupled at 2kHz to a 27mm fabric soft dome.

The DB1 and its amplifier housing are constructed in a very similar way to those
of the TB2, but with scaled-down dimensions. Again, the new rear panel is
equipped for wall mounting, but rather than a standard Omnimount system PMC
supply an optional bespoke wall bracket. This tiny speaker measures 290 x 155 x

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283mm (hwd) and weighs just 5kg. The soft-dome tweeter and crossover
frequency are the same, but the doped-paper bass driver is mounted in a 140mm
cast-alloy chassis. As you would expect, the low-frequency extension is not as
great, and neither is the power handling, with specifications of 50Hz and
108dBSPL, respectively.

The Flying Mole amplifier unit is fitted so that its power switch and volume control
are both at the top, while the bottom panel provides an XLR input, IEC mains
inlet, and the pre-connected Speakon output. The amplifier power output is
quoted as 120W when coupled to the PMC monitors, and the manufacturer
claims a power conversion efficiency of over 85 percent — hence a power
consumption of just 25W (6W when in standby mode with no input signal) and no
heat to worry about!

Flying Mole Amplifier

The name Flying Mole seems an odd one for a Japanese amplifier manufacturer.
Apparently it derives from the notion that the greatest goal in life for a mole
scurrying around underground might be to fly — well, it always has been hard for
Westerners to understand the Oriental mind-set! But regardless of the company
name, the products are impressive.
Digital amplifiers — Class-D amps — are nothing new, and lots of active
loudspeakers already use Class-D amplifier designs. However, most leave
something to be desired when auditioned; Class-D amps are often harsh-
sounding and lack the resolution and 'inky black' silences that a good analogue
amp can deliver. However, the technology has improved significantly in recent
years, and Flying Mole's unique Bi-phase Fusion Technology certainly seems to
be able to deliver the goods. In some ingenious but secretive way, this technology
integrates the amplifier's switched-mode power supply with the switching amplifier
circuitry, resulting in unusually high power efficiency and no audible artefacts.
Most analogue amps struggle to better 30 percent power efficiency and typical
Class-D amps are about 65 percent efficient. The impressive 85 percent
performance figure of the Flying Mole amps means that they always run cool and
don't require external heat sinks — they're just simple plain boxes the size of
paperbacks, but they drive loudspeakers very nicely indeed!
Strangely, the distortion figure given in the specifications (0.03 percent at 50W
output) appears rather unimpressive, yet auditioning the amp suggests it
compares very favourably to a Bryston — which quotes distortion figures with
several more zeros before the digit. The logical conclusion is that what we hear
and what we measure are not necessarily the same thing! In many ways, the
Flying Mole amps can be compared to a really good valve triode design — which
would share a similar, yet equally inaudible, distortion figure.
The DADM100pro reviewed here is available in three main versions with different
input and output connectors. The DADM100pro BI version used with the Activated
PMCs features a balanced XLR input and Speakon output, while the BB version
retains the XLR input but uses 4mm binding posts for the outputs. The last version
is the HT model, which couples an unbalanced phono input socket with 4mm
binding posts for the output. The last two models measure 43 x 132 x 238mm
(hwd), while the BI model isn't quite as deep (223mm). All weigh a ridiculous

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650g. In fact, the amplifier is so lightweight that during the review a heavy speaker
cable was easily able to pull it off the desk!
In addition, there are dual AC/DC versions of each of the above, equipped with a
12V DC power input in addition to the standard IEC mains inlet. The DC input is
useful for portable and outside-broadcast applications, and the amp draws a
modest 5A of current at full power! There are also various brackets and mounting
kits available to fix the amps to the rear of any speaker, to a 1U rackmount shelf,
or in a six-pack lump!

Listening Tests

I have owned a pair of TB1S speakers for many years and use them when
recording on location. Recently they were upgraded to the TB2+ spec with the
soft-dome tweeters and crossovers. As already mentioned, Bryston Powerpac
120 amplifiers are bolted to their back panels and provide plenty of clean,
articulate 'welly' to allow the monitors to deliver their best. As it happens, I also
have a pair of DB1S speakers (with the original metal-dome tweeters) and a pair
of stand-alone Flying Mole amplifiers which I use when editing — so all in all,
there are sufficient elements to hand to provide valuable references and
comparisons.

The first thing to mention is the significant improvement wrought by the new
tweeter and amended crossovers — to both the TB2 and DB1. The old metal-
dome tweeter was certainly no slouch, but the soft dome — which is admittedly a
much more expensive device — has clear advantages in terms of the precision
and naturalness of the high frequencies. The crossover changes that the new
tweeter required have improved the mid-range resolution too. If the TB2 speaker
rated at an eight out of 10 before, it's the full 10 out of 10 now, and the DB1 has
benefited from a similar improvement in resolution.

Furthermore, the matching of timbre

and imaging is now almost flawless
across the entire PMC range. I was
able to compare the DB1SA and
TB2SA not only with each other, but
also directly against the larger LB1 two-
way and huge IB1 three-way monitors.
Aside from the inherent increases in
headroom/SPL and bass extension
(and the additional resolution of the
three-way design), the sound character The amplifier's audio I/O and mains
barely changed at all — which is a connectors are on the underside of the
praiseworthy achievement. amplifier pack, and a sturdy right-angled
Speakon connector links the amplifier output
with the speaker's crossover.
The Flying Mole amplifiers appear to
defy the laws of physics — I can think of no other explanation! I compared the
Flying Mole DADM100pro BI monobloc amplifier (see the 'Flying Mole Amplifier'
box for more details) directly with the Bryston Powerpac 120 and found it hard to

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PMC TB2SA & DB1SA

tell the difference! The Bryston is large, heavy, and very expensive, while the
Flying Mole is none of those things — and yet they produced near identical
sound. Okay, so in the final analysis the Bryston retained the edge, with more
headroom, better bass control and sustain, and the ability to convey transients
and dynamic changes with an effortless ease that the Flying Mole couldn't quite
equal. But without a direct A/B comparison it would be very hard to tell them
apart. Not all Class-D designs are equal, and I share the view expressed by
Peter Thomas that the Flying Mole amps are in a class of their own — the first
switching amp that comes within a whisker of equalling the best analogue
designs.

Overall, then, the new Activated versions of the DB1 and TB2 are unqualified
successes. All of the qualities of the original monitors have been retained — the
superb bass extension and control, the consistent sound balance regardless of
monitoring level, the vast three-dimensional sound stages, the wide sweet spot,
and the neutral presentation with high levels of resolution. The '+' updates with
the better tweeter and revised crossovers have improved mid-range and high-
frequency resolution and accuracy, and have also made the sound character
more consistent across the entire PMC range. The Flying Mole amps are the
icing on the cake, matching a powerful, high-performance, high-resolution
amplifier to the speaker to form a convenient, effective, and affordable package.

The problem with passive speakers as good as PMC's is that they can reveal the
failings of inadequate amplifiers just as easily as they can poor mixes or mic
placement issues. Unfortunately, the cost of a really good amplifier often
matches or exceeds that of the monitors, so the complete package can appear
very expensive when compared to some of the active monitors aimed at the
home-studio market. The Flying Mole amplifiers, whether in stand-alone form or
fitted to the DB1SA and TB2SA, redress that balance very well, and allow the
PMC monitors to deliver a superlative performance at a far more reachable price.
It is hard to find fault with the combination in any way at all, and if you are in the
market for good nearfield studio monitors, these Activated monitors make ideal
reference points.

Published in SOS May 2005

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All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
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Web site designed & maintained by PB Associates | SOS | Relative Media

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 Propellerhead Reason v3

In this article:
What's New
Propellerhead Reason v3
A Remote Possibility Virtual Electronic Studio (Mac OS X/PC)
Combinator Published in SOS May 2005
MClass Mastering Suite Print article : Close window
Enhanced File Browser
Reviews : Software
Bits & Pieces
Conclusions

Propellerhead Reason
v3 £299
Astonishingly, Reason is now over four years old!
pros Version 3 adds performance-enhancing features and
Combinator. Full stop.
mastering facilities, losing only Mac OS 9 support on
Excellent mastering
processing. the way. We bring you the first UK review of the full
Close integration with some release version...
hardware control surfaces.
Expanded factory Refill
collection. Derek Johnson
cons
More demanding of host An incremental software update is a
computer. milestone for most software, especially
No new sound makers or if it's a really popular package. When
audio recording. that software is Propellerhead's
No real moves forward with Reason — the yardstick by which other
the linear sequencer.
virtual studios are measured — there
summary are plenty of users holding their breath.
Live or in the studio, Reason
remains top software, offering
an enviable combination of Reason's last update, to v2.5, was free
great sound and value for and introduced a host of powerful new
money. Existing users, devices. The latest release, v3.0,
upgrade now! doesn't initially seem to be in quite the
information same league (and it's not free!), but
Reason v3 £299; just a little investigation reveals that
upgrade from previous Propellerhead have enhanced their This overview provides a glimpse of the
versions £69; upgrade from Reason rack with a modestly equipped
flagship package in some unexpected Combinator installed — it takes up a lot of
Reason Adapted £199.
ways. room!
Prices include VAT.
M Audio UK +44 (0)1923
204010. If Reason is new to you, a quick recap is in order. Should you require more
+44 (0)1923 204039. detail, check out some SOS back issues: Reason was first reviewed in March
Click here to email 2001, v2.0 surfaced in September 2002, and v2.5 made a splash in December
www.maudio.co.uk 2003. In between, check out a 'Making The Most Of...' two-parter in November
www.propellerheads.se and December 2002, and have a gander at SOS's on-going Reason Notes
column, which started 11 months ago.
Test Spec

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Propellerhead Reason v3

PC REVIEW SYSTEM
For many readers, the ideal hardware electronic studio would be a mix of classic
analogue synths, drum machines, step sequencers, samplers, effects units and
3.06GHz Pentium 4 PC with an audio mixer. In essence, this is what Propellerhead took as their inspiration
512MB of RAM running
Windows XP.
when developing Reason. Its on-screen representation of these elements houses
them in a virtual rack and adds linear sequencing, automation and easily
MAC REVIEW SYSTEM configurable real-time control from hardware control surfaces. It also offers a
450MHz Apple Mac G4 with brilliantly elegant virtual jack-based interconnection system, a lot of knobs and
896MB of RAM running Mac sliders, and instant integration with a wide range of other software via the Rewire
OS v10.3.8. protocol.
Propellerhead Reason
version reviewed: v3.0.
What it has never had is an audio input, either to audio tracks or the sample-
based devices. I'll get the bad news over with now: this feature is still lacking in
v3.0. Anyone requiring linear audio tracks needs a MIDI + Audio sequencer that
supports Rewire to host Reason, and the creation of custom samples for
importing into Reason sample players requires separate sample-acquisition and
editing software.

What's New

There are two major developments in v3.0. First of all, the new Combinator turns
Reason into more of a real-time performance instrument than it has been before.
Just like the 'performance' level of a workstation synth, this new instrument
groups Reason devices into one super-device. You can layer and set up key and
velocity splits, but that's just the beginning: unlike most workstation synths,
Combinator puts no limits on the number or configuration of devices it holds,
save those imposed by the host computer's CPU and RAM. It sits in the rack as
one device, addressable from one MIDI sequencer track.

In the past, Reason owners have been

forced to go to great lengths to
repurpose existing Reason devices to
recreate the effects of integrated
mastering processors. This is no
longer necessary, thanks to v3's other
major innovation: the new MClass
collection offers four dedicated
mastering processors, and they're
really rather good.

Other new features which we'll address

shortly include a new, more accessible
patch browser (see the box at the end The new Combinator turns Reason v3 into a
of this article), built-in support for many super performance synth. Any device in the
Reason rack can be combined, layered and
hardware control surfaces, an controlled from one handy window. Here
enhanced linear sequencer, a neat there are three Subtractor modules visible in
little line mixer and improved sample the Combinator's programmer, with key
handling. splits, and three NN19s playing at different

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Propellerhead Reason v3

velocity levels. Note the Modulation Routing

window to the right.
Sadly, for some of us, there are
changes of a less positive kind.
Reason is now Mac OS X only — it ran quite well under Mac OS 9 until v2.5. It'll
also only run on Windows XP and 2000; 98 and ME are no longer in the picture.
Version 3.0 is much more demanding of its host computer, so although
Propellerhead claim that the minimum specs are a G3 running Mac OS 10.2 or a
300MHz Pentium III, think of those as absolute minimums. Reason is much
happier with faster processors (and tons of RAM). I was a little saddened that I
wasn't able to audition demo songs on my aging Mac without crackles and audio
dropouts, and even had to tweak the audio system of my 3GHz Pentium 4 laptop.

A Remote Possibility
Reason is not known for being buggy — release versions usually work first time. A
trawl around the Propellerhead web site, though, does reveal a problem with v3.
Basically, the problem is with Keyboard Control (previously known as Keyboard
Remote), the system whereby computer keys can be assigned to Reason
functions. What has been discovered is that if you set up mappings and then
change them, the mappings may change unpredictably and can occasionally
cause the software to crash. The big problem is that making changes in a song,
saving it, and then reloading it, could crash Reason. Songs with no mappings are
problem-free. Propellerhead advise users to not make any Keyboard Control
assignments in v3 until the problem has been solved. Visit the site for a full
discussion of the issues, a strategy to minimise problems if you have made
assignments, and await an update!

Combinator

It was always frustrating, in earlier versions of Reason, to be presented with four

incoming MIDI control busses (via the fixed 'Hardware Interface') that allowed
real-time layering of four devices — there was no way to use these layers in the
main sequencer except by a lot of messing around. The new Combinator
changes all that, adding many performance-level features. A Combi — the name
for the resulting agglomeration of devices — doesn't just layer sound-makers. It
allows them to be played in their own key range or velocity-triggered layers, and
you can add as many effects as you want, and integrate Matrix step sequencers
or Redrum drum machines. Not only can the result be saved as a recallable
patch, but all the virtual linkages between devices (audio, gate and virtual 'CV')
are saved as part of that patch. As existing Reason users will know, even simple
gate/CV links made within a single device cannot be saved, except in a full song.
As you might expect, any links from inside a Combi to the rest of the Reason
rack will not be saved in a patch. However, Combinator does have its own audio
inputs, so if you create effect-only Combis, you'll have one of the most user-
definable multi-effects processors you've ever played with (at least in this price
range!).

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At its simplest, Combinator is reduced to a single rack strip, like any other
Reason device, offering patch selection and name display, plus input and output
level metering. Clicking a little arrow causes the Controller panel to fold out. The
business end of Combinator, this panel offers four assignable rotary control
knobs and four assignable buttons, plus pitch-bend and mod wheels. These
controls can be assigned to any parameters in any device in the Combinator, so
one knob or button can control one parameter on each of several devices if you
wish.

Two more buttons let you run all the

pattern-based devices in the Combi
(ideal for auditioning independently of
the host song), and bypass any effects.
One of the remaining two buttons folds
out the Programmer, which lets you
define key and velocity ranges for each
sound generator in the Combi. It looks
rather like the Remote Programmer on
the NNXT sample player, and is logical
and straightforward. Key ranges are
shown by a tweakable bar under a mini All four MClass mastering processors
nestled in their very own Combinator setup.
keyboard, with an option to key in Of course, they can be used individually
exact values. Velocity ranges are anywhere in the Reason rack.
simply keyed in, and I did feel it might
have been useful if there had been a
more graphical way of showing a velocity-controlled layer than the shading effect
applied to the layer's key range bar, though this would have required much more
screen space. It would be handy to be able to apply a transposition setting to
devices in this display (if you wanted to have key-split instruments playing in
specific pitch ranges), and I'd quite like to see some sort of trigger delay option,
so that devices play a fixed time value after a note-on.

Devices appear in a list to the left of the keymap display, and highlighting a
device allows you to access the Modulation Routing section for that device, to the
right: the knobs and buttons on the controllers are assigned to device parameters
here.

The last Controller button — labelled 'Show Devices' — unfolds the Combinator
even further, so that it shows all of the devices that have been loaded into it.
Adding devices can be done in three ways. First of all, it's possible to highlight a
group in the main rack and use the Edit menu's 'Combine' command. A Combi
containing those devices is instantly created (Combinators can also be
'uncombined'). Devices can easily be added using the 'Create' menu or
contextual menu option. And finally, any device in the rack can simply be
dragged into the Combi. You can't drag one Combinator into another, though —
doing so simply adds all the modules from the one you're dragging to the other.
This is a nifty option, allowing you to combine, say, effects chains and layered
synths in a new Combinator. However, the new devices added in this way tend to
automatically link to any available inputs on the main Remix in your rack, if you

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Propellerhead Reason v3

have one, rather than the mixer in the target Combinator.

Each Combinator has a flexible audio routing system, consisting of two pairs of
stereo input jacks and two pairs of stereo outs. Devices in a Combi are routed to
a stereo pair labelled 'From Devices', and thence to the rack's main mixer from
the pair labelled 'Combi Output'. Thus you can see a mixer is needed in most
circumstances, to reduce the outputs from all the combined devices to a single
stereo stream, whether you're combining synths, samplers and drum machines
or the outs from a parallel effects setup.

There is one area in which the devices in a Combinator are still dealt with on an
individual basis: automation. Try to record a parameter change with your mouse
for a device in a Combinator patch and it won't work. The only parameters that
can be automated are the knobs, buttons and wheels of the Combinator's
controller panel (plus aftertouch, breath control, sustain pedal and expression
pedal). To automate the parameters of one of the devices, you have to assign
that device to its own sequencer track.

Combinator even has a collection of gate and CV connections: the four rotary
Controller knobs can be assigned a CV from elsewhere in the Reason rack for
programmable modulation, and sequencer control gate and CV inputs mean that
a Combinator can be played by a Matrix pattern sequencer. Finally, I note that
the rear panel of the Combinator Programmer is amusingly labelled 'TS8450
Touch Sensitive Display Unit. We wish, guys, we wish...

MClass Mastering Suite

The four-strong MClass suite offers a mastering EQ, stereo imager, compressor
and maximiser. Each is a full-sized device, with a full complement of controls.
And to make it even easier for you, Propellerhead have made a Mastering Suite
Combi available from the 'Create' menu; it has all four devices in it, and a
collection of specific factory patches to start you off (see above).

The MClass EQ is actually larger than the remaining three devices, to show off
its curve display, a larger version of that used by the original half-rack PEQ2. The
MClass EQ is worlds beyond that device, however, offering high and low shelving
bands, two parametric bands, plus a 30Hz low-cut switch. The range of the EQ is
quite impressive, with the low band operating from 30Hz. Though the high band
tops out at 12kHz, both mid bands have a range of 39Hz to 20kHz. I was slightly
disappointed to not be able to change EQ response by dragging the curve in the
display, but that's just a personal thing!

I can't be alone in playing with delays to increase the perceived width of mixes,
so the dedicated MClass Stereo Imager is most welcome. Not only does this
device control stereo width, but it operates on high and low frequency ranges
independently, with user control over the crossover frequency, and offers

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Propellerhead Reason v3

independent stereo outputs for each

range. Solo controls let you check out
the effect on each band.
The Micromix stereo six-channel mixer is a
small addition to the Reason rack, but is
It's great to see the MClass nonetheless welcome.
Compressor join the Reason family —
unlike the original COMP01 device, it offers the facilities of a real compressor,
such as standard or soft-knee operation (the more 'musical' compression), plus
side-chain access for ducking and frequency-conscious processing (such as de-
essing). The control set is very traditional, with knobs for Input Gain, Threshold,
Ratio (1:1 to Infinity:1), Attack (1 to 100ms) and Release (50 to 600ms), plus
Makeup Gain, but there are a couple of surprises — the Release parameter has
a programme-dependent option. If you enable this, the release time changes in
response to the length of incoming peaks. And the rear panel houses a CV
output which transmits a dynamic CV signal derived from the compressor's gain
reduction.

Lastly, the MClass Maximiser allows you to make your mixes as loud as possible
without clipping. There's a large, detailed peak/VU meter display and control over
input and output gain, with a three-way switched control over Attack and Release
(Release again has an adaptive programme-dependent option, for a more
natural-sounding result). A soft clipping control appears after the final gain stage,
and enabling the 4ms look-ahead option lets the Maximiser examine audio
before it's processed, and limit it if you wish.

Enhanced File Browser

New for v3.0 is a reworked file-browser system, letting you more easily navigate
your hard drive and any Reason-related files and Refills it might contain. The
redesign lets you search for patches by name or type, and the database folders of
earlier versions have been replaced by a sub-window called 'Locations' — in fact,
the database folders from earlier versions will be placed in the Locations list when
you upgrade. You can add any file location to this to make it easily accessible.
One great new feature is in-browser auditioning of both samples and patches: as
long as your master keyboard is routed to the device patches are being selected
for, they can be played in the browser without loading first. A particularly welcome
file-browsing option, available from the Create or contextual menu, is 'Create
Device by Browsing Patches': you just pick the patch you want, and the device
that plays it loads immediately, ready to go. As an extension of that idea, it's also
possible to open the browser from one device and look for patches for any other
device; select another device's patch, and it will automatically replace the original
device.

Bits & Pieces

The MClass suite is fab — I've already imported a couple of mixes into NN19,
just to process them with these effects — but there is still more. One further

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Propellerhead Reason v3

device has been added to the rack: the Micromix stereo line mixer. While it's
ideal as a sub-mixer inside Combis, it has, of course, a life elsewhere in the rack.
Many of us have been using v2.5's Spider audio device as a simple audio mixer/
combiner, and will welcome the option to add a bit of panning and an effect send
for more demanding applications that don't require the facilities or CPU overhead
of the full-blown Remix module.

Each of Micromix's six channels is

equipped with a stereo input jack pair,
mute and solo buttons, and level, pan
and a single pre/post fader stereo aux
send (to suit the stereo input of the
RV7000 advanced reverb device).
There's even space for a miniature
LED bar-graph level meter, scribble
strip and (rear-panel) CV control for the
pan parameter.

Other changes are more subtle. A new

Preferences page labelled 'Control
Surfaces and Keyboards' helps you
select the hardware controllers you The new Preferences window, where MIDI
intend to use, and the software control surfaces and your master keyboard
communicates instantly with many are selected and assigned.
current examples, complete with ready-
made mappings of controllers to Reason parameters. Multiple controllers can be
used if you need to control more than one device at once. There's even a 'MIDI
Out' element to the MIDI routing for controllers, but this link is solely for
controllers that require bi-directional MIDI connections, like moving-fader
surfaces — there's no way to send MIDI note data outside Reason this way.

Another MIDI enhancement affects the main linear sequencer: it's possible to
record automation to several tracks at once, though you can still only record one
note-based MIDI performance at a time. The sequencer itself benefits from a bit
of tidying up: it now has sensible Mute and Solo buttons, a much clearer 'MIDI In'
indicator column, and record-enable switching. There are no major new editing
options, though, and neither is there a tempo track yet, nor a score display.

The two supplied Refills (Orkester and the Factory Sound Bank) have been
enhanced and enlarged — they both now total 1.22GB in size, and take
advantage of new features such as the Combinator and the MClass processors.
Some of the stacks and layers sound fabulous! Refill installation now occurs as
part of the overall installation, too. The fab Electro Mechanical Refill added some
months ago is not part of the Reason v3 package, but is still a free download for
registered users.

At the other end of the process from installation — bouncing a finished mix to
disk — dithering has been added to the audio export options, and sample load
times for Redrum, NNXT and NN19 have been sped up.

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Propellerhead Reason v3

Conclusions

As ever, I had a ball with Reason. It's just so much fun to work with, and the new
features enhance that feeling. You'll never want to be without Combinator again,
and the mastering suite really is in a (n M) Class of its own. Overall, the software
remains the one to beat just for sheer facilities and user-friendliness.

To temper this slightly, I'm not so delighted about the way v3 will only run on up-
to-date operating systems and exhibits sluggishness on older computers, even
with simple Songs. Perhaps Propellerhead are readying their software for some
as-yet unrevealed features.

Furthermore, my own bias is towards sound sources, and there are no new ones
in this update (although admittedly Combinator offers a powerful way of applying
what's already there). And Reason remains a closed zone to plug-ins, so you
can't add any yourself. I also have a soft spot for arpeggiators, and no matter
how existing devices can be pulled together to create arpeggiator-like effects, it's
not the real thing! Finally, there will always be users who wish audio recording
was available, and that goes for me too, if only to see how Propellerhead's
engineers would present the feature.

But as much as I try to play devil's advocate with Reason, it remains the music
software I use most. Whenever I feel nostalgic for all the old synths and effects
units my studio no longer has to accommodate, I fire up Reason, and I'm in the
future. Now.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Propellerhead%20Reason%20v3.htm (8 of 8)9/27/2005 9:23:03 PM

 RME ADI2 & ADI4 DD

In this article:
Case & Connections
RME ADI2 & ADI4 DD
RME ADI4 DD: AES-ADAT A-D/D-A Converter & Digital Format Converter
Digital Format Converter Published in SOS May 2005
Front-panel Controls Print article : Close window
Technology & Specifications
Reviews : A-D/D-A Converter
Audible Benefits

RME ADI2 & ADI4 DD

£445/£455
pros Two new half-rack boxes offer considerable
Flexible analogue operating operational flexibility and pristine sound in both the
levels and excellent clocking
performance for the ADI2. analogue and digital domains.
AES, S/PDIF, and ADAT
formats supported.
24-bit capability and Hugh Robjohns
elevated sample rates.
cons I have reviewed several RME
In-line mains PSU modules. hardware products in the past, and
summary have always been impressed with the
The ADI2 is a compact but
high audio quality they seem able to
high-performance two- provide despite their very moderate
channel A-D and D-A pricing. Recent technological
converter with useful I/O developments have made this quality-
flexibility. Improved clock price ratio even more favourable, with Photos: Mark Ewing
circuitry helps bestow class-
leading technical performance
some impressive advances in word-
which belies its modest clock stability which are claimed to put the latest-generation of RME converters in
pricing. The ADI4 DD offers Apogee territory in this regard — an intriguing thought!
flexible format-conversion and
routing options, and could be
the ideal solution for anyone One of the latest RME offerings is the ADI2, a relatively straightforward two-
facing format incompatibilities. channel A-D/D-A converter housed in a half-width 1U rack case. In essence, it is
information a development of the previous ADI1 which I reviewed way back in SOS August
1999, sharing a similar (but updated and extended) feature set and mechanical
ADI2, £445; ADI4 DD,
£455. Prices include VAT. footprint. Technological advances have been incorporated to support 24-bit
Synthax Audio UK +44 conversion and all standard sample rates up to 192kHz, with significantly
(0)1664 410600. improved jitter rejection.
+44 (0)1664 410999.
Click here to email
www.rme-audio.com
Case & Connections

The case of the ADI2 is constructed from three folded-steel panels to provide a
very strong and fully screened enclosure for the internal circuitry. The front panel
is made from aluminium sections, using blue-painted panel areas and clear silver

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RME ADI2 & ADI4 DD

legends to highlight the various controls. The rear panel is painted black with
large white legends, and the connections are all very obvious. The analogue line-
level inputs to the A-D are catered for with Neutrik combi jack/XLR sockets,
accepting either XLR or TRS connectors. The input circuitry employs clever
'servo amplifiers' that can accommodate balanced or unbalanced signals equally
well, automatically compensating for the 6dB lower signal level that can
otherwise occur when connecting an unbalanced signal to an electronically
balanced input.

The unit can be configured from the front panel for one of three operating levels,
where 0dBFS (digital peak) equates to +19dBu, +13dBu, or +4dBu. These are
designed for compatibility with professional high-level, standard +4dBu, and semi-
professional -10dBV equipment respectively. The first two options effectively just
change the available headroom from 15dB to 9dB (relative to +4dBu), while the
third reduces the sensitivity to accommodate the lower semi-pro operating levels.
In this case, the stated +4dBu peak level equates to +2dBV, thus offering a
nominal 12dB of headroom above -10dBV.

The A-D's digital output is presented on both a coaxial phono connector and a
Toslink optical port, although you can choose different data formats — a front-
panel selector enables the output format to be switched between Consumer (S/
PDIF), Pro (AES), and ADAT. The phono connector is used to output S/PDIF or
AES data: changing the format from Consumer to Pro increases the signal level
(from 0.8V to 2.3V peak-to-peak) and changes the status flags to comply with the
relevant standards. The phono socket is electrically isolated from the chassis and
connected via a transformer so that it is fully floating. Thus, a properly balanced
AES output can be obtained simply by wiring the tip and sleeve of a phono plug
to pins two and three of an XLR plug respectively. Valid audio data appears to be
present at this output regardless of whether the optical socket is configured for
Pro/Consumer or ADAT operation. All sample rates are catered for up to 192kHz,
using the single-wire double- or quad-speed transmission method which is now
more or less the standard.

The optical output can be switched

between S/PDIF or ADAT modes from
the front panel, but at standard sample
rates only channels one and two carry
the data, the other six channels being
mute. However, at the doubled sample
rates the widely supported SMux data
format is used, so that the first two
channel pairs are used to convey the A-
D's stereo signal. Surprisingly, no audio output is provided at all at the quad
rates, although a standard-rate embedded word clock is still conveyed in case
the ADAT signal is being used for clocking purposes, which is a thoughtful
feature.

Moving over to the D-A side of things, digital inputs are catered for once again
with a phono socket and a Toslink optical input. The data format arriving at each

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RME ADI2 & ADI4 DD

connector is detected and decoded automatically, but a front-panel selection

determines which input is used to feed the D-A (and act as external clock
reference, when appropriate). Again, the phono socket is electrically isolated
from the chassis and connected via a transformer so that balanced AES signals
can be accommodated with a suitable converter lead wired in the same way as
described earlier for the AES output.

The line-level analogue outputs are provided on both XLRs and TRS sockets,
featuring the same kind of servo amplifier idea so that the correct signal level is
maintained regardless of whether the output is connected to balanced or
unbalanced equipment. As with the input-sensitivity selection, a front-panel
control allows the output level to be switched between the same three settings,
giving peak signal levels of +19dBu, +13dBu, or +4dBu (+2dBV).

The ADI2, like its predecessor, is too compact to allow a mains power supply to
be incorporated, so a coaxial power connector is present on the rear panel,
complete with a metal hoop through which the power lead can be threaded to
prevent it falling out if accidentally tugged. Like many other RME products, this
unit is remarkably flexible in its powering options. It can accept DC supplies
between 8V and 28V DC (of either polarity), or low-voltage AC supplies between
8V and 20V AC. So battery powering from a range of sources is perfectly viable.
A compact third-party mains power unit is also included in the package, and this
can accept mains voltages between 100V and 240V AC at 50Hz or 60Hz, to
produce a 12V DC output suitable for powering the ADI2.

RME ADI4 DD: AES-ADAT Digital Format Converter

The RME ADI4 DD has been designed to translate between arguably the two
most common digital interface formats, AES and ADAT; and to do this in both
directions with up to 24-bit word lengths and sample rates up to 96kHz. Much of
the functionality can be assumed from a glimpse of the rear panel, although there
are several hidden features that only become apparent after reading the
comprehensive manual (supplied as a PDF file on a CD-ROM).
So, first let's look at the rear panel. Two XLRs provide a pair of AES (or S/PDIF)
channels in and out, and these are supplemented with a 25-pin D-Sub connector
which duplicates that first pair of channels and adds three more pairs. (In other
words, there are eight channels in and out altogether.) Each AES input and output
is transformer coupled and can operate in single- or dual-wire modes up to 96kHz.
There are numerous wiring formats for AES signals on D-Sub connectors —
Yamaha, Tascam, Euphonix, and Genex are just four that come immediately to
mind. However, the RME designers have been remarkably helpful by allowing the
user to re-configure the ADI4 DD to operate correctly with any of the first three
listed above. Inside the case a ribbon cable can be plugged into one of three
sockets, re-configuring the D-Sub to suit the Tascam, Yamaha, or Euphonix
house standards, respectively. For this review, I re-configured the unit from the
default Tascam setting to the Yamaha convention, to interface directly with my
equipment.
Above the D-Sub connector, a BNC socket accepts an external word clock,

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RME ADI2 & ADI4 DD

complete with a 75(omega) termination switch and LED indicator. A word-clock

output BNC is also provided. Power is connected through a coaxial socket with
similar AC/DC voltage flexibility as provided on the ADI2, and an in-line switched-
mode mains PSU is also provided. Finally, the first of two pairs of Toslink optical
connectors supply dedicated ADAT inputs and outputs, with the second pair being
switchable between S/PDIF and ADAT formats.
There are only four buttons on the front panel, the first of which selects the source
for the first pair of AES channels, the options being the XLR, the first two channels
of the D-Sub, or the S/PDIF optical connector. The next panel section carries an
array of LEDs indicating clock status and audio data — each AES input has its
own pair of LEDs, and two more pairs cater for the two optical inputs. Another
LED illuminates when double-rate AES inputs are detected.
Next up are buttons to select the clock source (external word clock, ADAT, or
AES) and to convert SMux ADAT inputs to double-rate AES outputs. Finally, the
AES outputs can be configured to operate with professional or consumer (S/PDIF)
voltage levels and data flags, and the second Toslink socket switched to provide
S/PDIF data.
The 'hidden' features provided by the ADI4 DD include the ability to convert four
channels between 'double wide' and 'double speed' data formats (in both
directions); the facility to split incoming eight-channel AES or ADAT data to both
eight-channel ADAT outputs; and the option to function as a 'two input to eight
output' AES splitter. This is a very well engineered and flexible little box that may
well provide the ideal solution to anyone facing format incompatibilities.

Front-panel Controls

Having completed the geek's tour of the connectors, the front panel needs little
explanation, as most of the facilities have already been mentioned. There are
only five buttons, a rotary control, and a quarter-inch stereo headphone socket to
master — although the 31 LEDs make it look a little more complex than it really
is! Starting at the left-hand side, the first button cycles through the three input-
level options, each with a green LED to indicate the current setting. Next up is a
vertical bar graph meter with two columns of six LEDs, showing the A-D's digital
output level. The bottom four lights are green, followed by a yellow and a red
LED, and the ensemble is scaled -60, -30, -12, -6, -3 and 'Over' (although,
strictly, this is not an 'over' warning at all since it can only indicate digital signals
reaching 0dBFS.) The meter is actually a lot more informative than this
description implies, as each LED has several intensities which help to bridge the
gaps between adjacent LED levels. AT the bottom end, signals as low as -
76dBFS can be identified, while, at the top, peaks of -2dBFS or -1dBFS are
easily distinguished. Only signal peaks that actually hit 0dBFS cause the peak-
hold function to activate, maintaining the red LED's illumination for one second to
attract attention.

The central section of the front panel is concerned with clock-rate and I/O
selections. The first button cycles through the clocking functions, with six LEDs to
indicate the status. There are three internal crystal-based sample-rate options —
32kHz, 44.1kHz, and 48kHz — followed by an external sample rate derived from
the selected digital input. Further pushes of the button cycle through these four

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RME ADI2 & ADI4 DD

options again, but with a 'x2' multiplier for the internal rates, and then again with
a 'x4' multiplier. So getting to 192kHz is a tedious process, but simple enough
and with clear indications along the way.

The next two buttons configure the digital I/O, the first selecting the coaxial or
optical digital input, and the second configuring the digital outputs for ADAT, Pro,
or Consumer formats, as described earlier. Once again, LEDs indicate which
input connectors and output modes have been chosen. The last button selects
one of three analogue output levels, again with LEDs to show the current status.

The final control is a rotary volume knob for the built-in headphone monitor which
auditions the output from the D-A converter. The quarter-inch socket sits
alongside the control for easy access, and there is sufficient 'oompf' from the
amplifier to enable this output to be used to drive unbalanced line inputs, should
that ever become necessary.

Technology & Specifications

The earlier ADI1 employed 20-bit Crystal and AKM converters and operated at
only the standard sample rates — although this was the state of the art for
'budget' converters back then. Such limitations wouldn't be countenanced these
days, of course, and it is interesting to see just how far RME have evolved in the
intervening six years. The current unit uses the latest high-resolution AKM chips:
the 5385 A-D and 4395 D-A converters (the latter with built-in automatic de-
emphasis facilities in the unlikely event that they are needed).

The analogue side of things appears to

be handled by the ubiquitous 4580 op
amps, in the same circuit topology as
used in several other RME products
including the previous ADI1 and the
flagship ADI8DS (reviewed back in
SOS September 2003). All of the unit's
control functions, as well as the audio
data manipulation and formatting, are
handled by a Xilinx Spartan IIE FPGA
(Field Programmable Gata Array). Three different pin-out configurations for the
ADI4 DD's digital D-Sub connector are
Everything has been put together very supported, as used by Tascam, Yamaha,
nicely, on one fairly dense printed and Euphonix. You can change the format by
circuit board using mainly surface- re-plugging an internal ribbon cable.
mount components. The front-panel
LEDs and buttons are mounted on a second vertical card, linked via a ribbon
cable. All the rear-panel connectors are soldered directly to the main board, but
are fixed to the rear panel to help minimise any mechanical strain being
conveyed to the PCB connections.

The published specs make impressive reading, with an A-D signal-to-noise ratio

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RME ADI2 & ADI4 DD

of 113dBA and distortion below 0.0003 percent. The D-A boasts a signal-to-noise
figure of 119dBA and distortion below 0.0007 percent. Compared to the ADI1,
the signal-to-noise ratios are nearly four times better, and the distortion has
improved by an order of magnitude! Perhaps even more impressive are the jitter
figures, which are roughly half that of any comparable product — less than 0.8ns
on the internal crystal and only 1ns when slaved to an external clock reference,
with around 30dB of jitter suppression from wobbly external sources! Apparently
the Steady Clock technology that provides these impressive results was
developed originally to recover stable clocks from the multi-channel version of
the AES interface — MADI — which typically suffers a monumental 80ns of jitter!

Audible Benefits

The ADI2 is a very straightforward and compact unit, the simplicity (and
comparatively low UK price) of which belies its superb quality. This really is a star
performer which compared favourably against my trusty Apogee PSX100, as well
as the rather nice converters in the Drawmer DC2476 mastering processor.
Noise and distortion are completely inaudible, even when working with generous
headroom; the audio spectrum is open and airy (especially at the double and
quadruple sample rates); and the stereo imaging is three-dimensional and
completely stable — always the sign of a good clock — with wide and deep
sound stages. The only relevant negative comment I could raise is that the low
bass seems a little thin when compared to some more expensive converters —
but you have to move a very long way up the price scale before the RME's
inevitable shortcomings start to become noticeable at all.

Leaving the sonic merits to one side, the ADI2 has some practical limitations
compared to many state-of-the-art converters. For example, the A-D and D-A
stages can't be used as completely separate units, running at different clock
rates; the output cannot be dithered down to lower word lengths to suit CD-R and
DAT recorders; there is no external word-clock input (separate from that
embedded in the digital inputs); and there is no 'soft-limiting' facility to minimise
peak overloads when working with minimal headroom. However, these features
are not important — let alone necessary — to the vast majority of semi-pro users
in our modern 24-bit world.

On the plus side, the I/O flexibility in terms of connections, formats, and operating
levels is excellent, the simplicity of configuration a joy, the built-in headphone
monitoring a useful facility, and the cost-performance ratio astounding. I'm not a
fan of in-line or wall-wart power supplies, but given the sonic and fiscal benefits
of this unit, I feel pleasantly disposed to overlook such a minor issue!

Published in SOS May 2005

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RME ADI2 & ADI4 DD

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/RME%20ADI2%20&%20ADI4%20DD.htm (7 of 7)9/27/2005 9:23:08 PM

 Roland Fantom Xa

In this article:
Previous SOS Fantom
Roland Fantom Xa
Reviews Workstation Synth
The Synthesizer Published in SOS May 2005
Keyboard Fantoms Print article : Close window
Compared
Reviews : Keyboard workstation
The Sampler
The Sequencer
In Use
The Fantom Editor
Conclusions At £1099, the Xa is the most affordable keyboard in
the Fantom range. But inevitably, features have been
Roland Fantom Xa £1099
removed to make it such a bargain. Have Roland
pros
thrown out the works from the workstation?
The cheapest Fantom
keyboard to date.
Still an extremely powerful
synth.
Gordon Reid
Built-in sampler, expandable
to 516MB of RAM. The Fantom Xa is the eighth model to
appear in Roland's Fantom range of
Many additional facilities,
such as arpeggiators, chord workstations and is, with the exception
generators, rhythm patterns,
of the Fantom XR rack module, also
and more.
the cheapest. To achieve this, many of
Included Editor makes it a
pleasure to program.
the facilities found on the other
Sounds superb.
Fantoms have been removed. In
principle, this makes it immediately
cons attractive, but in some ways it's clearly Photos: Mike Cameron
Lower-quality keyboard than less powerful than its forebears.
any other Fantom.
Much smaller display than
on previous Fantoms. Most obviously, the 320x240-pixel colour screen of the X-series has gone, to be
Only one SRX expansion replaced by a 240x64-pixel greyscale screen. Ironically, given that this is the first
slot. Fantom keyboard to sport a small LCD, the soft-keys and the on-screen
Sampler still has limitations commands now line up perfectly. There's even a bit of panel artwork to ensure
of previous models. that you notice this (on all previous Fantoms, the buttons and commands were
Non-touch-sensitive sample offset from one another). Reducing the screen resolution by 80 percent is a
pads. significant change, but, thanks to a cleverly implemented set of zoom pages for
Outdated effects structure. graphical displays and grid representations for programming, I was generally
External power supply. able to obtain the results I wanted on the Xa itself.
summary
Like its siblings, the Fantom Gone too are the likeable semi-weighted keyboards of the Fantom S, X6 and X7.
Xa is a well-specified
Roland have removed the weights from underneath the keys, which makes the
combination of an expanded
XV synth engine, an MRC Pro keyboard feel springy and unpleasant, and they've also removed pressure
sequencer, and 'groove'-style sensitivity. Peculiarly, although the keyboard can't generate aftertouch, the
sampling, with oodles of keyboard's real-time controllers can be assigned to do so, and the Xa will both

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Roland Fantom Xa

extras such as user- transmit and respond to channel aftertouch. Even more peculiarly (and perhaps
programmable arpeggiation, as a leftover trait from the full X-series workstations), the sequencer can also
rhythm patterns, and an
included software Editor.
generate polyphonic aftertouch. This implies that the Xa will respond to poly-
However, it lacks several of aftertouch generated internally or received over MIDI, but I can't find any
the niceties of previous modulation destination that responds to it.
Fantoms, the removal of
which has made it both
cheaper but also less usable In short, Roland have made the Xa unsuitable for piano and, to a lesser extent,
for some players. I suspect orchestral playing. Given that the Xa still costs over a thousand pounds, this is
that it will be seen as an good rather disappointing, unless you view a keyboard as a set of buttons for triggering
price/performance arpeggios and patterns. But perhaps this is in keeping with some of the more
compromise for cash-
strapped, dance-oriented
Groovebox-like aspects of the Xa. For example, some of the X-series' PCMs
musicians and producers, and appear to have been replaced with drum and percussion samples aimed
that it will be a success. squarely at dance music production, and then there are the illuminated pads on
information the right-hand side of the panel, introduced on the Fantom S. The Xa has 10 of
these compared to the S- and X-series' 16, but they can still operate as drum
£1099 including VAT.
pads, as well as triggers for other features such as Real-Time Phrase
Roland UK +44 (0)1792
515020. Sequences, rhythm patterns, and samples. This all seems to be... well, Groovy.
+44 (0)1792 799644. In the past, I have complimented Roland on focusing products at their target
www.roland.co.uk markets, and in this light, the more 'dancy' sounds make sense — you just need
to be aware that if you're after an orchestral or rock & roll synth, the Xa is
www.roland.com
perhaps less suitable than previous Fantoms.

But groove instruments still need to be capable of expression, and the Xa's 10
pads not only lack the poly-aftertouch of the previous incarnations, they are not
capable of outputting any aftertouch. Nor are they velocity sensitive; you can set
up fixed MIDI velocities for each, or a common velocity for them all. In fact, the
pads have been reduced to the role of mere switches, not the sensitive
performance controls they were. Further casualties of the transition to the
Fantom Xa include expandability and the S and X-series' internal power supplies.
The Xa has just a single SRX card expansion slot, and an external 9V AC/DC
converter (boo!). Despite all this, though, there's still much to be praised...

Previous SOS Fantom Reviews

Much has already been written in Sound On Sound about the Fantom series, so I
suggest you take a look back at Nick Magnus's review of the Fantom S and S88
(in Sound On Sound October 2003: see www.soundonsound.com/sos/oct03/
articles/rolandfantoms.htm), and Paul Nagle's review of the 'X' series (in Sound
On Sound September 2004: see www.soundonsound.com/sos/sep04/articles/
rolandfantomx.htm) and the original FA76 (SOS February 2002: see www.
soundonsound.com/sos/feb02/articles/fantom0202.asp). In many ways, the Xa is
identical to these, so if I appear to have skipped over something in these pages,
the chances are that it's already been covered in Nick's and Paul's previous
articles.

The Synthesizer

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Roland Fantom Xa

As far as the sound engine is concerned, the Xa retains the maximum polyphony
of 128 voices and the 16-part multitimbrality of the rest of the 'X' series, but the
ROM has shrunk back to the 'S' specification, with 1228 PCM waveforms rather
than the full complement of 1480. Lest you think that this is a problem... it isn't.
The Fantom S was and remains a phenomenally powerful synth, and I defy
anybody to plumb fully the depths of the Xa's ROM, its 10 types of patch
structure, its cross-modulation (FM) capabilities, the boosters, the ring
modulators, the hugely flexible modulators and modulation matrix, and the many,
many ways that patches can be modified with up to six (reasonably) assignable
effects units.

The number of programmable

memories is the same as on other Xs,
with 256 patch memories, 32
programmable rhythm sets and 64
Performance memories in addition to
the usual bucketful of presets. But the
Xa's bucket is smaller than that of the
other Xs — there are 768 preset
patches rather than 1024, and 36
Unlike on all previous Fantoms, the 'soft'
rhythm sets rather than 40. Happily, keys beneath the LCD line up with the on-
the number of effects algorithms and screen functions they access! It's a shame
the MFX architecture remain that this also happens to be the Fantom with
unchanged from the rest of the X the smallest display, but you can't have
everything...
series. This means that there are 78
algorithms including COSM
(Composite Object Sound Modelling) and RSS (Roland Sound Space) effects.
Unfortunately, it also means that Roland's aging MFX/chorus/reverb system is
retained. Flexible though this is, it's dropping further and further behind the
flexible Insert and Master effects structures being used on other companies'
workstations, and it's time that it was replaced.

Like the 'S' and 'X' series, the Xa also has a three-band compressor/limiter
placed across its main outputs, which Roland call the 'Mastering Effect'. Three-
band compressors like these are becoming ubiquitous on hard disk editors as
well as keyboard workstations, but I think that unless they're used with great care
and subtlety, they can do more harm than good — and they're certainly no match
for the handiwork of a good mastering engineer. Still, you don't have to use them!

Also unchanged from the Fantom S spec, the Xa offers the full complement of
rhythm patterns and memories, as well as preset and user-programmable
arpeggios. Nevertheless, some functions have been lost, and not just because
the hardware has changed. For example, the 'Live Setting' section of the full X-
series instruments is gone (which is strange, given the performance orientation of
the Xa) as is the ability to use the pads to jump to favourite edit screens, and the
voice monitor that showed how the polyphony was used. But when it comes to
the crunch, the patch, performance, and effects structures of the Xa are all but
identical to those of a full 'X', so we need say no more about them here.

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Roland Fantom Xa

Keyboard Fantoms Compared

FA76 S & S88 X SERIES XA
KEYBOARD — 61-note semi- 61-note semi- 61-note
weighted (S) weighted (X6) unweighted
76-note semi- — 76-note semi- —
weighted weighted (X7)
— 88-note 88-note —
hammer-action hammer-action
(S88) (X8)
VELOCITY Yes Yes Yes Yes
SENSITIVE?
PRESSURE Yes Yes Yes No
SENSITIVE?
DISPLAY 320x240 mono 320x240 mono 320x240 240x64 mono
LCD LCD colour LCD LCD
PADS None 16 16 10
MAXIMUM 64 64 128 128
POLYPHONY
MULTITIMBRALITY 16 parts 16 parts 16 parts 16 parts
ROM SIZE 64MB 64MB 128MB 64MB
equivalent equivalent equivalent equivalent
NUMBER OF 1083 1228 1480 1228
WAVEFORMS
EXPANSION (SR JV 1 slot None None None
TYPE)
EXPANSION (SRX 2 slots 4 slots 4 slots (6 slots 1 slot
TYPE) in XR)
PRESET MEMORY 640 Patches 640 Patches 1024 Patches 768 Patches
256 GM2 256 GM2 256 GM2 256 GM2
Patches Patches Patches Patches
16 rhythm sets 32 rhythm sets 40 rhythm sets 36 rhythm sets
9 GM2 rhythm 9 GM2 rhythm 9 GM2 rhythm 9 GM2 sets
sets sets sets
64 64 64 64
Performances Performances Performances Performances
— 8 piano — —
Patches (S88
only)
USER MEMORY 128 Patches 256 Patches 256 Patches 256 Patches
16 Rhythm 32 Rhythm 32 Rhythm 32 Rhythm
Sets Sets Sets Sets
64 64 64 64
Performances Performances Performances Performances
CARD MEMORY — Smart Media PC card slot PC card slot
slot (up to (up to 1GB) (up to 1GB)
128MB)

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Roland Fantom Xa

EFFECTS 3 MFX slots, 3 MFX slots: 3 MFX slots: 3 MFX slots:

90 types 77 types (78 78 types 78 types
on S88)
Global chorus: Global chorus: Global chorus: Global chorus:
2 types 3 types 3 types 3 types
Global reverb: Global reverb: Global reverb: Global reverb:
4 types 5 types 5 types 5 types
Global EQ Input effects: 6 Input effects: 6 Input effects: 6
types types types
— Mastering Mastering Mastering
effects: effects: effects:
compressor compressor compressor
SAMPLING — 16-bit linear, 16-bit linear, 16-bit linear,
44.1kHz 44.1kHz 44.1kHz
— File type: WAV File type: WAV File type: WAV
& AIFF & AIFF & AIFF
SAMPLE MEMORY — 32MB 32MB 4MB standard
standard (up to standard (up to (up to 516MB)
288MB) 544MB)
SEQUENCER Tracks: 16 Tracks: 16 Tracks: 16 Tracks: 16
100 patterns 100 patterns 100 patterns 100 patterns
Resolution: Resolution: Resolution: Resolution:
480ppqn 480ppqn 480ppqn 480ppqn
Capacity: Capacity: Capacity: Capacity:
120,000 notes 120,000 notes 400,000 notes 400,000 notes
ARPEGGIATOR 88 preset, 0 128 preset, 128 preset, 128 preset,
user 128 user 128 user 128 user
RHYTHM PATTERNS 50 preset, 0 256 preset, 256 preset, 256 preset,
user 256 user 256 user 256 user
CHORD MEMORY — 64 preset, 64 64 preset, 64 64 preset, 64
user user user

The Sampler

Apart from the loss of the user interface supported by the large screens of
previous models, and the lack of the other Fantom Xs' digital inputs, sampling on
the Xa appears to be largely unchanged from the 'S' and 'X' series. The means of
accessing it is different (there's no Input Setting button, either), but the basis is
the same. You can sample, import WAVs and AIFFs, truncate, loop, normalise,
amplify sections, stretch, chop and recombine to your heart's content. I
particularly like the Auto Divide function, which breaks samples up at moments of
near silence and assigns the next sample number to the ensuing audio. This is
an elegant way to sample sources such as drum kits, but I also find it excellent
for resampling phrases played on the Xa itself. I am less enamoured of the
sampling input effects; I'd rather apply effects after sampling, and thereby use
the more sophisticated MFX algorithms. It's nice to see skip-back sampling here,
though — once you get used to this, you'll never want to work any other way.
Introduced on the Fantom S and retained on the Xa, this feature simply means
that the sampler is continually recording whatever you present to its input. If you

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Roland Fantom Xa

like something that you hear, you can save it even though you haven't told the
sampler to start sampling.

The Xa's sampler lacks a couple of features that I really miss. Alt (bi-directional)
looping is one; I use this extensively on my S700-series samplers. It would be
simple to implement this on a processor as powerful as the one in the Xa, so why
haven't Roland done it? Secondly, the Xa is still unable to import existing
multisamples correctly. Overcoming this by converting and loading individual
samples, re-looping and then re-allocating them to the keyboard is too long-
winded for words. Just as I was finishing this article, Roland resolved this
problem for the rest of the Fantom X series, but the fix does not work on the Xa
— see the box on page 60.

The Sequencer

The Xa's sequencer appears to be identical to that of the other Fantom Xs, with
16 channels, each of which can hold data across all 16 MIDI channels. Song
data can be directed to the internal sound engine, to external MIDI products, or to
both simultaneously, and live 'mutes' allow you to inspect what's doing what to
what. All the expected editing and MIDI data manipulation tools are included, but
I won't list them all; if you know the way that Roland write sequencing software,
you'll know what's here. Nevertheless, one facility deserves special mention; the
provision of 71 templates for quantising and 'shuffling' selected data in various
musical styles. That's an excellent touch.

You can only edit songs in a

'temporary' area that's lost on power-
down, so you have to save your work
at the end of each session. Four
formats are supported, these being
SVQ (the format used by all of
Roland's recent MRC Pro sequencers),
the older standard MRC format, and
Standard MIDI File (format 0 and
format 1). If you save a song in SVQ
format, the resulting files can include There's no digital I/O on the Fantom Xa, but
the performance/setup data as well as otherwise, there's a fine complement of I/O
and facilities, including Expression and
any associated samples.
Sustain pedal jacks, five-pin MIDI In, Out,
and Thru, a Headphones socket, stereo
The sequencer is powerful, but I feel master analogue inputs with an associated
input level control, and stereo master
that this is the area that suffers most
analogue outputs on jacks with two further
from the loss of the larger 'S' and 'X' independent outs. Out of shot, to the left of
screens. Real-time recording is a what you can see in this picture, are the
doddle, and the step-time recording power switch and DC In connection to the
and editing is well implemented given external PSU, the PC card slot, and the USB
socket for connecting a computer and/or the
the limited amount of screen space
software editor.
available, but much of the operating

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Roland Fantom Xa

system is text-based. This won't be a

problem if you're used to working with older Roland sequencers, or other
keyboard workstations, but if you have always used computer-based
sequencers, and you're accustomed to large, graphical representations of
controllers and other data, you may find it difficult to come to terms with the Xa's
operation.

If you're not comfortable using the sequencer in a linear mode, you may enjoy
the provision of short sequences called 'phrases' that you allocate to pattern
memories and then trigger to build tracks and songs. I'm not a fan of phrase
sequencers on 'pro' instruments because the philosophy feels too similar to that
of auto-accompaniment keyboards. But if you're tempted by this way of working,
the Xa offers everything that you need. You can trigger and loop patterns by
pressing a single note or pad, and combine up to eight of them simultaneously to
create music that you couldn't play 'live' with just two hands. You can also
include the patterns within longer sequences.

In Use

Given the clear focus of the Xa on dance music and Roland's proven expertise in
this area, it's not surprising that it excels in creating and manipulating all manner
of beats and grooves. The step LFOs, arpeggiated patterns and synchronised
rhythms make the Xa an instant groovebox — the Xa's sales blurb even speaks
of 'instant dancefloor inspiration'. Of course, the danger with these kinds of
sounds is that they can quickly become last summer's craze, unless you're
prepared to delve deeper than the presets. On the other hand, the list of
percussion samples, rhythm sounds and patterns listed at the back of the manual
is huge, so there's plenty of scope for you to plough your own furrow.

Of course, there is also a superb selection of lead sounds, pads, guitars, organs,
and orchestral instruments, so the Xa is capable of acting as a first-class
expander. Unfortunately, I found that layering my patches in a Performance led
to a great deal of note stealing as the polyphony dropped. This is because, while
the Fantom has a maximum polyphony of 128 notes, this is obtainable only if you
restrict every sound to a single, monophonic PCM. If you use the full XV-engine
capabilities in patch mode, the polyphony drops to just 32 notes. If you then start
to layer patches, the polyphony can drop to 16, 10, eight or even fewer notes.
So, while you may have 128 individual tones available, it is how you use them
that will define the polyphony. Of course, this limitation is not unique to Roland's
workstations and, to their credit, they explain it in the manual. Furthermore, to
minimise the unpleasant side-effects of note-stealing, they have introduced an
innovative note priority mode called 'Loudest'. With this selected, the quietest of
the current notes is silenced, rather than the oldest, or the highest, or the lowest.
This is an excellent innovation.

Whichever types of sounds you use, the Xa provides a great deal of control, both
over the patches themselves and the effects that you apply to them. You can

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Roland Fantom Xa

assign MIDI CCs to patch parameters

and effects, apply MIDI aftertouch using
the knobs or a control pedal (not the
same thing, I know), and assign the
real-time knobs and the D-Beam to a
huge variety of destinations, either
directly or via a control matrix. If you're
prepared to take the time and trouble, you can create some amazing textures by
modulating the initial patch and effects combination.

I found Solo Synth less inspiring. It's a dual-oscillator, monophonic sound

generator first introduced on the Fantom S that shares its performance
philosophy with that of a single-aerial Theremin. In other words, move your hand
towards the D-Beam source and the pitch rises; move it away and the pitch
drops. But although you can resample its output, you can't store patches, nor can
you sequence them.

I'm not usually a fan of polyphonic arpeggiators or auto-chord generators, either.

But the chord generator can strum up, down, or up and down, and act as the
note source for the arpeggiator, both of which can be synchronised with the
rhythm unit. With suitable sounds, this can sound excellent, and if it's a bit too
sedate for you, you can add a bit of chaos using the Solo Synth as a sort of 21st-
century Theremin. I expected not to like this, but I did. When used to the full, the
chord generator, arpeggiator and rhythm patterns provide miniature sequences
that you can trigger, transpose and combine into complete musical backings.
Neat.

