Sound on Sound (September 2004)

In This Issue September 2004 In This Issue Click article title to open Reviews BONUS REVIEW: Line 6 Vetta II & Variax ...

282 downloads 5091 Views 5MB Size Report

This content was uploaded by our users and we assume good faith they have the permission to share this book. If you own the copyright to this book and it is wrongfully on our website, we offer a simple DMCA procedure to remove your content from our site. Start by pressing the button below!
Report copyright / DMCA form

DOWNLOAD PDF

In This Issue

September 2004 In This Issue Click article title to open Reviews

BONUS REVIEW: Line 6 Vetta II & Variax Electric 700 Modelling Guitar Amp & Guitar This is a BONUS online article not available in the printed magazine The Vetta II amp range embodies the most sophisticated implementation of Line 6's guitar amp modelling technology to date. It delivers more amp models than the original Vetta, offers more user control of the sounds, and it also features an optional digital interface for use with the Variax Digital Modelling Guitar. Second opinion by former Be Bop Deluxe guitarist Bill Nelson.

Ableton Live 4 Loop-based Sequencer (Mac/PC) As a creative, stable and above all fun way of combining audio loops in real time, Ableton's Live has proved an ideal counterpart to Rewire-compatible applications such as Cubase. With version 4 offering a wealth of new features, including MIDI sequencing, could Live now replace traditional sequencers altogether?

People

Business End Readers' Tracks evaluated Hear the tracks while you read constructive comments from MPG (Music Producers Guild) members who assess the latest batch of SOS readers' submitted recordings.

Crosstalk Readers Writes Replies to more of your emails, letters and faxes.

Dweezil Zappa: 64-bit Computing & The Frank Zappa Archive DAWs Of Future Past Frank Zappa produced a vast amount of recorded material during his career, often pushing both his musicians and the technology of the day to extremes. The task of editing, mixing and archiving these recordings is being undertaken by his son Dweezil, with the help of some cutting-edge computing hardware.

Jan Hammer

The Synthesizer Performer As a member of the Mahavishnu Orchestra, a composer for Analogue Systems RS300 film and TV and a performer in his own right, Jan Hammer helped to define the role of the synthesizer player in rock and CV-to-MIDI Converter No, we haven't got that the wrong way around... it really is jazz. a CV-to-MIDI converter. But why would you want such a Jim Sutherland: Producing Aberfeldy thing? Read on... Mono Forever Behringer B2030A Truth Recent years have seen a revival in back-to-basics recording techniques, but few engineers or producers have taken things Active Monitors as far as Jim Sutherland did with Edinburgh folk-pop band The new baby in Behringer's Truth monitor range offers Aberfeldy... great value for money and sleek good looks, but how good do they sound?

Carillon Core 4 Music PC Computer specifications have changed radically in the three years since Carillon launched their first music PCs, but the company's signature metal rackmount case and attention to detail still sets them apart. Their powerful Core 4 machine offers support for three monitors, plus a huge 400GB RAID 0 hard drive array for audio data.

Cycling '74 Mode

Let The Public Decide...

Leader Editor In Chief Paul White evaluates how best to define your music, and whether it is of commercial merit or not.

Sounding Off: Online Music Daniel James Online music: are we missing an opportunity to make audio distribution a better deal for musicians?

Studio SOS TV Composer David Lowe

file:///H|/SOS%2004-09/In%20This%20Issue.htm (1 of 4)9/26/2005 12:03:37 AM

In This Issue

Plug-in Suite (Mac OS X) If you are looking for effects and instruments to give your mixes a sideways slant, Cycling '74 might have just the thing...

Edirol FA101 Firewire Audio & MIDI Interface (PC/Mac) Edirol's first foray into Firewire audio is a compact 10-in, 10-out device with microphone preamps, S/PDIF and MIDI.

Edirol PCR1 USB MIDI Controller Keyboard & Audio Interface There has been fierce competition over the past couple of years to create the ultimate portable MIDI controller for use with laptops. Edirol's PCR1 is one of the lightest and thinnest yet seen. Is it what we have all been waiting for?

Emu 0404 PC Soundcard Emu have already taken the market by storm with their range of affordable, high-performance soundcards, and their new stereo PCI card offers impressive features and audio specifications at a budget price.

Hot new Sample CDs on test Sample Shop Reviews/appraisals of the latest sample CDs: PCP-80 Electric Grand ***** MULTI-FORMAT; Solo Strings ***** GIGASTUDIO+EXS24 MKII; Tool Shed Percussion ***** AUDIO+ACID/REX; Woodwind Collection Expanded Edition **** GIGASTUDIO

Line 6 Variax Acoustic 700 Modelling Guitar Extending the modelling approach they pioneered with the original Variax electric, Line 6 have now taken a shot at reproducing the sounds of different miked acoustic guitars. Almost completely immune to feedback, the Variax Acoustic will clearly be a hit in some live applications, but does it sound good enough to record?

This month, we help television composer Dave Lowe transform a cavernous-sounding spare bedroom into a usable home studio.

Technique

CLASSIC TRACKS: 'Anarchy In The UK' Artist: The Sex Pistols; Producer: Chris Thomas; Engineer: Bill Price When punk rock broke in 1976, the Sex Pistols caused panic in establishment Britain — and more than a few raised eyebrows in Wessex Studios, where Chris Thomas and Bill Price recorded the band's milestone EMI debut album.

Cubase SX: Using MIDI Device Manager Cubase Notes Cubase SX's MIDI Device Manager provides a way of choosing patches and banks on your external MIDI hardware by name rather than by number. We look at how to create your own MIDI Patch Name Scripts to make this possible.

Demo Doctor Analysis of Readers Recordings Think your own music is good? Listen to these tracks from SOS readers and see if you agree with the good Doctor's prognosis...

Instant Track Creation Reason Notes We explore an advanced technique for creating "instant" tracks using Propellerhead's Reason software.

Layering Sounds In Logic's Virtual Instruments Logic Notes Layering hardware sound modules to create new sounds is a simple matter, but things are not nearly as easy when you are working with virtual instruments in Logic. Here are a few useful workarounds.

Live Drum Triggering

Mic Preamp & A-D Converter This stylish new eight-channel preamp and converter offers front-panel instrument inputs and M&S stereo decoding.

One Drummer's Search For Perfect On-stage Sample Triggering When the epic, studio-created drum sounds of the 1980s had to be translated to a stage setting, sample triggering from drum kits came to the fore — and our man Gavin Harrison began his personal quest for the ultimate setup...

Native Instruments Guitar Rig

Music PCs: Manufacturers' Round Table

(Mac/PC) Guitar Amp Modelling Software & Foot Controller The market for software that emulates guitar amplifiers is pretty crowded these days, but Native Instruments have found a new angle with a modular program that comes with a dedicated hardware controller and interface.

PC Musician Buying a music computer from a specialist builder is an increasingly popular option for the PC musician. We rounded up some of the leading names in music PCs to find out exactly what they offer.

M Audio Octane

New Intel Chipsets: Good For Musicians?


In This Issue

Novation X-Station 25 Controller Keyboard, Audio Interface & Synth After releasing their Remote 25 MIDI controller keyboard, Novation released the Remote Audio 25, rapidly following it with the Remote Audio 25 Xtreme. Now, the X-Station has replaced both of these. Can we disentangle it from its convoluted beginnings?

Pearl CC22 Condenser Microphone This innovative rectangular-capsule design aims to combine the desirable attributes of small- and largediaphragm sounds.

Real Traps Mini Trap & Micro Trap Acoustic Panels These acoustic panels not only offer sonic improvements over simple acoustic-foam treatments, but are also very easy to install.

Roland Fantom X

PC Notes Intel's latest chipsets are finding their way into new PCs — but do they offer any advantages to the computer musician?

OS X Tiger, Cinema Displays and new iPods Apple Notes New Cinema displays, a new operating system upgrade, and a new iPod revision — all in this month's Apple Notes column.

Recording Violin & Piano Pro gear vs. Project Studio gear Recording chamber music in Birmingham's Symphony Hall is not something you get to do every day, so SOS decided to make the most of the occasion by using two completely different setups, one based on professional equipment (manned by Hugh Robjohns) and one on project studio gear (operated by Paul White). The results were interesting...

Stereo Mixdown of Projects Digital Performer Notes This month we're feeling bouncy, as we explain how to use DP's facilities to mix down Projects into stereo files for CD burning, MP3 conversion or mastering processes.

Synth Workstation Roland's original Fantom workstation wasn't really much competition for Korg's Triton, so it was no shock to see it Using SampleCell libraries with Pro Tools revamped last year as the hugely improved Fantom S. But Pro Tools Notes surprisingly, Roland have done it again. What's new this Digidesign's SampleCell products once had a huge army of time around? users, many of whom still have well-loved sample libraries stranded in this format. We look at the options available to Soundfield SPS422B & SP451 continue using these sample patches in Pro Tools. Surround Recording System This system can record in mono, stereo, or any current or Using Storm with Sonar future surround format — using just one mic. Sonar Notes We bring you yet more Sonar stratagems, including further TL Audio Ivory 2 5060 adventures in envelope copying, some handy track tricks, and Valve Compressor This new Ivory-series preamp and compressor combines advice on using Arturia's Storm with Sonar. the ease of presets with the flexibility of manual operation. Music Business

Ultimate Sound Bank Charlie

Getting Indie Music Online

Sampled Tonewheel Organ Plug-in (Mac/PC) Sampled tonewheel organ sounds have been around for many years, but more recently we have seen the arrival of modelled virtual tonewheel organs. Can a sample-based Hammond emulation still cut it?

Paul Sanders of Consolidated Independent In the age of iTunes, if you want to sell your music, you have to be able to get it on-line. But to do this, you need to understand a ton of Internet technology. Does this man have the answer?

URS N, A & S Series

Presenting Your Demos

Classic Console EQ Plug-ins (Pro Tools) For all those Pro Tools users who pine for the sound of analogue EQ, URS have recreated the equalisers from three classic mixing desks.

Getting Your Music Heard Find out what you need to do — and avoid doing — if you want your demo to stand out from the crowd.

Voxengo Pristine Space Multi-channel Convolution VST Reverb Plug-in (PC) Surround-capable reverbs don't usually come with a price tag of £100. However, nobody seems to have told Voxengo. So just how good is their Pristine Space PC file:///H|/SOS%2004-09/In%20This%20Issue.htm (3 of 4)9/26/2005 12:03:37 AM

In This Issue

VST plug-in?

Competition

WIN Tannoy Reveal 5.1 Surround Monitoring system Sound Advice

Q. In what order should I compress and EQ? Q. Can I remove the vocals from a track using phase? Q. Do I really need to buy a channel strip? Q. How can I stop my guitar buzzing? Q. How do I transfer SysEx files to my Korg Wavestation synth? Q. What can I use to trigger drums? Q. What is the whining noise coming from my soundcard? Q. Where can I get a hearing test? Q. Why don't my mixes sound big in mono?


BONUS REVIEW: Line 6 Vetta II & Variax Electric 700

In this article:


This Modelling Stuff Two Heads Better Modelling Guitar Amp & Guitar Published in SOS September 2004 Than One? Vetta II Operation Print article : Close window Amp Models Reviews : Modelling Guitar Do You Want II? Variax 700 Modelling Guitar The Vetta II amp range embodies the most sophisticated Opinion II: Bill Nelson on Vetta and implementation of Line 6's guitar amp modelling Variax technology to date. It delivers more amp models than the

Line 6 Vetta II Amp pros Vetta is the best and most flexible Line 6 amp so far. Excellent range of effects and stomp box models. User-friendly controls and displays. Variax guitar can be directly connected using digital I/O.

original Vetta, offers more user control of the sounds, and it also features an optional digital interface for use with the Variax Digital Modelling Guitar. Paul White

While the Pod and Pod XT are undoubtedly the high sales volume success story for Line 6, the Vetta II amplifier range embodies the most cons sophisticated May still appear implementation too complicated for some guitar players. of their guitar amplifier Fairly costly. modelling summary technology to The Vetta II date, some of represents another which has step forward in guitar amp modelling already filtered down to the Pod XT. Fortunately for existing Vetta owners, the software is upgradable (free via the Line 6 web site) so they don't get left out in the cold. and as well as sounding great, it Essentially, Vetta II delivers more amp models than the original Vetta, it offers more user also feels control over the sound, and it also features an optional digital interface for use with the responsive to play. If Line 6 Variax Digital Modelling Guitar or for connection to external digital devices. This is you want one great amp sound, buy one very neat if you're a Variax owner as connection is every bit as 'plug and play' as great amp, but if you plugging in a conventional guitar lead. need to have instant access to lots of

Though it looks much like a traditional 2x12 guitar combo from the front (around the size

file:///H|/SOS%2004-09/BONUS%20REVIEW%20%20Line...%20Vetta%20II%20&%20Variax%20Electric%20700.htm (1 of 9)9/26/2005 12:03:51 AM


sounds (both live and in the studio) or to create a unique trademark sound of your own, the Vetta II seems like the way to go.

information Line 6 Europe +44 (0)1788 821600. +44 (0)1788 821601. Click here to email www.line6.com

and shape of a Vox AC30), the Vetta's control panel leaves you in no doubt that Kansas has been left far behind. However, the familiar Drive, Volume and EQ knobs (Bass, Middle, Treble and Presence) of a conventional amplifier remain, so it's not all uncharted territory, and as with the Pod range of products, the controls have a logical simplicity to them. There's a rotary control for direct selection of the amplifier model plus four further Global knobs that are designed to allow instant tweaking during performance to provide global control over reverb level, low and high EQ, plus overall volume. These global controls operate in addition to the stored parameters, so if you find yourself in a room where the amp sounds a bit dull, simply turn up the High global EQ knob and all your patches will be affected by the same degree. This type of global function was part of the feature-set of their very first Axsys modelling amp and turned out to be very practical. The control panel features two displays: the lefthand display shows the patch name and location and has an associated rotary control that lets you scroll through the 64 factory patches and 64 user patches; the righthand main display shows the current values and positions of the controls, and may be edited using the four encoder knobs directly beneath the displayed virtual knobs. A fifth encoder scrolls through the pages whenever more than one page of edit parameters is available. There are two rows of 'soft' knobs relating to the two amplifier models, and when you move onto editing an effect, the knobs again relate to whatever parameter is displayed above as a virtual knob. Vetta is designed so that you can choose (in the system setup) how it should respond when the amp knobs are turned. In Absolute mode, the parameter changes to match the physical knob position as soon as you move the knob. In Relative mode, however, moving the control simply adds more or less to the current value. This means the stored value of the knob and its physical position probably won't agree, but the on-screen knobs always show the actual value.

In addition to the usual array of effects and the amp EQ (which changes its character to suit the amp model selected), there's a further EQ section that can be switched to either graphic or parametric modes offering four bands of control per amplifier in either case. Each stomp box and main effect type has its own access button and pressing any of these followed by Edit gets you quickly into the appropriate editing page without having to scroll through menus. There's also a button here to turn the effect loop on/off.



The Vetta's rear panel houses the IEC mains inlet and switch, two external speaker jacks and an impedance switch for use with external speaker cabinets where you might want a wider stereo image. There's also a stereo effects loop (on two tip-ring-sleeve jacks), just in case you want to add external effects to the compendium of onboard modelled effects, plus both balanced (XLR) and unbalanced stereo (jack) Direct outputs. Digital I/O comes as standard on the Vetta II providing both coaxial S/PDIF and AES/EBU in and out, plus a CAT5 Ethernet-style socket for the direct connection of Line 6's own Variax modelling guitar (see review below). Of course a Variax can be plugged into the analogue inputs instead, but using the digital link cuts out an unnecessary stage of D-A and A-D conversion. Another very neat feature of the digital Variax link is that you can store your Variax setting as part of the Vetta preset, so that when you dial in a new preset, the guitar automatically switches to the appropriate model and pickup selector settings. And talking of neat features, you can even configure the digital I/O to work as a digital effects send/return loop if you own an effects unit equipped with digital I/O. Completing the connections rundown are MIDI In and Out sockets and a connector for an FBV foot controller. Note: the Vetta II isn't compatible with the Floorboard or FB4 range of controllers.

This Modelling Stuff The modelling concept will be familiar to anyone who's used a Pod or one of its rivals before &Mdash; the signal path comprises software models of popular guitar stomp pedals, amp modelling to recreate a number of 'classic' amp tones, speaker cabinet modelling (with mic type and position simulation), delay, modulation and reverb effects, and also tremolo, gating and compression. Rather than confuse the user with multifunctionality, all the control panel buttons have a dedicated function, clearly marked on the button cap. A quick glance at the control panel reveals that up to three different modelled stomp effects can be used at once — and that's in addition to the four main effects blocks plus compression, tremolo and gate. Nice! Further dedicated buttons allow tempo 'Tap' input for delay times or modulation rates, Amp/Cab selection, Effect Routing and a neat Double Tracking effect that's not available on Line 6 products further down the food chain. There are also extra buttons for setting up the main system parameters (display contrast, MIDI channel, etc) and one for setting the output configuration. Vetta includes Line 6's AIR modelling, which seems to create file:///H|/SOS%2004-09/BONUS%20REVIEW%20%20Line...%20Vetta%20II%20&%20Variax%20Electric%20700.htm (3 of 9)9/26/2005 12:03:51 AM


early reflections to give a room ambience effect when direct recording, and also models specific microphones and their position relative to the speaker. The room AIR effect can be switched on or off for the direct outputs as well as adjusted in intensity. There's also a digital output, a stereo width enhancer, control over the direct output levels and the option to turn the internal speakers on/off. There's also a headphones jack on the top panel when quiet practice is required, though this amp is far too much fun to use quietly!

Two Heads Better Than One? Where the Vetta differs from earlier Line 6 amplifiers, other than in its more sophisticated modelling, is that it can generate two different amp models at the same time. You can then opt to use just one of these models or both at once, each panned to its own position in the stereo mix. The stereo mix then feeds the two internal speakers via solid-state power amplifiers. The Vetta II Combo provides 75 watts per channel (150 Watts total), while the Vetta II HD head produces double that (150W per channel, 300W total, into selectable 4, 8, or 16 Ohm loads). It's also possible to take a speaker simulated DI for recording or for feeding to a full-range PA system. Four buttons close to the input jack allow the individual amp models to be turned on/off and also toggle the display between the upper and lower rows of knobs on screen, so that both amps can be adjusted using the same four rotary controls. When both knobs buttons are lit, the amp controls adjust both models simultaneously. For the more computer-centric user, there's a free software editor available from Line 6 (www.customtone.com) with a graphical interface that shows all of your routing and effects. It neatly lets you store banks of sounds on your computer and download from an online library of thousands of Line 6 sounds (Vetta sounds, as well as Pod, Pod XT, Flextone III, GuitarPort, etc). The latter are all upwardly compatible with Vetta II and the editor (which also works with Pod XT).

Vetta II Operation The most basic requirement is to select a preset, which in the case of Vetta II stores the amp and speaker model along with all its effects and settings. Once the Factory or User buttons have been pressed, these may be selected directly using the rotary knob adjacent to the patch name window, but it's also possible to use buttons A through D to select any patch in the currently active bank (patches are arranged as 16 banks of four). Additionally, the handy Favourites button recalls whatever patch you decide is your favourite. Patches normally load as soon as they are selected, but there is a user option that lets you 'defer' loading so that you can preset an effect, then load it when you want to. FBV owners can also select patches using their feet by stepping through the banks and then selecting patch A through D in the selected bank. Anyone into MIDI control from a sequencer, say, can also call up patches directly using MIDI Program Change commands. file:///H|/SOS%2004-09/BONUS%20REVIEW%20%20Line...%20Vetta%20II%20&%20Variax%20Electric%20700.htm (4 of 9)9/26/2005 12:03:51 AM


General operation is much like a Pod XT but with more parameters and choices. All the buttons have an orange backlight to show when they are active and everything is labelled in large, musician-friendly legending. The individual stomp effects can be turned on or off using their respective buttons, as can the mainstream effects, tremolo, gate and compressor. Hitting the Tuner button brings up a large and extremely accurate tuner display in the main window, and the tuning reference can be changed from the default 440Hz if required. Many of the effects will already be familiar to Pod and Pod XT users, but many new ones have been added, including a new simulated double-tracking feature that doubles the guitar sound and widens the stereo image &Mdash; though it helps to have an extension cab to get the most out of the stereo aspect of this effect. The connection order of the modelled stomp effects can be changed and the graphical display icons make this a simple and very visual process. Similarly, the main effects can be configured in series, parallel or series/parallel configurations, again with a nice graphic display of what's connected where.

Amp Models As Line 6 point out, their manual acknowledges particular amplifiers and effect devices that inspired their models, but there's no suggestion of endorsement by these manufacturers. and all trademarks are acknowledged as property of their registered owners. In all there are 74 different amp models, so I don't plan to list them all by name, but they include 29 Line 6 specials based on what their engineers consider to be the best characteristics of existing hardware amps plus 45 models based on specific amplifiers. All the big UK and US names are represented, but there are also numerous boutique models inspired by the sounds of Bogner, Budda, Cornford, Diezel and ENGL amps as well as some based on lesserknown vintage models.

Other notable additions are the Pod Purple X sound effect generator (Star Wars Pod racer anyone?), Each amp model boots up with Random Sample and Hold filtering, a chewed tape a suitable cabinet, but there are simulator called Tape Eater and some new chorus 28 cabinet models to choose variations, including one that features three different from and any cab can be used with any amp with no fear of modulation sections running at random rates. frying the cone! So, if you fancy There's also a Synth Filter effect that seems to have hearing what a Marshall stack taken its inspiration from the Line 6 Filter rack might sound like played flat-out processor, where single notes are reshaped into through a Supro practice amp harmonically rich waveforms and then filtered 'synth speaker, no problem. Again all style'. This section includes Synth Lead, Synth the big names in speakers are Strong, Synth Analogue, Synth FX, Buzz Wave, Rez there, in sizes from six inch to Synth, Saturn 5 Ring Mod, Double Bass, Synth 4x12s and 1x15s, and some of Harmony, Auto Wah, Dingo Tron and Sparkle the vintage British cabs even Sweep variants, all of which sound suitably have a choice of speaker colour electronic. However, I found that even when and power rating that relate to specific models or periods of carefully playing monophonic lines, the synthesized manufacture. sound pitch occasionally yodelled uncontrollably, so it's definitely a case of picking what works rather than expecting the synth sounds to follow everything that you play. What I wasn't expecting was how much better some of the sounds were when combining



amp models. This is particularly evident when you want a more or less clean sound that's warm but also slightly jangly and with a decent amount of sustain. The patch 'Another Brick' shows this off to advantage and is a million times more Floyd-like than the similarly named factory patch on my Pod XT. The Vetta II also comes close to the warmth and edge of the Dire Straits guitar sound, should you be into that, but it also captures all the classic rock sounds with better than expected authenticity. Some of the new effects are great, others are interesting in the 'kippers and custard' sense, especially Tape Eater and Bubble Echo, but most have their uses, even the rather odd Pod Purple XT. The sound from this amp is certainly large and 'produced' sounding, but I also felt the Vetta II was more responsive to play than some of the earlier Line 6 amplifiers I've tried. I'm not sure if die-hard tube amp players will be completely won over by it, but it gets very close and I appreciate being able to dial up great sounds at any volume level.

Do You Want II? The Vetta II isn't a low-cost amp, and you really need to budget for the FBV floor controller to get the most out of it in live performance, but it delivers on sound and is very loud. It also makes a great companion for the Line 6 Variax, as it enables all the guitar and amp settings to be saved in a single Vetta preset. Only the pitch shifter disappointed me &Mdash; I've yet to hear one that really works well, though every manufacturer insists on putting one into their effects boxes. Perhaps one day someone will come up with a new, real-time resynthesis method that does away with 'lumpy' looping? The synth emulations turned out to be rather hit-and-miss, but played with care, they are capable of some great sounds that should turn a few heads, so consider them a bonus rather than a limitation. So who needs a Vetta II amp? Personally I can get by quite happily in the studio using my Pod XT, but if you play live as well as record, and need to change sounds between songs with the minimum of fuss, the Vetta II is a great tool and is much more flexible than the Pod XT both in the number of amp models available and the number of effects types on offer. It also has that wonderful dual amp facility, which sounds a lot more impressive in the flesh than it does on paper. The other feature that should appeal to guitar players is that from the front view, the Vetta looks very much like a traditional guitar combo.

Variax 700 Modelling Guitar We've reviewed the Variax (500 model) before so 'theres little need to go into detail about what it does other than to say its onboard electric wizardry can make it sound like a whole collection of classic electric, acoustic, 12-string, banjo, resonator and sitar models. Furthermore, it's easy to use Ñ you simply dial up the guitar type using one knob, then use the volume, tone and pickup select controls as normal.



Since our initial review, the company have launched their more upmarket 700 series guitar that features exactly the same electronics and modelling, but built into a rather more sophisticated, Japanese-made guitar. The reason for the 'mystery' socket next to the jack is now revealed, as that's where the CAT5 cable plugs in if you want to connect the guitar digitally to a Vetta II amp. The 700 series comes with the option of a fixed 'hardtail' bridge or a custom Baggs two-point adjustable floating tremolo not dissimilar in principle to those made by Wilkinson or the stock hardware fitted to the new Fender Strat Deluxe design. It has individual, adjustable saddles (both height and intonation) and an adjustable collet to help prevent the tremolo arm from flopping around. Overall, this seems to be a very nicely engineered little trem system that builds on traditional principles rather than seeking to replace them. The review model has a bolt-on 22 fret maple neck with a rosewood fingerboard and medium/narrow frets little wider than those fitted to most Fender guitars. The scale length is a fairly standard 25.5 inches while the rosewood fingerboard has a 10inch radius that should make it feel comfortable to both the Fender and Gibson school of players. The 'three a side' headstock is fitted with Gotoh tuners and gives the neck a sort of 'Gibson meets PRS' look. The frets and fingerboard edges are beautifully finished while the solid mahogany body on this model is faced with a shaped and contoured slab of ash, coated in a flawless honey-coloured lacquer (Line 6 offer a range of finishes/ colours) that looks all the more impressive because the guitar has no body hardware other than the controls and the tremolo bridge. A bone nut and 'pearl' inlays complete the picture. Just about everything about this guitar oozes quality, which makes it all the more unfortunate that this particular model had been delivered for review without being set up, making it almost untunable and awkward to play! The main problem was that the nut slots weren't cut nearly deeply enough, so as well as this making the action higher than necessary, it also meant that fretted notes were slightly sharp compared to unfretted notes. Any guitar technician could sort this out in a matter of minutes (and a good guitar store should set up new guitars of this class as a matter of course, one would hope), but on a guitar that's so good in all other respects, receiving it in this state was very disappointing. In fact, given that guitars with tremolo systems can be hard to keep in tune at the best of times, I feel Line 6 ought to consider fitting a graphite or teflon impregnated nut and locking machine heads as standard. They would certainly be the first things I'd change if I bought one of these instruments. When notified, Line 6 were disturbed and apologetic when informed that the instrument we received was not set up properly, and assured us that all instruments are carefully set up and re-checked in the UK prior to delivery to the dealer. Despite this gripe, this guitar is such a step up from the basic 500 file:///H|/SOS%2004-09/BONUS%20REVIEW%20%20Line...%20Vetta%20II%20&%20Variax%20Electric%20700.htm (7 of 9)9/26/2005 12:03:51 AM


model that I feel it well worth paying a bit extra for — just insist on a setup before parting with your cheque!

Opinion II: Bill Nelson on Vetta and Variax "It hasn't taken me long to decide that the Vetta II amp is a beauty. As soon as I tweaked a couple of things and happened across a particular tone that inspired me to romp through 'Maid In Heaven' (and I haven't done that for a long time), I knew right there and then that it was more than capable of delivering the goods. There are absolutely hundreds of possibilities for building wildly different sounds and I'm sure I will be able to come up with a whole set of useful ones for the band's concerts. It's loud too! I won't need to worry about hearing my guitar in the monitors or through the on-stage drum thunder... enough power to knock down walls! It really is impressive. It is a digital amp though and there's inevitably something of that in the nature of the sounds but this is perfectly acceptable to my ears and not at all 'unmusical' (a common criticism from the more luddite guitarists out there when they are confronted with digital equipment). The point is that it sounds great, regardless of modelling accuracy or whether it's digital or analogue or steam-driven. It is its own thing and within that identity has great flexibility. I'm tremendously impressed. "The Variax is, for me at least, less successful. I initially tried the Vetta out using the Variax and went 'Hmm...' The amp immediately sounded more convincing when I put one of my own guitars through it. Not an expensive instrument either but a Chinese manufactured 'Switch' guitar, designed by UK designer Trevor Wilkinson, made from a plastic called Vibracell. The guitar cost less than £200 but is amazing value for money. This guitar sounded tremendous through the Vetta, much sweeter, less 'brittle' than the Variax. "I think the Variax would benefit from a more expensive, up-market approach to perfect it. Either that or Line 6 should talk to Trevor Wilkinson and make a 'Switch-Line 6' collaboration/hybrid. I think the Variax could feel more playable and look a lot more attractive. A bound fretboard, a smoother, 'classier' playing feel plus a touch of glamour and identity would help its appeal. After all, there's a romance that happens between players and their instruments and the Variax should be seductive, individual and yes,



'sexier'. I can sort of understand the approach Line 6 have taken, kind of minimal, no scratchplate, no pickups, no frills, etc, but the guitar, particularly in the natural/amber finish, looks dangerously like a school woodwork project, and feels a little like one too. I'm sure this isn't what Line 6 intended. The modelling technology, again, whilst interesting and very clever, is sort of defeated if the guitar itself lacks the magic that makes you want to play it and be seen with it. Good ideas don't have to be presented in dull packages to overcome perceived prejudices about new technology. A little more flair in terms of styling and a greater attention to what makes a good guitar feel good would make all the difference. "I did like the inclusion of the Coral Sitar sound. Not something you'd use often but good to have on tap. The Gretsch simulations were good too, as were the Rickenbacker 'Chime' models, particularly the 12-string. I realise that guitarists' likes and dislikes are purely subjective and it may be that many players would find my comments quite different from their own. There are thousands of players out there with Stratocasters, for instance, yet I've never been particularly comfortable playing one, as much as Hank and Jimi inspired different periods of my own musical development. Stratocasters are obviously not 'bad' guitars, it's just one of those inexplicable, personal things. "So, the Variax guitar isn't quite for me, just yet. I do hope that they continue to develop it though and would be more than happy to provide input and feedback if it should be of any help. I think it's important that this concept succeeds, as it's a brilliant idea and really should be taken to the limit." Bill Nelson will be using the Vetta II amp extensively on his October 2004 'Be Bop and Beyond' UK tour. More info: www.billnelson.co.uk Published in SOS September 2004 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


Ableton Live 4

In this article:

The Story So Far Live On Stage Getting Installed Two Views What's In The Box? The Session View New Instruments The Arrange View The Clip View The Mixer MIDI Clips Warping Time Keeping It Live Causing A Scene Final Thoughts

Ableton Live 4 £299

Ableton Live 4 Loop-based Sequencer (Mac/PC) Published in SOS September 2004 Print article : Close window

Reviews : Software

As a creative, stable and above all fun way of combining audio loops in real time, Ableton's Live has proved an ideal counterpart to Rewire-compatible applications such as Cubase. With version 4 offering a wealth of new features including MIDI sequencing, could it now replace traditional sequencers altogether?

pros An incredibly fast way of throwing together audio from diverse and different sources. Innovative, performancebased approach to writing and arranging loop-based music. Very stable and reliable, good for on-stage use. Well designed, intuitive, easy-to-understand interface.

Ingo Vauk

In their company profile, Ableton cofounder Gerhard Behles describes the idea of Live as software that provides musicians with a 'studio as instrument' — a concept that many of us from the era of dance music are very familiar with. The way they have gone about cons implementing this concept in fact Solo function would be incorporates a lot of the methods '90s improved by true solo-in-place dance music has developed since the and PFL modes. With the introduction of MIDI arrival of the S900 and the Atari Live's Arrange view. computer. Live is essentially a hard and the two proprietary samplers, a sample editor disk-based audio player and recorder would be useful. that takes advantage of today's fast processor speeds and disk access times, Limited MIDI editing. and combines them with some blistering time-stretch ('warping') algorithms to deliver what can only be described as an astonishingly fast, innovative and summary Live 4 is an intuitive audio and intuitive tool to compose, arrange and play music. It allows musicians to combine MIDI sequencing application almost any audio, be it on their hard disk or recorded from scratch, with anything that opens up a different and else, virtually 'on the fly'. You can audition material from hard disk alongside exciting new dimension to running program sequences, drop in live overdubs that can be triggered back off hard disk recording. disk in an instance (remember 'Frippertronics'?), change tempo while all this is information going on, and a lot more. All of this can be done mixing most common file £299 including VAT; formats, sample rates and bit depths for maximum flexibility and speed.

file:///H|/SOS%2004-09/Ableton%20Live%A04.htm (1 of 13)9/26/2005 12:03:55 AM

Ableton Live 4

upgrades from all older versions £99 (boxed) or 119 Euros (download). M Audio +44 (0)1442 416590. +44 (0)1442 246832. Click here to email www.maudio.co.uk www.ableton.com

Test Spec Ableton Live 4. Apple G4 tower 933MHz with 1.2GB RAM running OS 9.2.2, with Digidesign Digi 001 interface. Apple iBook 800MHz with 640MB of RAM running Mac OS 10.3.3, with Emagic EMI 2| 6 interface.

The Story So Far Much of the basic functionality was present in version 1 of Live, but Ableton have extensively developed its automation, audio processing and interface facilities through releases 2 and 3. Version 3 saw the introduction of envelopes for manipulating audio data, with the facility to copy and paste envelopes between parameters speeding up the process of creating more complex automated tasks considerably. Building on their intuitive basic interface, Ableton have added functions such as keyboard and MIDI triggers for audio clips as well as scenebased automation. Facilities such as the legato mode allowed users to superimpose melody on to audio via MIDI keyboards, or create unique grooves by switching between loops part-way through. This intuitive interface convinced a lot of people in the performing arts to start using the software, with modern dance and theatre companies now being able to perform live to music that was being created 'on the fly' while providing a reliable time base (if desired). Mixing and audio processing, meanwhile, was enhanced by introducing more proprietary plug-ins such as compressors, more sophisticated DJ-style EQs and creative effects like the resonator, which generates harmonics around a chosen fundamental note and is useful for both percussive and atmospheric sounds. VST handling is possible from multiple editor windows and the managing of sounds and banks is comfortable, with a logical filing system. From the basis of the solid 'audio sequencing instrument' that was version 3, Ableton have launched Live 4, which is now described as 'real-time music production' software for Mac and PC. Version 4 introduces MIDI sequencing for the first time, along with Ableton's own plug-in instruments, the appropriately named Simpler and Impulse sample players. Apart from the MIDI sequence editing facilities (which I will look at in more detail later), MIDI clips are handled in much the same way as audio, being stored as MIDI files that can be dragged on to the control surface into clip slots for instant combination with any other material running in the application. VST and Audio Unit instruments are supported, so that the user has access to his/her existing library of sounds in these formats. External MIDI I/O is also supported via the usual interfaces.


Ableton Live 4

Live On Stage I have used Live live on several occasions recently, and its simplicity, reliability (the only real crashes I had throughout the reviewing period were Rewire-related, in complex situations with Logic running as the host under heavy CPU load) and overall ease of operation make it the ideal tool on stage. The fact that scenes can now store tempo information opens up a whole new application for the software: triggering loops and any additional backing tracks for bands, since each song can be stored as a scene, providing multiple outputs for headphone feeds and so on. For this I would recommend it to anyone who needs a light and robust tool for showtime. The beauty of using Live in this fashion is that you can now store the tempo in the scene title, which means it can be changed on a nightly basis as the set develops. Nothing is more annoying than having to re-record the backline sequencing on tour because it needs to be sped up by a couple of bpm, and this is precisely what you have to do if you are using traditional hard disk systems or tape-based media.

Getting Installed Ableton specify their minimum requirements as any G3 Mac or faster with at least 256MB of RAM, running OS 9.2 or later or Mac OS X 10.1.5 or later; for Windows, the software requires a 400MHz CPU or faster with 128MB RAM, running Windows 98, 2000 or XP. As usual, PC users will also need a Windowscompatible soundcard, preferably with Direct X or ASIO drivers. With a program such as Live there's limited point in running bench tests of the 'how many plug-ins can the software run on how many tracks while standing on its head before crashing' variety, because the goal here is creative manipulation of audio files rather than simple multitrack playback. However, Live 4 is a processorintensive application, and anyone coming from 'traditional' HD recording The Session view provides mixing facilities, systems should not expect to be able but also allows you to prepare clips for to run the same number of tracks triggering via MIDI or using the QWERTY simultaneously as on their other keyboard. At the bottom of the screen is the clip view, where parameters for individual DAWs. A closer look at what Live is audio and MIDI clips are adjusted. actually doing makes it easy to see why this should be so, since most functions in the audio domain are related to real-time time-stretching, or 'warping' as Ableton prefer to call it. And this is offered in the same form by no other application for Mac or PC I know of. Installation could not be easier: starting from a free demo download, you simply


Ableton Live 4

drag and drop the folders onto your destination hard drive, which installs all the relevant elements into their respective folders. As normal with audio applications, you have to point the software towards the audio hardware by putting the necessary drivers into the usual folders. Since Live is VST-compatible, you might also want to put a copy of your VST plug-ins folder into the Live folder. I found that this is a more reliable method than just pointing it towards an existing VST folder using an alias (in fact this caused all sorts of problems with my existing Logic setup, resulting in the corruption of the Logic preferences). Audio and MIDI defaults are set in the Preferences dialogue once the software is up and running, along with the record path, default file format and resolution. When these steps are taken you can launch Live, which will open up with a dialogue asking you for the serial number, or alternatively give you the option to run in demo mode. Demo mode does not allow you to save a set, or render to disk, something that quickly becomes very annoying because you are very likely to come up with something you'll want to save in a very short time. Once you have purchased the software, the startup dialogue responds to the serial number by issuing a challenge code (generated from the serial number and your CPU identity), which in turn will be used by Ableton to issue an unlock code. If your machine is on-line Clips in a Live arrangement can be triggered this happens automatically and takes a from QWERTY key-presses (left) or MIDI couple of minutes; if not, it can be notes (above). entered into a form on Ableton's web site from another machine. All in all the process is very quick and painless, and you are working with your new toy well within 10 minutes of opening the box. Live comes with a single-user multiplatform licence, which means that you are allowed to use your software on only one machine at a time, but you are allowed to install it on more than one machine.

Two Views The user interface has been kept simple, and in version 4 there are still only two main windows at your disposal. You can toggle between the Session and Arrange views using the Tab key, and these main display modes have in common all the other displays, such as the browser, clip views, plug-ins, and so on. The bottom left corner of both screens houses the 'info view' help text that


Ableton Live 4

explains whatever the mouse is pointed at at any one time and virtually makes the manual redundant. Most elements of the screen can be optionally switched on and off, but the default layout is a good starting point for anyone beginning to explore the application. To the left of both views is a browser which gives the user access to up to three different favourite locations (for example two different dedicated audio drives and your sample library) as well as the Ableton plug-in folder and any VST folder you might have defined in the preferences dialogue. The top bar in both pages hold the transport, locator and tempo controls, all self-explanatory, clearly labelled and along long-established conventions.

What's In The Box? Live 4 comes with a well-written manual which is easy to understand and comprehensive. While trying to stay entertaining and brief enough to hold the impatient would-be-producer's attention it does, for example, go to the trouble of explaining in detail the phenomenon of latency and its significance to recording. It offers a step-by-step procedure for setting up your 'overall latency settings' which is very useful in order to fine-tune your system when you're recording live audio and monitoring through the computer. I have found that other manuals often leave out this kind of detail altogether or get lost in technical jargon that is hard for the novice to understand. Also to be found in the box is a CD of ready-made loops from Big Fish Audio containing all sorts of goodies for fans of prefab music, sorted into categories such as Bass, Drums, Horns, Percussion, Synth and Miscellaneous, the latter consisting mainly of scratching and vocal snippets. The quality of these sounds is good, and it's very easy to build an instant track using them — a good way of demonstrating the power of the software, though anyone who builds their entire tracks on the bundled samples can probably only be classified as a sad case!