On the other hand, if you see a synth primarily as a keyboard instrument, the fact
that the Fantom Xa offers many other performance options does not make up for
the low-quality keyboard; this makes the Xa far from ideal as a conventional
performance synth. This brings me to the subject of the Xa's piano sounds. The
grand piano sound of the S88 is not included, but given the price point of the Xa
and where it seems to be aimed, that's not surprising. However, that's no reason
to make the piano sounds that are included less than overwhelming. To be fair to
Roland, I am not greatly enamoured of the pianos provided by anybody's
workstations, but I find the Xa's to be particularly lifeless. The best have only four
velocity layers, and most of them have only two, quiet and loud. What's more, the
velocity splits are obvious, with no attempt to crossfade from one to the next. If
you require a realistic piano sound and action, this is not the instrument for you.

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Roland Fantom Xa

The Fantom Editor

If you want to delve deeper than the preset sounds and groove-style facilities,
you're going to need to dig into the Xa's editing system, at which point you'll
become inordinately grateful that the Xa is supplied with a computer-based editor/
librarian. This cross-platform application is compatible with a wide range of
platforms, from Windows 98, 2000, ME and XP to Mac OS (8.6 to 9.2.2, and 10.2
upwards). Roland recommend that PC-based users run it on a 400MHz Pentium
or higher (500MHz for a Pentium III), and that Mac users should consider a
233MHz G3 the minimum spec for OS 8/9 usage, and a 500MHz G3 the minimum
for running the editor under OS X. 128MB of RAM is considered the minimum for
both PCs and Macs, with 256MB or higher recommended.
The editor can communicate with
the workstation via MIDI and USB,
and although you have to load the
appropriate driver for your PC or
Mac, you'll find that this is
straightforward and that, once it's
done, the two devices talk to each
other without difficulty. However,
while you can use the USB port on
the Xa to transfer samples to and
from computer, and to communicate
with the editor/librarian using MIDI
over the USB connection, you The main patch screen from the Mac OS X
cannot do both simultaneously. Editor.

The Editor is a boon, wholly

overcoming the limitations imposed by the Xa's reduced screen. Parameters are
laid out clearly and — as far as is possible with such a huge sound generator —
intuitively, with detailed graphics where required. Once you start to use this,
everything drops into place, and XV-style synthesis is no longer daunting, it's a
pleasure to use.
Unfortunately, the editor only knows about Patches and Performances. The
librarian part of the package handles rhythm sets, rhythm patterns, rhythm groups,
chords and arpeggios in addition to Patches and Performances, but it can't edit
them. This is a great shame. If the package included a sample editor, and the
attached PC or Mac could manipulate and control the entire workstation, it would
make the Fantom a real contender at the high end of music production.
Now, if I were reviewing any other Fantom with an 'X' in its name, this is where I
would say, "Ah yes, but..." and go on to discuss the FANX UP1 Fantom X Audio
Track Expansion Kit, which was released while I was writing this review. Not only
does this add eight stereo audio tracks to the Fantom's sequencer, it includes a
sample editor, a multisample editor, and a utility that claims to convert S700-
series sample libraries for use on the Fantom X. This appears to be dreamland
except that... it doesn't run on the Xa!

Conclusions

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Roland Fantom Xa

Much of what made the Fantom X6 good has survived in the Fantom Xa. But
nevertheless, the Fantom Xa has left me in a bit of a quandary. You can look at it
like this:

Roland have taken the rather tasty Fantom X6, and removed all the bits that
gave it an edge in a highly competitive marketplace. Gone is the very playable
keyboard, to be replaced by something that lacks aftertouch and does not permit
the highest standards of keyboard playing. Gone are the touch-sensitive pads, to
be replaced by mere switches. Gone is the large, colour LCD that make editing
so simple and such fun, to be replaced by a monochrome screen. And gone is
the huge ROM that made the other Fantom Xs special, to be replaced with the
sound engine of the previous generation of Fantoms.

On that basis, who could recommend the Fantom Xa? But you can also look at it
like this:

Roland have taken the rather tasty but over-specified Fantom X6 and stripped
out several expensive and largely unnecessary facilities to create a lower-cost
workstation while sacrificing none of its essential features. Gone is an expensive
keyboard unnecessary for producing and triggering dance grooves, to be
replaced by something that does the job just as well at a fraction of the price.
Gone are the unnecessary touch-sensitive pads, to be replaced by cheaper ones
that still trigger all your favourite rhythms and samples. Gone is the large and
expensive colour LCD, to be replaced by something that still allows you to tweak
your sounds. And gone is the overblown ROM full of unnecessary concert piano
sounds, to be replaced by an otherwise equally powerful but cheaper sound
engine more in tune with current trends.

On that basis, who could fail to recommend the Fantom Xa? I would say, though,
that since the Xa seems in many ways to be a groovebox at heart, and is aimed
squarely at dance music production, it might have been better to strip it out still
further; in its current form it may still be too complex (and expensive) for its
prospective market.

Clearly, the Fantom Xa is not going to appeal to all potential customers, but in
many ways it is still a well-crafted instrument that offers astonishing power for a
price that (as always) would have been unthinkable a few years ago.

Published in SOS May 2005

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Roland Fantom Xa

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
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Web site designed & maintained by PB Associates | SOS | Relative Media

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 Roland VC1

In this article:
VC1 & VariOS
Roland VC1
Schizo Synth D50 RAM Card for Roland V-Synth & VariOS
Patches Published in SOS May 2005
Virtual Digital? Print article : Close window
Editor & CD-ROM
Reviews : Sound/Song Library
Conclusions

Roland VC1 £99

pros
Authentic D50 sound — Roland's innovative V-Synth can now be
your V-Synth becomes a D50!
reprogrammed with a RAM card, effectively turning it
Offers you a better-sounding
D50 (thanks to the V-Synth's into another instrument. The VC1 card puts the clock
more modern audio back to 1987, perfectly recreating the S&S tones of
specifications), one which is
fully compatible with existing
the Roland D50...
D50 patches.
Fully self-contained — the
card stores all settings and Paul Nagle
user patches.
cons Launched in 1987, the D50 was Roland's
Changing between V-Synth answer to a growing perception that analogue
and D50 modes requires a was dull and a thing of the past, whilst FM
power-off. programming, the bright new concept that had
summary dominated synthesis throughout the mid-'80s,
If you have a V-Synth (or a could leave permanent furrows in the brow.
VariOS module), the VC1 Although Roland chose the acronym LA (Linear
card allows you to transform it Arithmetic) to describe its synthesis, the D50
into a D50 for about half of
the second-hand price of the
embodied what we now refer to as sample and
real thing. With an impressive synthesis, or 'S&S', its distinctive short samples
storage capacity and a useful (breathy chiffs, percussive plinks and so on)
supplied editor, this small supplementing synthesizer waveforms. This
card offers a fun diversion for lent the D50 a realism that's easy to overlook in
V-Synth owners who can now
carry two classic Roland
these days of massive samples.
keyboards in one flightcase.
information Paul Ward's D50 retrospective in July 1997's SOS, (which can be read in full at
£99 including VAT. www.soundonsound.com/sos/1997_articles/jul97/rolandd50.html) offers an
Roland UK +44 (0)1792 evaluation of this ground-breaking synthesizer, so for our purposes it's sufficient
515020. to note that a D50 patch consists of two tones known as Upper and Lower. Each
+44 (0)1792 799644. tone in turn comprises one or two sound sources (or 'partials' as Roland refer to
www.roland.co.uk them) which can either be PCM or 'analogue' waveforms. With enough
www.roland.com programming depth to keep the tweakers happy and an effects section supplying
a glossy finish, the D50 deservedly did well — at least until the Korg M1 came
along the following year...!

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Roland VC1

Fast forward to 2003, when Roland launched a very different flagship, the V-
Synth. My first encounter with it came at a trade show and by May, when the
Sound On Sound review was published (see www.soundonsound.com/sos/
may03/articles/rolandvsynth.asp), I had fallen under its spell. The V-Synth
bundles together virtual analogue oscillators, user sampling and COSM
processors; it also renders PCM waveforms malleable in ways they usually
aren't, by cunning use of Variphrase processing. Add to this a superior user
interface and a set of expressive performance controls, and you can see why I
was hooked.

Naturally, when I heard of a means to transform the V-Synth into a D50 without
any kind of invasive surgery, I was intrigued. I also wondered whether the
transformation would be of novelty value only, or whether 'digital retro' might truly
have something to offer.

VC1 & VariOS

I don't have a VariOS module, but according to Roland's documentation,
everything stated in this review with regard to using the VC1 with the V-Synth also
applies to VariOS, with the obvious exception of the things I've said about the V-
Synth's physical controls (because the rackmount VariOS doesn't have these).
Slot in the VC1, boot VariOS and you have a D50 in a rack.
Helpfully, the VC1 manual contains separate sections describing the operation
from the perspective of both VariOS and the V-Synth, because the respective
user interfaces differ so much. Judging by the VariOS section in the manual, I
would recommend using the software editor, which would make everything far
easier to operate!

Schizo Synth

The PC Card slot at the rear of the V-Synth (and at the front of the VariOS
module) is a valuable means of storing data, exchanging files and so on. With
appropriate software present, it is also the means by which the V-Synth can boot
with an entirely different personality. This was already a familiar concept on the
VariOS, but it is rather unusual for a flagship synthesizer to be capable of such a
drastic 'brain swap'.

The first of the mooted 'alternative identities' to be available for the V-Synth is
provided in the form of the VC1 or 'V-Card' (a VC2 Vocal modelling card was
announced at this year's Winter NAMM, but was yet to be released when this
was written). The VC1 card looks similar to a laptop PCMCIA adaptor, and you
insert it into the V-Synth's card slot. Installation is now complete — you simply
power on and the V-Synth boots as a D50.

In order to achieve this, Roland engineers apparently descended into the vaults
to retrieve the original D50 code, which was stored on 5.25-inch disks. Running

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Roland VC1

this code on the V-Synth produced a much cleaner

sound than the original D50, because the D-A
converters of the V-Synth are better than those of
the D50 (not surprising when you consider how
technology has marched on since 1987). To get
around this, the card offers a 'D50 mode' for those
that want it, which restores some of the grunge and
aliasing of the original. A system menu offers the
choice of the much-improved version or something
closer to the sound owners of the original are
familiar with. However, even in D50 mode, the V-
Synth's emulation sounds better to my ears, and
the reverbs are more acceptably quiet too.

Once you are up and running, you navigate using

the V-Synth's touchscreen, make edits with the knobs and have a fair degree of
flexibility with respect to assigning the V-Synth's performance controls. However,
since this is not a rewrite of the D50's OS — it is the D50's OS — the limitations
of that instrument still apply, and you can't create assignments that exceed its
original capabilities. Nor can you apply V-Synth capabilities to the VC1's D50 —
the V-Synth is either a V-Synth without the VC1 card in the slot, or a D50 with the
VC1 installed, but never a mixture of the two.

The V-Synth's Time Trip Pad is an intuitive replacement for the D50's joystick
and can be used to vary the levels of partials or the balance between Upper and
Lower tones. Similarly, you can assign the D-Beam to take on the duties of the
mod wheel, aftertouch or pitch-bender, or to act as a volume control or even a
hold pedal.

The D50 was not blessed with knobs — you had to purchase the PG1000
programmer separately for that kind of real-time control. Even with the
programmer, editing wasn't exactly a joy because you had to constantly retrigger
notes in order to hear the changes made by the sliders. With the V-Synth's
onboard knobs, you can tweak happily with no such restriction. You select a
partial or partials for editing by means of the touchscreen or via the V-Synth's
Structure buttons, and from then on it's simply a matter of turning knobs.
Although these don't map exactly to D50 functions, they are mapped out logically
enough and, in conjunction with the screen (because there are far more
parameters than V-Synth knobs), editing is way easier than it ever was on the
original. Most of the controls can even be remapped. If you select the small
controller icon in the bottom left-hand corner of the main Play screen, you can
then navigate around the V-Synth panel, plucking any parameters you like from
the huge selection available. Provided you ignore the physical labels on the V-
Synth, you can quickly memorise your way around.

The V-Synth has an arpeggiator, but the D50 did not, so nor does the VC1-based
version, as you would expect. The V-Synth's Arpeggiator button is therefore used
to activate the D50 Chase function, which is like a built-in MIDI-based delay; the
V-Synth's Hold button sets portamento on or off and the Arpeggio Speed knob

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Roland VC1

sets the Portamento or Chase (delay) rate.

Patches

The VC1 is supplied with all of the D50's original presets, plus the sound libraries
for the D50/D550. In addition, 64 new patches are supplied using 28 extra
waveforms that were too large to fit into the original synth (these include a
passable mellotron string sample, and several electric pianos). There are an
impressive 384 presets — but that isn't the end of the story. It's worth repeating
that the VC1 is functionally identical to the real thing, meaning you have
complete SysEx compatibility with any existing D50 patches. Therefore it's
helpful that the card has a RAM area into which you can place up to eight banks
of your own D50 sounds — that's another 512 patches on tap! If you don't have
enough original creations to populate these banks, a quick Google search on the
phrase 'free D50 patches' should yield enough to keep you happily occupied for
weeks.

Virtual Digital?
Was the D50 the world's first virtual analogue? I occasionally hear people
mistakenly suggest that it had true analogue oscillators and filters when, in fact, it
was an entirely digital machine. This is a testament to Roland's programmers,
who emulated analogue oscillators and filters with uncanny skills. Ironically, in
1987, the synth world was looking to distance itself from analogue; it was another
eight years before the circle was complete and the Clavia Nord Lead appeared on
the scene.

Editor & CD-ROM

The supplied CD-ROM contains a Uniquest editor for both the V-Synth and
VariOS. It's compatible with a variety of operating systems, from Windows 98
through to XP and also Mac OS X or OS 9, and is an excellent means of editing
patches or managing user banks. As well as graphical editing of parameters,
options such as randomise, blend, mix and morph ensure you should never get
bored when seeking new patch ideas. I was surprised to be able to throw up
plenty that sounded genuinely different using these morphing techniques. I also
generated several that were silent or constantly droning, but as you can produce
whole banks with indecent haste, this isn't exactly an issue!

Logically, since there is only one card slot in the V-Synth, the VC1 has to be a
RAM card in order to store user patches and be fully self-contained. But this also
means that you might inadvertently delete some of the files needed for the VC1
to operate. For this reason, Roland include on the supplied CD-ROM a copy of
the all-important program files and details on how to restore the card, should that

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Roland VC1

ever be necessary.

Conclusions

Unlike synthesizers that use modelling technology, the VC1 is an absolutely

faithful reproduction of an earlier digital instrument. This means that the V-Synth
can import D50 banks, can exchange patches with a D50, and yet sounds
cleaner and better than the original. If you turn off its 'D50 mode', you have an
even more polished sound — the V-Synth's signal path conveys the D50's
personality much better than the original hardware. I'm no purist, though, and
personally, I would have been happy if the VC1 had offered you a D50 with the
full V-Synth stock of effects and an arpeggiator. And although the Upper and
Lower tones can be split, layered or played from separate MIDI channels, the
D50 remains a bi-timbral instrument, whereas I'd have loved access to more
modern levels of multitimbrality. But as this is not an emulation, you are given
exactly the D50's capabilities and sounds, and nothing more (other than an extra
28 waveforms).

I don't think for a moment that I'd want a D50 to permanently replace my V-
Synth, but of course that's not the idea — when you want your V-Synth back
again, you simply remove the card and reboot as normal to restore the V-Synth's
functions. However, I really enjoyed my refresher course in LA synthesis —
there's still plenty of mileage left in it, especially when you delve into the supplied
editor. If that isn't enough to convince you, it's worth remembering that strings,
brass, and organs were not really the focus of the original V-Synth factory set,
and given the abundance of free D50 patches on the Internet, the VC1 is a great
way of adding more conventional sounds to the Roland V-Synth.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Roland%20VC1.htm (5 of 5)9/27/2005 9:23:17 PM

 Sample Libraries: On Test

In this article:
Jam Pack 4: Symphony
Sample Libraries: On Test
Orchestra **** Sample Shop
Sonic Boom Box **** Published in SOS May 2005
Downbeat Leftfield **** Print article : Close window
Platinum Essentials *****
Reviews : Sound/Song Library
Star Minerals
***** Cummingtonite
**** Khanneshite
Jam Pack 4: Symphony Orchestra ****
*** Dickite
APPLE LOOPS
** Fukalite
* Sillimanite Garage Band may not be considered a serious piece of recording software by
some, but in reality it does most of what the majority of us actually need from a
sequencer, and Apple Loops, far from being gimmicks, have lots of applications
in songwriting and even in the creation of original music. Building whole songs
from 'cut and paste' elements may seem a bit of a cop-out, but many pieces of
TV music are already made that way and, after hearing this library for myself, I
can understand why this is possible — the quality is seriously impressive.

Once the installation has been carried out from the two
DVDs, containing around 10GB of material, you'll find
that you have over 2000 orchestral loop elements that
can be used in Garage Band 2, Logic Pro 7, or Logic
Express. There are also over 30 sample-based
orchestral instruments, which Logic users can load into
EXS24 or EXS24P (though some of the clever mod-
wheel/pitch-bend tricks may not come over with the
samples). The loops are mainly solo instrument phrases
covering a number of classical and film styles, and all
the ones I auditioned sounded smooth and lush, with
exactly the right character and an appropriate amount of
added ambience — I can envisage many of the parts sitting nicely in pop songs
or soundtracks. In addition to the vast range of loops, there's also a great
selection of sample-based orchestral instruments divided into strings, brass/
woodwind, and keyboards/percussion, again with a suitable ambience added at
source.

All the main orchestral instruments are represented, and the sounds have a
mature, mellow quality which contrasts with some of the strident samples that
I've heard from other libraries. The strings are particularly good, though it's hard
to fault the wind samples either. There may not be a great depth of multi-velocity
sampling going on (usually just two layers), but the sounds just seem to work
together, which is really all that matters. Furthermore, the designers have made

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Sample Libraries: On Test

full use of the modulation and pitch-bend wheels to add expression to specific
instruments — for example, the orchestral oboe changes timbre depending on
the mod wheel position and eventually produces staccato notes at the extreme
mod wheel position. The pitch-bend wheel sets the starting volume, so it can be
used to dynamically control notes. I did manage to get some of the loops to click
by wiggling the mod wheel excessively during playback, but reloading the
instrument seemed to cure this problem.

The pianos and harps also sound the part, and while this pack may not give
Vienna Symphonic Library a hard time, it's incredible value, both for its loops and
its sample-based instruments. Only some audible sample looping (which would
probably go unnoticed in a real composition) prevents me giving this the full five-
star treatment. As it is, it's a worthy four. Paul White

Apple Loops 2-DVD-ROM set, £65 including VAT.

Apple Store +44 (0)800 039 1010.
www.apple.com/uk

Sonic Boom Box ****

APPLE LOOPS

This is a cost-effective Apple Loops product from Ultimate Soundbank which,

rather than being too tightly themed, sets out to give the users a broad palette of
styles, both in Apple-Loop form and as software instruments that can be used
within Apple's Garage Band software. The Apple Loop section, which also works
in Final Cut Pro, Logic Pro 7, Logic Express, and Soundtrack, comprises over
4000 loops, both instrumental and percussive, and these are augmented by in
excess of 220 software instruments.

The collection comes on a single DVD and requires 4GB of

hard drive space for installation. Included instrument genres
encompass drums, percussion, basses, guitars, and a large
number of synths, as well as some more traditional
instruments. A lot of the rhythm loops are aimed at
contemporary dance and hip-hop styles, but there's also
some traditional pop/rock material as well as riffs and
phrases played on guitar, bass, wind, strings, and
synthesizer/keyboard. The blurb on the back of the box
says that 'your composition will sound as good as a
professional record in a couple of seconds'. Personally, I'd
say this claim errs on the side of exaggeration, but I have to
agree that the quality of the material provided here rivals that of far more costly
sample CD libraries.

While many of the Apple Loops products I've looked at so far concentrate on a
specific genre, Sonic Boom Box could be considered the Apple Loops equivalent
of a workstation synth, albeit with a dance-music bias, and as such it's a good

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Sample Libraries: On Test

starting-point for your collection, especially if you work in diverse pop genres.
The loop material is nicely produced, rarely needing additional processing to
make it sound good, and while the included instruments are pretty simple, they
still manage to sound strong and 'right for the job'. I particularly like the quality of
the percussion and rhythm loops, though some are a bit too 'off the wall' for my
taste — words like 'chemical' and 'acid' come unbidden to mind. In all, there's
some great stuff in here and it's certainly easy to get the basis of a groove
together to stimulate your creativity, which is where I think these packages score
best. If you like variety and have a penchant for loops that are a bit 'out there',
Sonic Boom Box is a bargain. Paul White

Apple Loops DVD-ROM, £66 including VAT.

Time + Space +44 (0)1837 55200.
+44 (0)1837 55400.
Click here to email
www.timespace.com
www.usbsounds.com

Downbeat Leftfield ****

MULTI-FORMAT

The Loopmasters Origin Series, of which this library forms a part, aims to offer
100-percent copyright-free loops, multisamples, and single sounds in a value-for-
money format. All the CDs in the series include WAV loops, REX 2 files for all the
loops, and some Propellerhead Reason instrument patches. As suggested by the
title of this CD, the collection of samples here were all recorded at lower tempos
(100bpm and slower), while the musical content is both dark and somewhat
quirky — hence the 'leftfield' tag. The library totals some 700MB of data, with 275
REX 2 loops, 700 WAV files, and samples for 21 different drum kits included.

A good chunk of the collection is made up by

a mixture of over 100 live and programmed
drum loops recorded at 100bpm, 90bpm, and
80bpm. There is some good stuff amongst
this lot that would suit anything from hip-hop
through to chill-out. The sounds are nicely
produced and not over-processed, with some
of the percussive elements lending a darker
feel to the grooves created. The tone is not lo-
fi gritty, but it does conjure up a 'serious' hip-
hop mood, so think Nelly or Snoop Dogg
rather than Goldie Lookin Chain or The
Beastie Boys — although you could get to the latter with suitable extra
processing. A separate folder of percussion loops, drum fills, and other drum
'extras' is also included.

Aside from the drum loops, the rest of the collection provides a diverse set of

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Sample Libraries: On Test

instrument loops and sounds. These include a folder of bass loops (plus a small
collection of Reason NNXT bass patches) whose moods match the drum loops
very well, so there is plenty of scope for mixing and matching between these. A
further folder of 'instrument' loops is subdivided into guitar, keyboard, pad/
atmosphere, strings, vocoder, and woodwind (dominated by solo sax lines)
groups. This is a pretty mixed bag. For example, the keyboard group includes a
good number of chilled Rhodes loops while, in contrast, the pad/atmosphere set
provides a mixture of both chill-out and much darker moods. Two further folders
complete the collection. Firstly, a sound effects folder contains various bleeps,
bloops and other noises — all very disturbing. Secondly, the Vibes & Atmos
folder contains a mixture of bed-style pads and sound effects, plus a collection of
turntable scratches.

In use, there is plenty to get your teeth into in this collection, and the loops from
the various groups can easily be mixed together to produce a complete musical
arrangement. The single hits, REX 2 loops, and instrument patches (for Reason
users) add extra flexibility. For styles moving from Zero-7 chill-out through to
darker hip-hop moods, Downbeat & Leftfield has plenty to offer. The only
downside is that, while the material is all very useable from a musical
perspective, I'm not sure there is anything radically new offered here. That said,
the sample set is competitively priced and certainly does offer good value for
money. John Walden

WAV, REX 2, and Reason CD-ROM, £39.95 including VAT.

Time + Space +44 (0)1837 55200.
+44 (0)1837 55400.
Click here to email
www.timespace.com
www.loopmasters.com

Platinum Essentials *****

AUDIO+WAV

This title aimed squarely at fans of West-Coast hip-hop has been produced by
Keith 'Clizark' Clark, a man who already has 10 million record sales to his name
having worked with such names as Snoop Dogg and the Eastsidaz. The single
split-format disc acts as both a normal audio CD and a WAV CD-ROM (all
samples are duplicated in both formats), and the audio comprises 20
construction kits followed by a small group of additional synth samples.

The construction kits cover tempos of 93-103bpm, and each track begins with a
mixed track featuring MPC-style sampled drums, bass, synths, and some guitars
— this lasts less than a minute. Most of these mixes exude an understated yet
powerful menace which would be well complemented by the kind of smoothly
murderous rapping that Snoop Dogg has nailed. The rhythm programming is
beautifully done, creating that head-nodding momentum that so often seems to
separate the hits from the 'B' sides — the fact that Clizark has made this difficult

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Sample Libraries: On Test

task sound so easy here should leave no-one in

any doubt as to his production chops!

Although the style of all the tracks is quite

sparse, this is definitely part of Clizark's
trademark sound, and you'd have no trouble
transforming any of these mixes into a finished
production with just a bit of editing and the
addition of some vocals. The solid basses
doubtless account for some of this gravitas,
especially as they have been shrewdly chosen to
complement the kick drum in each case.
However, it's the meaty drum sounds which really carry each track and which are
the obvious highlight of Platinum Essentials to my ear: just the right amount of
attack to sound punchy, just the right amount of mid-range and sustain to provide
body, and just the right amount of low end to anchor everything firmly. You could
spend a great deal of time trying to obtain drum sounds as good as on even one
of these construction kits, and you get 20 different sets here. If they had Desert
Island Discs with sample CDs, including this one would be a no-brainer as far as
I'm concerned!

Clearly, then, I was delighted to find that all the drum loops and single hits from
each full mix were included in each construction kit. However, I wasn't quite as
delighted to find that this was pretty much all that was provided in the way of
separate elements — none of the guitar licks, synth pads, and other pitched
samples which had helped round out the mixed tracks so successfully. Anyone
expecting a more traditional construction-kit format would justifiably feel a little
cheated. The small selection of synth multisamples at the end of the CD help
make up for this a bit by including ten rather nice sets of 'one note per octave'
bass-synth samples, which are pretty much to the standard of the drum sounds.
However, the other synth sounds are a wash-out, and don't sit well against the
rest of the library content.

If it weren't for the bargain UK price, I'd be rather in two minds about giving
Platinum Essentials as a whole our top star rating. On the one hand there are
drum sounds here which would outshine many five-star releases; on the other,
the incomplete track breakdowns, measly 32-minute running time, and wafer-thin
documentation leave me with a sneaking feeling that someone's having a laugh!
The bottom line for me, though, is that the drums and basses are worth the
mockery, and at this price that's got to mean five stars. Mike Senior

Split-format Audio CD & WAV CD-ROM, £32.95 including VAT.

Time + Space +44 (0)1837 55200.
+44 (0)1837 55400.
Click here to email
www.timespace.com
www.bigfishaudio.com
Published in SOS May 2005

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Sample Libraries: On Test

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Sample%20Libraries%20%20On%20Test.htm (6 of 6)9/27/2005 9:23:24 PM

 Steinberg Groove Agent 2

In this article:
Back To Basics
Steinberg Groove Agent 2
Installation Virtual Drummer Instrument (Mac/PC)
New Styles Published in SOS May 2005
New Sounds Print article : Close window
Copy Protection
Reviews : Software
More Control
In Use
Verdict
Using Groove Agent 2 Under
Logic From the same team that brought you Virtual
Guitarist, Groove Agent has had an impressive
Steinberg Groove Agent
2 £170 upgrade, and is now claimed to work better under
pros
non-Steinberg hosts. We put it through a proper multi-
Incredibly easy to use with
platform test.
Cubase SX.
Wide range of usable styles.
Martin Walker
Loads of different, easily
tweakable kits.
cons From Bornemark, the same Swedish
As our Logic-based tests developers that gave us Virtual
show, users of other hosts Guitarist and Broomstick Bass
may face a frustrating time! (reviewed elsewhere in this issue),
Sounds have fewer velocity Groove Agent is essentially an 'instant
layers than some competitors. drummer' for those that either can't or
summary don't have the time to program their
While it doesn't offer the most own drum rhythms, or who want some
sophisticated multi-layered kit rhythmic inspiration. Plenty of us fall
sounds, Groove Agent 2 is into these categories, yet when I first
perfect (at least under a
saw version 1 displayed at the Frankfurt
Steinberg host) for any
musician who wants to Musikmesse in 2003 I was tempted to
explore a huge range of dismiss it as a gimmick — that is, until I
drumming styles with the was given a proper demonstration of its
minimum of fuss and the capabilities. The sounds are all high-
maximum of enjoyment.
quality 24-bit audio (mostly recorded
information onto analogue tape), and although you
£169.99; upgrade from can use it as a simple drum machine, it
version 1, £55. Prices actually provides far more creative
include VAT. possibilities.
Arbiter +44 (0)20 8970
1909.
+44 (0)20 8202 7076. Version 2 offers many more rhythm
www.arbitermt.co.uk style options (81 instead of version 1's
www.steinberg.net 54) including grunge, punk, and trip-
hop, plus nine new kits, and there are

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Steinberg Groove Agent 2

Test Spec now up to eight stereo outputs available Groove Agent 2 looks very similar to v1
for more refined mixing options. What's apart from the extra Solo buttons for the
PC REVIEW SYSTEM
more, you can now bypass the internal eight groups.
2.8GHz Intel Pentium 4C PC sample-playback engine and output
with 1GB of DDR400 RAM
MIDI data to trigger your own preferred drum sounds, and you can capture MIDI
running Windows XP with
Service Pack 1, and based on performances to a MIDI file independent of any host application. Let me fill in the
an Asus P4P800 Deluxe details...
motherboard with an Intel
865PE chipset, running an
800MHz Front Side Buss.
Steinberg Groove Agent v2.0. Back To Basics
Steinberg Cubase SX v3.01.
Cakewalk Sonar v4.0. We reviewed version 1 back in SOS July 2003, but here's a brief recap. Groove
Agent can play drums in a variety of styles, arranged chronologically across the
Tonewise DirectiXer v2.5.
upper curving timeline slider, starting in the 1950s with 'Swing' and moving
MAC REVIEW SYSTEM through five decades to the 21st century and 'Mini Club'. Each style has an
Dual 2GHz Apple Mac G5 associated drum/percussion setup, but you can unlink the lower part of the slider
with 1.5GB of RAM running to marry any kit with any style. Furthermore, each style provides 25 complexity
Mac OS v10.3.8. levels from laid-back to incredibly busy, chosen from the lower slider, each with
1.3GHz Apple G4 its own unique 'fill', and once again you can link/unlink the fills from the
Powerbook with 512MB of complexity level.
RAM running Mac OS v10.3.8.
Apple Logic Pro v7.01 (both Other refinements include the choice of snare or sidestick, buttons to trigger
machines).
accents or fills on demand, a half-tempo feel, and a random option that plays
Apple Logic Pro v6.4.3 (both slightly varying patterns. A set of rotary knobs down the left-hand side of the
machines). window lets you create a triplet shuffle feel, loosen the timing, add simple limiting,
and control the Ambience balance by mixing together dry, two-metre, and seven-
metre distant miked versions of the same sounds.

Installation

Besides the jump in required hard drive space from 300MB to 450MB, system
requirements are much the same as for version 1 (a 1.4GHz PC Pentium or
Athlon processor for PC users, or a Power Mac G4 Dual and 1.25GHz), although
Windows 2000 and Mac OS 9 support have been quietly dropped. 512MB of
RAM is also recommended, but I'd increase this to 1GB for a typical user who's
running a host sequencer and other softsynths alongside.

Like most dongle-protected software, Groove Agent 2 should be installed without

the dongle, to avoid your computer finding this new hardware device before the
relevant drivers have been installed. Other than that, I found installation an easy
ride, although I'd have liked an option to install the plug-in in my usual 'vstplugins'
folder and the 450MB of audio files on another partition — some of us prefer to
keep our Windows partitions as small as possible. PC owners get the option to
install the DXi and Rewire versions along the way if they need them.

Those upgrading from version 1, like me, will be forgiven for initially thinking that

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Steinberg Groove Agent 2

they've been sent the old version by mistake, since until you notice the '2'
appended to the Groove Agent logo, there's very little visual difference between
the two, and exactly the same list of Styles appears across the top slider. The
secret is that many are now displayed in a different colour, and if you right-click
on these options, a pull-down menu appears with further related style options
(see the 'New Styles' box above for full details).

New Styles
Unlike the historical linking of kits and styles to the timeline in version 1, the 27
new styles included in version 2 are somewhat more arbitrary, and are tucked in
among the originals as right-click menu options. Here's the complete lowdown in
the order in which they appear along the timeline.
First up is 'Bop' for jazz standards, then there are two strange ones in 1953 under
'Paint' (presumably a Jackson Pollock reference?). 'Ominous' uses a palette of
heartbeats, morse code, and deep mechanical noises for some unsettling
soundscapes, while 'Machinery' mixes thuds, rattles, taps, and escaping steam to
form relentless, pulsing rhythms ideal for future city soundtracks.
Rather more traditional are '6/8' (this handy time signature was missing altogether
from version 1), 'Slow Blues' for jam sessions, 'Steady Beat' for straightforward no-
surprises drumming, and the latin-influenced 'Mozambique'. Next up are 'Bombay
Dance Hall' and 'Roots' (the latter utilising the new 'Noisy' kit), both filed under the
'Reggae' style, but with very different flavours.
The Hard Rock section has been fleshed out considerably, with extra entries for
'Grunge', 'Indie Punk', 'Unplugged', and 'Ballad', while 'Basic Hip-Hop' has been
supplemented with a simpler 'Live' and a 'Sloppy' version inspired by the Beastie
Boys. Meanwhile, analogue drum machine freaks will rejoice that the rather
generic 1981 'Elektro' style is now joined by 'Vintage FR3', 'TR7', 'CR8', 'Meek
Ballad' (where the 25 complexity levels simply add more and more beats to the
same pattern), and the harder sounds of 'Axis Y'.
The hard rock 1984 'Arena' style now has 'HM Straight' and 'Triplet' feels for
metalheads, the busy eighth-note feel of 'Grind', and the technique and busy fills
of 'Progressive'. 'Daft' provides simple rhythms, and 1994's 'Trip-Hop' style is now
supplemented by the acoustic sounds of Portishead-inspired 'Bristol Trip'. Finally,
'Kelly' is an additional R&B style that's slow and heavy with double-tempo hi-hats.

New Sounds

New Solo buttons feature alongside each instrument, making it far easier to
tweak individual sounds in the mix. Version 1 already offered plenty of internal
editing possibilities, including velocity response for selecting the softer or harder
end of each instrument's sample splits, ±1 octave tuning, decay, individual
ambience, volume, and output selection. However, version 2 expands the
number of possible stereo outputs from four to eight, so that you can now treat
each of the eight 'groups' (kick, snare, toms, hi-hat, ride, crash, percussion 1, and
percussion 2) through separate effects if you wish.

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Steinberg Groove Agent 2

Because you can choose from one to eight stereo outputs in version 2, the
'Ambience to output 4' option from version 1 (located under the Setup lid at the
bottom right of the window) has now become 'Ambience Split', which routes the
ambient and reverberated versions of all parts to the highest available audio
output (dependent, of course, on how many you've chosen).

And the sounds? Version 1 offered '50s jazz, '60s pop, '70s rock, and '80s studio
kits, plus various percussion instruments and extras such as brush and mallet kit
sounds. Version 2 adds a top-of-the-range Studio kit for clean, modern sounds
including three new snares (a Slingerland Radio King, a Slingerland copper, and
a model handmade in Prague), and a hard-sounding Heavy Kit designed for
metal styles with busy kick-drum patterns, and with ride and crash cymbals
specially selected to cut through the sound of multiple distorted guitars.

The 'Noisy' Kit uses tiny traditional drums including a 10-inch mini-snare, treated
through digital effects to give them a modern and much bigger lo-fi sound, plus a
mixture of rare vintage and knackered modern cymbals. There's also a handful of
electronic drum sounds including the Simmons SDS9, TR7, and CR8, treated
with ambience from a vintage EMT plate reverb, and combined into a versatile
selection of electronic kits.

This new selection of sounds adds freshness and variety to the new styles, but
given that you can bypass any or all of the instruments pre-selected for each kit
in favour of any other allocated to the same sound 'group', Groove Agent offers
incredibly versatile options — for instance, there's now a total of 36 snares, 25
kick drums, and 28 toms on offer, while the two percussion groups together
encompass 76 instruments from tambourine, triangle, and shakers to handclaps,
djembe, bottles, and tabla.

Copy Protection
Although I'm always loath to devote review space to discussing copy protection,
this is the first time I've been sent a product that requires a dongle but doesn't
ship with one, so here are the pertinent details. Like the majority of Steinberg's
latest applications and software synths, Groove Agent 2 won't run unless you plug
in a properly-licensed Syncrosoft USB dongle. Anyone who already has products
such as Cubase SX, Nuendo, Hypersonic or Virtual Bassist will already have one
or more of these, and thankfully, a single dongle can hold licences for multiple
Steinberg software applications, so you won't have to invest in an extra USB hub
to plug them all in at once! You can also transfer licences from one dongle to
another at any time.
However, Steinberg don't include a dongle with Groove Agent 2, so anyone who
normally uses other host applications such as Logic or Sonar may have to buy
one separately — they cost £20 in the UK. All that is supplied is a 32-digit
Activation Code, which you must use with the supplied Syncrosoft Licence Control
Center software that is installed along with Groove Agent 2. You run its 'Licence
Download' Wizard, enter the supplied code, and must then go on-line so it can
interrogate Steinberg's database, declare the code a valid one, and then

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Steinberg Groove Agent 2

download the corresponding licence into the dongle. Once you've done this
successfully, Steinberg's on-line database will no longer accept the same code to
prevent those with multiple dongles licensing them all, although safeguards are
built in to ensure that if for any reason your download fails part of the way through,
you can try again.
In fairness to this approach, licensing for those whose music computers are
without an Internet connection has been improved — you no longer have to install
the Syncrosoft software on another computer with Internet access, as I described
in PC Notes in February this year. Instead, you can now generate a 'Pending
Licence' with an associated Challenge File, copy/paste this document into a
computer with Internet access, and then copy/paste the resulting on-line
generated Response file back into your music computer and use that to generate
the licence.
However, I already know of some non-Cubase users who have been extremely
disgruntled to find themselves expected to fork out an extra £20 to run the
software they have just bought in all good faith. Given that old-timers may now
have several redundant dongles lying around, Steinberg perhaps ought to provide
a recycling scheme to bring down the cost to new users.

More Control

Although you can use Groove Agent as an excellent jamming companion,

changing patterns in real-time and adding accents and fills, those writing songs
will find the version 2 capture features much more versatile. You can capture
your 'performance' as a MIDI part in your host sequencer as before, which as you
might expect works fine with Steinberg's own Cubase and Nuendo. However,
outputting MIDI data from a VST Instrument is exploring the boundaries of the
VST 2.0 protocol, and other hosts including Logic Pro can't cope with it (for more
on using Groove Agent 2 with Logic, see the large box over the page by Dave
Lockwood). In an attempt to overcome some of these limitations, version 2
provides a 'Live Ð ð File' option under the Setup lid to output the data directly to a
MIDI file so that you can later import into any host for further tweaking.

If you find a pattern that works well in a particular song apart from one particular
group part, you can mute this and play in your own manually from a MIDI
keyboard. All the individual drum sounds are available from MIDI keys, as are the
pattern selection, stop/run, accent, fill, and various other settings, while MIDI
controllers can be used to alter the rotary control settings. If you feed Groove
Agent from an even-numbered MIDI channel rather than an odd one, the keys
used for selecting pattern complexity instead control the mute buttons for the
eight groups.

So far this is as it was in version 1, but version 2 adds further muting options —
the toggle mutes of version 1 can be velocity switchable (playing the key softly
acts as a mute, while hitting it hard unmutes it), or operate 'while held', so that
their individual mute status toggles to the opposite as long as you hold down a
key. The Stop/Run buttons now also have a new option to pause or stop Groove
Agent's output when you stop your sequencer.

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Steinberg Groove Agent 2

In Use

Cubase SX3 proved to be the perfect host, and I had absolutely no problems
running Groove Agent 2 within it, recording its MIDI output directly into a Cubase
MIDI track, controlling it remotely from an external MIDI keyboard, or directly
automating any of its controls by moving them on the VSTi and recording these
moves directly into Cubase in multiple passes. In fact, I found the whole
experience a rewarding one, and when I really tried to find limitations, such as
some styles with busy ride cymbals or tom rhythms that became a little
mechanical, a little automation of the Velocity Offset knob worked wonders in
adding a little variation.

The Sample Engine bypass is

apparently not a new feature, but
rather a clever workaround discovered
by enthusiastic users and publicised in
v2.0. By setting the input of your MIDI
track to Groove Agent, so that it
receives the drum note and controller
data, and the output of that same track
to another drum synth or sampler with
identical key mapping, you can trigger
any combination of sounds from the
internal patterns. Again, this worked
fine for me in Cubase, but of course it
Here you can see the MIDI loopback option
wouldn't from other host applications
provides by Tonewise's DirectiXer, which
that can't receive MIDI output from a makes it much easier to use Groove Agent 2
plug-in. within Cakewalk's Sonar. The new 'Live < >
File' option can also be seen under the Setup
lid on the bottom right. This allows you to
Using Cakewalk's Sonar as a host capture your performances as an external
proved to be a slightly tougher MIDI file, making Groove Agent 2 potentially
experience, and it took some head- compatible with a wider range of host
scratching before I worked out how to applications. It still didn't work reliably under
Logic, though!
capture MIDI note performance data to
an external file — what you have to do
is change the MIDI Output option from 'Live Output To Host' to 'Record to File'
before activating the MIDI Output toggle under the Setup lid. This starts the
recording process, but you must then stop it manually by returning Groove Agent
2's MIDI Output to Off, whereupon a MIDI file is created and placed on your
desktop. I dragged this into the appropriate MIDI track within Sonar with no
problems.

Automation proved more difficult to master, but during the course of my

investigations I discovered such an elegant solution that I abandoned further
tests. The answer is to purchase Tonewise's $49 DirectiXer wrapper, which has
a MIDI loopback function. Using this with Groove Agent 2 let me record its
performance directly into a Sonar track, and record automation, just as in

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Steinberg Groove Agent 2

Cubase. Nevertheless, during my time with Sonar I experienced a few random

stops during playback, plus missing initial notes, so I can't claim 100-percent
success for this combination.

Verdict

I was a fan of Groove Agent, and version 2 has more of everything. Its '1970s hi-
fi' user interface isn't very inspiring, but don't let looks put you off — there's an
impressive engine under the bonnet! Audio quality is very good, and the kits
range all the way from ultra-traditional through analogue to occasionally weird. I
can recommend it unreservedly to Steinberg users, although those running other
plug-in hosts may not have as smooth a ride.

It's great for any musician who may get called on to

explore unfamiliar territory, including those working in film,
computer games, and library music, where its extensive
range of styles is unlikely to let you down. However, even
if you mostly work within a narrower range of tempo and
genres, the mix-and-match approach to styles, kits, and
instruments means that you can rapidly come up with
something new, exciting, and sometimes unexpected — I
tried it out on a friend with a string of trance albums to his
credit, and he was soon a convert.

So what competition is there for Groove Agent 2? Well,

it's not a beat-slicer or sample loop player, which makes it
rather different from Spectrasonics' Stylus RMX with its
7.4GB library, and the 8GB of Submersible Music's Groove Agent 2
Drumcore, which may both have more initial attitude to provides many more
jump-start new songs, but might be harder to fit into kit sounds, as you can
existing ones (although Groove Agent 2's eight outputs see from these lists of
make it easy to apply radical processing to each group Kick and Snare
options — the new
separately if you wish). drums for version 2
are in the bottom
FXpansion's BFD is probably a closer competitor with its section.
Groove Librarian and 9GB of authentic acoustic drum kit
samples, and since these provide many more velocity layers than Groove Agent
2, they will sound more realistic. Nevertheless, Groove Agent's much smaller
450MB set of sounds means you can switch between kits and create new
combinations almost instantly, rather than having to wait several seconds for
huge samples to be loaded in before you hear the difference, and there are
almost limitless kit combinations available to experiment with, especially once
you start exploring the individual tuning, decay, and ambience controls. It also
provides dance and analogue kits that BFD doesn't cover. Of course, if you wish,
you can use Groove Agent 2's sample-bypass mode to trigger BFD and get the
best of both worlds!

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Steinberg Groove Agent 2

For me, Groove Agent 2 excels in its immediacy and in its much larger range of
instantly available styles, while its interface makes it incredibly easy to generate
and capture your real-time performances. If, like me, you want to be able to work
quickly when inspiration strikes, Groove Agent 2 can be up and drumming in no
time at all. I loved it!

Using Groove Agent 2 Under Logic

Groove Agent 2 is now available in
Audio Units format, so there's no
more VST wrapper to worry about
and no text-file kludge just to allow
the program to find its samples, but
whilst the program itself has been
significantly enhanced, it seems that
many of the issues surrounding its
operation within non-Steinberg
sequencers remain.
I like Groove Agent a lot. I like its
sounds, I like its immediacy and
controllability, and above all I like Manually written automation driving Groove
the musicality of the playing — and I Agent 2.
guess that's the reason why I persist
with trying to use it inside Logic, my
sequencer for the last 10 years or so, despite the issues that this brings up.
Groove Agent resides at the 'outer limits' of the VST2 instrument specification. As
a plug-in, it transmits not only its control data, but also note data, writing its output
to a MIDI track within Cubase or Nuendo, as Martin Walker details in the body of
this review. This allows you to generate a performance with the 'broad-brush'
controls of the GUI, but then edit that performance, right down to individual hit
resolution, to achieve something quite specific. You don't have to work that way,
though. Within Steinberg hosts, you can choose to simply record the output from
the interface as automation data and let the program run in real time. Failing that,
you can trigger the Groove Agent plug-in with MIDI note and controller data, and
edit the resulting performance at 'control' level, rather than at drum-hit level.
Within Logic, however, plug-ins can't transmit note data, so writing directly to a
MIDI track within your sequencer is not possible. Unfortunately, Groove Agent's
on-screen controls won't send any automation data to Logic either, so you are left
with a choice of manually writing automation or note data into Logic or using an
external keyboard or control surface. Having exhaustively explored all the options
over the last few weeks, I can heartily recommend... well, none of them, actually!
Each offers something, and each causes sufficient frustration to have at least one
committed Logic user thinking, "Hmm — maybe I will learn Nuendo."
The simplest option is to use a MIDI keyboard. You record MIDI note and
controller data to a Groove Agent-enabled audio Instrument in Logic, with the
notes above B3 each triggering a specific Groove Agent pattern. Fills are
triggered either by using the mod wheel or entering a pattern-select note with a
velocity of 90 or more. You hear the result in real time and you can subsequently
edit the performance, at least to some degree, in that you can go in and change
which pattern is playing by altering the MIDI note number, and affect when fills

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Steinberg Groove Agent 2

and other events occur by changing the timing of controller data. Editing can be
rendered a lot more intuitive by setting up a Mapped Object in the Environment,
which allows you to rename notes with their Groove Agent function, but you can't
easily address all of the parameters within Groove Agent using this method alone.
At the other extreme, you can choose to
abandon any semblance of real-time
interraction with Groove Agent and just
manually write automation data directly into
Logic to control it. When you activate
automation on a Groove Agent audio
instrument track, all the automatable
parameters appear listed in alphabetical order
(see right). You just choose what data you
want to write, and enter the necessary nodes
to create the required data values where you
want them to occur. It's far from intuitive, but
you can be very precise. Square vector points,
for 'clean' transitions between discontiguous
values (say, switching from Pattern 5 to 19)
can be created by 'rubber-banding' the
automation line while holding down the Shift, Activating automation on a track
Alt, and Control keys. This creates a pair of containing Groove Agent 2 brings
nodes, and you can then grab the line in up all of the instrument's
between and move it up or down, which is parameters ready for automation.
much quicker than trying to get four nodes in
exactly the right place yourself! Curiously, some of Groove Agent's switched (On/
Off) functions seem to need to flip from 0 to 127 rather than just between 'zero
and anything above zero', while others seem content with any value above 63. In
some instances, I found that any 'above zero' value would be enough to switch
Groove Agent's user interface, but that a 127 value was actually required to
activate the function. All the parameters are named in their separate automation
lanes (as shown in the screenshot on the left) so you can see their potential
interactions, such as when fills are triggered with repect to changes in the fill
pattern number, and you can easily move their relative positions for working on
specific combinations of parameters.
Mixing the automation-based and the real-time, keyboard-based working methods
described here seems to offer the best combination of flexibility and reliability. It
makes sense to use note data for basic pattern selection and fill triggering, to
maintain a degree of performance spontaneity, whilst manually written automation
takes care of everything else. If you're going to do this, though, it doesn't seem to
be a good idea for Logic users to heed the advice in the manual about setting up
Groove Agent to follow your sequencer's Start button. Once you've connected the
plug-in's Start parameter to Logic's transport, it no longer seems to be solely
under the control of the programmed note or automation data, and you will often
find that it 'free runs' (albeit in perfect beat sync) when you shuttle back to an
earlier point in the track, or even past a programmed Stop instruction.
The best way I found to work with notes triggering patterns was to use Force
Legato (Select All, then Shift, Tab) combined with Logic's Chase Notes facility
(File>Song Settings>MIDI>Chase:Notes). This ensures that Groove Agent
receives a Start instruction for the correct pattern wherever you choose to restart
the sequencer within the song.
Another possible working method would be to use a keyboard or hardware mixing
surface mapped to the plug-in's automation parameters, but the Groove Agent/

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Steinberg Groove Agent 2

Logic combination managed to frustrate most of my best efforts in this area.

Assigning external controllers to Groove Agent parameters is easy enough using
the automation Learn facility, and elicits the desired response from the
instrument's user interface, in that faders will fade and switches will switch.
However, it is sometimes possible to modulate on-screen controls without
affecting the parameters (ie. producing no audible change) and it seems to be
somewhat hit and miss whether automation data will be recorded or not.
Occasionally, fader movements were simply not recorded, and in the end, I felt I
had to abandon this option as too unreliable for serious work.
My next thought was that it might be
possible to use Environment objects
to send automation data to the
sequencer input. You can work out
the necessary definitions for the
Environment faders by examining
the Automation Event list after
manually writing some automation
moves. If you are not familiar with
this Event list (shown above), it is a
really useful Event editor dedicated
to automation data. Prior to v7 of
Logic, it could be accessed by
simply holding down the Alt and
Logic's Automation Event List.
Command/Apple keys and hitting
'E'. It may have lost its default key
assignment in the latest version, but the editor is still in there — you just have to
set up the key command yourself. Rather as with using a hardware controller,
however, I found I could affect on-screen controls, but not record automation data.
A quick check with one of Logic's own plug-ins in this configuration established
that they don't work that way either, so this one is not actually a Groove Agent
problem. It just doesn't feel right to me to conclude that there is no way round
Groove Agent's inability to output data within Logic, so I'll go on seeking a reliable
solution. In the meantime, if any other Logic-using Groove Agent fans have further
suggestions or cunning plans they'd like to share, these would be more than
welcome via email or in the SOS Mac forum.
The infamous random dropout that was present when running version 1 of Groove
Agent under Logic is still evident to some extent. In some songs the audio may
simply stop for a bar and then resume, perfectly in sync. Usually, if it has done it
once, it will go on doing it at precisely the same place, although sometimes it will
stop dropping out in one place and start in another, but never for more than one
bar. Sometimes it will stop doing it if you close the song and reopen it, or even
simply start playback from a different point. To sidestep this problem, both the
manual and the Bornemark web site (www.bornemark.se) suggest that you should
open Groove Agent as a multi-channel instrument and always make sure that it is
not the active (ie. Thru) track in Logic whilst playing back a Groove Agent track. In
my experience, the problem can still occur using this configuration — but it
certainly occurs more often if you don't follow that advice. Cycling a section of a
Logic arrangement which includes one of these dropout bars will make the
problem disappear on all but the first pass and, strangely, delaying the whole
Groove Agent instrument track by a few ticks can reduce the length of the
dropout, sometimes down to a single beat. Nothing else seems to affect it; not
buffer size, other audio activity, nor timing and density of automation data. And it
is as likely to happen in the middle of a section where Groove Agent is trundling
along repeating a pattern that was triggered 16 bars ago as it is when you hit it

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Steinberg Groove Agent 2

with a fill, a pattern change and an instruction to drop to half time and use the
sidestick with a hint of triplet shuffle, all at the same time. Nevertheless, using
note-triggering, Force Legato and Chase Notes, combined with Groove Agent
running as a multi-channel instrument and deselected as the MIDI Thru
instrument, I have gone up to three days without a dropout! I'd hesitate to call it a
cure, but it's looking like an acceptable workaround.
Version 2 does offer a workaround for
Groove Agent's inability to send MIDI
note data to a non-Steinberg host in the
form of the 'Write to File' facility. As
Martin Walker says, if you flip the 'Live <
> File' switch under the Setup lid, the
plug-in is supposed to generate a MIDI
file of its note data that you can import
into the song for editing. In practice, I
found this to be a rather hit-and-miss
affair in Logic. You can't determine
where the file goes or what it is called,
and sometimes I got one and sometimes
I didn't. Sometimes it would only appear
when I quit Logic, and other times not at
all.
Once you've established a working
method that suits you, I do recommend
making use of Logic's Aux Environment
objects to give independent control over
Groove Agent's output groups,
separating out kick, snare, hi-hat, and
perhaps virtual overheads (toms and
cymbals) for individual treatment. Logic
7 creates two Aux objects by default, so
you will need to create some more (by
selecting 'Aux' from the Audio objects list
under the 'New' menu in the
Environment) and then set Groove
Agent's outputs as the Aux objects' input
and your audio mix output as their
output (see the screenshots below). Selecting (right) inputs to Aux objects
from Groove Agent and (far right)
Note that Groove Agent numbers its
individual outs from Groove Agent to
outputs from one to eight (each one a Logic.
stereo output), while the Aux objects
number them from one to 16. Once
you've done that, you can insert individual processing into the different streams,
treating the snare and toms to EQ, compression and a reverb, say, while the kick
and hi-hat remain dry. You can also send Groove Agent's own ambience to a
separate output for even more mixing control.
I must also add a caveat for Logic users to Martin's thoughts on Steinberg's new
copy protection system (see overleaf). If you save a Logic song with Groove
Agent 2 in it and then try to open it without the Steinberg key plugged in, not only
will the song not open, but Logic will crash. Hardware key or no, Groove Agent
really should fail a little more elegantly than this — if the key isn't present, why
can't it simply disable itself within the song like most other plug-ins?
One final anomaly; Groove Agent occasionally sounds as though it is gently

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Steinberg Groove Agent 2

phasing or flanging, almost as if two versions of it are running at the same time,
although my Mac's processor metering proves that this cannot be the case. This
seems to happen most often during fills (maybe it is just more audible during
busier passages), and suggests perhaps that some samples are being triggered
twice. 'Suggests'... 'maybe'... 'sometimes'... I accept that that this all sounds
somewhat unscientific, but it is almost impossible to be definitive about something
as seemingly random as these phenomena. In the end, I just got on with using it
and accepted the possibility of having to edit a couple of bars of the final audio to
excise any compromised bits and fill in any gaps. On the whole, I still consider the
rewards of using Groove Agent to be worth the effort — although I would entirely
understand if others didn't, for it manages to be simultaneously the most
stimulating and frustrating software tool in my studio. Dave Lockwood

Published in SOS May 2005

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 Tascam FW1082

In this article:
Installation
Tascam FW1082
Front Panel Firewire Audio & MIDI Interface / Control Surface (Mac/PC)
Supported Applications Published in SOS May 2005
Rear Panel Print article : Close window
Bundled Software
Reviews : Effects
In Use

Tascam FW1082 £639

pros
Easy installation. Joining the growing market for one-box devices
High-quality multi-channel combining audio and MIDI interfacing with control-
audio, plus comprehensive
MIDI I/O. surface functionality, Tascam's latest Firewire unit
Faders are very responsive, might be all you need for multitrack recording and
and quiet when moving. mixing.
Cubase LE is a bonus.
cons
Multi-purpose controls can Paul Sellars
be a bit confusing to begin
with.
The Tascam FW1082 is a combined
A more comprehensive list
Firewire audio and MIDI interface and
of compatible applications
would be useful. mixer-like control surface for Digital
Audio Workstation (DAW) applications
summary
such as Steinberg's Cubase and
The Tascam FW1082 is a
MOTU's Digital Performer. According
well-built, logically designed
device, which provides high- to Tascam, it's intended to be 'the
quality audio I/O plus effective nerve centre of your digital audio
and comfortable hands-on environment', and it certainly looks the
control over your DAW. Hard part: a solidly built device, boasting
to fault.
banks of illuminated buttons, numerous
information knobs and nine motorised faders.
Photos: Mike Cameron
£639 including VAT.
Tascam +44 (0)1923 The FW1082 is not just a pretty face, however. Its audio facilities support 24-bit
438880.
recording and playback at sampling rates up to 96kHz, and allow for 10
+44 (0)1923 236290.
simultaneous input channels (eight analogue, two digital). In addition it features a
Click here to email
2 x 2 MIDI interface, basic analogue audio mixing, and a versatile programmable
www.tascam.co.uk
control surface. Steinberg's Cubase LE is also included, providing a complete
www.teac.co.jp multitrack audio and MIDI recording package (see the 'Bundled Software' box for
more details).
Test Spec
Tascam FW1082 software
version 1.40.
Athlon 1.8GHz PC with
Installation
512MB RAM, running

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Tascam FW1082

Windows XP Service Pack 2. Installing the FW1082 is straightforward. The necessary drivers are supplied on a
Steinberg Cubase LE version CD-ROM, and after running the installer program it's simply a question of
1.07. rebooting your machine and attaching the device. Under Windows (2000 or XP)
the FW1082 supports the ASIO and WDM (MME) standards. Under Mac OS X
(10.2.8 or later) Core Audio and Core MIDI are supported.

Full marks to Tascam for supplying proper, printed documentation. The user
manual and a useful setup guide are both presented on paper, which makes the
business of finding your way around a lot easier. Some other useful documents
(as well as duplicates of the printed booklets) are supplied as PDF files on the
installation CD, but all the most important stuff is covered in print.