The Session View The Session view looks like a traditional mixer page at first glance, and this is indeed part of its function. However, above the usual mixer controls is a section of 'clip slots' into which audio and MIDI files can be dragged from the browser. Once these clips are in place, their playback parameters can be set in the clip view below. Each track defaults to having 20 clip slots, though any number can be added, and can play one clip at a time. These clips are, in a sense, sitting dormant, waiting to be activated through mouse clicks, MIDI or keyboard triggers. It is the fact that an audio/MIDI arrangement can virtually be played from a keyboard in this way that makes Live so different from normal hard disk recording systems. The trigger quantisation can be set individually per clip (from 32nds to once per bar in doubling increments) or from the global quantisation parameter, which resolves triplets as well and goes up to every eight bars. In the global menu we also find a 'no quantise' setting.


Ableton Live 4

An interesting new feature in version 4 is 'follow actions', which essentially allow you to predetermine what should happen in this row of clip slots once a clip has been triggered. You can set a time in beats and bars that determines how long the clip plays for. After that the clip will do one of the possible actions that can be selected in a pulldown menu: go to the next or previous clip, go to the last or first clip of the The clip view for a MIDI clip in Live 4, group, play the clip again or stop it, or showing the MIDI editor. go to any (random) clip in this group. A group in this context is a column of clip slots that are separated from the next group by an empty clip slot. There are two follow-on actions per clip, and you can set the probability of each happening compared to the other. This sounds a lot more complicated than it actually is, and playing around with these settings generates more or less subtle variations on repetitive patterns that would take ages with conventional beat-slicing methods or MIDI editing. Another use for this function would be the creation of nested loops, which can be achieved by cleverly combining follow-on actions with varying start points in separate clips, using the same piece of audio. I'd imagine this to be very useful for DJs wanting to create extended mixes of tracks.

New Instruments Live 4 offers two samplers, Impulse and Simpler. As the name suggests, Impulse is a drum/percussion sampler, which allows eight samples to be loaded at a time. The interface is simple and intuitive, using drag-and-drop to assign samples from the browser and featuring a set of controls for each sample. There is a resonant filter, plus drive and tune functions, with additional controls including sample start and decay. Velocity can be set to control volume, tuning, filter and stretch settings. Again the simplicity of the device is very much in line with the overall design philosophy behind Live, and it has to be said that this is a very effective little sample player. Live does not provide a sample editor, and although most people probably have a stand-alone application like Peak for this purpose, I did miss it at times, especially when my ASIO driver would only allow for one application to address the audio interface. Shutting down the application in order to load Peak, which then has to be shut down in order to return to Live, seems rather a palaver if all you want to do is extract 2ms worth of noise to use in a sampler. However, you will encounter the same problem with a number of programs, such


Ableton Live 4

as Reaktor, and it's not a problem in OS X. Each instance of Simpler will only play back one sample at a time, but it provides more sophisticated sound manipulation, such as an ADSR filter envelope, key tracking, sample looping and an LFO which is assigned to pitch. This is obviously designed for melodic or pad sounds, but don't throw out your copy of Reaktor just yet. Since Live supports VST, AU and Mach-O (watch this space) plug-ins there is a lot more potential in the MIDI department then meets the eye. And, it has to be said, no sequencer application gives you all the sound-generating facilities you'll ever need. There are also a number of MIDI effects, including one for rescaling incoming MIDI notes in order to make shifting the pitch easy, a chord effect which builds up to six notes around any incoming note, and a self-explanatory Random device that works on pitch. The Velocity plug-in looks and acts like a sophisticated compressor with a random element thrown in — very slick! These effects can be dragged very much like audio plug-ins into the signal chain prior to the MIDI instrument, with the audio plug-ins being inserted post-instrument.

The Arrange View The arrangement chooser looks very much like a conventional desktop hard disk recording package, but it still integrates all the same functions as the Session view. The main difference is that it is used for a more off-line approach, in the way that you might use a Pro Tools setup with copy and paste functions to assemble loops into a structure. It still allows you to trigger audio using the same sources, but it renders the results visually in the more conventional horizontal track view we all know from other packages. This mode could be used to construct a track from scratch by dragging and dropping parts from the browser onto the track slots, or to fine-tune and edit a performance that was created using triggers. It also allows us to superimpose more linear graphic envelopes that don't repeat with each trigger of a sample as they do in the Session view. Another, obvious reason to use this view is to copy and paste, cut and generally mess with an arrangement. On the subject of copy and paste, though, there were a couple of situations when I was the Arrange view and I had worked on a section in detail using envelopes in the clip view, but then wished for the possibility to paste them back into the Session view. This doesn't seem to be possible, and would be on my wish list for further updates.

The Clip View Live calls the pieces of audio or MIDI data used in a session or arrangement 'clips', and in order for clips to work in the context of a Live arrangement they have to be prepared in the clip view, which — like the effects and instrument editing pages — uses the lower section of the Session or Arrange page. Here, we decide how a piece of audio is being triggered. There are four launch modes: trigger, gate, toggle and repeat. Trigger is what is also known as 'one-shot' in the


Ableton Live 4

world of sampling, whilst gate plays a clip for as long as the key is held down, toggle uses one key-press for on and the next for off, and repeat plays the looped clip until another clip is triggered in the same track. For audio clips, the clip view shows a waveform display, and is where playback parameters such as 'warp mode' (the type of time-stretching algorithm that is most appropriate for Live's ability to time-stretch audio in real time the job in hand), level and tuning can depends on preparing clips and storing be set. A clip is more than just a bit of information about the transients they contain audio, though, since it can be prepared as 'warp markers'. for playback in the clip view in a multitude of (non-destructive) ways, setting a host of playback parameters such as pitch, trigger points, panning, levels, warp modes, warp markers (which can be thought of as quantisation anchors within a piece of audio, corresponding to slice points in an application such as Recycle) and even the grain size of the timestretching cycles over time (interesting for textural variation), with the aid of controls as well as envelopes. This way, the same piece of raw data can be made to sound completely different without the need to create a new audio file, and the variations are virtually limitless. A nice feature is that the length of the envelopes can be decoupled from the sample length, in order to have variations in the envelopes over multiple cycles of the sample. This way a one-bar audio loop can be turned in something more interesting using envelopes that repeat over a longer period. Since there is no need to render each variation to disk, Live is very efficient in terms of hard drive load. For some more extreme manipulations, and when doing a lot of crazy warping to a lot of material, it becomes necessary to switch clips to RAM mode, which stores them in RAM ready to be triggered, thus reducing disk traffic and CPU load. This is a very useful function introduced with the release of version 3.

The Mixer The mixer part of the Session view has undergone a significant revamp since version 3 and allows for more complex routing. Audio channels can get their signal from a number of sources such as the individual outputs of Impulse or any other multiple-output plug-in, other audio Mixing and routing is handled in the Session window. channels for subgrouping purposes, the master out for resampling, external sources, and so on. This routing is done via the In Out section of each track and in general works very well. One aspect of the mixer that caught me out is the solo/cue button. By default, solo cuts all other signals and sends the file:///H|/SOS%2004-09/Ableton%20Live%A04.htm (8 of 13)9/26/2005 12:03:55 AM

Ableton Live 4

selected track to the main output, which is quite restrictive, meaning that you cannot switch a number of tracks into solo mode together, and you never hear the effect returns when in solo. However, this behaviour can be changed by disabling the Solo Exclusive preference, whereupon any number of solo buttons can be activated and these signals are all mixed together. Alternatively, the Solo Exclusive preference can be overriden by using the Command (Mac) and Ctrl (PC) modifiers when soloing multiple sources. This works fine but I still feel that there is room for a couple of improvements: when multiple signals are routed to a group track both the source and the group track need to be in solo in order to hear anything, and the same applies when you want to listen to a signal with its aux effect. A real solo-in-place configuration would be more comfortable in this case. Conversely, if you want to hear only the Aux return you need to solo it with the source and then lift that from the mix buss, and listening to just a group return entails selecting solo on all its sources as well as the return channel. I feel that a true PFL mode would be the more efficient solution. The MIDI routing works along similar lines to the audio, with the inputs and outputs of each track effectively functioning like a patchbay. This I found an elegant resolution, which makes doubling up MIDI signals to two sound generators as easy as sending multiple MIDI tracks to the same sound source. It is also possible to bounce several MIDI tracks together in this way, but interestingly this is the only way of combining MIDI tracks destined for the same sound generator. There is something to be said for the good old glueing tool to do this job, since that doesn't have to be done in real time.

MIDI Clips New to version 4 of Live are MIDI clips, which are handled in a similar way to their audio counterparts, with the clip view offering some of the same parameters such as volume and pan, plus MIDI-specific parameters, some of which depend on the sound source. MIDI clips can be created by playing in MIDI from a master keyboard as with any other MIDI sequencing software, and you can draw the notes into a keyboard-matrix-style window that looks familiar, too. You can also view Standard MIDI files in Live's browser, and import individual parts from them as MIDI clips. Ableton's design philosophy of simplicity and instant gratification certainly comes into full force here, and they have improved on some aspects of traditional MIDI editing windows. I like the way the pencil tool acts as an eraser when you click on a space that's already taken — why should you have to switch tools for such an obvious task? Also very cool is the way the grid resolution and quantise/drawing resolution can be changed with simple keyboard shortcuts. The 'fold' button is also worthy of mention: this condenses the display matrix to show only the MIDI notes that are actually used in the clip, thus reducing scrolling up and down to get to two adjacent notes that might be a couple of octaves apart. Notes can be selected in all the usual ways, including rubber-banding the mouse over an area of multiple notes, or shift-selecting them individually. It is also possible to click on the field to the left of a row of notes and select all notes of the same pitch, which is useful, especially when you want to adjust the overall level


Ableton Live 4

of one drum sound in a part while retaining the dynamic variation between individual hits. MIDI controllers are manipulated using the envelopes, which can be drawn in using the pencil tool from the main control bar. Despite Ableton's claim to have developed a completely new way of handling MIDI, though, it has to be said that it is hard to completely reinvent a format so well established, and as a user of Logic who is used to having at least two simultaneous displays, I find Live 4's MIDI slightly on the frugal side. I really do miss having some numeric information about position, velocity and so on, since purely graphic manipulation is not my thing. But this is a personal choice, and I know plenty of musicians who would say the opposite. Put it this way: it is hard to imagine Sir George Martin doing his next orchestral arrangement using Live, but then again I don't see him scratching beats either. Live's MIDI is great for simple, pattern-based music such as is found in most dance styles, while people who want to be able to approach the subject from a more musicianly point of view will want to stick to Sibelius or Logic. They will, however, still love the possibilities Live gives them in the audio domain, and of course it's still easy to run Live as a Rewire slave to a more mainstream MIDI program.

Warping Time Also in the clip view section, we prepare the timing and tuning aspects of a clip. Depending on the nature of the source material Live gives you four different algorithms for the tuning/stretching of audio: Beat mode slices up a rhythmic loop in a similar way to Recycle, two other modes called Tone and Texture offer different algorithms for granular pitch-shifting and time-stretching, which suit different material, and Repitch behaves like a conventional sampler, where playing a clip back at a new tempo alters its pitch. One drawback of Beat mode is that it introduces artificial transients on hits that are longer than the selected resolution. If, for example, a loop is essentially made up of 16th notes with the occasional eighth in between, the longer notes will acquire a slight peak halfway through. This leads to audible signal degradation, depending on how extreme the time-stretching is. In some cases this can be a good thing when you are trying to create interesting sounds, but the compound effect can be quite severe, and anyone trying to achieve natural-sounding tracks needs to watch out. I found that one way around the problem is to layer stretched loops with clean samples in order to mask the effect. This effect also becomes quite obvious when dealing with clips that consist of whole tracks when these contain some non-rhythmic passages, such as ambient drops or a cappella vocals. Setting so-called 'warp markers' within a piece of audio determines which points of the clip will be quantised to the tempo. For this purpose Live writes another file, associated with the original audio file, onto disk. Having prepared clips in this way you can combine any piece of audio with any other and the software will automatically play them in time with each other. It's in this instant matching of file:///H|/SOS%2004-09/Ableton%20Live%A04.htm (10 of 13)9/26/2005 12:03:55 AM

Ableton Live 4

material that Live is so different to any of the other software around. Even when auditioning clips from disk that have not been made part of the session in the browser window, Live Live supports VST and Audio Units plug-ins, will play them back in time, provided and also includes a selection of proprietary they have been truncated and warpeffects. marked at some point. If they have not been prepared the software takes an educated guess, which sometimes works if the sample has at least been cut to the nearest bar. Apart from the DJing applications, these features make Live a very powerful composition and beat-generating tool as well. In the so-called legato mode, for example, all loops get triggered at the same time, and switching from one clip to another in this mode means that playback switches from one clip to another midbar rather than retriggering the new loop on the downbeat. Combining a number of drum loops in this way makes Live a great groove-creation tool. Another welcome addition to the feature list in Live 4 is the swing function, which globally adds swing in percentage points to all MIDI clips and audio that has 'warp' enabled and isn't set to Repitch mode. Unfortunately Live still doesn't allow for user-built groove templates, but compared to the options available in previous versions (straight quantise or no quantise) this is a vast improvement that in all fairness will be good enough for most situations. Loops with an inherent swing need to be straightened out first (by setting the warp markers to each swung hit), because otherwise there will be a compound effect when you apply additional swing. While we're on the subject of new features, Live now allows for reverse playing of clips by writing temporary files onto the hard disk.

Keeping It Live Once all your clips have been set up, they can be triggered using either MIDI and/ or the numeric keyboard. Using the Key Map Mode and MIDI Map Mode switches it is very easy to assign trigger sources to clips. Apart from the abovementioned launch modes there is also the legato setting in the clip view, which allows the player to vary the pitch of a clip during playback — if you like, superimposing a melody onto a piece of audio that is continuously playing. The RAM play option comes into its own here because it cuts down the latency and allows for smoother disk handling, which greatly increases the overall performance. With a combination of alphanumeric key strokes and a MIDI keyboard or controller set up to trigger parts that require more dynamic and/or melodic modulation, Live thus becomes an 'on the fly' arrangement tool. It is in this area file:///H|/SOS%2004-09/Ableton%20Live%A04.htm (11 of 13)9/26/2005 12:03:55 AM

Ableton Live 4

that dance companies and more adventurous DJs get a lot from the package.

Causing A Scene Another way of assembling an arrangement very quickly is to prepare a number of combinations horizontally across all the channels that then can be triggered as a scene using a single master switch. In a way this is the opposite of creating arrangements using mute groups; you are, if you like, using 'play groups'. Again, this can be extremely useful when working on an extended arrangement on the fly, or in the context of a live set where you need the timing of a machine combined with the flexibility to improvise structure. With the 'capture scene' command you quickly build up a complex set of arrangement sections that can be triggered instantly via MIDI, all on the fly, without stopping the music once. With the global Record button enabled in the main transport bar it is possible to record any such performances on to a more conventional sequencer layout: all the different triggers from all the clip slots in a track will be recorded and displayed on a track in the Arrange view. If you have a MIDI control box connected and assigned to any of the parameters, the global Record switch allows you to record your automation. Any data added in subsequent passes replaces the original, which is a clean and effective way of storing automation.

Final Thoughts What sets Live apart from all other audio packages is its immediacy: it really feels more like an instrument than a sequencer or recording system. It has a highly intuitive interface and, in its simplicity, is the ideal tool for musicians who want to have a performance-based computer system. I found Live 4 to be very stable on both Mac OS 9 and OS X, and I would therefore not hesitate to use it in a live situation. Although the newly incorporated MIDI editing capabilities don't offer the same depth as their counterparts in more conventional sequencers, Live 4 puts a different spin on MIDI sequencing that can open up new ways of working. Above all it's great fun to use and absolutely belongs in any creative music environment.

Published in SOS September 2004


Ableton Live 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



Analogue Systems RS300

In this article:


Description & Menus Memories, Special Options & CV-to-MIDI Converter Published in SOS September 2004 Show Inputs Applications Print article : Close window Conclusions

Reviews : Format converter

Analogue Systems RS300 £340 pros Converts eight analogue voltages simultaneously to a variety of MIDI messages. Allows you to generate notes from any voltage sources, including LFOs and envelopes. Can create MIDI Clock from analogue sources. 40 user memories.

cons Priced a little beyond the casually curious. No MIDI in/merge.

summary A remarkably useful module in ways you might not initially guess. When you start to use it, you find new applications constantly springing to mind — although you have to be careful not to clog your MIDI cable with more data than your receiving synth can handle.

No, we haven't got that the wrong way round... it really is a CV-to-MIDI converter. 'But why would you want such a thing?' you might ask. If so, read on and find out... Paul Nagle

Hands up those of you who, on skimming the contents of this magazine, assumed this review involved a MIDI-to-CV converter. If so, take a closer look. Analogue Systems have produced something rather more unusual here; something that might require a little thought before its potential is appreciated. If you have preMIDI synth gear that works on Control Voltages (CVs), you may well find this latest addition to the RS Integrator modular synth interesting.

information £340 including VAT. Analogue Systems +44 (0)1726 850103. +44 (0)1726 850103. Click here to email

Description & Menus

Test Spec

The RS300 module is just under five inches wide, or 24 units of horizontal pitch, the measurement system used by various modular manufacturers including Doepfer and Analogue Systems (1HP equals a fifth of an inch). A MIDI Output socket and no less than 10 voltage inputs provide clues to its raison d'être. Of these, eight are CV inputs and the remaining two are trigger inputs. When in use, a small red data LED flashes to confirm that MIDI data is being generated.

RS300 firmware reviewed : v1.1 (dated 7/7/04).

The module is crowned by a 2x20 backlit screen which, in combination with the

www.analogue systems.co.uk

file:///H|/SOS%2004-09/Analogue%20Systems%20RS300.htm (1 of 7)9/26/2005 12:03:58 AM


Edit knob and Cancel button, serves to access a series of menus. Rotating the Edit knob displays menu pages and to enter each page, you simply push the knob. On reaching a page in which values may be updated, turning the Edit knob performs this action. At all times, the Cancel key returns you back one level. Despite the interface's menu-driven nature, the navigation system remains quick and effective to use, and will be instantly familiar to anyone who has ever used a modern digital effects unit or synth; it's just that this style of user interface is much less commonly seen on Analogue Systems' gear. The final item on the panel is a Trigger Mode switch, which comes into play when MIDI notes are generated — more of this later. The top-level menus are named as follows: Input 1-8 Trigger/Switch Memories Special Options Show Inputs With only a couple of exceptions, each of the CV inputs behaves identically. Submenus are provided to specify the input voltage range and the type of MIDI message that should be generated. Each may be defined independently, with four possible voltage ranges to choose from: -10V to +10V, -5V to +5V, 0V to +10V, and 0V to +5V. From these voltages, a variety of seven-bit MIDI messages can be generated, such as pitch-bend wheel messages, aftertouch data, MIDI Continuous Controller (CC) messages from controller 0 to 127, MIDI note data (although from CV 1 only — see below), and velocity information for the aforementioned note data (again, see below). If you want an accurate translation of your input voltage to MIDI output, you should specify the range correctly. For example, if you are using the signal from an envelope whose level is between 0V and +5V, specifying a range of 0V to +10V will effectively compress the output into a range covering only half the possible values, because your envelope will never reach the peaks you have told the RS300 to expect. The first two CV inputs have additional functionality not present in inputs three to eight. CV 1 offers the intriguing possibility of generating MIDI notes, whilst voltages received at CV 2 can be translated into velocities for those notes. If you do not use CV 2 in this fashion, all the output notes will have a fixed velocity specified within the menu. In order to gain the maximum useable range from your input, notes can be transposed by ±36 semitones. The Trigger Mode switch (see inset picture, right) on the module's front panel governs how notes are triggered. Set this switch to Trigger and note length is



determined by the duration of gate signals received at the Note Trigger input socket. Each gate signal creates a new MIDI note whose pitch is obtained from CV 1. In this way you can play MIDI synths via any synth that works on the octave-per-Volt CV system, provided that they are equipped with CV and Gate outputs. In practice, the RS300 did not respond well to fast trills from my Roland SH101 keyboard, but from an analogue CV sequencer such as my ARP 1621, triggering was precise and reliable. Creating MIDI notes in this way is straightforward enough, but wouldn't it be great if you could do so from a pure voltage source, one that lacks an associated trigger — like an LFO? With the RS300, you can. You simply flick the Trigger switch to its 'Free Run' setting and then navigate through the CV 1 Keyboard Settings menu until you arrive at the 'Note Sequencing' option. This sub-menu contains two settings for trigger generation. In the first instance, whenever a new note is detected, the previous note is turned off before the new one is played. In the second instance, the new note is turned on before the previous one is released. Thus, the second method generates legato or overlapping notes, which some synths can use to introduce portamento effects. To set the length of these 'free-running' notes, a 'Free Run Note Hold' option is provided with values from 0.1 to 12.7 seconds, plus an 'infinite' setting — when this option is selected, the last note generated will receive no note-off event. One bonus of the note-hold parameter is that you can make a note that lasts a specific time, which is great when used with effect-type patches or with slow attack envelopes. I found that triggering was far more precise when the RS300 generated the note on/off events in this way, as opposed to receiving the gate information as described earlier. Those trills were faithfully rendered every time! In cases where the module is having difficulty interpreting an unstable input voltage, the Note Delay parameter allows a period of between three and 50 milliseconds for it to stabilise before deciding which note to generate. In most cases, I found I was able to use the minimum setting. Finally, there is a calibration routine for notes, based on a fixed range of three octaves (note that this is designed for octave-per-volt analogues). The module arrives ready-calibrated, so you may not need to do this. However, if you do, and have a three-octave C-C keyboard, simply select the 'Apply 0V To Input 1', press the Edit button and play the lowest note. Then select 'Apply 3V To Input 1' option, press Edit again, and play a note three octaves higher. My Roland SH101 has only 2.5 octaves, so I employed its transpose function to send the required range. You can't fool the calibration with voltages in a different range; I contemplated creating an inverted keyboard effect by sending a high voltage first and a low voltage second, but the module rumbled me and aborted the



operation. The calibration is for generated notes only — CV 1's Input Voltage range selection is ignored when notes are being produced. Finally, a 'Show Keyboard Input' sub-menu displays the note detected. The Trigger/Switch input (shown in the inset pic above) is the final one to require explanation. It is activated by a voltage blip such as a pulse wave or short envelope, and is capable of generating MIDI program changes, various MIDI on/ off pedal controllers (sustain, portamento, soft pedal, and so on), a percussion note, or MIDI clock. If you choose to generate a program change, you can specify one program to be sent when the voltage is high and another when it is low. Of perhaps more value, you could generate a specific MIDI note complete with velocity and MIDI channel (this option defaults to MIDI channel 10; the designers clearly had percussion triggering in mind). For me, the most useful option in the Trigger/Switch menu is the ability to generate MIDI Clock. This covers an impressive range of tempos from 0 to 900bpm and has a variable 'Clocks Per Beat' parameter to set the number of incoming pulses per output MIDI beat. I used it to synchronise MIDI drum machines to my vintage sequencers and use all the tricks analogue offers in terms of tempo modulation.

Memories, Special Options & Show Inputs There are 40 user memories in which to store your configurations. As the function of specific MIDI CC numbers varies from synth to synth, you will probably allocate one or more of these memories for each synth you plan to control with the RS300. As the RS300 has no MIDI In socket, the only way to select programs is via the menu system. I did wish for some means to do this remotely, but at least the menu is uncomplicated, with Read, Write, Edit Name and 'Erase All' options, as you'd expect. The Special Options menu allows you to set the MIDI channel of the module, which is a global setting and not stored within user memories. You must therefore change the channel manually when controlling different synths or, alternatively, use a rechannelise function on your sequencer if you have one. Also located here is the MIDI Data Rate parameter, with settings of low, medium or high. The RS300 produces controllers, notes and aftertouch as accurately as MIDI allows. But since MIDI is a serial protocol, your synth might struggle under the onslaught of eight streams of dense information. If your source voltages are derived from riotous LFOs, wild Sample & Hold outputs or quickly sweeping envelopes, and then you further add to this load by banging in some notes on the keyboard, timing may suffer. In such cases, experimenting with lower data rates can be a lifesaver.



This menu also allows you to call up the current version of the operating system (version 1.1, dated 07/07/04, in the case of the review model) or reset the unit to its factory defaults. The Show Inputs menu option, as the name suggests, brings up all eight CV inputs on a single screen. If the RS300 is being used to produce MIDI notes, the note value is shown. An asterisk appears in the top left-hand corner of the display whenever a gate is received at the Note Trigger input. Similarly, a small 'T' appears in the bottom right-hand corner when a voltage is detected at the Trigger/Switch input — a valuable visual reference. In all other cases, the numeric value of the event being transmitted is shown.

Applications Now we're familiar with the inputs and menu system, let's take a look at some of the things you might do with the RS300. I've always shied away from the phrase 'limited only by your imagination', but the possibilities offered by this module are truly far-reaching. A few examples might make the value of this module clearer... I've always felt that analogue sequencers had many advantages over (the majority of) their digital counterparts, so the opportunity to generate notes on my modern gear from my venerable ARP 1621 was one not to be missed. The ARP is a 16-step sequencer that can be divided into two rows of eight steps, which I did here. I connected the first row into the CV 1 input and connected the ARP's Gate output to the Note Trigger jack. Finally, The review unit was supplied in one of I connected the second sequencer row Analogue Systems' stand-alone wooden to the CV 2 input of the module, cases, with its own power socket. Of course, if you're wiring the module into an existing specifying velocity as its output. Hitting modular system, a version without the case Run on the sequencer produced a and power supply is also available. stream of MIDI notes and gave me full hands-on control of the note values and velocities via the ARP's sliders. When I drafted in a second sequencer generating MIDI controllers via the other CV inputs, all the marvels of analogue sequencing were at my fingertips — but they were now recordable into Cubase for further tweaking and arranging. I then moved on from notes to controller generation. The possibilities offered by having up to eight analogue modulation sources generating MIDI controller curves are fascinating. I dug out the manuals for various synths in the studio — including my Access Virus, Korg Prophecy and Novation K-Station and Supernova II. Having decided on the parameters to control, I connected a series file:///H|/SOS%2004-09/Analogue%20Systems%20RS300.htm (5 of 7)9/26/2005 12:03:58 AM


of outputs from my Digisound analogue modular synth. I then sat back agape at the most unusual and complex sounds I had ever heard emerging from my virtualanalogue synths. For example, directly modulating eight of the Virus' filter parameters using a series of LFOs resulted in a swirling pad to die for — the Virus is great, but doesn't have eight LFOs of its own! In some cases, I had to reduce the RS300's MIDI Data Rate before my synths drowned under the amount of data my analogue modular was sending them. Some of the most outlandish and experimental sounds were achieved by selecting controller numbers at random, connecting my voltage sources, then tweaking liberally. Forbidden Planet, here we come! Lastly, I tried running some of my MIDI gear on Clock information generated in the analogue domain. I love my collection of Korg Electribes, but often wish they had a clock output with which to synchronise my older gear — such as the sequencer in my faithful Roland SH101. As the SH101 transmits gate information when its sequencer is running, I connected this output to the Trigger/Switch input. I was then able to generate MIDI Clock and run my drum machines at the LFO rate of this simple monosynth, although I had to manually hit Run on the drum machine, since the RS300 does not provide a MIDI start command.

Conclusions Analogue Systems have produced a module that I always hoped would exist one day. True, Doepfer's A192 also converts voltages to MIDI controllers — but to my knowledge, the RS300 is the first to generate MIDI notes in this way as well. Producing streams of notes from analogue origins opens up a whole world of extravagant sweeps and warbles which can make you look again at even the blandest and most forgotten-about MIDI gear in your studio. Similarly, the ability to generate a clock signal, then slave MIDI drum machines and sequencers to it has advantages that modular freaks will already be well aware of. Best of all, from my perspective, is the RS300's ability to generate MIDI controllers. By directly controlling synth parameters, you can create mutant sounds and modulations that are unachievable with the synth alone. If you want an LFO to sweep the filter overdrive of a Supernova, you now have a means to do it. And there are many more similar examples, depending on the gear you have. You can probably tell that I'm impressed, and I won't deny it — but to make the RS300 even better, I'd have gone the whole hog and included a MIDI input capable of merging incoming data with that generated by the CV sources. You could then have used MIDI program changes to select user memories. Of course, this would have driven up the cost of a module that is already far from 'bargain basement', so Analogue Systems leave you free to add your own merge if you deem it necessary. I'll end by saying the RS300 is a module whose value will be best appreciated by file:///H|/SOS%2004-09/Analogue%20Systems%20RS300.htm (6 of 7)9/26/2005 12:03:58 AM


those of you with a diverse selection of voltage sources on tap. Combine it with a modular synth or analogue sequencer and you can expect hours of happy experimentation with your MIDI gear. If my examples have inspired you to think of better ones, then you are probably just the sort of person Analogue Systems had in mind when they brought this module into the world. You might discover a whole new perspective on driving synths (or software synths) from your analogue gear, which can't be a bad thing. Published in SOS September 2004 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



Behringer B2030A Truth

In this article:


Construction & Specifications Active Studio Performance

Monitors

Published in SOS September 2004

Behringer B2030A £234 pros Good value for those on a tight budget. Magnetically shielded. Flexible frequency-tailoring functions.

cons Not as much bass as I'd hoped for. The sound's a little light on detail, but nothing to stop you setting up a decent mix balance.

summary You generally get what you pay for when it comes to monitors, but these new monitors do seem to offer just a little bit more than you'd expect for the price.

information B2030A, £234 per pair; B2031A, £281 per pair. Prices include VAT. Behringer UK +49 2154 9206 6441. +49 2154 9206 321. Click here to email www.behringer.co.uk www.behringer.de

Print article : Close window

Reviews : Monitors

The new baby in Behringer's Truth monitor range offers great value for money and sleek good looks. Paul White

Behringer have updated their Truth monitor range to offer two models, the B2030A reviewed here and the larger B2031A. Both are ported, active twoway nearfield systems, the main difference being that the B2031A has an 8.75-inch bass/mid-range driver and a one-inch tweeter, while the B2030A has a 6.75-inch bass/mid-range driver teamed with a 0.75-inch tweeter. In both instances the speakers are built symmetrically, so there's no need for Photos: Mike Cameron separate left-hand and right-hand speakers. As you'd expect, the larger monitors have more firepower in the amplifier department (150W and 75W) and generate a slightly greater Sound Pressure Level (SPL). The B2030A has a 75W amplifier driving the bass/midrange unit plus 35W powering the ferrofluid-cooled, soft-domed tweeter. Like the earlier Truths, both drivers are set into a moulded baffle trim that incorporates a tweeter waveguide to control directivity, and both drivers are magnetically shielded for use near TV monitors or the few antiquated glass CRT computer screens not currently awaiting disposal at a car-boot sale.

Construction & Specifications Clearly the speakers are built to a price and are economically constructed from plastic-laminated E1 MDF, though this is actually a good cabinet material because of its density and self-damping properties. You might not get natural file:///H|/SOS%2004-09/Behringer%20B2030A%20Truth.htm (1 of 4)9/26/2005 12:04:01 AM


cherry-wood veneer, but the overall impression is smart. The long-throw bass/ mid-range driver features a polypropylene cone with roll surround mounted in a cast aluminium chassis. The port exits comprise separate slot-shaped moulded inserts set into the baffle on either side of the tweeter. The active crossover, which is set at 2kHz, utilises fourth-order (24dB/octave) Linkwitz-Riley filters, and the speakers are also designed to be used with or without an optional subwoofer, though they manage a perfectly decent low end without one. Separate high- and low-frequency limiters look after the selfpreservation interests of the drivers, and there's an automatic standby mode that puts the amplifiers into sleep mode if the speakers aren't used for more than five minutes. They wake up almost instantly on receipt of an audio signal. At the rear of the cabinet, which measures 12.5 x 8.4 x 8.33 inches, is a choice of XLR or TRS jack servo-balanced inputs, plus an IEC mains connector and a power switch. These are all mounted on the amp chassis, with the mains and input connectors on the underside. An LED set into the front-panel baffle surround comes on when the speakers are powered up, and a second LED indicates that the limiter threshold has been reached. An input gain trim control allows level adjustment from -6dB to +6dB, and there are slide switches to adjust the frequency characteristics of the speaker to suit different environments. At the low end, the response can be flat, -2dB, -4dB, or 6dB, which the manual tells us is to accommodate an optional subwoofer or to simulate the response of smaller speakers. A further switch compensates the low end to take account of room placement, and this again offers flat, -2dB, -4dB, or -6dB options. Normally bass attenuation is only needed if the speakers are located close to a wall or corner, as the boundary effect causes a rise in bass energy, but it may also be useful to reduce the bass output slightly in smaller rooms. At the high end, the switch positions are +2dB, flat, -2dB, or -4dB to suit the HF reflectivity of the room in which the speakers are used. Unfortunately the manual doesn't give any details of what frequencies these filters are set at, so all adjustments must be made by ear. Although quite different in size, both models have ostensibly similar frequency responses extending from 50Hz to 21kHz. However, no decibel limits are given, so this means relatively little. The provided frequency-response plot is also an obviously 'smoothed' response curve, and so doesn't show up all the narrow humps and bumps that every speaker exhibits. Where size does matter is in sound level — the B2030As can generate up to 113dB per pair at one metre, while the larger B2031As manage an extra 3dB.

Studio Performance file:///H|/SOS%2004-09/Behringer%20B2030A%20Truth.htm (2 of 4)9/26/2005 12:04:01 AM


In my own studio, with all the EQ switches set flat, my first impression was that the monitors sounded slightly brash, with less depth of bass than I expected. This situation improved noticeably when I dropped the tweeter level by 2dB, so it is important to adjust the EQ settings to match your room and monitor positions. Once optimised, the speakers delivered a very decent level of performance given their budget price, though their shortcomings were still evident when they were tested alongside my Mackie HR824s. I found that the subjective depth of bass was less than I expected based on the technical spec, while the tweeter sounded slightly 'forward', but they still delivered a fairly good overall balance and proved capable of discriminating between good mixes and not-so-good mixes. Of slightly more concern to me is the way these monitors, and some other budget monitors I've tried, seem to filter out some of the more subtle elements of a mix, leaving the principal parts sounding rather exposed, not unlike listening to an MP3 and then comparing it with the original. The finer points of the reverb and the air between the instruments tend to be diminished, which reduces your ability to hear 'into' a mix, but you have to bear in mind that these comparisons are being made with much more expensive speakers. As low-cost speakers, they still perform rather better than you could reasonably expect for the price, and it shouldn't be hard to set up a decent mix balance using them. If you are on a very tight budget, then the B2030As offer very good value and they can be made to perform perfectly adequately in smaller rooms, provided that you take due care with their placement and experiment with the EQ switch settings. I think the designers have used their available manufacturing budget extremely well in producing a useful and good-looking monitor at such a low UK price point, though there's no denying that spending more money will buy you better performance and greater accuracy. With monitors you generally get what you pay for, but in this case it's probably fair to say that you get slightly more than you pay for! Although clearly aimed at the budget-limited home-recording enthusiast, the B2030As could also serve a valuable part in the more professional studio by acting as a secondary monitor for checking how mixes might translate to a domestic playback system. Finally, the B2030As are designed as nearfield monitors, and as such should be used fairly close to the listener, where they deliver plenty of level. If you feel you need more depth of bass and more level, then the B2031As may be a more suitable choice. Published in SOS September 2004

file:///H|/SOS%2004-09/Behringer%20B2030A%20Truth.htm (3 of 4)9/26/2005 12:04:01 AM




file:///H|/SOS%2004-09/Behringer%20B2030A%20Truth.htm (4 of 4)9/26/2005 12:04:01 AM

Carillon Core 4

In this article:

Carillon Core 4

Heat And Noise Music PC Closer Inspection Published in SOS September 2004 Matrox P650 Triple-head Graphics Card Print article : Close window Performance Reviews : Computer Recording System Specifications Of Review PC Silk & Steel Carillon Extras

Carillon Core 4 £2199 pros Impressive build quality. Fast and vast 400GB RAID 0 hard drive array. Massive 2GB RAM. Versatile DVD/CD writer. Free 6GB sample library.

cons Very quiet in a rack, but noisier than some other PCs when desktop mounted. 320 Watt PSU must be running close to its limit. No floppy drive option.

summary With the largest and fastest audio drive array I've reviewed to date, triple monitor capability, DVD writer, and loads of RAM, this Carillon Core 4 system should prove ideal for any musician who needs to work with video and loads of audio tracks.

information Basic system as reviewed without monitors, music hardware or software £2199 including VAT. Carillon +44 (0)20 7692 7628. +44 (0)20 7836 7089. Click here to email www.carillondirect.com

Computer specifications have changed radically in the three years since Carillon launched their first music PCs, but the company's signature metal rackmount case still sets them apart. Their powerful Core 4 machine offers support for three monitors, plus a huge 400GB RAID 0 hard drive array for audio data. Martin Walker

I first reviewed a Carillon music PC in SOS July 2001 (read the review on-line at www.soundonsound.com/sos/jul01/articles/carillonpc.asp), and the 866MHz Pentium III processor, 128MB of RAM, and single 20GB hard drive of that machine provide a reminder of just how rapidly PC technology moves onwards. However, the company's custom-designed rackmount case has stood the test of time, and I suspect will still be going strong for many years to come, its 2mmthick steel sleeve making it one of the most rugged designs available. Carillon have recently expanded their range to include cheaper tower case systems from £599, miniature shuttle systems from £669, and both Centrino and desktop replacement laptops, but the core of their range is still based on the AC1 (Audio Computer) rackmount case. Models in the AC1 range start at just £699 for the AC1 LE and move up through the Core systems 1, 2, 3 and 4 Photos: Mike Cameron with faster processors and more hard drives. There are also more specialised machines such as the blue-panelled AC1HD — the only Digidesign HD-certified audio-specific PC in the world — and the AC1X, fitted with dual Xeon processors.

file:///H|/SOS%2004-09/Carillon%20Core%A04.htm (1 of 8)9/26/2005 12:04:07 AM

Carillon Core 4

Core systems can be supplied without music hardware and software, but you can also select these on-line to add to your system before it's delivered, or choose from various pre-configured solutions for applications such as sampling, beat and loop creation, and recording guitar-based music. The subject of this review is the Core 4, which is the fastest single-processor PC in the range, fitted with a 3.4GHz P4 processor, 2GB of RAM and three hard drives: an 80GB EIDE model for system use, and twin 200GB SATA models set up as a RAID 0 array for audio purposes. With a Matrox P650 triple-head graphics card, a versatile DVD-RW writer and the usual Carillon extras, it has an impressive spec.

Heat And Noise Because components such as the 3.4GHz processor and P650 graphics card generate a lot of heat, the Fanmate controller in the Core 4 is turned up beyond the minimum setting adopted by many retailers on other systems, while the AC1 case has large side intake vents that let some of the fan noise out, so it's not the quietest PC I've ever used. However, this system has the huge advantage that the majority of users will bolt it into a rack, which would render the remaining side and rear fan noise virtually inaudible, so in practice it will be blissfully quiet. After some hours in active service, the processor stabilised at a slightly hotter than normal 57 degrees Centigrade, so the Fanmate controller setting seemed wise. I was however a little concerned at the 320 Watt capacity of the PSU, and while I ran into no problems during the review period, I think a 400W model might be more appropriate for a system fitted with three hard drives and a DVD burner.