The FW1082 communicates with the host computer via the IEEE 1394 Firewire
protocol. All recent Macs will have suitable connectors already built-in, and PC
Firewire support is becoming more universal. It should be pointed that the
FW1082's manual specifically warns against connecting the device to the
smaller, four-pin variety of IEEE 1394 connectors found on some laptops,
recommending that the larger six-pin type be used instead. I experimented very
briefly with attaching the FW1082 to my laptop's four-pin connector, and didn't
encounter any immediate, obvious problems. Nevertheless, I would be inclined to
heed Tascam's warning, and to use a computer with a six-pin connector for all
serious work. For the purposes of this review, I installed an inexpensive Belkin
PCI IEEE 1394 card in my desktop PC, and it worked without a hitch.

Once the drivers are installed, a software control panel is available, from which
it's possible to adjust various settings, including ASIO buffer size and sample
rate. The control panel can be launched either from within your DAW software, or
simply by pressing the Control Panel button on the FW1082's front panel.

Front Panel

The FW1082's front panel resembles a conventional mixer in many respects. At

the top is the analogue input section, with gain Trim knobs for each of the unit's
eight analogue inputs. Each knob is accompanied by a pair of LED indicators.
The first of these lights up green to indicate the presence of a signal, while the
second lights up red to warn when the input is overloaded (the warning threshold
is adjustable).

Over in the right-hand corner are three 'mode' keys, which are used to switch the
FW1082 between its three different modes of operation. The precise functions of
many of the FW1082's other controls depend upon which mode is selected. In
Computer Control mode, the FW1082 closely integrates with your computer and
DAW software. All fader and knob movements and button presses are
transmitted to the computer, and interpreted by the software. Data can be sent
back by the computer to adjust fader positions, or illuminate LEDs. Several
different control protocols are available — see the 'Supported Applications' box
for more details.

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Tascam FW1082

In MIDI Control mode, the FW1082's

buttons, knobs and faders can be used
to control external MIDI devices, and
MIDI-capable software applications.
The FW1082 has two physical MIDI
output sockets, and two 'virtual' output
ports which are visible to the host
computer. MIDI data is sent to the
virtual ports via the Firewire
connection. By default, a reasonable
set of MIDI controllers is already
assigned to the FW1082's controls, but The FW1082's faders are all touch-sensitive,
motorised devices.
these are all editable via the software
control panel.

In Monitor Mix mode, the FW1082 can be used as a straightforward audio mixer.
This allows for easy, zero-latency monitoring of the input signals, and the main
audio outputs from the DAW software — ideal for overdubs. You can choose
between monitoring only the inputs, only the DAW outputs, or both together.

The lower half of the front panel is dominated by the main channel controls. Each
of the eight channels has its own fader (a ninth fader controls the master level),
illuminated mute and solo buttons, a Select button, and a red LED to indicate
recording status. In Computer Control mode, pressing a Select button simply
selects the relevant channel in the DAW software. Pressing the red Rec button
and the Select button together arms that channel for recording. In Monitor Mix
mode, the Select button causes the selected channel's pan position to be
indicated by the channel record LEDs: channel 1's LED indicates hard left,
channel 8's LED indicates hard right, while channels 4 and 5 illuminated together
indicates centre. In MIDI Control mode, each Select button can be programmed
to send a control message of your choice.

Above and to the right of the channel

controls are four encoder knobs, and
several associated buttons. In
Computer Control mode, these knobs
can be used to adjust pan and EQ, and The FW1082 provides eight analogue inputs,
aux send levels for the selected four with phantom-powered mic preamps.
channel — the exact implementation
may vary depending on the DAW software. In MIDI Control mode, they can be
programmed to send continuous controller messages, while in Monitor Mix mode,
the bottom knob serves as a pan control for the selected channel, and the others
do nothing.

The FW1082's different modes and multi-function buttons and knobs can be a bit
confusing to begin with, but I've probably made them sound more complicated
than they are. The front-panel layout is quite logical, and with a little practice it
quickly becomes second nature.

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Beneath the encoder knobs is a collection of fairly self-explanatory transport

controls, which have the same functions in all three modes. There are
straightforward rewind, fast-forward, stop, play and record buttons which mirror
the DAW software's transport controls, together with a dial for scrubbing
backwards or forwards through tracks.

There's also a pair of bank selection keys, which can be used to switch the
channel controls between banks of eight DAW channels. For instance, if your
DAW project has 32 tracks in total, you can use these keys to switch between
four groups of eight channels. The motorised faders really come into their own
here, swiftly updating their positions to reflect the selected group's settings. Other
useful buttons include Locate, Set, In and Out keys, which can be used to move
location markers and set punch-in points in DAW software, and four cursor-key
buttons for navigating around other on-screen parameters. Finally, there are
knobs for setting monitor and headphone levels, plus LEDs to indicate MIDI
activity, Firewire connectivity and so on.

Supported Applications
A device like the FW1082 is really only useful in conjunction with suitable DAW
software, and Tascam have attempted to make it compatible with a variety of
different applications. When in Computer Control mode, the FW1082 supports
several different control protocols. 'Native Protocol' is the default, and this can be
used with Cakewalk's Sonar and MOTU's Digital Performer, via special software
plug-ins for each. These plug-ins are included on the installation CD, and in both
cases there are detailed PDF installation guides explaining how to get things up
and running.
'Mackie Emulation Protocol' allows the FW1082 to imitate the control messages
sent by Mackie's Mackie Control device. Theoretically, therefore, any application
which supports the Mackie Control should be able to understand the FW1082. In
fact, the special 'Cubase LE' protocol built into the FW1082's software control
panel appears to be a variant on the Mackie Emulation Protocol, since activating it
requires you to add a Mackie Control device in Cubase's preferences.
Presumably the same applies to the full Cubase SX, should you have it.
'HUI Emulation Protocol' is much like Mackie Emulation Protocol, except designed
with a different Mackie device in mind: the Mackie HUI (Human User Interface).
Neither the FW1082 manual nor the Tascam web site makes specific claim to
support any other applications apart from Cubase LE, Digital Performer and
Sonar, although this may change. Nevertheless, if your preferred application is not
one of these, but supports either the Mackie Control or Mackie HUI (or, as in the
case of Logic, both), it seems reasonable to assume that it will be possible to get
it working with the FW1082. That said, you should certainly ask your local Tascam
dealer for a demonstration before parting with any money, just to be sure it works
as you want it to.

Rear Panel

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The rear panel is where the FW1082's various sockets and connectors can be
found, and there's no shortage of these. Two MIDI In and two MIDI Out ports are
available, and there are two Firewire ports, although the manual advises against
daisy-chaining Firewire devices. Each of the eight input channels has a
balanced, line-level quarter-inch jack socket, and channels 1 and 2 also have
quarter-inch TRS (tip, ring, sleeve) insert sockets, which can be used to insert an
external processor such as a compressor or EQ into the signal path. The input
for channel 8 offers switchable impedance suitable for DI'd guitars and basses,
while channels 1 to 4 provide XLR connectors and built-in microphone preamps.
These sockets can also deliver 48 Volts of phantom power to any mics that
require it.

The only analogue outputs are a pair of monitor outputs on quarter-inch balanced
jacks, and a headphone socket which mirrors these monitor outputs. Digital I/O is
provided by a pair of stereo co-axial S/PDIF sockets, on standard RCA
connectors. The digital output can be set (via the software control panel) to
simply mirror the analogue monitor outputs, or it can be used independently. A
further jack socket allows you to use a footswitch to punch in and out of record
mode.

Bundled Software
Tascam have bundled a couple of useful, mid-range software applications along
with the FW1082 hardware: their own Gigastudio 3 LE, and Steinberg's Cubase
LE. Gigastudio LE is a 64-voice version of the popular, hard disk streaming
software sampler application for Windows XP. It offers many of the same features
as the full version including Rewire support and VST plug-in capability, and
represents a nice bonus for Windows users. No Mac version is available.
Cubase LE, on the other hand, is available for both Windows and Mac OS X. It's a
cut-down version of Steinberg's flagship Cubase SX sequencer, and although the
bundled version is based on version 1 of SX rather than the current version 3, it's
still a powerful audio and MIDI recording, editing and sequencing package in its
own right. Up to 48 audio and 64 MIDI tracks are available, and audio tracks can
take full advantage of the FW1082's 24-bit, 96kHz capabilities. It also includes
Rewire and VST plug-in support, reasonable score editing and printing features,
and a very nice time-stretch processor for audio parts. Hardware control surfaces
including the FW1082 are, unsurprisingly, supported.
An upgrade path to Cubase SX is available, if and when you feel you've outgrown
the limitations of Cubase LE.

In Use

For all its complexity, the FW1082 is easy to get along with. The analogue signal
path is clean and clear, without a hint of noise or interference. The mic preamps
sound good, with no obvious coloration. Monitor Mix mode makes zero-latency
monitoring easy, but I also found I was able to get sufficiently low latency out of

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the FW1082's ASIO drivers that monitoring via software could usually be
managed quite comfortably.

Integration with Cubase LE is seamless and easy, and recording and playback of
fader movements is a no-brainer. The shuttle dial works well, and provides a
convenient way to quickly seek back or forwards through a track. (The faders
ignore any automation data while the dial is turning, and update themselves
when the song position indicator comes to rest, which seems sensible.)

A hardware control surface is arguably

only as good as its faders, and the
FW1082 doesn't disappoint. The eight
60mm channel faders and single
master fader are touch-sensitive,
allowing the most minute fingertip
adjustments to be made, predictably
and reliably. No unwanted twitching or
sticking here; your virtual faders go
where you put them. All nine faders are
motorised, well engineered, and very
quiet, even when switching banks and
sending all eight channels from one
extreme to the other. There's a soft The control panel is straightforward, but does
everything you need it to.
click as the servos activate, before the
faders flick up or down to their new
positions with barely a sound. On slower automated fades, they're as good as
silent.

In the past I've worked with cheaper control surfaces, and found myself still
habitually reaching for the mouse to make mixer adjustments. Not so here. The
FW1082 is very comfortable and responsive, and with judicious use of the
various function keys it's possible to get plenty of work done — at least in terms
of recording and mixing, if not editing — without touching your computer. Once or
twice I found myself wishing that Tascam had fitted independent EQ and aux
send knobs for each channel, but to do so would have required a lot more space
on the front panel, and doubtless bumped up the manufacturing costs and retail
price considerably.

MIDI Control mode is a bonus, and some users may seize upon the opportunity
to start programming custom controllers for their favourite hardware or software.
Many, I suspect, will be more than satisfied with the basic DAW control functions.

It's hard to find fault with the FW1082. It's easy to install, quite straightforward to
use, and works well. In the course of the couple of weeks I was testing it, it
sprung no nasty surprises on me, and gave the impression of being very solidly
built and dependable. All in all, it does a convincing job of providing high-quality
multi-channel audio, comprehensive MIDI I/O, and a very usable control surface.
If you're in the market for a hardware DAW controller, the FW1082 deserves
serious consideration.

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Tascam FW1082

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Terratec Phase 88 Rack FW

In this article:
PCI And Firewire
Terratec Phase 88 Rack FW
Overview Firewire Audio Interface (Mac/PC)
The Phase 24 Published in SOS May 2005
Installation Print article : Close window
Terratec Phase 88 Rack FW
Reviews : Computer Recording System
Brief Specifications
Summing Up

Terratec Phase 88 Rack

FW £360
Terratec's range of affordable audio interfaces now
pros includes two attractive Firewire options: the eight-in/
Excellent value for money.
out Phase 88 Rack FW and the two-in/four-out Phase
Individual gain controls on
every input. 24 FW.
Good subjective audio
quality, particularly at 24-
bit/44.1kHz. Martin Walker
cons
Relatively poor 96kHz Terratec's Phase product line aims to offer professional features at bargain
performance. prices, and with the introduction of the Phase 88 Rack under review here, it now
Hum on analogue input encompasses PCI, Firewire and USB formats. It offers the popular arrangement
channel one. of eight analogue inputs and eight analogue outputs, plus a healthy complement
Confusing list of differences of digital I/O, all for a very competitive £360.
between the Firewire and PCI
versions.
summary Although it's based on the Phase 88 card I reviewed in SOS April 2004, this time
round the I/O is housed in a very professional-looking rack module rather than a
While I have a couple of
reservations about the audio desktop or drive-bay-mounted breakout box, and it also offers new features, such
quality of the Phase 88 Rack as balanced analogue inputs and outputs, gain controls for all eight inputs, two
FW, it certainly provides mic preamps, two MIDI inputs and outputs instead of one, plus the addition of
excellent value for money, word clock in and out. For just £60 more than the retail price of the Phase 88, this
and I suspect it will sell and seems excellent value for money — and for those who don't need so much I/O,
sell.
Terratec also offer the even more affordable Phase 24 FW (see box overleaf).
information
Phase 88 Rack FW
£360; Phase 24 FW £165.
Prices include VAT.
SCV London +44 (0)20
8418 1470.
+44 (0)20 8418 0624.
Click here to email
Photos: Mark Ewing
www.scvlondon.co.uk
The Phase 88 Rack, in both PCI and Firewire variants, offers eight analogue ins and outs
www.terratec.com at a highly affordable price.

Test Spec

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Terratec Phase 88 Rack FW

Terratec Phase 88 Rack FW

Windows XP driver version
PCI And Firewire
1.19.
Hardware: Intel Pentium 4C Hopefully, not too many musicians will get confused between the existing Phase
2.8GHz processor with 88 and the new Phase 88 Rack, but things get more complicated: the Phase 88
Hyperthreading, Asus P4P800 Rack is itself available in two different models, which outwardly look almost
Deluxe motherboard with Intel identical. The packaging simply states 'Alternative PCI or Firewire interface', but
865PE chip set running
you don't get both options in the same box — whereas the rack unit of the PCI
800MHz front side buss, 1GB
DDR400 RAM, and Windows version has a D-type connector on the back panel and is bundled with the
XP with Service Pack 2. associated Phase 88 PCI card interface, the Phase 88 Rack FW sports two
Firewire connectors and a front-panel Firewire logo.
Tested with Cakewalk Sonar
4.0, NI Pro 53, Rightmark
Audio Analyser 5.4, Steinberg If you buy the PCI version you can
Cubase SX 3.0 and Wavelab
retrofit Terratec's optional Firewire
5.00a, Tascam Gigastudio 160
v3.04. interface (and vice versa according to
the manual), but the box I received
showed no indication of which model it
contained. There are quite a number of
differences between the two, as I'll
explain.

The PCI version has drivers that With a comprehensive monitor mixer, clear
support Windows 98SE, Me, 2000, XP display of clock settings, and software
and Mac OS X Core Audio/MIDI, and switching of mic/line input sensitivity, the
its S/PDIF I/O is provided in co-axial Terratec Control Panel provides
comprehensive control over the Rack FW.
format on the PCI card. This card is
identical to the one bundled with the
original Phase 88 model, which means that you can internally synchronise the
Phase 88 Rack PCI with other Terratec products including the Phase 88 and
EWS 88MT/D and EWS Mic 2/8, using the on-card Sync In and Out connectors.
The PCI drivers support up to four cards, so you can achieve up to 40
simultaneous audio inputs and outputs. However, if you want to switch between
word clock or S/PDIF external clock, you have to do it using a jumper on the PCI
card — not much fun once it's fitted inside your computer!

Meanwhile, the Firewire version only supports Windows XP SP1 and higher or
Mac OS 10.3.4 or higher, cannot at present be cascaded to expand the I/O
totals, and loses the card's co-axial S/PDIF I/O in favour of a pair of optical
Toslink connectors on the rack's rear panel. This time you can freely switch
between word clock and S/PDIF clock using the Control Panel utility. Finally, if
you power up the unit without a Firewire connection, the Rack FW version can be
used as a stand-alone A-D/D-A converter box, and you can even alter its digital
mixer settings in real time using MIDI controller data, instead of using the
software Control Panel.

Overview

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We received the Firewire version of the Phase 88 Rack for review, although as I
noted, there was nothing on the box to indicate which version was inside. It was
bundled with a generous four-metre-long six-pin to six-pin Firewire cable, a four-
pin to six-pin adaptor for laptop use, and associated 12V AC wall-wart power
supply — like many others, this interface can't be powered from the Firewire
buss.

The front panel contains a pair of XLR mic input sockets, a pair of five-pin DIN
MIDI In and Out sockets, eight input gain controls with associated signal and
clipping LEDs, a button providing global 48V phantom power for the mic inputs,
and an LED power indicator. The rear panel houses 16 quarter-inch jack sockets
for unbalanced/balanced operation of the eight line inputs and outputs, the
second MIDI In and Out, S/PDIF optical in and out, a 15-pin connector for the
supplied word clock breakout cable which terminates in a pair of BNC sockets,
the PSU input socket, and the interface panel, which is different for FW and PCI
versions.

On the Firewire review model this panel houses an identical pair of Firewire
sockets, plus four LEDs to indicate successful IEEE 1394 communication, word
clock selected, external clock sync (from S/PDIF or word clock), and valid sync.
My only layout quibble is that these indicators would be much better placed on
the front panel; thankfully, the software utility displays successful sync each time
you change sample rate, which does provide reasonable feedback.

The Phase 24
Along with the Phase 88 Rack FW
described in the main text, Terratec
also sent their Phase 24 FW, a
compact interface with a single pair
of balanced analogue inputs and
outputs on the back panel, plus S/
PDIF co-axial in and out, single
MIDI In and Out, and a further
unbalanced stereo output on the
front panel with thumbwheel level
control, which can either be used for
headphones or another stereo line
output. It's housed in an attractive
and robust metal case and bundled
with its own Velcro-sealed carry-
case. Unlike its stablemate, the Unlike its larger brother, the Phase 24 FW
Phase 24 FW can be powered from can be powered from the Firewire buss.
the Firewire buss, but a 9V AC wall-
wart is also supplied for those whose computers only offer four-pin Firewire ports.
Unlike the Phase 88 Rack FW, the 24 FW's converters support sample rates up to
192kHz, and it provides a typical 109dBA signal-to-noise ratio for its A-D
converters — 9dB quieter than the Rack — and 111dBA for the D-A converters,
again slightly better than its big brother. Its drivers are also totally different,

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Terratec Phase 88 Rack FW

although they too support only Windows XP SP1 or Mac OS 10.3.4 or later. The
Control Panel provides a useful relay-switched low/mid/high sensitivity option for
the inputs to cope with different signal levels, and a similar range of monitor mixer
settings to its bigger brother.
I confirmed the 109dBA dynamic range at 44.1kHz sample rate using Rightmark's
Audio Analyser, and as expected, this dropped slightly to 103dBA at 96kHz and
101dBA at 192kHz, although the analogue bandwidth didn't extend at all at these
higher rates, with a -0.5dB point just above 20kHz in each case. Nevertheless, my
usual double-blind listening tests against my own Emu 1820M and Echo Mia soon
showed that once again the Emu's low-jitter clock provide the most focused and
warm sound, closely followed by the Phase 24 (giving it similar audio quality to the
192kHz-capable converters of both M Audio's Audiophile 192 and ESI's Julia),
with the Mia slightly behind with a touch of harshness from its its AK4528
converters.
Driver quality seemed good too: as with the Phase 88 Rack FW I managed a
2.2ms ASIO latency in Cubase SX 3, while the GSIF driver once again worked
faultlessly, even when I allocated the same pair of outputs to each application.
The MME-WDM drivers were again multi-channel, but performed well. Overall, I
was impressed with this robust and straightforward interface, which provides
excellent audio quality for the price, and as many ins and outs as many musicians
will ever need for either a desktop or laptop setup. It looks good, sounds good,
and is easy on the pocket — what more could you ask?

Installation

As usual, I installed the drivers for Windows XP, and for the first time ever found
that the manufacturer had supplied me with a CD-R containing drivers newer
than those on the web site. Terratec are unusual in providing their GSIF drivers
for Gigastudio owners as a separate install option, but I had no problems getting
the MME-WDM, ASIO and GSIF drivers installed, along with the associated
Control Panel utility.

Although the software installation appeared to be successful, the review unit

initially refused to record or play back any audio at all. After much investigation I
eventually traced the problem to a bad connection between the circuit boards,
which was resolved by re-seating the digital board into the main circuit board.
Terratec subsequently confirmed that this was due to a design flaw in early
production units, which they have now solved by adding an additional bracket
inside the case. If you find that your unit is affected by this problem, contact your
country's Terratec distributor and they will replace it with a new one, free of
charge. I also encountered some hum on analogue input one — not enough to
be audible at normal listening levels, but significantly worse than the other seven
inputs. This might have been a fault specific to the review unit, but shouldn't
happen.

Opening the case did let me confirm that the Phase 88 Rack uses the same
AK4524 converters as its EWS88MT and Phase 88 predecessors. The Control
Panel of the Phase 88 Rack is likewise almost identical to that of the Phase 88,
apart from extra options to switch inputs 7/8 between the rear-panel Line sockets

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Terratec Phase 88 Rack FW

and the front-panel Mic sockets. However, for the Firewire series Terratec have
streamlined various features.

The largest area is devoted to the Digital Mixer, used for monitoring purposes,
with level sliders, mute and stereo link buttons for the eight analogue inputs, the
stereo digital input (greyed out until a valid signal is detected), and for any two of
the eight analogue or two digital WAV playback channels. These signals are
mixed together and then sent to the Master section, which has its own master
fader, mute and stereo link buttons, while its routing selector lets you decide
where the stereo output of the mixer is routed, chosen from the four stereo pairs
of analogue outputs, the stereo digital output, or 'Mixer Off', which leaves all
physical outputs routed by default to the appropriate WAV playback channel.

The peak-reading level meters of the

Phase 88 and Phase 88 Rack PCI are
omitted on the Firewire rack version,
as are the multi-card selections, both The Phase 88 Rack becomes a Firewire
presumably because of bandwidth device thanks to the inclusion of an interface
considerations. The Master Clock board with two Firewire sockets and some
section provides a useful extra S/PDIF sync LEDs.
Detection indicator, but its Internal/
External, sample rate, and external clock rate detection displays are almost
identical.

Terratec's previous EWS 88MT and Phase 88 have suffered from ground-loop
problems in some setups, and one of the benefits of the Phase 88 Rack's fully
balanced analogue I/O is that for those with compatible gear, most ground-loop
problems disappear. I certainly found it quiet and relatively clean, and despite
using exactly the same converters, the Phase 88 Rack FW managed a dynamic
range of 100dBA at 24-bit/44.1kHz, exactly in line with the manufacturer's spec,
and 4dB better than the Phase 88. At 96kHz both models had slightly worse
background noise levels of 99dBA, but the dynamic range of the Rack model
dropped to 93dBA due to the presence of a crop of low-level spectral lines above
1kHz. Its 96kHz frequency response was also -0.5dB down at 20kHz compared
with the 44kHz of the Phase 88, although at the low end it was more extended,
being only -0.2dB down at 4Hz. Stereo crosstalk figures for the Rack model
measured about 4dB better, although 50Hz hum levels were about 8dB higher —
not that this will usually be audible.

So how did it sound in subjective terms? My double-blind listening tests of the

Phase 88 Rack FW against my own Emu 1820M and Echo Mia cards produced
some interesting findings. I consistently picked out the Phase 88 Rack as having
the warmest sound, possibly because its frequency response is the most
extended at the low end, albeit by only 0.05dB at 20Hz. I could also pick out the
slight harshness of the Mia, and the most natural sound of the Emu. Overall, the
Emu came out on top yet again — as you might expect, given its more expensive
converters and low-jitter clock — but I preferred the warm yet clear sound of the
Phase 88 Rack to the Mia, even though the latter has a significantly better
dynamic range. Technical specs don't always provide the final word.

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Terratec Phase 88 Rack FW

I had no problems using the mic inputs, although there's a hefty click when you
switch between mic/line sensitivity using the Control Panel, so turn your monitors
down first. The S/PDIF and MIDI I/O also worked fine. Driver quality seemed
good too — I didn't quite achieve the lowest available ASIO latency setting of
2.0ms on my PC when running Cubase SX 3, but did manage a very close
2.2ms, and the GSIF driver also worked faultlessly with Gigastudio, even when I
allocated the same pair of outputs to each application.

The MME-WDM drivers are multi-channel, like those of many recent interfaces.
This makes them more difficult to use with applications that prefer their drivers to
show up as multiple stereo pairs, but they managed a very typical 45ms Play
Ahead setting in NI's Pro 53, while the Direct Sound drivers achieved a better
than average 25ms. Sonar's lowest successful Effective Latency with the WDM
drivers proved to be a reasonably good 10.2ms, but in ASIO mode I once again
achieved 2.2ms.

Terratec Phase 88 Rack FW Brief Specifications

Sample rates: 32, 44.1, 48, 88.2 and 96 kHz from internal clock.
Analogue inputs: eight, balanced/unbalanced TRS quarter-inch jack, 33dB gain range,
inputs 7 and 8 switchable to balanced mic sensitivity on XLR sockets, with optional global
+48V phantom power.
Analogue outputs: eight, balanced/unbalanced TRS quarter-inch jack.
Digital I/O: S/PDIF in and out on Toslink optical (FW model) or phono co-axial (PCI
model), word clock in and out, two MIDI Ins and Outs.
A-D converters: 24-bit 64x oversampling (part of AK4524 codec chip).
Input SNR: 100dBA typical at 48kHz sample rate.
D-A converters: 24-bit 128x oversampling (part of AK4524 codec chip).
Output SNR: 109dBA typical at 48kHz sample rate.
Frequency response: not stated.

Summing Up

Terratec always seem to provide great value for money with their products,
although sometimes the clever shortcuts they take to keep their prices keen can
end up being confusing to the user. I was, for instance, a little puzzled about the
differences between the PCI and Firewire variants, and slightly concerned by the
problems I had getting the review model to work. Nevertheless, the Phase 88
Rack FW provides good subjective audio quality, even though its AKM 4524
converters are outclassed by many more recent designs (including its cheaper
Phase 24 stablemate!), and £360 is an excellent retail price for an eight-in/eight-
out analogue product.

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Terratec Phase 88 Rack FW

If you're prepared to pay a little more, you could consider the greater number of
inputs provided by Guillemot's Hercules 1612 FW or the ADAT expansion
potential of M Audio's Firewire 1814, but the closest competitor to the Phase 88
Rack FW must be Edirol's FA101, now selling on the street at around the same
price. It also offers fully balanced analogue I/O with two mic input options, but in
its favour one of these can be switched to high-impedance instrument mode for
use with guitars; it also has an additional headphone output, supports 192kHz
sample rate, can be powered from the Firewire buss, and its case is half the size,
making it more suitable for location work. On the other hand, the Phase 88 Rack
FW offers twice as many MIDI inputs and outputs, handy individual level controls
on each analogue input, and it looks more impressive when mounted in a rack!
Only you can decide which feature set is more appropriate to your needs —
Terratec have just made your final choice just that little bit more difficult.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Ursa Major Space Station

In this article:
Connectivity & Internal
Ursa Major Space Station
Processing Digital Reverb & Effects Processor
Control Layout Published in SOS May 2005
In Use Print article : Close window
Sounds Of The '70s
Reviews : Effects
Ursa Major Space
Station £899
pros
Accurate recreation of the The original 3U monster that was the SST282 Space
original SST282 algorithms.
Station has been reissued by Seven Woods Audio as
Vintage sound character
and effects. this small desktop unit. Not content merely to offer all
Attractive control surface. the classic sounds, though, the new incarnation has a
Sophisticated new Room number of fresh tricks up its sleeve...
program.
cons
Only digital I/O. Hugh Robjohns
No preset memories.
Imposing handbook. The Ursa Major Space Station — the original
No block diagram to explain SST282 3U rackmounting monster — was launched
controls. in 1978 as one of the first generation of digital
summary reverbs, competing directly with the classic EMT
An accurate recreation of the 250. The Space Station was a multitap delay-based
original Space Station's device providing echo, ambience, and reasonable
effects using modern reverb effects, and it remained on sale for the best
technology in a very compact
part of a decade, until technology advances allowed
and elegant package.
Supplemented with a brand- far more sophisticated products at much lower
new and highly sophisticated prices.
Room algorithm, plus updated
wide-bandwidth versions of
the original SST echo and To remove any confusion, let me just clarify that
reverb programs. Ursa Major was the name of the company (based in
information Belmont, USA) which was set up by Christopher
Moore to produce the Space Station. It quickly
£899 including VAT.
became a classic reverb — more than a reverb really — and its contribution
DACS Audio +44 (0)191
438 2500.
became an identifiable element of much music of the time, especially as a guitar
+44 (0)191 438 2511.
and vocal effect. In fact, the mystique of the Space Station has remained, and
Click here to email
Chris Moore, working with Princeton Digital, has already produced a software
replica of the Space Station as a TDM plug-in for the Pro Tools platform.
www.dacs-audio.com
Princeton Digital have been involved in the recreation of several vintage digital
www.seven
products, including the Eventide SP2016 reviewed back in SOS May 2004.
woodsaudio.com

The new hardware incarnation of the Space Station has been produced by a
company called Seven Woods Audio, again with the direct involvement of the

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Ursa Major Space Station

originator Chris Moore. It has acquired a new model number along the way —
SST206 — and has shrunk from a 3U rackmount beast to what initially appears
to be a remote control but is, in fact, the entire unit! It has also gained a new
state-of-the-art algorithm to supplement the original echo and reverb effects. The
handbook includes a note from Chris Moore to explain that the new Room
program is the 'best reverb that I know how to create with today's best hardware'.

Connectivity & Internal Processing

The heart of the new Space Station is a single 140MHz Motorola DSP chip
programmed with faithful recreations of the original Space Station's algorithms,
some updated versions, plus the new Room reverb. The unique aspect, though,
is that the unit is packaged in a slim panel with wooden side cheeks, which is
designed to sit on a desk. It measures just 165 x 127 x 150mm (hwd) and weighs
next to nothing, with a four-metre connecting cable which terminates in a pair of
XLR connectors and a compact universal mains power supply. The latter accepts
the usual IEC mains lead and operates on AC voltages from 90V to 260V,
consuming less than one Watt of power.

Audio connections are provided only in digital form, with stereo AES input and
output provided on flying XLR connectors. The SST206 is a stereo output device,
like the original, but where it differs is that it accepts a stereo source where the
original was a mono-input device. However, the stereo input is a convenient by-
product of adopting the AES input format, and the stereo input is summed to
mono to feed the delay/reverb processor. A dedicated Dry Level control allows a
percentage of the stereo source signal to be passed through to the stereo
outputs, if required.

The unit is optimised for use at a 48kHz sampling rate with 24-bit resolution. It
will also operate happily at 44.1kHz and even 32kHz — although the control
calibration will be inaccurate at these lower sample rates since the delay and
decay times are related to the sampling rate. Similarly, it can also be used at
88.2kHz and 96kHz (but only with the SST Reverb and Echo programs), with
proportionally shorter delay and decay times again.

Control Layout

The original Space Station (and its TDM plug-in version) was driven via a
combination of rotary controls and buttons which configured the delay line's
outputs and feedback paths in various ways. The fundamental design involved
24 separate delay-line taps. One was used for the echo effect, feeding back to
the input through a decay control. In the reverb mode, the delay time of 15 taps
was modulated before being fed back to the input. The last eight taps were called
the Audition Delay Taps and were configured in pairs to feed the processor's
stereo outputs. The buttons selected from 16 different tap configurations to

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Ursa Major Space Station

provide different effect characters and overall decay times.

The new version shares the same delay-line topology, but employs twelve rotary
controls to configure the unit: two black, three red, four blue (all with pointers and
calibrated from zero to 10), and two without pointers. This last pair select the
required program and delay tap pattern, which are indicated with LEDs. There is
also a small bar-graph meter to show the summed input level, calibrated for -30, -
15, -6 and 0dBFS levels. The controls are all clearly labelled in white against a
dark-grey background, making the unit very easy on the eye.

The first black control sets the input level, while a blue control underneath sets
the level of dry input signal sent to the outputs. The next black control sets the
echo delay time (and doubles as the pre-delay time control for the new Room
program). The red knobs adjust the overall decay with separate tweaks for the LF
and HF decay times. The decay time is essentially a feedback control, passing
some of the delay-line's output back to the input to extend the effective decay
time. The input to the delay line passes through a simple bass/treble equaliser
controlled with the LF and HF decay-time parameters, so these enable the
character of the reverb or echoes to be tailored to simulate dark or bright spaces.

The last four blue knobs control the

eight delay taps which generate the
stereo output signal. The actual delay
times associated with each pair of
delay taps are governed by the
Audition Delay Pattern, as set by one
of the plain black knobs. The options
include echo, two slap effects, two
delay cluster settings called 'fatty' and
'cloud', three space repeats, four A multicore cable protruding from the top of
the new SST206 Space Station transports
rooms, and four comb-filter effects. power and AES digital audio signals.

The tap outputs are controlled in pairs, with the odd-numbered taps feeding the
left channel and the even-numbered taps feeding the right channel. By varying
the levels of the different taps, the character of the effect can be changed
dramatically, and this is completely independent of the Echo Delay and Decay
Time parameters — the virtually unlimited combinations make this a very
versatile machine.

However, it is worth pointing out that the SST206 has no facility for factory or
user presets — you have to create and adjust the desired effect manually each
time you use the machine. For some, that will be a creative pleasure, while
others will find it an intellectual challenge, depending on their respective points of
view! Furthermore, the operation is not particularly intuitive at first sight, and the
new model lacks the block-diagram graphic of the original to help explain what
each control is doing. However, with a little practice and familiarity it soon
becomes easy to adjust the parameters as required, and it is rewarding to create
and shape the wanted effect.

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Ursa Major Space Station

The two original progams — the SST Echo and SST Reverb — each have two
variations, as mentioned earlier. The accurate original algorithms impose a 7kHz
bandwidth to the delayed signal with an 80dB dynamic range (11 bit), indicated
with a steady-mode LED. The updated version is indicated with a flashing LED
and provides a full 20kHz (or greater) bandwidth with 120dB dynamic range. It
also boasts less modulation noise in the reverb mode and an 'infinite delay' in the
echo mode.

The all-new Room program is a more conventional (in the modern sense) reverb
algorithm featuring a true stereo input. When in this mode some of the controls
take on new functions — Echo Delay becomes a Pre Delay, and the four output
tap controls adjust early-reflection delay time and level, reverb level, and room
size. The Delay Pattern control determines the length of the early reflections.
This algorithm uses the full power of the Motorola DSP to generate a very
sophisticated and natural-sounding reverb, which compares very well indeed to
the professional Lexicon and TC effects. However, the mode can only be used at
44.1kHz an 48kHz.

In Use

The supplied handbook takes quite a bit of reading, made harder by the dense
and uninviting layout. However, there is a lot of helpful information in there on
how to get the best from the SST206 — which is needed given the less-than-
intuitive controls. It is a great shame that the very clear and simple block diagram
that featured so prominently on the original could not be squeezed onto the new
control surface, and its ommision from the handbook is inexcusable. They say a
picture is worth a thousand words and that was never truer than in this case!

Hooking the Space Station up was simple enough. I used it in conjunction with a
Yamaha DM1000 console, configuring a pair of AES inputs and outputs from one
of the interface cards to act as an aux send and effects return. Since the device
is optimised for operation at 48kHz, I performed most tests at that sample rate,
although some material required me to use 44.1kHz, and I also ran a brief test at
96kHz. Although the parameters all become 10 percent larger at 44.1kHz, this
has little practical effect on the character of the processing. With double-sample-
rate operation, the new Room reverb mode is unavailable, of course, and all the
other parameters have half their original delay lengths — which does have a
significant knock-on effect for the sound and character of some effects,
especially the longer delays and reverbs, for obvious reasons.

It's not impossible to use the SST206 at 88.2kHz or 96kHz, but it becomes a lot
less versatile and flexible. As a result, I ended up using the slightly bizarre
combination of a D-A/A-D converter combination on the input with a sample-rate
converter on the output to allow me to operate the Space Station at 48kHz with
96kHz source material. Given that I wanted to use a particular effect with a 7kHz
bandwidth, the lack of 24-bit/96kHz integrity hardly seemed to matter!

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Ursa Major Space Station

Sounds Of The '70s

The sound of the SST206 is impressive, both in the original 'vintage' mode and
with the full bandwidth and dynamic range. There is something special about the
way the delay effects are generated and controlled that gives it a unique
character — perhaps it is the slightly imperfect and grainy quality. Similarly, the
SST Reverb mode has a quality that is identifiably 'vintage' yet eminently usable,
especially for ADT and flutter echoes on keyboard and guitar parts, and for
applying grungy, analogue-like delay effects to vocals.

The original Space Station was designed and used at a time when recording
engineers were technically minded and were used to having to configure
equipment by hand to create the desired effects. These days, I suspect 80
percent of home-studio users rely entirely on factory presets, particularly in the
case of reverb and delay effects — partly through laziness and partly because
the sophistication of modern effects processors makes manual adjustment a
complex business. However, the new SST206 demands personal creative input
to create and shape every single effect. Given the digital technology involved, it
does seem strange that factory and user presets are not available, and that
favourite settings therefore have to be noted on the supplied control template.

A technical understanding of what is going on under the panel is not essential for
creating interesting and appropriate effects, but it certainly helps. Having said
that, I dare say many will discover suitable effects simply through semi-random
knob-twiddling — the range and diversity of sounds available in the SST206 is
vast.

The new room reverb is a very classy effect, with a very lifelike quality. A good
example of this can be heard when you adjust the room size — as the size is
reduced the sound takes on more distinct room-mode colorations that really do
give the impression of a small, believable space. The ability to adjust the delay,
length, and level of the early reflections separately from the main body of the
decay allows a wide range of room characters to be simulated. More importantly,
however, the reverb can be tailored to sit nicely in a mix without clogging up the
space between instruments. I would rate this program to be the equal, on quality
grounds, of many of those found in devices like the Lexicon PCM90 and the
more sophisticated TC products, although the fact that each effect has to be
dialled in from scratch makes it a little less usable than those more familiar
references.

Overall, the Space Station is a well-designed, superb-sounding effects processor

with a unique and attractive profile. It is relatively expensive and functionally
limited in comparison with typical multi-effects processors, but it offers a range of
effects which have a unique and attractive sound character and are highly
customisable. It won't suit everyone — not least because of the need for hands-
on control — but if you like rich, interesting delay effects and vintage-style
reverbs, this is a fascinating product that delivers satisfying results.

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Ursa Major Space Station

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Yamaha Motif Rack ES

In this article:
A Quick Recap
Yamaha Motif Rack ES
About Studio Connections Workstation Synth Module
What's New? Published in SOS May 2005
The Benefits of USB Print article : Close window
Preset & User Memories
Reviews : Sound Module
Motif Rack & Motif Rack ES
Compared
The Arpeggiator
The PLG150AP Piano
Expansion Board Yamaha's original Motif Rack was a fine-sounding,
The Big Issue well-specified synth module, but it suffered from
Conclusions MIDI timing problems when reviewed in SOS. Two
Yamaha Motif Rack ES years on, we put the follow-up Motif Rack ES to the
£949 test...
pros
Does not suffer from the
Nick Magnus
MIDI timing problems
affecting the original Motif
Rack. When SOS reviewed the Motif Rack
An extremely versatile synth back in June 2003, we were left in a bit
module. of a quandary. Whilst being a great-
High-quality effects. sounding, versatile synth module with
Insert effects doubled to bags of creative potential, the Motif
eight in Multi mode.
Rack suffered from some serious
Multiple USB output ports Photos: Mike Cameron
timing problems. Specifically, this
allow up to 33-part
multitimbrality with PLG meant that the unit played out of time
boards installed. in response to incoming MIDI messages — not only relatively within itself, but
Bundled Studio Manager very noticeably when played from within a MIDI sequencer in parallel with other
and editing software allow full MIDI instruments (see the original review at www.soundonsound.com/sos/jun03/
integration with compatible articles/yamahamotifrack.asp for full details). Yamaha acknowledged the
DAWs.
problem, and undertook to find the cause and fix it. Changes were subsequently
cons made to the unit in a bid to alleviate the problem, and although matters were
No editing of parts at slightly improved, the problem was not solved beyond all reasonable doubt. Now,
element level from within the
Multi-part editor software.
nearly two years later, Yamaha have released an updated model, the Motif Rack
ES. The ES features a number of enhancements over the original Rack — but
No per-key editing of drum
parts within a Multi. have Yamaha successfully addressed the crucial issue of the MIDI timing?
summary
Two years ago I liked the
original Motif Rack's vibrant
sounds and its potential for A Quick Recap
expandability via PLG boards.
The many minor
improvements in the Motif
The Voice architecture of the ES has been carried over from the original Rack,
Rack ES, such as the details of which you can find in that June 2003 review (for more details of the

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Yamaha Motif Rack ES

doubling of simultaneously Motif concept, see the review of the Motif 7 keyboard in SOS September 2001,
addressable insert effects, viewable on-line at www.soundonsound.com/sos/sep01/articles/yamahamotif7.
have made the Rack ES an
even more versatile creative asp). Briefly, the Motif Rack is a 16-part multitimbral synth module, employing
tool. But the real winner is the Yamaha's long-established AWM2 sample-based subtractive synthesis, and
resolution of the original capable of playing a maximum of 128 notes of polyphony. Each complete synth
Rack's MIDI timing problems,
Voice (or Patch) consists of up to four split or layered Elements (think of these as
allowing me to give the Motif
Rack — in its ES incarnation, oscillators) and up to two different insert effects. Voices can either be played
at least — a thoroughly singly, or from within Performances or Multis. Performances layer up to four
enthusiastic thumbs-up. Voices together, either split, layered or a combination of both, on a single MIDI
information channel. When played from within a Multi, Voices can be assigned their own
£949 including VAT. MIDI channels for a full 16-part multitimbral sequencer-driven performance.
Yamaha-Kemble
Brochure Line +44 (0)1908
369269.
+44 (0)1908 368872.
About Studio Connections
www.yamaha-
music.co.uk The Studio Connections Initiative comes as a by-product of the long-standing
alliance between Yamaha and Steinberg (now cemented by the Japanese
conglomerate's recent purchase of the German firm). The initial aim is to integrate
hardware-specific editing applications within software DAWs using Yamaha's
Open Plug-in Technology (OPT) software format. In effect, this allows compatible
hardware devices to appear within the DAW, and be edited or automated as if
they were software plug-ins. All the hardware settings (or simply those for a single
device) can be stored along with the sequencer song data and recalled instantly.
This concept is referred to as Total Recall.
The software that makes this all
possible, Yamaha's Studio Manager
(currently at v2.1.2), is bundled on a
CD-ROM supplied in the box with
the ES. When installed, it can be
directly invoked from within an OPT-
compatible DAW, providing
transparent access to, for example,
a Motif ES voice editor, without
having to run the editor as a
separate, external application.
Additionally, devices' editing
applications can be operated
Yamaha's Studio Manager software — if you
remotely, meaning that a Yamaha have Steinberg's Cubase SX3, Nuendo 3, or
01X control surface, for example, Yamaha's own SQ01 sequencer, you can
could be configured as a hardware use them with this application to recall
control surface for the Rack ES. So settings on compatible hardware.
what products are able to make use
of this system? Currently, the DAWs
compatible with Total Recall are Steinberg's Cubase SX3 and Nuendo, while the
hardware devices currently supported are Yamaha's DM2000, DM1000, 02R96,
01V96, the 01X and the Motif ES keyboard and Rack synths (note that the mixers
are only compatible from OS v2 onwards).
Also bundled on the CD-ROM are the Motif ES Voice Editor and the Multi-part
Editor. At least they are both meant to be on the disk according to the manual —
only the Multi-part editor was included on the review disc. In case you suffer from
a similar problem, the missing ES Voice editor can be downloaded from www.

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Yamaha Motif Rack ES

yamahasynth.com, but if you're on a dial-up connection, be aware that it is a

substantial 29MB download!
The Multi-part editor, which provides on-screen editing of multitimbral
performances, or Multis, did not exist at the time of the original Rack review, and
appeared two months later, in August 2003. It's very welcome nevertheless, and
is now at version 2.1. Curiously, Yamaha's OPT-compliant SQ01 sequencer for
Windows, which was bundled with the original Motif Rack, is not included in the
Rack ES's software bundle either. This can be downloaded from www.yamaha.co.
jp/product/syndtm/dl/sq01.html, but it seems that you have to have installed all the
previous revisions in order to upgrade to the latest version (v2), so again, be
prepared for a lengthy download session! The software bundled with the Motif
Rack ES is compatible with both Windows XP and Mac OS X (10.3 or later), and
the USB MIDI driver is compatible with Windows XP/2000 or any Mac with a
Power PC processor and a USB port.
Users of other sequencers can still take advantage of Studio Manager's Total
Recall and editing functions by running it as a stand-alone application. Of course,
if you're using it like this, it's not integrated into your main DAW application, so
Studio Manager's settings have to be saved as a separate file and reloaded
manually when the song they refer to is required again.
Plans are currently afoot to make the Studio Connections interface protocols
available to other manufacturers so that they can develop compatible products. If
everyone buys into the idea, then Yamaha/Steinberg's concept will be an
attractive proposition; and even if they don't, it will remain a useful tool, if one
limited to a narrow range of products. For more information, visit: www.
studioconnections.org.

What's New?

Apart from the new grey livery, there are no obvious visual clues to suggest that
the ES is significantly different from the original Rack. The panel legending sports
only two minor additions, and the red LEDs are now yellow. The rear-panel
connections remain unchanged — however, a journey through the Edit pages
reveals a number of improvements. Possibly the most significant changes
concern the effects. Firstly, the number of insert effects has risen from 107 to
116, whilst five new choruses have been added to the global effects, taking their
total to 49. The most telling improvement is the way in which insert effects are
deployed within Multis. On the original Rack, up to four parts could be specified
to use the insert effects pair originally programmed for their assigned Voices.
This is a good system, in that those particular Voices are guaranteed to sound
pretty much the same in a Multi as they do when played in single Voice mode
(although the Multi's global effects will also have a bearing on the way the parts
eventually sound). Having only four insert-effectable parts could be seen as a
little restrictive, so the ES has improved on this situation, allowing up to eight
parts the luxury of their own insert effects. Add to this the global reverb and
chorus effects, which are accessible by all 16 parts, and the ES begins to look
extremely well endowed, with 18 simultaneously available effects.

Further improvements include a new Master Effects section; this can be applied

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Yamaha Motif Rack ES

across the combined stereo output of

the ES. Eight types of Master Effect
are provided, including a multi-band
compressor as well as various options
such as a delay, lo-fi processing, The back panel seems unchanged from the
distortion, a dynamic filter, a ring days of the original Motif Rack, featuring the
USB computer connection, MIDI In, Out and
modulator and an audio 'slicing' effect. Thru, main stereo outputs and four individual
When the ES is in Voice mode and the outs on analogue jacks, and co-axial and
Master Effect is engaged (via the Utility optical stereo digital out.
menu) this effect will be applied to any
Voice that is subsequently selected. In Multi mode, each Multi can utilise (and
store) its own Master Effect, or bypass it altogether. Of these effects, the multi-
band compressor would seem the sensible choice with which to treat a complete
multitimbral performance, whilst the other effects are arguably more suitable for
playing single voices. Nevertheless, having played complete mixes through the
each of the effect types — the lo-fi and distortion were rather fun — I can see
why the more radical and experimental amongst you would disagree! In addition
to the Master Effect, a five-band master EQ (situated after the Master Effect in
the signal path) is similarly applied across the combined stereo output, and this is
permanently 'on'. The low and high bands can be switched from peaking to
shelving types, whilst the middle three bands are all peaking types. In fact, the
EQ facilities of the ES are extremely comprehensive and flexible — the new
Master EQ is in addition to the existing two-band per-Element EQ and the three-
band per-Part EQ in Multis — to say nothing of the additional EQs found on the
insert effects.

On the original Rack, only basic offsets (cutoff, amplitude envelope, filter
envelope) could be applied to each Voice as a whole in Multi mode. The ES,
however, allows for detailed Multi-part editing at the Element level, in the same
way as on Roland's JV, XP and XV synths, for example. Drum parts can still only
be edited from within a Multi at the Common level though, unlike the Roland
synths. If you want to edit the drums on a per-key basis, or create customised
drum layouts, you must do this in Voice mode and store the results to a User
drum patch first.

The Benefits of USB

An unexpanded, out-of-the-box Motif Rack ES is a 16-part multitimbral unit, as
was the original Rack. However, the USB driver supplied with the ES has a new
trick up its sleeve: multiple output ports. Eight virtual ports are made available
from within your host sequencing program — so what are they for? If your ES is
fitted with PLG expansion boards, the main unit and each of the PLG boards can
be addressed on its own virtual port, and therefore its own unique MIDI channels.
In the case of the multitimbral PLG100XG board, this means a completely
independent set of 16 extra MIDI channels.
So, with the PLG100XG installed, together with one other PLG board, the ES is
capable of playing on up to 33 MIDI channels at once, clearly demonstrating the
advantage of using the USB connection. However, it should be borne in mind that
if you intend to do SysEx dumps from the ES into, for example, Sonar's integral

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Yamaha Motif Rack ES

SysEx librarian or a non-USB hardware sequencer or data filer, these still have to
be performed over a standard MIDI connection.

Preset & User Memories

The number of Preset and User banks has been augmented on the ES — there
are now six Preset banks plus one GM bank, giving a total of 896 presets. A third
User bank has also been added, bringing that total to 384. The number of Drum
kits remains the same, at 65 presets and 32 User kits. A further change has been
implemented in the arrangement of Performances and Multis. On the original
Rack, there was a library of preset Multis, split into two banks. Of these, Bank 1
contained 59 Performance types, while Bank 2 provided 65 Multi types. Any of
these presets, when edited, could then be saved to the User Multi bank. If this
sounds a little clunky, then Yamaha obviously thought so too — the ES now has
separately defined Performance and Multi banks, each filled with 128 presets, all
of which can be overwritten. This makes much more sense; preset Multis are
useful for demonstration purposes, but are highly likely to be overwritten in the
long term... right?

Motif Rack & Motif Rack ES Compared

MOTIF RACK MOTIF RACK ES
Preset Arpeggios 256 1787
Global Effects Reverbs (x20), chorus (x44) Reverb (x20), chorus (x49)
Insert Effects 107+107 116+116
Master Effects None 8
Master Equaliser No Yes
Preset Voices 128 (x5) 128 (x7)
User Voices 128 (x2) 128 (x3)
User Performances 59 128
User Multis 65 128
Detailed Part Edit In Multi Mode No Yes
Total Recall Compatibility No Yes
Multiple USB Ports One only Yes (x8)
Timing Problems Over MIDI Yes Cured!

The Arpeggiator

The arpeggiator on the original Rack was fairly well stocked with 256 assorted
patterns, but unlike its keyboard version, had no user-definable arpeggios. The
ES still has no user-definable patterns, but instead has upped the number of
patterns to a whopping 1787. These are grouped into 18 different categories,
including not only 'traditional' arpeggios, but complete musical phrases, lead

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Yamaha Motif Rack ES

lines, bass lines, guitar strums, drum patterns, and even some 'noteless'
arpeggios containing only controller data — think gated effects and the like. Up to
five different arpeggio patterns can be stored along with each Voice,
Performance or Multi. Any one of these five patterns can be selected while the
arpeggio is playing, either manually using the Page buttons or by external control
change messages, allowing for automated pattern changes in a sequence. Five
little boxes at the upper right of the display indicate (with a note icon) which of the
five slots have arpeggios assigned to them. If a new slot is selected while the
arpeggio is in the middle of a phrase or pattern, its note icon turns to an arrow to
indicate that it will be taking over at the beginning of the next measure, allowing
for seamless changes. This is a really neat idea, and makes it all the more of a
shame that Multis can still only use one global arpeggio at a time. If each part
could run its own patterns, there would be much potential multitimbral fun to be
had. Nevertheless, the arpeggiator's output can be recorded into a sequencer, so
you could always build up a song in this way part by part.

The PLG150AP Piano Expansion Board

Like the original Motif Rack, the ES can be
expanded with one or two PLG expansion
boards that work in tandem with the module.
These boards host completely independent
synths, increasing the total polyphony
(dependent on the board type) and utilising their
own independent effects DSP. The review Motif
Rack ES was supplied with the recently released
(and usually optional) PLG150AP, a dedicated
piano board boasting a 64-voice polyphonic,
29MB reproduction of a Yamaha CF3S Concert
Grand, employing three velocity layers to cover
the dynamic range from pianissimo to fortissimo.
The board comes with 32 presets, with room to
store 64 of your own custom variations. The
presets cover the ground you would expect —
bright, medium and dark versions of the basic piano, along with honky-tonk, EQ'd,
chorused and variously effected renditions. So how does it sound? Piano sounds
are invariably subjective, and while the pre-packaged pianos usually found in
synth modules are useful, they are rarely comparable to dedicated stage pianos
or the giga-sized piano libraries we've now become used to, especially when
played in isolation. That said, the piano that comes as part of the standard Motif
waveform set is easily amongst the better offerings.
Comparison with the PLG150AP board reveals the PLG version as being
generally louder, having a bigger, rounder low range, more body in the mid range
and slightly less 'fizz' in the upper frequencies. It responds well to EQ, as the
presets demonstrate — this piano can be made to sit at the back of a mix or to
force its way to the front without too much difficulty. My personal taste would lead
me to wish that the tonal difference between the quietest and loudest samples
was more dramatically stated — playing hard doesn't quite provide enough
excitement or 'energy'. Although Yamaha claim that the samples have 'long
loops', the notes' decay seems a tad on the short side, and trying to compensate

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Yamaha Motif Rack ES

by increasing the Amp EG Decay time doesn't actually make any difference, as
the samples' decay amplitude seems to be written in stone — a characteristic I
seem to remember also affecting Roland's SRX piano expansion board. You
could always apply a little compression to flesh out the body of the tone, but you'll
never make the notes any longer. As a consequence, I was reaching for the
reverb send level to try and extend the notes' length by any means possible!
Despite these niggles, both the
Motif's internal piano and the
PLG150AP stand up very well
against the competition, and the
PLG board has the benefit of its
own 64-note polyphony, which will
be indispensable to anyone using
any of the Motif range to produce
piano-heavy arrangements.
The modular plug-in boards
currently available for the Motif The new PLG150AP optional piano
Rack are as follows: expansion board.
PLG150AP (Yamaha's CF3S concert
grand piano).
PLG150AN (analogue physical modelling).
PLG150PF (pianos).
PLG150DX (FM synthesis).
PLG150VL (acoustic-modelling synthesis).
PLG150DR AWM2 (drums with dedicated effects).
PLG150PC AWM2 (percussion with dedicated effects).
PLG100XG (XG/GM synth).

The Big Issue

But for many reading this, the big question is: does the ES play in time over
MIDI? Well, I'm delighted to report that the ES's timing is good. In fact, it's
astonishingly good. In the previous tests on the original Rack, its timing was
compared to that of other MIDI modules, which generally have an inherent
latency to them, as we know, but which nevertheless performed very favourably
compared to the Motif Rack! On this occasion, the Rack ES was subjected to a
potentially cruel and unfair test — its timing was compared to that of a virtual
instrument plug-in (in this case Cakewalk's TTS synth). For those who aren't
aware, virtual instruments have virtually no latency when played back from a
sequencer. The ES was bombarded with one of my mind-numbingly trite, yet
action-packed multitimbral test ditties. The drum part was copied to a separate
track and assigned to the TTS synth, which played along with the ES. The result?
The two drum parts stuck to each other like glue. No obvious flamming, no
hesitations. I wouldn't go so far as to claim they were phase-accurate, but they
were as close as you could reasonably expect — and certainly as tight or tighter
than the other hardware MIDI instruments in my rack. Bearing in mind that
sequencers prioritise tracks according to their position in the track list, I moved

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Yamaha Motif Rack ES

the ES's drum part from track 2 to track 16, at the bottom of the list. Incredibly,
the timing remained just as solid as before. These tests were performed using
the ES's five-pin MIDI connections and the USB connection. In order to
accurately measure the difference in response time between the TTS virtual
synth and the ES, the drum part from the TTS synth was rendered to audio, as
were the various MIDI/USB/track number variations of the ES drum part.
Surprisingly, there was virtually no difference in timing between the ES's MIDI
and USB port outputs. And best of all, the average difference in timing between
the ES and TTS was in the order of between 4ms to 6ms. So yes, it looks like
Yamaha have got the timing problem well and truly licked.

Conclusions

Despite plenty of competition, the Rack ES offers a great deal to sway

prospective buyers. Fully expanded, it can perform not just 16 but 33 parts
simultaneously if a PLG100XG board is present, with a theoretical maximum
polyphony in excess of 192 voices. These same boards can provide additional
forms of synthesis — Virtual Analogue and Virtual Acoustic for example (see the
box above). The global and insert effects are of exceptionally high quality, with a
generous eight simultaneous insert effect pairs available in Multi mode, and the
overall sound quality is very appealing, aided in no small part by the well-
compiled, colourful sample waveform set. Whether you're into prog rock, acid
trance, tote bag or geek pie, the ES comes loaded with presets that will fit the bill.
For sound designers, the bundled (or freely downloadable) editing software
makes creating your own custom sounds a breeze. Most importantly, the timing
problems that beset the earlier Rack have finally been vanquished — the ES now
plays back as tightly as a ...well, a very tight thing. The UK price of the ES is
£949 — just a tad over the £925 of the original Rack — which means you get all
the new enhancements for a mere £24 extra. If you have been teetering
indecisively on the knife-edge of purchasing a Motif Rack, then teeter ye not; go
out and get the Motif Rack ES.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

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 Q Can I use a mono compressor for stereo compression?

Q Can I use a mono compressor for stereo compression?

Published in SOS May 2005
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Sound Advice

I was wondering if it's possible to use a single mono compressor that can
be stereo linked (the Focusrite ISA430, for example) as a stereo
compressor using the following method. Play a mono mix of the stereo
signal through the unit and record the Link Out signal; play the left
channel through the unit at the same time as the recorded Link Out signal
is fed into the Link In input, and record the output; do the same thing
with the right channel; combine the two mono recordings back into stereo.

I guess I'd have to be pretty careful about compensating for any delays. Also, the ISA430 (for one) doesn't
specify what levels the link signals work at. But might this work?