Closer Inspection The AC1 rackmount case was as impressive as I remembered it, and considerably sturdier than any PC tower case. Its aluminium front panel offers up to three 5.25-inch drive bays and two 3.5-inch ones; the three internal hard drives and NEC DVD-RW drive already fitted meant that only one of the (5.25-inch) bays was still free in the review system. The handy extras are still there, including the modular front-panel options for transport controls or MIDI controllers, Sorbothane vibration-damped feet for desktop use, and the front-panel stereo jack Patch socket, which is routed through to an identical rear-panel socket so you can have permanent front-panel access to your choice of audio I/O. Carillon only sell this case as part of a complete PC system. Inside, the interior was extremely tidy. Carillon are still staunch supporters of Intel's own motherboards, and while these may not have the range of overclocking features favoured by some enthusiasts, Intel boards are renowned for their rock-solid stability and reliability, which is far more important for the serious musician. The D875PBZ board fitted in the Carillon PC is well laid out, has a massive passive heatsink on its Northbridge chip set rather than a fan, and the various connectors are positioned such that there's still plenty of space left file:///H|/SOS%2004-09/Carillon%20Core%A04.htm (2 of 8)9/26/2005 12:04:07 AM

Carillon Core 4

around the most critical component — the CPU. In this machine a 3.4GHz Pentium 4 had been fitted, along with Zalman's popular Flower cooler. The cool air The Carillon AC1 case features built-in intake is provided by twin side vents, transport/MIDI controls, and an audio socket while a 80mm case fan sits adjacent to which is patched through to the rear panel. the CPU to extract the warm air via an adaptor tube to the 60mm exit grille. A 320 Watt version of Carillon's own Ultramute brand of quiet PSU completed the power and cooling arrangements. The D875PBZ has one AGP and five PCI expansion slots. A Matrox P650 triplehead graphics card had been fitted in the AGP slot (see box) while PCI slot three housed a Conexant Systems 56k modem card. Many specialist music retailers don't fit modems to their music PCs, but Carillon rely on one for their Carillon Fix service (see Carillon Extras box). Slot four housed a VIA Technologies Firewire Controller with two six-pin and one four-pin Firewire ports, overcoming the lack of integral Firewire support that is this motherboard's only real limitation. Six USB 2.0 ports appear as standard on the rear panel, a total that's reached on many PCs only by installing a dummy backplate and thus losing a PCI slot. The motherboard supports eight USB 2.0 ports in total, and one of these was used for the optional RTM1 Transport Panel (£69) with its five large positive transport buttons and metronome clicker, which I described in detail in my original Carillon review. The remaining ports are one serial, one parallel, PS/2 mouse and keyboard, and LAN. The BIOS is proprietary to Intel, and while there are few overclocking options, there are still plenty of performance-related tweaks on offer for those who wish to delve deeper. Carillon had left hyperthreading disabled for maximum compatibility, but are happy for users to enable it if The Core 4 provides six rear-panel USB ports without using up a PCI slot, although they wish, which I later did for my CPU tests. There's no onboard audio chip to there's no integral Firewire support on the Intel motherboard, meaning that a PCI disable on this motherboard, and for Firewire card is included. the review model Carillon had fitted one of Emu's new 1212M audio interfaces, with its 1010 PCI card in slot two and 0202 daughterboard in slot one. This left only slot five empty for future expansion, but the Emu cards do provide a Firewire port, so you could probably do without the separate Firewire card if necessary. Drive-wise, there are two ATA100 headers and two SATA 150 headers on the motherboard. Carillon had fitted Seagate Barracuda hard drives throughout, with an 80GB PATA model connected as Primary Master for System duties, and a


Carillon Core 4

pair of 200GB SATA models connected as a RAID 0 array for audio work. All three drives had been fitted into Silentdrive sleeves inside a robust internal cage, using a small modification to accommodate the two SATA connectors. In the topmost 5.25-inch drive bay a black NEC ND2500A DVD-RW drive had been fitted and connected as EIDE Secondary Master. This is apparently the first writer to manage 8x DVD-R and 4x DVD-RW speeds while also supporting DVD +R and DVD+RW formats for maximum compatibility, as well as providing good CD-R writing quality at a good price. Carillon had disabled the floppy drive controller in the BIOS, as no drive was fitted, and I suspect few users will miss one. Finally, for the purposes of this review Carillon also provided me with three Viewsonic VP181b flat-screen monitors, along with Steinberg's Nuendo 2.0 software, so I could perform some practical tests of the P650 graphics card's triple-head capabilities (see box).

Matrox P650 Triple-head Graphics Card For this particular system Carillon had fitted a Matrox Millennium P650 graphics card. Unlike its Parhelia stablemate, this model doesn't require a cooling fan, but still has dual-head support for two monitors with up to 1920 x 1440 pixel VGA displays, or 1600 x 1200 with digital connections. Moreover, once fitted with the upgrade kit (a one-foot DVI-to-dual-HD15 splitter cable) it provides triple-head support for the ultimate three-monitor display setup. Matrox unveiled this combination at the 2004 NAMM show specifically for audio professionals, and it's a good sign that we're being taken more seriously by the PC component industry. With one or two monitors connected you can use any combination of analogue or digital models, with the added versatility that the second connection can also be to a TV or video recorder. With three monitors the first can again be digital or analogue, but you need to connect the dual-monitor adaptor cable for monitors two and three, which restricts them to analogue or TV. This latter restriction is slightly disappointing when you've been sent three digital-capable monitors, but it did give me the opportunity to compare the analogue and digital connections — the difference was subtle but noticeable, with the digitally connected monitor image being slightly sharper and clearer, but I managed to minimise this with a tweak of the analogue contrast settings. The Matrox P650 also has various multi-display setups including a 'feature' option when using more than one monitor, accessible via the Matrox Powerdesk-HF utility. I could run the three supplied monitors as one 'Stretched' mode display, all using the same resolution and colour depth, which is ideal if you want to run your sequencer with a wide two-monitor arrange page and its software mixing desk on the third monitor. Alternatively you could choose the two-display Stretched mode for the sequencer, and use the feature display switched to PureVideo/DVDMax mode to run full-screen video in the third display. This could be ideal for anyone writing film or TV music, and worked really well with Nuendo. Finally, the twodisplay Independent mode lets you specify a different resolution, colour depth and relative virtual position for the first two monitors, and use the third as feature display. The Matrox card also provides a hardware-based video overlay feature to


Carillon Core 4

reduce CPU overhead when playing back digital video. I also found the Multi-Display Zoom functions very useful. You can select any region in your main monitors and display a larger version in the feature screen, either leaving the original region fixed or having it move with your mouse — very handy to create huge audio level meters, for instance! There are also numerous other smaller features such as the ability to decide where dialogue and message boxes appear, to avoid them being split across two screens. I was well impressed with the Matrox P650 from a musician's point of view, and apart from the onedigital-plus-two-analogue restriction mentioned earlier, the only negative point I could find was that the P650's heatsink became the hottest item in the entire PC, forcing the system cooling fans to work a little bit harder.

Performance As expected, all the tweaks for optimum Windows audio performance had been carried out; with 2GB of RAM, the page file had wisely been set to a custom Initial size of 2GB, with a Maximum size of 4GB. SiSoftware's Sandra test suite measured 4765MB/second and 4763MB/second for Integer and Float memory bandwidths, showing the slight performance boost of Intel's PAT technology over Asus' Hyperpath equivalent, which it beat by just a few percent. However, for most musicians running loads of plug-ins and soft synths, the single most important parameter is likely to be CPU performance, and this 3.4GHz P4 system measured 34 percent with the Steinberg/Fivetowers performance test with 23ms latency in Play mode, and 41 percent at 4ms latency. These figures dropped to 33 percent and 38 percent once I'd enabled hyperthreading in the BIOS, all of which tie in fairly closely with the expected improvement over my own P4C 2.8GHz system dictated by the 21 percent increase in clock speed. With the HT left enabled, Sandra's CPU Arithmetic benchmarks measured 10477MIPS Dhrystone, 3444MFLOPS Whetstone, and 7428MFLOPS for iSSE2, while CPU Multimedia measurements were 26074it/s integer and 37157it/s floating-point. Once again these results tallied quite closely with my own 800MHz FSB 2.8GHz Carillon's Pentium 4C 3.4GHz PC is the system multiplied by 3.4/2.8, proving fastest P4 system I've reviewed to date, and that the increased performance is its result sits exactly in line with other P4C solely due to the higher CPU clock systems of different clock speeds, beaten speed — in other words, you get about only by dual-processor setups based on 21 percent better performance than a Intel's Xeon. 2.8GHz P4C system, and about 13 percent more than a 3.0GHz P4C system. If you want even faster CPU performance then Carillon now have dual-Xeon systems in their range. The PATA hard drive had been split into a 10GB C system partition, with the file:///H|/SOS%2004-09/Carillon%20Core%A04.htm (5 of 8)9/26/2005 12:04:07 AM

Carillon Core 4

remaining 65GB or so devoted to a D partition for audio sample data, both being formatted with FAT32. Dskbench measured 55.6MB/second sustained read and write speeds for C and a slightly slower 50.3MB/second for the inner D partition. However, on this system I was far more interested in the results for the SATA RAID E drive, where the sustained read performance was a staggering 124.2MB/ second, and the write a still-amazing 95.0MB/second, giving a potential 412 16bit/44.1kHz playback tracks with a 128k block buffer size, which equates to 126 tracks at 24-bit/96kHz. Real-world figures are likely to be rather lower than this theoretical figure, but you ought to get a hundred tracks with few problems. To complete my findings, I got on well with the cordless mouse/keyboard pair, even up to distances of three to four metres away from the base unit, although I did find the scroll wheel a little 'scratchy' in use, and due to an oversight someone at Carillon had mistakenly stuck the wrong overlays on the left Shift and Windows keys.

Specifications Of Review PC Case: Carillon AC1 4U rackmount, with die-cast aluminium front panel, 2mm thick steel sleeve and 320W Ultramute PSU. Motherboard: Intel D875PBZ socket 478, with Intel 875P 'Canterwood' chip set running 533/800MHz system buss, with four DDR DIMM sockets supporting up to 4GB of dualchannel DDR333/400 DDR SDRAM. Processor: Intel Pentium 4C 'Northwood' 3.4GHz 512k L2 cache, 4 times 200MHz front side buss. CPU heatsink and fan: Zalman Super Flower Cooler CNPS7000-AlCu with Fanmate controller. Case cooling & silencing: Nexus quiet rear 80mm case fan. System RAM: 2GB of Infineon PC3200 CAS3 SDRAM, running as DDR400 dualchannel. System drive: Seagate Barracuda ST380011A, 80GB, 7200rpm, 2MB buffer, Parallel ATA. Audio drive: two Seagate Barracuda ST3200822AS set up as RAID 0 array, 200GB, 7200rpm, 8MB buffer, Serial ATA. Graphics card: Matrox Millennium P650 triple-head with passive heatsink cooling and 64MB RAM. Floppy drive: none fitted. DVD-RW drive: NEC ND2500A, ATAPI Ultra DMA 33 Mode 2 interface, 40x CD-ROM, 32x CD-R, 16x CD-RW, 12x DVD-ROM, 8x DVD+/-R, 4x DVD+/-RW, 2MB buffer. System ports: PS/2 mouse and keyboard, serial, parallel, RJ45 LAN, six USB 2.0, two six-pin and one four-pin Firewire, modem line and phone, audio patch socket. Keyboard & mouse: black Logitech cordless desktop with custom coloured key caps to match installed audio software, ball mouse. Installed operating system: Windows XP Professional Edition plus Service Pack 1.

For this particular system:


Carillon Core 4

Monitors: 3x Viewsonic VP181b Thin Edge Ultraslim, black, with 18.1-inch diagonal, 1280 x 1024 native resolution, 30ms response time, Xtreme View 160 degree horizontal and vertical viewing angles, DVI-I and analogue input connectors. Soundcard: Emu 1212M with version 5.12.01.0488 drivers.

Silk & Steel Carillon's heavy-duty aluminium and steel rackmount case will never be a cheap item to manufacture, but it means that their PCs are undoubtedly among the most rugged systems around. They have achieved an excellent reputation for quality in the three years since their launch, yet their cheapest rackmount system is just £699; and while £2199 may sound expensive for a 3.4GHz Pentium 4 PC without monitor, music hardware or software, this Core 4 system features a huge 2GB of RAM, a powerful Matrox P650 graphics card and a versatile DVD writer, plus three hard drives totaling 480GB capacity. Where I managed to find any other specialist retailer who could offer a computer with a similar specification, the price ended up quite similar. Although you can buy quieter desktop systems now that other manufacturers are using acoustic foam case linings, this Carillon Core 4 PC is an extremely attractive proposition for anyone who wants a reliable, desirable, and quiet PC to bolt into a rack, and it comes with an impressive list of extras including a 6GB sample library. It doesn't have the fastest CPU I've tested (that crown currently belongs to Red Submarine's dual 3.06GHz Xeon system), and it's fair to say that if your budget is over £2000 and you're aiming to run the maximum number of soft synths and plug-ins, rather than hundreds of audio tracks, a dual-Xeon system might suit you better. However, the Carillon Core 4's 2GB of RAM and amazingly fast and vast 400GB audio RAID drives make this system a perfect partner for someone who wants to run bucketloads of 24-bit/96kHz audio and sampler tracks, particularly with synchronised video alongside. Keep on tracking!

Carillon Extras With every Core system Carillon provide a Pinnacle software pack consisting of the Instant CD/DVD suite of burning and backup applications, Steinberg Clean 5 restoration and CD/DVD burning application, Wavelab Lite audio editor, and WinDVD 4 player. All systems also come with the Carillon Fix software that enables one of their engineers to remotely diagnose problems on your PC (I described a sample session in my July 2001 review), plus a full backup of your Windows OS and programs in the D partition along with a bootable Carillon recovery disc to restore the backup if you ever need to return your PC to its original shipped state. Carilon also go to great trouble with their documentation. There's a 16-page printed Getting Started Guide, while Carillon How is a system-specific help file that provides a handy socket-by-socket guide explaining how to connect up your audio interface and how to test your audio software with it. It provides a string of


Carillon Core 4

tutorials explaining various aspects of using them together, and has an extensive glossary of terms. Carillon Help offers a handy set of audio FAQs, plus details of their support web site's on-line help facility. In addition, you get the Carillon Loopstation 6GB sample library already installed on the Audio D partition. This is a big improvement over the original idea of letting registered users download samples free from the on-line version of the library, although this option is still available, and its contents will be growing in the months to come. Previewing and downloading samples is easy thanks to the supplied Loopstation search engine software, which lets you search by instrument or style. There's a huge range of multi-part acoustic, electric and synthesized instruments on offer, as well as loads of loops in a variety of styles and tempos. These are all of high quality, and the library should prove a valuable resource to dance and rock musicians, although there are no choral or orchestral sounds. Published in SOS September 2004 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



Cycling '74 Mode

In this article:

Bang For The Buck Mode Mono Mode Poly All In A Spin Mode Wash In Use Using Modulators Summing Up

Cycling '74 Mode £146 pros Quirky without being overcomplicated. Lighter versions of most modules are available to conserve CPU power. Some interesting timebaserelated effects not available elsewhere.

Cycling '74 Mode Plug-in Suite (Mac OS X) Published in SOS September 2004 Print article : Close window

Reviews : Software

If you're after effects and instruments to give your mixes a sideways slant, Cycling '74 might have just the thing... Paul White

Mode is a bundle of plug-ins from the same stable as Pluggo, and comprises cons three instruments and two effects Rather boring user interface processors plus 18 further modules design. derived from these. The additional Synths tend to be very CPU- modules include 'light' versions of the hungry. plug-ins, effects spun off from the summary instrument plug-ins, and modulators for use as modulation waveform sources. Mode is a useful grab-bag of effects and instruments, and Mode supports all the currently popular though some of these are plug-in formats on the Mac platform greedy on the CPU front, the (VST, AU and RTAS) and runs under availability of light versions helps offset this shortcoming. OS X only, requiring version 10.2.8 or later. Authorisation is via a code information available on-line when registering; until you register, the plug-ins will work in £145.99 including VAT. demo mode, where the audio is disrupted and replaced by a bleep every minute DACS Audio +44 (0)191 or so. 438 2500. +44 (0)191 438 2511. Click here to email www.dacs-audio.com www.cycling74.com

Test Spec Apple G5 dual 2GHz with 2GB RAM, running OS 10.3.3. Tested with Emagic Logic Pro v6.4.1.

Many of the the plug-ins enhance an existing sound in some way through the use of arpeggiators, sequencers, delay, distortion, filtering, feedback and modulation — but not always in the most obvious way. There are also modulators that can be used just as they would be in a modular synth, by patching them to control parameters of a different plug-in. Unusually, the majority of parameters are accessed via knobs that have no numerical display, which is intended to encourage adjustment by ear, and nestling alongside familiar and predictable parameters are oddball creations added to inspire weirdness — which is no bad thing. Furthermore, parameters are often provided with a greater control range than would normally be considered useful, simply so that the user can

file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (1 of 7)9/26/2005 12:04:11 AM

Cycling '74 Mode

experiment with excess. The full 'heavy' versions of the plug-ins include inbuilt effects, such as delay.

Bang For The Buck The first virtual instrument on the list is simply called Bang, and as its name might suggest, it is a drum/percussion sound synthesizer. It responds to MIDI notes by changing pitch and combines elements of subtractive synthesis, FM and simple sample playback. It can only create one sound at a time, so if running multiple instances of it turns out to be too CPU-intensive, it may be easier to create percussion sounds one at a time and then sample them for playback in a soft sampler. The instrument panel is divided into three sections, corresponding to the three synthesis types on offer. The sample playback section has its own five-stage envelope generator and tuning controls, plus bit-depth reduction for distortion purposes. The FM section is a fairly simple two-operator affair with modulation envelope, adjustable carrier and modulator pitch and waveform. Envelope and pitch modulation is also possible, with MIDI tracking of modulator pitch available, and of course everything is velocity-sensitive. There's also plenty to play with in the subtractive synthesis section, including the usual oscillators, filters and envelopes. All three sections have sends to feed a delay processor. The factory presets demonstrate this module's ability to create a vast range of non-emulative drum and percussion sounds, from kicks and snares to toms and synthetic cymbals plus plain weird noises. The three modules that make up Bang are also available as separate light plug-ins, to save on CPU resources where only one type of sound synthesis is required. Mode's Bang Delay is also available as a separate delay-with-mod effects plug-in, as is its filter/distortion module.

Mode Mono As the name suggests, Mono is a monosynth, but in this case using FM oscillators featuring extensive envelope control. Aliasing is turned into a virtue and feedback enables the creation of some very non-FM-style sounds. A very sophisticated distortion section is followed by a more familiar resonant filter to file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (2 of 7)9/26/2005 12:04:11 AM

Cycling '74 Mode

skim off the spiky bits, and as with Bang, a digital delay is built in. There's also a simple but effective monophonic arpeggiator with adjustable direction, octave-shift and timing controls, where the note duration can be extended to exceed the length of individual steps allowing for creative abuse. There are separate oscillator, amp and mod envelopes as well as LFO modulation, distortion and that handy arpeggiator, which can lock to the host's tempo. I think the distortion section also deserves special attention as it also includes a waveshaper, which sets the distortion transfer characteristic and can be used to radically change the harmonic structure of any waveform to the point that it becomes unrecognisable. Because this section has the ability to transform even silence into a noise, a safe Zero button is included to ensure that no input always results in no output. This synth has a fairly hard, contemporary sound — sort of FM meets Oscar meets PPG — and it is capable of huge variety despite its apparently straightforward user interface. Used with the arpeggiator, it can create both contemporary dance sounds and vintage electro-pop sounds. A light synth-only FM module based on Mono is also available as a stand-alone plug-in, as is the arpeggiator, which it is claimed can be routed to modulate any parameter of any of the other Mode instruments or effects. The manual points out that Arpeggiator probably won't work when the host sequencer's Freeze Track function is being used so it's best to render these tracks as audio files instead if saving CPU resources becomes necessary. Mono's delay section also shows up as a separate plug-in, as do its distortion/filter components.

Mode Poly Poly has a very simple user interface and uses digital waveform oscillators as sources, augmented by built-in delay and chorus effects, a resonant filter and a polyphonic arpeggiator that can be locked to the host's tempo. There are separate amplitude and modulation envelopes and a comprehensive display for interacting with the arpeggiator. An LFO Designer allows the user to create more file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (3 of 7)9/26/2005 12:04:11 AM

Cycling '74 Mode

complex modulation waveform shapes and each of the two oscillators can select from a list of 24 different 12-bit waveforms. Delay and chorus add further interest to the sound, but there is an 'effectless' version of the plug-in available if you'd rather add your own treatments. The arpeggiator is interesting as it automatically splits any played chords into three parts (high notes, low notes and middle) and then arpeggiates these independently. Setting octave shifts or high/mid/low settings for the arpeggiator steps is simply a matter of clicking in a grid where active cells light up as blue 'LEDs'. There's also a step display that shows whereabouts you are in the arpeggiator sequence. Polyphony can be set from four to 12 voices and the sounds on offer range from organesque to more PPGlike sounds. A further variation on this plug-in is a synth-only version with no effects and no arpeggiator. Spin-offs include the LFO as a separate modulation module and the chorus module.

All In A Spin That about wraps it up for instruments, but Mode also includes some unusual effects, the first of which is Mode Spin. This is a multi-effects unit where parameters may be sequenced in steps locked to the host tempo. Effects include a multi-mode filter, distortion and bit-crunching effect, auto-panner, volume sequencer and stereo delay. Modulating levels via a line of miniature faders allows rhythmic gating effects to be set up, but having other effects on board means the end result can be far more complex. The step size can be adjusted from 32 steps per 4/4 bar to one step per bar, while the slew rate between steps can be changed to offer hard stepping or smooth gliding. The tempo gate effect is set using a simple volume slider for each step. and of course the delay effect (which has filtering in the feedback loop) puts in an appearance to make things more interesting. This delay is also available as a separate tempo-sync'able plugin. Distortion/Bit Crunch, Spin Filter, Spin Sequencer and Spin Pan are also available as separate plug-ins if you only need to use one element.

Mode Wash It all starts to sound a bit like a launderette running in reverse at this point, as Spin is followed by Wash. Wash is certainly an oddball application of delay, and comprises six interacting delay lines controlled by a grid-like user interface. There's a Lock button on each of the delay lines which, when activated, sets the feedback to 100 percent and shuts off the input to create an infinite loop of whatever sound is trapped in the delay line at the time. This loop can be filtered, time-shifted and volume/pan controlled to create a continuous, smooth sound, so it is useful for sound design as well as performance. Each delay can be set in steps up to 10 seconds, while a delay time knob sets the delay time to a percentage of this maximum figure — much like any other coarse/fine arrangement. A resonant filter is available following each delay line with adjustable frequency, resonance and mode (low-pass, high-pass, band-pass and notch). Additional audio can be added to the loop 'sound on sound' style, and by file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (4 of 7)9/26/2005 12:04:11 AM

Cycling '74 Mode

using the matrix, you can send any output into to any of the delay inputs. The audio output can be any combination of the delay line outputs so you can get anything from subtle echoes to dense clouds of regeneration. You can also end up with a swirling mess, but that's part of the fun! Spin-offs from Wash include a single-channel looping delay module, where double-clicking the delay time indicator will enter a tap tempo rate dependent on the timing between the two clicks. This also works in the full version of Wash.

In Use Mode might not have the most attractive plug-in windows, but they are on the whole clearly set out and easy to control. The synths have a distinctively digital edge to them and provide more sonic flexibility than their meagre parameter sets might indicate, though they tend to be quite processorintensive, so it makes sense to use the lighter versions wherever you don't need all the bells and whistles. The arpeggiators are extremely useful and Poly Arpeggiator in particular can create some very complex-sounding results from relatively simple musical inputs. As well as the full versions of Bang, Mono, On the effects side, everything works well but Wash is of particular interest as Poly, Spin and Wash, the Mode package includes smaller plug-ins that are derived it allows the looping and freezing of from elements of these instruments and audio for the creation of unusual drones effects. or textures. It's not always clear what the end result will be until you try it, but experimentation is easy thanks to the visual grid interface. However, I found Spin more to my liking as I've always had a soft spot for timebase-controlled effects that allow you to sequence levels or filter settings. The effects on offer here are extensions of the rhythmic gating idea but with the option to change the rhythm and levels of the individual steps. Overall, the extra quirkiness of some of these effects is nicely balanced by the controllability of the interface while the ability to apply modulation sources to other Mode plug-ins also opens up some very file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (5 of 7)9/26/2005 12:04:11 AM

Cycling '74 Mode

creative possibilities. My only real criticism of this suite of goodies, other than the slightly unimaginative graphic design, is that the synths in particular are very processorhungry, so anyone not using a G5 machine will need to deploy them with care. If anything, I'd like to see even more made of the timebase effect possibilities as I really see that as a great way to add interest and rhythm to tired old sounds, but what's available here is a step ahead of what I've seen elsewhere and still very easy to use.

Using Modulators When inserted in an audio channels, modulator plug-ins such as Mode Poly LFO pass audio signals through them without changing them, but they do function as modulation sources that may be used to dynamically modulate the parameters of other Mode plug-ins. For example, you could insert Poly LFO in an audio track, then use it to control a parameter in a Poly Chorus plug-in inserted in the same channel. In this example, the Mode Poly LFO Edit window would allow you to select a destination parameter to control — for instance, the chorus plug-in's depth setting. Once set, you can confirm that all is working properly by looking at the chorus window controls, where you should see the depth control moving. You can also adjust the ratio of LFO value to parameter change to adjust the depth of the modulation effect.

Summing Up Mode is a useful blend of the familiar and the quirky, where anything too confusing or unnecessary has been stripped away to keep operation simple. The synths have a sound all their own rather than being anybody's clone, and while their thirst for CPU resources worries me a bit, the provision of 'light' versions goes some way to lessening this concern. My first impression was that the synths are probably best suited to dance music production because of their hard, slightly digital edge, but they can be coaxed to work outside this genre without too much effort. The same is true of Bang, which is great for creating electronic-sounding drum or percussion sounds but also handy for creating abstract sound effects. On the effects front, the various permutations and derivatives of the Spin and Wash modules are capable of producing musically useful but artistically different sounds, and as I've already hinted, there are some good tools here for sound design as well as straight ahead music performance. Ultimately then, Mode is a file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (6 of 7)9/26/2005 12:04:11 AM

Cycling '74 Mode

mixed bag and intentionally so, but if you're fed up with me-too sounds, Mode might be all it takes to dig you out of your rut and set you thinking in different directions. Published in SOS September 2004



file:///H|/SOS%2004-09/Cycling%20%2774%20Mode.htm (7 of 7)9/26/2005 12:04:11 AM

Edirol FA101

In this article:

The Box System Requirements Installation Getting Going Practical Performance FA101 Brief Specifications Parting Thoughts

Edirol FA101 Firewire Audio & MIDI Interface (PC/Mac) Published in SOS September 2004 Print article : Close window

Reviews : Computer Recording System

Edirol FA101 £429 pros Good sound quality. Capable low-latency performance with WDM or ASIO drivers. Versatile analogue connectivity. Supports six-channel, fullduplex 192kHz operation.

cons No custom monitor mixing software is provided, so a host application that supports ASIO 2.0 Direct Monitoring is required for more than basic zero-latency hardware monitoring. Changing the sample rate and buffer size settings requires restarting the device. No support for legacy versions of Windows or Mac OS.

conclusion A few omissions notwithstanding, Edirol's first foray in Firewire audio is a very successful one. If you need a lot of analogue connectivity in a small and hotpluggable package, the FA101 makes an excellent choice, with good sound quality and solid low-latency performance.

information £429 including VAT. Edirol Europe +44 (0)20

Edirol's first foray into Firewire audio is a compact 10in, 10-out device with microphone preamps, S/PDIF and MIDI. Mike Bryant

When Edirol released the world's first professional USB 2.0 audio interface late last year in the form of the UA1000, I remember thinking that it wouldn't be long until this newer, Photos: Mark Ewing higher-bandwidth format found its way down to the more affordable end of their range. That may still happen, but for the moment at least, the company have gone with the majority consensus and produced a Firewire device: the FA101. Numerous reasons could account for this decision, but perhaps the most obvious clue might be that whilst the UA1000 still does not work on the Mac platform, the FA101 comes with native OS X Core Audio support from the word go. Users of either platform will need a reasonably up-to-date system, however: versions of Windows prior to XP are not supported, whereas Mac users need to be running OS 10.3.4, or 10.3.3 with the Firewire audio update from Apple.

The Box The FA101 boasts 10 inputs and 10 outputs, crammed with little space to spare into an attractive, metallic red half-rack casing. With a front-panel template clearly derived from Edirol's UA5 USB 1.1 interface, among the first things that come to the attention are the twin Neutrik combo sockets for the microphone preamps on inputs 1 and 2. These can supply 40dB of gain along with +48V phantom power if required, and the second input is also equipped with a highimpedance option for DI'ing electric instruments. Three further analogue input

file:///H|/SOS%2004-09/Edirol%20FA101.htm (1 of 6)9/26/2005 12:04:15 AM

Edirol FA101

8747 5949. +44 (0)20 8747 5948. Click here to email www.edirol.co.uk

Test Spec FA101 driver version 1.00. IBM Thinkpad T30 with 1.6GHz Pentium 4-M and 512MB RAM, running Windows XP Professional with Service Pack 1a. Tested with Steinberg Cubase SX 2.02, Wavelab 4.01b, Propellerhead Reason 2.5 and Cakewalk Sonar 3.1.1.

pairs with balanced jack connectors are present on the unit's rear, with channels 7 and 8 providing adjustable input sensitivity between -10 and +4 dBu via a small knob placed alongside. All eight analogue outputs are also provided on balanced jacks with nominal levels of +4dBu. Further items of interest include the usual MIDI I/O pair, a headphone socket and associated volume control, two Firewire ports, and a small raft of controls related to monitoring and clocking, which I'll come to shortly. Metering for both the combo jack inputs and the main output pair is served by four LEDs — three green and a red clip indicator — below which lurk a pair of Toslink optical connectors for S/PDIF I/O on channels 9/10. As befits something with the potential to be dragged from the confines of the studio, the FA101 is capable of being powered via the Firewire buss when connected to computers with a suitable six-pin port. Many PC laptops ship with the four-pin variety — also known as iLink or DVLink — and will require the unit be powered with the supplied AC adaptor brick. It's also worth noting that PCMCIA Firewire adaptors, such as the one I was using for this review, are not able to supply buss power despite generally employing six-pin connectors. Either way, one-metre-long cables of both types are supplied with the FA101, so you won't get stuck with the wrong one.

System Requirements Mac Processor: 800MHz G3, 700MHz G4 or better; 1.2GHz G4 or better recommended for 192kHz operation. RAM: 256MB or more. Operating system: Mac OS 10.3.4, or 10.3.3 with Firewire update.

PC Processor: 900MHz or better Pentium, Celeron or Pentium-compatible processor; 2GHz or better Pentium 4 recommended for 192kHz. RAM: 256MB or more. Operating system: Microsoft Windows XP Home Edition or Professional.

Installation The FA101's installation procedure isn't the most elegant I've seen, but Edirol have done a good job in making it pretty foolproof for the user. They also seem unduly concerned that we temporarily configure Windows XP's hardware settings to ignore the installation of 'unsigned' drivers, perhaps because seeing Microsoft's apocalyptic warning message more than once might perturb users not already accustomed to ignoring it.


Edirol FA101

After running the driver setup program (the supplied version 1.00 was the most recent at the time of writing) you're presented with a checklist of steps to run through, starting with plugging in and powering up the device. Windows then detects and (hopefully) installs the driver components FA101 'Sub 1' and 'Sub 2', with the 'Install Automatically' option taking care of things at the New Hardware Found prompt. You can then dismiss the setup program, after which the prompt appears for a third and final time, to install the FA101 proper. It sounds a little long-winded, but there's not much the user can do wrong, providing nothing untoward happens during the process. Once installed, the FA101 makes its presence felt in a pretty minimal way, with a single entry appearing in Windows' System Hardware applet, and a new link added to the Control Panel for opening the device's somewhat sparse driver settings dialogue. The 10 input and output channels appear straightforwardly as stereo pairs 1 to 5, and the unit doesn't bother making itself the default playback device for system sounds and standard Windows applications.

Getting Going As with several of the company's other Audio Capture products, Edirol have taken a 'hard-wired' approach to the FA101's digital clocking, with a sample-rate selector knob and external sync button adorning its front panel. This approach has its down sides, chiefly the fact that it's not possible for software applications like Cubase or Sonar to automatically switch the unit's sample rate as required on the fly. Instead you must quit all your software applications, set the knob to the desired position, and cycle the unit's power using the tiny switch located on its rear. Rebooting the FA101 is also necessary each time you change the buffer size setting, so if this is something you're likely to do on a regular basis you might want to think twice about shoving it in a rack. Another little inconvenience for impatient types is that this procedure tends to make Windows XP rather unresponsive while it reinitialises the FA101's drivers — on my PC this took about a minute or so before things could get going again. The FA101 caters for five sample-rate settings between 44.1kHz and 192kHz, with unrestricted 10-in, 10-out performance available at up to 24-bit, 96kHz. Unlike Edirol's UA1000, which offers only a single stereo output at 192kHz, the FA101 provides six-in, sixout capability, with the additional caveats that you lose the output level metering and input monitoring on all but the first two channels. I suspect that not too many people will use the FA101 at this super-high sampling rate, but it's always good to have compatibility with


Edirol FA101

those just-around-the-corner formats we've been hearing about for years now.

The FA101's software side is pretty minimal, with just a simple driver settings panel for adjusting the ASIO and WDM buffer size.

One commonplace software component Edirol haven't provided is a utility for controlling the FA101's onboard monitor mixer, though users on the Windows platform can take advantage of the 'Soft Ctrl' feature, which enables applications that support ASIO 2.0 Direct Monitoring to control the level and pan of each monitor signal independently before they are fed to outputs 1/2 and the headphones. Routing monitor signals to busses other than the main stereo output can't be done, however. With the Soft Ctrl function disabled, rough-and-ready monitoring is possible with each channel either hard-panned or in mono, with a single Mix knob determining the overall level. The Mix knob is actually a rather unusual design, with the left side of the centre detent controlling the level of the input monitor signal and the right side that of the main output, making it easier to achieve a good relative balance without having to do any fiddling about with faders in software. Overall, whilst the monitoring facilities allow reasonable flexibility depending on the specific host application used, a dedicated monitor mixer would undoubtedly have made better use of the FA101's 10-in, 10-out potential, and I missed the ability to define a number of preset configurations for different gear setups. That said, I was using version 1.00 of the device's software, so there's still a glimmer of hope that Edirol will come up with something like the Control Panel mixer supplied with their UA1000.

Practical Performance The FA101's ASIO driver defaults to an overall latency of roughly 20 milliseconds at 44.1kHz, with equal-sized 440-sample buffers on both the input and output. Edirol's driver control panel is about as vague as can be with the specifics, providing 'larger', 'normal' and 'smaller' settings along with an unmarked fiveposition slider for the buffer size. In practice, this allows for a minimum of 104 samples each way, equating to a pretty handy 5 millisecond latency at 44.1kHz for the purpose of monitoring via your host application, or half that for soft-synth use. Performance was very respectable on my 1.6GHz Pentium 4-M laptop. With the smallest 104-sample buffer size, Cubase SX 2 provided solid performance until CPU usage drifted over about 50 percent, at which point clicks and pops would begin creeping in. This nonetheless made for reliable glitch-free tracking whilst monitoring through VST or Direct X effects, providing certain rather demanding plug-ins (like Izotope's Spectron, for example) were steered clear of. At higher latency settings, the FA101 provided ASIO performance that matched or slightly bettered my trusty Echo Layla 24 and Indigo PCMCIA interfaces in terms of CPU efficiency. file:///H|/SOS%2004-09/Edirol%20FA101.htm (4 of 6)9/26/2005 12:04:15 AM

Edirol FA101

Running Cakewalk's Sonar 3.1.1 with the FA101's WDM drivers also turned in good results. With the WDM buffer set to its smallest setting the program The FA101 scores over some of its competitors by having 10 individual analogue reported a mixing latency of just 1.1 milliseconds, and played back a simple outputs, though it lacks the sophisticated monitor mixing capabilities of Edirol's own test song with no problems other than UA1000. the anticipated huge increase in CPU consumption. For general usage, a 4 millisecond mixing latency provided a decent balance of speed and stability on my hardware — performance which helped make up for the fact that users of Sonar are stuck with the FA101's rather basic hardware-controlled monitoring capabilities unless they monitor through the computer. Breaking out the indispensable Rightmark Audio Analyser and hooking up a loopback connection on the FA101's first I/O pair produced some quite impressive results. Noise levels at 16-bit, 44.1kHz measured a steady -96.1dB(A), which at 24-bit decreased to a very respectable -102.3dB(A), tallying precisely with the figures cited by Edirol. The preamp-less inputs 3 to 8 were even better, averaging -103.9dB(A), whilst harmonic distortion of 0.004 percent and a very flat frequency response of about +0.18, -0.06 dB from 40Hz to 15kHz shouldn't draw any complaints either. In subjective listening tests, this certainly seemed to translate into audio quality that compared very favourably with my Layla 24, with a clean, open sound and good stereo imaging.

FA101 Brief Specifications Inputs 1 and 2: balanced or unbalanced XLR or TRS quarter-inch jack, nominal level -50 to -10 dBu (XLR) or -35 to +4 dBu (jack), high-impedance instrument option on input 2. Inputs 3 to 6: balanced or unbalanced TRS quarter-inch jack, nominal level +4dBu. Inputs 7 and 8: balanced or unbalanced TRS quarter-inch jack, nominal level -10 to +4 dBu. Outputs 1 to 8: balanced or unbalanced jack TRS quarter-inch jack at nominal level +4dBu, headphone socket for outputs 1/2. A-D converters: 24-bit. D-A converters: 24-bit. Digital I/O: Toslink optical type S/PDIF (channels 9/10), plus MIDI In and Out. Supported bit depths: 16, 24. Supported sample rates: 44.1, 48, 82, 96 kHz (plus 192kHz with six-channel full-duplex operation). Interface: Firewire (IEEE 1394). Power: Firewire buss or supplied AC adaptor.


Edirol FA101

Parting Thoughts Since manufacturers are just starting to provide Firewire devices at the more affordable end of the market, it's not surprising that the FA101 undercuts everything else for the number of inputs and outputs it provides. To a certain extent, it looks like Edirol have had to make some sacrifices to reach this price point, because the FA101's routing and monitoring facilities lack the flexibility generally found in more costly multi-channel interfaces. At the time of writing, the Edirol's stiffest competition looks set to come from M Audio's newly released Firewire 1814, which, though equipped with fewer analogue outputs, does include such niceties as work clock and ADAT I/O. Yet although desktop users not averse to installing PCI cards have a lot more choice available to them, the FA101 still compares well on price with a number of similarly specified PCI-format products, including Edirol's own DA2496 and MAudio's Delta 1010. Despite a couple of inconveniences, the FA101 is a tremendously capable device, and one that I'd be happy to use as my main audio interface. Versatile analogue connectivity and a very compact format make it well equipped for use outside the studio, whilst no-compromise 24-bit, 96kHz operation and good low-latency performance mean it'll be able to handle most demanding recording applications with ease. All in all, Edirol's first Firewire product is a strong contender, and one that deserves to sell extremely well. Published in SOS September 2004 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



Edirol PCR1

In this article:

The Thin Controller Let There Be Audio Complete Control Conclusions

Edirol PCR1 £229

Edirol PCR1 USB MIDI Controller Keyboard & Audio Interface Published in SOS September 2004 Print article : Close window

Reviews : MIDI Controller

pros Size and design. Excellent, unusually versatile MIDI-control possibilities. 16 memory slots. Digital optical output. Bundled Editor software. Support for Mac and Windows, in both current and legacy versions.

cons

There's been fierce competition over the past couple of years to create the ultimate portable MIDI controller for use with laptops. Edirol's PCR1 is one of the lightest and thinnest yet seen. Is it what we've all been waiting for? Simon Price

The Pitch and Mod 'buttons' are not very well suited to their respective tasks. Limited audio interface, with line-level input only. Audio performance struggled at useable latency settings. No five-pin MIDI interfacing capabilities.

It must say something about the musictechnology scene (or me) that when I visited the UK Sounds Expo show earlier this year, the two things that most interested me could be mistaken for toys! One was the Korg Legacy Collection, or specifically the diminutive controller keyboard that comes with it, and the other was the Edirol PCR1, summary which, at first glance, looked as though Innovative design makes this Photos: Mark Ewing the sexiest and most portable someone had sunk a portable of the mini keyboard controller keyboard halfway into a desk! The PCR1 is very slim (just 40mm high, controllers. The PCR1 has a including the knobs, at its thickest point), with similar dimensions to a standard minimal audio interface, and computer keyboard. In fact, it's very much a computer peripheral, designed to be lacks normal pitch and mod an input device for software synths and MIDI software, and adding audio I/O. controllers, but its size and excellent MIDI implementation This emphasis away from the traditional MIDI controller keyboard is particularly give it its own distinct appeal. evidenced by the lack of standard five-pin MIDI connections, or even a power supply. Instead, all power and communication is handled via the USB connection information to the computer. £229 including VAT. Edirol Europe +44 (0)20 8747 5949. +44 (0)20 8747 5948. Click here to email

The Thin Controller

www.edirol.co.uk

In their attempt to create a really small and portable controller, Edirol have looked

file:///H|/SOS%2004-09/Edirol%20PCR1.htm (1 of 5)9/26/2005 12:04:18 AM

Edirol PCR1

Test Spec 800MHz Apple iBook with 640MB of RAM running Mac OS 10.3.4. Propellerhead Reason, Ableton Live, Digidesign Pro Tools LE v6.4, Emagic Logic Express v6.

at every aspect of the physical design. In addition to the main housing, the keys, knobs, buttons and pitch/mod controllers have all been rethought to minimise size, and to withstand the rigours of travel. The keys feature a new mechanism that Edirol have given the highly contrived acronym SLIM, which (they claim) stands for Short-stroke Low-profile Impact Mechanism. The keys have about half the normal travel, allowing the keyboard to fit into such a small space. Despite this, the keys have standard width, and normal velocity sensitivity. The keyboard doesn't generate aftertouch, but you can set one of the top-panel knobs to stand in for this. The keys take some getting used to, and may not be for everyone, but should be an acceptable compromise for those who are not traditional keyboard players. In fact, for playing percussive instruments like drum machines, you may even prefer them. The knobs also have a low profile, but although they are small, and squeezed into a small space, they are cleverly shaped, with a small diameter that allows enough space between them for your fingers. I liked the buttons, which are a made from rubber and reminiscent of those on Sinclair's ZX Spectrum. Although you might think this gives the PCR1 a bit of a calculator-like feel, I much preferred them to typical clunky plastic buttons. The last innovation is the one that doesn't work so well for me: the concept for the pitch and mod controllers. The Edirol PCR1 does away completely with the usual joystick or wheel controllers. Instead, the pitch controller is an oval rubber pad, while the modulation controller looks like a small metal rod sunk into the PCR1's top panel. These both seemed like excellent ideas until I realised that they weren't quite what I was expecting. Each controller really just consists of Up and Down buttons, and you have to specify the speed at which they operate as part of the unit's preference settings. I really didn't get on with this method, and I don't think there's any substitute for 'real' controls for these parameters.

Let There Be Audio In addition to being a controller, the PCR1 doubles up as an audio interface, capable of 24-bit/96kHz operation. The stereo analogue inputs and outputs are on RCA phono connectors, and there's also a minijack headphone output. This side view shows just how thin the PCR1 Communication with the host computer really is — it's just 40mm at its thickest point. is taken care of by the same USB cable that the controller side uses. A wide range of host systems are supported: WDM, MME and ASIO 2.0 on Windows 98/2000/XP, ASIO/OMS on Mac OS 9, and Core Audio/Core MIDI on Mac OS X. I was testing on OS X, so after installing the drivers I went into the Mac's Audio MIDI Setup control panel, and confirmed that the unit was being 'seen' as both a MIDI controller and audio interface. At this point I jumped straight into Propellerhead's Reason to have a play. In Reason's Preferences, the PCR1 was seen as two separate MIDI devices, because the controller's USB connection is divided into two virtual ports. This provides a degree of flexibility when setting up your controller templates, allowing you to output different file:///H|/SOS%2004-09/Edirol%20PCR1.htm (2 of 5)9/26/2005 12:04:18 AM

Edirol PCR1

controls via different MIDI ports. I was also able to assign Reason's audio output to the PCR1's, but noticed straight away that the Latency slider was displaying a very high value. For a while I couldn't understand why there was a consistent overhead of 38ms latency over that of the Mac's built-in audio outputs at any given buffer size. What I hadn't noticed at this point was that the installer CD adds a specific PCR1 control panel to the system prefs, which has its own buffer setting with a very high default value. By reducing this buffer considerably, I could match the kind of latency I'm used to getting when using my Mac's own headphone output, but audio performance was considerably reduced. This was a pretty harsh compromise, so I was keen to get back to my Mac's headphone socket as soon as possible. That said, of course Apple have not seen fit to put an audio input on my iBook, so the PCR1's audio connections would come in very useful for recording on the road, or whenever a grander audio interface was not available. A limitation is that the audio inputs are only line-level inputs, so they cannot take the signal from a mic or guitar. An impressive inclusion, however, is the fact that the headphone output doubles up as an optical digital out, the same as you find on some Sony Minidisc products.

Complete Control While the audio side of the PCR1 is fairly basic, the MIDI controller role is clearly the more developed side of the product, and you can feel the Roland heritage backing it up. The manual is clear and accessible, and should get even a MIDI control novice up and running quickly. 16 memory slots are available for storing control assignment sets, and these come stocked with various pre-programmed templates. The factory templates are mostly aimed at controlling the popular sequencer packages, but you can use the memory slots to store your own settings, and set the unit up to control any MIDI-capable software or device connected to the computer. Two sets of card overlays are provided, one printed and one blank. It's a bonus that they are included, but they are a bit annoying as they don't sit securely on the panel, and get knocked around when you use the controls. Although there are eight physical knobs available, a Shift button gives access to a second bank of controls. Most of the time the unit defaults to Play mode, where all the controls transmit their current parameters, and the keyboard sends Note messages. The two other transmission modes are 'Bank Sel', and 'Prg Change', which are used for selecting sounds on the destination device. Other modes are available for changing settings, editing control assignments, and loading memory sets. Like M Audio's Oxygen/Ozone controllers, the main keyboard is used for numeric entry in these modes. As well as all the basic control possibilities, there are various other options available, such as an all notes off Panic command, and my favourite, the MIDI Clock output option. This last trick allows you to transmit Beat Clock from either of the two virtual MIDI ports, and assign one of the knobs to control the tempo. This is excellent for live situations and laptop DJ sets, and it worked perfectly in Ableton Live. Finally, there is a Snapshot command, which transmits the current position of all the controllers to the destination, so all controls become 'in sync'. Editing the functions to which the controls are assigned is quite easy, and follows file:///H|/SOS%2004-09/Edirol%20PCR1.htm (3 of 5)9/26/2005 12:04:18 AM

Edirol PCR1

the model of entering Edit mode and moving or pressing the control you wish to program. There is a comprehensive list of MIDI message types available in addition to the standard CC (Control Change). The full All the PCR1's MIDI (and audio) interfacing is list is note, aftertouch, CC, program handled via its USB connection — there are no five-pin MIDI sockets at all. The remaining change, RPN, NRPN, SysEx, and tempo change, which makes the PCR1 features on the unlabelled back panel are: (left-right) the display contrast control, the considerably more versatile than many stereo analogue ins and outs on phono of its rivals. A nice idea is that there jacks, and the mini-jack headphone output. are Basic and Advanced edit modes, which have different numbers of parameters you need to set for each controller. Buttons can be set to Latch or Toggle mode — with corresponding LED behaviour. The knobs can have their range adjusted, and can be switched between transmitting absolute values and emulating a rotary encoder. This is a clever idea that I've not seen before. To help me understand this, I set one of the knobs to this mode and assigned it to a pan pot in Ableton's Live. When you turn the knob to the right and stop, the panner in Live moves right continuously until you return the knob back to the 12 o'clock position. Moving the knob further away from the centre results in a faster movement in Live. Unfortunately, there isn't a 'pick-up' mode, where the controller only starts operating when it passes through the current value on the destination device. If hands-on control editing sounds like a lot of hard work, you can turn to the cross-platform PCR editor software. This utility displays a graphical representation of all the available hardware controls. Clicking a knob or button opens another window with drop-down menus and numerical fields for settting up the control. Once you've entered all the values, you transmit the patch to your PCR1. Simple as that. Even if you don't use the editor for setting up the PCR1, the software is the easiest way to manage all your banks, as each setup can be saved as a document for later loading.

Conclusions As you can probably surmise, I'm really impressed with the PCR1's comprehensive MIDI control abilities, and this is what you should concentrate on when considering going for this controller keyboard. The audio interface is a useful addition, especially the digital out, but is quite limited and should be looked on as a bonus to the MIDI side. The lack of mic or instrument input capability means that the PCR1 is not a 'one-stop shop' like some of its competition, but then this is reflected in the price. What the PCR1 really excels at, of course, is portability — the audio connections will be a welcome extra outside the studio, and Edirol have even thrown in a padded cloth 'gig bag' in the box. The amazing compactness, and touches like the beat clock/tempo controller, make this an obvious piece of kit to pack for a live laptop performance.


Edirol PCR1

Published in SOS September 2004 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



Emu 0404

In this article:

The Grand Tour In Use Emu 0404 Brief Specifications Cream Of The Crop

Emu 0404 PC Soundcard Published in SOS September 2004 Print article : Close window

Reviews : PC Soundcard

Emu 0404 £70 pros Amazing value for money. Incredible dynamic range for the price. Same DSP effects suite as the more expensive Emu cards.

cons Not currently compatible with Emulator X soft sampler. Emu's MME-WDM and Direct Sound drivers still don't support 96kHz operation. No GSIF driver support.

summary Emu's 0404 is absolutely stunning value for money at just £69.99, and with a dynamic range of 112dB it blows away most stereo soundcard competitors at double the price.

information £69.99 including VAT. Emu Europe +353 1 433 3201. +353 1 806 6788. www.emu.com

Test Spec

Emu have already taken the market by storm with their range of affordable, high-performance soundcards, and their new stereo PCI card offers impressive features and audio specifications at a budget price. Martin Walker

Joining the excellent 1212M, 1820 and 1820M in Emu's soundcard range, the new 0404 is also a PCI card, and features the same E-DSP chip capable of running up to 16 DSP effects simultaneously. The 0404 boasts stereo analogue inputs and outputs, plus both co-axial and optical S/PDIF digital input and output, and a MIDI input and output, for an amazingly low UK price of just £69.99. This time the analogue I/O is unbalanced, but the software bundle seems to be exactly the same as for the 1212M, consisting of Steinberg's Cubasis and Wavelab Lite, the Sound Guy's SFX Machine LT effects plug-in, and a trial version of Minnetonka's Discwelder Bronze DVD-Audio burning application, plus the same Patchmix DSP utility software as the rest of the range. Once again this product is currently for PC users only, and runs either on Windows 2000 or XP.

Emu 0404 Windows XP driver version 5.12.01.0495. Intel Pentium 4C 2.8GHz processor with hyperthreading, Asus P4P800 Deluxe motherboard with Intel 865PE chip set and 800MHz front side buss, 1GB DDR400 RAM, running Windows XP with

The Grand Tour All the 0404's circuitry, including the Burr-Brown PCM1820 ADC and AKM AK4395 DAC chips, is on a single PCI card. Both converter chips are 192kHzcapable, but only 44.1, 48 and 96 kHz sample rates are supported here, although

file:///H|/SOS%2004-09/Emu%200404.htm (1 of 4)9/26/2005 12:04:22 AM

Emu 0404

Service Pack 1.

at this price I don't see this as much of a restriction. As on the other Emu soundcards, the mysterious Xcard In and Out connectors are present for Tested with Steinberg rumoured multi-card support, as well as a Sync socket so you can attach the Cubase SX 2.2 and Wavelab 5.0, Native Instruments Pro 53. optional Sync daughterboard via an internal ribbon connector to add word clock in and out, SMPTE in and out, and MTC (MIDI timecode) out. I suspect these functions will be overkill for the majority of potential 0404 users, but it's a welcome upgrade option. All I/O is on flying breakout cables, with the audio emerging from a nine-pin Dtype connector to four quarter-inch unbalanced in-line jack sockets, and the digital I/O using a separate 15-pin D-type connector terminating in two inline phono sockets for co-axial S/DPIF, a tiny in-line box with two Toslink optical sockets, and two in-line five-pin DIN MIDI sockets. As its 0404 nomenclature suggests, both S/PDIF inputs and outputs carry the same signal, so you can record and play back up to two mono analogue and two digital signals simultaneously.

Emu's 0404 has the same multi-channel DSP mixer and effects as its more expensive cousins.

Installation of the 0404 drivers and Patchmix DSP software went without a hitch, although for any existing Emu users who are wondering, you can't use the 0404 alongside a 1212M, 1820 or 1820M, since the drivers in each case look for their specific hardware and don't currently support multiple cards. However, although the 0404 Patchmix DSP software seems tweaked to only look for the 0404 card, it's functionally identical to the version supplied with Emu's other, more expensive cards, with exactly the same selection of 32-bit DSP effects that run at either 44.1 or 48 kHz (but are disabled at 96kHz), and once again there is no hidden sample-rate conversion going on behind the scenes.

In Use Since I devoted so much space to explaining the ins and outs of the Patchmix DSP software in my extensive review of the 1820M in SOS June 2004 (available on-line to subscribers at www.soundonsound.com/ sos/jun04/articles/emu.htm) I won't repeat myself here, except to reassure new users that while it may seem impenetrable at first, loading in the various Session templates should soon make things clearer. Running my audio tests using Rightmark's Audio Analyser v5.3, the 0404 turned in a truly superb performance for a £69.99 soundcard, with a frequency response that was +0.14/-0.13dB from 20Hz to 20kHz, and total harmonic distortion of just 0.002 percent. However, it's the dynamic range measurements that are most file:///H|/SOS%2004-09/Emu%200404.htm (2 of 4)9/26/2005 12:04:22 AM

Emu 0404

remarkable: at an astonishing 112dBA, this is an amazing 16dB better than M Audio's Audiophile 2496, 12dB better than their Firewire 410, and 4dB better than Echo's Mia. Only when you compare it with Emu's own 1212M and 1820M do its results fall behind by about 6dB. Of course background noise levels aren't the whole story, and in listening tests the differences between the 0404 and my benchmark Echo Mia were extremely subtle, whereas to my ears the 1820M was noticeably more clear and focused. However, given that Echo's Mia MIDI currently sells for about £160, and apart from its balanced I/O provides roughly the same features as the 0404 minus the DSP effects, this is still an excellent result. Emu's ASIO drivers once again worked very well inside Cubase down to latencies of 4ms on my PC at 44.1kHz, and I managed a very good 10ms Play Ahead setting with NI's Pro 53 soft synth using the Direct Sound drivers, and the usual indifferent Windows XP result of 45ms with the MME ones. Also, although I read claims on a web forum that Emu's drivers aren't yet multi-client, I had no problems running the MME-WDM drivers and ASIO ones simultaneously with different applications, or allocating different pairs of ASIO drivers to multiple applications. Like the 1010 PCI card of the more expensive Emu cards, the 0404 also supports up to 32 ASIO inputs and outputs, so you can mix down the outputs from these multiple applications internally using the Patchmix DSP utility and route them all to the 0404's stereo analogue or digital I/O. On the minus side, while the ASIO drivers can be used at 44.1, 48 and 96 kHz, Emu still haven't released MME-WDM drivers for any of their range that support 96kHz sample rates, and using Gigastudio isn't an option either, since there are no GSIF drivers at present.

Emu 0404 Brief Specifications Sample rates: 44.1, 48 and 96 kHz from internal clock. Analogue inputs: two, unbalanced quarter-inch jack at -10dBV sensitivity. Analogue outputs: two, unbalanced quarter-inch jack at -10dBV level. Digital I/O: S/PDIF in and out on phono co-axial and Toslink optical, up to 24-bit/96kHz, MIDI In and Out. Frequency response: 20Hz to 20kHz, +0.2/-0.1dB. Dynamic range: 111dBA (analogue inputs), 116dBA (analogue outputs). THD + noise: -100dB (0.001%), 1kHz signal at -1dBFS.

Cream Of The Crop At this price and with this spec I suspect most other soundcard manufacturers will have shed a few tears behind the scenes when Emu released the 0404. I can't think of a single alternative at a similar price that provides remotely similar


Emu 0404

functions and performance. In fact, I suspect the 0404 may tempt some potential purchasers away from Emu's own 1212M model, which is more than double the price, since the 0404 provides exactly the same DSP effects and versatile Patchmix DSP mixer. So what compromises would there be? Well, you lose the balanced analogue inputs and outputs (which, as many musicians are finding, can help solve a lot of ground loop problems when connecting to other mains-powered gear), the Firewire port, ADAT functions and 192kHz sample-rate support. Also, while you do still have the Sync board as an option, for the time being at least the excellent Emulator X soft sampler isn't compatible with the 0404, and without GSIF driver support you can't use Gigastudio either, leaving Halion, Mach V or Kontakt as the only real softsampling options. However, as long as you're happy with unbalanced analogue I/ O and don't need more than two analogue ins and outs, there simply isn't anything available to my knowledge that provides such professional-sounding audio at this price. Emu have done it again! Published in SOS September 2004 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



Hot new Sample CDs on test

In this article:

PCP80 Electric Grand ***** Solo Strings ***** Tool Shed Percussion ***** Woodwind Collection Expanded Edition ****

Hot new Sample CDs on test Sample Shop Published in SOS September 2004 Print article : Close window

Reviews : Sound/Song Library

PCP80 Electric Grand ***** MULTI-FORMAT First manufactured in 1976, Yamaha's CP70 and CP80 electric grand pianos were a practical solution to the problem of using acoustic piano in a live rock group. By putting individual pickups on each note, Yamaha created a hybrid instrument that sounded pretty good when played acoustically, but could also be cranked up through a PA with no fear of feedback. While the CP-series pianos' somewhat cantankerous amplified sound wasn't to everyone's taste, their sturdy, biting, instantly recognisable tone went down well with discerning musicians — Peter Gabriel, Eddie Jobson, and George Duke learned to love the piano, and many keyboardists (myself included) continue to use a CP70 as a writing tool. Composer/sound designer Akihito Okawa obviously shares in the nostalgia for this vintage keyboard, as he has lovingly sampled a CP80 in all its 88-note glory. The specs make impressive reading: Prominy's multi-format release comprises 24-bit/96kHz EXS24, Kontakt, and Halion versions with up to 20(!) velocity layers, plus a 16-bit/44.1kHz, 16-layer Gigastudio version comprising 2GB of samples. The two-DVD package also offers some lower-sampling-rate and 'lite' options. Loading the 16-velocity Gigastudio instrument takes a while, but that's understandable given the number of samples involved — most of the 88 notes use 16 dynamic layers plus a further 16 release triggers, resulting in a 256MB program containing nearly 2600 samples! The musical effect is superb — having so many velocity splits makes transitions between dynamic layers totally undetectable. The CP80's range is perfectly replicated, from tiny, delicate tinklings to aggressively poky loud notes, the latter just the ticket for a high-decibel belt through 'Hit Me With Your Rhythm Stick'! The Gigastudio version offers only three programs: the 16-layer full Monty, a RAM-conserving eight-velocity version, and a small 'pedal-on' layer, designed to simulate the ringing effect introduced by the sustain pedal. (I found this had a file:///H|/SOS%2004-09/Hot%20new%20Sample%20CDs%20on%20test.htm (1 of 6)9/26/2005 12:04:25 AM


negligible effect on the sound.) Due to programming limitations, the piano's builtin mono tremolo effect can only be enjoyed by Halion users. From a pianist's point of view, performing with these samples feels just like playing the real thing, with none of the depressing artefacts one associates with sampled pianos. The multi-dynamic samples respond admirably to expressive and improvisational playing, and the provision of extremely quiet notes encourages a subtle touch. It takes a brave piano tuner to deal with the CP80's wayward, occasionally unmatched upper partials, but this piano has been brought bang into tune by an immaculate, expert tuning job. Certain endearing idiosyncrasies (the clanky, slightly reedy attack and thunky bass notes) are still in evidence, but they're part of the charm! Once again, a fine, endangered keyboard has been rescued from extinction by a fanatically detailed and diligent sampling job. Dave Stewart EXS24, Gigastudio, Halion, and Kontakt 2-DVD-ROM set, $199 (around £108). Prominy +1 949 451 1753. +1 949 451 1753. Click here to email www.prominy.com

Solo Strings ***** GIGASTUDIO/EXS24 MKII VSL's Pro Edition covers most angles of the orchestra, but failed to include solo viola and double bass. The Viennese completists will no doubt rectify this in a future update — meanwhile, anyone who wants to put together a sampled string quartet without first remortgaging their house should be aware of Solo Strings, which, in true Christmas spirit, brings the four members of the strings family together under one roof. The solo violin and solo cello samples on Solo Strings are identical to those in Strings and Performance Set Pro Editions, reviewed in SOS March 2004. Back then, I singled out those two instruments for special praise, and can only repeat that their samples are absolutely stunning. Solo Strings' viola closely matches the solo violin's array of musical styles, the exceptions being that the viola plays no spiccatos, grace notes, runs, or glissandi. Having suffered my fair share of tired, uninspiring solo strings samples, I'm pleased to report that this solo viola is the best I've heard. The detailed, noisefree recording captures the instrument's rich, vibrant, autumnal tone in all its glory, and release triggers add a luxurious extra dimension. The player's subtle vibrato imparts emotion without going over the top, but for all you slaves of passion there's an expressive 'strong vibrato' option.

file:///H|/SOS%2004-09/Hot%20new%20Sample%20CDs%20on%20test.htm (2 of 6)9/26/2005 12:04:25 AM


Also played with vibrato, the double bass pizzicatos might not be everyone's first choice for a jazz walking bass part, but their big, plump sound should be weighty enough for most applications. The arco samples pack power — there are some fierce bow attacks hiding amongst the diminuendos, and the sustains can comfortably handle all the subtleties of melodies, chords, and bass lines. This stately, versatile double bass perfectly complements the other three solo strings in a quartet setting. Like the violin and cello, the viola and double bass work through a large number of stylistic variations (details of which appear in the March 2004 review), the only notable omission being harmonics. Things get even better when we move on to the performance samples — the viola's 'performance legatos' offer loud, portamento, accented, and quiet options, all of which sound convincingly organic when played under the control of the VSL Performance Tool's Horizon Legato mode. The double bass performance legatos work well too, though for some reason the loud version utilises portamento on some intervals. Altogether, the four solo instruments supply more varieties of note repetition than you can shake a stick at. Solo Strings' huge size (over 30GB) should give a good indication of its musical scope. In effect, it costs around £10 per gigabyte in the UK — when you consider that 512MB Akai ROMs used to retail for £200, £296 seems a very fair deal for a library containing arguably the best collection of solo string samples in the world. Dave Stewart EXS24 MkII and Gigastudio 5-DVD-ROM set, £296 including VAT. Time & Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.viennasymphoniclibrary.com

Tool Shed Percussion ***** AUDIO+ACID/REX As I'm sure I've said before, you can never have enough drum sounds, but sometimes getting new or original samples to play with can be tough. The nice people at Big Fish Audio have decided to help by putting together this library, which mixes some standard (and not so standard!) drum kits, standard percussion, and 'found' percussion into loops and hits. In total there are 49 construction kits, as well as an additional group of over 500 single hits. Each construction kit follows the standard format of a blast of the full drum loop, file:///H|/SOS%2004-09/Hot%20new%20Sample%20CDs%20on%20test.htm (3 of 6)9/26/2005 12:04:25 AM


followed by the individual parts. The kits are a great mixture of drums, odd/exotic percussion, and bits of junk or tools. For example, the Roller kit contains a Gretsch bass drum, a metal pipe, a parade snare, sandpaper blocks, maracas, a selection of toms, a wooden block, and a slap on someone's 'large round belly'! Other sounds that sit on the stranger end of the scale include saw blades, buckets (both full and empty), tape measures, and a metal leaf rake. With such a mismatch of sounds, there is a danger that the end results could be a complete din, but the full mixes are actually really good — with a little editing these would work well as percussion beds for most musical genres, especially for soundtrack work. Tempos cover 50-170bpm, and there's also a nice mixture of time signatures, but for me it's the component hits, shakes, and crashes that make Tool Shed Percussion really interesting. These individual hits and loops can be chopped up and slotted into just about any type of rhythm track. Containing both vintage 'normal' kit, some interesting percussion, and other 'found' instruments like crisp packets, suitcases, and paint rollers, these samples can be taken as is or processed to give some some really original-sounding percussive sounds. I must also mention the excellent included booklet, which is clear, uncluttered, and full of information. Along with the track and bpm listings, it includes detailed information on all the drums and strange percussion used. Size, construction, type of beater, and age of kit are all listed. This is a real plus point that helps to underline the thought and care that the producers, Kent Carter and Pat Campbell, have put into this collection. Other sample CD companies please take note! On the downside, a few of the samples are quite noisy, with some unpleasant audio side effects thrown in, although that's hardly unexpected when thumping merry hell out of a huge oil tank outdoors! Also, perhaps there are slightly too many standard drum kit sounds for a sample CD with this title, but these really are only very minor niggles that fail to put a dampener on what is a high-quality and well-thought-out collection of truly original percussion sounds that can't fail to inspire and add some much-needed interest to the dullest of rhythm tracks. Oli Bell Audio CD + Acidised WAV/REX CD-ROM 4-CD set, £56.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.bigfishaudio.com

Woodwind Collection Expanded Edition **** GIGASTUDIO US composer Tim Smith set up Westgate Studios to distribute his custom sound file:///H|/SOS%2004-09/Hot%20new%20Sample%20CDs%20on%20test.htm (4 of 6)9/26/2005 12:04:25 AM


libraries. Tim's Woodwind Collection started life as a single CD, but with the addition of new instruments and articulations, it grew into a 1.52GB Gigastudio two-CD set. The expanded edition includes all the first version's samples, and the two editions are separately available — for a comparison of their contents, check out his web site. A quick listen to the oboe and English horn gives a good insight into the library's character — both have a bright, clear tone, but, being dryly recorded in mono with no release triggers, they lack the sumptuous depth of more expensivelyproduced libraries. However, slap on a little reverb and you're in business; the instruments' vibrato sustains have a lovely lyrical quality, their positive attack and incisive timbre being ideal for poignant, expressive lead lines. Moving on, I also enjoyed the sweet, pastoral tone of the flute, and its marcato short notes sounded excellent in chordal passages. The flute and piccolo both play a selection of fast upward runs spanning an interval of a fourth — a useful extra, as such musical flourishes usually sound bogus when programmed with single notes. Sadly, there's no alto flute — Tim explains that this was omitted because no-one asked for it. D'oh! I've put in my request, but its probably too late now... The whole clarinet family — 'E' flat, 'B' flat, and bass — is here, each instrument sounding bright and effective, with nice snappy staccatos and rock-steady sustains. Combining jocularity with a sad lyricism, the bassoon also sounds in the best of health, and a rich-toned, powerful contrabassoon provides some profoundly deep support. Great stuff, but there's more: the instrumentation here extends beyond the orchestral, supplying a recorder family (sopranino, two types of soprano, and alto), an ineffably sweet sound nowadays not much heard outside of historical documentaries. The library also has a tin whistle and a penny whistle (or flageolet), the latter contributing bends, mordents, and grace notes, very handy for programming Irish folk tunes. Overall, I was impressed by all the instruments' accurate tuning and controlled delivery, especially evident in their beautifully precise trills. This well-organised, comprehensive package offers fifteen instruments, a wellchosen selection of performance styles and up to four dynamic layers. None of the long notes are looped, but with sustains lasting around nine seconds that's not the end of the world. this library proves you don't need a million-dollar budget to create a good sample library, just good players, a decent recording setup, and the musical experience to know which articulations will work when sampled! If you're looking for reasonably-priced woodwind samples to supplement your favourite orchestral brass and strings, this musically solid set comes highly recommended. Dave Stewart Gigastudio 2-CD-ROM set, $199 (around £108). Westgate Studios +1 817 281 6212. Click here to email www.westgatestudios.com

Star Fawlty Towers Signs ***** Flowery Twats



**** Farty Towels *** Fatty Owls ** Watery Fowls * Flay Otters Published in SOS September 2004 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



Line 6 Variax Acoustic 700

In this article:

Acoustic Guitar Modelling Connecting Up The Models Variax On-body Controls So How Does It Sound? Variax Verdict

Line 6 Variax Acoustic 700 Modelling Guitar Published in SOS September 2004 Print article : Close window

Reviews : Modelling Guitar

Line 6 Variax Acoustic 700 £899 pros Virtually immune to acoustic feedback. You can switch instantly between 16 different guitar and instrument models. Controls straightforward.

Extending the modelling approach they pioneered with the original Variax, Line 6 have now taken a shot at reproducing the sounds of different miked acoustic guitars. Almost completely immune to feedback, the Variax Acoustic will clearly be a hit in some live applications, but does it sound good enough to record?

cons Sound needs external reverb to sound convincing. Internal compressor isn't as flattering as it could be. Storing patches means overwriting the factory patches.

summary Like all new Line 6 products, the results can be impressive. However, the company has shown in other areas that they can almost certainly make further improvements given time. In its current form, I feel the Variax Acoustic 700 has the most to offer in the area of live sound, where its immunity to feedback is a huge bonus.

information £899 including VAT. Line 6 Europe +44 (0) 1788 821600. +44 (0)1788 821601. Click here to email www.line6.com

Paul White

Few people will need reminding that Line 6 were the company who brought guitar modelling into the mainstream market with the Pod, even though, to my knowledge, Roland were the first to test the waters in this area with their more complex V-Guitar systems. There was a lot of speculation about areas into which the company might diversify once it had grown sufficiently, and grow it certainly did, but rather than trying to cover all aspects of the music market as some companies do, Line 6 stuck to the guitar market, moving first into amplifiers and guitar effects, then most recently into producing their own guitars that use integral modelling technology to emulate a number of Photos: Mike Cameron classic instruments. Once again, Roland offered guitar and pickup modelling, but the Line 6 approach is very different. Instead of creating a special pickup to be bolted to your guitar, Line 6 provide a range of guitars of their own, with everything built into the guitar body and operated by simple, guitarist-friendly controls. The necessary piezo pickups are built into the bridge, one for each string, and the end result is very simple to use, which counts for a lot when you're on stage with no time to mess around. The first of these Variax models was based on a solid-body guitar (which is

file:///H|/SOS%2004-09/Line%206%20Variax%20Acoustic%20700.htm (1 of 8)9/26/2005 12:04:30 AM


visually distinctive because of its lack of visible pickups) and most of the sonic emulations were either of solid-body or semi-acoustic models, though there were also 12-string, banjo, and sitar models and the odd acoustic guitar. Soon a version with a tremolo followed, but at Winter NAMM in 2004 we had a sneak preview of the Variax Acoustic, the production version of which is reviewed here.

Acoustic Guitar Modelling Like the electric models, the Acoustic Variax 700 is a complete guitar, rather than a system relying on add-on pickups. It is a shallow-body instrument with the look and feel of a steel-strung acoustic guitar, but it's designed to have less natural body resonance so as to help avoid feedback problems in live performance. The design brief was to produce an instrument that not only offered a choice of sounds, but that could also be recorded or DI'd into a PA system to produce better results than a conventional acoustic guitar fitted with pickups. Indeed, the aim seems to have been to get as close to the miked sound of the real thing as possible. The guitar has an attractive naturalfinish cedar top with a fixed mahogany neck and a more-or-less-solid mahogany body which has chambers machined into it rather than using a conventional acoustic construction. Looking through the soundhole shows that the cavity beneath it isn't much bigger than the soundhole itself. I can't say what other cavities there are without sawing the guitar in half, and I don't think Line 6 would appreciate that! A headstock reminiscent of the solid-body model houses acoustic-style tuning machines, and a cutaway The DSP engine room of the Variax Acoustic provides good access to the upper is hidden beneath a plate in the back of the guitar. frets of the 24-fret, 25.5-inch-scale neck, which has a 17-inch radius fingerboard and cross-shaped marker inlays. The setup of the guitar was fairly good straight out of the box, though, as is so often the case with new guitars, the nut slots could do with being cut deeper to really optimise the action. Variax Acoustic 700 comes with its own DI/power-supply box, which provides both quarter-inch jack and balanced XLR outputs that can be fed into an acoustic guitar combo, a PA mixer, or a recording system. The guitar may also be powered by onboard batteries or directly from a Line 6 Vetta II amplifier (or other Line 6 products that have a Variax socket). A custom, heavy-duty gig bag is included.



As with the solid-body models, the string sound is picked up via a piezo pickup beneath each bridge saddle, after which it is subjected to a modelling process to make it sound like one of a number of classic instruments that can be selected by the user. The modelling process even includes a Mic Position slider, so that you can adjust the guitar tone by moving the virtual mic closer to or further from the soundhole. Because acoustic guitars are often compressed during recording, a simple onboard compressor is also included and, because of the proclivity of acoustic guitar players for using odd open tunings, a selection of these are available which use pitch-shifting technology to allow the player to switch tunings without retuning the instrument. Custom tunings can also be set up by the user and the same pitch-shifting technology is used to generate the twelve-string sound, where the top two strings are correctly tuned in unison and the remaining four in octaves.

Connecting Up While the easiest way to use the Variax Acoustic is to connect it directly to a mixer or amplifier using a standard guitar cable, this means running the guitar from batteries, and a set of alkaline batteries only lasts 10-12 hours. There's also the facility to use a standard 9V battery in emergencies, but this will only run the guitar for an hour or two — maybe just enough to get you through the night. A cheaper way to run is to use the included XPS power-supply/DI box, which connects to the guitar via a TRS jack twin-core cable. A standard The output socket can feed an amp directly if mono guitar cable then connects the you're running the guitar from batteries. XPS to your amp or mixer. The box Alternatively you can power the guitar from also has an XLR output that the included footpedal/PSU by connecting to provides a balanced feed at +4dBu it using a TRS cable, or from a Vetta amp by line level, and the unit can also connecting to it using the Ethernet-style socket (hidden by its dust cover in this function as either an A/B selector picture). (for sending different sounds to different amps via the jack and XLR output connectors) or as a standby switch. A further RJ45 connector is fitted to the guitar body and allows the guitar to be connected directly to those Line 6 amps and other products that have a dedicated Variax input. Via a standard Ethernet cable, such amplifiers can also provide power directly to the guitar. I believe the same socket can also be used to install Variax software updates.

The Models



With an instrument like this, which seems to be designed predominantly with live performance in mind, simplicity of use is paramount. To switch between the various guitar models, a rotary knob is built into the top edge of the body, and short versions of the instrument names are printed onto the knob so that it's easy to see what you've selected. After selection, the sound can be modified or additional functions activated by means of just three sliders. In all, there are 16 instrument models: the majority are steel-strung acoustics, including resonator guitars, plus there's a nylon-string classical model and a few rather more obscure instruments thrown in for good measure. As with all the other Line 6 modelling products, the Variax Acoustic 700 is the result of extensive listening tests using selected acoustic instruments. According to their web site, the models Line 6 used for their research were: 1941 Martin 517, 1946 Martin 00018, 1960 Martin D28, 1954 Gibson J45, 1951 Gibson SJ200, 1933 Selmer Maccaferri, 1951 D'Angelico New Yorker, 1958 Manuel Velazquez classical guitar, 1973 Guild F412, 1935 Stella Auditorium 12-string, 1939 National Reso-Phonic Style 'O', 1937 Dobro Model 27, Gibson Mastertone banjo, Mandola, Japanese shamisen, and Indian sitar. The models were based on the sounds of these instruments, though as always there's a disclaimer that informs the users that there is no connection between any of these companies and Line 6, and that all trademarks are respected. Unfortunately, no manual was provided with the review sample, so all the information had to be gleaned from the Line 6 web site. A number of products are already on the market that use modelling to improve the DI sound from existing acoustic guitars fitted with piezo or magnetic pickup systems, some of which sound very good indeed. Pickup systems tend to change the sound of the instrument, as they don't capture the full contribution of the body resonance, but, provided that you The included PSU pedal powers the guitar know what kind of signal the pickup is via a TRS cable, and provides an producing, modelling can be used to re- unbalanced guitar feed on a further jack synthesize the missing body socket. There's also an XLR socket which resonance and also to EQ the overall offers a balanced DI signal. sound to better match what a microphone would hear. For the results to be anything more than an approximation, you'd need to have measurements made on your own instrument both from the pickup and via a microphone, so that a corrective EQ curve could be set up (using many frequency bands). Body resonance models based on other instruments can then be modelled to give your instrument a greater range of sounds, again provided that the resonance characteristics of your own instrument are known so that the modelling process creates only the difference between your guitar and the sound you're aiming for. Most add-on systems assume a generic model for the guitar being used, so the



modelled results can only be approximate, though I believe that one manufacturer already offers a customisation facility to those prepared to pay for it. Clearly, the benefit of the Line 6 approach is that every Variax Acoustic 700 guitar will sound ostensibly the same (via its pickups) so the modelled results can be predicted and optimised with much greater accuracy and consistency than if you used your own guitar. Furthermore, because the body doesn't resonate to nearly the same extent as a real acoustic guitar, feedback problems are virtually eliminated.

Variax On-body Controls Moving the volume slider towards the model-selection knob turns it up. The Mic Position slider is closest to the front of the guitar, and it moves the imaginary mic closer to the soundhole as you move the physical slider towards the soundhole. This even works for those modelled instruments that don't have a soundhole. In effect, when the slider is moved towards the Model Select knob, the strings and overall brightness are emphasized, whereas moving towards the soundhole emphasises lower frequencies. The slider closest to the back of the guitar changes the amount of compression — all the compression parameters are optimised for acoustic guitar, leaving you just to set how much compression you want to use. Moving the slider toward the Model Select knob adds more compression. Although it looks simple enough, the Model Select knob actually performs different functions, the most obvious of which is switching between instrument models. A little illumination is provided by a green LED, so you can select sounds on a dark stage with some degree of certainty. The same knob also selects the Alternate Tunings function, as it incorporates a push switch for changing its mode, whereupon the LED changes from green to red. A different alternative tuning is preset for each guitar model, and tunings include drop 'D', open chord, and dropped octaves on the lower strings. However, your guitar must be conventionally tuned for these to work properly, as all the shifted pitches are referenced to standard guitar tuning. To set up your own custom tuning, you can retune any of the strings up by as much as seven semitones or down by as much as an octave to create your own alternate tuning. To get into this mode, you have to press the Select knob twice, rather like double-clicking a mouse button. Once in this mode, you select the string you want to tune using the compressor slider. A nice touch is that the selected string is louder than all the others, making it easy to know which one is currently active for editing. The Mic Position slider then selects the degree of shift for each string in turn. Once all the strings have been set, you can go back to the normal playing mode using your new alternate tuning.



Of course you need to have a way to save your customised instruments, and the way this works is that you have to overwrite a factory patch after modifying it — there are no user memory slots. Once all the tweaks have been made, you hold down the Model Select knob until the LED stays solidly lit. Your patch is then saved.

So How Does It Sound? When I first plugged the Variax Acoustic into my studio monitoring system and played, I was somewhat disappointed, as I thought the sound was rather hard and unyielding, almost like a guitar that had been miked far too close or had been recorded using a contact mic on the body. However, the direct sound from the strings can be confusing when heard in conjunction with the modelled sound, so I recorded a few bars of guitar using each of the classic acoustic sounds and then listened to them played back without me twanging along. The impression was slightly better, but not much. Then it occurred to me that the modelling didn't include any room ambience, so what I was hearing was more like an acoustic guitar recorded in an anechoic chamber, so I patched in Emagic Space Designer and added some natural room ambience. Immediately the result improved significantly, though I still got the impression that the dynamics of the instrument weren't quite right and the sustain characteristics didn't match the real thing. What's more, adding more Variax compression didn't improve the sound in the way I expected it to. As an experiment, I turned off the Variax compressor, and instead used Logic's own plug-in compressor with an attack time of 3ms and with Peak rather than RMS sensing. No more than 6dB gain reduction was applied on signal peaks. The improvement this time was very noticeable, with the sustain improving and the sound becoming more even and less honky — ever since hearing the banjo preset, I felt I could hear a hint of banjo in all the other presets too! Switching back to the Variax compressor confirmed that I couldn't achieve the same result using that. Although this test was very subjective, I couldn't help feel that the raw Variax sounds were rather harsh without the Emagic compressor.

The three sliders to the right of the Model Select knob control the position of the 'virtual' mic (top), the output volume (centre), and the amount of compression (bottom).