SOS Forum Post

Technical Editor Hugh Robjohns replies: In a word, no! There are

a number of problems with the process you are proposing.

Firstly, you are assuming that the signal used to link two units
together for stereo operation is a normal audio signal. It might be, The Focusrite ISA430 MkII can be linked to a
but equally, it might be a DC-referenced control signal, and the DC second unit for stereo compression.
reference would be lost if you were recording the signal into a DAW.
Similarly, any gain changes that occur anywhere in the recording and replaying of the link buss signal will
upset the compression settings.

Next, assuming that you can record and replay the link signal, there is the danger of disturbing the delicate
phase relationships between the left and right channels when each is processed separately and re-recorded.
This will upset the stereo imaging. Remember that both audio channels are going through two A-D/D-A stages,
both subject to random jitter effects controlled by different clocks at different times. Furthermore, the link buss
signal is going through another two conversion stages twice.

Furthermore, there's the delay introduced by the A-D/D-A conversion process to take into account. Remember
that the side-chain control signal will have to pass through an A-D stage on recording, and then a D-A stage on
replay to control the compressor. The left or right channel passes through a similar pair of D-A and A-D stages.
But in order to create the side-chain control signal, the mono sum track used also passes through a D-A stage.
Hence, the control signal will be one converter delay out of sync with the original audio, and hence you risk

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 Q Can I use a mono compressor for stereo compression?

transient compression errors.

When you also factor in the practical difficulty of optimising the compressor settings, plus the huge amount of
time and effort this process will take, it appears to be a futile triumph of technology over sense! Why bother
trying to benefit from the sonic quality of a unit like the Focusrite ISA430 when you are inherently trashing it by
using the process you describe? Given that pretty much everything you produce will need a pass through a
compressor sooner or later, why not simply buy or hire a decent stereo compressor?

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

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 Q How do I convert a split-stereo file to interleaved stereo?

Q How do I convert a split-stereo file to interleaved stereo?

Published in SOS May 2005
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Sound Advice

What's the best way to convert a split-stereo audio file into an interleaved
stereo file? I use Cool Edit, Wavelab and Logic. Do I just bring up the left
and right mono tracks in a Cool Edit multitracker window and bounce
them to stereo? What would be the best way to do this?

SOS Forum Post

Reviews Editor Mike Senior replies: You can do this directly from
Logic's Audio window.

First you have to make sure that the files have the '.L' and '.R'
suffixes on them as these are required for Logic to recognise them as
left- and right-channel files. This is the case on the Mac; if you're
using an older version of Logic on a PC, if I remember correctly,
there's a similar system, and I suspect you may be using the PC
version given your other choices of software.

Import the two files into the Audio window by choosing Add Audio File
from the window's Audio File menu. Once the files are in the audio
window, select Reconnect All Split Stereo Files from the Audio
window's Edit menu (see screenshot, left) — this step may not be
necessary, because Logic may do this by default as you import. Then select the connected split-stereo file in
the Audio window and choose one of the Convert To SDII/AIFF/WAV Stereo options from the Audio window's
Audio File menu.

This should save the interleaved stereo file in the same folder location as the split file, although it won't add the
interleaved file to the Audio window automatically. If you want to use the interleaved file in the Audio window,
then use the Add Audio File menu option again, but make sure that the Force Record & Convert Interleaved
Into Split Stereo Files box in the Audio Preferences is unticked, or it'll split the file again!

Published in SOS May 2005

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Q How do I convert a split-stereo file to interleaved stereo?

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

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 Q How should I sync up my digital inputs?

Q How should I sync up my digital inputs?

Published in SOS May 2005
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Sound Advice

I want to sync the converters in the following system — is this possible?

My Focusrite ISA428 is connected to my Emu 1212M soundcard via ADAT,
and my SPL Channel One is connected via S/PDIF. If I sync the clock on
the Emu to the SPL for example, will I then need the 1212M's sync board
to send a Word Clock signal to the Focusrite ISA428?

SOS Forum Post

Technical Editor Hugh Robjohns replies: If you are planning to

use all this gear at the same time, all three units must be in sync with
each other. That means making the one with the most stable clock
the system master, and then locking everything to that.

The Emu card can obviously be made a slave sync'ed to either the
ADAT or the S/PDIF input, so that's easy. You could then use a
Word Clock cable to couple the SPL and Focusrite, making either
one the master and the other the slave, and setting the 1212M's
sync source accordingly. That way, both the S/PDIF out of the SPL
and the ADAT out of the Focusrite will be sync'ed to the same clock, The Emu 1212M soundcard provides a
and everything will work happily. range of sync'ing options.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

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 Q Is there any advantage to using two subwoofers?

Q Is there any advantage to using two subwoofers?

Published in SOS May 2005
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Sound Advice

I have heard of having a subwoofer in your studio in combination with

your monitors to reach the extra-low frequencies. However in the Bob
Katz book [Mastering Audio, reviewed SOS October 2003 — www.
soundonsound.com/sos/oct03/articles/bobkatz.htm], there was a
picture of his studio where he has two subwoofers, one below the right
monitor and one below the left monitor. Sub frequencies in stereo? This
was a first for me. I never thought of the low frequencies being in stereo.
Usually you think of keyboards, voices, and guitars in stereo, but not bass
frequencies. What are your thoughts on this?

SOS Forum Post

Technical Editor Hugh Robjohns replies: This is a very complex subject

and we'll need to separate the real issues from the urban myths if we're to
get to the bottom of it.

There is quite a lot of evidence to suggest that we cannot perceive what

direction low-frequency sound is coming from, and hence a single
subwoofer would appear to make sense in the context of a sub-equipped
stereo or surround system, where full-range speakers aren't practical (or
affordable).

However, the upper frequency limit of the sub (and thus the lower limit of
the satellites) has to be set very carefully. A lot of home theatre systems,
for example, use ludicrously small satellites and thus require the sub to
operate well into the range of directional frequencies.
Because the EMES Black TV Active
monitors are full-range speakers,
There are two 'proper' standards for crossovers in the home cinema world when they're used with the Amber
(for sensible-sized speakers, not mini satellites) — one is 120Hz and the subwoofer, the crossover can be
other is 80Hz. Personally, I favour the latter, as I think it is possible to comfortably set at 80Hz rather than
locate 120Hz fundamentals. 120Hz.

However, the major fly in this particular ointment is that distortion in low-frequency speakers is inherently quite
high, especially in the case of ported cabinet designs. Distortion produces harmonics, and those harmonics,

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 Q Is there any advantage to using two subwoofers?

although low level, are in the directional frequency range and hence the location of the sub becomes very
obvious.

The result is that unless the sub is located close to the centre of the frontal sound stage, low-frequency content
will tend to produce harmonics which will pull the stereo image towards the location of the sub. And you can't
normally place the sub close to the centre of the frontal sound stage, because that will tend to excite the most
pronounced room modes and produce a very lumpy and uneven bass response.

So, one way around this problem is to ensure that the crossover point between the satellites and subwoofer is
as low as possible (80Hz for example), even though that means that reasonably-sized satellite speakers are
needed (not usually a problem in music studios, but not common in the cheap home theatre systems), and that
the subwoofer has to have extremely low distortion figures.

Using two subwoofers is an alternative way of tackling the problem.

By driving the room from two points instead of one, two different sets
of room modes are excited, which can result in a smoother overall
low-frequency balance. Also, any tendency for the harmonics
produced by distortion to skew the image can be balanced out and
thus effectively nulled, though I'm ignoring the unwanted masking
effect that such harmonics would have. We could also go on to
discuss the issues of actually trying to match the on- and off-axis
energy responses of the satellites and subs, which makes the use of
satellite and subwoofer systems seem even less attractive to me.

Of course, there are some systems that do work very well, and there
are undeniable practical and fiscal advantages to 2.1 (or 2.2) setups
in certain situations, but they all take a huge amount of careful Subwoofers which employ a closed cabinet
setting up, both in calibrating and positioning the subwoofer(s). design, like the Dynaudio BM9S, can offer a
We've touched on these issues in these pages and elsewhere in more precise low-frequency response, but at
the expense of efficiency.
Sound On Sound on a number of occasions — have a look at
Mallory Nicholls' article on subwoofers from SOS July 2002 (www.
soundonsound.com/sos/jul02/articles/subwoofers.asp) and the Studio SOS feature in SOS April 2003 (www.
soundonsound.com/sos/apr03/articles/studiosos0403.asp). Ideally, I'd always prefer a full-range stereo
system, but the constraints of space and budget mean that for most people this is rarely practical in a domestic
situation.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Q Should I buy a vintage analogue synth or a modern modelling synth?

Q Should I buy a vintage analogue synth or a modern modelling

synth?
Published in SOS May 2005
Print article : Close window

Sound Advice

I want to buy a 'knobby' synth because I am fed up of setting up sounds

with a data wheel. I also want a very analogue sound. I am thinking of
buying a modelled synth but, at the same time, I would really like a
genuine vintage synth to get a 'real' analogue sound (and because they
look so cool!). Any pointers would be appreciated.

Ben Slater

SOS contributor Steve Howell replies: As was pointed out in Sounding Off in SOS February 2004 (www.
soundonsound.com/sos/feb05/articles/soundingoff.htm), analogue synths are not without their pitfalls. Firstly,
assuming you can actually find a good example of the synth you favour, they can be costly to buy but, more
importantly, they can also be costly to maintain.

When buying a vintage synth,

you should check for noisy
pots and switches. Whilst
these can often be fixed with a
squirt of an appropriate contact
cleaner, replacing them can be
expensive, especially if the
pots are surface-mounted to
the PCB and/or the switches
aren't now available. You
Korg's analogue-modelling MS2000B synth (left) is more flexible than the vintage MS20
might think that noisy pots (top) and still offers plenty of 'tweakability'... but is it as desirable?
aren't really a problem, but a
large part of the appeal of a
knobby synth is the ability to tweak controls during a performance — there's nothing worse than your ripping
solo being spoilt by the intrusive sound of crackles!

You must also check out the keyboard. Often, the keyboard mechanism on these old synths is very simple and
it is all too easy for the contacts to break (or become bent or twisted so that they don't make contact). You can
sometimes fix these yourself if you're handy with a soldering iron, but getting them repaired or replaced by a
specialist is likely to set you back a few bob! And what about MIDI? Most old synths don't have it, although
they can usually be triggered by control-voltage (CV) and Gate signals. So if you want to integrate the vintage

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 Q Should I buy a vintage analogue synth or a modern modelling synth?

synth into an existing sequencing setup, you're going to have to seek out an example that has a MIDI retrofit,
or budget for some kind of MIDI-to-CV converter.

Then there's the sound-generating circuitry itself. By definition, it's going to be old, and components may be
failing, leading to tuning and other instabilities as well as noisy outputs — I once tried an ARP Axxe that
sounded as though someone was frying bacon in the background! Not only are these problems potentially
costly to repair but it could well be that some components are simply not available any more, especially if the
manufacturer used any integrated circuits that are now in short supply, or worse, custom components.

Of course, if you buy the synth from a reputable dealer who specialises in vintage synths, a lot of these issues
can be avoided, as the stuff they sell will invariably be refurbished (or at least serviced prior to sale) and will
often carry some form of warranty. You will pay a bit more for that peace of mind, understandably, but it can be
worth it.

You should also listen carefully to anything you are thinking about buying — or even do a blindfold test — and
ask yourself, "Does it actually sound good?". Do not allow yourself to be deluded by the attractive retro looks
or the allure of owning a genuine analogue. Due to component tolerances (and failing components), not every
analogue synth sounds good (or even the same as another identical model). And just because it has a Moog
badge on it (or whatever), don't consider that a guarantee of 'fatness', 'warmth' or any other adjectives that are
applied with dewy-eyed nostalgia to anything vintage.

I had lusted after an ARP

Odyssey since the time I tried
one as a teenager in Rod
Argent's Keyboards back in the
mid-'70s, and when one was
offered to me many, many
years later for a very silly price,
I bought it on spec without
checking it out first — bad
move! When it arrived, it
looked gorgeous — a prime If you have to have a true analogue synth, the Voyager by Bob Moog (above) might be
example of a white-faced expensive, but you won't find a MIDI-equipped original Minimoog (top) in pristine condition
original, with all its sliders and perfect working order for less, if at all.
intact — but it was a totally
underwhelming example of the instrument, and not at all what I had been remembering so fondly. I guess what
I am saying is, don't buy an old synth wearing rose-tinted spectacles. If you do, you may well be in for a
disappointment!

Modern, modelled synths are often a much better bet as a long-term investment. To all intents and purposes,
and perhaps contentiously, they sound equally as good as the majority of vintage synths, if not better in some
respects. They are inherently more flexible, are usually polyphonic, and are often more versatile, with sound-
shaping facilities that the originals could only have dreamt of. They are also usually multitimbral, come with
effects to polish the sound built in, and may have sophisticated (and often programmable) arpeggiators. They
might not sound exactly like a vintage Moog, ARP, or Roland, but they're pretty close, and (unless you're very
unlucky) won't spend much time being serviced.

I guess the only slight downside to these modern, modelled synths is that whilst many have plenty of knobs,

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 Q Should I buy a vintage analogue synth or a modern modelling synth?

they don't always have a control or switch for every parameter, unlike original analogue synths. Often, the less
frequently used parameters on the modelled versions are accessed via an LCD and menus. However, it's
perfectly possible to create very vibrant and convincing analogue synth sounds without ever having to delve
into the more obscure aspects of the synth's programmability.

No-one has a greater respect for old synths than I do — after all, they paved the way to the technology we
enjoy today. But just because a synth is old and carries a badge doesn't make it good. Witness the Polymoog
— what a weak-sounding, unreliable crock! There are some great old synths out there if you can find a good
example of one that satisfies your requirements and budget, but don't dismiss the more recent modelled
hardware synths.

If you're still in the market for analogue, check out Gordon Reid's guide to buying a vintage keyboard from
SOS September 1994 — see: www.soundonsound.com/sos/1994_articles/sep94/vintagesynths.html. And for a
more detailed idea of some of the things that can go wrong with vintage gear, check out the two-part feature on
equipment

servicing that appeared in SOS in March and April 1996 (see www.soundonsound.com/sos/1996_articles/
mar96/servicing.html and www.soundonsound.com/sos/1996_articles/apr96/servicing2.html).

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Q%20Should%20I%20buy%20a%2...20synth%20or%20a%20modern%20modelling%20synth.htm (3 of 3)9/27/2005 9:24:39 PM

 Q Will my PC run Garritan Personal Orchestra?

Q Will my PC run Garritan Personal Orchestra?

Published in SOS May 2005
Print article : Close window

Sound Advice

I am very interested in purchasing Garritan Personal Orchestra as

reviewed in the October 2004 edition of Sound On Sound. My concern is
whether my machine will be up to the task of running it. I have a PC
running Windows XP with an AMD processor at 1.47GHz and 768MB of
RAM. On the Garritan web site it says that the recommended spec is
1.8GHz, but no minimum spec is provided. However, I notice that the test
spec in your review was an IBM Thinkpad T40 with a 1.3GHz Pentium M
processor and 768MB of RAM. This leads me to think that my machine
should be OK. I just wanted to check I wasn't missing anything — does
the Pentium M range of processors have an extra edge over my AMD chip,
for example?

Rob Shillito

SOS Reviews Editor Sam Inglis replies: Clock rates can be a useful guide to the performance of different
CPUs, but they can also be confusing. Apple have spent much PR effort debunking the so-called 'Megahertz
myth', but it's perhaps not so well known that this is applicable to Intel and AMD processors as well. The
Pentium M processor does indeed use somewhat different technology from other Pentium CPUs, which
enables it to carry out more instructions in each clock cycle. A 1.3GHz Pentium M is, in performance terms, the
equivalent of a standard Pentium 4 running at 2.2GHz or thereabouts. See Mark Wherry's article on Centrinos
in SOS February 2004 for more details (www.soundonsound.com/sos/feb04/articles/centrinos.htm).

For the last few years, AMD have also been engaged in a marketing campaign designed to show that their
CPUs are equivalent to Pentium 4s running at much faster clock speeds. For instance, AMD's Athlon XP 2800
+ is supposed to be the equivalent of a 2.8GHz Pentium 4 (hence the 2800+ in the name), even though it is
actually clocked at 2.25GHz. Similarly, AMD's Opteron 64-bit processors have relatively low clock speeds but
offer performance comparable to the very fastest Pentium 4s.

It sounds as though your computer probably has AMD's 1700+ CPU, which in theory should offer performance
comparable to a 1.7GHz Pentium 4, closer to the recommended requirement. Running any piece of software
on a machine that barely reaches the recommended spec is never very much fun, and you may find yourself
running out of notes before you'd like, or being unable to do much else within your sequencing host, but I
would think you would be able to get some use from it on your machine — the recommended spec is to 'get
the best out' of the library, not a minimum spec required to use it at all.

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 Q Will my PC run Garritan Personal Orchestra?

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Q%20Will%20my%20PC%20run%20Garritan%20Personal%20Orchestra.htm (2 of 2)9/27/2005 9:24:43 PM

 Business End

In this article:
Jacoba
Business End
David Kenny Reader Tracks Evaluated
This Month's Panel Published in SOS May 2005
Print article : Close window

People : Miscellaneous

Business End enables you to have your demo

reviewed by a panel of producers, songwriters,
musicians and managers. If you want your demo to be
heard by them, please mark it 'Business End'.

Jacoba
Track 1 -
Coral Worman (CW): "This is very well recorded; it's
2.3Mb
very professional-sounding. I just don't think that it
has a unique sound. Although I like it, it's just too
derivative of other bands.

"They obviously play live a lot and I reckon the stuff they're playing on this goes
down well live — I think that would have been their yardstick for including these
tracks."

Gavin Nugent (GN): "I think the first track could

be tidied up a bit at the beginning. It's reasonably
immediate, but it's just not structured enough to
be a lead track on a demo. There are some great
sounds in here, though; it's very polished."

Michael Nielsen (MN): "To me, it sounds like

they're really good players — they've obviously
played together a lot. But it's all about
songwriting.

"When you're sending a demo, it's like first impressions when you meet
somebody."

GN: "You have to be so ruthless about it, and think that it's the first and possibly

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Business End

only chance you'll have to get something out of this. That first song isn't going to
get you anything apart from somebody going 'I've heard it before'. We're
reasonably humane about these things, but we get so many of them that you
start to wonder why you should take it seriously if the band themselves aren't.
And if they haven't figured out that most rudimentary of things, as Michael says,
the first impression is 'why bother?'"

MN: "The first track is so strongly reminiscent of other groups who, in their turn,
are reminiscent of other things. Sometimes you think 'I'll make this sound like
such-and-such, because it's familiar, and other people will feel that', but it can
count against you if you're sending it to industry people. I'd recommend trying to
do — or at least trying to present — something more original. It's no less risky
than sending something that everybody already knows. And then if people love it,
they're going to really love it, because it's individual."

GN: "Last year, in one of the band databases, one of these listings agencies,
there was this figure — apparently, there are about 10,000 working bands in the
London area alone. I mean we, Double Dragon, are very small and in one week,
we will still receive maybe 60 to 80 demos. I know exactly how this would be
treated — 10 seconds into the first song, I'd hear the voice, and skip on a track.
The second song would probably get about five seconds. A publisher might give
it a little bit longer, because, like with the first song, there is something of a
melody there, some structure. It's not particularly well thought-out, but there is
something."

MN: "I think differently — because I don't get 60 or 80 demos sent to me a week,
I'd listen to the whole thing. You might find track three should have been the first
track, and that could make a difference.

"They might be at a stage where, as they say in artistry, the first stage is
emulating your heroes. Well that's fine, it's a good thing; you have to master
everything, your sound and your intentions, and then you can move on to
becoming yourself. So with a third or fourth track, you might hear a glimpse of
that. But I've probably got more time to give it than someone from a record label.
For me it's some combination of sometimes an interesting lyric or a twist in how a
song is presented or a structure, someone who's writing in an unpredictable way,
not a predictable verse-chorus thing."

CW: "The things that make a demo stand out for me are the song and the voice.
I'll always go for the voice. It doesn't have to be good, just distinctive."

MN: "But then on the other hand, complete uniqueness isn't always a virtue
either, because no-one knows how to deal with it. It's a fine enough thing to be
good example within your genre. You could say that Radiohead, for example,
have influenced Muse, who, in turn, have influenced the first track on this.
There's one Radiohead album that's influenced loads of bands, but that doesn't
automatically make those bands bad or terrible. Muse have made a whole career
so far out of one, to my ears, slice of that sound and they do very well, so you

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Business End

can't knock it. But you have to be really, really good within your area. So
uniqueness isn't everything, but you still have to be good and bring something to
the party."

GN: "There's nothing here that says 'we know what we're about'. They are very
good players but, well — 10,000 other bands..."

MN: "If we were on the Isle of Man and saw them play a whole gig, we might say
something different but we've only got a CD with three songs. It's quite a hard
call in a way to have a full perspective from that, but they have to realise that
when they send a CD in, people like Gavin are getting 60 or 80 of them a week,
so they have to think about that a bit more."

GN: "I've been in publishers' offices when they've been listening to stuff, and
their ears are a little bit more open. I have to look at it and say 'where's the
uniqueness?' With something like this, where they obviously have a very
commercial sound, there needs to be something to make it stand out. Michael
mentioned Muse, and I remember when I first heard their early recordings,
there's just something, you know, like the X-factor, something not quantifiable —
and there's nothing here which says that to me."

CW: "Muse have unusual arrangements, though. When I heard them, it didn't
seem to me like they cared; it wasn't standard."

GN: "They can be almost orchestral."

CW: "Yes, exactly. Long verses and a lot of space in the middle. There's no
space in this; it's quite traditional. I think that's what I mean when I say 'be more
original'. You can get away with it with guitar music, with punky guitar music, it's
not pop, where you're made to work to very strict formulas.

They're obviously a good band — they can play and he can sing. Try not to be so
much like other bands, find that one thing for you and concentrate hard on the
writing.

"Nothing is ever truly original — someone really cool said that, but I can't
remember who it was! The real problem here is that each track here seems to be
derived from a different band that they're influenced by — they need to be
consistent. If you're marketing to media, no matter how much you like to not be
pigeonholed, the buying public need to think 'I like this because it's like T Rex
used to be' or 'I like this because it's punky' or 'I like this because it's like the
Strokes'."

David Kenny

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Business End

MN: "Tragic Shell-Suit Disaster — great title!"

Track 1 -
4.7Mb
GN: "When you hear the guitar at the start of the first
track, you think it's going to be like Beck or
something, but then you hear the lyrics, and they're just awful."

CW: "He could have almost pulled that off, but he so can't sing. Listening to the
second track, though, I can say right now that he could have a real future in
stand-up comedy."

GN: "OK, the second track — I take it all back.

This guy is actually a genius. This is absolute
genius!"

MN: "I like the fact that his voice sounds English."

GN: "It's just like a credible Goldie Looking

Chain."

MN: "If he just had some melodic changes and

sang a bit, this could be a single."

GN: "With the third song, you can't really hear the lyrics because of the guitar
sound."

CW: "Yeah, it's really distorting."

MN: "Maybe that's part of his sound."

CW: "I don't think that's deliberate."

GN: "You know that old trick of using a headphone as a mic when you've got no
microphone; I think he could be doing that. He's got some really strong
production ideas in here — there's a real identity."

CW: "The lyrics are really, really good."

GN: "It's a terrible phrase to use but the fourth track's like protest music, like
protest rock or hip-hop."

MN: "It's realist sort of stuff — it's good. I'd say that this is a successful demo,
because it makes me want to listen to the whole lot. I know we haven't got time
right now, but it would definitely make me want to hear more.

"He's got confidence, he sounds like where he comes from, which gets my vote

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Business End

— you know what I mean? He's not got an American accent and he's not rapping
in an American style; it's realist and I like that. It doesn't sound premeditated, it
sounds like something that's just got to come out."

GN: "I can think of labels in Europe and America who would release this. Very
simply because it gives the finger to just about everything and it doesn't give a
fuck about airplay. It's what it is. It's the Ronseal of music — it does exactly what
it says on the tin. You could start a great ball rolling by just getting it into an MP3
chart where nobody cares about how much swearing there is."

CW: "It's very clever lyrically and it's completely real. I'm sure there are so many
people who would like this."

MN: "It's tongue-in-cheek, it's got a bit of a swagger. It makes me think of things
like the Fall and the Happy Mondays, a real interesting mixture of things."

GN: "Yeah — Shaun Ryder. I think that on its own is enough to get me
interested. They were one of the last bands in a long time where you thought
'they really don't care'. They don't care what people like me at a record label
think, they just think 'Fuck you!' Which is great!"

MN: "It's not as primitive as that, though, is it? It's the way he does it; there's an
engaging aspect to it as well."

CW: "There's a lot of distortion on the recording but for me, with demos, it really
doesn't matter how bad the quality is, you can hear through that. At least he's not
sitting at home whining because he hasn't got the money to go into a studio."

MN: "No, you're right. It sounds like he's into some simple beats, a bit of funk,
and he's into playing guitar and singing and all of those things come across.

"On a recording, the technical quality is down to taste really. You could say that
this isn't a hi-fi, audiophile thing, but I don't think that's his intention at all. I don't
think he's got techno-lust as much as a desire to write music, which is the right
way around."

GN: "This is one to watch, I think."

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Business End

This Month's Panel

Gavin Nugent is the Label Manager for Double Dragon Music

(doubledragonmusic.com), an independent label which has released music by
Ash, The Crimea, Fi-Lo Radio and Charlotte Hatherley.
Gavin's music career has been interesting and varied; in and out of bands since
1987, he went on to specialise in studio design and construction and
subsequently became involved in designing bespoke music computers. He ran his
own studio in Dublin for four years before moving to the UK to work for Double
Dragon.
Michael Nielsen joined Strongroom in the early '90s where he trained as an
assistant engineer. He soon went on to become a recording and mix engineer and
began producing records in 1993. His production credits to date include two
Jamiroquai albums and he has been responsible for the mixing on four albums by
Underworld.
Lately he has become interested in 5.1 mixing and has worked on several albums
(including Underworld's groundbreaking Everything Everything LP) and two
movies using this format.
Coral Worman is a Director of a management company for artists, producers and
composers based at Strongroom Studios, and entitled (perhaps unsurprisingly)
Strongroom Management.
Coral has been involved in the music business for more than 20 years. In that
time she has worked in A&R at both RCA and Polydor. Her long experience in
artist and producer management led her to work for Orinoco Management in the
early '90s and latterly to her current position at Strongroom.
Many thanks to Nina Mistry and Strongroom Studios (www.strongroom.com) for
organising and hosting the session.

Published in SOS May 2005

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Business End

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Business%20End.htm (7 of 7)9/27/2005 9:25:02 PM

 Crosstalk: readers' writes

In this article:
Stars & Stripes
Crosstalk: readers' writes
Live: Going Solo Your letters, emails, faxes
Everything To Gain Published in SOS May 2005
Print article : Close window

People

Stars & Stripes

We've received quite a few letters concerning a news item in the April issue,
which unveiled our new 'sound-stripe' technology. While some readers seemed
to think we were pulling their legs, perhaps due to the timing of the
announcement, others seem to have more serious problems:

Well... I tried the new 'audio barcode' technology, and I have to say I'm a little
confused! Unfortunately, being left-handed I instinctively disregarded the stated
instructions and used my right forefinger to scan the code from right to left, rather
than left to right. All I heard was a garbled mess with faintly distinguishable Latin
phrases in a rather demonic-sounding voice — very disturbing, to say the least! I
would suggest displaying a warning in the future for your younger readers.

A Concerned Reader

Another reader seems to think that a prominent breakfast cereal company has
beaten us to the punch by some 35 years! Back to the drawing board we go...

With regard to the April Fool strip gag in the April issue of SOS, I'm afraid you're
a few decades late — not only is that technology real, but I remember it from the
1960s/very early '70s. OK — it wasn't exactly the same, but I do remember that
Kellogg's did a Frosties promotion involving a strip of red plastic, with a long
series of raised ridges, similar to the raised 'bar code' you made up. The
instructions were to make a small slit in the Frosties box, insert the red strip and
then pull it out through the slit as fast as possible. The result, using the box to
amplify the sound, was that you heard Tony the Tiger saying 'rrroooaarrr'! Well, it
was more of a fart than a tiger roar (akin to running a knife along a plastic round-
wound bass string), but it was an early design...

Wil Walker

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Crosstalk: readers' writes

Live: Going Solo

In his 'Pre-producing Live Loops in Cakewalk Sonar' technique article in the April
2005 issue, Craig Anderton states that, "Another advantage to using Sonar is
that the Solo buttons are 'latching' — you can have multiple Solo buttons
activated at once, whereas with Live, soloing any track 'de-solos' other tracks."

This isn't true. A quick read through the

Live manual will show that by using the
Ctrl/Apple key, multiple channels can
be set to solo. Furthermore, a single
click on one of these channels' Solo
buttons without Crtl/Apple pressed will
de-solo all channels... Or clicking on the
Solo of a non-playing channel will de-
solo all others and solo only that one.
Alternatively, 'Exclusive Solo' can be
deactivated in the Misc Preferences for a more traditional approach.

All this makes for an incredibly flexible and sensible soloing system, one that
allows me much more freedom, and less accidental ear-blasting, than any other
DAW I've tried.

Ben Dunkley

SOS contributor Craig Anderton replies: Ben is correct. As to how I could

make this kind of mistake, I've used Live in live performance for several years
now, and have assigned just about everything I use to a hardware controller.
Because Live can't record solo button presses, I adopted the Exclusive approach
for remote control a long time ago; it has become such a part of what I do that I
completely forgot about the other options. I greatly appreciate his passing along
this information to set the record straight, but better yet, it inspired me to re-read
the manual to see what else I once knew but had atrophied through lack of use,
which was an exercise well worth doing!

Everything To Gain
In reviewing the Phonic T8100, T8200, and T8300 valve processors [see SOS
April 2005], Paul White takes the manufacturer to task for the meters used on the
units, the legending of which is described as "ridiculously small". How about "just
plain wrong"?

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Crosstalk: readers' writes

Describing the metering facilities found

on the T8200 Tube Optimizer, Paul
writes, "[it] is generously equipped with
eight retro-style, illuminated circular
moving-coil meters showing the input/
output level and amount of gain
reduction on both frequency bands."

The picture of the unit in question,

found immediately above the quoted text, clearly shows the T8200's gain
reduction meters [pictured below] indicating a value of +3dB on both of the unit's
channels. By definition, gain reduction can only have negative values. Phonic
should have changed the metering scale on the dedicated GR meters, or
calibrated them so they would rest at 0dBVU.

While I do understand the decision to lower manufacturing costs by employing as

few different parts as possible, it is sadly obvious (and obviously sad) that the
Phonic engineers cut one corner too many.

Andre Knecht

Editor In Chief Paul White replies: If the meters had been large enough to fit
on a scale you could read easily, then it would have made sense for Phonic to
put on a second gain reduction scale, which most dual-purpose (level/gain
reduction) meters have. However, the unit in question is fairly inexpensive, and
adding a second scale would have probably made the meters even more
unreadable, so I can understand why they chose not to.

In reality, the position and dynamic behaviour of the meter needle should give
enough feedback to set the compressor adequately, but I have to agree that such
sloppy metering wouldn't be acceptable on a more professional unit. Despite this
shortcoming, the Phonic units still delivered a musically viable sound at an
attractive price, so the lack of a gain-reduction scale shouldn't put you off
checking them out.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Crosstalk%20%20readers%27%20writes.htm (3 of 3)9/27/2005 9:25:06 PM

 Greg Ladanyi

In this article:
Going To (Southern)
Greg Ladanyi
California Jackson Browne, Don Henley & The SoCal Sound
Running On Empty Published in SOS May 2005
Other Records, Other Print article : Close window
Continents
People : Artists/Engineers/Producers/Programmers
Toto IV
Escaping The Eagles
Empty Again

Greg Ladanyi showed up at the right time in rock

history to chair sessions for Jackson Browne, Don
Henley, Warren Zevon, Toto, Fleetwood Mac and the
Jacksons — but while 50 percent of life may be simply
showing up, the other half requires a lot of hard
work...

Dan Daley

The so-called SoCal Sound of the

1970s owed its existence, in large part,
to a producer and engineer who was
himself a Southern Californian. Greg
Ladanyi's parents moved from the
flatlands of Midwestern Indiana to
sunny Los Angeles when he was one
year old. "California is all I'd ever really
known," Ladanyi recalls of a youth
spent divided between accordion
lessons and high-school sports. By the Greg Ladanyi backstage with the Studer 24-
track used for recording the live material on
time he was 17, the accordion had Jackson Browne's Running On Empty.
morphed into keyboards and athletics
gave way to evenings spend playing,
and later managing and bartending, in clubs like the legendary Gazzarri's on
Sunset Strip, hard by the Roxy and the Whiskey A Go Go, the centre of the LA
music universe in the 1960s. Van Halen was the house band (the only one
allowed by the owner to play all original songs), and the Doors, Janis Joplin and
other icons of the era passed through the club while Ladanyi worked there,
sparking a more enduring interest in the music business.

In the early 1970s, he had shifted to doing live sound for bands as well as
managing a few of them. His first position was at a studio called Stronghold in
LA. Initially, he was responsible for booking and marketing the facility as well as

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Greg Ladanyi

assisting at engineering. His partner at the time, Al Thomas, was engineering

what would become Captain Beefheart's Blue Jeans And Moonbeams LP when it
became apparent that there was a personality clash in the making. "They'd
already given us $5,000 for the studio and we'd already spent it," Ladanyi recalls,
still aghast at the memory of his first studio crisis. "I said to the producer, Andy
DiMartino, 'I can engineer it.' I got behind the console and was calling my partner
every 20 minutes for instructions on this or that about running the console. But I
got through it, and by the time I was done, I was thoroughly in love with the idea
of engineering records."

In 1974, he got a job as a runner at Sound Factory, which along with Wally
Heider's, Goldstar and Sunset Sound formed the recording nucleus of Southern
California pop music. The first key thing that took place there was that Ladanyi
found himself being mentored by David Hassinger, the studio's owner and
engineer for dozens of major rock and pop artists, including the Four Seasons,
the Rolling Stones, the Grateful Dead and producer Phil Spector. "This guy was
a true pioneer of music recording," he says. "I don't think I could have gotten the
same education anywhere else but working with David day after day."

Going To (Southern) California

At Sound Factory, Ladanyi assisted with a number of major productions,

including Linda Ronstadt's Hasten Down The Wind album, produced by Peter
Asher and engineered by Val Garay, a pairing that would combine for most of the
singer's heady years of hits in the 1970s and '80s. Through them, Ladanyi was
exposed to the fundamental human elements of the Southern California sound, a
blend of folk, rock and country that engineers like Garay and ultimately Ladanyi
would learn to capture and put a light gloss on, but never a thick pop sheen.
"After Linda they'd work on JD Souther, then Andrew Gold," he remembers. "It
was on Souther's Black Rose that I moved over into engineering."

It's rare when someone can put their

finger on the exact moment when their
career fell into place. Ladanyi is one of
the lucky few. "When Jackson Browne
came in to Sound Factory, Val had
already been hired to mix the
Pretender album, recorded by John
Haney. Something happened with
scheduling and Jackson had to make a
change. I'd been developing a
relationship with him in the studio and I
got up the nerve and asked him if I
Danny Kortchmar and Jackson Browne on
could take a shot at mixing the record. Browne's tour bus, recording 'Nothing But
He gave the record to me and a couple Time' from the Running On Empty album.
of other engineers and when I came
back and played my mixes for him, he

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Greg Ladanyi

said 'You've got the gig.' I was elated, I was nervous, I was everything you'd
expect at a moment like that. I'd learned a lot by assisting with David Hassinger
and Val Garay; they gave me my technical backbone as an engineer. But this
was the moment when I had connected one-on-one with the artist."

At this point, Ladanyi makes a particularly trenchant observation. "I had the
technical background. But I had also learned engineering while working on a
group of very particular artists, like Jackson. They were meticulous song-crafters.
The song was the centre of every project, the jumping-off point for every record
and session. That was the genius of Southern California rock at the time — it
was all about the song. Jackson Browne, the Eagles, Linda Ronstadt, Warren
Zevon — they all worked off the song. The sound built itself up around the song.
If I had learned to make records in a more pop-music culture, I would have had a
whole different character as an engineer and later as a producer. The sounds
stemmed from the songs. The sounds came later. It was the writing, the chords,
that determined the sounds. If I had started out working on Metallica, my
understanding of the basics of audio technology would have been the same, but I
would have looked at sound differently."

The SoCal Sound was characterised by vocals that rode clearly above the track,
recorded into warm German condenser microphones with lots of low end, and
with organic instruments like acoustic guitars playing chords that often contained
suspended seconds and fourths and lots of open strings. The capo — referred to
as 'the cheater' by the Wrecking Crew generation of guitarists who prided
themselves on being able to play Monkees records in the key of 'E' flat —
became the equivalent of an effect on SoCal sessions. "The musicians were
incredible players — Lee Sklar and Craig Doerge and Danny Kortchmar and
Russ Kunkel," says Ladanyi. "But they were about playing together around the
song and the artist. No sequencers, no synthesizers. The energy at the time was
of a nucleus of a band playing, as much as possible, what the song was about.
There were overdubs and fixes, but the records really got developed as they
were played down by the band."

Running On Empty

How much of a role the band played on the records is exemplified by Jackson
Browne's Running On Empty, a chronicle of a band on the road in 1977,
recorded in concert halls, hotel rooms and buses during a US tour. It's one of
only two live albums of non-repeat material ever to have spawned a US hit single
(the other was Frampton Comes Alive).

"It was to be a total concept record," says Ladanyi. "When Jackson first brought it
up, people thought he was crazy. But he was determined he was going to do it."

Ladanyi was able to revive his old live mixing capabilities and combine them with
his newfound studio skills. "The guitars played through the same amps as they
used in the studio, which were little Vox and Fender Bassmans," says Ladanyi.

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Greg Ladanyi

"That way, we achieved some degree of

isolation on the instruments. If we had had
concert-level volume, we would have had
tremendous leakage. This is where close-
miking was essential. The audience
microphones — we usually used a pair of
Sennheiser shotguns pointed out from the
stage towards the crowd and sometimes a
U67 in the centre at the mix position —
became the ambience microphones. When
we had to do a fix on something later in the
studio, we'd just duck the audience
microphones a bit. We didn't put a lot of the
audience mics in the final mix — compare
that to the higher level of audience you hear
on Frampton Comes Alive — and the fixes Greg Ladanyi today.
were never anything substantial anyway.
These guys were good players.

"The concert dates were recorded to a Studer A80 24-track two-inch at 15ips
Dolby A — we just took a line-level feed from the stage directly to each track on
the Studer, with not much control over levels. Jackson still owns that same
Studer machine. We ended up with around 150 reels of tape.

"The sessions recorded in the bus [a Continental Silver Eagle tour bus on the
road in New Jersey], like 'Nothing But Time', were done in a little control room we
had set up in the rear, with a Technics 12-channel mixer and a Revox two-track.
It was a simple mix of Jackson and Danny [Kortchmar] on guitars, and Craig
[Doerge] on Wurlitzer; we overdubbed bass drum — Russ [Kunkel] playing a
cardboard box — and other percussion after bouncing the two tracks onto the
Studer 24. But the lead vocals and guitars were all live. There was a lot of low-
frequency rumble on those tracks, but that's what it sounds like on a
bus." (Ladanyi says even more detail will be heard on the forthcoming DVD-A
version. "At the beginning of 'Running On Empty', for example, you can faintly
hear Jackson having a conversion with Danny Kortchmar. He sings the first line
of the first verse to let him know what's going on, and then counts 'one, two,
three, four' into the song.")

"Other tracks were recorded in hotel rooms and lounges — 'The Road' was
recorded in Room 301 at the Cross Keys Inn, Columbia, Massachusetts;
'Cocaine' and 'Shaky Town' were done in Room 124 at the Edwardsville, Illinois,
Holiday Inn — recorded by the Record Plant truck parked outside with a Studer
24-track running at 30ips, non-Dolby."

Ladanyi recorded the entire record over the course of numerous performances in
several months on the road, using no monitor speakers and only the occasional
headphone. "Everything was happening so fast, I just used the meters to tell me
if the levels were right," he remembers. "I had to have total faith in my recording
techniques. It was quite a risk, but I wasn't really nervous about it — except when

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Greg Ladanyi

the signal would disappear from the

meter for a second, which happened a
couple of times. We were taking a split
from the stage and another was going
to the FOH position, where Buford
Jones was mixing the live show. So I
would run back and forth between the
mixing position and the backstage area
where the tape machines were,
constantly checking levels and how the
show was going."
Greg (right) with Mick Fleetwood and Billy
Burnett, working on Fleetwood Mac's Behind
Running On Empty was, in many ways, The Mask album in 1988.
the perfect recording storm — the
excitement of a live performance
before an audience, done with seasoned studio musicians using the same
equipment they had in the studio along with studio-quality microphones and tape
machines. "I mean, you might get that one night at a live concert recording, but to
have it every night, night after night on the road was amazing," Ladanyi says.
"You never knew where the magic night was going to be."

Other Records, Other Continents

As the SoCal musicians moved on to other genres, they often took Ladanyi with
them. One of the furthest stretches in the '80s was several sessions he
engineered for the Jacksons' 1984 Victory LP. "I was really getting out of the
California Sound box now," Ladanyi says. "The Jacksons were much more pop,
more so than they were R&B. Lots of sequencers. It was very melody-driven, but
with an R&B feel. It was a marriage of black and white pop, really.
"Michael [Jackson] was very hands-on on the songs he sang lead on. I used a
U67 on his vocal; it was a tube microphone that had a warmer kind of bottom end
to it. Michael's voice has less low end than Jackson's or Henley's. I used
compression on an API 525 to soften the attack of the mid-range, setting it for a
faster attack and with a slower release. The slower the release, the warmer the
sound you get — it works for horn parts, too."
Ladanyi's geography also broadened. In the late 1980s, he travelled to Europe to
work with General Public and Clannad. He began visiting Latin America regularly,
working with Latino rock bands like Engenheiros Do Hawaii in Brazil and Caifanes/
Jaguares in Mexico, which led to work with Latino artists in the US including Los
De Abajo and the Cruzados.
"They had heard of me through my work with Jackson, Don Henley and Toto,"
Ladanyi says. "It was a time when American music was being more heavily driven
by rap, which frankly I didn't have a lot of emotional connection with. The Latino
rock artists wanted to bring that pop sound like what we had gotten on Don
Henley to their records. They were doing rock records with percussion influences
of salsa, samba and reggae. It was a real eye-opener to me, all of it. New
cultures, new musical influences, stuff I could apply what I knew to. I mean, the
basics always apply — if you encounter a new instrument, or a new way of
playing a familiar one, use your ears to find the right place to put the

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microphones."

Toto IV

Ladanyi was to become involved in plenty of other enduring recordings. He had

moved into the co-production chair with Jackson for the singer's Hold On album
in 1980, and produced records for Phoebe Snow, Warren Zevon, the Church,
Fleetwood Mac and Jeff Healy over the next few years. But he was still in high
demand as an engineer, and when another group of ex-studio musicians called
Toto went back into the studio for their Toto IV album in 1982, they called on
Ladanyi to mix the album, which contained hits including 'Rosanna' and 'Africa',
and would win seven of nine Grammy nominations that year.

"Toto IV was by far the most extensive mixdown I

had been involved with," says Ladanyi. "The band
had three 24-track machines on some songs — up
to 72 tracks. I thought 'My God, how do I make all
this fit in two speakers?' The record was mixed at
the Sound Factory, which was a 32-input fader/24-
monitor API console — no automation and pairs of
hands moving faders — Steve Lukather, David
Paich, Steve Porcarro, Jeff Porcarro, and me. This
was the real deal. 'Rosanna' and 'Africa' took three
days each to complete the final mixdowns. These
songs were mixed in sections and then cut
together on the two-track. I balanced the track with
EQ, compressors and effects, then we all made
the fader rides together.
Greg (right) with Saul
Hernandes from Jagaures at
"I went on to record six more records with Toto. the console at Record Plant in
King Of Desire was the last one. We made that 1999, recording an album titled
record at Skywalker Ranch and did it in about Bajo el Azul.
three or four weeks. The thing about that record
was how dynamic the playing was in the era before CDs and how much fun it
was to capture that. When it came time to mix, I was asked to compress it pretty
heavily — that was the way you made it work for radio, or so a lot of people
thought. I had to compress it more than I thought was reasonable given how
dynamic the record was. Sure, compression makes it louder and more in-your-
face, but you just don't experience the dimension that the band played with. Too
many great records lost that element when they had to undergo compression for
radio's sake."

As the '80s went on, the demand for more tracks increased, and Ladanyi was
also employing more automation. "What everyone was looking for in the 1980s
was ways to experiment more, and more tracks were necessary for that," he
declares. "However you got them — sync'ing two analogue decks, MIDI, going to
digital decks — everyone wanted more flexibility than having to combine 10
tracks of drums to six just to open up four more tracks."

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Greg Ladanyi

Of course, it's a common complaint that this dramatic increase in the number of
available tracks led to deferred decision-making on the part of engineers and
producers, resulting in records that presented dozens if not hundreds of options
when it came time to mix. Ladanyi's response is surprising: "You have to balance
that against the enhanced level of experimentation that additional tracks affords
you. Experimentation that can take a record or an artist to a different place.
Granted, if there's too much stuff, the record can lose its focus and organic core.
There are pros and cons to what happened. It wasn't all good or all bad."

Escaping The Eagles

In 1984, Jackson Browne introduced Ladanyi to Don Henley. The former drummer/
vocalist/ composer for the Eagles — the band that was the essence of SoCal rock
— was preparing for his first solo album. "Don had come in to sing on a Glenda
Bickel record I was working on, and I was blown away by his presence," Ladanyi
says. "I was very forward about it — I told him that if he ever wanted to make a
record, I'd love to make it with him."
Ladanyi eventually did get called to engineer and co-produce, along with guitarist
Danny Kortchmar and Henley himself, on I Can't Stand Still and Building The
Perfect Beast. The overarching issue they faced was creating a sound for Henley
as a solo artist that would enable him to escape from the shadow of the Eagles,
one of the most successful rock bands of all time. "That was the question: how do
you take Don Henley out of the context of the Eagles?" Ladanyi says. "Danny was
instrumental in that; he was writing the record with Don and he was bringing in a
lot of blues and R&B influences, which pulled Don away from the country and pop
sounds."
Another way to do it was by using sequenced drums and synthesizers. "You
never heard any of that on those Southern California recordings," says Ladanyi.
'Dirty Laundry', I Can't Stand Still's biggest hit, had a looped keyboard part that
guitar parts were built around. The song's keel was a sequenced Linn Drum part
that Toto drummer Jeff Porcarro would later play over, with both live and machine
drums mixed together in the end. Ladanyi also altered the philosophy of sound
being subordinate to the song that had guided so many of the 'SoCal Mafia'
records. "We used a lot of reverbs on the drums to create live-sounding
ambiences," he explains. "But we also used a lot of gating — I'd been listening to
what Phil Collins had been doing with that effect at the time. We used the robot-
looking version of the EMT 250 digital reverb a lot. There was a real evolution in
how records sounded going on then. Big reverbs and fast gates were a central
part of that. It was very aggressive-sounding stuff. The 250 also had a short 0.8-
second setting that was great for drums."
It was, in other words, the antithesis of the Southern California laid-back approach
to records, and Don Henley's solo records are a case study in how an artist can
create an almost entirely new persona by changing sounds and attitude. Tellingly,
the hooks on most of Henley's biggest solo hits — 'Dirty Laundry', 'Boys Of
Summer' and 'Sunset Grill' — are virtually devoid of the soaring three-part
harmonies that characterised his earlier records with the Eagles.

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Greg Ladanyi

Empty Again

Today, Ladanyi works in a small home studio based around a Nuendo system. "It
was the first hard disk recording program that worked at 96kHz with 24 tracks,"
he says. "I do all my recording and mixing inside Nuendo with a front end
comprised of the Groove Tubes Vipre mic pre and Groove Tubes compressor,
Tube-tech EQ/compressor, and assorted microphones."

Ladanyi has been quite active in recent years. Recent successes include The
Crickets & Their Buddies, where he engineered and produced performances by
Eric Clapton, Grahan Nash, John Prine, Waylon Jennings, Nancy Griffith, Albert
Lee, Peter Case, Johnny Rivers and others with members of Buddy Holly's
original band, and Joe Cocker's Heart & Soul, which Ladanyi mixed.

He has also completed a 5.1 remix of

Running On Empty begun three years
ago. Transfers of the original tapes
were done through a Neve 8078
console at Jackson Browne's
Groovemasters Studio. "We broke out
different elements to give us around 40
tracks on some songs," Ladanyi says
of the mix. "For example, we needed a
guitar ride at a certain point, so we just
copied that section to a spare track, Greg Ladanyi at the Complex in 1984.
processed it and wrote separate fader
moves. We took some of the longer intros from the master tapes, and used some
of it as ambience. But we have material that is more audible on 5.1-channel DVD-
A, because you have more discrete locations. On 'Cocaine', 'Shaky Town',
'Rosie' and 'Nothing But Time' there are conversations amongst the band that
you would not hear on the stereo versions. We mixed on Westlake monitors
powered by Hafler amps. The LCR and surrounds are BBSM-5s, while the LFE is
an Lc8.1 subwoofer.

"For the live concerts we wanted to put the listener inside the audience — with
the band coming from the front speakers, just as they would at the concert —
while for the tracks recorded in hotel rooms [and on the bus] we wanted to
duplicate the musicians' positions around the microphone, with the listener in the
very centre of room.

"We had to clean up some buzzes on the bass track, for example, using tight EQ
notches — you would not hear them in the stereo mix, but here with six channels
you cannot hide them any more, so we needed to do a little cleaning up. I like to
use compression as an 'enhancer' of an overall performance, particularly on
vocals, bass and guitars, where it can bring up the bottom end. We used different
mix techniques, capturing early reflections from the recording environment and
placing them in the mix to add a reality around the audience."

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Greg Ladanyi

The only thing he says he would have changed 27 years later is the addition of a
few more audience microphones. Given Ladanyi's remarkable career, that's
about all one would change.

Published in SOS May 2005

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 Manny Marroquin

In this article:
Breaking In
Manny Marroquin
Rules Of The Road Mixing For Kanye West & Alicia Keys
One On One Published in SOS May 2005
Kanye West Print article : Close window
From Maroon To Pink
People : Artists/Engineers/Producers/Programmers
What It's All About

You might not know his name, but you've definitely

heard his work: Manny Marroquin is the mix engineer
of choice for leading artists in both urban and rock
music.

Dan Daley

In an industry where specialist mixers

sometimes achieve celebrity in their
own right, Manny Marroquin has
purposely kept his profile low.
However, the 33-year-old Guatemalan
native has created a discography that
makes one wonder if it took more effort
to keep it quiet than it would have done
to promote it. Marroquin's credits
include not only R&B and hip-hop, but
more than a sprinkling of rock and pop.
He's worked on recent albums from
Kanye West and Alicia Keys, Grammy Manny Marroquin at the SSL 9000K desk in
winners for Best Rap Album and Best Larrabee North Studios.
R&B Album respectively, and he has
garnered previous Grammy nominations for tracks for Cher, Whitney Houston
and Toni Braxton. Producers including Babyface, R Kelly, Rick Nowels and Daryl
Simmons have turned their work over to him to be mixed, most often on the SSL
9000K desk at his 'home' studio, Larrabee Studios in North Hollywood. Other
artists whose records his deft touch has enhanced include Usher, Mario, Maroon
5, Carlos Santana, Janet Jackson, the Nappy Roots, Lee Ann Rimes, Will Smith,
Cher and Seal.

Breaking In

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Manny Marroquin

Marroquin began mixing full time about six years ago, after a stint engineering at
various Southern California studios. He moved from Guatemala City to Los
Angeles with his family when he was nine, at the height of an intense civil war
that gripped Guatemala and much of the rest of Central American in the 1970s
and '80s. "I don't think it affected me like it could have," he says. "I don't
remember it as shocking or scary. War... was just a part of life then."

In LA, Marroquin was a student at Hamilton High

School, a magnet school for students showing
talent in the arts. He went into high school as a
drummer; by the time he was a year away from
graduation, he had the keys to the school's
recording studio, where he would record demos
for other students on its Tascam eight-track deck
and Ramsa mixer. "I knew that after graduation I
wanted to... mix," he says. "I didn't really know
what a mixer did, but I knew that I wanted to mix."

Despite several offers of college scholarships,

Marroquin followed a teacher's advice to get an entry-level job at a studio. Upon
graduation, and much to the consternation of his mother, he began working as a
runner at Enterprise Studios in LA. "My first day there was the best and worst
day of my life," he recalls. "The best because I was at a huge studio; the worst
because I was cleaning bathrooms. At that point I made it my mission to get out
of the 'runner business' as soon as possible."

There was no one magic moment that enabled him to transition from assistant
engineering to the first chair. Rather, he remembers it as a progression of
projects, each more complex than the last, and meeting a succession of
producers to whom he could prove himself. As a result, he says, "I treat every
mix as if it is my last and never take anything for granted."

Rules Of The Road

Rap and hip-hop seem to Marroquin to have more facets than other genres.
"Rock, for instance, has certain rules," he explains. "You have guitars, drums and
vocals. If the kit is recorded well, you're not going to change the sound of it that
much. Why would you want to, unless you were trying for a specific sound?

"Urban music, though, that's the Wild West, man. The sounds can be very varied,
from so many different sources, and all of them encourage you to get more
creative with how you put them together. The percussion is made up of a lot of
samples from a lot of different sources, so, unlike a drum kit, if you change the
balance slightly between kick and snare and hi-hat — for example, if you make
the hat 2dB hotter than it would normally be — the feel of the track changes. You
move the fader a half dB and nothing's the same. Rock's about sound and sonics

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Manny Marroquin

and the song; urban's about the feel and the

vibe."

Rap and hip-hop mixes are almost like

mysteries to be solved with non-verbal clues,
which Marroquin enjoys following. "It's often
about using samplers and loops to capture
an older feel," he says. "Like Alicia Keys.
She's a big fan of old R&B sounds. On 'You
Don't Know My Name' [one of four hit singles
on Keys' Diary... LP that Marroquin mixed]
there was a sample in there from the original
recording of the song by the Moments. It's
surrounded by a bunch of other tracks and
the misconception is that all you're supposed Some of Manny Marroquin's 'favourite
to hear is the 'boom'. But with this particular toys', including Neve 1066 preamps
song it wasn't about the boom, like other upgraded to 1073s by Brent Averill.
more pronounced hip-hop tracks. You dig
deeper and you find that sample and that's what makes it all make sense. If one
of those tracks isn't balanced right, you lose it."

Interestingly, Marroquin liked the way that that LP's progress through numerous
studios in LA, New York, London, Paris and Amsterdam upset his routine — just
enough, he says, to put an edge on his mixes that he wouldn't have gotten
staying in his own studio. "It's good to get out now and then," he says. "I was
following the production around, and I would ride the cab at night through the
streets of the cities. You get inspired in a different way."

One On One
The discography on Manny Marroquin's web site at www.mannymarroquin.com
takes the trouble to list the specific tracks he worked on for each artist's project.
This degree of transparency is unusual these days, especially in urban music
genres, where credits are often a meaningless recitation that fails to separate
posse members and hangers-on from front-line engineers and musicians.
"It seems like it gets out of hand sometimes," he says, on that topic. "I think it has
to do with being organised — or not. When you're young you don't really know
how the whole record-making process works, and more people can learn it on
their own now because they can record at home. There are so many people
involved in productions that it's hard to know who to give credit to. They don't
realise that you can create a better sound and a better feel in the studio with one
or two guys working consistently on the same project together. Really great music
is the result of collaboration, but there's a point at which too many collaborators
can diminish the outcome, lose some of the focus. I think sometimes that the
business gets to the point where lives get so crazy that they just don't have the
time to work like a team. It's like a luxury these days for producers to be able to
bring one engineer everywhere he has to be to work on an entire album from start
to finish."

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Manny Marroquin

I mention that mixing is something of a solitary profession. Marroquin, however,

views it as something more akin to a one-on-one tutorial, an opportunity to spend
time with an artist and a producer, learning about the nature of the project and the
aspirations for its outcome as much by listening to them talk as listening to the
track elements.
"Every time I mix with someone, I learn so much just by talking with them and
getting a sense of the personality — theirs and the project's," he explains. "If I'm
going to style their record — and that's really what a mixer is, a stylist for records
— I need to know something about them as people and as artists. The mix is an
extension of the artist's or producer's vision, and I have to become a bit like a
mind-reader to understand what it is they're trying to get across."

Kanye West

Kanye West has become a household name as an artist — his very public
frustrations at missing out on new artist honours throughout 2004 seemed to
have made his Grammy win in February as much a vindication as a prize. But
Marroquin regards West first as the exceptional producer that he is, citing his
work with Jay-Z in particular. Marroquin mixed virtually all of West's Grammy-
winning The College Dropout LP, including the singles 'Through The Wire', 'All
Falls Down' and 'Jesus Walks'. "He wanted one person to mix the album," he
says. "That's a rare thing these days, but I hope it becomes more common again
because of the consistency you get in the end. I see this as a trend — in the
1990s I used to get one or two songs on a record; now more often than not I'm
getting half the record or more. It's not me — it's the idea that working with a
consistent team can give you a better result."

Kanye West's 'Through The Wire' would have

been a challenge to any engineer. The track
was about the serious car accident that left the
artist's jaw wired shut during the recording
schedule. "It does sound unique, yeah — it's
hard to sing when you can't open your mouth,"
Marroquin chuckles. He used a Sony C800
condenser microphone to take advantage of
the Sony's higher sensitivity under the
circumstances, but running it through his usual
signal chain of a Neve 1073 EQ and a Tube-
tech CL1B compressor straight into Pro Tools.