Putting this into perspective, the sound of the Variax Acoustic is still more authentic than simply using an acoustic with a piezo bridge pickup, and the resistance to acoustic feedback is undeniable, but unless you add your own



compression and reverb/ambience, you probably won't want to record with it. Checking out the other models showed the resonator guitars to be pretty convincing, and though the banjo probably wouldn't stand up to close scrutiny, it would certainly pass muster in a mix. The same goes for the sitar and shamisen — you're unlikely to hear Ravi Shankar playing a Variax, but if you want to add a bit of George Harrison nostalgia to a track, it should work fine. I was impressed with the quality of pitch-shifting used to generate the 12-string and custom tuning sounds — the fact that each string is a single-note source has obviously enabled the algorithms to be optimised — and the ability to move the 'virtual' mic position is interesting as it enables you to get closer to the sound you want before resorting to EQ. The custom tunings can feel odd if you aren't monitoring loudly enough to hide the acoustic sound of the guitar, but that's also true of any other guitar system that achieves retuning using pitch-shifting.

Variax Verdict Inevitably some compromises have to be made to keep the operating system simple, but I'd like to be able to apply any preset alternative tuning to any instrument, and I'd also like to be able to save more user tunings and patches without having to overwrite the factory sounds. Also, given that some reverb or artificial ambience is needed to make the sound breathe, it would have been useful to include this — most of the stand-alone acoustic guitar modelling preamps do. Indeed, if you don't need accurate modelling and aren't troubled by acoustic feedback, the subjective results obtainable from these external preamps used with a conventional guitar (fitted with suitable pickups) sound more musically acceptable to my ears than the untreated Variax sound. The Variax Acoustic may need a little tweaking to get the best out of it, but currently it provides the only practical way to get an acoustic guitar sound live at high volumes without feedback, and of course you can switch between sounds very quickly. While I wouldn't use the instrument for recording solo parts, it works well in a mix provided that you add some external compression rather than relying on the onboard compressor, and I really like the resonator-guitar and mandola presets. Live, it should be no problem as part of a band, though some players may not feel comfortable using it for solo performances. As with all firsts, there is bound to be room for improvement, but because the instrument is software upgradable this doesn't necessarily mean waiting for the next model. The Variax Acoustic 700 is a very clever piece of technology that meets a real need — a bit of fine-tuning is a small price to pay for freedom from feedback and the ability to switch guitars on a whim. Published in SOS September 2004






M Audio Octane

In this article:

Flexible Front-panel Design What Is M&S Miking? Connections High-octane Performance?

M Audio Octane £449

M Audio Octane Mic Preamp & A-D Converter Published in SOS September 2004 Print article : Close window

Reviews : Preamp

pros Sensibly priced in the UK. Clean-sounding mic preamps. Analogue outputs/inserts. Includes M&S decoding and instrument inputs.

This stylish new eight-channel preamp and converter offers front-panel instrument inputs and M&S stereo decoding.

cons Mic amp gain limited to 50dB, and mainly controlled by the top few degrees of movement of the rotary control.

summary A good-sounding multichannel mic preamp with some useful added features.

information £449 including VAT. M Audio +44 (0)1442 416590. +44 (0)1442 246832. Click here to email www.maudio.co.uk www.m-audio.com

Paul White

The M Audio Octane seems to be one of a growing range of products aimed at extending the I/O capability of any digital recording system that happens to have ADAT optical connectors built in. Very simply, this 2U box houses eight mic/line preamps that feed both Photos: Mark Ewing individual analogue outputs and an ADAT digital lightpipe connector, the latter via onboard 24-bit converters. There are many hardware and software DAWs with ADAT input capability, so the idea is that you can turn that unused ADAT port into eight more useful analogue input feeds. Unlike the Behringer ADA8000, which offers similar functionality (but without the separate analogue outs), the M Audio Octane is strictly a one-way device — the unit won't give you extra outputs as well. However, most people now mix within their DAWs, so that's probably not a serious restriction for the majority of potential buyers.

Flexible Front-panel Design The Octane is housed in a very stylish 2U rack case with a heat sink on the back, and draws power from the included 18V AC mains adaptor. A power switch and LED are located on the front panel, but of course any PSU of this type should be unplugged or turned off when the unit is being left idle for long periods. Phantom power is switchable onto the mic inputs in two groups of four via buttons to the right of the front panel, and this supplies a full 48V. All connections other than

file:///H|/SOS%2004-09/M%20Audio%20Octane.htm (1 of 4)9/26/2005 12:04:35 AM

M Audio Octane

two instrument jacks are on the rear panel, and all the controls are on the front, which keeps everything very tidy. To offer the maximum flexibility without adding to the cost, only channels one and two have the aforementioned front panel instrument input as well as a gain control for that input. The main mic input controls operate similarly across all the channels; there's a gain control, a 20dB pad button, and a three-segment LED level meter, where the top LED indicates the onset of clipping. Channel one also has a switchable low-cut filter, and all the even-numbered channels have phase-reverse buttons. Channels seven and eight are special cases because, in addition to functioning as regular mic/line channels, they can be switched to decode the feeds from an M&S microphone pair into stereo via an internal sum-and-difference matrix. When M&S decoding is switched in, the Width control may be used to adjust the width of the stereo image. These two channels are set up so that the Middle signal feeds channel seven and the Sides signal channel eight. This is a nice feature if you happen to have a figure-of-eight mic you can use, because recording in M&S stereo can produce excellent results. Because the conventional ADAT format was designed to operate at 44.1kHz or 48kHz, those are the only two sample rates supported here. A rotary selector switch chooses one of the two internal sample rates or external clock (via the rear-panel word-clock BNC socket) and a green LED shows when the unit is locked to an external clock. Both word-clock ins and outs are provided, but there's no provision to sync to S/PDIF or ADAT inputs, as there are no digital inputs on this unit. That's no problem if your DAW interface can provide word clock, but it means you have to use the Octane as the master clock for your system if you don't have a source of word clock to feed it. However, if you're using the Octane as your main high-quality input device, then running it as your clock master may well be the best option for minimising clocking jitter anyway.

What Is M&S Miking? The M&S mic technique is an interesting way of recording in stereo. A cardioid mic is pointed directly at the centre of the sound source, and a figure-of-eight mic is set up in a coincident configuration so that its null points in the same direction. The cardioid mic provides the Middle signal, essentially just a mono recording of the sound source. The figure-of-eight mic collects the Sides signal, which can be combined with the Middle signal to create a stereo signal. However, combining the Middle and Sides signals to create traditional stereo requires a bit of extra routing and the use of a mixer with phase-inversion facilities, so M Audio have made things simpler here by including the necessary M&S matrixing within the Octane. This means that you can set up an M&S rig, plug the two mics into channels seven and eight, and then simply hit the M&S Matrix button to output a


M Audio Octane

normal stereo signal for recording purposes. The reason why many engineers go to the trouble of recording M&S stereo is that it has some practical advantages. Firstly, by adjusting the levels of the Middle and Sides signals, you can adjust the stereo width of the combined stereo signal, which can be very useful in some circumstances — the Octane's Width control works in this way. Also, when M&S stereo recordings are collapsed to mono, the Sides signal completely cancels itself out, leaving the Middle signal intact — other stereo recording techniques can be compromised by phase cancellations when collapsed to mono. Mike Senior

Connections The rear panel has balanced XLR inputs on all eight channels, and in conjunction with the pad switches these can also accommodate line levels. However, you must be very careful not to apply phantom power to any inputs that have line sources connected via the XLR. A safer bet is to use the rear-panel TRS A-D Line Input jacks, where inserting a jack plug overrides the XLR input. The instrument inputs on channels one and two also override the XLRs and have a very high input impedance of 3M(omega) so as not to load passive guitar pickups. Each channel also has a direct analogue output, again on a TRS jack, that may be used balanced or unbalanced. Plugging into these doesn't disconnect anything, so they can be used as regular outputs to feed some other analogue system or they can be used in combination with the inputs to provide seminormalised insert points. Of course if you do decide to connect the line inputs to a patchbay, the XLR inputs will be bypassed, so you need to plan your wiring needs in advance and ideally allow easy access to the rear panel for when those inevitable non-standard jobs come along. Both the word-clock connectors are standard BNC sockets.

High-octane Performance? The success of a product like the Octane relies largely on the quality of its mic preamps. These provide up to 50dB of gain, which is a little short of the 60dB provided by most preamps and consoles, but it's more than enough for capacitor mics, DI boxes, or dynamic mics used up close. The frequency response is flat from 20Hz to 20kHz, within a little over a tenth of a decibel, and the signal-tonoise ratio is 133dBA at a mid-gain setting. The balanced line inputs have a fairly standard 20k(omega) input impedance, and the line output impedance is 600 (omega). Channel one's bass-cut switch has a 12dB/octave slope and an 80Hz cutoff frequency, which is useful if your mic doesn't have a roll-off switch and there's some low-frequency background noise you'd like to get rid of. As with most such equipment, there is plenty of headroom to accommodate the peak levels needed by digital systems. The Octane was checked out using my MOTU 828 audio interface, with the file:///H|/SOS%2004-09/M%20Audio%20Octane.htm (3 of 4)9/26/2005 12:04:35 AM

M Audio Octane

Octane used as the master clock. After setting my MOTU to external sync, the Octane was recognised without any fuss and it functioned perfectly. The instrument inputs checked out fine and these were also good for recording the outputs from my Line 6 PodXT, so they seem to be able to handle modest line levels. Other than the limitation of 50dB of gain, the mic amps worked very well, imparting no obvious coloration to the sound. However, they do suffer from a problem I've come across on several other mic amps in recent weeks and that is that the gain is all bunched together at the top of the control's range, and I think I'm right in saying that a special pot is needed to avoid this — regular pots with a logarithmic law just don't do what you need. Because you need to be working in the high gain range for most studio vocals through a typical capacitor mic, setting the gain with any precision is trickier here than it should be. The M Audio Octane is a straightforward and well-thought-out product that has no obvious shortcomings other than perhaps its less than generous maximum preamp gain of 50dB and the bunched-up gain-control law. It is cheaper than the nearest equivalent Presonus model, but still over twice the UK price of Behringer's offering in this area. However, the Octane does have the M&S decoding matrix and the instrument inputs on the front panel, as well as analogue inserts/outputs, so it offers a comprehensive solution to most people's input expansion needs. It will of course also work with a hardware ADAT recorder, which makes it a useful front end for location recording. Published in SOS September 2004 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




In this article:

Friendly Fire Decks & Strums & Rock & Roll Rig Kontrol The Kontrol Centre What's In The Rack? Doing The Splits The Sounds The Flexible Guitar

Native Instruments Guitar Rig £350 pros Excellent user interface. Controller/preamp hardware included. Unequalled ability to combine elements in unusual ways. Good basic sounds.

cons May be too 'hungry' for some older computers.

summary Guitar Rig might have a friendly interface but it is no creative lightweight. Whether you're into good basic guitar sounds or imaginative weirdness, it's a great tool.

information £349.99 including VAT. Arbiter Music Technology +44 (0)20 8970 1909. +44 (0)20 8202 7076. Click here to email www.arbitergroup.com www.native-instruments. de

Test Spec Guitar Rig v1.1.

Native Instruments Guitar Rig (Mac/PC) Guitar Amp Modelling Software & Foot Controller Published in SOS September 2004 Print article : Close window

Reviews : Software

The market for software that emulates guitar amplifiers is pretty crowded these days, but Native Instruments have found a new angle with a modular program that comes with a dedicated hardware controller and interface. Paul White

Available for both Mac and PC, Guitar Rig takes the now-familiar concept of amplifier and effect modelling and delivers it in the form of a hybrid hardware/software package, where the hardware acts as preamp/impedance matcher and floor controller, while the software provides the necessary modelling and graphical user interface. PC users will need at least a 700MHz Pentium or a 1.33GHz Athlon XP machine to run Guitar Rig under Windows XP, while Mac users can run it under OS 9 or OS X on anything faster than a G3 800MHz machine. Guitar Rig can run as a stand-alone application, which conjures up visions of a tatooed biker metal guitarist with an iBook perched on top of his stack, or it can work as a plug-in within host software that supports DXi, VST 2.0, AU or RTAS. Authorisation is via a challenge and response code handled by the NI web site or by mail, and the software will run for 30 days prior to authorisation. When used as a plug-in, Guitar Rig is inserted into the signal path as an effect, not as a software instrument, and obviously you need to adjust your audio driver settings for minimum latency in order to be able to play guitar parts comfortably. Though Guitar Rig will work with MME drivers, it is not recommended due to the high latencies these introduce. ASIO or Core Audio are much safer bets.

Apple G5 dual 2GHz with

file:///H|/SOS%2004-09/Native%20Instruments%20Guitar%20Rig.htm (1 of 9)9/26/2005 12:04:41 AM


2GB RAM, running Mac OS 10.3.4, with MOTU 828 MkII running at 256-sample buffer. Tested with Emagic Logic Pro v6.4.

Friendly Fire The software amp modelling element of Guitar Rig is similar to many existing products except that the designers have tried to make it more user-friendly by using a drag-and-drop virtual rack interface that looks not unlike a guitar player's version of Reason. Essentially you pick the amps, speaker cabinets and effects you want to use and then drop them into the rack, where the signal always flows from the top unit to the bottom. Those who like to create stereo effects will find split modules for parallel signal routing and there are also two virtual tape decks, one optimised for playing audio that you might want to play along to, such as backing tracks, while the second allows you to record your playing and also to overdub additional parts. The rack includes a very accurate guitar tuner and a metronome that can lock to MIDI Clock or the host sequencer tempo. As you might imagine, the processing modules include emulations of classic and modern amplifiers, cabinets, microphones and effects, with the promise of more to be added in future revisions. The initial version of Guitar Rig offered three tube amplifier emulations plus more than 20 effects including models of some well-known hardware pedals plus studio-style effects such as delay, reverb and EQ. The Cabinets and Mics part of the program has 14 speaker types, As of version 1.1, four amp modules are including the wonderful B4 Rotary available: Instant Gratifier, Plexi, Twang Speaker taken from NI's very popular Reverb and AC Box. tonewheel organ plug-in. These are modelled as though they had been miked up using a choice of five microphone types, each with four different positions, and more flexiblity is offered by the ability to feed your virtual amps into up to eight miked speaker cabs at the same time. Apparently the modelling also takes into account the electrical reaction between the power amp and the speaker, though when multiple speaker cabinets are added, this is based on the first cabinet in the rack. The Split module allows parallel signal processing, where different amplifiers, cabinets and effects can be included in the two arms of the chain, which allows for the creation of very dramatic stereo effects and layers. In fact this is a concept that many sequencer designers would do well to include in their plug-in management sections. Further Splits can be dragged in to create multiple parallel signal paths and the output of each chain can be panned independently. Just before I completed this review, a pre-release software update was sent to me which added an AC30-inspired amp to the existing roster, as well as a treble booster and a psychedelic delay that can produce reverse delay effects as well as more conventional echo treatments.



The package ships with a large number of presets in various styles so that you can experience instant gratification before going on to create your own custom patches, and be warned that the designers haven't imposed many limitations on what you can string together, so you have full freedom to create outrageous patches as well as sensible ones. You can also tweak the factory patches and save as many versions as you'd like.

Decks & Strums & Rock & Roll While most of the elements of Guitar Rig will be fairly familiar to anyone who has used either software or hardware amp modelling devices before, the tape decks deserve a little more explanation. Deck number one can play back audio files and time-stretch them or change their pitch without changing speed, whilst files or sections thereof can be looped and recordings can be made using the New button, though you can't use loop and record at the same time. It accepts any audio files in WAV, AIF, AIFF or MP3 formats, though for some reason it has problems handling 24-bit WAV files so these are currently limited to 16-bit. This is a known 'issue' and will be resolves in a forthcoming update. All recording is limited to 16-bit at the moment, but that doesn't seem to cause any quality issues. Tape deck two is more geared towards overdubbing so you can create riffs to play over, and because it comes at the end of the signal path, it records all the amp modelling and effects that you've applied in glorious stereo. The time and pitch features of deck one are missing but otherwise the idea is generally the same. Overdubbing is of the very simple 'sound on sound' type (using the phrase in its traditional lower-case meaning), which allows you to play something in deck one while playing along and then recording the results to deck two. There are no multiple tracks or means of changing anything after recording, but if you simply want to get musical ideas down and you don't have a sequencer, it's fine.

Rig Kontrol The hardware foot controller bundled with Guitar Rig is called Rig Kontrol and has four footswitches and a pedal. Unusually, it outputs its control signals not as MIDI, but in the form of a modulated audio tone that registers all the switch and pedal moves, which is recorded on the other half of a stereo sequencer track. The obvious advantage of this control system is that no MIDI interface is needed to use Guitar Rig. The only influence Rig Kontrol has over the guitar signal itself is that it provides a high-impedance, low-noise input and so saves the user having to find a suitable DI box, though having said that, I thought I got a better sound using Rig Kontrol than when I tried the software with a conventional active DI box. The guitar connects to the line input on the Rig Kontrol foot controller and the two line out jacks on Rig Kontrol connect to the line ins of your audio interface. Alternatively, there's also a two-channel mode where the two inputs can be used to process two different guitars, which could come in handy for jamming or recording with a colleague, though the Rig Kontrol pedal/switch functions don't work in dual-channel mode as both audio channels are taken up with guitar signals. file:///H|/SOS%2004-09/Native%20Instruments%20Guitar%20Rig.htm (3 of 9)9/26/2005 12:04:41 AM


Physically, Rig Kontrol is built into a tough metal case, and can operate from almost any AC or DC power adaptor that will physically fit into the socket. It can also run from batteries, and the current consumption is very low so they should last a reasonably length of time. As you'd expect, Rig Kontrol includes the necessary high-impedance input circuitry to match the output of a typical guitar fitted with passive pickups, and it is equally happy with active guitars and basses. The functions of the pedal depends on how you set up your patch: it is typically used to control volume or wah-wah, though it is also useful when assigned to operate the crossfader in the software's Split module (see box), allowing you to crossfade between two signal paths during performance. Patch and Bank switching is handled by the footswitches.

The Kontrol Centre The Guitar Rig screen is divided into two main sections, with the virtual rack below the Kontrol Centre on the right of the window. Kontrol Centre takes the form of a toolbar to access key functions such as the tuner, metronome, patch naming, tape decks and so on, while the rack part is where you place your virtual modules to build presets. To the left of the window are three more sections that deal with bank and patch management, the palette of available rack components and global settings plus controller assignments, which are set up using simple multiple-choice menus. Because Guitar Rig offers a lot of modules and more are coming along, these are organised by type so that, for instance, when you want an amp emulation, you only get to see the amplifiers. Though most modules can be freely moved around in the rack, there are certain elements that always go in certain places, specifically the input section, the tape decks, the tuner and metronome and the output section. The amp, effect and speaker modules then slot in between these using simple drag-and-drop routines. Of course software wouldn't be software without a Preferences section and here you can choose from three window height settings (to suit your monitor) Although the Rig Kontrol doesn't output MIDI, and also select a quality mode for the all parameters within Guitar Rig can be time-stretch option that can be used selected for MIDI control on a patch-by-patch with tape deck number one. This basis. allows you to work out exactly what was played during that amazing solo by slowing it down, and as this kind of processing also takes up CPU resources, you have the option to trade quality against computing power.



Rig Kontrol has no truck with MIDI, but Guitar Rig can and does. So, as well as controlling it via the physical controls on Rig Kontrol, you can also feed it MIDI from your sequencer, enabling you to automate more parameters. Normal plug-in parameter automation is also available to the extent that your host software supports it. Even keys on your computer keyboard can be used as real-time controllers, and to save reinventing the wheel, Controller Sets can be created and stored to be used again. In the Preferences section dealing with controllers, you let Guitar Rig know what controllers you have available and whether they are switched or continuously variable. You can then assign these to individual parameters within the preset you're working on. This is fairly flexible, but there are rules to help clarify the operation. If a controller is being used for patch or bank selection, then it can't also be used for parameter control, which seems sensible enough. A controller can only control one parameter within a preset, which is an obstacle to doing clever things like reducing chorus depth at the same time as you increase the rate, but this shortcoming is apparently due to a limitation in the way host programs deal with plug-ins and not a design oversight. Of course the same controller can control different parameters in different patches and there's also a Learn function that overrides any previous assignment made for the controller involved. As you'd expect, a patch is essentially a snapshot of all the control settings and module placements for the setup in question, but there's also a Template facility that allows you to save subThere's no MIDI on the Rig Kontrol: instead, components such as effect settings it puts out a stereo audio signal with your and EQ settings for incorporation into guitar on one channel and control data on new patches at a later time. the other. Furthermore, each module comes with its own presets list and each amp has a default mic and cabinet arrangement available. It is also easy to categorise patches according to musical style or other criteria if you feel the need, and if you are really intent on creating so many patches that its hard to keep track of them, there's also a database search facility where you can search by title word, style, author, instrument, 'before' date or bank. Favourites can be added to a category called, not surprisingly, Favourites.

What's In The Rack? The rack always contains Input and Output components; the former has an inbuilt noise gate with adjustable threshold. The Output component has knobs for both master volume and the volume of the currently active preset. The Tuner and Metronome are also constant presences; the metronome syncs to tempo when plugged into a host sequencer that provides tempo information, and the tuner has an adjustable reference pitch as well as a mode for drop-tuning for the benefit of all those musicians hailing from Seattle.



The tape decks make up the other permanently elected members of the rack but after that it's pretty much up to you. There's a choice of cabinets from 1 x 12 to 4 x 10 and 4 x 12, based on popular models from the major UK and US manufacturers, as a well as a less-than-authentic parameter that can be used to change the apparent size of the cab. Virtual mics can be virtually positioned in front of the virtual cab on-axis, off-axis, at the edge of the speaker cone, at a distance or even at the back of the open-backed cab models. There are five mic models from which to choose, including the ubiquitous '57', '421' and a couple of capacitor models, and there's also an Air control that adds synthesized early reflections to mimic what might occur in a real room. When two or more mics are being used together, the relative distance between them can be tweaked to change the phase-cancellation effects, there's a mic phase button and some basic treble EQ. Though there are relatively few amp models at the moment, some come with a choice of preamp models, which can be thought of as amp channels, so for instance the Instant Gratifier solo head has Clean, Raw, Vintage and Modern variants. There are also two alternatives for the AC30 and Twin Reverb-inspired modules as well as the usual Drive, EQ and Presence controls. If you're into deeper-level tweaking, you can also press the Maximise button to see more controls, which allow you to adjust the imaginary Effects options include the new psychedelic delay, as well as more conventional modules bias of the imaginary tubes, mess with such as a spring reverb. the power-supply capacitor values, drop or raise the supply voltage with a virtual Variac and even change the mains supply from 50Hz to 60Hz, all without the risk of electric shock! If you need searing shred that will sustain until your guitar rots, Instant Gratifier on the Modern setting is probably the one to go for, though it can turn its hand to clean sounds too. No prizes for guessing what Twang Reverb is based on either — and it has a tremolo plus authentic spring reverb. Once again, you can change the tube amp and power-supply parameters and, unlike on the original, you can adjust the reverb decay time and even the length of the virtual spring in the reverb model. Plexi is based on the obvious British classic with Normal and Bright channels, the usual controls and the now familiar amp and power supply adjustments. You can also mix the two inputs. The late arriving AC Box has all the familiar Vox AC30 controls, though it has been tweaked to let you mix the Normal and Bright channels, and two extra tone knobs have been grafted on. It also has a tremolo plus those under-the-hood controls for tinkering with the power supply and tube bias.



Teamed with these amp models are two distortion devices, Skreamer and Distortion, plus separate stereo Tremolo, Stoned Phaser, Chorus/Flanger, Ensemble, Rotary Speaker, Oktaver, Filter (which handles wah-wah) and Talkwah. The latter is a voice-box filter effect that even has a variable mouth size parameter! Then there are a couple of variants on the EQ theme, a volume pedal, a limiter and another noise gate. Tube compressor emulation is available for clean sustain along with a four-tap delay/echo unit, a separate spring reverb unit and a studio reverb with variable room size and basic parameter adjustment. Interestingly, the Oktaver breaks up on anything but monophonic melody lines, just like real octave pedals. Delays can be sync'ed to tempo where needed, and unlike real gear where you have to observe such niceties as impedances and levels, you can connect almost anything to anything without worrying about cooked speakers or the smell of burning transformer! The update I installed also added a treble booster and the aforementioned psychedelic delay, plus Pitch Pedal for polyphonic pitch-shifting.

Doing The Splits Combing amp, speaker and effects to form a simple chain is as easy as you'd expect, but Guitar Rig also allows you to create stereo patches with different components in the left and right signal paths using the Split modules. Furthermore, although the vast majority of electric guitars have a mono output, many of the modules are stereo — not just the effects but also the speaker cabinets when using pan or the Air option that creates the ambience of a cabinet miked in a real room. If a A typical split setup. The Rig Kontrol's pedal stereo module is fed into a monocan be used to crossfade between two only module, the output will be elements in a split sound. summed to mono, examples being mono amp models or speaker cabinets used without Air modelling. Splits are set up using the Split A and Split B modules plus the Split Mix module. Elements inserted between the two Split modules comprise one arm of the split while those inserted after Split B but before the Split Mix module comprise the other arm. Panning and A/B mix is independently adjustable for each channel using the Split Mix controls and there's also a Dual mode that keeps the two channels completely separate for when two guitars are being processed at the same time.

The Sounds The factory presets give a good account of what Guitar Rig is all about, but as ever, you really need to fine-tune them to your own taste and to match your own file:///H|/SOS%2004-09/Native%20Instruments%20Guitar%20Rig.htm (7 of 9)9/26/2005 12:04:41 AM


instrument before they really shine. Certainly creating your own custom stack isn't difficult, though with so many presets already on offer, it's often easier to modify something that already exists rather than start from scratch. The sound of guitar amplifiers, or models of guitar amplifiers, is very subjective, and every hardware and software modelling system I've tried to date has had its own character. Almost as important as the sound is the way the guitar feels when you play it, as there are some very complex things that happen between the ear, brain and fingers when you're playing. Most of this can be attributed to the touchresponsiveness of the amp emulations, and though I've yet to find anything that exactly replicates the experience of playing through a real amplifier, some models are extremely good. The physical feel of Guitar Rig when playing is comparable to the other better amp modelling software packages out there, and while it still doesn't match a real amplifier in this respect, the sounds are nicely touch-responsive and also respond well to backing off the guitar volume control. The amp sounds themselves are, quite literally, what you make them, because in addition to the usual controls, you have those extras for emulating a clapped-out or badly set up amp that could just provide the magic overdrive sound you were looking for. Likewise, being able to hook up a compressor before the amp allows patches to be created that are fairly clean but have lots of sustain. I liked the rack effects a lot, particularly the psychedelic delay, which produces as good a backwards guitar emulation as I have heard if set to 100 percent wet. However, what Guitar Rig is best at is excess. Forget all you've learned about gain structure, impedance matching or signal levels, because in Guitar Rig's virtual world, you can feed 4 x 12 cabinets directly from compressors, you can daisy-chain different amps and you can have a floor full of virtual speakers and microphones. Some of the combinations yield quite unexpected results — don't be afraid to use one amp as the preamp for another, for instance. The user interface is a doddle to use, with thoughtful touches like Minimise buttons to squash the rack modules if your screen gets too busy. Conventional window buttons can be used to remove instruments from the rack, and when you drag in a new one, a red line appears between existing modules to show you where the newcomer is about to go. Simple amp sounds are simple to create and the spring reverb module is disturbingly authentic, but because of all the creative options available, you can go far beyond the norm to create sounds that are unique to you. Not all of these are pretty, but with just a little effort, you should be able to to get close to just about any guitar sound you've heard on record. Furthermore, no matter what amps and effects you use, the guitar signal is recorded clean and processed on playback, so you can change your mind about what to have in your guitar rack right up until the time you come to do the final mix.

The Flexible Guitar



Guitar Rig isn't so much a new concept as a friendlier and more customisable version of what's gone before. While the playing experience may not be quite the same as with a real amplifier, the results you can get are excellent and make recording the electric guitar a pleasure. On the whole, Guitar Rig is very well thought-out and incredibly flexible. The floor controller is a nice touch for anyone who needs a volume/wah pedal or who is brave enough to use a virtual amplifier live, and its novel system of recording automation data certainly worked flawlessly. In addition to this, you can automate the parameters much as you would with any other plug-in when working within a sequencer. For me, though, the real beauty of the package is the straightforward drag-anddrop method used to create custom racks of amps and effects. Being able to ignore the laws of physics when interconnecting amps and effects comes a close second! The only law of physics you have to remain aware of is that of required CPU resources, and according to the CPU meters in Logic, a fairly busy Guitar Rig rack required around one-sixth of the power of a dual 2GHz Apple G5 computer. Published in SOS September 2004 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



Novation X-Station 25

In this article:


Enter The X-Station From Remote Audio Xtreme Controller Keyboard, Audio Published in SOS September 2004 To X-Station MIDI Control Print article : Close window Audio Reviews : MIDI Controller And There's More... Conclusions

Novation X-Station 25 £499 pros Semi-weighted keyboard with aftertouch. Large selection of controls, laid out like a traditional synth panel. Dual-format audio inputs for guitars, mic- and line-level signals. Built-in effects (great for nolatency monitoring). KS-series synth with audio routeable either to outputs or computer. MIDI interface, audio interface and controller keyboard all from one USB cable. The keyboard can be powered over USB, and there's a built-in battery charger for emergency or stand-alone operation.

Interface & Synth

After releasing their Remote 25 MIDI controller keyboard, Novation released the Remote Audio 25, rapidly following it with the Remote Audio 25 Xtreme. Now, the X-Station has replaced both of these. Can we disentangle it from its convoluted beginnings? Simon Price

I'm sure many hi-tech music-related products have complex design histories, and we simply never get to hear about them, but the one behind Novation's X-Station takes some beating. Following the release of the well-specified Remote 25 MIDI controller keyboard in Autumn last year (see the review in SOS August 2003, or at www.soundonsound.com/sos/aug03/articles/novationremote25.htm), Novation announced a similar product which was to incorporate an audio interface. In this, they seemed to be following a similar path to that taken by M Audio with their Ozone controller, which appeared when someone realised it would be great to have a USB audio interface built into the company's Oxygen 8 controller keyboard. Novation's Remote Audio 25 was a separate product, which co-existed with the MIDI-only Remote 25. The Remote Audios were manufactured, and one was sent to SOS for review.

cons Poor Mac OS X drivers require high system specs for useable results, and there's no OS 9 support. If you use the built-in synth, template memory is reduced, one of the effects units is used up. Synth and MIDI control can't be used together. Although the effects are stereo, they are only recorded in mono.

summary

Before that article was completed, however, Novation announced at the start of this year that the Remote Audio would be available in a variety of sizes (49- and 61-note versions), and then that the product was to be renamed the Remote Audio Xtreme, and bundled with various pieces of music software, including a cut-down version of Ableton's Live and Steinberg's Cubase, and Novation's own V-Station software synth. Furthermore, a software

file:///H|/SOS%2004-09/Novation%20X-Station%2025.htm (1 of 8)9/26/2005 12:04:46 AM

Photos: Mark Ewing


An impressive desktop-music solution built into a keyboard synth. The Mac OS X implementation isn't great on the audio side, but as a controller it's extremely well thought-out and generously equipped. Perfectly suited for controlling software synths, and a great option for live laptop giggers.

upgrade was planned which would turn the controller into a fully fledged synth. We elected to wait until the synth upgrade became available before completing our review. Just after that happened though, Novation elected to clear the decks with the whole project, and relaunch the product as the X-Station, losing the software bundle from the package in the process. With the arrival of the completed X-Station 25 at the SOS offices, we've finally been able to complete the review, which has been a work-in-progress for nine months at the time of writing!

information £499 including VAT. Novation +44 (0)1494 551270. +44 (0)1494 551280. Click here to email

Enter The X-Station

Physically, the X-Station 25 is very similar to the Remote 25, sharing all the same parts for the knobs, sliders, and keys. It also has the same attractive silvery-grey matt finish. However, the X-Station is not simply a Remote 25 with an audio www.novationmusic.com interface bolted on. The controller layout is quite different for a start, and of course, there's that built-in synth, of which more in a moment. What the X-Station Test Spec does share with the Remote 25 is its build quality — as soon as you get hold of it and hit a few keys, you get the feeling you're dealing with a different class of 800MHz Apple iBook with 640MB of RAM running Mac device to the super-budget MIDI controllers out there. The unit is quite light, but OS 10.3.4. feels solid and substantial — more an instrument than a computer peripheral. The two-octave keyboard is semi-weighted, and there are smooth 'clicky' dials, a Propellerhead Reason, Ableton Live, Digidesign Pro nice blue backlit LCD, and lots of lights. Lights is good... Tools LE v6.4, Emagic Logic Express v6. X-Station OS reviewed: v2.0 (and all previous versions!).

Connection to the computer is via USB, which takes care of both the MIDI and audio sides of things. There are also two standard MIDI Out connections, and a MIDI In, so on top of everything else you can use the X-Station as a simple MIDI interface. The controller treats all the MIDI outputs (the two standards, plus the USB connection) separately, and any control can be set to output via any combination of the three. As you can see from the back-panel picture over the page, you also get sustain- and expression-pedal inputs. You'll notice that there's a power-adaptor input next to these, but before you groan, the unit does take power from the USB connection, so you won't need an adaptor most of the time. In fact there's a very neat extra when it comes to power: the unit can be run from batteries, and if you use rechargables, they will be charged while the unit is plugged in. Novation warn that not all laptops supply enough power to run the XStation, and in fact you may recall that in the SOS review of the Remote 25 controller, the test desktop machine had inadequate power to drive the unit. However, there may have been progress in this area, as even my 12-inch iBook running from its battery was able to power the X-Station. What I did find was that the unit is not happy when connected to a USB hub, even when powered from the adaptor. The lights all pulsed menacingly, and it disturbed the whole buss. The X-Station is absolutely bristling with controls: there are nine sliders, and tons of knobs, rotary encoders and buttons. While the Remote 25 has its controls set out by type in three rows, the X-Station takes a different approach, grouping them together into zones that are likely to correspond with the panel modules on software synths. So, you have a group of knobs and buttons laid out as an



oscillator/mixer section, a filter section, LFOs, arpeggiator, and effects. The sliders are grouped as two ADSR envelope modules, but if you ignore the labelling, the sliders still form a row, with buttons above each. In other words, as well as controlling plug-ins and synths, the sliders are equally suited to a mixer or drum machine. Similarly, the eight knobs along the top of the synth are sometimes assigned to pan/send controls in mixer templates. However, they form a broken line that doesn't line up with the sliders, so you have to use your imagination a bit. The memory came loaded with 40 templates for controlling various bits of software, with the more common ones benefiting from glossy card overlays that transform the keyboard's labelling. Many synths don't really require overlays because the panel is printed with such common parameters. Some buttons, or groups of buttons, can be programmed to step through different pages, with corresponding LED feedback: a classy addition. The centre section has standard transport controls, the LED display, and the Data/Value encoder that dials up the stored templates. This is also the place where you can get under the lid of the X-Station to start editing controls, and making your own templates. The far left of the panel is devoted to the audio interface, which I'll come back to in a bit. The bottom half of the keyboard is identical to the Remote 25, with its aftertouch-sensitive keyboard, pitch/mod joystick, and X-Y touchpad. As on the Remote 25, the joystick's mode of operation can be switched at the back of the unit between sprung and free movement. I liked the feel of the keyboard a lot, although the keys still pivot at the back unnervingly as noted in the original Remote review. Another criticism from that review was that the pitch/mod joystick didn't travel all the way across, and would always be a few cents out of tune at its limits. The initial SOS review model (which was still entitled the Remote Audio 25) had the same problem, although I found it possible to force the joystick to its full range by pushing it hard. Later on, the version 2.0 operating system which came with the Xtreme upgrade (and the X-Station which we had last of all) added a joystick-calibration option to fix this.

From Remote Audio Xtreme To X-Station The original Remote Audio is no longer available, but those who purchased one during its short time on the market can still upgrade it to equivalent X-Station capability (although the upgrade uses up some memory in the unit that was originally available for storing user templates, and upgrading wipes these user memory slots, as I found out the hard way!). With the X-Station, to save a user template you must overwrite one of the factory ones. The only other changes in the move from the Remote Audio Xtreme 25 to the X-Station 25 are minor cosmetic ones on the front panel to reflect the presence of the built-in synth. The software bundle has also been dropped, which is a shame, as it made the keyboard package a neat desktop-music solution, but on the other hand, the price of the bundle-less X-Station is also lower than that of the Remote Audio Xtreme.

MIDI Control file:///H|/SOS%2004-09/Novation%20X-Station%2025.htm (3 of 8)9/26/2005 12:04:46 AM


The X-Station's 40 factory templates cover pretty much all of the well-known audio programs and virtual instruments. The templates assign MIDI CC values to all of the controls, based on what a particular device is expecting. This is a different approach to that of using a controller that just kicks out any old set of values, and relying on the software's learn function to map to them, although you can of course do this too. Using templates has the advantage that you don't have to set anything up, but there can be disadvantages. Propellerhead's Reason can work in either way, and so serves as a good example. Each device in Reason is factory-set to respond to certain CC messages when that device is selected to receive MIDI input in the sequencer. This means that for the mixer to respond to the X-Station's Reason Mixer templates you have to create a sequencer track for it, and arm it to receive MIDI. In Reason, only one device at a time can receive MIDI in this way. For studio work, it's probably fine to use the template approach, as you tend to be working on one instrument at a time. Playing live, you probably want to think about doing things differently, forgetting about the templates and using Reason's MIDI learn function to assign controls as you see fit. The main advantage of this is that learnt MIDI controls stay active regardless of which device is armed for MIDI input, so you can control key parameters on various devices at once. However, this can introduce the problem of overlapping controls, due to Reason's preset remote mappings. The best way around this is to avoid MIDI channel 1 for your controls, as Reason's built-in control mappings default to channel 1. The X-Station's control layout allows for some streamlined templates. For example, Reason synths like Malström can now be controlled from one template, instead of being split across two. Now that the layout mimics many Every connection suggests versatility — an of the clusters of controls found on the optional power socket, MIDI sockets for synths, it's easier to use. You no longer flexibility or interfacing duties, digital as well have to memorise which knob or fader as analogue outs, expression-pedal is mapped to which parameter, or keep connections, and the audio interface. having to search the overlay card. This is particularly useful for live applications, where it's likely to be too dark to read the front panel. I spent a lot of time playing with the Subtractor and Malström templates (for Reason), thinking that these were the modules that would most benefit from a more 'knobular' approach. Even so, the X-Station isn't an instant improvement on the mouse, as it takes a little time to learn how things are mapped, and even though there are an unusually large number of hardware controls available, the Reason synths still have more parameters than can be addressed at once. The X-Station gets around this by using some of its buttons to toggle controls between different functions. For example, the second ADSR section on the X-Station's panel is toggled between controlling Subtractor's mod and filter envelopes. With a little practice, the system works well. Some controller sets are split over two templates, such as the Redrum and Reason mixer. This became viable with the v2.0 X-Station OS, which allows the template settings to remember their settings when switching about. However, the keyboard octave range resets when switching between templates, which is annoying. The v2.0 OS



also added a control 'pick-up' option, so that a controller doesn't become active until it moves through the controlled parameter's current value. This solves the age-old problem of parameters 'jumping' suddenly when you move knobs or sliders. While not as elegant a solution as motorised controls, or a rotary-encoder system that uses plus/minus values instead of absolute values, it's definitely the next best thing. At the time of the original Remote 25 review, Novation promised an on-screen editor, which has not yet materialised at the time of writing. To be honest, it's no big deal if the same happens for the X-Station, as the template-editing system is pretty easy. Many aspects of the controller side, including the template editing, remain the same as they were on the Remote 25. I managed to create a Pro Tools control bank by configuring a template to emulate the JLCooper CS10 controller. The whole thing only took a few hours, and worked perfectly (for more detail, see the Pro Tools Notes column from May this year).

Audio The simplest way to think about the audio side of the X-Station is as a separate two-in, two-out, 24-bit USB audio interface that just happens to be built into something else. As I was testing on a Mac, I used the Core Audio drivers, which provide system-wide access to the interface, with control from the Audio MIDI Setup control panel. My test applications, Cubase SE, Logic Express, Reason, and Live, were able to 'see' the X-Station in their setup pages, with the usual latency controls, and everything else you would expect. There are two inputs, with the dualformat quarter-inch jack and XLR connections that are becoming so common. The inputs can accept line-, mic-, and instrument-level signals, with a continuous, wide-ranging gain control rather than switches. Phantom power can be switched in and out individually on the two channels. The Input button selects which input is being addressed The X-Station's controls strike a successful by all the controls, or you can link the balance between dedicated ones for the inputs for stereo operation. There's a two-segment LED meter for each input, onboard functions, and assignable ones for other uses. with green to show the presence of a signal and red to show clipping. This might not seem like a lot, but it's more than you get on many small USB interfaces. There are separate level controls for the line-outs and headphones, all of which are on quarter-inch jacks. Zero-level latency (direct in-out) monitoring is accommodated with a pot that mixes the input levels with the output from your software. However, for this method to work, your software has to have a 'disable direct monitoring' option so that you don't hear your inputs twice with a delay.



The sound quality of the input preamps is pretty good, but I had an on-going issue with the output from software, where I had intermittent clicks, pops and glitches. This seemed to be unrelated to buffer sizes, and I suspected it was an issue with the drivers. Novation suggested that my test computer was underpowered, as they recommend a minimum 1GHz G4. This is a whopping requirement for a two-in, two-out audio interface — my Digidesign Mbox and M Audio Quattro have no problems running on my 800MHz iBook. In fact, they run fine on my ancient 400MHz blue-and-white G3! In any case, I duly tried it out on a 1.25GHz G4, and the dropouts went away, so it would seem that you really do need a 1GHz G4 or better to use the X-Station's audio interface. That wasn't all, either; the Mac would also freeze about every third time I connected the unit via USB, and sometimes Cubase SE would see two units in the audio setup. All this led me to develop a lack of confidence in the quality of the Mac OS X drivers, letting down the audio-interface side of the X-Station for Mac users. A bonus with the audio side of the package is the inclusion of built-in effects on the input path. There are two separate multi-effects units for independently treating your input signals. The available effects are Delay, Reverb, Chorus, Compression, Distortion, and EQ, all of which can be used simultaneously. The implementation of this feature has been thought out well, allowing flexible use. Each input's effects routing can be switched to choose whether the signal is recorded with or without the effects. This is really useful, as you can, say, record dry vocals, using zero-latency monitoring, and still hear reverb and delay to help your performance. The distortion effect is of the guitar overdrive pedal variety, giving you the same luxury when plugging your guitar directly into the interface. There is an issue with recording mono inputs with effects though, which becomes particularly apparent when you're using the built-in synth, as you'll see in a minute...

And There's More... With the first version of this product, the Remote 25 Audio, our review would have stopped here. However, the X-Station concept was born when Novation realised they could use the DSP chips in the unit to add a hardware synth to the feature-list! From what I can tell, the synth side is somewhere between a KStation and an eight-voice version of the KS-Rack (using a single KS sound engine). In other words, it's a pretty decent analogue-modelling synth, and by no means a toy or gimmick. The synth is eight-part polyphonic, monotimbral, has three oscillators, and a multi-mode filter. It has frequency modulation, pulse-width modulation, two LFOs (with single-shot mode), ring modulation, and envelopes: it even has effects. The synth is accessed by pressing the top 'Play' button, which toggles between synth and controller modes. There are 200 preset patches accessed from the



Data controller, which are in fact the same as the factory presets on the 'K' synths. User patches must overwrite the factory ones, as memory is a little scarce with all this stuff crammed in. Sound-wise, the best description is that it sounds like a Novation synth, and of course in particular a K-Station or VStation. By this, I mean that the sound There are plenty of what seem like dedicated is clean, rounded and creamy, and to synth-editing controls, and of course they my tastes, a little too polite. While can be used in this way, but they can all be basically modelling an analogue reassigned if you wish by editing the subtractive synth, it has more in templates. common sonically with basic danceorientated sound modules, reminding me of my Yamaha CS1x. This means that it's perfect for pop and anthemic dance music-type stuff. In addition to the normal oscillator waveforms and noise generators, there are several more characteristic waves, such as 'Organ', 'Rhodes Piano', and 'Analog Bass' all of which get a good look in throughout the presets. Luckily, programming patches is dead easy given that you've got so many knobs, and I was able to start replacing the presets with some nastier sounds! While ingenious, the synth functionality comes with some compromises. For a start, the synth takes over input 2's audio and effects hardware. This means that while the synth is activated, you can only use input 1. This does not affect the audio-output side, which can be mixed with the synth by positioning the monitor knob halfway between the input and output positions. An extremely cool feature is that the audio from the synth can be routed directly into your host software via the USB connection, because it's effectively behaving the same as an instrument connected to input 2. However, there's one flaw in this plan, which is that (as with any other mono input) you can't record stereo effects, and the synth leans quite heavily on the effects. Thus, what you hear when monitoring the input path is stereo — but what you record isn't. Obviously, you can't record the synth from the main outs (instead of via USB) because that would require another audio interface. Of course, this problem affects any mono input, but is more of an annoyance with the built-in synth, because you tend to think of the effects as an integral part of the sound, whereas they are really just a guide when recording other instuments or vocals. The other limitation is that the built-in synth is inactive when you are in controller mode. This means that you can't play back the synth from a MIDI track while using the controller to do something else. All you can do is record the synth part to an audio track, and then continue to use the keyboard to play other parts.

Conclusions On paper, the X-Station is almost too good to be true. You get a really nice controller keyboard with semi-weighted, aftertouch-generating keys which feels a file:///H|/SOS%2004-09/Novation%20X-Station%2025.htm (7 of 8)9/26/2005 12:04:46 AM


class above the competition. You also get an unusually large number of varied hardware controllers. You get a USB audio and MIDI interface, with mic/ instrument inputs and a built-in multi-effects unit. And it's also a hardware synth! All this makes the X-Station the perfect choice for use as a live controller. The main reason I say this is that in addition to handling the audio, MIDI, and controller tasks for your laptop live rig, it's perfect in an emergency. For a start, you get the best of both worlds in terms of power supply. For the most part you'll get your power from the USB connection, but if your computer packs up, you can revert to the rechargeable batteries. Then the built-in synth means that you will still be able to make some kind of noise, instead of just standing there staring at the boot screen. Other than this, the X-Station must be a seriously tempting option for anyone looking for an integrated desktop-music system (although my ardour has been slightly dampened by the withdrawal of the software bundle promised with the Remote Audio Xtreme). In any case, it's ahead of the competition in the smallformat controller-with-keyboard market. However, if the main thing you need is the knobs and sliders, the price puts it up against some serious competition with motorised faders and genuine rotary encoders. The downside is that Mac support appears to be an afterthought, and is not even mentioned in the manual. If Novation address this, though (or if you're a Windows user!), the unique combination of features in a portable package is going to make this a desirable bit of gear. Published in SOS September 2004 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



Pearl CC22

In this article:

Pearl CC22

Rectangular Capsule Design Condenser Microphone Facts & Figures Published in SOS September 2004 On The Session Conclusions Print article : Close window

Pearl CC22 £648

Reviews : Microphone

pros Smooth and extended frequency response. Generally neutral and accurate tonality. Very consistent polar response with frequency. Side-address configuration and small body make this a discreet mic to place.

This innovative rectangular-capsule design aims to combine the desirable attributes of small- and largediaphragm sounds. Hugh Robjohns

cons Noisier than some. No pad option to increase SPL rating. Mic case cannot contain shockmount.

The Swedish microphone manufacturer Pearl Mikrofonlaboratorium (PML) was founded in 1941 and was the first and only microphone manufacturer in the country until the early '80s. Pearl's first products were mainly crystal and dynamic microphones, but by the end summary of the '50s their 'trademark' rectangular condenser Using Pearl's unusual capsule was introduced. The development of this unusual rectangular capsule, this fixed- design has continued to this day, and Pearl claim the cardioid condenser mic design's unique characteristics provide one of the finest delivers a neutral and capsules available for music recording. This dualaccurate sound, combining the consistent polar response membrane rectangular capsule is currently used in of a small-diaphragm mic with several studio microphone models, including the TL4, the clarity and body of a large- TL44, DS60, CC30, and CC22. A new omnidirectional diaphragm mic. model based on the same capsule is also about to be information launched, called the CO22. CC22, £647.65; CC22H £647.65; Type 1927 shockmount, £43.63; Type 1928 shockmount, £179.90. Prices include VAT. DACS +44 (0)191 438 2500. +44 (0)191 438 2511. Click here to email www.dacs-audio.com www.pearl.se

Photos: Mark Ewing

Rectangular Capsule Design The rectangular capsule is unusual for more than just it's shape. The membrane is coated with vaporised aluminium rather than the sputtered gold employed by most other condenser capsule manufacturers. With different dimensions in each axis, the diaphragm is relatively free from resonances, compared to traditional circular membranes, and Pearl claim that the capsule therefore achieves a very flat and extended frequency response, quoting figures of 20Hz-25kHz (-3dB).

file:///H|/SOS%2004-09/Pearl%20CC22.htm (1 of 5)9/26/2005 12:04:50 AM

Pearl CC22

The CC22 is a fairly compact microphone and, although its size is more usually associated with small-diaphragm condensers, it's really a large-diaphragm design. It measures 145mm long, and the main part of the body is 28mm in diameter, increasing to 32mm for the capsule grille. The mic weighs a modest 270g, and is supplied in an aluminium case with a hard moulded-foam liner. A shockmount is supplied separately in a cardboard tube, but unfortunately the case has not been designed to accommodate the mic installed in its shockmount. It would even be a tight fit if you were prepared to modify the foam liner! The microphone is a side-address design, which feels a little odd with such a slim body, but because the capsule is so visible the correct orientation is quite obvious — more than can be said for many microphones! The CC22 has a shiny black paint finish over a brass body, with the company and model name etched on the side to indicate the front of the polar pattern. The etching is in gold — presumably the underlying brass metal — but on the opposite side the serial number is etched and filled with black paint so that it is readable but not obvious. Just below the capsule section is a reasonably subtle gold-plated ring. The capsule, which measures roughly 32 x 18mm, is supported on a compliant suspension and is protected behind a 45mm-long wire-mesh grille. The aluminium diaphragm is visible in the right lighting conditions behind gold-plated front and rear plates drilled with an array of holes. The audio output is provided through the usual male XLR connector at the base of the microphone, and it requires standard 48V phantom power, from which it draws a modest 2.7mA current.

Facts & Figures The CC22 is intended for both studio and live-sound applications, and features a fixed cardioid polar response. The mic has a fairly typical sensitivity of 18mV/Pa with a self noise of 13dBA — not the quietest of large-diaphragm mics, but comparable with a lot of highly regarded products from other manufacturers. The maximum SPL figure is given as 126dB for 0.5 percent distortion, and there is no pad facility to increase the headroom. The head amp is a transformerless design, but there is also a variant model called the CC22H, with an alternative preamp. This provides a much greater output level — up over 13dB on the standard model to a sensitivity of 100mV/Pa. This version of the mic can also accommodate 6dB more SPL (132dB), although it has a slightly higher noise floor (up 3dB). The alternative design has a slightly higher output impedance of 500(omega), but requires a load of over 6k(omega) and draws less phantom current (1.3mA). The review model was equipped with the Type 1927 shockmount, which is basically a soft rubber doughnut about two inches long, supported in the centre by a metal ring connected to the stand adaptor (supplied with 5/8- and 3/8-inch thread barrels). The microphone slides into the rubber doughnut, and the size of the aperture and the 'stickiness' of the rubber ensure a safe and reliable file:///H|/SOS%2004-09/Pearl%20CC22.htm (2 of 5)9/26/2005 12:04:50 AM

Pearl CC22

mounting. I found this shockmount to be reasonably effective, as well as being particularly compact and discreet. However, there is also an alternative design called the Type 1928 which is a more traditional 'cats cradle' affair. This version comprises a square metal frame connected to the stand adapter, and the microphone is supported in a tubular centre section which is suspended from the square frame by a quartet of 'O'-rings above and below. It looks simple and is probably very effective — the mechanics of the design will inherently afford far greater mechanical isolation than the doughnut Type 1927.

On The Session I compared the Pearl CC22 with a variety of large-diaphragm mics from my collection, most of which come in at a slightly higher recommended retail price in the UK. These included the AKG C414B-ULS, the Blue Baby Bottle, the Neumann TLM103, and the Microtech Gefell M930. I also compared the Pearl with the Sennheiser MKH40 as a gold-standard reference, even though the latter is considerably more expensive. Other models which are broadly comparable on price (but which were not available to test) include the Audio Technica AT4047 and AT4050, the Beyerdynamic MC834 and the Rode NT2. The Pearl CC22 sounds very smooth, with just a hint of a presence lift centred about 6kHz. The high-end response seems to roll of very gently above about 15kHz, which provides plenty of detail and clarity without the 'in yer face' brightness associated with so many condenser mics. The bottom end is equally well extended, with a similar gentle roll-off below about 40Hz — it's about -4dB at 20Hz, but if used close to a source the proximity effect will lift the low end response considerably. The polar response is actually very tidy and consistent with frequency. The frontal area is wider than many, only dropping sensitivity by 3dB or so out at 45 degrees off-axis. The frequency response is also very consistent off-axis, and the rear rejection is extremely good at more than 25dB — although there is a pronounced high-frequency tail, giving a more hypercardioid response above about 4kHz.

The CC22 in the Type 1927 rubber shockmount.

I think part of the reason for the very consistent polar response is the relatively narrow width of the capsule. In the horizontal plane the diaphragm dimension resembles that of a small-diaphragm condenser capsule, which generally exhibits a consistent polar pattern with frequency. In the vertical dimension, the capsule is relatively large — larger than most large-diaphragm circular capsules, in fact — and this imparts a characteristic 'larger than life' element to the sound quality as well as helping to minimise self noise. file:///H|/SOS%2004-09/Pearl%20CC22.htm (3 of 5)9/26/2005 12:04:50 AM

Pearl CC22

The CC22 is an honest-sounding microphone, tending towards neutral rather than coloured. In my comparisons it sounded closer to the AKG C414B-ULS than any of the other mics to hand. The Microtech and Neumann models were noticeably more coloured and heavier sounding. In terms of resolution and the ability to capture complex detail the CC22 performed very well, capturing the complex harmonics of a 12-string acoustic guitar quite creditably. The Sennheiser MKH40 set the benchmark here — as usual — and none of the test microphones really came close; but the Pearl, AKG and, in third place, the Blue all delivered neutral and acceptably accurate results. I grew to like the Pearl a lot, and found it delivered great results on almost everything acoustic — piano, percussion, guitars, and strings. I didn't get a chance to try it on brass, but I would expect it to deliver a smooth, clean sound if placed carefully. It worked well on trained voices too, although it isn't a mic to flatter the vocalist in the way a big Neumann does. If the star quality is not already present in the source, the CC22 won't invent it! The accurate polar pattern was useful in multi-mic situations, ensuring that the inevitable spill was captured cleanly and without undue coloration. As far as separation was concerned, the relatively high front-to-back ratio of the CC22 is better than many well-regarded mics, although care is needed to angle the high frequency hypercardioid tail away from bright sources! The side-fire configuration generally required a vertical orientation, which I found made the mic relatively inconspicuous in many cases — far more so than the bulky AKG C414B-ULS or Neumann TLM103. In my mic box, only the Microtech Gefell M930 is as discreet!

Conclusions The Pearl CC22 is a very fine microphone, matching or exceeding the sonic capabilities of its marketplace peers. Although seen relatively rarely in the UK, this microphone deserves careful auditioning, because it offers a unique blend of properties making it a worthy alternative to the more familiar items. It is quite an expensive microphone, but it delivers a smooth, extended, and neutral sound quality, seeming to combine the resolution and precision of a small-diaphragm mic with a hint of the 'larger than life' character and body of a large-diaphragm mic. It is particularly well suited to classical or folk-music recording, where accuracy and detail are key attributes. Published in SOS September 2004


Pearl CC22




Real Traps Mini Trap & Micro Trap

In this article:


Modular Acoustic Treatment The Long & The Short Of It Acoustic Panels Published in SOS September 2004 Installation Standing Waves Print article : Close window Testing The Space Reviews : Accessory Acoustic Absorption Trap Star Or Pony & Trap?

Real Traps Mini Trap & Micro Trap pros Mini Traps offer very effective bass trapping down to 90Hz and below. Stylish looks in both black and white. Easy to fit.

These acoustic panels not only offer sonic improvements over simple acoustic-foam treatments, but are also very easy to install. Martin Walker

It always amazes me that so many musicians are prepared to spend Good value in the US, but thousands setting up their studios expensive in the UK and Europe. without investigating any sort of Much of the performance of treatment to sort out their acoustic Micro Traps can be duplicated problems. If there are peaks and with foam tiles at a lower UK troughs in the frequency response of cost. your monitoring environment, then you'll summary be misled while you're mixing and your Mini Traps look good, are finished tracks won't sound good on extremely easy to fit, and will other systems. cons

significantly improve the sound of any studio currently without acoustic treatment.

information Mini Traps, £280 each; Micro Traps, £220 each. Prices include VAT. Discounts are available when purchasing four or more units, as described in the main text. Sonic Distribution +44 (0) 1525 840400. +44 (0)1582 843901. Click here to email www.sonic-distribution. com www.realtraps.com

In the past, acoustic treatment often had to be done before installing your studio equipment. However, several companies including Auralex, Primacoustic, and Real Traps have now designed easy-to-use modular acoustic treatment systems which can be added after your gear has already been set up. While such systems don't provide a ruler-flat frequency response, they are a vast improvement over nothing at all!

Modular Acoustic Treatment Real Traps have been offering acoustic treatment panels in the US for a little while now, under the direction of Ethan Winer. Their first design was a panel absorber with a solid wood surround and back panel, and a flexible plywood front panel. Within the case, a fairly rigid fibreglass block (similar to loft insulation, but

file:///H|/SOS%2004-09/Real%20Traps%20Mini%20Trap%20&%20Micro%20Trap.htm (1 of 8)9/26/2005 12:04:56 AM


highly compressed) absorbed the incoming sound energy, as well as lowering the resonant frequency and 'Q' of the trap to make it useful over a wider frequency range. Real Traps made the absorber available as a set of stand-alone modular products with a standard two-foot width, tuned to centre frequencies of 60Hz, 100Hz, or 180Hz. Each was available in three heights to accommodate different ceilings. This design proved very popular in the US because the units could be easily hung on the walls of an existing studio from mounting bars, providing acoustic treatment with the minimum of down time. However, the panels were, of necessity, rather heavy, making them virtually impossible to ship around the world. The Mini Traps under review here take the modular concept further, but this time the panels have been made only 3.25 inches thick, bringing the weight down to just fifteen pounds, which is why Sonic Distribution here in the UK have been able to import them for the European market. This also makes them light enough to be hung from a picture hook, so installation becomes even quicker and easier. This time Ethan abandoned the low-Q tuned resonator approach in favour of using a hybrid of rigid fibreglass and membrane absorber that looks like a huge acoustic tile. The fibreglass block is mounted in a slotted metal frame with an feltlike acoustically transparent fabric covering, designed to expose the maximum amount of block to incoming sound. Since the fibreglass is three times as dense (six pounds per cubic foot) as most competing foam products, it's also claimed to be far more effective at absorbing lower frequencies. Mini Traps can absorb a wide range of frequencies right down to about 90Hz, below which their effectiveness falls off, although they still provide some useful absorption right down to 50Hz. Although understandably cagey about revealing his design secrets, Ethan did mention to me that his custom-manufactured fibreglass panels have a dual-layer membrane bonded to them that increases their absorption up to three times at the lowest frequencies. At the same time, the front surface is intentionally semireflective at higher frequencies, so you can use enough traps to cure your lowend problems without killing the high end (as many people do by covering their studio walls with egg boxes or thin acoustic tiles). In fact, the Mini Trap design has proved to be so effective that Ethan has now discontinued his original panel traps!



The Long & The Short Of It This review took me longer than expected, because after some initial acoustic tests I ended up turning my entire studio contents round by ninety degrees! I hasten to add that this wasn't because of the Mini Traps, but it certainly helped them to perform even better. Since you may face the same scenario, let me explain why I did this. To get the best stereo imaging, you should always set up your gear as far as possible such that each side of the room is a mirror image of the other about your listening position — this ensures that the reflections from the side walls are similar. However, this often leaves a choice of whether to point your speakers down the short or long dimension of the room.

Before you get serious about acoustic treatment, it's worth trying your gear in different orientations. As you can see quite clearly from this frequency response plot in my studio using minimal acoustic treatment, pointing the speakers down the shorter dimension (green trace) resulted in terrible dips at 70Hz and 120Hz, and these remained no matter how much I adjusted both speaker and listening positions. Turning the studio round 90 degrees so the speakers fired into the longer dimension (white trace) reduced these frequency dips significantly, even before acoustic treatment.

Firing into the short dimension means the side walls are further away from the speakers, thus reducing the impact of side reflections, and with my original setup it also meant I could face a huge double-glazed window, giving me a nice view of the garden! However, it also meant that the rear wall was closer to the speakers and, sadly, a few measurements with Rightmark's Audio Analyser showed that the reflections from this wall were causing problems in my small 12.5 x 10 x 7.25-foot studio. Therefore the garden view had to go, and the local plumber had to be called to move a radiator so that I could set up my gear firing down the long dimension. The difference was remarkable, giving the Mini Traps a much flatter initial response to work on — sometimes some initial reorganisation can reap huge acoustic dividends!

Installation Sonic Distribution sent me four Mini Traps in two huge boxes, plus a further box of five even lighter and thinner (1.35-inch) Micro Traps. Both traps are available in either black or white to suit your particular studio decor, and I chose the white. The traps looked very smart once I'd got them unpacked, and were light enough for one to be carried in each hand. I also found them less obtrusive compared to some of the more colourful moulded-foam designs, which is an important consideration if you want to install them in a living room, for instance. The ideal place for the Mini Traps in most small studios is in the corners, since file:///H|/SOS%2004-09/Real%20Traps%20Mini%20Trap%20&%20Micro%20Trap.htm (3 of 8)9/26/2005 12:04:56 AM


this is where you normally find the most bass energy for them to work on. However, unlike panel traps (the main working surface of which is the front panel and which are therefore best placed flat against a wall), Mini Traps work more effectively if you place them across a corner, since the air can then get at the back as well as the front, thus doubling the active surface area and extending absorption to lower frequencies. Because of their comparatively light weight, the traps can be hung horizontally or vertically using normal picture wire threaded through one of the six slots spaced across the rear mounting bars. They will need a fixing more substantial than a picture hook to support them, but a single screw and wall plug did the job for me. These waterfall plots from 20Hz to 200Hz were taken before (left) and after (right) installing four Mini Traps. You can see the initial ringing at 43Hz, 86Hz, 130Hz, and 180Hz — there are long 'plumes' at these frequencies carrying on forwards into the plot. After the Mini Traps had been fitted, the top three of these all but disappeared, and even the one at 43Hz became noticeably shorter. The dips between the various peaks were also less pronounced, resulting in a tighter and cleaner bass end.

In many rooms, placing the traps vertically across the four corners at about head height is the best option, but they can also be hung across the upper wall/ceiling boundary if you've got a door in the way, or in a ceiling corner. If you need more than four Mini Traps (and the more you install, the flatter your bass response is likely to get) you can fit two in each corner by mounting one vertically at head height and a second horizontally at ceiling height. You can also mount them on side walls or the ceiling, to treat the main wall reflections. Micro Traps are suitable for this role if your bass end is already OK, even though their effectiveness falls off significantly below about 300Hz. Once again, wall-mounted traps will be significantly more effective if you space them away from the wall so that their rear surfaces become active. With ceiling traps you can just suspend them using longer wires, but for the walls the easiest thing is to place the three-inch-thick foam blocks found at each end of the packaging behind the traps. There's a detailed instruction sheet with each box of Mini Traps to explain the various options.



Standing Waves Nowadays there are some excellent nearfield monitors available, but you won't really hear their quality in an untreated room. Each of the room dimensions (width, depth, and height) will result in standing waves at a specific frequency, along with all its harmonics. A simple formula to calculate the lowest mode in Ethan Winer's Modecalc utility will plot your Hertz is 6720Hz/L, where 'L' is the room modes and highlight any problem length (in inches) being considered. areas. In my studio the dimensions are such So, for instance, if your room is eight feet wide, it will have its lowest that in the most difficult area to treat (below about 200Hz) the modes are reasonably mode at 6720Hz/96, or 70Hz, plus evenly spaced, and bunching only occurs at there'll be modes at 140Hz, 210Hz, around 220Hz, 400Hz, and 450Hz. The 280Hz, and so on. You get another worst shape is a cube, since all three modal set of modes determined by the sets will be identical, so you'll get huge room depth, and a third set problems at these frequencies. according to its height. Apart from these axial modes, you also get tangential modes that involve two dimensions, and oblique modes that involve three, but the axial ones are the strongest). Since the wavelengths at these frequencies fit exactly into the room, you'll find much higher sound levels at the room boundaries (and particularly in the corners, where two or more sets of modes coincide), plus a series of low and high levels (minima and maxima) spaced between the boundaries. If you're lucky, or have built the room with acoustics in mind, the three sets of Axial modes will be smoothly spaced from the lowest room-mode frequency to several hundred Hertz, beyond which their numerous peaks and troughs can be effectively treated using acoustic tiles. If not, some frequencies from the different mode series will coincide, causing even larger peaks and troughs in your frequency response. The very worst case is a cubic room, since all three mode series will be identical, but thankfully there are published lists of preferred ratios for room dimensions that ensure relatively smoothly spaced modes. Ethan Winer himself has written a tiny Modecalc utility for Windows and DOS into which you can enter the dimensions of your room and see a simple graphic display of the axial mode spacing (and any pile-ups), as well as a selection of preferred ratios. You can download Modecalc for free from www.ethanwiner.com. This will help you identify the main problem areas, although your bass response will also depend on where you sit within the room, and therefore the distance of your head from the various room boundaries. Whatever frequency you choose, there will be a partial null point at one quarter wavelength from each wall, which explains why moving your listening position can also shift the frequencies of peaks and dips in the measured response — not all problems are solely due to room modes.

Testing The Space



Because I was reassembling my studio from scratch, I had the opportunity to measure the room acoustics in a fairly empty state, using some sine-wave sweeps courtesy of Rightmark's Audio Analyser. I could also perform waterfall tests using ETF Software's ETF 5. My ATC SCM10 speakers have a very good reputation for flat response, so I wasn't concerned about them clouding the results. As I had expected, my small studio sounded pretty 'honky' without any acoustic treatment, ringing like a bell at some frequencies. The worst resonant peaks proved to be at 43Hz, 130Hz, and 180Hz (all related to the front/back room dimension), with a further peak at 80Hz (because of the ceiling height) and dips at both 56Hz and 113Hz (because of the side-to-side room dimension). One of the beauties of the Mini Traps is the instant gratification — you can prop them roughly in position within a few seconds and listen to the improvements. According to Ethan's lab tests, they work particularly well between about 90Hz and 150Hz if placed across the corners, so I wasn't expecting them to cure the 43Hz problem, but even after placing just two Mini Traps across the front corners, the modes at 80Hz, 130Hz, and 180Hz were far more controlled, showing a much shorter overhang on ETF 5's low-frequency waterfall plot. Adding the other two traps in the rear corners improved these results still further. However, the traps also made a tiny, but nevertheless measurable, improvement to the 43Hz mode, which is pretty impressive for such a compact solution! A flatter response would need significantly more treatment, but subjectively my room had already been transformed — the bass end was now significantly tighter and smoother. Most users will get good performance from the Mini Traps with this corner placement, but since I was already impressed enough to seriously consider buying the review models I spent a considerable amount of time fine-tuning their positions, and found moving them by even six inches resulted in slightly different results. After further careful adjustments in my studio, I ended up with the first two traps in the front corners suspended about eighteen inches from the ceiling; the third slightly higher in one of the rear corners; and the fourth midway between the speakers on the front wall. A door in the fourth corner meant that I couldn't place a Mini Trap there, and putting it across the ceiling/wall boundary proved less effective. The Micro Traps were very useful in taming mid-range and high-end reflections above about 500Hz, and are rather less obtrusive than similarly performing acoustic tiles of several times the thickness, although they are significantly more expensive here in the UK. However, for me, the Mini Traps are the stars of the range.



Acoustic Absorption Unfortunately, however well spaced they are, your room modes will still cause peaks and dips in your frequency response unless you treat them. This needs absorption to soak up some of the extra energy at problem frequencies (mostly below about 250Hz), stopping it being reflected back into the room. Many musicians place a comfy sofa at the back of the room, which certainly helps with the rear reflections, but the best place for absorption is where the energy levels are highest, which tends to be in the corners of the room where modes from the different dimensions 'pile up'. Anything placed here will improve ringing at these frequencies and even out the frequency response.

As you can see from this graph of Absorption versus Frequency, the response of the Micro Traps is similar to normal three-inch thick foam tiles, so these are therefore more suitable for wall-mounting to tame side-wall and ceiling early reflections. The Mini Traps, on the other hand, are effective down to about 200Hz when mounted flat against a wall, and down to below 100Hz when mounted across a corner.

Measuring the absorption of a particular material at different frequencies isn't easy without proper lab facilities, and even then differing testing methods can give rather different results, particularly at frequencies below 250Hz, where the effectiveness of thinner and more lightweight acoustic materials falls off. As I had some Melatech acoustic foam tiles in my studio of the same dimensions (four foot by two foot by four inches thick) as the Mini Traps, I tried them in exactly the same positions, and can confirm that they had little effect below 250Hz, although I can't personally comment on other products using higher-density foam. However, when having his Mini Traps and Micro Traps lab-tested, Ethan went to the extra expense of having some rival foam-based products measured under exactly the same conditions to provide the more comparative results you can see in the graph below.

Trap Star Or Pony & Trap? Over the last fifteen years I've tried a variety of modular solutions to bass trapping, including resonant tubes and panel traps, and none of them have had performance anywhere near that of the Mini Traps. The Mini Traps looked good, were incredibly easy to fit, and made a significant improvement to my studio acoustics at the most troublesome frequencies — they even had some effect down to about 50Hz! The Micro Traps are also very useful for treating wall reflections, and don't have to be glued in place, making both products ideal for studios where appearance is important and where the treatment may need to be taken down or moved at some point.



In the US you can buy Mini Traps direct from Real Traps for $179.99 each (about £100 in the UK), and Micro Traps for $119.99 each (around £67), plus shipping charges at cost (generally around $20 per trap). This seems very reasonable. Unfortunately, European readers face much higher shipping costs, because the Mini Traps are just so bulky to import, resulting in a UK price of £252 each when buying four Mini Traps at a time (the realistic minimum quantity), or £238 each for eight or more. Even if you were able to import them into the UK yourself, the price for each of eight Mini Traps would end up about £222 once you added shipping, duty, and VAT. However, Sonic Distribution provide free shipping anywhere in the UK or Europe, and they also offer a free planning service to work out the best places for traps in your own studio. Furthermore, they have a loan scheme so that you can try out any of their products that don't need permanent fixing, so you can see how effective the MIni Traps are in your own studio at no risk. Ultimately only you can decide how much the Mini Traps are worth to your studio. Despite costing about the same in the UK as an esoteric mic channel, four Mini Traps would probably prove far more beneficial overall for your studio. You can of course try a DIY approach by buying four-inch-thick high-density rockwool slabs and covering them in muslin or hessian. However, the most appropriate materials often seem to be hard to find locally, can cause skin irritation until covered, and furthermore (judging by various comments I've found on web forums from users who made DIY versions and compared them to the real Mini Traps), the latter are significantly more effective, considerably smarter, and far more convenient! I've put my money where my mouth is, and not only are the four review Mini Traps staying put in my studio, I've also added another two!




Roland Fantom X

In this article:

Roland Fantom X

X-Rated Models Synth Workstation What's New? Published in SOS September 2004 Polyphony Fantom X Editor/Librarian Print article : Close window Pads Reviews : Keyboard workstation Fantom Memories Live Setting In Use Layer Screen & Mixer Screen Chord Memory Roland's original Fantom workstation wasn't really Sampling much competition for Korg's Triton, so it was no shock Sequencer to see it revamped last year as the hugely improved Effects Fantom S. But surprisingly, Roland have done it again. Conclusions

What's new this time around?

Roland Fantom X6/X7/ X8 £1599/£1999/£2299 pros 128-note polyphony. Sample memory expandable to a massive 544MB. 32MB Flash memory provided. Trigger pads are a joy to perform with. Quality piano samples.

cons Fewer effects than rivals. Loop points of imported samples are not recognised. Too many presets and too few user locations — especially in the case of multitimbral Performances.

summary The Fantom X has grown into a well-specified workstation with enough individuality to distinguish itself from the other 'big two' Japanese rivals. Its pads, in particular, offer the performing musician that little bit extra and the colour screen is easy on the eye.

Paul Nagle

When Roland released the Fantom workstation in 2002, it seemed destined for an uphill struggle against rivals that offered user sampling and a greater number of simultaneous effects. Undeterred, Roland returned from the drawing board the following year with the Fantom S — declaring, in no uncertain terms, that the Sampling angle was now Sorted. However, the story didn't end there: this year marked a new chapter, with the Fantom X entering the fray. Have we reached the final stage in this powerful workstation's evolution, and should the others tremble on their X-stands? Let's find out... Photos: Mark Ewing

X-Rated Models The new Fantom is available in three keyboard models, the X6, X7 and X8, which are named according to keyboard length (61, 76 or 88 notes). Although I didn't have this for my review, a 1U rack equivalent, the Fantom XR, is also available, and this has several architectural differences — there's no sequencer, two more expansion slots, and half the amount of flash RAM. The earlier Fantom S lacked a 76-note version, and its 88-note hammer-action

file:///H|/SOS%2004-09/Roland%20Fantom%20X.htm (1 of 10)9/26/2005 12:05:01 AM

Roland Fantom X

information Fantom X6, £1599; Fantom X7, £1999; Fantom X8, £2299; Fantom XR (rack version), £999. All prices include VAT. Roland UK +44 (0)1792 515020. +44 (0)1792 799644. www.roland.co.uk www.roland.co.jp

model was paired with a dedicated Piano Mode not present on its five-octave sibling. This mode offered parameters such as 'Sympathetic Resonance' and a simulation of the piano lid opening and closing. Happily, all three Fantom X keyboard models have identical specifications — and these include Piano Mode. I recommend that you read Nick Magnus's comprehensive write-up of the Fantom S (see SOS October 2003, or surf to: www.soundonsound.com/sos/ oct03/articles/rolandfantoms.htm). I will concentrate here on the key aspects that have been revised, and my comparisons throughout this review will be with the S model rather than the original, non-sampling Fantom (which I reviewed back in February 2002 — see www.soundonsound.com/sos/feb02/articles/fantom0202. asp).

Test Spec Fantom OS version reviewed: v1.0.

What's New? The central 320x240 display is angled, so there's no need to lean forward to view it clearly. But it's when you power on that the X factor really makes itself known — in a swirl of animated colour graphics. Roland claim this as a first for high-end synth workstations, which I believe it is, although well-specified arranger keyboards have had them for quite a while — I seem to recall the Technics KN5000 having a colour screen as far back as 1998. Prior to receiving the synth, I wondered whether colour would be more of a gimmick than a real step forward. However, when faced with the reality, I quickly acknowledged the benefits — not least because each screen can pack in more information before it looks cramped or confusing. Even the Performance Mixer screen, with its 16 sliders and many small buttons, is useable, but had it been rendered in monochrome instead, I doubt this would have been the case.

It's nice to see a colour display finally reach a series of high-end synth workstations, as they've been on high-spec arranger keyboards for a few years. The function buttons still don't quite line up under their onscreen tabs, though!

Other than the aforementioned Piano Mode button and its paler shade of silver, the remainder of the front panel is very similar to the Fantom S. Thus Roland's standard pitch-bender and a DBeam are present, along with four real-time control knobs, two assignable buttons, octave-transpose buttons, backlit trigger pads, and sequencer and arpeggiator controls (see the pictures later in this article). Unless my eyes deceive me, the alignment of the 'soft keys' F1-F8 beneath the display seems once again to have been improved fractionally. I wonder if, in some future Fantom, they will finally line up perfectly?


Roland Fantom X

Turning to the rear panel, we discover the remaining external changes. Firstly, the digital optical output has been dropped and the co-axial output is now joined by a co-axial input, addressing one of Nick Magnus' complaints about the Fantom S and giving the X models true 24-bit S/PDIF I/O. The four audio outputs and control pedal inputs are unchanged and, of course, the five-pin MIDI trio is present. The Smart Media slot of the S model has been replaced by a more versatile PC Card slot, into which peripherals such as Smart Media or Compact Flash can be inserted (with a suitable adaptor). Finally, the USB connection looks unchanged, but it can now be used in either data or MIDI modes. You can therefore use it to transfer files to and from your computer, or as a MIDI interface to communicate with the supplied Editor and Librarian programs.

Polyphony Internally, the Fantom X's polyphony has been hiked up to 128 notes, courtesy of a new processor chip at the heart of the instrument, Roland's most powerful to date. For many, this will be the single most significant improvement over the older 64-note Fantoms. Some of the advertising blurb is slightly misleading, though. Take the following example; "it is possible to play expressive four-tone Patches with 128-voice polyphony". In fact, this doesn't mean you can layer four tones without penalty, but if you translate it to read "it is possible to play four tones set to non-overlapping velocity ranges with maximum polyphony" it is accurate — although a lot less snappy. Stereo samples result in halved polyphony, as you'd expect. A useful so-called Voice Monitor is accessed using Shift and function key F4. This is a dynamic representation of polyphony and one of several animated displays that really show off the screen graphics.

Fantom X Editor/Librarian Roland supply a software editor on CD with the Fantom X, plus a basic librarian. The software requires at least a 400MHz Pentium with 256MB of RAM, and will run on any operating system from Windows 98 onwards. Mac users should have at least a 233MHz G3 for Mac OS Classic and a 500Hz G3 if using OS X. Again, 256MB of RAM is recommended. The graphic editing of parameters is well implemented, although personally, I thought the synth's screen did the job just fine. Listed under 'Accessories' in the manual is a 'sample data convert tool'. Unfortunately, this is not yet available at the time of writing, so whether it will address Nick Magnus' complaints in the Fantom S review about the workstation's inability to import existing sample libraries, I can't say.

Pads


Roland Fantom X

The 16 backlit trigger pads of the Fantom S were (literally) a hit, and their sensitivity has been improved for the Fantom X. I certainly found them responsive enough in a variety of roles. They can trigger up to four samples in separate velocity zones and, better still, they generate polyphonic aftertouch (the keyboard itself transmits the more common channel aftertouch). The pads are not confined to triggering percussion or samples — they can play any synth patch you select. By programming each pad to trigger a specific note, you can build up chords, and here that polyphonic aftertouch can work wonders, emphasising notes within the chord in ways not possible on the keyboard. The Pad Hold button allows notes to be sustained so you can trigger drones or ambient washes of evolving sound. Or, when triggering sample loops, activating Hold will keep the loop running. You can record such performance tricks into the sequencer too, so building up a song that features a selection of loops is remarkably fast. RPS mode — Realtime Phrase Sequencing — is another useful performance aid. Once activated, the top eight pads can trigger RPS loops. Some useable patterns are supplied, or you can program your own. It seems Nick Magnus' complaint about the way pads are allocated globally in Patch mode has been partially answered by implementation of the Live Setting mode, which I'll discuss in more detail later. Nevertheless, it's a shame that pad assignments and note values cannot be made for every Patch.

Fantom Memories It is now possible to expand the sample memory to an impressive 544MB without compromising the fitting of four SRX-series expansion boards. With just one memory slot available, you will need to choose between adding a 64, 128, 256 or 512MB DIMM. As before, 32MB is supplied, giving approximately 180 seconds of stereo sampling straight out of the box. If you install a 512MB DIMM, you get a total of 54 minutes stereo or 108 minutes in mono. Then there are the ROM sound sources, which have been increased to 128MB, with 1480 waveforms to choose from. If this still isn't enough, don't forget those expansion slots. A fully maxed-out Fantom X can have almost 1GB of wave memory to draw from! User memory (separate from Sample memory) has been doubled: 32MB is provided for storage of samples, patches, Performances and Songs. As this is Flash memory, it remains populated after power-down and a System option determines whether user samples are automatically loaded when you switch on. Preset memory of an unexpanded Fantom X stands at 1024 patches, divided into banks A-H, each with 128 patches. On top of this, there are 256 GM2 (General MIDI 2) presets, nine GM2 kits, 40 preset and 32 user kits, with a further 32 available from a PC card.