On West's record and others, Marroquin finds

that creating stems as he mixes makes for At the age of 16, playing the drums.
more efficient mixes later on. Based on an
audio post-production methodology, he'll create
submixes of guitars, drums, percussion, keys and even background vocals. The
key, says Marroquin, is to compress these stems, rather than the overall stereo
mix. "I'll send individual elements, like the kick and snare, to their own
compressors," he explains. "I'll put them on separate busses and send them to a

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Manny Marroquin

Fairchild 670. For example, I'll put all the drums through that to give them some
'glue' and then make a stem out of them with the bass. You can also get a
compression-like effect without squashing the tracks by using EQ. As you know,
compression brings out certain frequencies in different instruments, so you find
them and you tweak them with EQ. I like the way guitars sound through a Neve
33609 and a Motown EQ to bring out a pleasant high end."

For Usher's 'Can You Handle It' from the Confessions album, he applied an anti-
sibilance technique he says is painstaking but worth the effort. "One of the
hardest things to do is to get rid of sibilance using only EQ without affecting the
presense of the vocal," he says. "It's an art. I use a Dbx 902 de-esser, which is
one of the best-sounding de-essers out there. But it only has one frequency per
curve. So I do the de-essing using the SSL EQs through a side-chain. They allow
you to really key in on the affected frequencies. Then you send the output to a
compressor preset for the frequencies you've been tweaking, and when those
frequencies trigger the compressor, they get backed down into the track where
they belong. I learned that technique from [the late] Barney Perkins, who used to
use it on a lot of the Babyface and LA Reid stuff he worked on. I was a huge
Teddy Reilly fan — the New Jack Swing sound, I loved that sound even before I
started working on SSL consoles and I found that he built his sound around the
G-series compressor. I started using that compressor and it was love right from
the beginning. Now, the XL reduces the amount of outboard gear that I need to
use because of its great musical-sounding dynamics section."

From Maroon To Pink

Unlike some hip-hop and R&B mixers, Marroquin also feels quite at home on
rock tracks. He recorded and mixed Maroon 5's contribution to the Spiderman 2
soundtrack, and their track on a Sly & The Family Stone tribute record, the
classic 'Everyday People'. "I tell you, it was hard to do because of Arrested
Development's version of that song, which really set the bar," he says. "So we
decided to take the real indie-rock approach to the sound, putting a lot of the
tracks through foot pedals and distortion. And we used programmed drums, so
we have this contrast of the essence of garage rock and the essence of hip-hop
on the same track. The tricky thing is to do all this and keep them sounding like
Maroon 5 — a pop band doing a soul music track with a garage-band vibe."

Carlos Santana's 'Game Of Love' was a huge European hit that Marroquin says
had nothing to do with hip-hop or Latin genres. "We kept the emphasis off the
kick and snare and put it on the guitars and bass," he says. "With the drums, the
kit sounds more cohesive, with less individual emphasis on the kick and snare,
like you get in urban music. I'll use the sub-compression a lot more, to give the
'glue' effect and also to add the kind of analogue tone that compression brings to
a track."

Marroquin also worked on singer Pink's first album, produced in 1997 by

Babyface and the hip-hop team Presidential Campaign. The record, which

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Manny Marroquin

spawned the minor hit 'Most Girls', was

"Pink before she was Pink", he says,
as the singer was developing both a
persona and a performance style. In
keeping with her own sharp edges,
Marroquin used a combination of pro
reverbs, including an AMS and Lexicon
480L and 224XL units, and the clangy
spring reverb from a Fender guitar
amp. "Hip-hop can tend to have very
little reverb, and what there is is short,"
he says. "When I get kind of off-the-
wall records, I like to play with spring Manny Marroquin got into engineering at
reverbs. They're cool-sounding. You high school.
can make it short and tight with a gate
and it adds tone and depth to the sound without washing it out. I also EQ the
reverb return. I listen for the frequency where the reverb matches the input signal
and tweak that. When you have one thing in a crowded mix you really want to
bring out but adding EQ would make it sound too harsh, put it through a spring
reverb. They're noisy, but thank God for gates."

Marroquin picked up a Grammy nomination for Cher's 'Love One Another', which,
like all Cher records, required vocals very far out in front. "The problem with that
is that it's easy for the vocals to get separated from the rest of the track," he
cautions. "What I would do is add tube compression to the vocal. That adds
warmth on the low end, around 150Hz, better than EQ can give you there. With
Cher, she already has a lot of low-mid tone to her voice, so I would go to the high-
mids with an Avalon 2055 equaliser and add a little there."

Lee Ann Rimes' vast dynamic range led

Marroquin to automate a slew of EQs and
compressors even as he manually rode the fader
on her vocal. "The problem really was that the
vocal recording was overcompressed in places,"
he says. "Which I can understand because of
how big and dynamic her voice is. Like many
women singers with big voices — Toni Braxton is
also like that — the louder they get, the less you
have to do to the vocal. By the finale I'd have
nothing on the track at all. You can often fix over-
compressed vocals, but it's tedious. You EQ the
problem areas out and then compensate for the lost volume with more level. The
key is to pay attention to the lines leading up to the problem spot and the one
after it. Think of them as a line on a curve instead of a square. You trick the ear
with gradual changes before and after the problem part."

What It's All About

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Manny Marroquin

Marroquin cites a number of influences as a mixer, including Bob Powers and

Bob Clearmountain, who he says "changed the game forever on how mixers
work and are regarded by the rest of the industry". What seems curious, though,
is why Marroquin's discography, so full of urban rock and pop artists, lacks Latino
names. "I'm not sure, I never really thought about it," he replies. "It's funny — I
worked with Ricky Martin, but I did his English-language stuff, not the Spanish. I
remember when I was first starting out, people would tell me 'Don't do this, you
speak with an accent, you're not white, you're a foreigner.' Some people thought
I should be a technician; they didn't think I could hold my own with someone like
Madonna in a control room. But I never thought like that, and I never took any of
those comments personally. I think people were trying to help me avoid getting
hurt or disappointed. But I'll tell you what — it really comes down to what comes
from the heart. It's not about money or status. It's about music."

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Manny%20Marroquin.htm (7 of 7)9/27/2005 9:25:11 PM

 Sounding Off

Sounding Off
Mr X
Published in SOS May 2005
Print article : Close window

People : Sounding Off

Soon all software could be Free — like it or not...

Julian Bentwood

When most of us think of free software, we simply think of software that doesn't cost us anything; those useful
VST plug-ins we can download from the web without first having to supply our credit card details, for example.
However, there is another, more specific definition of free software which has been attracting a lot of attention
in recent years, thanks to a steady growth of interest in the GNU/Linux operating system.

Advocates of this kind of free software make a distinction between software that is
available at no cost, and software that is Free in a larger sense. For software to be
truly Free, the argument goes, a specific set of freedoms and rights must be granted
to its users — including the right to redistribute copies.

Many supporters of this species of Free Software argue that all software ought to be
Free — not because this represents the most efficient or cost-effective method of
distribution, but because it is morally better; better for society as a whole.

I'm not writing to argue either for or against Free Software in principle, although I think
there are some interesting arguments to be had. I'm writing this piece because I'm
About The
increasingly convinced that, regardless of the arguments for and against, those of us
Author
who use computers to compose, record and produce music may well find ourselves
dependent on Free Software before too much longer. Julian Bentwood is a
pseudonym. The
man in the bag has
This may seem like a strange claim to make. At present only a minority of enthusiasts worked for several
use exclusively Free Software for anything, and a still smaller minority is of the major music
experimenting with Free Software for music. Even these 'early adopters' would software companies
probably concede that Free audio applications still lag some distance behind their and wishes to
remain anonymous.
commercial counterparts, in terms of both features and usability.

So why am I convinced that Free Software is our future? To use a military metaphor, while commercial
software may be winning the features arms-race, it has already lost the political battle for hearts and minds.

file:///H|/SOS%2005-05/Sounding%20Off.htm (1 of 2)9/27/2005 9:25:13 PM

 Sounding Off

One of the central tenets of the Free Software philosophy is that users ought to be free to make and distribute
copies of software, so that more people may benefit. This is condemned as piracy by commercial software
developers — and yet it's exactly what a great many musicians and producers do already, without giving it a
second thought.

If you don't know what I'm talking about, and if neither you nor any of your friends has ever used any pirated
software, then you can congratulate yourselves on belonging to a virtuous minority. To say that, in my
experience, the use of pirated software is 'widespread' would be an understatement.

The people involved in this unauthorised redistribution may have given little thought to the arguments on either
side of the Free Software debate. However, whether they realise it or not, their actions represent an implicit
endorsement of one of the key claims of Free Software advocates — namely that the social benefits of sharing
software outweigh any harm done by refusing to recognise the intellectual property rights of commercial
software developers.

A great many computer users, perhaps even the majority, apparently do not feel that the limits these
developers seek to impose on them are reasonable. Consequently they abide by neither the letter nor the spirit
of the End User Licence Agreements they supposedly consent to by clicking 'OK'.

You may feel this is lamentable. You might even point to it as evidence of the intrinsically iniquitous and selfish
nature of human beings (although you'd be open to an accusation of cynicism if you did). I would ask you what,
practically speaking, you think can be done about it.

Copy-protection is not the answer. I'm talking about changing people's minds; persuading them that it's actually
more important to respect the wishes of software developers than it is to allow their Internet peers to upload. I
don't know how this can be done, or even if it's possible. As things currently stand, commercial software
developers appear to swimming against the tide, and I see no sign that the tide is about to turn. If unauthorised
software copying is anything like as widespread as I suspect it is, it must represent a considerable disincentive
for programmers to continue developing commercial products.

Whether you're convinced by the arguments of Free Software advocates or not, it's hard to deny that their
vision of how things ought to be done much more closely resembles the reality of what actually happens than
the average End User Licence Agreement. In the end, Free Software may win by default.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Sounding%20Off.htm (2 of 2)9/27/2005 9:25:13 PM

 Studio SOS

In this article:
Synchronising Cubase SX
Studio SOS
To A Roland VS2480 East Norfolk Sixth Form College
VS2480 Tips Published in SOS May 2005
V-Fader Control Print article : Close window
Mixing Horns & Rhythm
People : Studio SOS
Guitars
Carl's Comments
Homeward Bound

East Norfolk Sixth Form College needed help

integrating their hardware multitracker with their
computer sequencing system, so the SOS team
travelled over to Great Yarmouth to lend a hand.

Mike Senior

Back in the January and February

issues of SOS I wrote a pair of
workshop articles about the Roland
VS2480 recorder. Within a couple of
weeks of the second article going to
press, I'd had two separate reader
inquiries at the SOS office asking for a
tutorial on the multitracker's remote
control functions. Having a dedicated
programmable fader bank in my own
setup, I'd never needed to experiment
with this aspect of the VS2480's
operation, so I couldn't really provide
any useful hands-on advice. However,
I knew how the remote control
functions were meant to work in theory,
so I offered to head over to the studio
of one of the two readers, Carl
Simmonds, to help him get things
worked out in practice.

Carl teaches music technology at East

Norfolk Sixth Form College, where he
manages a studio setup based around
a Triton LE keyboard workstation, a PC
running Steinberg Cubase SX, and a

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Studio SOS

Roland VS2480 recorder. There were

two main things which Carl wanted to
do. Firstly, he wanted to configure the
system such that the computer
sequencer and the hardware
multitracker would run in sync, under
the direction of the hardware transport
controls. Then he wanted to be able to
use the VS2480's faders to control
mixer parameters within Cubase SX.
There is a dedicated fader layer within
the Roland digital mixer for this
purpose, but Carl was unsure how to
get this working properly.

I enlisted the help of SOS contributor

Tom Flint, himself an experienced user
of digital multitrackers, and we headed
over to Great Yarmouth where the
college is based. On our arrival we Photos: Tom Flint
found the small control room already
In order to synchronise the Roland VS2480
half full of people, as some of the with Cubase SX, Mike first adjusted the
Music Technology 'A'-level students Utility settings of the multitracker as shown in
were keen to be in on the session, the insets (above): MIDI Out Sync Gen was
despite our visit coinciding with the half- set to MTC so that the VS2480 would send
term holidays! MIDI Time Code to the sequencer; MMC
Mode was set to Master so that MIDI
Machine Control messages would also be
sent; and SysEx Tx Switch was set to On so
that these messages would not be filtered
Synchronising Cubase out before reaching the multitracker's output.
Finally, Mike hooked his own Yamaha
SX To A Roland VS2480 QY700 (below) to the VS2480 to test that the
setup was correct before starting work on
Cubase SX.
With barely a pause to inhale our cups
of tea, we set to work on synchronising
the multitracker with the sequencer. The first thing to do was make sure that the
multitracker was set up right. I explained to Carl that the most sensible method of
synchronisation in his case would probably involve using MIDI Time Code (MTC).
The other option would have been to use MIDI Clock, but that requires identical
tempo and time-signature data to be set up in the sequencer and the
multitracker, so can be a hassle if you have changes of time signature or tempo
within any song. I also suggested that it would make most sense to have the
sequencer as slave and the VS2480 as master, as most of the audio recording is
done on the multitracker.

Dealing with the VS2480 was fairly quick — in the Utility menu the Sync page
needed its MIDI Out Sync Gen parameter setting to MTC so that the multitracker
would output MIDI Time Code, and I also checked that the MMC Mode switch in
the MIDI settings page was at its default Master position. Transmission of SysEx
messages is usually switched on by default, but I also checked this while I was in

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the MIDI settings page. In my own studio I run a VS2480 synchronised to a

Yamaha QY700 hardware sequencer, which I'd brought along so that we could
check that the college's VS2480 was working correctly before we delved into
Cubase SX. Having connected the multitracker's MIDI Out to the MIDI In of my
QY700, the sequencer synchronised immediately, so we re-plugged to the studio
computer's MIDI In and set about sorting Cubase SX.

Being more of an Apple Logic user

myself, I asked Carl to navigate to
Cubase SX's Synchronisation Setup
window, where we selected MIDI Time
Code as the Timecode Source and
specified the MIDI interface input port
to which the VS2480's MIDI Out was
connected. Finally, MIDI Machine
Control was activated so that Cubase
would respond to the VS2480's
transport controls. Pressing Play on In order to get Cubase SX synchronising to
the VS2480 at this point had no effect the VS2480 (which was now sending MIDI
on Cubase SX, so we had a quick look Time Code and Machine Control messages),
at the Transport bar and realised that Cubase SX's Synchronisation Setup window
was configured as shown. In addition, the
the Sync button wasn't yet switched Sync button on the Transport panel was
on. However, even switching that on switched on (inset).
yielded no result.

Knowing that we'd tested the VS2480 with the QY700, there was little doubt that
the multitracker was set up correctly. Furthermore, MIDI was definitely reaching
the sequencer, as could be seen from the MIDI input meter on Cubase SX's
Transport panel. This led us to suspect that we hadn't routed the MIDI Time
Code correctly in the sequencer, so we tried setting different MIDI interface input
ports from the Synchronisation Setup window. All of a sudden Cubase SX sprang
into life, synchronising perfectly with the VS2480. When we'd first configured the
Synchronisation Setup we'd set the wrong MIDI input port! It's at times like these
that it pays off to have worked methodically — if we hadn't tested that the
VS2480 was set up correctly before tangling with Cubase SX it could have taken
us much longer to get to the crux of the problem.

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VS2480 Tips
While I was working on Carl's
VS2480, I noticed a number of things
which would help him work more
efficiently. For example, he had
paired the channels of very few of his
stereo signals, because he wanted
independent control over the
individual pan parameters. However,
you can still access individual pan
settings for stereo-linked channels if
you cursor over to the Pan control in
the channel parameter screen and
press Enter — this brings up a little
window with the separate pan values
and the communal balance.
Another thing which speeds up the
mixing process is setting the Knob/Fader Assign Switch in the Utility menu's
Global pages to its Fader setting. This means that you can easily view and adjust
aux send levels across sixteen channels at once, which is very useful when you're
trying to remember which channels are sending to which effects. I also showed
Carl how the User Knob/Fader Assign mode can be used to transform the channel
fader into an EQ bypass switch for when you're setting up the channel equalisers
— after all, it's very good practice to keep switching the processing in and out of
circuit while you're deciding on the right setting.
The multitracker's internal patchbay was another source of frustration. Carl had
been having to keep resetting it for each new project, because the default
template wasn't suitable for the college's setup. Fortunately, the VS2480 patchbay
has an option to save routing templates, and I demonstrated that these could be
shared between different projects to solve his problem.
Finally, I demonstrated how, when using the VGA monitor option, you can lock the
unit's LCD to show the waveform display permanently — very useful if you do a
lot of editing. First you have to press the Page button by the LCD until the IDWave
option appears over the F3 button. This switches you to waveform view, and then
pressing F6 (IDHold) locks this view to the LCD — however, this waveform
display will always show the currently selected channel.

V-Fader Control

The second task was to sort out the best way to control Cubase SX's mixer from
the VS2480's faders and rotary controls. The VS2480 has a dedicated fader
layer (called V-Fader) within its digital mixer specifically designed for controlling
external units. You enter the V-Fader mode by holding down the Shift key and
pressing the bottom right-hand one of the Fader buttons above the master fader
— the button has the V-Fader label underneath it. Each channel of the V-Fader
layer sends out MIDI data on a separate MIDI channel, so the eighth fader and
rotary control both send out MIDI on channel eight, for example. The controls can

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Studio SOS

only send MIDI Continuous Controller messages, but you can specify which one
each individual control transmits from the Utility menu's V-Fader pages. To start
with we left the controllers at their default settings, so the faders and rotary
controls were sending out Continuous Controller numbers seven and ten
respectively.

Once again I connected the QY700 to

the output of the VS2480 to confirm
that the faders were indeed sending
out the data that we had specified. This
all checked out fine, so we plugged the
VS2480's MIDI output cable back into
the computer's MIDI interface before
opening up the Cubase SX Device
Manager window. Since there was no
pre-programmed template designed for
the VS2480, Carl used the Generic
Remote mode to set things up from
scratch. After choosing appropriate
MIDI ports, Carl created an audio
Fader in the Mixer window and this
could immediately be controlled from
the first fader in the V-Fader layer —
by default Fader 1 was set to be
controlled by Continuous Controller
seven messages on MIDI channel one.
So far so good. Because no pre-programmed control-surface
template was available for the VS2480 within
Cubase SX, Carl and Mike decided to set up
Next we tried to get the hardware
remote control from scratch using the
faders to reflect movements of the Generic Remote profile in the Device Setup
software faders, so that there would be window.
no mismatches between hardware and
software fader positions when using
Cubase SX's automation at mixdown. (Although the VS2480 has an internal mix-
automation system, this doesn't extend to controlling the V-Fader layer). Again, I
connected my QY700 up to the VS2480's MIDI Input and sent it MIDI Continuous
Controller number seven messages. Although the VS2480's MIDI indicator LED
showed that the data was reaching the VS2480, none of the V-Fader controls
budged. Referring to the VS2480's manual, there was no mention of the V-
Faders responding to incoming MIDI messages — only a list of which Continuous
Controller assignments were fixed for external control of the multitracker's audio
mixer. Just to test that the VS2480 was actually receiving the right messages, I
switched on its external MIDI control facility (using the Mixer Control Type
parameter in the Utility menu's MIDI pages), and sure enough one of the audio
mixer's input faders began following the movements of the assignable controller
wheel I'd set up on the QY700. (On my return to the SOS office, I contacted
Roland UK's technical support team, who confirmed that the V-Faders only send
MIDI data.)

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Studio SOS

Knowing that the VS2480's audio-mixer faders can send and receive MIDI
controllers, and that Carl was only using a handful of these for mixing the
VS2480's analogue inputs, I decided to quickly try using these to control Cubase
SX in place of the faders on the V-Fader layer. Unfortunately, the sequencer's
MIDI Thru function caused the hardware faders to 'fight' me whenever I
attempted to move them, because the Cubase SX audio Faders were re-
transmitting every Continuous Controller message back to the VS2480 as they
were received. In the end, Carl settled for controlling the Cubase SX mixer from
the V-Fader layer — at least this allowed sensible hardware control for projects
up until the point at which software mixer automation was used.

Mixing Horns & Rhythm Guitars

Before we headed back, Carl played us one of the

recordings that the students had made on the
VS2480 — a funk-style track with drums, bass,
guitars, and horns. The quality of the production
was pretty impressive, with a particularly nice drum
sound despite the main recording space being a 4 x
4m Esmono isolation room. This is a soundproof
metal booth which can be built inside your own
room to decrease sound leakage while recording. I
encountered one of these rooms for the first time
while interviewing Andy Cross for a Readerzone
article back in SOS January 2003, and was struck
not only by the effectiveness of the soundproofing,
but also by how dry the internal acoustics were. The The college's main recording
college's room was no exception — the room area was a large Esmono
absorbed so much sound that you even felt yourself isolation room. Although this
having to work hard to hold a conversation in there! provided great soundproofing,
the internal acoustic was
exceptionally dry, and this
Admittedly, the drum sound was quite dry, but it meant that Carl and the
suited the funk style of the track. I mentioned to students were having to rely
Carl that he might have problems capturing any quite heavily on delay and
reverb effects at the mixdown
more roomy rock sound, and suggested that stage. Tom & Mike suggested
recording drum ambience from out in the corridor that hardboard panels could
would probably help in such situations. I also be used to reintroduce some
suggested that he look at investing in one of the acoustic reflection when
latest convolving reverb plug-ins — with such dry recording things like drums
and acoustic guitars to make
recordings he has to rely quite heavily on artificial this less of a problem, and
reverb and delay, so I can imagine that high-quality also recommended that Carl
reverb would make a real impact at the mixdown look at acquiring one of the
stage. Another alternative where a more natural recent high-quality
recorded result was required would be to use convolution-based reverb
plug-ins.
hardboard panels to make the walls and floor more
reflective when recording acoustic instruments.

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Studio SOS

Speaking of reverb, Tom and I felt there was too much of it on the horn tracks,
giving rather a long reverb tail which seemed rather out of keeping with the
overall sound of the mix. The first thing to do was to take off all the processing
and check the balance of the three mic signals. After a small bit of track
rearrangement we managed to get the three channels up on adjacent faders, and
it turned out that only a little re-balancing was required to get the horns to sound
more 'authentic' — funk horn sounds are often pretty dry, so there was little need
to add reverb. I also thought that the compression settings used were a bit harsh,
and compromised the punchiness of the original dynamics. I felt that switching off
the dynamics processing was an improvement, and that where the odd phrase
poked a little too far out of the mix it would probably be best to sort this out with
the VS2480's automation at mixdown.

Tom pointed out that the rhythm guitars sounded quite middly and were having to
compete directly with the horns, muddling the overall mid-range. Carl and the
students had already EQ'd the guitars quite severely — low-end shelving and
high-frequency boost — to get them to cut through more, but there was more that
needed to be done. The EQ processor's high-pass filter proved a better tool for
the job, allowing us to progressively remove low end until the mid-range cleared
up satisfactorily.

Carl's Comments
"I really appreciate the guys from
SOS coming down to the college to
help us out. We all learnt an awful lot
about the capabilities of MIDI, and
also received some helpful tips
regarding our studio setup. Since the
visit we have progressed on from
just using the VS2480 as a mixer
surface to control volumes and
panning, and we're now using the
faders to adjust individual channel
settings (EQ settings, effect sends,
dynamics controls, and so on) and the parameters with Hypersonic and Reason.
We're also going to begin to investigate using the VS2480 to control the
parameters in our Korg Triton as well. Is there no end to this unit's capability? I
hope not, as it gives us something to do in the classroom... Thanks again!"

Homeward Bound

The flagship digital multitrackers are complex beasts which present a serious
learning curve to the home studio owner, and computer sequencers are even
more complex, trying to be all things to all people. So it's hardly surprising that
getting a multitracker to work with a software sequencer can be a headache.
Most apparent hardware/software faults in the home studio are the result of
incorrect setup — indeed, it was an erroneous MIDI port setting that threw a

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Studio SOS

spanner in the works when we were trying to synchronise Carl's PC sequencer to

the VS2480. However, the good news is that you can solve many studio
configuration problems just by approaching them methodically, and a good
technique here is to eliminate individual elements of a malfunctioning system
from consideration until you can home in on the exact culprit, just as we did by
testing the Cubase SX and VS2480 MIDI functions with the QY700 this month.
Even the most complicated studio conundrums can usually be resolved if you
can whittle them down to their root cause.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
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Web site designed & maintained by PB Associates | SOS | Relative Media

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 The Dust Brothers

In this article:
Early Beginnings
The Dust Brothers
Boutique Sounds Sampling, Remixing & The Boat Studio
The Boat Published in SOS May 2005
The Benefits Of Print article : Close window
Sampling
People : Artists/Engineers/Producers/Programmers
Old Methods, New
Tools
Boat For Hire
A Little Home Studio
A Piece Of History The Dust Brothers changed the course of record production
Things To Come with a new approach to sampling. In their first ever in-
depth technical interview, John King and Mike Simpson
explain their unique way of making records and open the
doors of their remarkable LA studio, The Boat.

Paul Tingen

In 1989, the Beastie Boys' Paul's Boutique single-

handedly redefined a whole musical generation's
approach to sampling. The musical masterminds
behind the album were the Dust Brothers, two hitherto
unknown college whiz kids who had created the
musical backings from collages of their favourite
recordings. Paul's Boutique was awash with innovation
— it reputedly featured the first recorded instance of
intentionally added vinyl crackling noises — and it
turned the Dust Brothers into the Godfathers of
sampling.

Since then, Mike Simpson and John King's career has

taken in a diverse succession of projects including
Technotronic's Trip On This (1990), the Rolling Stones' Photos: Mr Bonzai
Bridges To Babylon (1997), Hanson's Middle Of The Dust Brothers: John King
Nowhere (1997), Santana's Supernatural (1999), Linkin (left) and Mike Simpson.
Park's Hybrid Theory (2000) and Tenacious D's
eponymous album (2002). As a staff producer for Dreamworks, Simpson also produced
Eels' Beautiful Freak (1996). Perhaps the most influential and artistically successful of
all, however, was Beck's 1996 album Odelay. Simpson and King later contributed to the
same artist's Midnite Vultures (1999), and their relationship continues to this day with
the brand-new Guero, which appears to set them on course for another round of
limelight-hogging in 2005.

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The Dust Brothers

Meanwhile, the Brothers have also been developing a state-of-the-art recording studio.
The Boat was opened for commercial use in 2003; based around a vintage Neve 8028
desk from 1969 and a Pro Tools HD3 system, it has become one of Los Angeles's most
happening studios (see boxes).

Early Beginnings

"My musical background came from collecting records," recalls Mike Simpson, "and sort
of studying the sounds and arrangements and the way they were recorded. I grew up in
New York listening to black music, and I was there for that famous summer in the mid-
1970s when hip-hop started. When I moved out to California in 1978 there was no hip-
hop or rapping culture here, so I lived on cassettes sent to me by friends. In 1986 I
enrolled in a local community college, where I did a class in electronic music. That was
my first opportunity to really do computer sequencing and work seriously with samplers.
I'd been doing a college radio show since 1983, during which I played hip-hop music,
and I began playing the music I was putting together in class on the radio show. I met
John in 1985, and he joined me in putting on the show and putting together tapes."

King and Simpson's hip-hop radio show caught the ear

of rapper Tone-Loc. He had just signed to the newly
formed record company Delicious Vinyl, who in turn
were busy setting up their own studio. Tone-Loc and
Delicious Vinyl invited Simpson and King to help out
producing records and setting up the company's studio.
When their name was about to appear on a record
sleeve for the first time, on a single by Young MC, the
duo decided on the name the Dust Brothers. Reputedly
it's a reference to angel dust, the drug, but this turns out
to be only an aspect of the truth.

"King and Simpson are pretty common names," explains the latter, "and we decided that
we'd better come up with a cool name. At the time we were bringing back music that no-
one was listening to any more, so we wanted the name to be an anachronistic reference
to things of the past. While we were working for Delicious Vinyl, many people had been
describing our music as 'dusted,' and that's where we took the name from. The state of
hip-hop was pretty minimal at the time, and we were doing these very textural, tripped-
out, almost hallucinogenic remixes of things. Angel dust was just an additional whacked-
out reference that also fitted with what we were doing."

Boutique Sounds

During these first years in the recording studio, the Dust Brothers were predominantly
engaged in dusting down, or perhaps dusting up, old favourite records, and giving them
new leases of life. They applied their sampling skills with considerable success on Tone
Loc's Loc'd After Dark (1989) and Young MC's Stone Cold Rhymin' (1989). Then they hit

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The Dust Brothers

upon a project that became the landmark Paul's Boutique. Did the duo actually set out to
change the music industry, or did they just stumble into prominence? The latter, claims
Simpson, with estimable modesty.

"Sampling was just a hobby for us. It was just

something we did for fun while we were in
college. John was destined to become a
genius computer programmer, and I was
going to enrol in law school. We never had
any intention of making records. I didn't even
know what record producers did at the time. In
the course of doing samples for Delicious
Vinyl Records, every once in a while we put
something together that seemed just too
dense and too busy and too crazy for a rapper
to rap on, and we put these tracks aside as The Boat's impressive list of outboard
instrumental Dust Brothers tracks. Then the equipment includes a mouth-watering array
Beastie Boys wandered into the studio, and of vintage mics, preamps and compressors.
heard one of these tracks, and they loved it.
That's how the album got started.

"Up until that point in hip-hop, people had been using samples very sparsely and
minimally. If anything, they would use one sample in a song and take a drum loop and
that would be the foundation. But what we were doing was making entire songs out of
samples taken from various different sources. On Paul's Boutique everything was a
collage. There was one track on which the Beastie Boys played some instruments, but
apart from that everything was made of samples. But we never had a grand vision of
trying to make groundbreaking music. We just enjoyed making music in a way that was
an extension of our DJing, combining two or three songs, but with greater accuracy than
you could do with turntables."

The significance of Paul's Boutique is illustrated by a web site (www.moire.com/

beastieboys/samples) on which fans have collaborated in spotting all the samples on the
album. For the track 'Shake Your Rump' alone the web site lists samples taken from
records by Sugarhill Gang, Funky 4+1, James Brown and Afrika Bambaataa, Bob
Marley, Paul Humphrey, Led Zeppelin, Harvey Scales, Rose Royce, Ronnie Laws, Foxy
and Alphonse Mouzon. ("I think they got all of them," says Simpson.) Yet most of the
samples used on Paul's Boutique were cleared, easily and affordably, something that
Simpson says would be "unthinkable" in today's litigious music industry. The album will,
therefore, always be unique.

In the early 1990s, with anti-sampling legislation and attitudes tightening, the Dust
Brothers were mainly busy remixing, while cutting their teeth on engineering, composing
and producing. Their increasing fame offered them lots of opportunities to apply these
skills, but Simpson admits that they spent several years climbing a steep learning curve.

"It was tough. People asked us why our stuff from the late 1980s sounded so good, and
we said that it simply was because the original recordings that we sampled sounded so

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The Dust Brothers

good. After Paul's Boutique we signed a publishing deal that gave us some money to
live, and we took the opportunity to buy a house and build a home studio. We spent
three or four years there learning how to record and engineer stuff. Paul's Boutique and
Odelay were sort of the crowning achievements, but there were a lot less great records
in between."

The Boat
The Boat, in Silverlake, Los Angeles, was built in 1941 for live radio broadcast. The Dust
Brothers acquired it in 1997 and proceeded to completely renovate it. The building looks
like a boat — hence its name — and its striking architecture makes it a Silverlake
landmark. A quick look at the lengthy equipment list reveals the old-meets-new
philosophy behind the place. On the new side there's the Pro Tools Accel system and Pro
Control console, Ableton Live software, and a list of Pro Tools plug-ins so long you can't
even begin to shake a stick at them.
At the same time, pride of place goes to the 1969 56-input Neve 8028 desk, with 1073
and 1066 modules and four built-in Neve 2254A compressors. There's also a vintage
analogue MCI JH114 16/24-track tape recorder, and an astonishing amount of vintage
and/or valve outboard gear and microphones. The list is far too long to reproduce here,
but is available on the studio web site at www.theboatstudio.com.

"Combining old and new has been our goal

as musicians and producers and now as
studio owners," asserts Mike Simpson.
"We've made our name staying abreast of
the latest technology, but at the same time
we've used that technology to sample all
those brilliantly recorded recordings from
the 1970s. As it got more and more painful
to use samples, we realised that we were
better off creating those sounds ourselves,
and the way to do that is to get all the
equipment it was originally created on."
"I love collecting gear and have a ridiculous The Boat really does look like, well, a boat —
collection of outboard and microphones and right down to the portholes and gangway!
instruments," John King fills in. "After I
collected all the gear I could handle, I kept finding more, and that's how I started acquiring
what we have at The Boat. The old gear has the aspect of a vintage car. It's beautiful, it's
historic, there's a definite nostalgia to it."
Yet nostalgia is not the Dust Brothers' driving force. Their bottom line is that analogue,
vintage and valve gear still sounds better than even HD digital. What they aim to do with
The Boat is marry the convenience and functionality of digital with the superior sonic
qualities of analogue.
"The new Pro Tools HD system sounds a lot better than the old system," opines Simpson.
"But there's still a huge gap between analogue and digital. HD digital still lacks a certain
emotion. The late 1960s and early 1970s probably saw the pinnacle in sound
reproduction. The imaging and dynamics are just so much better. Also, I'm sort of a bass
junkie. I like it when you can really feel the low end, and with those late-'60s and
early-'70s records was the last time you really felt that, at least in the rock and soul stuff.
Now everything is so thin and brittle, it makes me cringe when I hear snare and kick

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The Dust Brothers

drums. Obviously the centrepiece of the studio is the wonderful Neve console. It's such a
nice-sounding board. Being able to record and pump channels back through the console
really makes a huge difference."
The Boat also sports an impressive array of monitors: Urei 813C, plus Genelec 1031A,
Yamaha NS10, Westlake Audio BBSM6 and 10, JBL 4408A, Tannoy AMS 10A and
Auratone 2B monitors. All this combines to make it the ultimate mix environment,
according to John King. "One thing is that the mixes we did here sounded fantastic
everywhere else. I really trust the room and the monitoring, especially the Urei main
monitors, which are great. The only thing we've mixed so far at The Boat is Beck's new
album and I'm so happy with how that came out. We didn't really use much outboard
during the mix, because it was already sounding so great. We used the SSL compressor
pretty much on every mix. If nothing else it's a security blanket, and it lets you adjust the
levels nicely as the mix is going back into Pro Tools."

The Benefits Of Sampling

The Dust Brothers' house was in Silverlake, Los Angeles. They created their studio in a
spare bedroom and, pushing the angel dust reference, called it PCP Labs. The studio
existed from 1991 to 2001, and sported a 24-channel Soundcraft Spirit desk. "We loved
this board," says Simpson. "We tracked a lot of great songs through this board, including
all the songs from Odelay." PCP was split into two control rooms in 1996, with two
Yamaha 02Rs in King's room and a 64-input Amek Einstein in Simpson's section.

Despite the legal issues, substantial elements of the Dust Brothers' college-era collage
approach to music continued to survive, and with Beck's Odelay they finally found the
perfect marriage between this and their newly acquired engineering and production
skills. Beck's attitude and way of working gave them a perspective on an additional
reason why previous efforts had met with such variable success. The Dust Brothers
found that musicians who were not familiar with the new technology often approached
recording in a manner that was at odds with their way of working.

"We sometimes would record musicians the way you would traditionally record a live
band, and then add samples," Simpson explains. "Not very successfully, I would say.
Because for some of the more traditional musicians we worked with, the idea of
sampling was sort of foreign, and they wanted to play things right. But we don't
necessarily want you to play things right, we want you to play things cool. You play over
a groove until you have a good bar, and then we take that bar and loop it. I always say
that our best music comes from mistakes that happen. You're trying to do one thing, and
then someone makes a mistake and that mistake ends up being the hook of the song,
the coolest part of the song.

"Beck really understood the benefits of sampling from the beginning, and he understood
all along what our goal was. It's a different mindset for a musician, and Beck really got
that. He's totally uninhibited, and not necessarily trying to play it right. He's just trying to
play it with attitude and flavour. That makes it easy for us, and it's why we have had
such great success in working with him. He really understands the medium and what we
do, and hand-delivers us these great out-of-control performances that leave us with
tracks that we can draw all these great loops from."

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The Dust Brothers

Old Methods, New Tools

Guero is Beck's eighth studio album, and as on Odelay, Simpson and King worked on
almost all of the album's songs. "Beck wanted to do more of a contemporary R&B
record," says Simpson. "To me it picks up where Odelay left off. There's a little bit of
everything: there are some rock songs, some great hip-hop songs, some great blues-
inspired songs, some 1980s dance-inspired songs, and so on. It's a melting pot of all the
types of songs Beck loves. Sometimes there will be a few genres within one song. But
some songs that were more rock were left off because they didn't fit the mould.

"The way it started was that we had worked with Beck on some songs for Midnite
Vultures, and we finished off only two in time to make the record. There were six other
songs that were pretty well developed, sometimes only needing Beck to finish his vocals
and some sprucing up here and there. Beck loved those songs, and wanted to revisit
them. So we pulled them up and took some of them apart and reconstructed them.
Pretty much the moment we came into the studio and heard the stuff, the feeling was
'Yeah, let's do new stuff too.' We began this the way we did with Odelay, pulling up
loops or samples, pulling out records, saying 'Oh yeah, I want to do a song that sounds
like that.' But whereas Paul's Boutique was made from samples, a lot of Odelay and the
new record is more based on sound than on the samples themselves. We were after the
sound and the vibe more than anything else.

"Our [non-record] samples come from years of tracking. Everything we ever tried or
worked on, apart from the Stones' material, which we were forced to turn over, ended up
on hard disk. When making backups we would pull out all the beats and other samples
and put those on a separate drive. At one point we had one of our employees compile
all the samples from throughout our history, and we now have one sample library called
Dust Beats, containing all the beats in one folder, and there's a folder with bass grooves,
and guitar grooves, and so on. Using Ableton Live you can so effectively scroll through
these sample libraries, and see whether they fit."

John King agrees that "the creative process in making the new album was very similar
to the making of Odelay," adding, "it was about Mike, Beck and me in a room, having
fun, coming up with ideas, then embellishing and finishing them." Yet King quickly goes
on to elaborate on the dissimilarities. "The major difference is that we're doing

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The Dust Brothers

everything with Pro Tools now. For Odelay we

used Studio Vision software and Digidesign
hardware, with a two-channel interface, so we
could only record or play back one or two
tracks of live audio at the same time. I had to
take everything that we did and convert it into
samples that then could be played back with
the Samplecell card, and make MIDI notes
that corresponded with wherever I wanted the
samples to happen. But for the new album we
had many inputs and outputs and as many
tracks as we wanted. We don't even use a The main live area at The Boat. Miked up at
sampler any more, because there are so the front is a Fender Rhodes Suitcase piano.
many tracks. And so we got to layer more
vocals and instruments, using multiple mics on instruments, which we couldn't do before.

"For this new album we began songs written from scratch in Ableton Live, running with
Pro Tools. I love Ableton. It's a quick way for me to get the ball rolling, and quickly make
ideas happen that Beck likes and then plays over. I get that going and then I set up
microphones, like the SM57 combined with Neumann 47 or 47 FET for electric guitars
— I tend to use 47s on almost everything — sometimes a Royer 122 ribbon mic, using
an LA3A compressor, and a 47 with Royer for acoustic guitars, and so on.

"I record all that stuff in Pro Tools, and pick out my
favourite things and cut and paste and create verses
and choruses. Then I see what Beck likes and start
some arrangement. We continue to go back and forth
with each other until I feel the song is there, at which I
hand things over to the studio's Pro Tools assistant,
Danny Kalb, who continues to work with Beck on
overdubs.

"On one of the songs, I think it was called 'Emergency

Exit', there are all these strange digital artifacts and
stretching noises going on that Ableton was making. I
think it has some loops that went at half speed. The average person would say 'That
sounds horrible, they need to improve their stretching algorithms,' but Beck was like
'Wow, that sounds amazing.' When he says that I just go with it. A lot of the exploratory
nature of the work we do with Beck comes from his open-mindedness and eagerness to
do new things. The same happened with several effecty plug-ins, like Sound Toys and
some of the GRM Tools stuff, which I used for creating crazy, freaky effects. Beck
always wanted me to record while I was doing that.

"In terms of the end result, there's more live playing, and it's thicker with sound, but the
spirit is similar. One thing Beck remarked on was that we did everything so fast this time.
He remembered with Odelay having a lot of time to sit around and write lyrics or
melodies, while I was converting playing into samples and thinking about how to make it
all work. By the time I was ready for him it seemed like he had a finished song ready to
go, and we'd do a first take. But this time he had to sit and listen more to what we were

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The Dust Brothers

doing, because we would accomplish everything so quickly."

Boat For Hire

The Boat studio was originally put together for the Dust Brothers' own use, but in 2003,
they decided to turn it into a fully commercial operation. "A year after we had The Boat up
and running, we found that neither us of was using it that much," explains Mike Simpson.
"Instead we spent most of our time at our own home studios so we could be closer to our
families. So we decided that it was a shame to have The Boat just sitting there, and
began interviewing studio managers."
Enter Adam Mosely, an engineer and producer with an impressive track record in his own
right. Cutting his teeth at the legendary Trident studios in the late 1970s, the Briton
worked with greats like Phil Ramone, Tom Dowd, Mutt Lange, Steve Lillywhite, Ken Scott,
Mike Stone, and many others, recording the likes of the Cure, Wet Wet Wet, Kiss, Rush
and so on. "Adam really brought The Boat to life," says Simpson. "He had great ideas and
started pulling clients in."
"I thought it was an astonishing place with
the most incredible potential," enthuses
Mosely. "The Dust Brothers had already
bought the most amazing equipment, and
would buy even more, and the Neve board
was great. It originated in AIR Studios in
London, where it had been the second
board George Martin ever bought at AIR.
Rupert Neve customised it for him. Then it
travelled to Sweden, where it was more
used in the dance arena. The Dust Brothers
located it, bought it, and shipped it back to
LA, where it was retro-ed back to its original Engineer and studio manager Adam Mosely
state." at The Boat's vintage Neve desk.
The Boat was opened as a commercial
studio in January 2003, and since then artists like Madonna, Avril Lavigne, Marilyn
Manson, Lenny Kravitz, Don Was, and many others have explored its best-of-the-old-
meets-best-of-the new characteristics. "I sensed that the industry was going back to a
more old-fashioned approach again," says Mosely. "They are wanting to get back to a
bigger sound, and away from the mid-range compressed sound. Seeing the equipment in
the studio, the opportunity was just a no-brainer. The Neve has such a huge, warm,
dynamic sound, it's phenomenal. Combined with the vintage tube gear and microphones,
it really enhances Pro Tools HD3, which already has an incredible sound at 88.2 or 96
kHz, with things coming back exactly as you hear them.
"A lot of modern boards don't have the dynamic range of the Neve, and there's a big
difference between Pro Tools Sessions that have been recorded here, and elsewhere.
But even when people have recorded elsewhere, when they put their tracks back through
the Neve for mixing, the sound becomes so much bigger. So what we have done is set up
a procedure that makes it possible for people to mix easily via the Neve and have recall.
We didn't want to introduce total recall on the board, because of the sound, but I realised
that we could create total recall simply by using the oscillator to align the monitor return
faders.
"What most people do is mix in stems, ie. mix in Pro Tools to stereo pairs, and send these
pairs through the 24 monitor returns on the board. We then align the monitor faders at

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whatever level people want it to come out at, and they can come back a week or month
later and do a recall. We also got the SSL X-Logic FX384 compressor, because it's such
a recognised industry sound, and the GML EQ, which is probably the cleanest clearest
EQ there is. With the rest of the vintage outboard, and us having every plug-in on the
market, people have a complete mix solution here."

A Little Home Studio

With The Boat being almost constantly booked out, the Dust Brothers can hardly get into
their own studio any more, and so both have their own, not-to-be-sniffed at home
facilities. Their gear mania doesn't only cover "every keyboard ever made", it also
extends to a huge collection of vintage and/or valve outboard gear. Much of it is located
at The Boat, but substantial amounts are also in use at their respective home studios.

Simpson's "little home studio setup" contains a full Pro Tools HD3 rig, "with a couple of
Neve mic pres and LA2A compressors. Basically all the stuff we have at The Boat,
minus the Neve desk. I have probably one third of what The Boat has in terms of
outboard gear."

"I have converted one of my two houses into a

studio complex," King chips in, "where I have
two studios. We moved here six months ago.
I've always had a studio in my house, and in
the last house I lived in we converted this
huge beautiful living room into a huge studio
[called The Medina]. I have Pro Tools HD3 at
my current house, with Pro Control, so I can
mix virtually. I also have various Pultecs, a
couple of LA2A compressors, a couple of
1176s, LA4A, RCA BA6A, Neve 1073, 1076,
Neve stereo compressor, Neve mastering EQ, Some more unusual keyboards in The Boat.
Manley massive/passive, Manley DI, Manley The grey instrument at the back is a Mattel
mic pres, Telefunken V72, V76, Mastering Optigan; at the front is a Wurlitzer organ.
Labs mic pres, Distressor, the SSL
compressor and all the great microphones.

"And we use tons of synthesizers. You name it, we have it. They are all hardware
synths. I don't like using soft synths. I like to have knobs. I don't really like presets, I like
to be able to tweak things. We have every keyboard ever made. Many of them are in
The Boat, but we also have them in storage. I have closets here at home that are
stacked floor to ceiling with all kinds of crazy keyboards. We have all kinds of Moogs
and I'm a big fan of the whole Korg line of keyboards, so I have Korg polysynths and
Monopoly. We mostly bought them via eBay, and few of them are MIDI-fied. They are in
their original state. I can play them well enough to get something into a computer and
make it sound good."

Despite their avalanche of rare and vintage gear, the Dust Brothers wax most lyrically
about Pro Tools and especially Ableton Live, repeatedly saying that they now finally

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have the equipment at their disposal that they have "always dreamed of". "Because of
the way I produce things and create things with samples and loops," states King,
"especially Ableton is what I dreamed of back in the mid-1980s, when I was using
primitive software with numbers flashing across the screen. I had to program it all and it
was just so complicated. I knew that the ability would be there to do what Ableton does,
which is that you can work with loops and time-stretching in real time. If I have a beat
going or even maybe just a tempo running, I can click on Files in my library and then on
Samples, and audition beats or music or guitars or basses or whatever, and they will
instantly play back to whatever I'm playing.

"In the past I had to pull the sample up,

choose which one might work, trim it, tune it,
sync it, and after a long process I could decide
whether it really was cool or not. Now I just
click and instantly hear things from my library
playing in sync with the song. It's exactly what
I need, and allows me to focus on the creative
aspect and not get distracted by technical
things."

"The very first sampler we had was a Roland

F10," recalls Simpson, "and then we went with Wurlitzer and Fender Rhodes electric pianos
the Akai S900. Those were still mono in the main live area.
samplers. Then we dabbled with the SP12,
the predecessor of the SP1200, and then we had a Roland S770, which I think was the
first stereo sampler. We did all of Paul's Boutique on an Emax HD, which was mono and
12-bit and had a 22kHz sampling rate. So we had plenty of experience of the primitive
domain of early sampling: low bit rate and low sampling rate. But we've never been in
love with the degraded sound of those early machines, we were always trying to make
samples sound better. We had Pro Tools in our heads before it even existed. Since both
John and I came from a computer background, we knew what computers were capable
of, and we were kind of bombed that the samplers were still so lo-fi or hard to use.

"The sequencer we used on Paul's Boutique was very primitive software called Texture
by a guy called Roger Powell. This was when computers still had no user interface, it
basically was just a bunch of letters and numbers across a green screen. After that we
used this very primitive sync box, the JL Cooper PPS1, that allowed us to sync the
computer to tape. We also had an Allen & Heath console with very primitive automation
with which you could create mute events. So we basically filled all tracks on a multitrack
with loops, and arranged songs by using these automated mute things. It was such a
painful process. I remember thinking 'God, why couldn't we just have a timeline across a
screen and chunks for each sample and a visual representation for the waveforms
across the time line? Why do I have to sit here and type all these numbers and MIDI
times?'"

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The Dust Brothers

A Piece Of History
John King and Mike Simpson are quite happy to see their old sampling and sequencing
gear relegated to the dustbin of history, but they had to go back to their bad old Emax HD
for a song called 'Hell Yes' from the new Beck album. "Beck was into a song that I had
carried around on a cassette since 1989," King elaborates. "It had been composed on an
MPC 60 and the Emax sampler, the same one we used on Paul's Boutique. At the time I
had just bought some new records and had pulled a few things and programmed this
beat. It was very hip-hop.
"Beck and I decided to use it, and started working with it from cassette, while my
assistants and I were frantically searching all storage areas for the original disks. When
we finally found them I had to contact the Experience Music Project Museum in Seattle,
because we had donated our Emax sampler to the hip-hop exhibit for its grand opening.
They sent the sampler back to us, and I popped in the disk and lo and behold, it worked!
We also managed to load the MPC60 disk into Mike's MPC2000, so we were able to get a
more pure sound than we had from the cassette, which had a lot of hiss on it and didn't
have a lot of dynamics."
This might sound like a lot of trouble, but attempting to recreate the original from scratch
would have risked losing the magic. "I certainly know better than to try to re-record or
recreate things that sound cool," says John King. "Record companies used to do demos,
and that's something Mike and I always fought against early in our career. When
something sounds great, it's done. You don't want to go back and re-record something
that sounds great. The way we recorded with computers in our history, the quality was
always good enough. You don't want to repeat golden moments. We always felt like 'We
don't do demos, we only do finished product.'"
King still has an MPC 3000 and an MPC 4000, and remarks "It's more fun to have pads to
bounce than mousing in notes. But to be honest, I rarely use it."
"We'll do a bit of MIDI programming," Simpson adds, "usually to augment a loop. We may
program in some 808 kicks or snares. We also use Reason sometimes to augment beats."

Things To Come

So if Ableton Live has finally made the Dust Brothers' dreams come true, what ambitions
do they still hold for the future? Above all, it seems, they'd like to do an album as artists
in their own right. One of their soundtrack albums, 1999's Fight Club, was released
under the Brothers' own name, but John King stresses that "Fight Club is not a Dust
Brothers album, it's a Fight Club album. It was music done for a film and not meant to
stand alone. We've been working on a Dust Brothers album since 1987, but songs
continually get given to artists we work with. And now we're both so busy with things
we're working on, and we both have families, and there's life, that it's hard to get round
to doing your own thing..."

Published in SOS May 2005

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The Dust Brothers

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/The%20Dust%20Brothers.htm (12 of 12)9/27/2005 9:25:20 PM

 The World At Your Fingertips

The World At Your Fingertips

Paul White's Leader
Published in SOS May 2005
Print article : Close window

People : Industry/Music Biz

I had planned to write a Leader on an entirely different and arguably less

contentious subject this month, but after seeing the response to last month's
offering, I couldn't resist continuing the theme, especially after one reader
suggested that I had, in effect, lit the blue touch paper and stood well back!
Predictably, there were missives from keyboard players who opposed my
viewpoint that the guitar is a far more expressive instrument, so I feel I should
explain some of my reasoning.

Nobody would argue that a well-played, well-composed piano piece can

convey a lot of musical emotion, but I still stand by my statement that keyboard
players have very limited means of injecting expression into the notes that are
played. In the case of a piano, you can control the intensity, timing and sustain
of a note, and you can play a great number of notes at the same time, but with
a guitar, you can control the level, the sustain, the timbrality via damping,
picking position and partially pinched harmonics, the precise degree of pitch-
bend and vibrato, and you can produce a number of non-pitched effects as well as electronically assisted
effects.

In fact, it's probably fair to suggest that the more polyphony an instrument offers, the less expressive it tends to
be (another great subject for forum debate). A guitar used to play solo lead lines, for example, is far more
expressive than one used to play chords, though even with chords you can go way beyond what a keyboard
can do. Same with a violin, and solo instruments such as sax and flutes can be enormously expressive, as
everything goes into that one note. There's simply no argument — pianos and organs are poor controllers
when it comes to putting expression into synthetic sounds that need articulation and subtlety. If you don't
believe me, try playing a sax sample from a MIDI piano and then from a MIDI guitar and see which is closest to
sounding like the real thing. At the risk of pouring more fuel on the flames, I'd suggest that synths became less
expressive and less interesting as soon as they became polyphonic. In their mono guise, they were
instruments of wonder, but once polyphony came along, we ended up with something little more expressive
than an organ played through a wah-wah pedal.

Of course keyboard players will tell us that they have MIDI controllers to help add expression, and they are
right, but the way these work is pretty limited and the expression has to come from juggling these additional
controls rather than from the way the note itself is played. Perhaps the most natural hardware controller is the
breath controller, but sadly these never really caught on, because drool isn't cool! If you need more convincing
that MIDI controllers aren't the same thing as a direct means of adding expression and timbral variation,

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 The World At Your Fingertips

consider the analogy of the automated sci-fi monster controlled by complex hydraulics. These remote controls
are akin to MIDI controllers and allow some semblance of natural movement to be emulated, but they still don't
come close to the fluid movement of a real person or animal.

Finally, I didn't write last month's Leader to run down keyboards — we all use them, and for some things they
are very good — but when it comes to injecting expression into a wide range of synthetic non-keyboard-style
sounds, I just don't think they're up to the job. They're also a lot more reliable than the current crop of pitch-
tracking guitar synths — a technology that I think is headed in the wrong direction. True expression comes
from fingers on strings or lips on reeds, not from add-on wheels, levers and optical beams — fun though all
these things are to play with. Given that the guitar is capable of translating so much of the player's actions into
timbral and dynamic variations, and accepting that a lot of people play guitar, it just seemed to me that a new
form of synthesis driven directly from the guitar strings and linked to their harmonic content would be a better
way forward than trying to force the keyboard into being something that it's not, and never will be.

Paul White Editor In Chief

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/The%20World%20At%20Your%20Fingertips.htm (2 of 2)9/27/2005 9:25:24 PM

 Digital Performer: EQing Tracks

In this article:
Keyboard Shortcuts
Digital Performer: EQing Tracks
Quick Tips Digital Performer Notes
Published in SOS May 2005
Print article : Close window

Technique : Digital Performer Notes

This month, how to tweak your tracks with MOTU's

answer to Sony's Oxford EQ and speed up your
workflow with the essential DP keyboard shortcut
selection!

Robin Bigwood

Users of Digital Performer who've upgraded to version 4.5 now have at their
disposal what I think is one of the best software EQ plug-ins out there:
Masterworks EQ. In sonic terms, it bears comparison with the very best offerings
in both native (Waves) and 'powered' (TC, Mackie) plug-in formats, but remains
extremely flexible and easy to work with. Use it on a track and its presence is
often very obvious. It can do 'warm and subtle' as well as 'harsh and edgy', but it
doesn't pretend to be phase-accurate, and the resultant 'smearing' gives a
reassuringly analogue-style effect that can add warmth, life and fluidity to many
sources. If you want phase coherence and ultimate accuracy (for classical or
mastering work, for example), you'd be better off looking at something like the
Waves LinEQ or the superb and under-rated Periscope, by Audio Ease. But for
general multitrack mixing, Masterworks EQ should find plenty of uses.

Fundamentally, MW EQ is a five-band
parametric design with additional low-
pass and high-pass filters, but each of
the seven bands can be enabled
individually — in fact, by default the
plug-in instantiates with none of the The characteristic 'overshoot' curve that
bands enabled — so things can be as Masterworks EQ's shelf filters can produce
simple or complex as you like. Also, when used with a high Q value — much
there's nothing pre-determined about smoother sounding than a conventional shelf.
the frequency ranges on which the five
parametric bands work so, in effect, MOTU's 'Low', 'Low Mid', 'Mid', 'High Mid'
and 'High' terminology is provided for convenience only. You could, if you
wanted, set the 'High' centre frequency at 20Hz! What is set in stone, though, is

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Digital Performer: EQing Tracks

the way in which only the Low Mid and High Mid bands are capable of being
shelving-type filters. The other three main bands are peaking (or 'bell') type only.
All in all, it's a very flexible system that provides plenty of open-endedness but
retains a link to the 'British large console EQs' that it's apparently modelled on.

What really sets Masterworks EQ apart, though, is the selectable filter

characteristics for each band. To start with, the low- and high-pass filters are
capable of cutting at anything from 6dB to 36dB per octave, and those high-value
settings are well into 'surgical' territory. Then, for the other bands, there are four
different filter types (plus a shelving option for LM and HM bands). These
primarily affect the bandwidth behaviour of the EQ boost (or cut), causing the Q
parameter to act in subtly different ways. Here's a run-down of the four filter
types:

I: This is the most like DP's long-

standing Parametric EQ plug-in.
Bandwidth increases as gain
increases, so at lower gain settings
bandwidth can end up very narrow. For
little tweaks here and there this is
probably the most 'surgical' mode, and
is the most digital-sounding of all the
filter types.

II: Here, bandwidth remains virtually

unchanged regardless of gain. This
type is noticeably 'warmer' than type I
but is capable of very narrow boosts
and cuts at extreme gain and Q MOTU's new Masterworks EQ in all its glory.
settings, so is particularly good at Here five bands are enabled, as well as the
taming troublesome hums or real-time FFT spectrum-analysis feature.
resonances.

III: Warmer again, bandwidth increasing with gain. At low gain settings, very
smooth, broad curves are possible. This type is excellent for dialling in relatively
small amounts of boost or cut on individual tracks. It's a contemporary-sounding
EQ that's most like the best analogue EQs and other decent digital EQs such as
Waves' Renaissance EQ.

IV: This filter type is capable of almost 'table-top' EQ curves and has the very
widest bandwidth for any gain or Q setting. It's something of a blunt tool for
making corrections to individual tracks but is superb for subtle treatments on sub-
mixes or even final mixes.

Even the shelving curves, available on the LM and HM frequency bands, are
quite 'grown-up' and analogue-like. The Q control remains active for shelving-
type curves and effectively determines the steepness of the start of the shelf
'curve', but high Q settings cause an 'overshoot'. Visually the effect is obvious

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Digital Performer: EQing Tracks

(see the screen below) and musically the overshoot leads to a smoother result.
The final trick up Masterworks EQ's sleeve is a real-time FFT frequency-analysis
display, superimposed on the EQ curve window. I'd like a bit more gain on this
display, so that really low-level sounds are as visible as loud ones, but just as
with Periscope, it makes zoning in on problem areas that much easier.