Roland Fantom X

To store your own patches, there are 256 user locations, but just 64 memories for multitimbral/layered Performances. And there are a further 64 preset Performances — but does anyone ever use preset Performances? Unlike on the Korg Triton (to take one obvious example) where every patch location may be overwritten, if there are entries in the Fantom's preset memory you don't want, there's no way to delete them. Fortunately, Roland score highly in the provision of reliable bread-and-butter sounds, although perhaps they lag behind Korg and Yamaha in the exotic or gratuitously 'showy' stakes. The drum samples are well up to the standard expected from a company who have consistently led the field in creating classic beatboxes. I mentioned the PC card option earlier, and it's a flexible way to expand the datastorage capacity — you can store up to 7000 samples on a single card, for example. The Fantom X supports cards of up to 1GB. Curiously, even with such a large card installed, you can apparently only access an additional 256 user patches. I didn't have such a large card to check this, but coupled with the internal memories, the expanded total should probably suffice. More restrictive, again, is the number of user Performances — a card adds the capacity for a mere 64 extra. If, after a while, all these large numbers start to blur, you might begin to wonder whether some kind of practical limit will one day be reached: a point where packing in more and more of everything doesn't result in a better musical instrument. Perhaps in recognition of the value of simplicity, the Piano Mode button does exactly what it says on the tin. It offers direct access to two lists of sounds catalogued as Acoustic or Electric Piano (although you can sneakily catalogue your own sounds to appear in these lists, even if they're not pianobased). The pianos range from good to very good; most impressive of all is an 88note multisampled acoustic piano, where every note has been sampled in stereo, complete with four velocity-switched layers. Roland proudly declare that more than 700 individual samples are used to bring you their most detailed piano recreation to date. It is a delight to play, especially from a weighted keyboard. As for the remaining patches, the many hundreds of them become an exercise in cataloguing and retrieval. Fortunately, Roland provide several ways to speed up access to those you want — including Favorites (sic), Patch Categories and the curiously titled 'Live Setting'.

Live Setting Live Setting (accessed via the menu system or directly using Shift and the Piano Mode button) is a means of registering up to 20 banks of 16 patch/rhythm combinations. For greater convenience, you can register Performances too, and access them in exactly the same way. In each bank, the eight function keys below the display are used to select the first eight Live Setting selections and using Shift, the second eight are available.


Roland Fantom X

Though the name may initially seem odd, it makes sense when you consider that Live Setting provides an ideal means of gathering together everything needed for a live set. Selection of the Performance or Patch/Rhythm pairing is made even slicker as text files can be imported and associated with each Bank. This is so handy, since the text can be anything you would find useful: lyrics, jokes, reminders and so on. The text appears in a scrollable window for each Bank and is far better and far easier to read than bits of paper stuck to your synth with gaffer tape!

In Use At every level, shortcuts are implemented to make operation as speedy as possible. For example, on many screens, a Zoom edit softkey opens an expanded graphical window of parameters, often adjustable with the four assignable knobs. The Fantom X makes extensive use of the Shift key to access menus you will need regularly. The pad Clip Board is another great timesaver, and may be customised to take you directly into 16 of your favourite screens. To register a shortcut to your current location within the menus, hold down the Clip Board button. Any pads already assigned are then illuminated, and you complete the assignment by touching the pad button of your choice. Later, to jump directly to any stored shortcut, hit and release the Clip Board button, then select the appropriate (flashing) pad. A visual representation of all 16 screen shortcuts is a handy reminder of the choices. I noticed that it was wisest not to try to navigate through some of the menus whilst the synth was receiving heavy amounts of data. At one point during the review period, I happened to be sending 16 MIDI channels of dense, quantised notes to the synth at the same time as I was poking about in various System menus. In this instance, the Fantom X struggled to juggle the incoming notes with the screen updates. Under more normal usage conditions, however, I had no complaints about timing or response, and this was after some pretty heavy-duty testing!

Layer Screen & Mixer Screen Multitimbral Performances are accessed by pushing either the Layer or Mixer buttons. The Mixer shows all 16 channels on a single screen and is packed with on-screen sliders, knobs and buttons. A small floating rectangle represents those parameters whose value may be tweaked by the four assignable knobs. As you file:///H|/SOS%2004-09/Roland%20Fantom%20X.htm (6 of 10)9/26/2005 12:05:01 AM

Roland Fantom X

can record the slider movements into the sequencer, you have an attractive and valuable visual reference of a complete automated mix. One final example of the way shortcuts are implemented is worth a mention here: hit the Mute softkey from the Mixer screen and the pads may then be used as a direct means to mute or unmute any of the 16 parts. The Layer screen is equally informative, although it shows just four layers at any time. An arrow indicates the currently selected layer and to play several patches at once, you add a tick to the 'Kbd' box. Velocity and keyboard zones can be set up quickly, although I'd have liked to have the ability to set these using the keyboard rather than the data-entry dial.

Chord Memory Chord Memory is a feature I've always enjoyed, and this implementation is quietly sophisticated. There are 64 Preset and 64 user-programmable chords, and chord selections may be stored with Performances — although not with Patches. To select a chord to play, or to edit the existing chord, use Shift and the Chord Memory button. In this screen also, you can activate 'Strum' mode to create strummed chords from a single key, the speed varying according to how hard you play. Chords can be strummed up, down or alternately up and down. If you create a complex chord, then apply it to an acoustic guitar or harp patch, the results are addictively pleasurable. Each note in a chord is represented on a music stave, so there's educational value too. Activating the arpeggiator and Chord Memory at the same time gives arpeggiated chords, and you'll be pleased to know that arpeggios and strums can be recorded into the sequencer.

Sampling Sampling is largely unchanged from the Fantom S, although the colour screen does make sample editing slightly prettier. In the Fantom X, you have the additional option to sample from the digital input, after which the comprehensive array of tools can be applied. As described in the review of the Fantom S, you can normalise samples, chop and divide them over the keyboard, combine them, stretch them, and do plenty more besides. Of particular note is the Real-time Time-stretch tool, which is perfect for stretching sampled loops. To use it, you must specify the tempo of the loop and the time-stretch type from 10 different choices which optimise the sample according to whether it contains fast or slow phrases. Then if you turn on the patch's Wave Tempo Sync parameter, the waveform playback speed can be altered without affecting the pitch. The results are pretty good in all but the most extreme cases, and even then, they often fall into the 'interesting' category. As on the Fantom S, there's also Skip Back Sampling, in which the sampler file:///H|/SOS%2004-09/Roland%20Fantom%20X.htm (7 of 10)9/26/2005 12:05:01 AM

Roland Fantom X

constantly records in chunks of up to 40 seconds. So if you play something of interest, hit the glowing blue button. You can then assign the resulting sample to a pad, to the keyboard, or just save it for later. The SOS Fantom S review covered issues and niggles regarding sample import (in WAV or AIFF format) and memory handling, and I've nothing to add here, except to echo the frustration that loop points in imported samples are ignored. The increases in User memory size and PC Card capacity rendered the whole issue of sampling so painless that I rarely considered the memory complications explained so thoroughly in Nick Magnus's review. I found the Fantom's implementation of sampling to be spot-on — and the automated loading of samples on boot-up was a huge improvement over the manual method required by my Korg Triton.

Sequencer Other than an increased capacity (now 400,000 notes), the sequencer maintains the reassuringly familiar 16-track format. Only one song may be present in memory at once, although up to 256 songs can be saved to a PC Card or to user memory. A 'Quick Play' function is a handy means to play such songs directly without loading them into volatile memory. The sequencer is well specified, and, other than graphic controller editing, has everything you would expect — and a few features you might not (for example Quantise Templates and SysEx editing). It is, however, MIDI-only. There's no attempt to incorporate audio, other than by triggering samples either from the pads or keyboard. However, you can sample an external instrument as the song plays back, creating a single WAV of the resulting output. It's not hard disk recording — and you certainly daren't make any mistakes — but it works.

Effects There's a selection of 78 multi-effects (MFX), three choruses, five reverbs, six input effects and a three-band mastering compressor. The eagle-eyed amongst you will have spotted here that the Fantom X gains one extra MFX over the S model. This is 'Sympathetic Resonance' — an integral part of the Piano Mode implementation, but here presented as a separate effect in its own right. The effects sound great, and include algorithms like COSM guitar-amp modelling, lo-fi processing, tempo-sync'ed delays; all Roland's usual high-quality favourites. In Performance mode, the three MFX can be configured for parallel or serial use — or various combinations in between. However, in Patch mode, just a single MFX can be assigned to the keyboard and a second to the pads, which is rather


Roland Fantom X

limiting. When you're ready to mix down a song, the dedicated Mastering processor, with its multi-band compression, works well to create tight, punchy mixes.

Conclusions Having used the Fantom X for some time, I have to admit to being a convert to its colour screen. It does make a difference, although given the choice between a colour screen and the excellent touchscreen of, say, Roland's own V-Synth, I'd still opt for the touchscreen! I'm sure the doubling of the polyphony is going to be the single most appreciated benefit over the previous model, but this was required in order to play catch up with Yamaha's Motif ES and Korg's Triton Extreme. In their quest to compete with Korg and Yamaha, Roland have looked at ways in which the workstation concept can be refined. The addition of the pad bank offers several performance advantages — whether used to trigger samples, drones or loops. For its pairing of Patches with Pad assignments, I therefore found the Live Setting feature very welcome indeed. My disappointments were few, but of these, having just three multi-effects compares poorly with the Triton and the Motif ES with their five and eight insert effects respectively; both Korg and Yamaha use their effects well to give a superior finish to every sound. I would also have liked a few more user Performances — or to dispense with the supplied preset patches entirely.

The X range is the first Fantom family to come in 61-, 76-, and 88-note versions, as well as the XR rack version (not pictured).

For those who still prefer hardware sequencers, the Fantom X boasts a tried and tested design that is both easy to use and well specified. However, it makes no attempt to integrate audio, and there's no equivalent of Korg's 'In Track Sampling'. Finished songs can be sampled directly though, and the resulting WAVs may then be transferred across the USB connection for burning to CD on your computer. If you are already using your computer for this purpose, it's quite logical. Yet other workstations offer the facility to create CDs directly, and this may be a purchase factor. Many will find Roland's implementation of sampling to be exactly to their taste — and I did. It's straightforward to use and the range of editing on offer covers all the bases. Addition of a large-capacity PC card would give access to a huge


Roland Fantom X

selection of user samples. In fact, the only gripe I had echoes that comment from the Fantom S review, about loop points of imported samples being ignored. For anyone already happy with their workstation, whether the Fantom S, Triton or Motif, the improvements present in the Fantom X will probably cause no great rush to the SOS Readers' Ads section. However, for those who are workstationshopping today, it is a classy-sounding, expandable instrument worthy of serious consideration. Published in SOS September 2004 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



Soundfield SPS422B & SP451

In this article:

SPS422B Microphone System SP451 Surround Processor The Soundfield System In Use Conclusion

Soundfield SPS422B & SP451 Surround Recording System Published in SOS September 2004 Print article : Close window

Reviews : Microphone

Soundfield SPS422B & SP451 £3143/£2321 pros Although there's only one mic to set up, you can instantly audition a variety of different coincident multimiking configurations. The B-format signal can be converted into any current or future surround format, and can be recorded using only four tracks. Versatile transcoding of Bformat signals using the SP451.

This system can record in mono, stereo, or any current or future surround format — using just one mic! Hugh Robjohns

The Soundfield microphone has, in the quarter century since its inception, acquired a reputation as one of the finest and most versatile microphones in the world. Although many wellcons known recording engineers seem content to use it as a top-flight mono The operation of the frontPhotos: Mark Ewing panel controls can take some vocal microphone — ignoring its stereo working out when you're using and multi-channel capabilities the unit for the first time. completely — it is as a multi-channel device that the Soundfield microphone summary stands head and shoulders above pretty much any other system. A versatile recording package which will record anything from mono to 7.1 surround. The B-format outputs provide future-proofing for your surround recordings.

information SPS422B, £3143.13; SP451, £2320.63. Prices include VAT. Drawmer Distribution +44 (0)1924 378669. +44 (0)1924 290460. Click here to email www.drawmer.co.uk

The subject of this review is a surround sound recording system comprising the SPS422B and SP451. The package is realistically priced in the UK, and provides a very versatile and easy-to-use system for the acquisition of surround sound, with the bonus of being track-efficient and both backwards and forwards compatible — recordings made in Soundfield's B-format can be used as the source for mono, stereo, LCRS, 5.1, 6.1, 7.1, 10.2, or any other format you care to dream up!

SPS422B Microphone System The SPS422B comprises a Soundfield multi-capsule microphone and its 1U rackmounting controller — the mic and controller are matched to one another during manufacture, so you can't swap mics and controllers. The microphone

file:///H|/SOS%2004-09/Soundfield%20SPS422B%20&%20SP451.htm (1 of 5)9/26/2005 12:05:08 AM


connects to the rear panel of the controller, and the balanced line-level outputs are presented on six XLRs: a main stereo output plus the four B-format signals. The front panel is laid out clearly, with just five rotary controls, four buttons, and a large dual bar-graph meter. To the left of the meter, two sets of control functions affect the B-format signal. A rotary switch and continuous rotary control determine coarse gain (switched in 10dB steps from zero to -30dB) and fine gain (over a ±10dB range), while two illuminated buttons configure the unit for the particular microphone orientation in use (inverted and end-fire modes). To the right of the meter, the controls adjust the stereo output. It is worth remembering that the unit actually works internally with M&S signals, so the two rotary controls to the right of the meter adjust the nominal polar pattern of the Middle channel, and the level of the Sides channel. Together these controls effectively determine the mutual angle and polar patterns of The SP442B microphone the equivalent stereo pair. The output can be left in system includes this multiM&S mode, or converted to normal left-right stereo, capsule microphone. The via a button — the current mode is indicated on the rack unit accepts the mic's bar-graph meters' legends. A second button selects output and converts it to an 80Hz high-pass filter. The final rotary control is a stereo and B-format signals. headphone monitor level control adjacent to the stereo headphone socket. The bar-graph meters have a VU ballistic and are scaled from -30dBu to +21dBu. The action of the Pattern and Width controls can be confusing to those not versed in M&S theory, but help is at hand in the form of a small program which can be downloaded from the Soundfield web site. This provides an interactive graphical illustration of the effects of these controls, both on the source M&S virtual setup, and on the equivalent left-right microphones. After ten minutes of playing, the system becomes blindingly obvious!

SP451 Surround Processor The SP451 has a very simple set of interconnections: four XLRs for the balanced line-level B-format inputs, and eight XLRs for the line-level transcoded surroundsound outputs. The review unit was set up to provide a 5.1 surround-sound output, so the last two XLRs are unused. However, with an appropriate decoder card (see below) they can be used to provide outputs for a 6.1 or 7.1 system. Like that of the SPS422B, the front of the SP451 is very clear and simple. Starting at the left, a master input level control adjusts all four B-format signals simultaneously (from full attenuation to 10dB gain), and the result is shown on file:///H|/SOS%2004-09/Soundfield%20SPS422B%20&%20SP451.htm (2 of 5)9/26/2005 12:05:08 AM


four bar-graph meters. To the right of these meters, three rotary controls adjust various aspects of the transcoding. The system uses the B-format input information to create six virtual coincident microphones — all with cardioid polar patterns by default — to feed the 5.1 surround outputs. The first knob is labelled Front Width and alters the effective mutual angle of the microphones which feed the left and right front loudspeakers. The range is from zero degrees (mono) through to ±90 degrees (relative to the centre), with a Cal position equating to ±45 degrees. The second control, Rear Width, does exactly the same for the rear left and right microphones, with the default position equating to angles of ±135 degrees. The third control is labelled Rear Focus and alters the effective polar patterns of the two rear microphones. The default is cardioid, but the range spans omni to figure of eight. Together, these three controls provide a remarkable degree of control over the nature of the surround sound stage by determining the frontal width, the rear width, and the ambience. The Rear Focus control is very useful to 'blur' the rearward definition by choosing a pattern which captures more spatial information, for example. The right-hand section of the front panel is concerned with the output signals. Six rotary controls, each with an associated bar-graph meter, are provided to adjust the output level (from fully off to +10dB) of each of the first six surround outputs. An alphanumeric display above each meter identifies each channel, depending on the currently selected transcoding mode. In the review model the controls were labelled Front Left, Centre, Front Right, Rear Left, Rear Right, and Subbass. The transcoding mode is determined by one of up to three plug-in MAP (Microphone Array Pattern) cards. The card supplied with the unit transcodes the B-format signals to five coincident cardioids plus an omni for sub-bass output below 120Hz. However, other formats are available to order, including options to generate 6.1 or 7.1. Three illuminated buttons are provided to select one of the three possible MAP cards, enabling instant comparison between different transcoding configurations, but since the review model only had the single card I was able to try this particular facility.

The Soundfield System In Use Since I have owned my own SPS422B Soundfield microphone system for a couple of years now, the basic operation of the microphone was already familiar to me. It is worth mentioning that for the novice there is a bit of a learning curve involved in getting to grips with the width and pattern controls, since these are not entirely intuitive. However, they are the key to controlling the stereo sound stage output from the SPS422B itself — although they play no part whatever in the context of the subsequent surround-sound transcoding from the B-format outputs. The other practical consideration concerns the meters. As standard, file:///H|/SOS%2004-09/Soundfield%20SPS422B%20&%20SP451.htm (3 of 5)9/26/2005 12:05:08 AM


these have a kind of VU response which can under-read considerably on fast transients. A rather percussive piano recording, which was peaking around +1dB on the meters, was actually causing transient overloads of the front end of the unit, so care should be taken with this aspect of setting the input level. The B-format signals can be recorded directly to a four-track machine for subsequent processing and decoding to whatever format is required. Clearly, The rear panels of the SP451 (top) and the four B-format signals have critical SPS422B (bottom) rackmount units. phase relationships which must be maintained, but this should not present problems with modern multi-channel digital recorders. In addition, if a further two tracks are available, then the stereo outputs from the SPS422B can also be recorded — perhaps providing a stereo track for editing and a B-format surround track for subsequent post-production. Needless to say, because the stereo (and surround) outputs are derived from a point-source microphone, the signals can be folded down without risk of phase errors: stereo to mono, and surround to LCRS, stereo, or mono. I recorded and replayed the B-format signals successfully using both SADiE and a Genex recorder, and hooking either up to the SPS422B and SP451 presented no problems at all. I used the SP451 as a kind of preamp, the B-format input coming from the recorder and the six outputs directly feeding the Bryston amplifiers of a surround-sound system comprising a collection of PMC speakers (IB1s left and right, LB1 centre and TB1s at the rear). This arrangement worked well thanks to the full attenuation capability of the input gain control of the processor, but in more normal usage the processor would probably be feeding a post-production console. With the default conditions I found the transcoding worked superbly, recreating a very precise and all-enveloping sound stage. However, adjusting the SP451's controls enabled a wide range of different sound stage characters to be created, from accurate and naturalistic to diffuse and ambient. It was also possible to create some dramatic effects — simulating walking through a long corridor and out onto a stage, for example — by starting with front and rear widths set to zero, and then winding them up to exit the corridor.

Conclusion The combination of SPS422B and SP451 is a very powerful tool for the capture and reproduction of surround sound, and is ideal for anyone involved in the recording of music or radio drama, for example, or indeed for the acquisition of atmospheres and sound effects, easily accommodating all present and future formats in a simple and compact form. Not only is the imaging extremely precise (if required), in much the same way as that of a conventional coincident stereo pair, but the surround stage can be recorded on just four tracks (instead of six or file:///H|/SOS%2004-09/Soundfield%20SPS422B%20&%20SP451.htm (4 of 5)9/26/2005 12:05:08 AM


more), and can be easily converted for any surround sound format, including formats involving a height element. Furthermore, although the facility is not present with this specific equipment, it is possible to post-produce B-format recordings to control the orientation and azimuth of the virtual microphone array, which could prove very useful in some circumstances. As with coincident stereo systems, additional sound elements can be mixed in with the decoded B-format signals to create a fully mono- and stereo-compatible mix — something which is often not the case when using surround atmospheres captured with spaced microphone arrays. Published in SOS September 2004 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


file:///H|/SOS%2004-09/Soundfield%20SPS422B%20&%20SP451.htm (5 of 5)9/26/2005 12:05:08 AM

TL Audio Ivory 2 5060

In this article:

Compression With Presets Front-panel Controls Session Impressions Dial-a-squeeze

TL Audio Ivory 2 5060 £469

TL Audio Ivory 2 5060 Valve Compressor Published in SOS September 2004 Print article : Close window

Reviews : Processor

pros Easy to set up, with welldesigned presets. Manual operation catered for. Includes a good mic preamp and two instrument inputs.

cons The use of a moving-coil meter makes it harder to judge the amount of gain reduction being applied.

This new Ivory-series preamp and compressor combines the ease of presets with the flexibility of manual operation. Paul White

TL Audio's latest Ivory-2 machine is once again based on a combination of summary things the company has done well in A great compromise between the past, packaged in a usable and a fully manual and a fully affordable form — not a bad formula, preset compressor — you get Photos: Mike Cameron come to think of it! Everything is very both! familiar, from the 2U ivory-coloured information panel to the legending and the style of the knobs and switches. Power comes in 5060, £468.83; DO2 via an IEC connector, not an annoying wall-wart, and the I/O levels are optional digital output board, switchable to accommodate both pro and semi-pro audio levels. A circular VU £149.22. Prices include VAT. meter, a circular mains switch, and milled ventilation slots give the unit a classy TL Audio +44 (0)1462 look without things becoming fussy. 492090. +44 (0)1462 492097. Click here to email www.tlaudio.co.uk

Compression With Presets At its heart, the 5060 is a stereo compressor using hybrid solid-state and tube circuitry incorporating a transconductance variable-gain element, but although it isn't designed primarily to be a 'voice channel' product, it includes one of TL Audio's very nice mic preamps with its input XLR tucked away around the back and with switchable phantom power. As a voice channel, the unit is strictly mono, and in stereo line or instrument operation it must function as a stereo unit, not as two mono units — there's only one set of controls and only one side-chain. Because the tube stages (half a dual-triode ECC83 in each channel) come between the input stage and the compressor, they can be deliberately driven hard by setting the input gain levels on the high side, if you want to add some of

file:///H|/SOS%2004-09/TL%20Audio%20Ivory%202%205060.htm (1 of 4)9/26/2005 12:05:12 AM


that old tube warmth. However, there's often a fine margin between the tube's Drive lamp flickering and the Peak clip LED coming on, so you do need to take care setting the input gain. As the tubes are properly run from relatively highvoltage supplies (100V stabilised), their overload characteristics are both subtle and authentic. There are two high-impedance instrument jack inputs on the front panel, as well as balanced jack inputs and outputs around the back with switchable +4dBu and 10dBV levels. Two further jacks provide additional unbalanced outputs, and in computer-based systems these additional outputs may be used to set up a latency-free monitoring path for recording. Some users may be miffed that there are no 'professional' XLR line ins and outs, but I for one much prefer balanced jacks on any piece of equipment that's likely to be used in a project studio, especially via a patchbay, as jacks are more compact, cheaper, and more commonly used in small studio and instrument-based setups. An expansion slot is available to fit the optional DO2 converter board, enabling signals to be sent to an audio interface, soundcard, or recorder in the digital domain. The DO2 offers 24-bit resolution, has a word-clock input BNC connector, and can be set to run at 44.1kHz or 48kHz sample rates. Another key ingredient of the 5060 is the option to use tailor-made presets for the compression settings, something that was first developed by TL Audio for their popular Fat Man processors a few years back. Now, I'm not a huge fan of presets as a rule, but I have to concede that a lot of users do struggle with compression and the settings here have been well chosen to cover the majority of requirements. Also, and very importantly, you can use the compressor manually, though the attack and release time constants are limited to switchable slow and fast settings only — they're not fully variable. If you find a preset you like and have the urge to tweak it a bit, all the preset control settings are shown in the handbook so you can recreate manually.

Front-panel Controls The input section of the 5060 features a four-way input selector (mic with 48V phantom power; mic; line; or instrument), with a further knob to adjust the mic gain in 20dB steps from -20dB to +40dB. The larger red input knob controls both the mic and line/instrument input gains and adds another 20dB of gain to the mic input. A low-cut switch brings in a 90Hz high-pass filter, and there are two status LEDs, a yellow one to show how hard the tube is being driven and a red one to warn of imminent clipping. Occupying a section all on its own is the Program knob, which offers 16 options if you include the Manual setting. The manual describes each preset and its applications in reasonable detail, but you can discern a lot from the names: Whisper Vox, Pop Vox, Rock Vox, Scream Vox, Keys, Bass, Ac. Guitar, El. Guitar, Snare, Kick, Loop, Pop Mix, Rock Mix, Dance Mix, and Slam Mix. The more conventional compression controls for threshold, ratio, make-up gain, file:///H|/SOS%2004-09/TL%20Audio%20Ivory%202%205060.htm (2 of 4)9/26/2005 12:05:12 AM


and switchable hard/soft knee are found in the next section, along with fast/slow attack and release switches. Whenever a preset is being used, all the physical compressor controls with the exception of the make-up gain are disabled. Clearly, a compressor preset only makes sense if the input level is exactly right, but fortunately there's enough gain leeway on the Input Gain control to handle this, otherwise you would have still needed manual threshold adjustment. In the output section there's the usual moving-coil meter, which can show either gain reduction or output level, and there's also a -10dB switch that affects only the way the meter reads. This means that when you're driving into a piece of digital gear that requires a very hot analogue level the meter needle isn't forever pushed up against the right-hand end stop. This meter is backlit and looks neat, but I can't help wondering if a fast LED meter would show the gain reduction more accurately. Also tucked away in the output section is the Fat EQ button, which functions not unlike a loudness control by creating a 'smile' curve, where 2dB of gain is added at 50Hz, 1.8dB at 10kHz and the mid-range at 720Hz is scooped by 0.9dB. This can sound quite effective on some material, particularly mixes, but it is perhaps best left switched off until you need it, as it's too easy to get used to the sound! This EQ curve comes after the compressor, so it won't interfere with the compressor operation.

Session Impressions Even though the compressor has only switchable attack and release times (both in the manual and preset modes), the presets work extremely well when used as directed, but can also work impressively on sound sources for which they were never intended. Overall, the sound is still typically TL Audio, by which I mean clear, solid, musical, and not at all gimmicky. I found the presets worked best with the input gain level set so that the Drive LED just flickered, backing off the input gain if the amount of compression was too much. As always, balancing the bypass level with the compressed level using the Gain Make-up control is a good idea when making comparisons, otherwise you tend to favour whichever setting is the loudest. That Fat EQ button is interesting, because, even though it does thicken the low end slightly, what I found most noticeable was the impression of air that was added to the top end. Overall the effect is very sweet-sounding, but subtle enough to be useful, so although I first thought it might be a bit of a gimmick it actually worked very well, especially on voices and complete mixes. The mic preamp also performs to a high standard, so you can use the 5060 as a recording front end as well as an insert processor in full confidence that you file:///H|/SOS%2004-09/TL%20Audio%20Ivory%202%205060.htm (3 of 4)9/26/2005 12:05:12 AM


won't be compromising the signal quality. Indeed, I had to keep reminding myself that I was reviewing such an affordable processor. As a confirmed control tweaker, I would have appreciated variable attack and release knobs rather than switches, but being realistic it's not much of a limitation, and considering that the unit is aimed at those users who feel more comfortable with presets I think it might actually make the manual mode less daunting.

Dial-a-squeeze As well as being an ideal processor for compressor-phobic musicians, the TL Audio 5060 is actually a very nice-sounding piece of kit and has enough flexibility to be used as a serious manual compressor as well, despite the apparent limitation of switchable attack and release times. The inclusion of a mic preamp and instrument inputs means the 5060 also works as a mono recording channel and as a pretty fine mono or stereo DI box. And we mustn't forget the tube circuitry — you may only get half a triode in each signal path, but it really works to give you that 'larger than life' sound without having to worry about noise — all the sensitive preamp circuitry is solid state. On balance, the TLA 5060 is a great addition to an already well-respected range and it should do well, particularly amongst those users who want quality compression but are a little unsure about how to set up the finer details themselves. Published in SOS September 2004 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


file:///H|/SOS%2004-09/TL%20Audio%20Ivory%202%205060.htm (4 of 4)9/26/2005 12:05:12 AM


In this article:

A Right Charlie Sound Shaping & MIDI Control Niggles Conclusions

Ultimate Sound Bank Charlie Sampled Tonewheel Organ Plug-in (Mac/PC) Published in SOS September 2004 Print article : Close window

Reviews : Software

USB Charlie £159 pros Great collection of instantly useable, highly-evocative organ patches. MIDI controllable. Excellent overall sound quality.

Sampled tonewheel organ sounds have been around for many years, but more recently, we've seen the arrival of modelled virtual tonewheel organs. Can a sample-based Hammond emulation still cut it?

cons No MIDI patch-changing. Some of the multi-mapped sample sets show inconsistencies from note to note. No controllable key-click. Slow-fast rotary speaker crossfade sounds as unconvincing as ever. The sample-based nature of the instrument restricts the amount of editing that is possible.

summary As a collection of 'wash-andgo' organ sounds, this will do the job admirably, providing my cons list doesn't overly concern you. Whereas sampling has the edge in capturing the basic tone of an organ, I believe modelling has the advantage in editability and the overall consistency of the sounds it produces.

Paul Ward

Ultimate Sound Bank's Charlie is the product of a considerable amount of work by Sound Engineer Stephane Briand and Sound Designer Geoffroy Soulaine. Utilising a studio full of the most desirable equipment, they have produced over 12,500 organ samples, taking in a wide array of microphones, guitar amps, rotary speaker, preamps, compressors, DI boxes and A-D converters. All this to arrive at a sample set that aims to be as painstakingly pure, 'authentic', warm or coloured as might be desired. This is no mean feat, but do the ends justify the means? The downside to sampling is that the captured tone is, by definition, a snapshot. So Charlie's basic tones are fixed — there are no virtual drawbars here, such as you find on modelled organ simulations like Native Instruments' B4. USB make no bones about this in the manual, feeling that the decision to use samples is justified by the quality and authenticity of the results using their sample set.

information £159 including VAT. Time & Space Distribution +44 (0)1837 55200. +44 (0)1837 840080. Click here to email

The result is a library of around 3GB of sampled material with which to assault your hard drive. Yes, this is going to eat up a lot of your disk space. I cringed as I watched the installation process eat up whole swathes of my recording time! On a PC you will need a VST 2-compatible host, whilst Charlie will run under Mac OS 9 as a VST 2 plug-in and under Mac OS X (10.2.3 or higher) as a VST

file:///H|/SOS%2004-09/Ultimate%20Sound%20Bank%20Charlie.htm (1 of 5)9/26/2005 12:05:18 AM


www.timespace.com www.usbsounds.com

2, RTAS, AU or MAS instrument. The documentation goes out of its way to warn specifically that Charlie will not run as an RTAS plug-in within Digital Performer when running in DAE mode with Pro Tools hardware. For the purposes of this review, I installed the plug-in on my studio PC and ran it as a VST instrument under Cubase SX v2.

A Right Charlie Installation was simply a case of manually copying files from the supplied DVDROM into my chosen VST plug-in folder. An automated method would have been slicker, but I doubt anyone should have any real problems here. Although Charlie arrives as an instrument in its own right, its presets can apparently also be called up into MOTU's Mach Five universal sampler, which would provide a wide array of additional sound-processing capability. Unfortunately, I didn't have a copy of Mach Five to check out this aspect. Authorisation is by way of a challenge/response system, which is as good a protection method as any. USB's web site seemed to be down for a couple of days when I first attempted to register, but I was able to make use of the 15-day grace period to keep using it. Once I gained access to the web-registration page, it was all fairly painless, though it did take me a while to discover the difference between a serial number (which was alphabetical in my case) and challenge code (which was numerical)... On first glance, I quite liked Charlie's darkly retro interface, though it did make reading some of the text quite difficult, on my monitor at least. Presets can be chosen numerically, or from a drop-down list. Charlie does not have the ability to respond to external patch change requests, either over MIDI or from the host program. Although USB justify this by mentioning the amount of time it takes to load sample material into RAM, and I do have some sympathy with this, I'm not wholly convinced; I would have liked to be able to call up presets from the main Cubase SX arrange page without having to open up the plug-in each time. On my PC, at least, the loading time seemed fairly quick, never taking more than five or six seconds before I was able to play the selected patch. Charlie is not multitimbral, but it does have the ability to layer two organ Programs, each of which can then be edited, muted, tuned, mixed and panned to taste. Layers can be edited individually, or you can link them, whereupon all edits you perform are applied simultaneously to both layers. A Program provides the basic tone of the organ, after which each is passed into the sound-shaping tools. Samples may be loaded in 16 or 32-bit formats — there is a RAM/processor power trade-off here, but USB give you the option to optimise usage depending on your particular circumstances. Overall controls are included to determine velocity curve (switched off in the majority of presets, which is just as it should be in my book!), pitch-bend range, polyphony (a processor-saving feature), and octave shift. A toggle switch brings file:///H|/SOS%2004-09/Ultimate%20Sound%20Bank%20Charlie.htm (2 of 5)9/26/2005 12:05:18 AM


in the percussion feature. It would be nice to have had a level control for percussion — and percussion works polyphonically, which may have some Hammond purists hot under the collar.

Sound Shaping & MIDI Control Given USB's take on authenticity and purity, I was initially surprised to see the inclusion of such a powerful resonant multi-mode filter. There are three low-pass modes, with varying degrees of strength, and a single high-pass mode. Cutoff can be modulated by key-tracking, or by positive/negative amounts by a dedicated filter envelope of the ADSR variety. A second 'master filter' allows for simple low-pass or high-pass filtration, also with a resonance control. None of the filters here fill me with awe, sounding somewhat 'lumpy' as they cross resonant peaks, but they do the job adequately for the job in hand. Both the filter and amp envelopes can respond to MIDI velocity to increase amplitude or filter cutoff. The decay or attack can also be modified by velocity times — it is not possible to have decay and attack both set to respond to velocity. There's also a simple depth/time pitch envelope, which adds a quirky twist to proceedings — I liked it a lot! A modulation matrix maps up to four modulation sources with four destinations in continuously variable amounts. The modulation sources include MIDI key, velocity, aftertouch, pitch-bend, mod wheel, or any single MIDI controller number. Four LFOs are provided, two of which are independent for each layer and two that are common to both layers. The amount of LFO modulation can be set to respond to the MIDI mod wheel. Other modulation sources include pitch, filter and amplitude envelopes. Last in the chain is a simple reverberation processor with time, damping and mix controls. This sounds good — too good, actually, if you expect to hear an authentically 'boingy' Hammond spring reverb! Most of Charlie's controls are assigned a MIDI controller number and it would be a simple job to create a mapping on your favourite controller surface to accommodate it. It's a shame, however, that the controller numbers themselves cannot be set by the user. There is no provision for adding key-click, but there is a control to shift the sample start point. To be fair, USB don't attempt to equate the two, but after seeing no click control I was hoping this would act as a reasonable substitute. I didn't find it particularly helpful in this respect, simply hearing a nasty, hard click at various settings, rather than the pleasing 'chiff' of a real Hammond. Should computer power become a problem, Charlie's maximum polyphony can be limited and USB also give a whole host of details on saving more processor



power by making careful choices when you create your Presets.

Niggles There is nothing at all wrong with the overall sound that Charlie produces; it all seems gritty and life-like enough. But there are a few things lacking. I miss keyclick; for me that is half the joy of the Hammond sound, although others may not agree. I noticed that some of the samples did exhibit some click, but this was not altogether appealing, since their neighbours on the very next key did not! I was quite enjoying the 'Lord Full' patch until I hit a bottom 'C', followed by a bottom 'D'... from loud key-click to no key-click at all! From that point onwards, I couldn't help hearing it each time I ran down the keys. Where Charlie wins in overall tone, it unfortunately loses in some of these details. I also dislike the crossfade from slow to fast rotary speaker settings; I have never liked this aspect of sampled Hammond emulations, and Charlie is no exception — it feels clumsy and sounds totally unconvincing. I'm also unhappy about multisamples of distorted Hammond tones. The beauty of a distorted Hammond lies in the interaction between the notes being played as they are mangled together by the distortion — here each note distorts in glorious isolation of any others being played. You could always add a distortion effect in the signal chain, but this is something I would have liked to see USB tackle before the sounds leave their VST instrument.

Conclusions I'm something of a Hammond freak, so I always feel a Pavlovian response coming on whenever I get the chance to play one. Due to space constraints in my studio, my own Hammond T-series organ has to suffice for me, but I still hanker after the full-blooded power of a C3/B3 from time to time. Until now, Native Instruments have satisfied that urge with their B4 emulation, for which I have nothing but praise. Whether it stands up at the side of the real thing is a moot point, but it certainly does the job for me. Charlie, with its use of real samples and attention to detail should, therefore, hit a similar spot. But, for me, it doesn't quite cut it. With an instrument as revered as the tonewheel organ, any manufacturer daring to tackle an emulation is always going to have a tough time of it. Comparisons are immediately available and any 'difference' will inevitably be seen as a 'failing'. Charlie is a fine-sounding instrument and in a mix will doubtless produce results indistinguishable from the 'real thing', but I feel it is slightly let down in a couple of key areas, most notably key-click, rotary-speaker emulation and distortion. These are areas where modelling might well have been able to help. But credit to USB where it is due. The collection of sounds here is very good file:///H|/SOS%2004-09/Ultimate%20Sound%20Bank%20Charlie.htm (4 of 5)9/26/2005 12:05:18 AM


indeed, the presets sound great straight out of the box, and the editing facilities are excellent. The price also seems fair when you consider the work that has gone into this product and the trouble that it could save you. Anyone looking for a collection of instant Hammond sounds will find Charlie a useful addition to their armoury, but anyone looking for true playable authenticity may well be a little disappointed with some of the performance aspects. Published in SOS September 2004 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



URS N, A & S Series

In this article:

On The Outside Kicking In High Standards

URS A, N & S Series

URS N, A & S Series Classic Console EQ Plug-ins (Pro Tools) Published in SOS September 2004 Print article : Close window

Reviews : Software

pros Good sound, with none of the deficiencies that plague some digital EQs. Efficient use of DSP resources.

cons Expensive. Dual-concentric controls are fiddly.

summary

For all those Pro Tools users who pine for the sound of analogue EQ, URS have recreated the equalisers from three classic mixing desks. Sam Inglis

The difference between URS's 'classic console equalisers' and generic digital EQ is immediately obvious, though it doesn't come cheap!

Unique Recording Software's aim with their Classic Console Equalizer plug-ins is to bring the sound of some very wellknown mixer EQ designs to Pro Tools information users. Although URS are cagey about Classic Console EQ naming the sources for their A Series, Bundle (includes N Series N Series and S Series designs, you and A Series) £586.33 don't have to be a genius to work out (TDM) or £292.58 (RTAS); which classic consoles are being emulated here; if you can't tell by the way they S Series £445.33/£222.08; Everything Bundle (includes sound, you would only have to look at a photo. Both the visual and operational N, A and S Series plus A10 aspects of the originals are faithfully reproduced, which means that some of the and N12 graphic EQs) features conventionally found on parametric digital EQs are missing from the A £1521.63/£757.88. Prices and N Series designs. Specifically, the centre or corner frequencies for these include VAT. equalisers are stepped, rather than continuously variable, and the peaking EQ Sonic Distribution +44 (0) bands are not truly parametric: control over bandwidth (Q) is limited to a simple 1525 840400. wide/narrow option in the N Series and is entirely absent from the A Series. Gain +44 (0)1582 843901. settings were also stepped in the first versions of these plug-ins, but a recent Click here to email update to v1.2 introduced continuously variable gain. www.sonic-distribution. com www.ursplugins.com

Test Spec URS plug-ins version 1.2. Beige G3 Apple Mac 300MHz with 256MB RAM, running Mac OS 9.2.