Keyboard Shortcuts

With the advent of the Consolidated Window, the whole DP user experience has
improved no end. It's undeniable, though, that it's still faster to work with most
software using good old keyboard shortcuts. In this regard DP is better than most
apps: keystrokes can be set up (or modified) for virtually all commands via the
Commands window (Apple-L). But there are literally hundreds of keystrokes
already in force by default. Here's a round-up of those I find utterly indispensable:

Transport/Control Panel Window

The
vast

The keyboard shortcuts for selecting editing windows and tools in DP are amongst
the easiest to learn, and using them can massively speed up workflow.

majority of these are based around the keypad:

Play: Enter

Stop: 0 (zero)

Record: 3

Return to zero (or start of memory cycle): 1

Fast-forward: +

Rewind: 4

Memory-cycle on/off: 7

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Digital Performer: EQing Tracks

Metronome on/off: num lock

Count-off: =

Overdub record: *

You can also turn on Auto-record (punch in/out) by hitting Alt-3. Then define
punch-in and punch-out points by locating to the appropriate point in your
sequence and hitting F3 and F4 respectively. Similarly, memory-cycle start and
stop points are defined by F1 and F2. Those are two shortcuts I couldn't possibly
live without!

The Dot Trick

This is absolutely the best way to quickly move the playback wiper from place to
place in a sequence. Hit '.' (the full-stop key on your keypad, otherwise known as
'dot') and you can type in a value for the first part of whatever time format you're
using. If it's measures, you'll be typing in a bar number. Hit the dot again and you
can enter beats. Hit it once more and you can enter ticks. When you've defined
your desired playback position, just hit the Enter key and the wiper moves
straight there. The joy of this technique is that as well as being quick, it's also
extremely flexible — you don't have to enter all information for any given time
format. If you want to locate to bar 36, for instance, no matter where you are
already '[dot] 36 [enter]' takes you there.

Speaking of time formats, I find myself using Alt-Apple-T a lot to bring up the
Time Format window and switch between the formats on offer, especially when
moving from time-based to measure-based projects.

Windows

These are amongst the easiest shortcuts to learn, and the most useful. Hold
down Shift first, and you then get single-key access to all the main DP windows:

Tracks Overview: T

Sequence Editor: S

Graphic Editor: G

Mixing Board: M

Soundbites window: B (think 'bites')

Event List: E

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Digital Performer: EQing Tracks

Drum Editor: D

Audio monitor: A

Markers: K (you just have to remember that!)

Less well known, but a lovely shortcut, is Shift-F. This brings up the window of
whichever plug-in is in the uppermost slot (slot A) for any track that is selected.
What this means in practice is that you don't even need to have the Mixing Board
open to gain access to plug-ins, and since all plug-in windows have both track
and insert-slot pop-up menus, it's easy to access any plug-in. This works
brilliantly when you're editing virtual instruments in DP: just select something in
the Instrument track, hit Shift-F, and there's your instrument.

While we're on the subject of windows, remember that the Mac-standard Apple-
W closes any DP window, and an offshoot from this — Shift-Control-Apple-W —
closes all plug-in windows. Plug-ins' graphic interfaces, particularly if they're
animated in any way, can consume precious processor power, so this one is
worth learning.

Tools

Also easy to learn, the keyboard shortcuts for DP's editing tools beat using the
mouse and Tools palette any day. They're all genuine single keystrokes (no
modifier keys at all) and have a 'momentary' action, so when you let go of them
you should revert to whatever tool you last selected with the mouse in the Tools
palette, which most often is the 'arrow' or 'pointer' tool.

Pencil: P

Reshape: R

I-Beam: I

Zoom: Z

Scissors: C (think 'cut')

Mute soundbite: M

Scrub: S

Loop: L

Pattern brush: E (think of it as three bristles of a paint brush!)

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Digital Performer: EQing Tracks

Arrow/Pointer: A

Standard modifiers

All the tool shortcuts work hand-in-hand with the Mac's 'standard' modifier keys
to achieve specific actions in DP.

Perhaps most useful is the use of the Alt key for duplicating data, although how
this applies to different types of data is a touch inconsistent. If you want to
duplicate some MIDI data, such as notes, pitch -bend or controllers, you can
simply select it, then point at the data while holding down the Alt key, and drag.
For soundbites, you can additionally start dragging first and then hold down the
Alt key. It is possible to hold down Alt first, but you must then drag the 'body' of
the soundbite, not its title bar. Fortunately, automation data isn't picky at all about
the use of the Alt key!

The Control key has two uses. First, you use it for 'throwing' soundbites, either
against other soundbites or to the beginning of the sequence. Just select the
soundbite (or soundbites), hold down Control, drag in the direction you want to
throw, and let go. When a soundbite is not selected, holding down Control and
dragging on the soundbite lets you make a time-range selection, even across
multiple soundbites.

The Apple key has only one main use, but it's a very important one. During any
editing action — moving, duplicating or selecting — holding down the Apple key
toggles the current state of the edit grid on or off. So if you're doing an edit and
everything's snapping to grid and you wish it wasn't, just hold down the Apple key
and un-snappy dragging behaviour is restored. The opposite of this situation
holds just as true.

Don't overlook the humble Shift key, either. Using Shift, you can add to a current
selection, whether that's data, tracks, soundbites or any other kind of selection.
By its nature this shortcut is 'non-contiguous' — so if you Shift-select events in a
list, say, that are not next to each other, you don't select all the ones between the
selected ones too. However, you can also de-select, or remove from a selection,
using Shift. If you've ever used the scissors tool to cut up a selected soundbite
you'll know that all the resultant soundbites become selected too. When you want
to then move just one of them, you can hold down Shift, click around to de-select
all the unwanted ones, and drag the remaining one, all in one action. This makes
more sense when you try it!

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Quick Tips
Yet more keyboard shortcuts...
New MIDI tracks, Instrument Tracks and mono and stereo voice tracks can be
created by hitting Shift-Apple-M, -I, -A or -S, respectively, while aux and master
fader tracks can be summoned up with Control-Apple-A and -M.
Alt-spacebar plays whatever is selected in your sequence: superb when you're
working with audio, so that you don't have to keep messing around with the
playback wiper.
Apple-/ (forward slash) clears all clipping indicators — in the Mixing Board,
Sequence Editor, plug-in windows and Performance Monitor window, amongst
others.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Digital%20Performer%20%20EQing%20Tracks.htm (7 of 7)9/27/2005 9:25:38 PM

 Pro Tools: latest news

In this article:
New Plug-ins
Pro Tools: latest news
Getting Convolved Pro Tools Notes
More Plug-ins Published in SOS May 2005
Education, Education, Print article : Close window
Education
Technique : Pro Tools Notes
Current Versions
Mac OS X (10.3.7)
Pro Tools HD and HD Accel:
v6.7cs8. Our new-look Pro Tools Notes column brings you all
Pro Tools Mix and Mix Plus: the latest news from the Digidesign universe...
v6.4.1cs3.
Pro Tools LE systems:
v6.7cs8. Mike Thornton
Windows XP (with Service
Pack 1 or 2) You might notice that from this month, the format of Sound On Sound's Pro Tools
Pro Tools HD and HD Accel: coverage has changed. From now on, this column will be devoted to bringing you
v6.7cs8. the latest news and updates on all things Pro Tools-related, and there'll be
Pro Tools Mix and Mix Plus:
separate workshop features elsewhere in the magazine where we can talk
v6.4.1cs3. technique in even more detail.
Pro Tools LE systems:
v6.7cs8. New Plug-ins

This month's news seems to fall into three obvious groups: modelling plug-ins,
convolution reverbs, and training (see box on the next page). On the first front,
we say hello to new versions of existing plug-ins from Line 6 and IK Multimedia,
and a new modelling plug-in from Waves.

Amp Farm 3.0 from Digidesign and Line

6 is claimed to provide a new level of
realism and dramatically improved
sonic power and flexibility, thanks to all-
new cabinet models and the ability to
select one of four virtual microphone
setups to use with the cabinet of your
choice. Compatible with Pro Tools 6.7
software on HD and Accel systems,
Amp Farm 3.0 supports sample rates
up to 192kHz.

The rival Amplitube 2.0 is a major

upgrade to IK Multimedia's amp and

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Pro Tools: latest news

speaker modelling plug-in. Amplitube

boasts separate preamp, EQ, power
amp and cabinet models which can be
mixed and matched to craft your amp
sound, and the new version includes no
fewer than 30 effects, combining
accurate models of boxes from the past
with the latest high-end technology.
Compressors, delays, pitch-shifters, octave dividers, modulation, reverbs, rotary
speaker, fuzzboxes, overdrives, stereo image processors, limiters and many
more are represented, and Amplitube 2.0 has two completely customisable six-
stompbox pedalboards. This means that by simply switching a preset you will be
able to completely repatch all of the stompboxes. The same is true for the rack
effects, which are now offered in a two-chain configuration with four
programmable effects each.

Waves' Q-Clone plug-in models hardware equalisers in real time. It's a common
problem: you have a classic outboard equaliser that sounds great, so you want to
use it on several tracks — but you only have one device. What do you do? Until
now, your only choice was to process and print each track separately, which
meant you couldn't hear the entire mix until you were finished, and that any EQ
changes were tedious and extremely time-consuming. With Waves' Q-Clone plug-
in, you insert an outboard equaliser on a channel, adjust it to get the sound you
want, then click a button to capture the sound of that equaliser with your chosen
settings and replicate it in plug-in form, thus allowing you to use a single
'hardware' EQ on as many tracks you need.

In addition to capturing your own equalisers, Q-Clone comes with a library of

presets captured from hardware equalisers. Users will be able to post their own
captured processor characteristics on Waves' web site, just as you can with their
IR-1 convolution reverberation plug-in.

Getting Convolved

Talking of which, the leading players in convolution reverb plug-ins for Pro Tools
have both released new versions of their products. TL Space version 1.1 and the
new TL Space on-line IR library are now available for download from Trillium
Lane Labs' web site at www.tllabs.com to registered TL Space users. The TL
Space on-line IR library features the first instalment of regular IR updates, a set
of 24 new impulse responses in a selection of categories, including churches,
concert halls, rooms, effects and vintage analogue reverb units. TL Space 1.1 is
a free upgrade for all TL Space users, and is available for Pro Tools LE and TDM
systems on Mac OS X and Windows XP.

Meanwhile, Waves have expanded their IR convolution reverb plug-in series with
three new or improved plug-ins. IR360 is for multi-channel surround sound.
Initially available only in HTDM format, it offers surround sound capabilities by

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Pro Tools: latest news

adding extra modules to IR-1. IR-1 itself

has been upgraded to version 2 in both
HTDM and Native versions, and now
lets users capture their own samples.
The new IR-L is an entry-level light
version offering simpler controls which
uses the same convolution engine and
samples but with a more basic feature
set, and is available in two versions:
Native and HTDM.

Waves have also developed www.

acoustics.net, a dedicated web site
offering a newly expanded library of
downloadable impulse response
samples created by the company and submitted by users. IR360 and IR-1
version 2 now let users capture the sound of acoustic spaces and hardware
devices by playing an included sweep signal into an acoustic space or through
an outboard processor and recording the result. The software then lets users
import the recording to create a custom impulse response that's ready to use.

In addition to the new releases, Waves is offering a set of IR libraries on DVD as

well as surround versions for most sampled venues. Existing and old IR libraries
are supported.

More Plug-ins

If that's not enough new plug-ins for you, Trillium Lane have also announced a
new product called TL Drum Rehab, a tool for augmenting or replacing drum hits
with samples whilst retaining the original dynamics, offering up to 16 levels of
multisampling. TL Drum Rehab will ship with an extensive library on DVD,
including samples from leading drum libraries, and additional drum samples will
be available on-line from the Trillium Lane Labs web site. It's an RTAS-format
plug-in for Pro Tools LE and Pro Tools HD systems on Macintosh OS X and
Windows XP and is expected to ship in March 2005.

Further plug-in action comes from Izotope, who have released all their plug-ins
for Mac OS X in HTDM, RTAS and Audiosuite formats. The release includes the
Ozone mastering suite, the Trash distortion, amp and cabinet simulator, and the
intriguing Spectron. Spectron is a versatile spectral-based effect which separates
incoming audio into thousands of frequency bands which can be processed
individually through combinations of effects such as morphing, filtering panning,
delay and feedback. Izotope also offer the free Vinyl record simulator, which
makes input audio sound as if it was a record being played on a record player. It
provides the user with control over parameters ranging from the amount of dust
to the year the record player was created.

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Pro Tools: latest news

www.izotope.com

Finally, to Finalis from Elemental Audio. Designed as a mastering-quality limiter,

the RTAS-format Finalis can be used anywhere audio levels need to be
controlled and/or prevented from exceeding a limit, and is designed to be easy to
use while still providing flexibility. It includes three different psychoacoustic
limiting algorithms and introduces a new Crest Factor I/O Meter.

www.elementalaudio.com/

Education, Education, Education

If you want to brush up on your Pro Tools skills or get a recognised qualification to
help in your career development, London-based Alchemea College are keen to
help. Following the success of their 101, 201, 210M and 210P courses, they have
been upgraded to the South of England's only 'Pro School', permitting them to run
the 300 expert-level training courses. Alchemea will also be the only UK training
centre to offer the 210P and 310P Post-production courses. They pride
themselves in offering small class sizes with the basic courses limited to four
students, the intermediate ones to two, and the top-end training one-on-one. All
classes takes place in Alchemea's custom-built HD suite, which includes two Pro
Tools HD Accel systems, control surfaces, Avid AV systems and 5.1 monitoring.
If you'd rather learn at home, Protoolsvideos.com (www.protoolsvideos.com) have
released Pro Tools 6.7-specific tutorials, covering all of the new features in Pro
Tools 6.7. The instructors behind Protoolsvideos.com are all Pro Tools experts
and provide advice, consultation, and instruction to Pro Tools users ranging from
beginners to studio veterans. It doesn't matter whether you are running Pro Tools
on Mac or PC, because all of the Protoolsvideos.com video tutorials support both
platforms.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Pro%20Tools%20%20latest%20news.htm (4 of 4)9/27/2005 9:25:41 PM

 Reason: New Refills & Tips

In this article:
Re-release Refills
Reason: New Refills & Tips
Going Flatpacking Reason Notes
Quick Tips Published in SOS May 2005
Bands Apart Print article : Close window

Technique : Reason Notes

This month, new Refills and tweaking techniques for

Reason v3, plus the usual haul of time-saving tips.

Derek Johnson

Of course, this month's most important Reason news is that v3 is now available.
A full review can be found starting on page 32 of this issue, so have a glance
over that to see if Propellerhead have been reading your mind as to which new
features to add.

The enhancements and additions, while not perhaps filling all the gaps in our
favourite virtual studio, are pretty impressive. Certainly, existing users should
order their updates as soon as possible: the new mastering tools and the
awesome Combinator are well worth the expenditure. If you need any persuasion
to make the upgrade, or simply like the sound of a good deal, surf over to
Propellerhead's web site and visit the shop (www.propellerheads.se). Here, you'll
not only be able to simply buy the £69 v3 update (which can alternatively be
sourced from your local hi-tech music dealer), but can also be tempted by one of
five special bundles featuring T-shirts and/or special deals on Recycle and the
Drum Kits and Strings Refills.

Re-release Refills

Recently I've really enjoyed reviewing Sonik Synth 2, a co-production between

sample masters Sonic Reality and IK Multimedia (the Sampletank people). I
found its wide range of raw samples very impressive. SR are also behind the
Sonic Refill range of Reason-compatible sound libraries, currently 20 strong, with
various compilations also in the catalogue. Two volumes of synths, plus rhythm
section, retro keyboards, acoustic folk, world percussion, pianos and organs,
bass, symphonic and Mellotron are just some of the collections' themes.

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Reason: New Refills & Tips

It just so happens that SR have

contributed patches and samples to
the new Reason v3 factory sound
banks. They might also be first off the
mark with new third-party content for
v3. Three volumes of the Sonic Refill
collection (Retro Keys, Vocal Textures
and Mello-T) have been updated for
A Sonic Reality combi patch from the Retro
v3, adding new Combinator patches to Keys Refill collection. Note the customised
already comprehensive libraries. The skin applied to the Combinator's controller
rest of the range will be updated over panel.
the coming months. Interestingly,
Sonic Reality v3 Refills will be available from Propellerhead's on-line shop as
keenly-priced downloads — but be warned that they're huge!

I've had a listen to the first three updated collections, and the quality and variety
is great. I'm sure I recognise some samples from Sonik Synth 2, but this is no
great problem: I liked a lot of that material, and having it available in the form of
NNXT patches is very convenient. The new Combis are also pretty good,
stretching the raw material into more sophisticated shapes. For example, the
electric pianos and organs of Retro Keys are given a classic or modern sheen
courtesy of appropriate layering and effect processing. Check out www.
propellerheads.se or the web site of Sonic Reality (and Propellerhead) UK
distributor M-Audio (www.maudio.co.uk) for more details.

Going Flatpacking

Another developer quick off the v3 mark is Lapjockey. You may recall their first
release, the Flatpack Refill reviewed back in July 2003's Sound On Sound. This
large and varied romp through the sounds of classic synths, keyboards and drum
machines scored with slick presentation and audio quality. Again, some of this
group's work can be found in the expanded v3 factory Refill, and a demo of the
forthcoming Flatpack 2 is included on v3's installation disks.

From the brief example supplied with

Reason v3, Flatpack 2 is going to be a
good collection. Combinator is
exploited to the max as a starting point
for some serious synthesis
experimentation, and the name of the
game seems to be sound design in
ways undreamed of with earlier
Lapjockey follow up their Flatpack Refill with
versions of Reason. FP2 really goes
Flatpack 2, a patch from which is illustrated
for it in terms of analogue emulation here. You'll notice the customised look for
and the creation of involving Combinator, a practice which seems to be
soundscapes. catching on...

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Reason: New Refills & Tips

The team have developed several 'shells' that form the basis of their patches.
Kilburn powers classic synth recreations: raw synth waveforms have been
sampled, and then a given instrument's signal path is emulated using Reason
devices in the Combinator. The Scope shell is dedicated to the generation of
"soundscapes, pads, textures and just about any kind of rich evolving sound
beds you can imagine". Boxmoor is the shell for the creation of Combinator-
based drum machines, and Rex Dex is a collection of REX-based loop players.
Existing and new effects have also been 'combined' into a new collection of
patches.

Flatpack 2 is due soon, and I'm sure SOS will give it the once over when it's
available. Until then, www.lapjockey.com is worth a visit for more info.

Quick Tips
If you already have your v3, here's a little routine you might like to try. If you've
used individual songs to collect chains of effects or other interconnected devices,
for re-use in other contexts, why not spend some down time turning the elements
you want into combis? It's literally as simple as highlighting the desired devices
and choosing 'Combine' from the Edit menu (see main body). After sorting out the
input and output audio routing, and perhaps adding a mixer of some sort to merge
any parallel audio streams that might be orphaned, name and save the result as a
Combinator patch for even easier access later. Of course, devices can be cut and
pasted from the main rack into Combinator, if desired. A development of this idea
is highlighting an entire rack of devices and turning it into a combi. Having such
'template songs' as Combinator patches rather than normal songs could save you
some time. It's still possible to address individual devices in a combi from the
main sequencer, and choosing 'Uncombine' from the Edit menu removes the
Combinator and places its combined devices in the main rack.
The new Micromix stereo line mixer in v3 has just one auxiliary send. If you'd
ideally like to feed more than one effect from a single instance of Micromix (and
don't just want to add a Remix to whatever session or Combi you're using), use a
Spider Audio to split the aux send to multiple effects. You may not have individual
control over each 'send', but it can be a good compromise for parallel processing.
A new test routine not previously available within Reason comes courtesy of the
MClass Stereo Imager. Both its upper and lower bands have a range of mono
through normal stereo to very wide stereo. Set both bands to mono, place the
device at the end of the audio chain, then use the device's bypass/on switch to
create an instant test for mono compatibility. Audio is heard in mono, or effectively
in mono, in many environments, and it's useful to have a handle on phase issues
(which can be caused by excessive use of the Stereo Imager's extreme width
settings) that can cause problems for a mix when played in mono.
If you own Propellerhead's Recycle, you may already know that some annoying
clicks can be quickly removed from slices that are not quite accurate by simply
adjusting the Attack and Decay parameters in the amplitude envelope. Doing so
can save lots of time over doing it properly in Recycle (by moving slices markers
individually). If you haven't done so and find a loop to be a bit clicky when loaded
into Dr:Rex, the Attack and Decay parameters can be used to eliminate clicks
without changing the character of the loop too much. Having said that, changing

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Reason: New Refills & Tips

the character of the loop may inspire you to go further with your envelope
experiments. A slightly soft attack allied to a fast decay can make a REX loop
sound very different from the original audio file. Tempo-sync'd LFO effects, routed
to pan or filter cutoff, can also change the feel of a REX loop immensely.
A simple way to create a fat, phasey sound in a device such as Subtractor or
Malström is to enable both oscillators and detune them. Adjust the fine-tuning
control in a positive direction for one oscillator and in negative values for the
other. Anything up to +7/-7 sounds rich and moving, but further detuning will start
to sound, well... out of tune! A similar idea could be used in Combinator, where
two whole devices could be detuned against each other in the same way. Panning
detuned devices can also add to the space and width of the resulting stereo
image.

Bands Apart

Reason v3's new MClass processor collection adds the kind of mastering tools
that many have tried to emulate with what was previously available in Reason.
Without further ado, here's a technique for creating a simple multi-band
compressor (just two independent frequency bands) with MClass devices. You
need a Stereo Imager and two Compressors from the MClass family, plus a
Micromix stereo line mixer. To create this ganged processor, we'll be exploiting
the Stereo Imager's 'separate output'. This device provides stereo width
enhancement on two bands, separated by a simple crossover-frequency control.
It's equipped with a 'separate' output, so that one band can be processed
separately from the other.

Flip to the back panel, and route the

main stereo out of the Stereo Imager
to the input of one Compressor. Then
route the separate out to the other
Compressor's input. It's up to you
whether the separate out is switched
to output the high or low band, though
this choice has an impact on the next
step of the technique. (The illustration
above has the low band chosen,
though it's not visible from the front!)
While you're at the rear, route the The final split-band compressor chain. These
outputs of each Compressor to an devices could easily be 'combined' as a
input each on the Micromix. Combinator patch. I'd loaded a piece of
mixed digital audio into the NN19 shown
here to test the chain's effectiveness.
Return to the front panel and enable
the Stereo Imager's 'hi' solo switch (if you've chosen the high band as your
separate out, solo the 'lo' band on the front panel). Set like this, both bands
have their own independent outputs, ready for compression. You might like to
label your Compressors in some useful manner — 'Low Band' and 'High Band',
for example.

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Reason: New Refills & Tips

That's pretty much all there is to it: the Stereo Imager's crossover-frequency
control determines the frequency range of each band. If it's set full left,
everything below 100Hz will be treated by the 'low band' Compressor and
everything else by the 'high band'. You still have full control over the stereo
image of each band.

Moving the low band's width control towards mono can give the processed audio
a more focused bottom end, while the opposite (extreme width enhancement of
the low-frequency range) creates a 'fuzzy' mix (low frequencies don't have much
directional information and make more impact if not over-processed by stereo
effects).

It would be possible to use a Spider Audio device to merge the two MClass
Compressor outputs, using their output gain controls to balance the frequency
bands, but the Micromix facilitates a tidier setup, complete with solo'ing and
muting options that you can use while you're setting everything up.

Effect chains such as this are ideal for converting into a Combinator combi. Make
this setup in Combinator to start with, or highlight all the devices and select
'Combine' from the Edit menu. The result is an instant Combinator patch.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Reason%20%20New%20Refills%20&%20Tips.htm (5 of 5)9/27/2005 9:25:46 PM

 Sonar: Studio or Producer?

In this article:
Red Power
Sonar: Studio or Producer?
Sonar Studio Or Producer? Sonar Notes
Published in SOS May 2005
Print article : Close window

Technique : Sonar Notes

It's time to address the Studio versus Producer

question, as well as looking into a new Sonar remote-
control option.

Craig Anderton

I'm a fan of hands-on control, particularly when mixing, but remote control is also
useful when tracking. With so many musicians now recording in the same room
as their gear, hard drives and other noise sources are problematic. And, for
guitarists, standard magnetic pickups can pick up hum and grunge, which often
necessitates standing in a 'sweet spot' (seldom next to your computer keyboard)
to minimise noise.

Although there are many comprehensive remote controllers, sometimes all you
want is something simple, inexpensive and small — like the ADS Tech Red
Rover. I originally bought one of these to control Cool Edit Pro (now Adobe
Audition), but the good news for Sonar users is that they can now download a
Red Rover control-surface plug-in from www.cakewalk.com.

Red Rover lists for around £150, but is also bundled with Audition for around
£270. Although you might think of choosing the less expensive option because
you already have Sonar, Audition's noise-reduction tools are excellent for
cleaning up tracks — and if you're into creating loops, Audition is one of the few
editors that leaves Sonar's groove clip markers intact even if you change bit
depth or sample rate. Sonar is a full-featured program but it's not a digital audio
editor, and Audition can fill that gap.

Red Power

Back to Red Rover, which connects to the computer via a 10-foot USB cable
(which you can extend with an active USB driver) and is buss-powered. The front

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Sonar: Studio or Producer?

panel has a backlit, two-line by 20-

character LCD, three knobs (master
volume, track volume and track select),
an eight-segment level meter for
record and playback, buttons for mute/
solo/record, and transport controls.
Another button adds cue markers and
brings up the marker dialogue, while
the remaining button turns the
metronome on and off during
recording. A footswitch jack on the Red Rover, although originally designed for
back allows hands-free recording. Cool Edit Pro, is now supported as a control
surface within Sonar.

Red Rover seems pretty undemanding.

It doesn't hog a lot of USB bandwidth, nor draw much power (about 350mA). If
your USB port can't provide enough juice, either use a powered USB hub or turn
off the LCD backlight (the main power drain). Incidentally, a driver CD is not
needed: I plugged Red Rover into Windows XP and recognition was automatic.
The icing on the cake is that Red Rover's footprint is small enough that you can
fit the remote just about anywhere. About the only weirdness for me was that the
control surface plug-in refused to let me specify USB as the MIDI port — it
insisted on listing my Creamware card's MIDI interface. But I could select either
that or 'none' with no problems. Apparently Red Rover simply communicates via
USB, leaving any other MIDI ports out of the picture. For more info on Red
Rover, visit www.adstech.com and go to the Professional Video/Audio section.

Sonar Studio Or Producer?

A few Sonar 3 users have asked whether to pay the premium for Sonar's
Producer Edition, or get the Studio Edition now, then upgrade to Producer when
Sonar 5 comes out. They've done the maths, and based on past US dollar
pricing it would cost about $70 less to buy Sonar 4 Studio Edition now and step
up to Sonar 5 Producer Edition later ($99 upgrade to Studio, then $229 to
Producer) compared to going for Sonar 4 Producer now ($199), then upgrading
to Sonar 5 Producer when it appears (presumably another $199).

Under some circumstances, the choice is obvious: if you plan on doing surround
or video, you'll need Producer. Beyond that, the decision hinges more on
subtleties. The next 'big-ticket' item in Producer is the Sonitus suite of plug-ins,
which lists for $299 if purchased separately. Although the plug-ins that come with
Studio Edition are decent, I prefer Producer's. There are some good free plug-ins
on the net, so perhaps a set of plug-ins isn't essential. But if you don't have a
good suite, the Sonitus set is a pretty good deal when bought as part of the
Producer Edition, especially given its quality.

The Prosoniq MPEX time-stretching algorithms are also a Producer advantage. If

you need to stretch audio with minimal artifacts, quality is superior to Studio's

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Sonar: Studio or Producer?

timestretching. Another 'pro' option is

POW-r dithering, which many people
consider to be about as good as it gets.

One more factor is that the Producer

Edition's mixer includes per-channel
EQ based on the Sonitus EQ plug-ins.
Of course, you can insert your EQ of
choice in Studio, so this is probably not
a deal-breaker. Producer also provides Installing Red Rover is simple: Just select it
from the list of supported control surfaces in
four assignable effects controls on the Sonar.
mixer panel, which is a convenience —
but if that's the only reason you have
for upgrading to Producer instead of Studio, it hardly seems crucial.

For most people, an upgrade to Producer would be fairly easy to justify. For first-
time buyers, the cost savings made by going with Studio are more substantial.
But as long as Cakewalk continue their relatively benign upgrade policies, you
can always step up when the next major revision hits.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Sonar%20%20Studio%20or%20Producer.htm (3 of 3)9/27/2005 9:25:50 PM

 Apple GarageBand 2, PowerBooks & iPods

In this article:
Knowing The Score
Apple GarageBand 2, PowerBooks & iPods
Garage Band In SOS Apple Notes
New Tools For The Garage Published in SOS May 2005
Powerbooks & iPods Print article : Close window
What A Performance
Technique : Apple Notes
Power Of Two

Although intended as an entry-level application to

introduce new people to computer-based music
making, Apple's Garage Band has received acclaim
from beginners and professionals alike. In this special
extended Apple Notes we look at version 2, which
adds score editing, multitrack audio recording, and
more...

Mark Wherry

Launched as part of the iLife '04

bundle, which was introduced at the
San Francisco Macworld show in
January 2004, Garage Band was the
first new music-making application
from Apple to take advantage of the
technology and people the company
had acquired with the purchase of
Emagic back in 2002. While Emagic's
Logic has now been assimilated into
Apple's line-up of professional Garage Band 2 can import MIDI files and
applications as Logic Pro, alongside tries to assign suitable instruments to each
Final Cut Pro and DVD Studio Pro, track automatically. Here you can see the
first movement of Bach's second
Garage Band 2 turned out to be a huge
Brandenberg concerto arranged for Live Pop
popular success for Apple, introducing Horns, Pop Flute, Alto Sax, Orchestral
more people to making music on their Strings and Smokey Clav. Time to install
computer — or, specifically, their Mac. Apple's Orchestral Jam Pack expansion for
Garage Band 2, I think!

While the functionality of Garage Band

2 is deliberately limited, to make the application appeal to beginners, the charm
of the program, for want of a better word, has certainly worked on professionals
as well. Although Garage Band 2 will hardly (and nor does it intend to) replace
Logic, I've heard a surprising number of established musicians and composers

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Apple GarageBand 2, PowerBooks & iPods

talk fondly of Apple's entry-level music application, including those singer-

songwriters who use it as a successor to the four-track tape recorder for trying
out ideas. For me, Garage Band 2 has come in surprisingly useful when I need a
quick sound source, or to check note pitches and so on for sampling sessions,
especially since you can rely on the fact that Garage Band 2 comes pre-loaded
on every new Mac.

One year on, Apple released Garage Band 2, featuring a basic score editor,
multitrack audio recording, new audio tuning tools, and more, so this month we'll
take a look at the new features in a little more depth.

Knowing The Score

One of the big new features in Garage Band 2 is a notation view for editing your
MIDI regions in traditional musical notation. Once the Track editor is visible, you
can toggle between the new notation view and the original graphic view editing
mode, which allows you to edit MIDI notes in a piano-roll-style editor or work with
a limited selection of MIDI controllers (modulation, sustain, expression, foot, pitch
bend). One change in the graphic view for is that the transpose and velocity
parameters now appear in the main Region section of the Track editor, rather
than in the Advanced section, and the Transpose parameter has been renamed
Region Pitch. This is for consistency with the new audio features, as we'll see
later.

The notation view displays MIDI notes

and any sustain pedalling of the
currently selected Track on a linked,
piano-style treble and bass stave pair,
where size automatically scales to fit
the height of the Track editor. While
you have almost no control over how Here you can see Garage Band 2's notation
view. Notice how the program illustrates the
the notes are translated into notation, MIDI lengths of notes when they're selected,
as you would with Logic's score editor, and the automatic cleaning up of overlapping
Apple have opted for the safest notes.
possible interpretation, to help the
notation display to look as clean and readable (and presentable) as possible. For
example, two overlapping notes (where one note starts and another begins
before the first ends) are displayed as two separate notes. The only influence
you can have over the display of the notation is to set a resolution for how the
display is quantised, from the pop-up menu available via the small ruler icon at
the top-right of the Track editor.

While it might seem odd, the feature I like the most in the Garage Band 2
notation view is the way in which the MIDI lengths of notes are expressed. While
the shape of a note head on the stave is what tells you how long that note should
be played for, this isn't particularly precise when you compare how a piano-roll-
style editor (like the graphical view) displays length. So when you select a note in

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Apple GarageBand 2, PowerBooks & iPods

Garage Band 2's notation view, a horizontal bar is displayed by that note to
illustrate the note's precise length. You can hover and drag the mouse over the
bar to adjust the length of the note, just as you would in the graphical view. As
you change a note's length, the actual note head (and any rests around that
note) are redrawn to keep the notation correct. This is particularly neat,
especially because of the built-in 'clean lengths' feature. It's a really nice addition
that overcomes a limitation found in the majority of notation editors — even in
professional applications.

As with the graphical view (and Regions in the Region editor), you can create
new notes in the notation view by Apple-clicking on the stave. The length of the
note you're adding is set in a pop-up menu in the Advanced section of the Track
editor. You can also select sustain-pedalling symbols from this pop-up menu, and
the menu itself can actually be opened from anywhere in the Track editor area by
Control-clicking in an empty space.

Garage Band In SOS

You can read our original Garage Band discussion at www.soundonsound.com/
sos/apr04/articles/applenotes.htm, and read about version 1.1 at www.
soundonsound.com/sos/jul04/articles/applenotes.htm. This revision added official
Rewire support (originally introduced in 1.01), better track management, and the
ability to use audio previews of MIDI loops as audio loops to the original Garage
Band release.

New Tools For The Garage

Following on from the notation view, Garage Band 2 adds the ability to import
MIDI files. To do this, you must first make sure you have a Song already open in
the application — you can't load a MIDI file from the 'Open Song' window. To
import a MIDI file, simply drag the required file from the Finder into the Garage
Band 2 window. As you drag the file, you'll notice a vertical line appearing in the
main area, to show you at what point in time the MIDI file will be imported into the
song. When you release the mouse button, the file is imported into newly created
Tracks and, rather neatly, Garage Band 2 will make an effort to assign the tracks
in the MIDI file to suitable instruments in the Tracks in your Song, by reading any
program-change numbers in the file, based on the General MIDI standard.

For those in education, the wealth of MIDI files of art music available online for
study means that the MIDI file import is especially valuable; but it's also useful for
those times when you might have forgotten your sequencer's copy-protection
dongle and you need to quickly check a couple of notes in a MIDI file. I'm sure
I'm not the only one who's ever used the Quicktime Player application to get a
basic idea of what was in a MIDI file...

For entering MIDI data, Garage Band 2 also features a mode called Musical

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Apple GarageBand 2, PowerBooks & iPods

Typing, which is basically a different

implementation of the Caps Lock
Keyboard in Logic, turning certain keys
from your QWERTY keyboard into a
musical keyboard instead. In Logic, the
Caps Lock Keyboard is so called
because it is enabled when the Caps
Lock key itself is enabled, while the Garage Band 2 features a slightly more
colourful version of a feature similar to
Music Typing function in Garage Band
Logic's Caps Lock Keyboard, enabling you to
2 is enabled by choosing Windows / use your QWERTY keyboard as a musical
Musical Typing, or pressing Apple-Shift- input device.
K. You're never going to get a great
musical performance using your
computer's keyboard, but Musical Typing, and Logic's Caps Lock keyboard, are
still surprisingly useful, especially when you're working on a laptop and don't
have a portable USB keyboard handy. In addition to basic note pitches, you can
also adjust the velocity used for note commands, and there's even pitch-bend
and modulation keys (that really do work!), a sustain pedal, and keys to adjust
the transposition of the keyboard's range. Speaking of the keyboard's range, both
the Musical Typing window and the older mouse-driven musical Keyboard now
feature a small illustration of the full range of a keyboard, with an area highlighted
in blue to show you the focus of the onscreen keyboard.

On the audio side, Garage Band 2 will now import ACID files, in exactly the same
way as importing MIDI files. Simply drag an ACID-format WAV from the Finder to
the place where you want it in your Song and Garage Band 2 will automatically
add a new audio track containing the converted audio file. Garage Band 2 won't
automatically add an imported ACID file to your Apple Loop library, but thanks to
the new 'Add To Loop Library' command, you can now make your own audio
recordings, along with imported ACID files, into Apple Loops. Simply select the
Region you want to add to the library, select Edit / Add To Loop Library, and a
sheet will appear where you can give the loop a name, set whether it's a loop or
a one-shot sample, and specify musical scale, genre, instrument and mood
descriptors for searching.

Two new audio-processing tools have been added to the audio view of the Track
editor to enhance the tuning and timing of audio regions. The Enhance Tuning
function works in a similar way to processes such as that offered by Auto Tune,
but has just one slider to specify how much the tuning should be 'enhanced' or
effectively quantised to the nearest chromatic note. There's also a 'Limit to Key'
toggle, which, if selected, quantises the pitch using only the notes in the key of
the current Garage Band 2 Song. And to help you stay in tune in the first place,
Garage Band 2 also features a tuner mode, which can be enabled by selecting
an audio track and clicking the tuning-fork icon on the transport area, pressing
Apple-F, or choosing Control / Show Instrument Tuner. Finally, the Enhance
Timing process attempts to quantise the timing of your recorded audio, based on
the tempo and time signature of the current Song. Like Enhance Tuning, it has
one slider for you to specify how strongly the process should be applied.

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Apple GarageBand 2, PowerBooks & iPods

Powerbooks & iPods

The demand for a Powerbook with a G5 processor is growing all the time, and you
didn't have to look for long on the web during February to read plenty of opinions
and news stories on this subject. One site claimed that Apple would have portable
G5s by the second quarter of 2005, while others restated the problems of cooling
and power consumption, something that even Apple briefly commented on at the
release of the iMac G5 last year, cautioning customers that the iMac G5 didn't
mean that all of the issues involved in producing a Powerbook G5 had been
resolved. It's also worth remembering that most people look to a G5-based
portable for a substantial performance increase over current G4-based portables,
and that a 1.8GHz iMac G5 offered over double the performance of a 1.5GHz
Powerbook G4, based on previous tests in SOS. However, given that it may be
necessary to scale down clock speeds and make other compromises to put a G5
in a Powerbook, I'll be curious to see how a Powerbook G5 ultimately performs.
But while the world waits for a Powerbook G5, Apple recently refreshed the
Powerbook G4 line-up with the usual expected increase in numbers, but also
some new features. The trackpad now features a scrolling option, where you can
drag two fingers together to scroll the window currently in focus, either
horizontally, vertically, or panning in both directions. Although I haven't tried this
out for myself yet, it sounds like a nice alternative to other ideas used on similar
devices, where the edges of the pad take on this function and it's consequently
easy to scroll by accident.
The second new feature is what Apple call a Sudden Motion Sensor, which
causes the heads of the hard disk to automatically park in response to any
sudden movement or change of axis, thus suspending read and write operations,
until the sensor detects that the laptop is level again and re-enables hard disk
activity. This type of technology isn't usually ideal for applications where you're
going to have a constant stream of data going to and from the hard disk, such as
most music and audio applications, so might be better left disabled. However,
again, I look forward to checking out the feature in more detail.
In terms of specifications, all
Powerbooks now feature 512MB of
DDR333 memory as standard (a
very welcome change), Airport
Extreme, and Bluetooth 2.0 support.
Apple claims to be the first
manufacturer to include Bluetooth
2.0 support in a portable range. This
revision to Bluetooth is backwardly compatible with Bluetooth 1.2 and offers three
times the bandwidth: up to 3MB/s instead of up to1MB/s. The 12-inch Powerbook
features a 1.5GHz G4 processor and NVIDIA GeForce FX Go5200 graphics with
64MB of DDR SDRAM, and is available with a Combo drive and a 60GB Ultra
ATA/100 5200RPM drive for £1049, or an 8x Superdrive (faster than ever before
in a portable) and an 80GB Ultra ATA/100 5200RPM drive for £1199.
All 15-inch Powerbook models now come with the illuminated keyboard, and
feature an 80GB Ultra ATA/100 5400RPM drive and ATI Mobility Radeon 9700
graphics with 64MB of DDR SDRAM as standard. These models also offer a build-
to-order option on their graphics hardware, to include a 128MB 9700 video system
instead, with dual-link DVI support — which means that, yes, you could now drive
a 30-inch Cinema Display from a Powerbook. There are still two 15-inch models:

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the first offers a 1.5GHz G4 processor and Combo drive for £1379 and the second
has a 1.67GHz G4 processor (the fastest yet seen in a Powerbook) and 8x
Superdrive for £1579.
At the top of the Powerbook family tree is, of course, the 17-inch Powerbook,
which retails for £1849 and features a 1.67GHz G4 processor, a 100GB Ultra
ATA/100 5400RPM drive, an 8x Superdrive, and ATI Mobility Radeon 9700
graphics with 64MB of DDR SDRAM and dual-link support. The new Powerbook
G4 line should tide Mac musicians over until a G5 model becomes available,
although it's perhaps disappointing that no Powerbooks feature 7200RPM drives,
especially the 17-inch, even in build-to-order configurations.
Apple also updated the company's popular iPod range this month, with changes
to both the Mini and Photo models. The iPod Mini is now available in a 6GB model
for £169, while a 4GB model (the previous limit for the iPod Mini) costs just £139.
Both iPod Mini models feature an improved specification, offering up to 18 hours
of playback on a single battery charge, meaning that the iPod Mini now offers the
longest battery life of any iPod (compared to 15 hours for the iPod Photo and 12
hours for the standard iPod and Shuffle models). Another interesting change is
that the iPod Mini is now supplied only with a USB 2.0 cable and is capable of
charging via this cable, not just via the now-optional Firewire 400 cable.
Originally introduced last October in both 40GB and 60GB configurations, costing
$499 and $599 in the US respectively, the iPod Photo is now available in either
30GB or 60GB configurations for £248.99 and £309, which makes the iPod Photo
far more affordable than previously. Like the new iPod Mini, the new iPod Photo
models also support charging via USB 2.0 and, again, only this type of cable is
now supplied with the iPod Photo — a Firewire cable is an optional purchase, just
like the dock that's also no longer supplied as standard.

What A Performance

One aspect that people noticed about the first release of Garage Band 2 was that
performance wasn't exactly great on slower computers, especially those with a
G4 processor. Part of the reason for this was perhaps the number of effects each
track uses by default, which is why the ability to use global reverbs and delays
was added to an interim release. On the subject of performance, Apple's Director
and Lead Architect of Audio and Music Applications, Emagic co-founder Dr
Gerhard Lengeling, has stated many times that Garage Band 2 uses no more or
fewer resources for audio processing than Logic.

At O'Reilly's Mac OS X Conference last year, Lengeling commented: "Processor

footprint between Logic Pro and Garage Band 2 are identical; they are actually
the same product at the core. So Garage Band 2 can't take more processor
power than Logic Pro. Garage Band 2's visual interface takes a bit more
processing, but it's at a lower priority than the audio, so the audio footprints are
the same. As far as performance efficiency goes, they are exactly the
same." (Source: www.auscillate.com/ post/29.)

To help improve performance even further, Garage Band 2 implements a Track

Lock facility that's basically the same as the Freeze facility available in Logic.

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Apple GarageBand 2, PowerBooks & iPods

Once you have a track that's pretty

much finished, you can enable the
Track Lock function on it and the next
time you press 'Play' the locking
process will be invoked, bouncing all
the real-time processing for that track
into a single audio file. After this, you
won't be able to make changes to the
track, although Garage Band 2 will
warn you of this and ask if you want to
unlock the track to make changes.

It's now possible to convert your own

recordings, or imported ACID WAV files, into
Power Of Two Apple Loops and add them to your library
directly within Garage Band 2.

Garage Band 2 is certainly a useful

tool for beginners, and is surprisingly handy for more seasoned sequencer users
too. The new features continue the same design philosophy established in
Garage Band 1, making music technology accessible for those with no previous
experience. The only disappointment for me in Garage Band 2, given what the
application is supposed to achieve for its audience, is the fact that there's still no
way to create key, tempo or time signature changes. This would be tremendously
useful, especially when importing MIDI files that might contain such information.

Some music industry developers seem

to resent how Garage Band 2 has
been accepted by the general public,
pointing that music software has been
around (or, at least, readily available) When you have an audio track selected, you
for the past 20 years. But this is can enable Garage Band 2's built-in tuner.
missing the point. While Apple don't Useful for guitarists — and singers...
get everything right, the company have
a certain skill in taking high-end technology and making it approachable for
beginners: in Garage Band 2 Apple have taken Logic and made it into a
sequencer my mum would want to use. And if they have indeed succeeded in
bringing computer-based music making to a larger audience with the perception
of Garage Band 2, maybe other companies need to rethink their approach.

Published in SOS May 2005

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Apple GarageBand 2, PowerBooks & iPods

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Catch and Link modes in Logic

In this article:
The Catch Button
Catch and Link modes in Logic
Link & Contents Link Logic Notes
Have Your Say! Published in SOS May 2005
Powerful Linked Screensets Print article : Close window
Contents Catch
Technique : Logic Notes
Current Versions
Mac OS X: Apple Logic Pro
v7.0.1
Mac OS 9: Emagic Logic Pro Combining Logic's Catch and Link modes can greatly
v6.4.2
increase the usefulness of multi-window Screensets,
PC: Emagic Logic Audio
Platinum v5.5.1
but it is not always clear, especially to new users,
how the various options work.

Len Sasso

The options during playback for each open

window are controlled by two buttons at the
top left-hand corner of the window: the
Catch button ('running man' icon) and the
Link button ('linked chain' icon). I'll start
with a brief description of those buttons'
various states, then move on to
demonstrate some multi-window examples
that make use of different modes.

The Catch Button

The Catch button has two modes: it is on The Catch mode is enabled when the
when the button is blue and off when it is 'running man' button at the top left-hand
gray. When Catch is on, the window corner of a window is illuminated blue.
automatically scrolls to keep the Song With Catch enabled, you can choose
Position Line (SPL) in view. Two methods between two different scrolling methods
by ticking or unticking the Scroll In Play
of scrolling are supported; one keeps the option in the window's View menu.
SPL centred while the data scrolls behind
it, and the other jumps to reposition the
SPL at the left-hand edge of the window each time it reaches the right-hand edge.

The first of these methods is called Scroll In Play, and it can be switched on via
each window's View menu. Although it can seem more intuitive to have Scroll In

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Catch and Link modes in Logic

Play switched on all the time, it does require more graphic processing, so may
cause problems in complex Songs where your computer's processor is already
under pressure. Note that scrolling in the Event List is vertical, and scrolling in
the Score window is page-wise when in Page Edit view.

The Catch mode is available in all windows that have a time dimension; that
includes the Arrange window and each of the editors. The Audio, Project
Manager, Track Mixer, Transform, Transport, and Environment windows have no
time dimension, and therefore no Catch button.

Link & Contents Link

The Link button has three modes: Link when the button is red, Contents Link
when the button is orange, and neither when the button is gray. Contents Link
mode is activated by double-clicking the button. The key to understanding the
difference between the Link and Contents Link modes is knowing which windows
display objects, which display the contents of those objects, and which display
both. Three kinds of objects are relevant for our purposes: MIDI sequences,
which hold MIDI data; audio regions, which hold audio data; and folders, which
can hold MIDI and audio regions as well as other folders.

The Arrange window, Event List,

and Hyper Edit window can display
audio regions and MIDI sequences
as well as Folders. The Event,
Hyper Edit, Matrix Edit, and Score
windows can display the contents
of MIDI sequences. (The Event list
and Hyper Edit window are the only
windows capable of displaying both
MIDI sequences and their Clicking on the 'linked chain' button in the left-
contents.) The Arrange and Audio hand corner of any window activates Link mode
windows and the Event List can (left), whereas double-clicking switches to
show audio regions. The Sample Contents Link mode (right).
Editor window displays the contents
of audio regions.

A window in Link mode will always display whatever data is selected in another
window, if it is capable of displaying that kind of data. For example, a Linked
Event List window will display any sequence, region, or folder selected in an
Arrange window or another Event List. Alternatively, it will display any MIDI data
selected in any MIDI editor, such as a Matrix Edit or Score window.

A window in Contents Link mode will always display the contents of an object
selected in another window, if it is capable of displaying that kind of data. For
example, a Contents Linked Arrange window will display the contents of any
Folder selected in another Arrange window or Event list. A Contents Linked

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Catch and Link modes in Logic

Track Mixer window will display the channel strips of tracks inside any Folder
selected in an Arrange window — a great way to set up submixes. A Contents
Linked Matrix Edit window will display the MIDI data in a MIDI sequence selected
in an Arrange window or Event List.

It is a common misconception that the window in which a selection is made must

be active (topped) for the Link modes to apply. Actually, the selection can be
made in active, background, or floating windows — it doesn't make any
difference. Also keep in mind that if you click and hold for a moment (called 'long-
clicking') in a window that is not topped, it does not become topped. You can,
therefore, make use of Linking and Contents Linking without ever changing the
topped window.

Have Your Say!

If you want to suggest changes or improvements to Logic, then here's your
chance! The Apple development team are inviting SOS readers to send in their
suggestions of what they'd most like added or changed in Logic. Email your top
five suggestions (in order of preference) to [email protected], and
we'll forward your lists on to the Logic team. We'll be asking them for feedback on
which changes users deem most important and how these might be addressed.

Powerful Linked Screensets

A common and very useful three-window Screenset consists of an Arrange

window and an Event list in Link mode, with another MIDI editor (Matrix, Score,
or Hyper Edit) in Contents Link mode. Selecting objects in the Arrange window
reveals them in the Event list, where their position and length can be numerically
edited. Selecting MIDI regions in the Arrange window or Event list makes their
data available for editing in the chosen MIDI editor. Finally, selecting MIDI events
in the chosen MIDI editor reveals them for numerical editing in the Event List.

For audio editing, replace the second

MIDI editor in the above Screenset
with a Sample Edit in Link mode.
Selecting audio regions in either the
Arrange window or Event list will reveal
their contents in the Sample editor.
You might also replace the Event list
with an Audio window in Link mode.
You lose numerical editing of the audio
region positions, but you gain access
to all audio files and their regions, not Here is a powerful setup for MIDI
just the ones being used. arrangement and editing. The Arrange
window at the top is set to Catch, so that the
window scrolls to follow the Song Position
Notice that the Audio window and Line (SPL). The Event List window on the left

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Catch and Link modes in Logic

Sample Editor don't have a Contents is set to Link, so that it shows the positions
Link mode. The Audio window only and lengths of the MIDI sequence objects in
the Arrange window, and also shows which
displays audio objects and the Sample are selected. The Matrix Edit window is set
editor only displays the contents of to Contents Catch mode so that it always
audio regions, so the Link button Links shows the contents of the current MIDI track.
the Audio window with other object-
displaying windows (such as the Arrange window and Event list) and it Contents
Links the Sample Edit with object-displaying windows (such as the Audio
window).

Contents Catch

Catch and Contents Link modes used together have a special function for the
MIDI editors (though not for the Sample Edit). Once a MIDI region is selected on
an Arrange track, the contents of other MIDI regions on that track will
automatically be displayed as the SPL passes over them — in other words, as
they play.

The basic rules of thumb when opening multiple windows in a Screenset are to
use Link mode to synchronise MIDI editors to each other so that each displays
the same data for editing. Use Contents Link mode to synchronise a window that
displays data (such as a MIDI editor or the Sample Editor) with a window that
displays objects (such as the Arrange window). Also, use Contents Link mode to
display the contents of Folders s

elected in an Arrange window or Event list in another Arrange window or Event

list. Finally, turn Catch on to keep the data under the SPL always in view.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 CLASSIC TRACKS: The Who 'Who Are You?'

In this article:
It's Who You Know...
CLASSIC TRACKS: The Who 'Who Are You?'
In The Kitchen Producers: Jon Astley, Glyn Johns
Drinking & Singing Published in SOS May 2005
On The RAK Print article : Close window
Taking Over The Reins
Technique : Recording/Mixing
Separate Lives
The Producer's Intuition
Flying Faders
Write, Demo, Edit
Taken Away The Who's final album with Keith Moon took almost a
The Missing Minutes year to record and pushed the band to the limit.
Engineer and producer Jon Astley tells the remarkable
story behind Who Are You's title track.

Richard Buskin

When the Who re-entered the recording

studio in September 1977 following a
couple of years on the road, it wasn't in
the best of circumstances. For one
thing, constant touring had been taking
a toll on Pete Townshend's marriage,
so the composer/guitarist determined to
turn things around by adhering to office
hours — hardly a recipe for no-holds-
barred creativity. And after more than a
decade of hellraising, Keith Moon turned up in no shape to drum at an
acceptable level.

Moon relocated from Los Angeles to London for rehearsals, which took place at
the band's Ramport Studio in Battersea, and right from the start it was clear that
he might just have an attitude problem. On one of the first days, bored with all the
hanging around, he used his gold-plated lighter to set a noticeboard on fire; and
when producer Glyn Johns' assistant, Jon Astley, assiduously set about miking
the drum kit with a setup far more intricate than the simple technique which
Johns had traditionally employed, his work was quickly undone in classic Moon
fashion.

"Glyn trusted my engineering, and for the first time he was interested in getting
away from big, open miking and actually trying something different with close
miking," Astley explains. "He was open to some of the things that I wanted to try
and do, and it was quite interesting that he would let me do that — I used a

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CLASSIC TRACKS: The Who 'Who Are You?'

bunch of different mics on Keith's kit, and Glyn

came in the next day and said 'OK, we'll try it.' Up
until then, Glyn had only ever used three mics on a
drum kit, so it was a big step for him to take. You
see, the sort of ambient recording thing had been
and gone, and this was the disco era when
everything was in your face and very dry and very,
very close-miked.

"On the bass drum I used an AKG D30; I had a

Neumann KM84 on top and underneath the snare;
I used a Shure SM58 with a pad in it for the hi-hat;
overheads were Neumann U87s; and for toms I
used anything I could find that matched, such as
Sennheiser 421s. Then, the day after I'd set all this Jon Astley today.
up, Keith came in and went around the kit for me. I
just wanted to make sure the stereo imaging was
OK, so he played and then asked 'Is that all right?' I said 'Yeah, that's fantastic,'
at which point he stood up and walked straight through the bloody kit. He
obviously was aware that I'd put a lot of work into it, balancing the whole thing,
but I just thought 'Oh, well, welcome to the Who. Here we go..."

It's Who You Know...

Not that Jon Astley was all that unfamiliar with the band members and their
idiosyncracies. His sister Karen was then married to Pete Townshend, who had
initially bonded with the teenage Jon by taking him to some of the group's gigs
back in the mid-'60s. "Pete was courting Karen at the time, so it was probably to
keep my parents happy that he'd take me off their hands," Astley surmises with a
smile. "I was a Who fan anyway, and although I was a bit young to be a Mod
[Astley was born in 1951], I embraced the whole Mod thing. Then, after my sister
married Pete and I finished college in 1971, I bought a house in Twickenham, not
far from where they lived. After working at the Radio Luxembourg studios for a
couple of weeks, I then got my big break, working as a tape operator at Olympic
in Barnes. For me, this was the home of rock & roll — the Stones had recorded
there and, very soon after I joined, the Eagles were there, too, and that's when I
met Glyn Johns."

During his time at Olympic Studios, Astley worked on David Bowie's Diamond
Dogs as well as Tim Rice and Andrew Lloyd-Webber's Jesus Christ Superstar. In
the mid-'70s, he became full-time assistant engineer to Johns, producer and/or
engineer of choice for everyone from Led Zeppelin, the Eagles, Eric Clapton,
Joan Armatrading and the Steve Miller Band to the Beatles, the Rolling Stones
and the Who. "Glyn and I did a couple of Eric Clapton albums together,
Slowhand and Backless, on which I was officially assisting, but some days Glyn
didn't show up and Eric would say 'Oh, let's do a whole track. We'll show him!' In
fact 'Tulsa Time' was all me — producing, engineering, the whole lot — and Glyn

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CLASSIC TRACKS: The Who 'Who Are You?'

was fine with that. He didn't mind at all.

He thought it was part and parcel of
the job, and we got on very well.

"In '77, Glyn came up to me and said 'I

don't think there will be a conflict of
interest, but I have been asked to do
the next Who record. Because you're
working for me, I can't see that it'll be a
problem. But if you do, then say so.' I
said 'No, that's all right. I'd love to do
that with you.' So we went to Photo: Harry Goodwin / Redferns
Battersea, to what we called 'The The Who, backstage at a BBC studio in
Kitchen', which was Ramport Studio, London. From left: Roger Daltrey, Pete
and started on Who Are You." Townshend, Keith Moon and John Entwistle.
This photo is believed to have been taken
during an appearance on Top Of The Pops
to promote 'Who Are You'.

In The Kitchen

Housing a newly built custom Neve 8088 black 40-input console, a 24-track 3M
tape machine and 16-track Studer, Ramport had what was purported to be one of
the first quadrophonic control rooms, with four huge JBL speakers at the front
and two at the back.

"Pete would come in with a new song, which would serve as the backing track for
the Who to perform on, and John [Entwistle] would do the same," says Astley.
"Pete had a Polymoog that was programmed to play his backing tracks, and then
the other guys would overdub their parts. The problem was, Pete was bringing in
24-track demos and Glyn wanted to work 16-track, because the sound coming off
a two-inch head block with 16 tracks was so much better than a 24-track with
Dolbys and everything else. In fact, the Dolbys at Ramport never seemed to be
lined up properly — one day it used to sound bright, the next day it would sound
dull, and I could never tell what was going on. Every night there was a different
line-up of maintenance men, and the result was that things never sounded the
same from one day to the next. It was very, very odd."

Not that this was the biggest problem during the recording and overdubbing
sessions that took place during the last third of 1977. "Every time we came to
overdub Keith, it wasn't great," remarks Astley with considerable understatement.
"His timing was out, which was unusual for him, and this became frustrating for
everybody. He was drinking a lot and taking drugs to stop himself putting on
weight — which wasn't making that much difference — and while he was still the
jovial Keith character, it sometimes wore a bit thin with everyone else."

With tensions mounting, something had to give, and this is precisely what
happened during a playback of 'Sister Disco' on Thursday, October 27, 1977.
"Roger leaned over the desk while Glyn was sitting there and he said 'Can I hear
a bit more bass?'" Astley remembers. "Glyn stopped the machine and said

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'What?' and Roger said 'I just want to

hear a bit more bass in the mix.' Glyn
said 'We're listening to all this fucking
work that they've done, and you want to
hear a bit more bass?' At that point,
things exploded. It was unbelievable.
They both stormed out, and then I heard
this kerfuffle in the corridor and Glyn
came back in the control room with
tears in his eyes, holding his nose and
saying 'That's it. I'm going home.' Roger
had nutted him and driven off in his
Ferrari."

According to Daltrey's own recollection

several years later, he'd told Johns that he thought the strings made the track
sound over-produced: "He called me a little c**t, so I thumped him."

Thereafter, by not turning up at Daltrey's night-time vocal sessions, Johns

basically delegated responsibility for producing these to Jon Astley. Not that
reborn family man Townshend was there, either. "Pete would often leave the
studio at four in the afternoon to pick up his kids from school, and we wouldn't
see him again," Astley states. "Roger, on the other hand, only wanted to do his
vocals in the evenings, so my days were very, very long. In fact, I remember Pete
leaving when I was preparing to do a vocal session with Roger, and he said
'Make sure he sings the right notes.' I thought 'Oh, is that my job? I suppose it is
now.'"

Drinking & Singing

The bar at Ramport Studios amounted to a fridge in the corner of the studio, out of
which most present drank port — rock & roll! Entwistle preferred wine, and Moon
indulged his taste for Coca-Cola mixed with whatever spirits were available, but
Daltrey asserted that the port was good for his vocal cords.
"His voice was very good," Jon Astley confirms. "It held up when we put him under
great strain, although he only did a lead vocal about once a week — it wasn't like
he was doing two or three a night — and the two of us had great fun
experimenting with different mics. I had cardioids and figure-of-eights, one above
the other, to put his vocal into stereo by bringing it up on four channels... of
course, you can buy a microphone now that does all that, but back then we had
quite a hoot trying different things. The only thing was, it kind of made you sick
when you listened to it, because if Roger moved slightly left or right the stereo
image would move around in the speakers. I remember thinking 'I'm not sure
whether or not this is a good thing,' so I ended up mostly using one Neumann U87
with a Urei 1176 on it.
"Still, something I didn't notice during the Who Are You sessions, but which
became evident when I worked with Roger during the '80s, was that if you go past
four or five takes with him, even though he's still singing perfectly, he does lose an

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edge. It's very difficult to say exactly what that edge is, but it's a really good tone
that cuts through stuff, and that disappears when his voice smooths out. Although
that initially sounds nice, it isn't nice at all in real terms. And unless he's been on
the road, singing every night, that happens in the studio after four or five takes.
Having said that, I did comp his vocal from multiple takes on 'Who Are You', and
he also had to come back in and sing 'Ah, who the hell are you?' for the radio
version, and I then matched it up. After the track had been picked as the single,
we were worried that radio wouldn't play it because of him twice singing 'Ah, who
the fuck are you?', so I had him come in and redo those parts when I was mixing
the record at CTS in Wembley.
"I also remember getting Roger back to try to do some harmonies with himself on
certain songs. However, he's one of those great singers who's note-perfect when
doing a lead vocal and knows what he wants to do, but struggles when it comes to
singing backing vocals. He'll pick harmonies that cross, and if you say 'I just want
the third harmony on this line,' he might have a hard time. However, this seems to
be a common thing with lead vocalists."

On The RAK

Things staggered on until the end of the year. That December, string sessions
took place at Olympic, with Jon Astley's father Ted — composer of the music and
theme tunes to such well-known British TV shows as The Saint, Danger Man,
Department S and Randall & Hopkirk (Deceased) — taking care of the
arrangements for 'Had Enough' and 'Love Is Coming Down'. Then it was time for
a very welcome Christmas break; a break that quickly evolved into an extended
sojourn after Pete Townshend put his hand through a window and Roger Daltrey
had surgery following a throat infection. Only in mid-March did the band
reconvene, this time at RAK Studios in St. John's Wood, where Glyn Johns was
interested in trying out the API console and, more to the point, experimenting
with a much-needed change of scenery. This lasted less than a week.

"I remember an extraordinary day at RAK when

Glyn brought in another bass player and I thought
'This is a bit funny,'" Astley recalls. "You see, John
wanted to play kind of lead bass and Glyn felt they
needed someone to play solid-bottom bass to nail
down '905', which was very John-orientated. As it
happens, John was fine about it. He went 'Yeah,
yeah, whatever. Have a go,' and so in came Dave
Marquee — who'd played on the Joan Armatrading
sessions that I had done with Glyn — and we then
had this very strange session where we tried to
record a backing track with drums, two basses and
keyboards. John had actually brought in a backing
track that consisted of all these synths, and it was
on two tracks of the 24-track machine... I'd left the
first eight tracks and was recording the band on the Jon Astley in his mastering
studio in Twickenham,
last 16 tracks, but the maintenance man had wired located in the same house
up the remote with the wrong cable and when I put that belonged to Pete

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the last 16 tracks into Record it wiped the first eight. Townshend for 12 years.
We were 10 seconds into it and I went 'Hang on a
minute! Stop!' We'd already wiped John's backing track, although by managing to
stop it before we got too far into the song I was able to copy a verse and splice it
onto the front.

"I was thinking 'Oh no, I don't believe this day, I don't believe this day.' Keith was
awful, Glyn went home early, the rest of us went out for a bit of dinner, and I sat
down with the Who in a restaurant in St. John's Wood where they told Keith that
he was out of the band. It was a case of 'Unless you do something drastic, Keith,
we've got to find a new drummer.' I think Keith thought he'd been playing all right,
but his attitude was like 'Oh yeah, OK. Whatever.' He probably went home and
got depressed about it, but at the time he appeared to take it in his stride."

Taking Over The Reins

Another break followed, this time for a few weeks, before the sessions resumed
at Ramport that April with Keith still behind the kit but without Glyn Johns behind
the desk. Officially, this was due to a prior commitment, but there can be little
doubt that he'd also had his fill of the Who. Suddenly, Jon Astley found himself
producing as well as engineering.

"It was strange," he recalls. "The band

came to me and said 'Er, we'd like you
to take over as producer... if that's all
right with you.' They were very
reasonable! They seemed to have
great faith in what I could do, although
I have no idea why. I had never
produced anybody before. I suppose
for all they knew I could have gone off
with Glyn, instead of which I thought
'Oh, yes please. Thank you very much.
I've always wanted to produce the Photo: Richard Buskin
Who. I'm your man!' The original master tapes for the Who Are
You album.

"To be honest, I'm sure they just

wanted to get the record finished. And I suppose my role changed, insofar as I
was suddenly putting forward some of my own ideas — we had a piano player in,
but I felt that Pete's piano on one of his demos was better. I said 'I'd prefer you to
play it, Pete,' and I also suggested that he sing the middle eight. So I was
contributing towards the production, which was something I'd always wanted to
do. Fortunately, they were very, very open to suggestions."

But didn't Astley's promotion irk his erstwhile mentor, Glyn Johns? "No. If
anything, it caused more of a problem between me and Pete. He said 'This is
work, that's family. They're different things.' He was trying to be a normal dad

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and have a normal relationship with his family while making a record for this
legendary rock & roll band. It was very, very odd. I remember him hating the
drive to and from Battersea, so he bought himself a speedboat, thinking he could
belt up and down the Thames at 90 miles per hour. He'd get to Battersea quicker
than if he was driving... of course, he was wrong.

"When it came to his guitar playing, he did let it rip every now and again. For
instance, when he did the main part with a Gibson Les Paul on 'Who Are You',
recorded in the control room by Glyn, using a Gelf preamp and some nice plate
echo, everybody stopped and went 'Shit!' He reminded us of how great he was.
And this must have been when Pete was starting to go deaf in one ear, because
he had a specially built headphone box that would cut out whenever anything got
too loud."

Separate Lives

Aside from the overdubbed lead vocal, lead guitar, bass, drums and Rod
Argent's piano, most of the title track comprises the demo that Townshend
recorded in his home setup, including the acoustic guitar, rhythm guitar,
keyboards, handclaps and omnipresent backing vocals that he tracked himself.
Forget the promo video, filmed at Ramport in early May of '78, which features the
band members all gathered around the mic and working together as a band — in
reality, their parts were overdubbed separately. On previous records the band
had played together, even when overdubbing, but this time around Keith's
faltering abilities dictated otherwise, and this was hardly aided by the six o'clock
nightly routine of a maintenance engineer announcing 'The bar is open,
gentlemen.'

Little was straightforward on the Who

Are You project, which saw the band's
collective spirit showing definite signs
of wear and tear. Little, that is, aside
from Mr. Reliable, John Entwistle; ever
present, ever consistent, he delivered
few suprises both in terms of his
demeanour and his bass-playing
virtuosity. "If there was nothing to do,
John would simply replace his bass
parts," Astley says. "He'd set up Photo: Richard E Aaron / Redferns
exactly the same sound and say 'OK, The Who live in New York, circa 1977.
run the tape,' and he'd play exactly the
same thing. You could A/B between his bass parts over the course of five months
and they'd all be exactly the same. Still, at least he was always there, staying on
in the evenings to give Roger encouragement when he was singing, and he was
just great to have around. It was also the first time he'd written three songs for an
album, although that was probably because Pete, quite incredibly, had quite a
few songs vetoed by Roger: 'Nah, I can't hear myself singing that.'"

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Meanwhile, given one more chance, and with his back now firmly up against the
wall, Keith Moon finally got his act together, laying down all of his drum parts
within about 10 days. "He was great," asserts Astley. "The band couldn't believe
it. When they did '905', which was bass drum, snare, off-beat, on-beat,
everybody went 'That can't be Keith playing!' It was so unlike him. The timing
was great and it was difficult to do, but he pulled it off. The only thing on which he
couldn't play, which Pete warned me about, was 'Music Must Change'. Pete said
'It's in 6/8 and he doesn't feel 6/8. He never has, he never will. Don't even go
there.' He was right. We ended up putting footsteps on the track. On Pete's demo
he was walking around in a circle, and had it been quadraphonic it would have
been wonderful to listen to — you could hear his squeaky shoes, and the sound
of him walking around in a circle was the pace of the record... I mean, never mind
6/8, Keith never really felt 2/4 either. He felt orchestra — timpani here and big
cymbals there. It was acting, it was theatre, and he really was great. I loved him.

"After completing all his drum parts, he got a job working for the Who as a PR
man. He used to come into the studio and announce [in a very authoritative,
upper-crust voice], 'Yes, well, I have another meeting today. I have to go and see
these people...' He'd also go riding in Hyde Park. He just loved playing the
English gentleman. Very odd."

The Producer's Intuition

"Glyn Johns is the only person I've ever worked with who knew when a hot take
was about to happen," says Jon Astley. "He'd look at me across the control room
or sit beside me at the desk and say 'Mark the next one on the box,' and nine
times out of 10 he was right. He just knew when the band was about to come up
with the right performance. With Joan Armatrading he was right every time. When
we were recording 'Love And Affection' he would say 'We're not quite there yet,
but the next one will be,' and then he'd say 'Got it.'"

Flying Faders

When it was time for the mix, Jon Astley initially got the cold shoulder as a result
of the Who getting cold feet. "Pete came to me with this extraordinary excuse,"
Astley recalls. "He said 'Jon, this is the first thing you've ever produced, and
we're worried that, if it backfires and becomes a complete flop, it won't be good
for your career.' I thought 'Oh yeah, you fucker. I know what you're thinking.' Of
course, I just said 'Oh, OK, Pete. All right,' and he said 'We've asked Glyn to mix
it.'

By this time, I was working with Glyn on Eric Clapton's Backless album, so I
asked 'When's he doing it?' 'He's doing it at Olympic next week.' 'Oh, right. I
wondered what I was booked for.' As a result, Glyn and I spent an excruciating
few weeks inside Olympic's Studio One, where Glyn would push up a fader and
give me this quizzical look across the room, as if to say 'What the fuck's that?'

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Then he'd push up the vocal and sigh 'Oh, you

used a compressor on the vocal.' It was awful.
Anyway, I sat through that, put together the master
and delivered it as per his instructions, only for the
Who's management to reject it and say to me 'Jon,
you've got to mix this.' After all, I knew the record
inside out, I knew what I was doing and I had an
idea as to where I was going with it, and they heard
that in my rough mixes. So, it was quite a step for
them to take, rejecting Glyn's mixes, but they were
very, very unsure about the record because it
signalled a big change of direction in terms of
Pete's writing and they were obviously quite worried.

"John was with me for the mix at CTS, and the Photo: Michael Ochs
Neve we were using was the first ever board to Archives / Redferns
have little faders on motors going up and down. It Keith Moon at KROQ radio
was incredible. Then again, thanks to the great big studio, Los Angeles, 1977.
motor, it was impossible to grab hold of a fader for
just a second. I kept thinking 'Oh, I wish I could lift that snare drum just there,'
and of course I couldn't. Anyway, after I had finished mixing the album, we had a
playback session at Ramport, everybody loved it, and [management exec] Bill
Curbishley came in and said 'Jon, I've got a pair of Concorde tickets here 'cause
I know you love flying, and I want you to go and master the record in New York.'
All my life I've loved aircraft, and the opportunity to fly on Concorde was
amazing, so off I went to Masterdisk. That was management's way of saying
'Thank you for getting us through this,' which was really nice."

Write, Demo, Edit

The Who Are You album marked a major departure in Pete Townshend's
approach to songwriting, and the title track in particular saw him using his 24-track
home studio in Goring, Berkshire to the full. "Writing-wise, I think he was limited in
terms of what he was trying to do," says Jon Astley, "because he was going
through what I refer to as his Gilbert & Sullivan phase — 'Guitar And Pen' and
'Sister Disco'; all these operatic parts that didn't lend themselves to rock guitar.
Still, he was really getting into his piano playing, and I thought his piano playing
was quite astounding. On the previous album [The Who By Numbers] he'd played
acoustic guitar almost all the way through, and now he was into the keyboards —
all the little guitar bits in the middle eight of 'Who Are You' are from his own demo
— so he was always looking for something more to learn and master for each
record.
"His demos were fantastic. He came in with a 24-track demo of 'Who Are You'
which was about 22 minutes long, and this consisted of one extra verse and
chorus and a middle section that just went on and on into Neverland.
Extraordinary. Glyn and I listened to it with the band, and he said 'I wonder what
we should do with this.' Pete said 'Well, I think the middle section probably needs
cutting down a bit,' so Glyn said 'Yes, that's a good idea... All right, we'll see you,
um, tomorrow then, Jon.' And everyone went home! Because I knew Glyn wanted

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to work on 16-track, I actually edited the 24-track demo that Pete had made, and I
had pieces all around the control room marked with Chinagraph, indicating where
the various sections came from. God, it was such an absolute nightmare.
"I put together the middle section and I thought 'Oh yeah, that works. That's quite
interesting.' Originally, it had gone on for about 15 minutes, whereas now it lasted
about 90 seconds, representing all my favourite little bits. The whole thing was
driven by Pete's angular rhythm guitar part, played through an ARP 2600 suitcase
synth which had an auto-pan and a filter that was opening in time with the auto-
pan. This created a kind of wah-wah synth sound, and since it was played in four-
bar sections it was easy to edit together. Anyway, I bounced the whole thing down
onto about six tracks of the 16-track, and when everyone returned we played the
16-track Studer and Glyn went 'Yeah, it sounds good. What do you think, Pete?'
'Yeah, it sounds good.' And that was it. That became 'Who Are You', along with a
slightly reworked intro. The song was down to around seven minutes, and there
was a further edit that we did later when we took out another verse, but the lost-
verse version has since appeared on the [1996 CD] reissue of Who Are You."

Taken Away

At a time when punk was questioning

the band's sell-by date, Who Are You
resisted any urge towards bandwagon-
jumping by diving head-first into synth-
based prog rock and symphonic
arrangements. In this regard, it's hardly
surprising that doubts arose, yet the
album was hugely successful following
its release in August 1978, rising to
number two on the American charts
The custom Neve desk that was built for the
and going platinum in the process. The band's Ramport Studios, as it appeared in a
title track, released as a single the Neve brochure of the time. The console,
previous month, reached number 18 in which later found its way to Bearsville
the UK and number 14 in the US, yet Studios in New York State, was unique in
all of this was overshadowed by the several respects: details can be found on the
web site of Phoenix Audio, the UK-based
untimely death of Keith Moon on specialists in Neve restoration, who also
September 7 of that year — ironically, build new products based on original Neve
on the album cover he is sitting in a designs (www.phoenixaudio.net).
chair labelled 'Not To Be Taken Away'.

"He had got quite excited towards the end of making the record," recalls Astley,
who is now a full-time mastering engineer for the likes of Tori Amos, Chris Rea
and the Go-Betweens, while his remastering credits include all of the Who's
recordings and others by Led Zeppelin, George Harrison, Abba, Them, John
Mayall, Tears For Fears and Level 42. "I remember Keith really, really liking
'Guitar And Pen' and the things we did on that, and he was generally very up
about the record. However, although he'd pulled himself together after being
given the ultimatum, as soon as we finished the drum parts I know he went
partying and clubbing. As it happens, John also went clubbing every night... but
he did it in a very quiet way."

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CLASSIC TRACKS: The Who 'Who Are You?'

The Missing Minutes

Lastly, whatever happened to the 15 minutes of middle section that were edited
out of 'Who Are You' (see box, below)? Sitting in the upstairs studio of his
Twickenham house that overlooks the home of British blues, Eel Pie Island —
the house formerly owned by ex-brother-in-law Pete between 1968 and 1980 —
John Astley looks pensive. "The stuff I cut out included Pete fiddling on piano
and more acoustic guitar parts," he says. "I'd love to know what happened to it
all. It must be on a reel somewhere, and it might be worth digging up... I bet it's
down the road from here. Can you imagine putting together the original 20-
minute version? The only problem is, most of it wouldn't have any lead guitar..."

Thanks to Andy Neill, co-author of Anyway, Anyhow, Anywhere: The Complete

Chronicle Of The Who 1958-1978, for providing background information used in this
article.
Published in SOS May 2005

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 Composite Vocal Recording (Using Sonar 4)

In this article:
Composite Recording In
Composite Vocal Recording (Using Sonar 4)
Sonar Masterclass
Editing Time Published in SOS May 2005
Just Because We Can, Print article : Close window
Should We?
Technique : Sonar Notes
Clean-Up Time
Additional Tips

The audio sequencing facilities we have at our

disposal these days make it easier than ever before to
produce world-class vocal recordings by taking the
best parts from a series of takes and producing a
composite from them. Here's how to do the job in
Sonar 4.

Craig Anderton

Composite recording is the process of

recording multiple takes — usually in
quick succession — then editing the
best sections together into a single,
cohesive part. This is extremely
common with singers, where, for
example, the best version of the first
verse might be in the fifth take, the best
version of the second verse in the third
take, and the best chorus in the first
take. You determine where the best
parts are and then edit them together
into a single vocal track. Of course, this
technique is not limited to vocalists (guitarists, for example, will also find it very
worthwhile for assembling solos), but it's particularly applicable to vocals,
because there are spaces between phrases that make it easy to split large parts
into smaller sections.

In the days of tape recording (analogue or digital), the singer often recorded new
takes on individual tracks. By a process of muting and soloing, the best sections
would be isolated and then bounced down, in real time, to an empty track. The
old tracks could then be erased to make space for additional instruments or
takes. Where track counts were limited, punching-in came into the picture as
well. Sections that were judged not good enough were punch-recorded and

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Composite Vocal Recording (Using Sonar 4)

replaced, leaving the rest of the track untouched. However, with today's DAWs
and their virtually unlimited track counts, it's now no problem to spread a vocal
over just about as many tracks as you'd like.

Sonar has always been able to do composite recording, but version 4 adds
several new tools that make the process far easier and quicker.

Composite Recording In Sonar

The first aspect of composite recording is loop recording, which is what allows
you to put down take after take without having to stop recording.

Begin by defining the region you

want to loop, using the right and left
loop-locator points. Generally, you'll
want to leave a measure or two
before the part begins and a measure
or two afterwards, to give the The Show Layers command, new to Sonar 4,
performer a bit of a chance to regroup is extremely useful with composite recording.
after doing each take.

Next, decide whether you want each take to go in its own track, or be layered
with other takes in a single track. You can set the desired option by going
Transport / Record Options, then selecting the one you want under Loop
Recording (the Recording Mode setting doesn't matter). Prior to Sonar 4, I
would have recommended storing takes in separate tracks, because takes
stored in a single track had to be separated out into tracks manually anyway.
But version 4 allows the track to be opened up so that each take is displayed as
a separate layer (similar to how loop recording is handled in Cubase SX). This
is very convenient for editing, so I'd advise ticking 'Store Takes in a Single
Track' (see screen on the first page of this article).

Record-enable the track, then click on the Transport's Record button.

Recording will commence, and each pass will create a new layer within the
track. However, you won't actually see the layers until you stop recording. While
recording, it simply appears that you're recording into the track, and new takes
wipe out any evidence of previous takes. (Incidentally, if you have an older or
slower computer, or are starting to really push your CPU, you can turn off
waveform drawing by going Options / Global / General and unticking 'Display
Waveform Preview While Recording'. This takes a bit of the load off the CPU.)

I suggest doing no more than a half dozen takes at a time, for three reasons:
first, you don't want to have to wade through a million takes when you're trying to
locate and isolate the best sections; second, if you can't get a good performance
in six or seven takes, you (or whoever is performing) may need to practice the
part more; and third, you'll want to hear what you've done before committing to

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Composite Vocal Recording (Using Sonar 4)

too many tracks. Otherwise you'll just

be wasting your time if you do 20 takes
and they turned out to be going in the
wrong direction.

When you click Stop, the track For editing composite takes, it's important to
automatically unfolds to reveal all the be able to mute over specific ranges of times
layers. Note that you can unfold and re- rather than the entire clip.
fold the layers anytime by right-clicking
anywhere within the track, and respectively ticking or unticking 'Show
Layers' (see screen above).

Editing Time

Before you begin editing, make sure Automatic Crossfading is turned off. You
can do this with the Auto Crossfade button (located to the right of the Snap-To-
Grid button in the Track View toolbar), or just use your computer keyboard's 'X'
key as a shortcut. The reason why you must disable automatic crossfading is that
Sonar has no way of differentiating between times when you want overlapping
takes within a track to crossfade and times when you don't. If you're doing a
composite recording, as described here, you'll have lots of overlapping takes, so
if you attempt to move one Sonar will instantly want to create a crossfade with
the other takes.

Now it's time to take advantage of Sonar 4's Mute tool. Generally, I expand the
track height to make it easy to see the waveforms of the various loop-recorded
takes. The Mute tool has two options, selected via the drop-down menu to the
right of the Mute tool icon: Mute Time Ranges, or Mute Entire Clips (see screen,
left). For this application, Mute Time Range should be selected. The Mute tool is
the only Sonar tool whose function changes depending on its position in relation
to the height of the clip. Dragging over the lower half of the clip mutes the audio
under the area over which you dragged. Dragging over the upper half unmutes
any audio that was previously muted.

Now it's time to compare sections of each take to decide which you like best.
Reset the loop points around the area being evaluated. Start with all the tracks
muted, then unmute the area you want to listen to in one track. Keep muting and
unmuting various sections until you isolate all the parts that sound good (see
screen above).

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Just Because We Can, Should We?

Before you get started on creating your composite recordings, I think it's valid to
take a quick detour to explore its artistic ramifications. Of course, there are no
rules in recording, and the following is personal opinion. Still, I hope you find these
thoughts helpful as you work with composite tracks.
Composite recording used to be done with fairly large chunks of music — for
example, whole verses or complete phrases. But nowadays I see engineers slave
over piecing together vocals on an almost word-by-word basis. Vocalists come in
for a session, sing a dozen takes or so, then let the engineers apply pitch
correction and composite editing to create the 'perfect' vocal.
There's only one problem: vocals aren't necessarily meant to be perfect at the
expense of expressiveness. Vocals should tell a story, and telling that story
involves flow. A good vocal has expression that evolves over a period of time, with
high points, and low points — not just in terms of dynamics, but in terms of
emotional impact.
There's a story about when Barbra Streisand was recording vocals. While she
was away, the engineers did some editing to remove the breath inhales. Streisand
insisted they be put back in — and rightly so. She understood that breathing is an
essential component of vocals. When there's a long, loud inhale, you know the
singer is trapping a lot of air for what's going to be a long or loud statement.
Breath sounds also add intimacy, giving the effect that the person is next to you,
rather than being some disembodied 'voice machine' that creates vocals from
behind a Wizard of Oz-type curtain.
We have technology that's very powerful, but it's important to use that power to
serve the listener even more than the vocalist. A good producer recognises a
great take, and knows when to leave well alone. You can't create a great take;
you record a great take. Any editing is at its best when it gives something already
wonderful the extra 10 percent needed to make it transcendent, rather than
salvaging a vocal from a series of haphazard takes.
OK, rant over!

Clean-Up Time

Now let's tidy things up a bit. Click at

the places where you want to cut the
phrases, then type 'S' (for 'split'). I
generally split at the beginning of an
inhale. Now delete all the sections you
don't want to keep. Note that if you Muted audio is outlined with a dashed line,
while unmuted audio is a solid waveform.
delete a layer where all the audio is Although it was mostly the last take that had
muted, the entire layer will disappear, the best performance, phrases from two
thus making track height more other tracks were used as well.
compact.

At this point, if you untick 'Show Layers', the remaining regions will collapse into
a single track. That's useful, but the splits will remain, so if you want to

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Composite Vocal Recording (Using Sonar 4)

manipulate the clips (process them, move them, and so on), you have to make
sure that you select them all. I find it most convenient to bounce all the pieces
into a single clip, which is easier to manipulate. To do this, select all the clips by
drawing a marquee around them, then go Edit / Bounce to Clip(s) (see screen
below).

Additional Tips

By this point, you have a track that represents the pinnacle of musical
expressiveness... or at least something worth hearing! But you may want to take
the process just a bit further.

Quite a few vocalists double their parts

(sing the same part twice), to create a
thicker, more animated sound.
Although this can be done
electronically, 'real' singing sounds
better to my ears. One advantage of
composite recording is that you can
easily find not only the best version of
a vocal, but also whether any of the
The good bits have been isolated, with the
other takes work with the chosen take. unused material cut away. Now it's time to
If so, simply drag the take to another bounce all these into a single clip.
track and assemble a second track
with the 'doubled' vocal.

However, because you don't hear previous takes as you sing new ones, it's by no
means certain that the phrasing of takes sung at different times will be exactly
the same. It's also possible that the reason why you chose a particular take as
the best is because it stood out from the others for one reason or another,
making it even more unlikely that there will be any other matches in the pool of
takes. One technique that works for me is putting together the composite vocal,
then listening to it and singing along until I've learned the 'new improved' part. At
that point, I do a second composite vocal to create the doubled track.

Whatever method works for you, the important thing to remember is that you
should always choose the phrases you want to combine based on musical
continuity, not simply musical perfection. The technology should be there to
serve you — not the other way around.

Published in SOS May 2005

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Composite Vocal Recording (Using Sonar 4)

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Demo Doctor

In this article:
Free Souls
Demo Doctor
Doctor's Advice: Marketing Reader Recordings Analysed
and Music Published in SOS May 2005
The Running Culprits Print article : Close window
QUICKIES
Technique : Recording/Mixing
How To Submit Your
Demo
Demos should be sent on CD
or cassette to: Demo Doctor,
Sound On Sound, Media Resident specialist John Harris offers his demo
House, Trafalgar Way, Bar diagnosis and prescribes an appropriate remedy.
Hill, Cambridge CB3 8SQ, UK.
Please enclose a covering
letter with details of your
recording setup and a band/ Free Souls
artist photograph and/or demo
artwork (which we may use Venue: Home
here and on our web site to Equipment: Apple Mac G5, M Audio Firewire Track 1 -
illustrate your demo review). 1.4Mb
Samples from the two main 410 interface, Apple Logic Pro, Edirol PCR80
Track 2 -
demos reviewed will be placed controller keyboard, Rode NT1A microphone.
on our web site. Including 1.4mb
contact information, such as a Track 3 -
telephone number, web site Composer Davey Walker creates his atmospheric 1.4Mb
URL or email address, will music using vocal samples, ethnic sounds and
enable anyone who is synthesizer textures. It's an approach particularly
interested in your material to
suited to the Logic Pro setup, where everything can be accomplished in one
contact you.
place using the tools provided. And in singer Jenn Muir, he has found a voice
sympathetic to his compositions.

She doesn't feature on the opening track, but I

particularly liked its use of vocal samples with a
dub theme. The disparity between these
samples in both level and tone could easily be
rectified using Logic's automation system,
which is fairly simple and intuitive to use —
don't be put off by the busy graphics! The all-
important bass sample is excellent and provides
the right low-frequency content for dub bass
without getting totally lost on smaller speaker
systems, although it could have been a touch
louder. I was a bit perturbed by the clicking at
the start of the mix which, I think, is part of a sample. It could probably do with
cleaning up to avoid distracting the listener. Otherwise the track sounded very
well mixed with some particularly good Japanese flute samples, skillfully treated
with echo.

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Demo Doctor

The second track builds nicely, showing skill in the arrangement and choice of
sounds. For instance, the use of a mellow organ sound with a simple chord
structure is an effective counter to the busy percussion and effects. It's also the
perfect backing for Jenn Muir, whose distinctively English voice adds a certain
vulnerability to the song. Even so, I don't think harmony vocals, if used sparingly,
would weaken the intimate atmosphere created by the song, and, in combination
with the existing lead vocal, they might help create a more polished and
commercial production sound. Incidentally, I liked the use of EQ on the echo
applied to the vocal, deliberately thinning out the repeats and giving them an
ethereal quality.

'Something Special' is the title of the third piece and once again it's the
percussion that tends to dominate. The introduction features some nice Gamelan-
style tuned percussion, cabasa, kick drum and claves. It's a combination that
works well and builds subtly, with the pattern changing and extra percussion
being added towards the end of the arrangement. This gives the track a hypnotic
quality and it wouldn't be out of place on the shelves of 'New Age' shops or in
chill-out tents at festivals.

Doctor's Advice: Marketing and Music

Some of the CDs this month are very well presented, with effective artwork and
excellent publicity materials backed up, in some cases, by on-line resources, but
then they don't quite deliver in the recording department. Others are the exact
opposite — nicely recorded but with no real effort made to get a decent sleeve
together, let alone set up a web site! The musicians who send in demos to SOS
all have one thing in common though — they are looking for some recognition for
their talents. In these competitive times, both the quality of the recordings and way
they are presented need to be up to scratch if you want to make real progress as
an independent artist.

The Running Culprits

Venue: Home
Equipment: Novatech PC running Steinberg Track 1 -
1.4Mb
Cubase VST v5.1, Emu 1820M soundcard,
Track 2 -
Behringer UB2442FX and UB1622FX mixers,
1.4mb
Alesis M1 Active monitors, Behringer B2 Pro, Track 3 -
XM1800 and XM8500, Red5 Audio RVD1 and 1.4Mb
RV6, Shure SM58 and T Bone SC400
microphones, Sennheiser RS85 headphones.

I'm not quite sure why the band chose to put the least finished track first on this
demo but then they do say they are looking for useful suggestions about the
song. It starts with a nicely recorded bass solo that might be good as a short

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Demo Doctor

track in itself on a longer album, but here it's out of context. What follows has a
chorus but no verses as such, just an instrumental groove. Without some singing
or an obvious melodic hook, there isn't quite enough going on here to sustain
interest. As for the sounds, the drums come off the best, although the toms aren't
mixed high enough. The natural-sounding kit has too much of the overhead
microphones in the mix and a relatively small room acoustic is thereby exposed.
A touch of longer reverb on the overhead mics can sometimes give the illusion
that the whole kit is being played in a bigger room. However, looking at the
equipment list, the reverb options are limited, so it might be worth investing in a
good plug-in or outboard unit some time in the future to rectify this. Generally
speaking, closer overhead miking in a small room will give you more control
when adding reverb later on.

There is a general lack of continuity about the

three mixes on this CD and it's most obvious in
the bass end of the frequency range. It's not
helped by the bass guitar sound, which has the
kind of lower mid-range content fairly typical of
this budget equipment. DI'ing the bass and
applying a conventional bass player's mid-range
cut between 150 and 500Hz using the graphic
EQ in Cubase will improve it enormously. As it
stands, the overdriven guitar and bass are
fighting for space in the same lower mid-range
area (around 300 to 400Hz) on the first song, so
this EQ will also help separate the sounds in the mix.

Listening to the guitar on the second and third tracks, I was impressed with the
way that guitarist Tom Clements uses echo, tremolo and volume swells to create
sound textures. He also manages to combine acoustic and electric guitar sounds
very nicely indeed, especially on the second song where, despite some buzz
from the acoustic, he's layered the two parts really well. I also liked the use of
high notes on the guitar with a tremolo effect applied. Yet it's the third song that
features the best acoustic guitar sounds, cleverly coaxing a classy sound from
budget instruments.

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Demo Doctor

QUICKIES
Boondocks
This German band have sent in a demo of
classic rock cover versions designed to get
them more gigs. It's more or less recorded live
with no attempt to add production sparkle, but
at least the promoters will know that what they
hear is what they're going to get. The playing
is tight with some good sounds in the right vein
for the genre. The Mesa Boogie-style amp
simulations produced by the Line 6 Pod are
especially convincing, and the drum sounds
are good, but could benefit from a tighter,
more punchy reverb sound, like a gated or short room program on the group's
Lexicon MPX100. For synth fans, there are also some classic keyboard sounds
with more than a nod to Jan Hammer. The arrangements have been interpreted
pretty well, with Whitesnake's 'Ain't No Love In The Heart Of The City' sounding
particularly good.

Michael Phillips
Michael has produced a demo CD of jingles and stings designed for commercial
use. These include orchestral hits, groovy 16th-note synthesizer loops with a
touch of filter modulation and shorter and longer versions of various themes. The
mixes are good enough but the production is all a bit lightweight, and some more
work on the basic sounds to make them bigger and more powerful wouldn't do
any harm. For example, the orchestral hits lack low-frequency energy — a dose of
EQ boost in that area wouldn't go amiss. 'House Heaven' is a tad polite, the kick
drum needing more edge and the mix lacking energy and attack. Conversely, a
mellow piano-based composition entitled 'Quiet Expectation' is the best thing on
the CD. I also really liked Michael's funky wah-wah style synthesizer loops even
though they are in danger of being over-used on this sampler. With a view to
selling this music, it used to be easier to get into this market but now personal
recommendations and a decent agent are essential, so get networking.

Rioja
While reminiscent of American punk and new-
wave bands like the Ramones and Television,
this band also have a sense of humour. Songs
like 'Getting Married For All The Wrong
Reasons' are sure to hit the mark with a large
percentage of the CD-buying public. The
recording itself is a bit rough and ready and the
playing isn't always the tightest, to the point
where at times it sounds like they're only just
holding it all together. I was particularly
impressed with the way they manage to get
their Roland V-Drums to sound punky yet tight. However, the erratic signal levels
from the bass player's spirited performance don't aid the timing, and are a
candidate for compression at the mixing stage. With such a dynamic performance,
it may be that two compressors will need to be used in series — there's a chance

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Demo Doctor

that the bass sound would be ruined by the extreme settings that would be
required if using only one compressor. The CD cover is fun, so it was a surprise to
discover that Rioja have a rather tame web site which sadly only features one
song for download. This means you won't be able to hear the highly
recommended and amusing 'Jenny's Got A Big Mouth'. So come on, guys — let's
have some more mp3 tasters!
www.rioja.org.uk.

Horizon 9
The copious reverb and harsh EQ on these
mixes emphasise a deliberately aggressive
style with more than a touch of goth and
industrial about it. Against this backdrop, the
electric piano and atmospheric storm sounds
which introduce the second song come as a
pleasant surprise. The bass line and piano
improvisation provoke comparisons with
'Riders on the Storm' by the Doors, but that's
not a bad thing. The aforementioned large
amount of reverb worked on everything but the
vocals in this song — they should be more intimate and upfront. I thoroughly
approved of use of textured synthesizer and sound effects using backwards and
heavily vibrato'd chords. This really helped the song's dynamics and, with a bit
more work on the vocal sound, could turn the song into a moody classic for the
band.

Stuart Churchill
Stuart's songs bear the hallmarks of the Welsh working men's clubs where
country music is still in favour. His backings are unfussy and would benefit from
some classy melodic instrumentation in the form of electric-guitar or pedal-steel
fills, which he could obtain from sample CDs. He should also try miking up the
acoustic guitar with his Rode NT1 mic instead of DI'ing the signal from the pickup.
This will give a more classy sound, even if it takes a bit longer to set up. The
vocals sound good, and I'm so glad he avoided the temptation to add too much
reverb, which people who've played the clubs for years invariably seem to do on
their demos. He also sings decent harmony vocals and arranges these well.
Some real percussion like tambourine or cabasa might spice up the basic drum
tracks on these generally well-mixed songs.

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

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 Maximum RAM

In this article:
Taskbar Tips
Maximum RAM
Awareness Counts PC Notes
Flat Issues Published in SOS May 2005
Tweaking Flat-screen Print article : Close window
Monitors
Technique : PC Notes
Native Resolution

If you're tempted to approach the dizzy heights of

4GB of RAM in your PC, as supported by most
motherboards, there are a few things to bear in mind.

Martin Walker

Many PC musicians are still quite happy using 512MB of RAM, although I
suspect that the majority have now upgraded to 1GB, like me, so that they can
load more soft synths. An increasing number of musicians want to install even
more, perhaps to achieve greater polyphony with soft samplers such as HALion
and Kontakt, or they may want to have multiple sampled loops in RAM for real-
time pitch-shifting and time compression/expansion. Once in RAM, such loops
can be accessed almost instantaneously, and you may be able to reduce your
soundcard's latency as a result.

It's comparatively easy to install and use up to 2GB of RAM with Windows XP,
but although most motherboards support 4GB and beyond, making best use of
more than 2GB can be tricky. The currently 32-bit Windows XP supports an
address range of 4GB, but if you install 4GB of RAM it will, by default, only give a
32-bit application half of this. Fortunately, Windows XP Professional SP2 (but not
Home Edition) users can add a '/3GB' switch to their Boot.ini file to force their
PCs to allocate up to 3GB to programs (see screenshot, overleaf).

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Maximum RAM

Taskbar Tips
In Windows 98, opening more and more instances of the same application
resulted in new taskbar icons appearing, and their width becoming ever narrower,
until eventually they spilled over onto a second or even third row of hidden icons,
accessed by a scrollbar, or by dragging up the top edge of the taskbar to increase
its height.
Windows XP introduced us to the
delights of 'Button Grouping', where
taskbar buttons for multiple
documents opened by the same
application are all displayed in the
same area of the taskbar, to help
you find them more easily. If you
open up more than a certain
number, Windows combines them
into a single taskbar button labelled You can decide when taskbar buttons will be
with the name of the application. Grouped with TweakUI, and re-order them
This can be expanded into a vertical with Task Arrange.
list of the associated documents, so
that you can choose which one to view.
You can also fine-tune this feature using Microsoft's TweakUI which, under its
'Taskbar and Start Menu' section, has a section for Grouping. Here, by changing
the number under the heading 'Group any application with at least XX windows',
you can decide just how cluttered your taskbar becomes before Grouping leaps
into action on any application. Alternatively, you can decide whether Windows will
arrange the Groups by 'least-used first' or 'most windows first'. If you right-click on
a Group button you can also minimise or close the entire Group of windows in one
hit.
However, if you're really used to the Windows 98 way of doing things you can
disable Button Grouping by right-clicking on a blank area of the taskbar and
selecting Properties, then unticking the 'Group similar taskbar buttons' option.
If, like me, you regularly open a large set of applications in a specific order and
you get used to their relative positions on the taskbar, it can be annoying if one
crashes and ends up on the far right when re-opened, or if they're shuffled during
an Internet glitch. For a quick and easy way to reshuffle your icons at any time, try
downloading the 80KB freeware Task Arrange (www.softpedia.com/get/System/
OS-Enhancements/TaskArrange.shtml). This easy-to-use utility, written by Elias
Fotinis, runs under Windows 95, 98, ME, 2000, and XP and lets you rearrange
taskbar icons into any order at any time.

Awareness Counts

Even when this switch has been added, however, only applications that have
been compiled with their LAA (Large Address Aware) header flag set can take
advantage of the extra RAM. Few developers mention whether their products
have this flag set, but I do know that Steinberg's Cubase SX, Spectrasonics'
Stylus RMX and Mackie's Tracktion all do, and that Tascam's Gigastudio doesn't.

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Maximum RAM

Cakewalk don't currently enable it, but say they probably will on their next
maintenance updates.

I'll do my best to extract this info from other developers, in order to compile a list
of music applications that can take advantage of up to 3GB of RAM. I'm planning
to post this list in the PC Music FAQs section of the SOS web site Forums (www.
soundonsound.com). However, if any of you have official confirmation or practical
experience of getting other apps to address this much RAM, do let me know.
Apparently, it's possible for advanced users to enable the header flag in question
using Microsoft's Editbin.exe tool, supplied with the Visual Studio v6
development software, but as I've only got 1GB of RAM in my PC I've not needed
to try this myself.

Anyone wanting to install 4GB or more should ideally wait for Windows XP x64
Edition to be released later this year, and make sure they have a compatible 64-
bit processor, such as AMD's Athlon 64 or Intel's Xeon 'Nocona', plus 64-bit
device drivers for all their hardware. You'll also need 64-bit-compatible software,
such as Cakewalk's Sonar x64 Technology Preview (see Sonar Notes in the last
issue of Sound On Sound). Your plug-ins and soft synths will need to be ported
to 64-bit too, so I expect that there will be chaos before calm in the world of PC
music.

Flat Issues

Many musicians will have drooled over Apple's new 30-inch Cinema display, with
its massive 2560 x 1600-pixel resolution, but fewer will be able to afford its £2500
price tag. Fortunately, most of us have far more modest visual requirements to
go with our shallower pockets. Judging by the various home studio pics posted
on the SOS Forums, plus my visits to local musicians, many of you are still
running bulky CRT monitors, but recent developments really do mean that even
the most impoverished should consider one or more TFT (Thin Film Transistor)
flat-screen monitors. For one thing, you'll never again see picture distortion at the
edges of the display, due to the geometric limitations of the cathode ray tube or
to magnetic fields from nearby unshielded loudspeakers. Guitarists will no longer
encounter problems with hum pickup when working within a few feet of the
screen either, especially if they make sure that the flat-screen monitor's line-lump
PSU is carefully sited away from audio cables. In addition, flat-screen monitors
don't suffer from flicker, since their image is created in a completely different way
to the rapid-scanning approach of the CRT.

For musicians, there's another important issue: a flat-screen monitor produces a

much smaller footprint on your desk than a CRT monitor. Any large object sited
between the loudspeakers will result in audio reflections that impair your stereo
image, so a flat-screen display can often result in an improvement in audio
quality, as it presents a much smaller 'obstacle'.

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Maximum RAM

So what has changed in flat-screen

technology over the last couple of
years to merit the present discussion?
Well, many modern flat-screen
displays have significantly thinner case
edges, such that a modern 19-inch If you want Windows XP to make the best
TFT display may only be an inch wider use of 3GB of RAM, you need to add this
than the 15-inch TFT models of '/3GB' switch to your Boot.ini file (see text on
yesterday, making them even easier to page 276).
site in the studio and to use in multiple-
monitor setups. Some can even be hung on the wall for minimum acoustic
disturbance. Viewing angles used to be significantly narrower than those of an
equivalently sized CRT model, but fortunately that has also improved, with typical
modern flat screens offering horizontal and vertical viewing angles of around 170
degrees.

However, the most important factor for me is that prices, after remaining fairly
static during 2004, have now dropped significantly. When I bought a 15-inch TFT
monitor (identical in image size to a 17-inch CRT model) back in November
2001, they typically cost between £300 and £600, but I've just replaced mine with
a 19-inch model, whose display is both sharper and brighter, for just £220.

Tweaking Flat-screen Monitors

Some musicians feel they have to
buy a monitor with a DVI (Digital
Video Interface) connection to
achieve the ultimate image quality,
but in my experience the standard
analogue VGA connection provides
superb quality on most modern flat
screens. However, for the best
analogue image quality, you do
need to tweak the monitor's Clock
(width) and Phase (height) settings.
Most monitors provide an Auto-
calibrate function. For the best text
For the best possible text display, particularly
display, try enabling Windows XP's
on a flat-screen monitor, Microsoft's free
Clear Type font-smoothing and fine- Clear Type Tuner is a very handy tool.
tuning it, either via Microsoft's on-
line Clear Type tuner (www.
microsoft.com/ typography/cleartype/tuner/ 1.htm), or downloading the Powertoy
version from the same web page.

Native Resolution

So if you're tempted to invest in a new flat-screen monitor, what else do you

need to know? Well, TFT monitors do have some limitations, as regards colour

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Maximum RAM

and grey-scale accuracy (have a look at the excellent CRT versus LCD
Comparison pages at www.displaymate.com/ crtlcd.html for more details), but in
my experience, and although professional designers still stick to CRT displays,
you can successfully design CD artwork and web sites using a flat screen.

The biggest decision, however, concerns size. One inherent limitation of flat-
screen displays is that they have a 'native resolution', and if you choose a
different resolution, using the Settings page of Control Panel's Display applet,
you may see vertical bands of distortion. For instance, a typical 15-inch TFT
display with a native resolution of 1024 x 768 pixels will lose its sharp image
quality if you change to 800 x 600 pixels. For this reason, it's important to choose
a display with the most suitable native resolution for the task, since you can't
chop and change as you can with a CRT.

The most suitable native resolution depends on where the monitor will be placed.
For instance, my 15-inch Centrino laptop screen has a native resolution of 1400 x
1050 pixels, yet I find its incredibly detailed image perfect because my eyes are
only about 18 inches from it. However, in a studio you'll generally get a better
stereo image with the monitor screen about three feet away so that it's not in
front of your speakers. This also makes it easier for several people to view the
screen simultaneously. With a 15-inch TFT screen at its native resolution of 1024
x 768 pixels, you may find yourself squinting to read text and other fine details
from three feet away. A 17-inch model provides a 1280 x 1024-pixel resolution
and a welcome increase in screen real-estate, but once again may prove
frustrating for viewing fine details and web pages unless you have it close by, as
its pixels are still so tiny.

I found that a 19-inch model with the same 1280 x 1024-pixel native resolution
gave me the perfect combination of extra size and resolution over a 15-inch
model, and although its 5:4 aspect ratio makes the picture 'squarer' (both 800 x
600 and 1024 x 768 displays have a 4:3 aspect ratio), I find the extra height
useful for viewing more audio tracks simultaneously.

The 1600 x 1200-pixel resolution of 20-inch models returns to a 4:3 aspect ratio,
but unless you've got really good eyesight you'll need to work closer to it again,
and this size is still considerably more expensive than a 19-inch (the cheapest I
spotted was £480 — more than double what I paid for my 19-inch Videoseven
L19PS model). Larger sizes, such as 21-inch, still have the same 1600 x 1200
resolution, again making them more suitable for 'distant' viewing, but prices
rocket to over £900, while 23-inch models typically cost £1300. Personally, I
consider a 19-inch TFT monitor the perfect combination of increased size and
competitive price. I'm certainly well pleased with mine!

Published in SOS May 2005

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Maximum RAM

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Maximum%20RAM.htm (6 of 6)9/27/2005 9:26:55 PM

 Pop Shields: Why You Need Them

In this article:
How Vocal Pops Occur
Pop Shields: Why You Need Them
What Is A Pop Shield? Recording Tips
High-frequency Losses Published in SOS May 2005
Wind Shield Or Pop Shield? Print article : Close window

Technique : Miking Techniques

Pop shields are essential for most modern studio

productions, but what are they, and why are they so
important?

Paul White

Everyone has heard microphone

announcements ruined by loud popping
and banging noises, but we never hear
these noises when people speak
normally. That begs the question, 'What
are microphones hearing that we're
not?' If these noises are inconvenient
during live announcements, they can be
disastrous in studio recordings, so how
do we go about avoiding them?

How Vocal Pops Occur

It turns out that these pops and thumps occur mainly on what are known as
'plosive' sounds, prime examples being words that start with the letter 'B' or 'P'. If
you were to hold a lighted candle in front of your lips while speaking or singing
'plosives', you'd see the flame flicker, because we tend to expel a blast of air
when making these sounds. By contrast, if you sing a sustained 'Ahh' sound, the
candle will barely flicker at all, because you're mainly just producing sound
vibrations with your vocal cords and expelling very little air in the process.

The problem is made considerably worse if the mouth is very close to a

microphone. The plosive air blast is obviously strongest close to the mouth, and
when it slams into the microphone diaphragm it produces a very large
asymmetrical output signal. This may be so large that it can saturate the
microphone's output transformer (if present) or overload the mic preamp, making
the sound even worse. Engaging the low-cut filter on the mic (ideally) or preamp

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Pop Shields: Why You Need Them

may ease the overloading problem, but the basic cause of the popping will still
remain.

The problem is made even worse

because all directional microphones
suffer from the 'proximity effect', a bass
tip-up which makes the microphone
considerably more sensitive to low-
frequency sounds from very close
sources. A plosive blast is essentially There's nothing special about the
low-frequency energy, and hence it construction of most pop shields — in fact,
translates into a loud, low-frequency you can even make one for yourself out of a
pair of old nylon tights and a wire coat
thumping sound. hanger.

Capacitor mics of the type we use in the studio tend to be particularly susceptible
to popping, because their diaphragms are very light, so some form of effective
pop shield (or pop screen) is essential. Dynamic mics are a little more tolerant
because of their more massive diaphragm assemblies, but they are by no means
immune.

What Is A Pop Shield?

Even the best controlled singers (who naturally turn to one side or back off from
the mic when singing loudly or plosively) tend to get microphone popping on
occasions, so in most studios you'll see circular nylon-mesh screens that clip to
the mic stand and sit a couple of inches in front of the mic. You can see how
effective these are by trying the candle trick again. A good loud plosive with a
pop screen between the mouth and the candle should barely disturb the flame.

There's nothing magic about these

screens, and you can use ordinary
stocking nylon to make one for yourself
if you can devise some way to support
it. At one time commercial pop shields
were quite expensive, but competition
has caused prices to drop to the point
where making your own really isn't
worth the effort.

The way the pop screen works is

simple — sound passes through the
Nylon-mesh pop shields can cause a slight
fine mesh with just a little high-
dulling of the sound at high frequencies, but
frequency reduction, but plosives are metal-mesh designs such as this one don't
stopped dead. As the puff of air from suffer as much.
the mouth hits the mesh, it breaks up,
becomes turbulent, and loses its
coherence, so what starts off as an organised mass of air ends up being

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Pop Shields: Why You Need Them

randomised so that the air molecules are no longer all pushing in the same
direction. It's simple, but it works! To make the screens even more effective,
many designs incorporate two layers of mesh a short distance apart, so that
anything that gets by the first layer is mopped up by the second. Such a screen
will tame even the worst plosives. However, it is crucial that the windshield is
spaced a couple of inches in front of the mic capsule — there has to be a volume
of still air between the pop shield and mic capsule.

High-frequency Losses

Although the amount of high-end loss is generally very small, some engineers
still feel that nylon-mesh pop shields have too much effect on the sound.
Fortunately there's an alternative, which is to use a slightly more widely spaced
mesh made from woven or perforated metal. The larger holes have less impact
on high frequencies, but the hole spacing is still small enough to convert blasts of
air to harmless turbulence. Even a metal kitchen sieve will work, though its looks
leave something to be desired!

The reason the wire basket covering the capsule of a typical mic doesn't usually
prevent popping is that it is usually too close to the capsule to be effective,
though some hand-held capacitor models have the capsule set further back to
make the mesh more useful — effective though pop screens are, they're too
visually intrusive for most types of live performance.

The bottom line is that you should always use an effective pop shield when
recording close-up vocals. You don't need one for most instruments (though they
can be useful near hi-hats, which expel gusts of air when closing), and you don't
need them for recording vocals at a distance, such as choirs, but for typical
studio recordings where the singer is only a few inches from the mic, they are
absolutely essential.

Wind Shield Or Pop Shield?

Some microphones come with foam
wind shields that fit over the
microphone grille, but in practice
they tend to be ineffective against
anything more than a gentle breeze,
and they are no match for a full-on
plosive. Furthermore, the thickness
of foam invariably absorbs some
high frequencies, causing the sound
to become noticeably duller than it
should be. Wind shields can be
handy in live performance to stop the mic filling with drool, but they have a very
limited effect on popping.

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Pop Shields: Why You Need Them

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Pop%20Shields%20%20Why%20You%20Need%20Them.htm (4 of 4)9/27/2005 9:27:13 PM

 Troubleshooting Your Pro Tools System

In this article:
Moving Regions Between
Troubleshooting Your Pro Tools System
Sessions Masterclass
Numbers Of The Beast Published in SOS May 2005
Super Get Info Window In Print article : Close window
OS X's Finder
Technique : Pro Tools Notes
Other Troubleshooting Tips
Wheely Useful
Music Math v3

Whatever computer recording system you use, the

chances are it's going to fall over sooner or later, and
Pro Tools is no exception. So before you call
Digidesign's technical support line, take a look at
Sound On Sound's guide to diagnosing and fixing the
most common faults.

Mike Thornton

The Notes pages dedicated to specific software packages have proved to be one
of the most popular features of Sound On Sound in recent years, so we've
decided to develop our sequencer coverage in more detail. From now on, Pro
Tools Notes will be devoted to news and announcements that affect users of
Digidesign products, but it'll be complemented by an ongoing series of separate
workshop articles. These will provide the same sort of hands-on practical advice
that has been so popular in our existing Notes columns, but in even more depth.

Moving Regions Between Sessions

Have you ever wanted to import a region that you edited in a different Session into
the Session you are working on, but discovered that only the whole audio file was
available? This is because Pro Tools stores all the region information within the
Session and not the parent audio file. You can copy the region data from a
Session into an audio file by selecting the appropriate region in the region list and
choosing 'Export Region Definitions' from the 'Audio' pop-up menu. The region
information will then be written into the parent audio file, and the next time you try
to import the file into a Session, you will have a choice of importing the whole file
or the exported region or regions.

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Troubleshooting Your Pro Tools System

You can force Pro Tools to write region information into an audio file, and
can then choose whether to import the whole file or a specific region.

Numbers Of The Beast

For the first of our Pro Tools workshops, we're going to look at some of the most
common problems that can occur when you're using the program — usually
when you're in a crucial session, or up against an immovable deadline! As you've
probably found, Pro Tools has a habit of generating some rather cryptic error
messages when things go wrong, so firstly, let's see if we can demistify some of
these.

* DAE error -9035

This can come up for one of two reasons. One is that the drive that you use for
audio (and you aren't using the drive that has your operating system and
applications, are you?) is getting close to full. The best way to resolve this is to
use a different drive with more space on it. Failing that, clear some space on your
audio drive by backing up some old Sessions and then deleting them. To stop
this error message returning you may also need to defragment your drive, so that
the empty space on your drive is all in the same place rather than scattered
across the drive in little pieces. However, if it's an emergency and you don't have
time to clean up your drive properly, you can get some short-term relief from this
error by reducing the Open Ended Record Allocation to a smaller value. This is
done by going into Preferences in the Setup menu; in the Operations tab, set the
recording allocation to 30 minutes or less.

The other reason this error message can appear is if a Pro Tools Session (or
files within a Session) has more than 31 characters for the name, or uses illegal
characters like circumflexes, exclamation marks, brackets, semicolons, forward
and backward slashes, or international characters from other languages (any
characters in the Extended ASCII set, including accented characters). This can
easily happen when you open a Session created on a Mac OS 9 system, where

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Troubleshooting Your Pro Tools System

the forward slash was not an illegal

character.

This should have been fixed by the cs5

update of v6.4 but still seems to crop
up on older systems, so make sure you
have the latest version of Pro Tools
software for your hardware. A quick fix for errors relating to lack of hard
drive space is to change the Open Ended
* Disk too slow-related error Record Allocation to a fixed value.
messages (including -9073, -9094
and -9136)

There are a number of these and they can crop up for a variety of reasons
relating to the computer's inability to get the data off the drives fast enough.

One reason, on Apple computers, is

Mac OS 10.3's journalled file system,
which will automatically 'log' any file
modifications. If your computer crashes
badly enough to require a restart (or
you suffer a power failure), the OS can
then use this journal to help fix any
drive problems caused by the crash.
However, because any writing to a Journalling should be turned off on media
drive will also require a journal update, drives in OS X.
this slows down disk performance and
can throw up 'disk too slow' errors. Mac OS 10.3 ships with journalling on by
default. Digidesign recommend that you format their media drives with journalling
turned off. If you are using Apple's Disk Utility program to format drives, choose
'Mac OS Extended' format, instead of 'Mac OS Extended (Journaled)'. It is fine to
leave journalling active on your startup disk, and this should be the default
setting.

Another reason, which crops up more often on slower machines, is a hardware

buffer size set too low. You can increase the Hardware Buffer Size in the
Playback Engine option of the Setups menu to 1024 or even 2048 for more
reliable playback and recording.

Finally, you can get 'disk too slow' error messages if you're using too many host-
based (non-TDM) plug-ins, because the plug-ins are taking too much computer
processing power for the computer to handle other tasks such as playing the
audio! If you're happy with your plug-in settings, one way of simplifying things is
to use the corresponding off-line version of a plug-in from the Audiosuite menu to
make the effect permanent, rather than using the real-time version. To do this,
open the real-time plug-in from the insert point in the Mix window and copy the
settings to the clipboard. Then open the same plug in from the Audiosuite menu
and paste the settings in. I also tend to use the 'Create Continuous File' and

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Troubleshooting Your Pro Tools System

'Entire Selection' options, which will

consolidate the track into one file as
well, thus helping the load on the
system even more because it's no
longer having to deal with edits. You
obviously need to be happy with the
edits on this track at this point, and it
only works with plug-ins that don't have
any automation on them, but for 'static'
effects like EQ and compression it's
great.

If you need to cut some plug-ins but

you're also mindful that you might want
to change things at a future point, you
can duplicate the Playlist on this track
and then change the plug-in to
Audiosuite and consolidate the copy of
the track that's on the duplicate
Playlist. That way, if you need to go
back you can flip back to the original
Playlist. Don't forget to remove or turn
off the real-time plug-in, though, or
your track will be processed twice! Settings can be copied between real-time
and Audiosuite versions of a plug-in, allowing
to you make an effect or process permanent.
* NeoAccess Error

This error can come up when you are trying to save a Session. The short term fix
is to Save Session Copy on another drive. It is usually caused by a corrupted
Digidesign Database. Locate the Digidesign Databases folders at the root level of
each drive and delete the volume.ddb file. Also, try deleting the Database files
located in /Library/Application Support/Digidesign/Database/Volumes.

* Firewire problems with 002 and 002R

If you have one of Digidesign's Firewire interfaces, it should ideally be on its own
dedicated Firewire port. If your computer only has one Firewire 400 port, as most
laptops do, the question arises as to how to connect your Firewire 400 drives to
the computer. The recommended practice is to make sure your 002 or 002R is
the last item on the Firewire chain. If you daisy-chain drives to the 002(R) and for
some reason the computer loses contact with the interface, the drives will
disappear off the desktop and you may well lose data or worse, depending what
was happening at the time.

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Troubleshooting Your Pro Tools System

Super Get Info Window In OS X's Finder

Until recently, I had never heard of
the 'Inspector', a sophisticated
cousin of the Mac OS X Finder's Get
Info window, so chances are some
of you haven't encountered it either.
Using it, you can not only get
information on the file in the normal
Command+I way, but also preview
and play movies, sounds and so on,
and it allows you to use List View if Column View isn't your preference. Simply
add Option to Command+I when selecting the first file from the Finder, and a
window will open with the usual Get Info material in it. Put it somewhere
convenient on the screen, and now the clever bit: as you select different files, they
will appear in that window, instead of cluttering up your desktop with more and
more Get Info windows.

Other Troubleshooting Tips

The following apply to Mac systems only. First, check that the following are set
correctly in System Preferences:

Energy Saver: disable sleep mode

by setting the slider to 'Never'. Untick
'Put the hard disks to sleep when
possible' and in the Options page, set
Processor Performance to 'Highest' if
this is available on your Mac.

Display: set your monitor resolution

to a minimum of 1024 x 768.

Classic, Start/Stop tab: Untick 'Start

Classic when you log in'. Pro Tools 6.
x is not supported when Classic Mode
is running.

Date & Time: Verify that the date is

correctly set and that you are not
using 24-hour time.

If you're experiencing problems, it can

be worth trashing Pro Tools's
preferences. Go to Users/your user
name/Library/Preferences and drag You can leave the original version of a track
DAE Prefs, DigiSetup.OSX and Pro

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Troubleshooting Your Pro Tools System

Tools 6.x prefs to the trash. Then intact by copying it to a duplicate Playlist to
apply off-line processing.
empty the trash and restart the
computer.

Something else that can often help when things go wrong is Repair Permissions.
Quit Pro Tools and run Apple's Disk Utility application (left), which you will find in
the Utilities folder inside your main Applications folder. Select your startup
volume, go to First Aid and select 'Repair Disk Permissions'. You should do this
every time you install and/or update any software.

Finally, the Digidesign 'Plug-in Validator' is a utility that will help you check that all
the plug-ins you're running are up to date and compatible with your version of
Pro Tools (the exception is Waves plug-ins). It's in the Applications/Digidesign/
Pro Tools/Pro Tools Utilities folder. Manually remove any unqualified plug-ins
from your plug-in folder and then go and get the latest versions.

Wheely Useful
A mouse with a scroll wheel is extremely useful on both Mac and PC systems, as
it allows you to adjust what you can see without having to move your cursor away
from where you are working.
Without any modifier keys, it will scroll the Edit and Mix windows vertically.
When you hold down Shift, the scroll wheel will scroll the Edit window horizontally.
When you hold down Option (Mac) or Alt (Windows), it will zoom tracks in the Edit
window in and out horizontally keeping the cursor in the centre of the screen.
When you hold down Shift and Option (Mac) or Shift and Alt (Windows), it will zoom
tracks in the Edit window in and out vertically.

Music Math v3
I have just got to tell you about this
latest find! Music Math is an
application which converts values for
musical use. You can calculate the
time-stretching to use on a sample if
you change the tempo, or the
number of semitones to transpose a
sample, the time-stretching to apply
if you transpose a sample and you
don't want to change the duration,
delay times in ms to synchronise
with tempo, and there's even a tap
delay function. Tap the beat on your
keyboard, and the software will tell
you what the tempo is. In addition,
there is a timecode calculator for those of us who can't add up time in their heads!
You can find it at www.macmusic.org/softs/view.php?id=1536 and the best news

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Troubleshooting Your Pro Tools System

of all is that it is free!

Published in SOS May 2005

Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK.
Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
electronic, is expressly forbidden without the prior written consent of the Publishers. Great care has been taken to ensure accuracy
in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
The views expressed are those of the contributors and not necessarily those of the publishers.

Web site designed & maintained by PB Associates | SOS | Relative Media

file:///H|/SOS%2005-05/Troubleshooting%20Your%20Pro%20Tools%20System.htm (7 of 7)9/27/2005 9:27:22 PM

 Using Folder Tracks in Cubase SX

In this article:
Making Folder Tracks
Using Folder Tracks in Cubase SX
Using Folder Tracks Cubase Notes
Introducing Folder Parts Published in SOS May 2005
What's New In 3.02 Print article : Close window
Editing With Folder Parts
Technique : Cubase Notes

Steinberg originally introduced the concept of Folder

tracks in Cubase VST, as a way of organising the track
list in the Arrange window. This month we look at
how this feature became even more powerful in
Cubase SX, and how to make the most of it.

Mark Wherry

Folder tracks are a tremendously useful way of organising the Project window,
allowing you to place regular tracks within specially created Folders, to create a
hierarchical structure in the Project window's track list — just as you might create
folders on computer disks to organise your files. This can be useful for a number
of reasons, including cleaning up the track list to make working with large
numbers of tracks more manageable, and as an editing tool for working with
groups of related tracks.

Making Folder Tracks

To create a Folder track and place other tracks within it, select Project / Add
Track / Folder to create a Folder track in the track list. You can now drag other
tracks from the list onto the Folder Track, releasing the mouse button when a
green arrow appears on the Folder track. To move a track out of a Folder again,
simply drag it outside (either above or below) the Folder. Folders can be deleted
in the same way as any other track, although you should bear in mind that
deleting a Folder track will also delete the contents of the Folder.

While dragging a track into a Folder is essentially a very simple thing to do, there
are a couple of quirky rules about how Cubase handles this procedure that are
worth bearing in mind. When you drag a track into a Folder, it will always appear
at the top of the list of tracks within the Folder, although it is possible to re-order
tracks in the Folder by dragging them, in exactly the same way you would
normally re-order tracks in the track list. A caveat to this comment is that you can

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Using Folder Tracks in Cubase SX

put a track into a specific location in a Folder by

dragging it between two tracks that are already When a Folder
track is
there.
selected, a
hierarchical
If many tracks are selected, you can drag them into overview of
the Folder's
a Folder in one go. The added tracks will appear at
contents is
the top, but in their original order. If you want to displayed in
create a new track within a Folder, select the track the Inspector,
you want the new one to appear beneath and use allowing you to
one of the 'Add Track' commands from the Project select a track
and see that
or Quick menu. You can't double-click in the track
track's info in
list to create a new track inside a Folder in this the lower part
situation. of the
Inspector.
Any type of track can be included in a Folder track,
although it's worth pointing out that when you create Group and FX channel
tracks Cubase automatically places them in a Folder called 'Group Tracks' or 'FX
Channels' respectively, creating the Folder if necessary. This is also true for VST
Instrument Output channels and certain types of automation track. Note that in
any of these cases, tracks automatically placed inside Folders by Cubase can
still be dragged anywhere you like in the track list, including into other Folders.

Using Folder Tracks

The simplest application of a Folder track is to allow you to reduce the number of
tracks shown in the Project window without having to delete anything. To open or
close a Folder track, simply click one of the Expand/Collapse Folder buttons on
the Folder track itself, in the track list: either the small +/- button at the bottom-left
of the track, or the small folder icon. Alternatively, to open or close all Folder
tracks in the track list at the same time, hold down Control/Apple when you click
the +/- button.

You'll notice that Folder tracks have similar controls to regular tracks, including
Mute, Solo, Record, Monitor and Lock buttons, and using any of these controls
affects all the tracks contained within the Folder. For example, pressing the Solo
button will activate the Solo button for all the tracks in the Folder, providing a
quick way to solo different groups of tracks in your Project. This is particularly
useful when you're working with a large number of audio tracks that all need to
be record- or monitor-enabled at the same time: place the audio tracks in a
Folder and use the Folder's record and monitor enable buttons. Finally, the
Folder track's Lock button is also a convenient way of locking the entire contents
of a Folder with a single click.

When you have a Folder track selected in the track list, you'll notice that the
Inspector displays a hierarchical list to represent the contents of the Folder. One
neat thing about this is that you can then select a track in this list in the Inspector,
to display the normal Inspector sections for that track.

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Using Folder Tracks in Cubase SX

Once you've been using Folders for a while, it's inevitable that a certain curiosity
will arise and you'll wonder if it's possible to drag a Folder track into another
Folder track. Fortunately, you can: dragging a Folder into another will move that
Folder and all its contents into the other one, with Cubase preserving a
hierarchical view in the track list and making it very easy to see how the tracks
are organised. But when you're working with sub-folders, bear in mind that any
operation you perform on a sub-folder (or a track within a sub-folder) will have
consequences for every Folder track above or below that track or Folder in the
hierarchy.

Introducing Folder Parts

Although the features offered by Folder tracks are useful, the most interesting
possibilities are offered by Folder Parts, which are the Parts shown on a Folder
track in the Event display. Folder Parts are created automatically based on the
contents of the tracks contained within a Folder, and because of this you can't
create them manually. For example, say you have a Project consisting of one
Folder track that contains one empty MIDI track. If you create a MIDI Part
between bars one and nine on the MIDI track, because that MIDI track is with a
Folder track, a Folder Part will be created on the Folder track between bars one
and nine.

Now, say you have another MIDI track

in the same Folder and you create
another MIDI Part from bars two to
eight. The Folder Part on the Folder
track stays exactly the same, and if
you move the Folder Part around, the
Parts on both MIDI tracks stay locked
to their relative positions within the
Folder. Where things can become a
little confusing is where you have
Notice how Folder Parts contain miniature
overlapping MIDI Parts with a Folder. representations of the Parts on the tracks
For example, say you created another within the Folder. If your Folder Parts appear
MIDI track with a Part from bars five to to be empty, simply increase the vertical
13, the Folder track would now contain height of the Folder track, by dragging the
two eight-bar Folder Parts that overlap. bottom of the track downwards.
If you moved the first Folder Part (that
lasts from bars one to nine), the MIDI Parts on the first two tracks (the ones
lasting from bars one to nine and two to eight) would move with it, but the
overlapping MIDI Part on the other track, that lasts from bars five to 13, would
remain untouched.

A useful way of looking at it would be to consider each Folder Part a container for
Parts in its own right, so Parts on tracks within a Folder track are stored in
different Folder Parts, which you can always see in the overview of the Folder

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Using Folder Tracks in Cubase SX

Part if the track's height is sufficient (see screen above). Since it can become
confusing to have different Folder Parts, you may want to merge Folder Parts
together, although unfortunately you can't use Cubase's Glue Tool to do this, as
you might expect. Instead, while it seems a little counter-intuitive, where you
have overlapping Folder Parts you drag the start point of the later Part a little way
back into the earlier Part and when you release the mouse button the Folder
Parts are merged.

Once you get your head around the initially confusing concept of Folder Parts,
they can pretty much be manipulated just like any other Part in the Event display,
and any operations you perform on a Folder Part affect the Parts on the tracks
within the Folder. If you move a Folder Part, the Parts in the Folder are also
moved, if you delete a Folder Part, all the Parts in the Folder are deleted... you
get the idea, although there are a few things to be aware of. Notably, it's easy,
when lassoing Parts, to include a Folder Part by accident and delete a host of
Parts you didn't intend to remove, especially in large Projects with nested Folders.

What's New In 3.02

Just as I was finishing this month's
column, Steinberg officially released
build 622 of Cubase SX as version
3.02. On the new features front, this
version of Cubase inherits some of the
additional Appearance preferences
from Nuendo 3, including the
Appearance-Meters page that allows
you to configure the colours used for
the level metering in the user interface.
The only other entry in the new
features list is that the status of the
Play Order track is now saved with the
Project, and although some might have
regarded this a bug-fix, it's still
welcome! On the bug-fix front, issues
that could have caused a crash, such Cubase SX 3.02 includes a new
as using the Delete Time command, Preferences page, enabling the colours
the Generic Remote, Quicktime movies of its level meters to be completely
on a Mac with Show Thumbnails customised.
enabled, and opening the Drum Editor
window after enabling Score drum notation, have been addressed. A full list of
fixes and the download itself are available now from www.steinberg.net.

Editing With Folder Parts

When you double-click a Folder Part, Cubase will open one or more editor
windows. For example, if a Folder contains only MIDI tracks, double-clicking a
Folder Part will open all the relevant MIDI Parts in the default MIDI editor, which

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Using Folder Tracks in Cubase SX

is normally the Key Editor. Similarly, if you double-click a Folder Part for a Folder
containing only audio tracks, and the audio tracks contain only audio Parts, these
Parts will be opened in a single instance of the Audio Part editor. Otherwise, if
the audio tracks contain audio Events, each Event will be opened in its own
Sample editor. If a Folder contains a mixture of audio and MIDI tracks, Cubase
will open all the MIDI Parts in a single editor, all the audio Parts in a single editor,
and all the audio Events in their own editor windows.

Folder Tracks can be really useful, for example, when you're working with
orchestral arrangements, since an orchestra consists of many recognised
instrumental groups. You can create Folders for each orchestral family, such as
Strings, Brass, Woodwinds, and so on, and place the tracks for each instrument
into the appropriate Folder. The best thing about this method is that when you're
working with the Score editor you can open the Folder Parts for each
instrumental group to see all the staves for the instruments within that folder,
without having to make manual selections of multiple Parts all the time.
Obviously, this example applies to any arrangement where you have a defined
group of instruments, such as a big band arrangement, or a string quartet within
a pop song.

Published in SOS May 2005

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The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
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 Why & How To Partition Your Music PC Hard Drive

In this article:
Back To Basics
Why & How To Partition Your Music PC Hard Drive
Location, Location, Location PC Musician
Suitable Partitioning Tools Published in SOS May 2005
Laptop And Single-Drive Print article : Close window
System Tweaks
Technique : PC Musician
Windows Activity
Multiple Drive Configurations
Potential Rewards
Further Reading
Final Thoughts Did you know that sensibly partitioning your hard
drive or, if it's already partitioned, simply swapping
the positions of your audio and sample partitions
could result in a significant improvement in PC audio
performance? We explain the whys and wherefores.

Martin Walker

Each time I return to the subject of hard-

drive partitioning, the goal-posts have
moved significantly. Back in August
2000, when I wrote the 'Divide And
Conquer' feature, many of us were still
using drives with a capacity of 8GB or
less and running Windows 98. By the
time I wrote 'The Great Divide', in SOS
March 2003, this had jumped to 80GB, and many of us were more interested in
splitting our drives to run Windows 98, 2000 and XP alongside each other in a
multi-boot configuration. Two years on, in 2005, I still find myself answering lots
of questions about partitioning in the SOS Forums, but while most of us are now
using Windows XP, there are several new issues to face and splitting drives has
almost become mandatory, as some of us install models with a capacity of
200GB or more.

Back To Basics

As most PC musicians already know, you don't have to leave each of your hard
drives as one huge and rather unmanageable storage area. It's generally far
more productive to split each one into several partitions, to keep your data better
organised, and therefore safer. This also helps to minimise any reduction in
performance due to file fragmentation, by keeping the fragments within a smaller

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Why & How To Partition Your Music PC Hard Drive

area of the drive so that the drive's read/write heads don't have to dart about so
much when accessing files.

Most people also agree that having two drives in a music computer is generally
better than one, since you can devote one to Windows and its applications and
the other to audio storage, safe in the knowledge that they won't interfere with
each other in any way that reduces overall performance. This approach also
offers a security benefit: you can back up the data from one drive onto a spare
partition on the other, so that if one drive fails your data can still be retrieved from
the surviving one (although this isn't a substitute for a proper backup regime onto
other media).

There's now a huge number of musicians who run PC laptops on stage, for
location recording, or simply for greater convenience. They generally rely on a
single drive, yet still want to achieve the maximum audio performance from it.
Many musicians are also now using streaming soft-samplers, which brings up
issues surrounding hard drives once again. If you're running one of these
samplers alongside a multitrack audio sequencer, for example, do you need
three drives (one for Windows and its applications, the second for audio tracks,
and the third for sample libraries) to achieve maximum soft-sampler polyphony
without compromising the maximum number of audio tracks? If not, what's the
best way to split your data requirements across one or two drives?

Location, Location, Location

Let's start by considering the implications of partitioning. While it can make our
lives a lot easier, it results in a set of storage areas, each performing slightly
differently, and with the possibility of interactions between them in real time. Let
me explain.

The maximum number of simultaneous

audio tracks your hard drive can
manage is directly linked to its
'sustained transfer rate', rather than the
burst speeds often quoted by
manufacturers, because streaming
audio is generally a continuous task.
This sustained transfer rate is, in turn,
directly related to spin speed: the
faster a drive spins, the more of its
data sectors can be read per second.
This HDTach graph of one of my Seagate
Barracuda 7200 SATA hard drives (see next
So the 4200rpm drives found in some page) shows how sustained transfer rate
laptops generally achieve the lowest varies across the surface of a hard drive
number of audio tracks, the 5400rpm from outside to inside. As you can see,
ones found in old desktops and many although the average sequential read speed
current laptops are better, and the has been measured as 46.8MB/second, the

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Why & How To Partition Your Music PC Hard Drive

7200rpm drives commonly found in graph shows that it actually varies from
more up-market laptops and the vast 58MB/second on the outside to 30MB/
second on the inside.
majority of desktop PCs tend to be the
most popular for audio work. There are
faster drives available (including 10,000rpm and even 12,000rpm models), but
these are significantly more expensive and often more noisy. In any case,
7200rpm models can already run more simultaneous audio tracks than most
musicians need.

The sustained transfer rate will also vary from the outside to the inside of any
drive, simply because the read/write tracks are arranged in concentric circles.
Since the outer tracks are longer, they contain more sectors, and thus at a fixed
spin-speed more sectors of the outer tracks can be read in a single revolution. So
the fastest area of any hard drive is always on the outside. With most (but not all)
drives, the sustained transfer rate falls steadily from the outside to the inside, and
may typically drop by half in the process (I have seen exceptions where the rate
suddenly jumps up again slightly in the middle, or falls in multiple steps like a
sawtooth waveform, but these seem comparatively rare).

Sustained transfer rates can be obtained from manufacturers' drive specifications

and are generally quoted as a range, such as "32-58Mb/second", or as an
'average' value, but it obviously is sensible to make best use of the fastest areas.
Fortunately, you can easily measure the sustained transfer rate of any partition of
your drives using the now-famous DskBench utility (a free download from www.
sesa.es/us/dl/dskbench.zip). Simply copy this tiny 36KB file onto the partition or
drive you want to measure and then run it from there. Among various other
results, DskBench will provide you with figures for Write and Read — the two
sustained transfer rates that ultimately determine audio recording and playback
performance — as well as approximate CPU overhead. If the latter comes in at
under five percent or so, you'll be reassured that DMA data transfers are
correctly set up for your drives.

If you want to see how the sustained transfer rate varies across the entire
surface of your drive, try downloading the 1Mb HDTach utility (www.
simplisoftware.com/Public/index.php?request=HdTach). Now up to version 3, this
free hard-drive benchmarking tool provides a useful graphic readout of
Sequential Read Speed from outside (left-hand side) to inside (right-hand side),
as well as Random Access Time (a measure of how fast the drive can retrieve
randomly located sectors) and CPU Utilisation (once again, to check the
effectiveness of your DMA transfers). It's also a Windows-based utility, which
makes it easier on the eye than DskBench.

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Why & How To Partition Your Music PC Hard Drive

Suitable Partitioning Tools

Microsoft include a handy
partitioning utility, Disk
Management, with Windows.
Accessed from Control Panel by
clicking on Administrative Tools and
then on Computer Management, the
Disk Management tool can then be
launched by clicking on Disk
Management in the Storage area of
the left-hand pane.
You can choose what you want the
top and bottom views to be in the
right-hand window, but by default Windows' own Disk Management utility is
these views display a list of very handy for basic partitioning, and for
viewing the current arrangement of partitions
Volumes (partitions) at the top, and
from the outermost (left-hand) to innermost
a Graphical representation of the (right-hand) positions across the surface of
same beneath, showing primary, your hard drives.
extended and logical partitions, plus
any free space. What many
musicians haven't cottoned onto is
that the lower Graphical view also
shows the layout of partitions from
outside (left) to inside (right) of each
drive.
The Disk Management tool is very
useful if you know exactly how you
want to split and format a new drive,
delete existing partitions or change
drive letters. However, you can't use
it to re-size existing partitions, so
unless you get it right first time and If you want to create, alter and maintain a
complex set of partitions, a dedicated utility
never subsequently change your
like Partition Magic will make your life a lot
mind as your drives fill up, you'll
easier.
need a more flexible tool, such as
the famous Partition Magic,
developed by Powerquest, but now owned by Symantec (www.powerquest.com/
partitionmagic).

As you can see from the screenshot, Partition Magic shows exactly the same
arrangement of partitions for my two drives as the Disk Management tool, with the
bonus that they are appropriately scaled to show what proportion of the total drive
they occupy. Partition Magic really comes into its own when you want to
reorganise your data: you can re-size, move, copy, or delete existing partitions at
will, split them or merge them together, convert them from one format or partition
type to another, change their cluster size, or create new partitions, ready to install
additional operating systems.
A few alternatives to Partition Magic are now available, including Paragon
Software's Partition Manager (www.partition-manager.com) and Acronis Disk
Director Suite (www.acronis.com/ homecomputing). Both can be ordered on-line

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Why & How To Partition Your Music PC Hard Drive

for $49.95 (compared with the $69.95 of Partition Magic), and both offer a
comprehensive range of functions. Although the Acronis product doesn't currently
offer a couple of options that the other two do, it does support Windows Server
Editions (you need to buy the more expensive 'Pro' versions of the competing
software to do this). On the other hand, Partition Manager does offer an advanced
Defragmentation tool, which looks very useful.

Laptop And Single-Drive System Tweaks

If you have a single drive in your PC and it's formatted as one massive partition
(the default in most mainstream systems), it will seem quite nippy when
delivered. However, it may gradually become more sluggish as you install more
applications and store more data as, inevitably, later storage will be physically
further towards the inside of the drive. Not only will files stored in these areas
suffer from slower transfer rates, but access times may also become longer, as
the hard drive read/write heads have to keep jumping longer distances during
normal use to reach all parts of the drive.

Windows does its best to help, by

periodically re-organising files so that
those accessed most often are moved
to the outer portion of the drive, using
its 'pre-fetch' optimisations (I discussed
this in 'Speeding Up Windows' in SOS
April 2003). You can force this re-
organisation to occur by running
Microsoft's Disk Defragmenter utility,
but you can also help a great deal by
splitting your drive into several
partitions, and placing data that's only This HDTach graph shows the performance
needed occasionally (such as large of my 40GB 5400rpm Seagate Momentus
image backup files, or your collection laptop drive. The read speeds are
of application update files) on the considerably lower than those of the
7200rpm Barracuda drive shown earlier, but
innermost partition. they reveal a similar percentage drop from
outside to inside.
In general it's best to make the partition
most often accessed the fastest one, and for most general-purpose software this
would be the one containing Windows and its applications. Your PC would then
boot up in the shortest possible time and launch applications as quickly as
possible. However, for musicians streaming audio files from their hard drives, it
makes more sense to ensure that the partition containing audio files has the best
performance, despite the fact that this will result in Windows and applications
launching slightly more slowly.

Here's a practical example, based on the 5400rpm 40GB Seagate Momentus

hard drive in my Centrino laptop. As you'll see from the HDTach screenshot
above, the performance of my laptop drive falls off from about 36MB/second to
about 29MB/second across the first 20GB — a 17 percent reduction. So if, like

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Why & How To Partition Your Music PC Hard Drive

many people, you have a huge 20GB partition devoted to Windows and its
applications, with the rest of the drive devoted to audio duties (Single Drive
Scheme two in the diagrams below, simply swapping the positions of the
partitions could potentially increase your maximum audio track count by 17
percent.

Rather than re-allocating a large

proportion of the outer part of the drive
to a single audio partition, it makes
more sense to create one partition just
big enough to house your current audio
project (say, 5GB), and another further
in for all your older, finished projects. In
this way you can achieve the maximum
track count without compromising
Windows performance too much (in my
graph, drive performance has only
fallen by about three percent at the
5GB point).

In this example (Single Drive Scheme

three), we've gained a 17 percent
improvement for audio tracks and only
lost about three percent for Windows,
simply by creating an outer 5GB With a single drive it's worth spending some
project partition (see 'Windows Activity' time thinking about the best way to organise
its storage, especially if it's a slower laptop
box for more details on whether the
model. Dividing the typical single huge
slight loss of Windows performance will partition of a mainstream PC (Scheme one)
be noticeable). You can extend this into Windows and audio partitions (Scheme
idea, if you've got a streaming sample two) may help you organise and back up
library, by creating another larger your data, but it may actually degrade audio
performance. A much better option is to
partition for the library, between the
create a small outer partition to house your
current project and Windows partitions. Current Project (Scheme three), moving the
In the case of this particular (rather completed project into the Audio Backup
small) 40GB laptop drive, you could partition when completed, and perhaps
perhaps devote 15GB to the sample adding a further outer partition for Streaming
Samples (Scheme four), if required.
partition, still leaving plenty of space
for Windows and an inner data and
backup partition (Single Drive Scheme 4). Although Scheme 4 further
compromises Windows loading times, placing the audio and sampling partitions
next to each other minimises access times. This arrangement will probably
provide the best audio and sample streaming performance on a single-drive
laptop.

By the way, don't confuse the requirements of streaming sample libraries with
samples that are always loaded into RAM in their entirety. The latter are no
different from patch-based libraries. These two categories (RAM-loaded samples
and patch libraries) should ideally be separated from your streamable sample
libraries and placed into yet another data partition further in on the drive. After all,

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Why & How To Partition Your Music PC Hard Drive

the only improvement you'll see if you place these libraries in an outer partition is
a slight reduction in their initial loading times.

Some of you may have noticed from the HDTach screenshots that since most
drive read speeds fall to approximately 50 percent as you move from outside to
inside, buying a larger-capacity drive will help you make the most of the fastest
outer area. On a 200GB drive, for example, there may be little fall-off in read
speed across the outermost 50GB, making this area ideal for large current
projects or sample libraries. So even if you only have one drive in your PC, the
larger its capacity, the less you'll compromise audio performance with the
sampling split of Single Drive Scheme four.

Windows Activity
Swapping the positions of your Windows/applications and audio data partitions
initially sounds like a great idea, until you start worrying that perhaps Windows
and application performance may, in turn, suffer if they're moved to a slower
inside partition. After all, running from a partition with a lower sustained transfer
rate is bound to make Windows boot up more slowly, since its system files will
take slightly longer to load into RAM. Applications will also launch more slowly,
albeit by a tiny amount.
But will this swap also result in
worse real-time performance of your
MIDI + Audio application? After all,
the whole point of the classic-twin
drive Windows + Applications and
Audio setup used by most specialist
music retailers and DIY musicians is
to remove any possibility of
interaction. Splitting a single laptop
drive, or indeed splitting the
Windows + Application drive of a
dual-drive setup to provide
streaming sample storage, would When you're recording or playing back a
seem to go directly against this song, the Windows C: volume (dark blue
philosophy. On the other hand, trace) is rarely accessed, with only two tiny
some PC system builders suggest blips during the several minutes for which I
was recording this disk activity. The vast
that once your MIDI + Audio
majority of disk accesses occur in the P:
sequencer has finished loading, and
projects volume (yellow trace) during
you've loaded whatever additional streaming of audio tracks, and the S:
files are required for running your samples volume (pale-green trace) during
choice of soft synths and plug-ins, streaming of Gigastudio sounds.
neither the application nor Windows
will need to access the drive more
than very occasionally, and often not at all.
I decided to perform some tests to throw some further light on the matter. I started
by running the Filemon utility from Sysinternals (www.sysinternals.com). This
monitors and displays all file system activity so that you can see how Windows
and your applications access and use files. You can decide which volumes to
monitor, and you can stop and start the capture process at any time. Although it's

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very easy to get swamped with data, with care you can learn a vast amount about
which files are being accessed on which drives.
After a run-through playing a Cubase SX song with a clutch of audio tracks, I
discovered (as I half expected) that for most of the time the only volume being
accessed was the one containing the audio files used by the song. Further
monitoring showed that only when opening or closing a file dialogue window (load,
save, import, and so on) or exiting Device Setup (when Cubase saved its new
Defaults.xml file) was there any obvious activity on the C: volume. Even when
Gigastudio was running alongside and accessing my dedicated Samples partition
there was still no additional activity on the C: volume. I made similar checks
running Cubase on my XP General partition and found similar results.
Sonar was slightly different: I found that it accessed the C: volume once every 2.5
seconds for a tiny amount of time even when songs weren't playing, and during
playback it accessed its huge TTSRES13.DLL file about five times per second.
However, each access lasted less than a millisecond, leaving the C: volume
inactive for at least 99 percent of the time. I think we can safely assume that
placing an audio or streaming sample partition on the same drive as the Windows
partition won't compromise performance very much.
Other audio applications may differ in their approach, but I suspect that you're
reasonably safe in creating an audio/sample partition on the same drive as
Windows if there's some reason why you can't devote another drive to this task —
just be aware of possible interactions and try to keep them to a minimum. While
you may not achieve 100 percent of the performance that you might get using a
separate audio/sample drive, it might stay near 99 percent most of the time — but
it could possibly plummet very occasionally if the drive is suddenly called on to
perform another task. One possible task might be Window's Paging File, created
by default on the C: volume for use as virtual RAM, particularly when you're
running low on the real thing. I've written about the page file on several occasions
in the past, and on my machine it doesn't seem to get accessed very much.
However, those running video applications often find that they need a huge page
file, which is accessed quite often. If you find yourself in this situation, another
option is to create a small extra partition (of several GB in size) near your audio or
sample partition, to use as a dedicated page-file partition. That should minimise
any disruption of streaming activity.

Multiple Drive Configurations

While some mainstream PCs may arrive with a single massive hard drive, most
PCs intended for audio will be better off with two smaller ones, or one small and
one large — the first for Windows, applications and other data, and the second
for audio duties. This not only removes the possibility of audio accesses being
compromised when Windows wants to read or write a system file, but also
means that you can back up the data from one drive to a partition on the other.

But even with two drives it pays to consider their partition splits carefully.
Although the commonly used split of duties shown in Twin Drive Scheme one
(above) generally works well, it still makes sense to split the Windows drive so
that you can separate your own data (documents, graphics, and so on). Then, if
anything ever happens to Windows, such as corruption or a bad virus attack,
your work will be unscathed, even if you have to re-format the Windows partition

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Why & How To Partition Your Music PC Hard Drive

and install your applications afresh. At the same time, it's a sensible move to
create a third partition dedicated to image files of your Windows partition (made
using a utility such as Norton's Ghost), so that you can restore it to full working
condition within a few minutes rather than having to reformat and start again. The
setup just discussed is, in fact, the backup split (Twin Drive Scheme 2).

If your songs contain a lot of audio files, the 'Current Project' audio tweak
suggested earlier for the single-drive setup will also benefit a twin-drive system
(Twin Drive Scheme three), as it ensures that you always achieve the best
performance for the files you're currently using, but you've now got the space to
make the Current Project partition much larger, if you're using the 24-bit/96kHz
format, without compromising Windows performance at all.

Twin Drive Scheme four is yet another

variant for those who want to stream
lots of samples at the same time as
playing back many audio tracks. I've
also moved the Data and Windows
Backup partitions onto the second
drive, to leave more space for
samples, and to keep it away from
Windows itself, for more security. This
time it's not so important to move the
Streaming Samples partition further to
the outside of the drive than the
Windows partition, since it's likely, with
a twin-drive setup, that each drive is
somewhat larger than my laptop
example. In my own desktop partition
setup, for example (shown in the
second screen on page 130), my first
drive is a 80GB model with three small
Windows partitions, each of 5GB, with
the remaining 60GB devoted to
Samples. Referring to the HDTach
graph for my drive, starting the sample
partition at the 15GB point only results
in a loss of the top five percent of the
drive's performance, which I can live
with. If I had a larger first drive, this
performance reduction would be even
less. On the other hand, placing the When you have two drives, it's tempting to
Windows partitions on the final (inside) leave them both as single huge partitions
15GB of the drive would result in at (Scheme one). However, separating your
data from Windows and backing both of
least a 33 percent increase in Windows these partitions up (Scheme two) is a far
file-access times. more secure arrangement. Audio
performance will undoubtedly be improved
by ensuring that the project you're currently
working on has its files on the fastest part of
Potential Rewards
the audio drive (Scheme three), while with a

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Why & How To Partition Your Music PC Hard Drive

little thought you can stream samples from

the Windows drive without too many
If you've just bought a PC it's sensible repercussions (Scheme four).
to make the most of its potential
performance by organising the drive or
drives into partitions. However, if you've already got a mature system with loads
of stuff installed on your drives, will re-organising Windows, applications and data
result in significant benefits? Well, it depends on how you work, whether or not
you've already hit the drive-performance ceiling of your existing setup, and how
much the sustained transfer rate of a particular partition could be improved by re-
siting it, if it's currently a limiting factor on performance.

If you work mainly with MIDI hardware and software synths and plug-ins, and
your MIDI + Audio sequencer application doesn't show a high reading for hard
drive activity then there's absolutely no point in reorganising — it may make you
feel that your PC is better set up, but you won't notice any improvement in
performance. Creating a special outer 'Current Project' partition might shave a
little off the initial loading time of your songs, while creating a special outer
'Samples' partition to store large non-streaming sample libraries (like those of
Spectrasonics' Atmosphere or Trilogy might instead shave a second off initial
loading times when loading a new patch, but you'll see no real-time
improvements. Even if you're running loads of audio tracks, if you've never found
your drives 'running out of steam', you won't notice any improvement.

However, for musicians running Gigastudio, HALion, Kontakt, or any other

streaming soft-sampler, spending a few hours reorganising your drives could
immediately offer benefits in polyphony terms, especially if you use instruments
such as piano and harp that can easily consume 64 notes or more by
themselves. It may even allow you to avoid having to buy and install a third drive
just for sample storage.

Reorganisation could similarly improve the lot of the PC musician running a

laptop with a single drive, when squeezing every last drop of performance out of
the internal drive may make the difference between being able to continue using
the laptop by itself, or having to cart around an external drive to beef things up.

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Why & How To Partition Your Music PC Hard Drive

Further Reading
I've written on various occasions in the past about partitioning hard drives. Here
are links to the main articles:
Backup Strategies For The PC Musician www.soundonsound.com/sos/aug04/articles/
pcmusician.htm (covers organising your data, file-naming tips, what to include in backups,
data recovery, different media, file compression, dedicated software, sharing data).
Speeding Up Windows www.soundonsound.com/sos/apr03/articles/pcmusician0403.asp
(covers chipset drivers, Intel Application Accelerator, Bootvis, boot-up time, device
initialisation, Windows XP boot time).
The Great Divide: Partitioning PC Hard Drives For Multi-Boot Systems www.
soundonsound.com/sos/mar03/articles/pcmusician0303.asp (covers reasons to split,
format & partition types, partition options, Boot.ini name syntax, Microsoft multi-boot
options, XP boot errors).
One Box Two PCs www.soundonsound.com/sos/may01/articles/pcmusician.asp (covers
swapping drives, multiple operating systems, putting up the partitions, installing a second
OS).

Final Thoughts

You don't have to religiously follow any of the schemes I've outlined here: once
you've grasped the reasons for each of my suggestions you can adapt them to
your needs. For instance, if you've already created dedicated audio and sample
partitions and are currently struggling to achieve a higher sampling polyphony,
but you have no problem running lots of audio tracks, it may benefit you to swap
their positions on your drive or drives. As long as you bear in mind the falling
transfer rate from the outside to the inside of a drive, the relative sizes of
partitions are also entirely up to you. The beauty of the partition utilities
discussed in the box on page 130 is that you can change partition sizes at any
time, if you run out of space in one and find you don't need as much space as
expected in another. It's not uncommon to shuffle partition sizes or even their
relative positions several times before finding the best arrangement.

My tests on Windows partition activity while running audio applications confirm

that laptop owners will generally achieve acceptable performance by splitting a
single drive. I've been successfully streaming Gigastudio from a partition on my
Windows desktop drive for several years, with no problems to date. But don't run
away with the idea that you should necessarily abandon the twin-drive approach:
using a single drive is always a slight compromise, so you should be prepared for
at least a slight drop in polyphony or maximum track count. You'll also have to be
careful to keep the Windows and audio partitions as physically close as possible,
to minimise the impact of extra drive activity, as well as avoiding any utilities that
regularly access files on the Windows drive.

The central issue is awareness of how placing partitions on a drive affects their
performance — and don't forget that it's still important to keep individual

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Why & How To Partition Your Music PC Hard Drive

partitions regularly defragmented and the data within them regularly backed up.
Finally, remember that, however capable the latest partition utilities are, no piece
of software is 100 percent foolproof, so make sure you have backup copies of the
data in each partition before you attempt to move them elsewhere. Then, if the
worst happens during the reshuffle, you won't have lost any data in the process.

Published in SOS May 2005

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All contents copyright © SOS Publications Group and/or its licensors, 1985-2005. All rights reserved.
The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or
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 Yamaha AW4416 User Tips

In this article:
Setting Levels For Mixing
Yamaha AW4416 User Tips
Automatic Muting Masterclass: Part 1
Managing Scenes During Published in SOS May 2005
Mixdown Print article : Close window
Fine-tuning Dynamic
Technique : Recording/Mixing
Automation
Using The Waves Y56K
Card With Scenes
Wave Hello
Saving Space: Editing & Combining the guts of an 02R mixer with a fully
Optimising featured multitrack recorder, the AW4416 ended up
Effective Track Naming being a prodigiously complex beast. Our hands-on
The Value Of
workshop shows you which of those snazzy features
Defragmentation
work best in practice, and how to use them efficiently.
Archiving Songs
Multitrack Mastery
Software Updates & Manual Tom Flint
Supplements

Arguably the biggest product release of

1996 was the Yamaha 02R digital
mixer, featured on the cover of SOS in
February of that year. The product
offered pro-quality automated mixing
with motorised faders, two onboard
effects processors, and EQ and
dynamics on all channels; and all for
the sum of £7049! In those days, that
seemed like an unbeatable batch of
features for such a modest price,
despite the fact that the various I/O
boards and accessories bumped the overall figure rather higher. Still, the industry
marches on, and just over four and a half years later, the guts of the mixer had
been bundled together with a hard disk recorder, sampler, and CD-RW drive to
create the considerably less expensive AW4416.

The more basic AW2816, and AW16G were released at a later date, each one
improving on the integration of mixer and recorder, and therefore becoming more
user friendly. By their standards, the AW4416 is a little daunting, but it offers a
great deal of functionality to anyone willing to spend the time getting to grips with
it.

This short series won't attempt exhaustively to describe all the workstation's
features in detail — the two manuals do a fine job in that department already.

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What I'll be doing is giving practical examples of how all these features can be
used and abused to best effect during recording and mixing.

Setting Levels For Mixing

Each of the AW's mixer channels has an attenuator, and these become very
handy when preparing for a mix, because they allow each fader to be used over
its optimum range. If you look closely at the faders you will see that they offer the
most control around the 0dB position. For example, moving the fader about a
centimetre from the 0dB position will cut or boost a signal by around 5dB,
whereas when the fader is at the bottom of its travel, the same movement
represents a 40dB change! As a result, mixing is easier and more accurate when
the fader movements are made near to the 0dB mark.

Unless all your sources happen to be at the correct volume straight away, the
chances are that after you've found the right mix balance some faders will be up
at +6dB while others may be hovering down at -40dB. By adjusting the
attenuators on each channel, all the faders can be set at a 0dB starting point.

To do this, start with the track which is mixed the loudest, and is fundamental to
the mix — often the lead vocal. If the track fader is at +4dB, then pull it down to
its unity-gain position, and then use the channel attenuators to bring all the other
channels down by the same amount. The quickest way to apply the change to all
the tracks it by pressing the Ch View button to bring up the channel pages and
then using the Sel buttons in each fader layer to select each track or pair of
tracks in turn, pulling down the attenuator for each one as you go. Once this has
been done, the previous mix balance will be regained, but most of the faders will
still not be at unity gain.

Any faders that are above 0dB should be taken down to unity gain, and their
channel attenuators increased by the same amount to retain the mix balance. For
example, a guitar track which is still faded 2dB up can be taken down by that
amount, and then have its attenuator increased by 2dB. Conversely, any faders
below 0dB gain need to be pushed up — if a fader is moved from -10dB to 0dB
then the attenuator can be lowered by 10dB.

Once this process has been competed for all tracks, the basic mix should sound
the same as before, but all the faders will be in a line at 0dB. When it's time to
create an automix, you will be able to boost and cut from this position. If you find
that one of the tracks needs more boost in places, just attenuate everything else
down a few decibels more to gain the extra headroom. This only takes a few
seconds to do, and it retains the relative balance.

Alternatively, consider using the channel dynamics processor to dial in a few

decibels of gain. The disadvantage of using the channel attenuators is that they
come before the dynamics processor, so any pre-existing compressor threshold

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Yamaha AW4416 User Tips

settings will require re-evaluation. Adjusting the output of the dynamic processor
avoids the problem; however, it also means that if you later decide to bypass a
redundant compressor, for example, then the balance will immediately change.
More food for thought is that the dynamics processors allow up to 18dB of gain to
be added, whereas the attenuators do not go above 0dB and therefore reduce
the chances of overloading the output.

If the overall balance is correct, but clipping is occurring on the main Stereo buss,
press the Ch View button and the Stereo track selection button. Here you will find
the attenuator for the stereo buss, as well as EQ and dynamics processing,
allowing the signal to be brought under control.

Automatic Muting
The AW4416 has
to juggle its If you try to record
resources on and play back too
occasion, many tracks
particularly when it simultaneously, the
AW4416 will
is attempting to
automatically mute
play back and some of the playback
record at the same tracks so that it can
time. In 24-bit cope with the
mode, for example, recording duties. If
it often has to mute you don't like its
some playback choice of muting,
tracks if too many then you can change
other tracks are it from the main track
already playing view screen.
back. If you're
using the Quick
Rec 2 page to route the audio for recording, this causes the tracks to mute
automatically, but the tracks that are muted are often ones you actually need to
hear when performing your overdub, so some user intervention can be required.
In the track view screen, the arrow keys allow any track to be highlighted and its
mutes to be turned on or off. However, to change a mute when a track is armed
for recording you will first need to mute an alternative track. At this point, a non-
vital track can be muted leaving the important one available for playback.

Managing Scenes During Mixdown

I've come to the conclusion that the most efficient way to automate a mix is to
start by laying down a series of Scenes, positioning them at key places
throughout a Song, and identifying their positions with markers. Each Scene can
be used to determine the basic mix palate for the block of audio it precedes, and
it creates a foundation from which smaller changes can be made using the
dynamic automation.

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Yamaha AW4416 User Tips

Of course, a whole Song can be automated without the use of Scenes, but
Scenes are a much neater way of working because they act like chapters in a
book, insomuch as they break everything down into manageable lumps. It is also
quicker to apply mix changes to Scenes than it is to work through and edit
masses of dynamic automation bit by bit.

I like to set up a Scene which is relevant to the largest portion of the Song and
then work from there, rather than starting at the beginning and creating a new
Scene for each section as it is encountered. For each section I load the core
Scene, alter it appropriately, and then re-save it with a different name.

It's a good idea to test

your basic Scene mix
on a variety of
systems before
creating numerous
edited copies, so that
they don't all need to
be changed if some
drastic alterations to
the overall setup are
required. If, however,
you've already set up
your Scenes and find
that you need to
change the same
settings across
several, you can cut
down on the work by
saving edits into the
various EQ,
dynamics, and effects
libraries. For
example, if the EQ on
a stereo drum track
needs changing for all
of a Song's Scenes,
first make the
changes for the main
Scene and save the
new setting into the
EQ library. When
editing the rest of the
Scenes, rather than
making all the same
changes by hand, just
recall the saved setup
from the library for the
drum part, and then
re-save the Scene.

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Yamaha AW4416 User Tips

Once a series of When you come to mixdown, any fader set far below its unity gain
carefully programmed position makes mixing less accurate, because the resolution of
Scenes is in place, a the AW4416's faders gets coarser at lower gain settings — you
Song actually can tell this from the scale printed to the side of each fader in the
left-hand picture. To bring the fader back to roughly unit gain
requires very few real- while retaining your mix balance, first press the Ch View button
time automation (centre picture) and use the cursor keys to navigate to the
moves before it's channel's attenuation control. In this example channel 11's fader
complete. is about 20dB too low, so the Att control should be adjusted to its -
20dB setting (inset picture) so that the fader can be raised back
to its highest-resolution position (right picture).

Fine-tuning Dynamic Automation

When it comes to mixing, there are often occasions when a small block of audio,
possibly just a note or two, gets in the way and needs to be removed. Perhaps
the timing is out, or the notes are just not right. In some circumstances in may be
possible to edit out the offending notes, or simply mute them. However, the
sudden absence of a track of audio can sound unnatural. In such cases a fader
move is a more appropriate solution, as it effectively allows the mix engineer to
bury the unwanted notes. Ducking just a note or two is usually difficult to get right
by hand, but the automation allows a fader move to be tuned almost to perfection.

To tackle the scenario outlined above, solo the offending audio and find the start
of the first duff note. Write down its exact location, using the Waveform display to
help if necessary. Then do the same for the end of the last bum note. Next,
defeat the solo, so that the whole composition is heard, and rewind to just before
the beginning of the section. Set the Automix facility to record fader data, make
sure that the Fader Edit Out mode is set to Return, and start the track playing. At
the start of the duff section, briefly duck the fader and stop recording — at this
stage it doesn't matter if the fader move starts at the right spot or goes to the
right level.

Make your way to the Automix Event List page (the F4 tab in the main Automix
screen) and find the fader data you have just recorded. Using the cursor keys
and data wheel, change the time of the first move to the start time you noted
down, and alter the last event to the end time. You will notice that the smallest
timing increment is 25 milliseconds, so data positioning is not as accurate as
editing, but that still means that there are 40 steps to every second, so you can
get things more or less right. The next step is to reposition the rest of the fader
increments between the two outside points. To keep things smooth, it's probably
best to have the volume drop rapidly in small steps to its benign level, and then
rise back up in the same fashion. The actual fade amount can then be tweaked
until the notes are suitably unnoticeable.

Such methods are time-consuming, but worth considering when a small and
accurate patch-up job is needed. The same basic principles can also be applied
to other automation data, like pan positioning, or the placement of Scenes.

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Yamaha AW4416 User Tips

Using The Waves Y56K Card With Scenes

I am among those AW users who have

invested in the Waves Y56K mini-
YGDAI card (see the Wave Hello box
for more information on this product).
This card offers eight mono effect
chains (any pair of which can be linked
for stereo operation) each with as
many as five effects blocks into which As you can see from this block diagram of
the various processors can be loaded. the AW4416's mixer (shown in full at the
back of one of the two manuals), the one
The effects chains can be inserted into disadvantage of using the channel Att
the path of any relevant monitor controls to bring channel faders into their
channel, or they can be fed from a area of greatest control resolution is that this
spare aux send buss and returned via may interfere with any dynamics processing
any vacant input channels. The former already set up. In such cases, the make-up
gain control in the relevant dynamics section
routing is most useful for the various can be used instead, as long as you don't
compressor, limiter, de-esser, and EQ bypass it at any later stage.
effects, while the latter suits the reverb
and delay algorithms. The Y56K's
dynamics processors and equalisers all greatly improve on the in-built
equivalents provided as standard by the AW4416's mixer.

Clearly, eight channels of processing offers a lot of flexibility, but I soon found
myself wanting to swap effects setups around during a Song by using Scenes
with different Y56K chain setups. Unfortunately this is not practically viable,
because you get significant drop-outs when recalling Scenes with drastically
different Y56K chains — the card has to reload its whole internal DSP setup. For
this reason, I plan my basic mix with the relevant channel insert and send
settings for the eight Waves channels, and then make sure they remain the same
throughout a mix. If needs be, I avoid changing Waves chains by editing audio so
that it physically moves to a track with a particular insert effect at the point where
a Scene change happens. That said, it is possible to change, say, an EQ setting
or compressor level within a Y56K effects algorithm under Scene control without
causing the card to reload its whole DSP setup. Another option, of course, if you
think you're going to run out of effects horsepower, is to bounce audio to disk
through the Waves processor, keeping the unprocessed audio safe on a spare
virtual track.

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Yamaha AW4416 User Tips

Wave Hello
The Y56K mini-YGDAI card offers
some of the company's most sought-
after processors and effects for use
in both mixing and mastering
situations. I'd certainly recommend it
to any committed AW4416 users
who feel frustrated with the in-built
dynamics, EQ, or delay/reverb
effects that come as standard.
The board does take up a slot that
could otherwise be occupied by a
mini-YGDAI I/O board, although the
Y56K does include eight channels
of ADAT lightpipe I/O. It's also
important to note that the Y56K has If you own the Waves Y56K board, you can
some compatibility conflicts with now download three new plug-ins from www.
other optional I/O boards. Firstly, it y96k.com, although you'll need to update
cannot be used when the AW4416's your card's operating system to run them.
other mini-YGDAI slot has either of
the Apogee converter boards installed — this risks damaging the multitracker. A
similar risk of damage prevents the simultaneous use of a second Y56K. In fact
only the MY4AD, MY8AD, and MY4DA can be used alongside the Y56K.
The standard Y56K tools are the L1 Limiter, L1 Ultramaximizer, Renaissance EQ,
De-esser, Renaissance Compressor, Multitap Delay, and Trueverb reverb.
Current owners may also be interested to learn that there are now three more
Waves products available for download and installation on a standard Y56K card:
the Renaissance Bass, Renaissance Vox, and L2 Ultramaximizer. The first of the
three generates harmonic sub-bass frequencies and is best suited to dance-music
applications. The Renaissance Vox is an easy-to-use compressor and gate
combination specifically designed for use on vocals, and the L2 Ultramaximizer is
an alternative to the L1 Ultramaximizer. All three processors can be bought
together as the Waves Power Bundle, and can be downloaded into the Y56K via a
computer.
Before any new updates can be made, however, the Y56K Maintenance Update
OS has to be installed. The software is also available in download format from the
same site. Whether or not you desire the three extra Waves products, the update
may well be worth installing anyway, as it is free and is said to increase stability
and crash recovery in case of power failures. The installer uses the Y56K's
RS232 connector for performing the update, so a standard RS232 nine-pin serial
cable and a PC with a COM port are needed. Unfortunately the installers are
available only for Windows machines.
www.y96k.com

Saving Space: Editing & Optimising

If you are recording your Songs at 24-bit resolution then you will almost certainly

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find that a four-minute composition

using all 16 tracks creates a file that is
hundreds of megabytes in size. If there
are additional virtual tracks too, then
it's possible to get a project of a
gigabyte or more. Obviously, big files
take up a lot of hard disk space, but
they also become problematic when
attempting to make backups. For
example most CD-Rs and CD-RWs
offer about 700MB storage, so a 1GB
Song requires two recordable CDs and
increases backup time considerably.

To help matters, the AW4416 has an

Optimise feature (press the Song key
and then the F3 tab to find it) which
allows unwanted data to be discarded,
thus reducing file sizes significantly. If
you are re-recording a track over and
over, you will notice that the file size
goes up rapidly as the AW stores each
take ready for undo requests. The Taking dead space out of your audio tracks
has the most immediate advantage that it
Optimise function does away with all makes it much easier to navigate through
undo layers in one go, although it your recordings — the solid black bars on
retains virtual tracks and all mix data, the left give no visual cues to your
so it certainly pays to check that a arrangement, whereas the edited tracks on
series of edits are successful before the right make the song layout obvious at a
glance. The other advantage of such topping
optimising. and tailing is that you can then use the
Optimise routine to reclaim disk space and
Song sizes can also be kept to a also to reduce the size of the project prior to
backing up.
minimum by trimming each track using
the digital editing options. A 24-bit
recording of silence uses just as much memory as a 24-bit full orchestral
recording, so erasing areas of recorded inactivity is advisable. If several seconds
are taken from both ends of all 16 tracks and their virtual tracks, it's possible to
claw back several minutes of 24-bit recording, thus reducing file size by hundreds
of megabytes. Remember, though, that the file-saving benefits of editing will only
be noticeable after optimisation.

Curiously, erasing audio from within a track, rather than from the start and end,
does not seem to reduce file size, and for some reason the activity often has the
effect of increasing it by several megabytes! Of course it's not totally necessary
to edit the gaps between vocal lines or guitar breaks, because you can automate
channel muting, and many people will do just that to save time.

Nevertheless, erasing silences is thoroughly worth doing because it dramatically

improves the visual landscape of the Track View page, making navigation
through a track far easier — it becomes possible to tell where you are in a Song

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just from glancing at the display.

Effective Track Naming

You can name tracks and virtual tracks using either a
connected mouse or the cursor keys and Enter button.
Although the track name can be up to 16 digits long,
it's unwise to have long track names, because the all-
important track view screen only displays the first eight
of these (including spaces). Long names are simply
truncated, so naming a track Elec Gtr Lead, and
another Elec Gtr Rhythm will, for example, read Elec
Gtr for both tracks, which is rather unhelpful! As you
can see, a naming strategy needs to be employed to
ensure that the important information is contained in
the first eight digits. I tend to abbreviate names as
much as possible, sometimes using product codes
when I know that certain sounds have been used from
a particular sound module. User settings for EQ,
dynamics, and effects can also be named using 16
digits, but as these are only ever viewed in the
relevant library lists where all 16 digits are visible, it's unnecessary to abbreviate
them.

The Value Of Defragmentation

In the AW4416's v1.3 OS update, a disk defragmentation facility was added to

combat the operational problems the AW4416 suffers when the disk becomes
fragmented. Given that there are so many ways you can edit and re-record data,
it doesn't take long before the information relating to a particular Song is
scattered far and wide across the hard drive, and this means that the recorder
has to work hard to reassemble it in real time on playback.

To get to the defragmentation option

you must press the File button and then
the F3 key to recall the Disk Utility
screen. The data wheel must then be
used to select 'Int.IDE', which is the
name of the internal drive, whereupon
you are given a choice of either Defrag
or Format options. The manual small
print advises that defragmentation
requires approximately one hour per
gigabyte of disk space, which is a little
worrying if you have a large hard disk
installed! However, I have found that even after months of use the whole process
takes something approaching a couple of hours rather than several days.
Nevertheless, if you do have an uninterruptible power supply then it might be

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worth running your recorder from it while defragmentation is in progress, as a

power cut during defragmentation could corrupt the drive.

As crashing is an occasional hazard, it pays to back up at the end of every

session. I ike to use CD-RW disks for this purpose until I am satisfied that a
recording is finished, at which point I make a slightly more permanent backup to
CD-R.

Archiving Songs

Although only one Song can be loaded at any time, you can transfer individual
tracks from one Song to another. There are many reasons why this could be a
useful procedure, one of which is Song archiving. For example, if a Song is being
developed over a long period of time it's quite possible that all the virtual tracks
could be used up, or at least a project could become confusingly cluttered with
tracks and virtual tracks in various states of development. This may also result in
a project becoming so large that it has to be saved onto several disks.

One solution is to save the Song with a new name, so that two versions of the
same thing exist. Name one as the archive and one as the working project. Go
through the current version and delete anything which is not immediately
required, freeing up as much space as possible. Work can then continue on the
project, but if a previous bit of audio is required, it can be imported into the
current project from the archived version. Both versions can, of course, be
backed up separately.

Multitrack Mastery

That's all for now, but there's still plenty more to be said about the AW4416, so
tune in again for the concluding instalment of tips next month, where I'll be
covering some methods for making creative use of the effects and signal
generator.

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Software Updates & Manual Supplements

Since it starred on the cover of
SOS's 15th-birthday Issue in
November 2000, the AW4416 has
had its software revised on a number
of occasions. The machine is no
longer in production (I am told that
this is because one of the key
components sourced from a third-
party manufacturer has been
discontinued), so it is unlikely that
the current version 2.12 OS will now
be superseded, even though there
are users who still continue to write
computer utilities for the product.
If you have just bought a second-
hand machine, or have one which
has an old version of the software, you should be aware that there were many
minor updates that happened between versions one and two, rendering the
original manuals rather outdated in many areas. Therefore it's worth getting hold
of the manual supplements for versions 1.2/1.3 and 2.0 if you don't already have
them — these are available for download from www.aw4416.com/e/download/
download.html. The last OS update can also be obtained from the same site
(www.aw4416.com/e/download/ver_up2.html), but no new features were added
since version 2.0, beyond the ability to work with a further six CD-RW drives.

Published in SOS May 2005

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