URS's interface for the A and N Series plug-ins is faithful enough to employ a virtual version of the dual-concentric pot found in API and Neve designs, where the inner knob sets gain level and the outer knob sets frequency, or vice versa. It must be said that this is rather less intuitive in plug-in form than in the real world, and trying to set the knob positions by clicking and dragging can be quite frustrating. Fortunately, URS have recognised that this is the case, and have done their best to alleviate it by providing other ways of setting the controls — for instance, you can cycle through the stepped frequency settings by clicking on the

Digidesign Mix system file:///H|/SOS%2004-09/URS%20N,%20A%20&%20S%20Series.htm (1 of 4)9/26/2005 12:05:25 AM

URS N, A & S Series

running Pro Tools v5.1.3.

filter icon below the dial in the N Series EQ, whilst numeric entry boxes were added in the v1.2 update.

On The Outside The A Series is the simplest of these plug-in designs, offering four bands of EQ, each providing seven different centre or corner frequencies and up to 12dB boost and cut. The low and high bands can be switched between peak and shelving operation. The N Series EQ is slightly more complex: a total of five bands is available, two of which are high- and low-pass filters controlled by a single dualconcentric knob. Two more are high and low shelving equalisers, with a single mid-band peak EQ completing the quota. The peak EQ offers a choice of six centre frequencies, the high and low shelving filters three and four, and the highand low-pass filters four and five respectively. Up to 18dB of boost and cut is available on the peak and shelving bands. The S Series equaliser is more comprehensive, offering two fully parametric bands with continuously variable frequency and Q, two more fixed-Q bands that can be switched to shelving operation if desired, plus highand low-pass filters. Up to 15dB of boost or cut is available. The S Series plug-in can be bought separately, and there are several bundle options: the Classic Console Equalizer Bundle collects together the A and N Series plug-ins, whilst the Everything Bundle adds the S Series and two additional plug-ins comprising 10- and 12-band graphic EQs, again paying homage to classic API and Neve units and also available as a separate bundle. As is often the case, buying the TDM/HD plug-ins gives you the RTAS and Audiosuite versions as well, whilst Pro Tools LE users can opt to pay less and get only the native versions. Copy protection is by iLok, and the plug-ins will work under Mac OS 9, OS X and Windows XP. I had no problems installing the plugins, though for some reason, installing the v1.2 upgrade seemed to leave the N Series plug-in at version 1.0 in my system.

Kicking In One thing that's sometimes said about the EQ on classic high-end analogue consoles is that simply switching it into circuit can have a noticeable — and sometimes beneficial — effect, even with all the gain controls set to zero. It's easier to test these things objectively in a digital system such as Pro Tools, and doing so made it clear that with all the gain controls flat, none of these EQ plugins has any effect on the sound apart from applying dither noise, presumably indicating that they use double-precision processing before dithering the output file:///H|/SOS%2004-09/URS%20N,%20A%20&%20S%20Series.htm (2 of 4)9/26/2005 12:05:25 AM

URS N, A & S Series

back to 24-bit. So what happens when the gain controls aren't flat? Well, since Cambridgeshire is not exactly awash with vintage API, Neve and SSL desks, I can't tell you whether URS's plug-ins exactly replicate their behaviour. Because there's restricted control over bandwidth it's also not always possible to reproduce settings from the URS equalisers precisely in other plug-in designs, but you can certainly compare the way they sound: and the results are impressive. Both N Series and A Series are the kind of equaliser that engineers tend to describe as 'musical' or 'characterful'. A Series in particular is perhaps the most 'rock & roll' plug-in EQ I've ever heard, with a tough, aggressive sound that's perfect for rock drums, vocals and electric guitars. N Series, too, is capable of delivering plenty of welly on all these sources, but tends towards a warmer sound, and is equally useful in more delicate applications. I was particularly impressed with its ability to add high-frequency air to fingerstyle acoustic guitar. Unlike most digital equalisers, both of them sound good enough when boosting to make this a viable option. What's more, even a couple of dBs of cut or boost makes an obvious difference to the sound, and I found that the subjective effect of a 2dB boost in the A or N Series plug-ins could be equivalent to as much as 8 or 10 dB in, say, Waves' Renaissance EQ. The limited choice of centre frequencies takes a bit of getting used to, but they're all sensibly chosen and musically useful — which is why they were used on those classic consoles in the first place — and I never found myself wishing for a wider range. The S Series EQ is a somewhat different beast. Here, excessive boost often leads to very obvious distortion, which might have musical applications such as livening up a limp snare sound, but couldn't really be described as 'warm'. Smaller boosts can provide a smooth-sounding, even lift in your chosen frequency area, whilst the fully variable Q makes this useful for surgical cutting applications, which aren't really the forte of the other two designs. For instance, S Series did an excellent job of notching out a boxy 250Hz room mode in a male vocal, without making it sound tinny or gutless as some EQs can.

High Standards There's lots to like about all of these EQs from URS. In a world where so many digital equalisers sound cold and sterile, the warmth and colour of the A and N Series designs is a breath of fresh air, whilst S Series has all the smoothness and precision you could want. The DSP loads vary, but are low enough that you can afford to make these your default channel equalisers — you can load several URS EQ plug-ins even on a single Mix card DSP — yet their support for the Multishell architecture means you're not wasting a DSP chip by only using one file:///H|/SOS%2004-09/URS%20N,%20A%20&%20S%20Series.htm (3 of 4)9/26/2005 12:05:25 AM

URS N, A & S Series

instance. The only serious bone of contention is likely to be the cost, which is certainly up there with other high-end equalisers such as Sony's Oxford EQ and Waves' products. If money is no object, URS's plug-ins would complement one or both of these competitors very nicely! Published in SOS September 2004 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


file:///H|/SOS%2004-09/URS%20N,%20A%20&%20S%20Series.htm (4 of 4)9/26/2005 12:05:25 AM

Voxengo Pristine Space

In this article:

Space Control Spaced Out Ready-made Rooms In Use

Voxengo Pristine Space £100

Voxengo Pristine Space Multi-channel Convolution VST Reverb Plug-in (PC) Published in SOS September 2004 Print article : Close window

Reviews : Software

pros Excellent sound quality. Great to have surround support, especially at this price. A large number of very usable impulse files available for download.

Surround-capable reverbs don't usually come with a price tag of £100. However, nobody seems to have told Voxengo. So just how good is their Pristine Space PC VST plug-in?

cons Not unexpectedly, quite CPU-intensive. Only available in PC VST format.

John Walden

Continuing increases in host-based processing power have now made summary convolution reverb a viable option for Pristine Space provides an anyone with a modern Mac or PC, but excellent introduction to many of the options available are pricy. convolution reverb for those on a budget and is capable of In the May 2004 issue, for instance, Paul White reviewed the Waves IR1 some excellent results — the fact that it can also be used in convolution reverb plug-in; this is clearly a surround-sound context an impressive product, but many people makes it a real bargain. will baulk at its £699 price tag. PC users information with a VST-friendly sequencer who would like to hear what convolution can $139 plus VAT (approximately £100). offer might, therefore, be interested to try www.voxengo.com Voxengo's Pristine Space, which has a price closer to £100 (depending upon Test Spec the current US-UK exchange rate). Keen readers of Mark Wherry's Cubase Notes column from May will also have noted that, remarkably, Pristine Space is Pristine Space v1.1 not just a stereo reverb — with support for up to eight channels, it is possible to PC with 2.4GHz Pentium 4 use it in surround-sound configurations up to 7.1. CPU with 1GB RAM running Windows XP Pro, with Echo Mia 24, Egosys Wami Rack 24 and Yamaha SW1000XG soundcards. Tested with Steinberg Cubase SX 2.0.1.

Paul's recent IR1 review provides a useful introduction to the convolution process and the idea of 'impulse responses' that capture the properties of a real reverberant space, so interested readers might care to take a look at that (subscribers can read it on-line at www.soundonsound.com/sos/may04/articles/ wavesir1.htm) if they need some background information. In addition to convolution and support for eight channels, the other key features of Pristine

file:///H|/SOS%2004-09/Voxengo%20Pristine%20Space.htm (1 of 4)9/26/2005 12:05:33 AM


Space include comprehensive routing options, 'low quality' and 'zero latency' modes that use less CPU resources, a large number of impulse files that are available as free downloads, some non-destructive editing of impulse files and a browser-based help file. Usefully, a fully working demo can be obtained as a 1MB download from the Voxengo web site and tested on your own system, the only restriction being that it cuts out briefly every 25 seconds. If you decide to purchase, a registration number is supplied to unlock the full version.

Space Control The screen shot shows the user interface for Pristine Space and, aside from some global settings (for example, for setting latency), everything is confined to this single screen. The Preset button provides a selection of default channel configuration options, covering common situations that Pristine Space might be used in. These include as a two-channel insert effect, two-channel send effect and a 'true' stereo reverb send (using four channels). These presets enable the appropriate number of input/output channels (situated in the bottom half of the display) but the user still has to load the appropriate mono or stereo WAV impulse file into each channel's Slot. Of course, the channels can also be configured manually, as for the configuration shown in the screen shot and used for 5.1 surround project within Cubase SX — the button immediately above the word Quality at the bottom left of the screen specifies the number of channels to be used. The Quality button for each slot toggles between maximum and low quality modes; as convolution reverbs are processor-hungry, I found I used lowquality mode while tracking in SX before switching to maximum quality when performing an audio mixdown. The Slot buttons along the centre specify which of the eight convolution channels is currently selected, and the envelope display and File dialogue change to reflect the currently selected slot. Once an impulse file is loaded into a particular slot, some non-destructive editing options are available. The Length knob allows the length of the reverb tail to be cut, while the Env Enable buttons can be used to turn on/off individual envelopes for volume, stereo width, stereo pan, low- and high-pass filters and EQ which are applied to the impulse. If several impulse slots are being used, the envelopes for each slot are independent of each other. To edit the envelopes themselves, the buttons above the impulse display are used to select the required function ('V' is volume, 'S' is stereo width, and so on). Double-clicking on an envelope adds a further edit point. Convolution

Spaced Out Some 180 impulse files are available from Voxengo's web site. The WAV file names give a clear idea of their intention and they range from small and intimate spaces through to vast halls or cathedrals. Some examples include: Acoustic Space. Acc Guitar Space. Ambient Hall. Ballad Vocal Hall. Big Empty Club. Bright Theatre. Close Small Room.



channels can also be linked via the Link To button. Overall, editing is straightforward and, while there is not the same degree of control that is available in a product like Waves IR1, the ability to edit reverb length and tonal characteristics will probably be sufficient to meet the key requirements for most potential users.

Coffee House. Drum Booth. Echo Hall. In The Silo. Live Wooden Room. Medium Warm Hall. Nice Drum Room. Parking Garage.

Ready-made Rooms

Scala Milan Opera Hall. Small Clear Room.

Of course, in order to use Pristine Space, some Studio 10 x 10 feet. suitable impulse responses are needed from Venue Warm 1. real rooms, hardware reverbs or modelling Vintage Snare Room. software. Voxengo have a large number of suitable impulse responses available for Wide Space. download via their web site. These are bundled into a number of collections, each stored in a RAR archive (but not a ZIP, so you will need something like WinRAR from www. rarlab.com to unpack them). The Spaced Out box lists a few of these to give a taste of what is available. Voxengo also provide links to other web sites with suitable impulse files. These include the Noise Vault (www.noisevault.com) where I was able to download an excellent small collection of impulses derived from a Lexicon Ambience algorithm — unfortunately for me, I don't own a high-end Lexicon reverb unit so I couldn't do a direct comparison but, subjectively, the Pristine Space version did sound pretty good.

In Use Throughout testing, Pristine Space seemed very stable, and setting up within SX proved simple enough. Used as a stereo send/return effect, it's no different from any other VST plug-in, although there are a number of ways to configure Pristine Space itself when operating in this way. Things are a little more complex if you want to use the plug-in in a surround sound project, but Mark Wherry's May 2004 Cubase Notes column (www.soundonsound.com/sos/may04/articles/ cubasenotes.htm) details how to configure a multi-channel send with a surround panner to place the resulting reverb within the surround space. In terms of CPU loading, like other convolution-based reverbs, Pristine Space is pretty demanding. On my test system, a six-channel configuration at maximum quality, on average, left the SX CPU meter at the 40 to 50 percent mark — the exact figure depends on buffer sizes and what else is being demanded of the



system at a given time. The low-quality setting reduced this down to about 30 percent and, while tracking, the quality of this was more than adequate. So how does Pristine Space sound? In Paul's Waves IR1 review, he commented that most convolution reverbs sound similar when fed with the same impulse files. In this sense, obtaining some good impulse files is key to getting the best out of Pristine Space. However, to my ears at least, Voxengo have done a very good job on this front and this plug-in can produce some seriously good results. With solo instruments or vocals in isolation, the sense of being in a 'real' space is particularly noticeable. Many might consider Pristine Space to be good value for money even if it was just capable of stereo use, but the multi-channel support turns it into a real bargain. If you have not, as yet, experienced what convolution reverb can produce, then the demo version is most certainly worth testing. At this price point, Pristine Space comes highly recommended. Published in SOS September 2004 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



Q. In what order should I compress and EQ?

Q. In what order should I compress and EQ? Published in SOS September 2004 Print article : Close window

Sound Advice

When using a compressor and an EQ as insert effects on something like a snare drum, is it better to use compression followed by EQ or EQ first then compression? SOS Forum Post Technical Editor Hugh Robjohns replies: Compressors usually react to the loudest part of the signal regardless of frequency (although many are designed to ignore the frequency extremes). So, if you use an equaliser first, then by boosting or cutting different parts of the spectrum, you will effectively alter the compressor's threshold at different frequencies, making it more or less responsive to those frequency regions. EQ'ing the compressor side-chain is the best

In other words, the equaliser changes the way the compressor works way to set up 'frequency-conscious' on the sound, compressing signals in the boosted regions more than compression, though an EQ can also be placed before the compressor in the signal the rest, and vice versa. It's usually a fairly subtle effect, but it's chain. worth bearing in mind. In general, though, equalising the compressor's side-chain is a much better way to set up 'frequencyconscious' compression, but then not all compressors have a side-chain input or a built-in side-chain EQ. The more usual scenario is to equalise a signal after it has been compressed, in which case you're simply altering the tonality of the compressed signal. Having said all that, when applying dynamics to drums most of the 'rules' can be thrown out of the window because often you are trying to create an effect, rather than simply modify the signal's dynamics in the traditional way. By boosting the high end of a snare drum prior to compression, for example, you can often control the amount of transient 'slap' that slips through, which is very helpful if the compressor doesn't have an adjustable attack time! Published in SOS September 2004

file:///H|/SOS%2004-09/Q.%20In%20what%20order%20should%20I%20compress%20and%20EQ.htm (1 of 2)9/26/2005 12:05:47 AM

Q. In what order should I compress and EQ?



file:///H|/SOS%2004-09/Q.%20In%20what%20order%20should%20I%20compress%20and%20EQ.htm (2 of 2)9/26/2005 12:05:47 AM

Q. Can I remove the vocals from a track using phase?

Q. Can I remove the vocals from a track using phase? Published in SOS September 2004 Print article : Close window

Sound Advice

I understand that it's possible to remove the vocals from a track by reversing the phase of one of the stereo channels. Is this correct and, if so, how does it work? Also, is the reverse possible — can you remove the rest of the track to just leave the vocals? SOS Forum Post Technical Editor Hugh Robjohns replies: Removing the vocals using the phase reverse trick is very easy. All you have to do is invert the polarity of one side of the stereo signal — usually the right — and then mix the left and right channels together in mono. This is pretty straightforward to do either with an analogue or digital mixer, or in a DAW, and is often referred to as the 'karaoke effect'. This trick works because vocals are usually recorded in the centre of the stereo image — in other words, the same signal is present at the same level in both channels — whereas the rest of the band is spread out across the entire image (with each element at different levels in the two channels). By inverting the polarity of one channel and adding the left and (inverted) right signals together, anything which is identical in both channels will cancel out, and hence the vocals disappear. The bass and kick drum are usually panned centrally too, and so these will disappear as well, which isn't very helpful! However, if you apply a high-pass filter to one channel (in other words, roll the bass off below about 120Hz) before you do the mono summing, you can retain most of the bass parts while still removing the vocals very effectively. This works because if the bass is removed from one channel, there is nothing to cancel out the bass remaining in the other channel. One common problem with this simple karaoke trick is that it can't remove the (usually stereo) vocal reverb, or any panned vocals (such as a chorus), so in most cases you'll only be removing most of the vocal, rather than all of it. Unfortunately, removing everything else to just leave the solo vocal cannot be achieved with any kind of variation of this technique, simply because of the variation in the stereo panning of every other source. In order for all of the (unwanted) backing to cancel out when the polarity of one channel is reversed, it would need to be identical in both channels, with the vocal panned to one side. Published in SOS September 2004

file:///H|/SOS%2004-09/Q.%20Can%20I%20remove%20the%20vocals%20from%20a%20track%20using%20phase.htm (1 of 2)9/26/2005 12:05:52 AM

Q. Can I remove the vocals from a track using phase?



file:///H|/SOS%2004-09/Q.%20Can%20I%20remove%20the%20vocals%20from%20a%20track%20using%20phase.htm (2 of 2)9/26/2005 12:05:52 AM

Q. Do I really need to buy a channel strip?

Q. Do I really need to buy a channel strip? Published in SOS September 2004 Print article : Close window

Sound Advice

I often see reviews for mic preamps and channel strips in SOS and I'm wondering if I need one. I currently plug a condenser mic into a small mixer, which provides phantom power, which then goes into my M Audio Delta soundcard. It seems to work reasonably well, though I normally have to crank the level up a bit to get a decent input. But am I missing something? Is there a good reason to pay money for a box that seems to do something I'm already able to do? SOS Forum Post Technical Editor Hugh Robjohns replies: From your description, it sounds like you are currently using a small mixing desk to boost and condition the mic signal before feeding it to the Delta's A-D converter. A mic preamp is simply the front end of a single mixer channel (like the one you're already using) in a convenient package. If you don't require the additional mixing, EQ and monitoring facilities usually provided in a small mixer, then a stand-alone mic preamp is often a good choice. Additionally, if you only require, for example, two highquality mic preamps to accompany your existing small-format mixer, an external two-channel unit is a sensible and cost-effective option. A channel strip, on the other hand, is essentially a mic preamp, an EQ and (usually) a compressor, all in one box, offering facilities similar to a single channel of a large-format analogue console. Again, a channel strip is often a good choice if you need only one or two channels with those kinds of facilities, at a reasonable price.

When converting an analogue signal to digital, a channel strip's own digital output board, such as the optional ADC card for SSL's XLogic channel, will often give better results than a computer soundcard.

Neither would offer fundamentally more facilities than you currently have — assuming you are happy to apply EQ and compression to the signal once it's in your computer — although suitable external units could offer better quality and/or greater functionality. One other aspect that might be worth considering is that some stand-alone mic preamps and many channel strips can be fitted with optional A-D converters. This allows the delicate task of A-D conversion to be carried out very close to the mic preamp, rather than in the electrically hostile environment of a computer. Often the file:///H|/SOS%2004-09/Q.%20Do%20I%20really%20need%20to%20buy%20a%20channel%20strip.htm (1 of 2)9/26/2005 12:05:57 AM

Q. Do I really need to buy a channel strip?

mic preamp or channel strip designer is able to use better components and extract better performance than is possible in a computer soundcard. The signal can then be passed to the computer as an AES or S/PDIF signal, in which form it is far more robust and virtually immune to the kinds of sonic degradations that are inherent in analogue-input sound cards. Published in SOS September 2004 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


file:///H|/SOS%2004-09/Q.%20Do%20I%20really%20need%20to%20buy%20a%20channel%20strip.htm (2 of 2)9/26/2005 12:05:57 AM

Q. How can I stop my guitar buzzing?

Q. How can I stop my guitar buzzing? Published in SOS September 2004 Print article : Close window

Sound Advice

I'm having serious earthing problems which are making it difficult for me to record a clean guitar sound with my Boss GT6 effects and Roland GR20 guitar synth. There's a bad earth buzz that only stops when I'm touching the guitar with both hands. As you can imagine, this is frustrating because every time I change a chord or lift my finger for a split second there's a 'click'. All my leads are good, and I dont have this problem with any of my other equipment — just the guitar stuff. The Line 6 Pod Pro has the same problem, which increases and decreases when I move around the room. Even if I do find a spot where the buzzing stops for a bit, the slightest thing will set it off again. The problem is there regardless of whether the units are connected to the mixer or not, and the power supplies are completely separated from any of the other equipment — they have their own dedicated, regulated plugboard going into its own dedicated mains outlet. Could it be a problem with my domestic mains supply? Any advice would be greatly appreciated, especially if there would be a cheap way of fixing it. Mike Keane Editor In Chief Paul White replies: You don't say what kind of computer monitor you are using, but it's a well-known fact that CRT (as opposed to flat-screen) computer displays interfere badly with guitar pickups, especially single-coil pickups, though humbuckers are not completely immune. If the hum changes in character and intensity as you move around the room or even when you face in different directions, this suggests that you are picking up radiated interference rather than something coming in via the mains — mains hum wouldn't be affected by your position in the room. Other sources of this kind of interference besides CRT displays are any pieces of kit with transformers inside, which includes powered monitors and power adaptors and especially room lighting dimmer switches. Incidentally, I've also come across some power adaptors that do put interference into the mains circuit when they're plugged in. So it could also be worth

Kinman's AVn pickups use a stacked-coil design to combat hum.

file:///H|/SOS%2004-09/Q.%20How%20can%20I%20stop%20my%20guitar%20buzzing.htm (1 of 2)9/26/2005 12:06:06 AM

Q. How can I stop my guitar buzzing?

unplugging as much gear as you can, testing the guitar signal path, then, if it is quieter, plugging the other gear back in a piece at a time and see when the interference comes back. You may well be able to identify a faulty or sub-standard adaptor this way. There's not much you can do to resolve the problem without incurring some expense! Obviously, the best outright solution would be to replace your CRT display with a TFT flat-screen model. Another, and probably cheaper option is to modify your guitar, installing humbucking pickups and lining the cavity which houses the pickups and electronics with insulating copper foil. If you don't want to give up the sound of your single-coils, we did review two sets of stacked-coil Strat pickups — Dimarzio's Virtual Vintage pickups and Kinman's AVn pickups — back in November 1998 (www.soundonsound.com/sos/nov98/articles/pickup.htm), designed to get rid of hum without compromising on the classic Strat sound, or looks, for that matter. The other factor that may be aggravating the situation is the quality of the mains earth connection itself. It is worth using an illuminated socket tester to check that you have ground in the first place, though that in itself doesn't guarantee good results. As always, if you feel unsure about anything electrical, get hold of a qualified electrician. In some circumstances, you may get better results by burying a separate earth rod in your garden or basement and using that as your studio ground. As a rule, the damper the ground, the better the earth. Published in SOS September 2004 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


file:///H|/SOS%2004-09/Q.%20How%20can%20I%20stop%20my%20guitar%20buzzing.htm (2 of 2)9/26/2005 12:06:06 AM

Q. How do I transfer SysEx files to my Korg Wavestation synth?

Q. How do I transfer SysEx files to my Korg Wavestation synth? Published in SOS September 2004 Print article : Close window

Sound Advice

I bought some CDs of new sound banks for my Korg Wavestation off Ebay recently, only to find that most of them are in SysEx format rather than MIDI files. I've loaded sounds into my synths from MIDI files before using Cubase, but I don't know how to do it with a SysEx file. Can you help? SOS Forum Post PC music specialist Martin Walker replies: All you need is a small utility to download these SysEx files into your Wavestation. I use MIDI-OX (www.midiox.com) on my PC, which is a multi-purpose MIDI utility and SysEx librarian. Similar shareware or freeware utilities are available for the Mac, including Snoize's SysEx Librarian for Mac OS X (www.snoize.com/sysexlibrarian) and SysEx for OS 9 (http://members.cox.net/sgrace9/sysex/index.html). A quick Google search will turn up several alternatives for either platform.

Korg's landmark Wavestation synth, as seen in the original review from SOS August 1990.

I'd also suggest that you read my Korg Wavestation Tips article from SOS June 2002 (www.soundonsound.com/sos/jun02/articles/wavestationtips.asp) for more details, because the Wavestation can be a little tricky to download SysEx to, as the files are so large by comparison with most other synths of the period. By the way, you can download a huge number of SysEx files for the Wavestation free of charge from SOS Forum contributor Ben Hall's web site: www.seriously-productions.freeserve.co.uk/wavestation/index.html. Published in SOS September 2004

file:///H|/SOS%2004-09/Q.%20How%20do%20I%20transf...0files%20to%20my%20Korg%20Wavestation%20synth.htm (1 of 2)9/26/2005 12:06:12 AM

Q. How do I transfer SysEx files to my Korg Wavestation synth?



file:///H|/SOS%2004-09/Q.%20How%20do%20I%20transf...0files%20to%20my%20Korg%20Wavestation%20synth.htm (2 of 2)9/26/2005 12:06:12 AM

Q. What can I use to trigger drums?

Q. What can I use to trigger drums? Published in SOS September 2004 Print article : Close window

Sound Advice

I love banging away on my thighs and knees, and I feel that I make some of my most interesting drum patterns in this way. However, I have never felt I could get the same level from using MIDI drum pads, or bashing away at my MIDI keyboard. How could I build a controller with a pair of panels that are velocity-sensitive to sit atop my thighs? Preferably these would be as thin as possible to maintain the feeling of my playing style. Any ideas? Ascher Nathan News Editor David Greeves replies: First of all, you're not alone! Lots of people find bashing out a rhythm on the edge of the desk or on their knees much more intuitive than using a MIDI keyboard or the tiny finger pads on many drum machines. There are plenty of alternatives out there, though which is right one for you will depend on your personal thigh-slapping style. Some favour dedicated drum pad controllers which offer larger pads than most all-in-one drum machines. Akai's MPD16 provides 16 MPC-sized pads, while Roland's SPD6 offers six large pads. Both Roland's HPD1 'Handsonic' is a great handswill sit happily on your lap. Roland's HPD15 Handsonic percussion on percussion controller. controller, which we reviewed in SOS October 2000 (www. soundonsound.com/sos/oct00/articles/rolandhpd15.htm) could be an even better option. It has a very large circular pad, divided into various different zones. If you just want something to bash away at, you could ignore these (along with the HPD15's other bells and whistles, like its ribbon controllers and D Beam), hit the pad wherever you like and then change the MIDI notes in your sequencer once they have been recorded. Their timing and velocity will be preserved, and that's the important thing. If you can't find a commercially manufactured device that meets your needs, you may have to take matters into your own hands. You could have a go at building your own controller using piezo transducers. Paul White's excellent how-to article on the subject is available on our web site — surf to www.soundonsound.com/ sos/1995_articles/aug95/diydrumpads.html — and is recommended reading. The suggested method is to attach a piezo pickup to the underside of an old coaster or table mat, the top of which is then covered with

file:///H|/SOS%2004-09/Q.%20What%20can%20I%20use%20to%20trigger%20drums.htm (1 of 2)9/26/2005 12:06:18 AM

Q. What can I use to trigger drums?

rubber to provide a comfortable playing surface. A second coaster or mat is attached to the underside, seperated by spacers and the piezo pickup connected to a standard two-wire jack socket. However, you'll need a MIDI drum machine which accepts a drum trigger pad input in order to turn the signal from the piezo pads into MIDI. Not many machines have trigger pad inputs these days — the Alesis DM5 and DM Pro modules are a notable exception, and are listed in their current catalogue. You could also hunt down a second-hand Alesis D4 or Akai ME35T. You will need to consider the issue of velocity sensitivity too. The pads which Paul White explains how to make are really intended to be played with sticks. You'll need to experiment with different materials to produce a pad sensitive enough to be played with your hands, whilst isolating the piezo tranducer sufficiently so it isn't triggered accidentally. In any case, home-made pads will be less velocity-sensitive than manufactured units, and even they can struggle to translate the subtleties of acoustic percussion (which includes knees!) into MIDI data. This is something you'll have to live with. One final suggestion is that you get hold of a USB-capable dance mat (of the sort which accompanies arcadestyle dancing games) and assign its sensors to trigger MIDI notes. This is easily done in Ableton Live and you could happily drape the matt across your knees. The sensors on this kind of device aren't velocity sensitive but timing could at least be recorded. You could even go one better, cut up the mat and incorporate the sensors in the world's first pair of MIDI lederhosen! Where you put the USB port is up to you... Published in SOS September 2004 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


file:///H|/SOS%2004-09/Q.%20What%20can%20I%20use%20to%20trigger%20drums.htm (2 of 2)9/26/2005 12:06:18 AM

Q. What is the whining noise coming from my soundcard?

Q. What is the whining noise coming from my soundcard? Published in SOS September 2004 Print article : Close window

Sound Advice

I noticed a quiet high-pitched whine coming from my Tannoy Reveal Active monitors, which are plugged into my Spirit Folio F14 mixer. Eventually I found that the whine stops when I press the button on my Edirol DA2496 soundcard to select an external word clock. I get the whine even if the Edirol card is not plugged into the mixer, but not if the monitors are not plugged into the mixer. I currently have unbalanced cables from the mixer to my monitors. I have to use the internal word clock. When any music is playing, it easily drowns out the whine, but I'd rather get to the source of the problem and sort it out. SOS Forum Post PC music specialist Martin Walker replies: If you're getting interference even when your soundcard isn't plugged into the mixer, but it stops when you disconnect the mixer from your speakers, it doesn't sound like a problem with the soundcard itself, although you may well be hearing sounds that originally come from the soundcard, probably from its word clock. As you can hear the interference even when the connection between the soundcard and mixer is broken, I suspect that you're suffering from a ground loop problem which is causing background digital noises to be heard — contrary to popular opinion, ground loops don't only cause background hum. Once there are multiple earth paths, noises from mouse movements, hard drive accesses, and graphic redraws can become audible in the background. Breaking the loop normally results in these all disappearing. If you have the option of fitting balanced cables anywhere in the chain (mixer to monitors, and soundcard to mixer) this should cure the problem, so try this first. It could also help if you plug everything into one mains socket via a distribution board, since this will generally make any existing ground loop smaller. It's also worth reseating your soundcard in its socket and tightening down its backplate screw. This ensures a good earth connection from soundcard to the PC chassis. Active monitors can be a source of problems because they are earthed via their mains lead and via the audio input lead. This is why using balanced connections helps.

file:///H|/SOS%2004-09/Q.%20What%20is%20the%20whining%20noise%20coming%20from%20my%20soundcard.htm (1 of 2)9/26/2005 12:06:23 AM

Q. What is the whining noise coming from my soundcard?

However, the only sure way to completely cure ground loop problems is to temporarily unplug all your audio cables and start from scratch. First listen to your powered monitors to make sure they don't exhibit any background noises with nothing connected to their audio inputs (sometimes US models with transformers designed for 60Hz will buzz when running on 50Hz). Then connect the stereo outputs from your mixer to the monitors and check again — if you have the option to use balanced I/O, do so. Finally, connect the inputs of your mixer one by one to your various synths and soundcard devices with them powered up, temporarily turning up the mixer output volume fairly high after each one is plugged in to see when any hums or buzzes appear. As soon as you hear any noises you've found the offending connection, and can either try making up special pseudo-balanced cables if your mixer has balanced inputs and the source is unbalanced, or trying a line-level DI (Direct Injection) box between the device and the mixer input. Published in SOS September 2004 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


file:///H|/SOS%2004-09/Q.%20What%20is%20the%20whining%20noise%20coming%20from%20my%20soundcard.htm (2 of 2)9/26/2005 12:06:23 AM

Q. Where can I get a hearing test?

Q. Where can I get a hearing test? Published in SOS September 2004 Print article : Close window

Sound Advice

I'm concerned about my hearing getting damaged — more at live gigs than in the studio — and I'm wondering how I'd go about getting my hearing checked. Also, is there anything I can do to reduce the risk of damaging my hearing if I'm attending or playing at loud gigs regularly? SOS Forum Post Technical Editor Hugh Robjohns replies: You can get your hearing checked either by asking your GP to organise a test with an NHS specialist for you, or at a hearing aid dealer on the local high street. However, the standard hearing test provided by these facilities (and the industrial audiometric tests used by some employers) are really geared towards determining speech intelligibility rather than music, and rarely go above 8kHz. If you don't want to go to the trouble of booking a professional hearing test at You can use DIY hearing test this stage, there is a useful DIY hearing test CD available, which I reviewed a CDs like this one to perform periodic checks on your hearing. few years ago (www.soundonsound.com/sos/Nov01/articles/ studioessentials1101.asp). I wouldn't rely on it for absolute accuracy, but it is certainly very usable and allows you to make regular comparative checks of your own hearing, and to compare it with your friends and colleagues in a reasonably consistent way. You may also be able to find some free hearing tests on-line (some are listed on www.digital-recordings.com, for example) but the variable behaviour of different computers and soundcards make them inherently less reliable than the DIY test CDs. Usually the clearest evidence of hearing damage for most of us is the so-called 'disco dip', which is a marked deterioration in hearing acuity around the 2-6kHz region that both deepens and broadens with age and, more importantly, with exposure to loud sounds over an extended period. While it is perfectly possible for a single extremely loud event to cause hearing damage, for most of us it is the regular exposure to loud sounds over extended periods that is the real culprit — six hours in a loud nightclub two or three nights a week, for example. Once your hearing has been damaged there is currently nothing that can be done to repair it. Hearing aids and cochlear implants have improved dramatically over the years, and will continue to do so, but not to the extent of restoring hearing to the resolution required for subtle music recording and mixing! So it really is essential

file:///H|/SOS%2004-09/Q.%20Where%20can%20I%20get%20a%20hearing%20test.htm (1 of 2)9/26/2005 12:06:31 AM

Q. Where can I get a hearing test?

that you look after your hearing as best as you can, and that means minimising your exposure to loud sounds. There are some excellent earplugs available these days, some marketed specifically as 'Musicians' Earplugs' with either flat (all frequencies reduced by a set amount) or a tailored response. These are ideal for use on a loud stage if you are a performing musician, or for use in a clubs or loud gigs. I usually wear earplugs if I am driving for more than a couple of hours, and when flying transatlantic too. Published in SOS September 2004 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


file:///H|/SOS%2004-09/Q.%20Where%20can%20I%20get%20a%20hearing%20test.htm (2 of 2)9/26/2005 12:06:31 AM

Q. Why don't my mixes sound big in mono?

Q. Why don't my mixes sound big in mono? Published in SOS September 2004 Print article : Close window

Sound Advice

I am part of dance music production team. We do get club and radio play, and, in the case of the latter, as far as I'm aware this will often mean mono playback. So I am always careful when mixing down a track that the track sounds similar in mono to the stereo mix. I have compared our mixes with others in our genre and, as you'd expect, our mixes sound better than some and not as good as others, but even in the ones which I regard as better than ours, there seems to be a something which separates the great mixes from the good ones: the really good ones sound big in stereo and in mono. I've been comparing two such records, and both sound massive in the studio compared with our mixes but when you press the mono button, on one it disappears, and when you press it on the other it doesn't change one bit! How do they do it? SOS Forum Post Technical Editor Hugh Robjohns replies: Mono mixes will always sound different to stereo ones, and there is little that you can do about that. On a technical level, the mono mix contains only the 'mid' information whereas the stereo mix has both 'mid' and 'side' information. The reason a stereo mix 'sounds massive' is because of the quantity and nature of the side signal. If there is a lot of out-of-phase information in the stereo mix it will tend to sound very big, but this information will largely be lost when listening to the mid signal only. The trick is to make the mix sound 'big' in mono, and not rely too heavily on the side signal. One element of this is the careful choice of reverbs and delays, making sure that they sound suitably big in mono before checking in stereo. It is very educational to listen to the return from a variety of reverb processors and compare the sound in both stereo and mono. Some all but disappear, basically because the left and right channels are extremely 'decorrelated' — the processor is relying on phase differences to create the effect — while others remain far more credible. I think this is where you will find the main difference. Published in SOS September 2004

file:///H|/SOS%2004-09/Q.%20Why%20don%27t%20my%20mixes%20sound%20big%20in%20mono.htm (1 of 2)9/26/2005 12:06:36 AM

Q. Why don't my mixes sound big in mono?



file:///H|/SOS%2004-09/Q.%20Why%20don%27t%20my%20mixes%20sound%20big%20in%20mono.htm (2 of 2)9/26/2005 12:06:36 AM

Business End

In this article:

Business End

Escobar Readers' Tracks evaluated Len Marsbleu Published in SOS September 2004 This Print article : Close window Month's People : Miscellaneous MPG Panel

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'.

Escobar Pete Gordeno (PG): "You could compare this to the Air album, which isn't that far away from this stylistically, but the difference between the two is that the Air album is chock-full of hooks and each track has a really strong identity. This sort of music is so easy to make these days; it has to be really, really good to stand out.

Track 1 3.2Mb

"It leaves me a bit cold, really. They sound like they're going for some sort of latin fusion thing in places, but the way it's put together — the programming and the dynamics — just feel a little bit too much like someone at a computer. I think I can hear the computer working a bit too much here. I spend a lot of time in front of computers, and I know it can be easy to get lost in what you're doing and lose sight of the original idea. This sort of music needs to have much more of a live feel to it, even if it is being made on a computer — that's what it's all about. "It does have its moments — the first track starts off really well, but then doesn't really develop. The second two tracks were more like songs than the first, but at the same time, they still didn't really reach out to me." Rennie Pilgrem (RP): "I think the trouble is that the first two tracks have quite a house feel to them, but there's no bottom end — which means you couldn't play them in a club, and they're probably not interesting enough to listen to at home. You wonder where it's supposed to fit in. The third track is alright; it's got slightly more interesting beats and you can hear a bit more emotion in there. "I agree with Pete about the computers; you do get a sense of 'look at the plug-ins we've got'. The third one has got a bit of soul to it, but with the other two, there's no point where you could say 'I like it when it did soand-so' because there's nothing there — there are no hooks. It's somewhere in between lots of different things, with no clear identity of its own. The third track is interesting, though — if they were all like that I could imagine putting it on and enjoying it.

file:///H|/SOS%2004-09/Business%20End.htm (1 of 5)9/26/2005 12:06:50 AM

Business End

"The other problem with this is that there are 12 tracks on the CD, which is just such a bad idea. A demo should be your two or possibly three best tunes, maximum." Nel Johnson (NJ): "This is background music really. The only place you can imagine this existing is in a bar somewhere. There's just not enough to hold your attention. "The one positive thing about these tracks is the percussion — some of it is really interesting. That's definitely something they should concentrate on and develop, that could really be their thing." Deirdre Melvin (DM): "I really don't understand what they're doing with the vocal sample at the beginning of the first track — it doesn't seem to connect in any way with the rest of the song. I think the whole point of having an intro like that is that it leads into something else later in the song, and that never comes. Having said that, I think the first song on this is better than the other two. "It's interesting what Nel said about it being bar music — I could really imagine this sort of thing playing in some slightly trendy bar for people in their late 20s and early 30s. I can imagine this going down really well in that kind of environment. "It's a shame there's no letter or information with this; it makes it difficult to know where they're coming from. I mean if this a first demo, then it's a really good effort, but if it's by some band that's been going for 20 years, then it's a bit poor."

Len Jamie Vaide (JV): "The sound and the production of this is very nice, but the songs just aren't doing it for me. His voice is really weak, and that's always going to be a problem for anyone doing this singer-songwriter type of music. It needs to be more distinctive.

Track 1 2.7Mb

"It's not bad, but it doesn't really have any hooks to it. You can see what he's trying to do, but he's just missing it really. The vocal doesn't move me, and the tune doesn't either. This is the sort of thing that we've all heard far too many times before. I think the first song's easily the best thing on this, but it does need that bit more to make it work, just some sort of hook." RP: "I don't usually get to hear this sort of thing, and I think it's pretty good actually. The production sounds a bit bright — it would be interesting if it was done in a slightly more edgy, darker way. This is almost too polished. It might be good if he was put with a producer who could make it a bit edgier. I wouldn't have thought this was a demo if I'd just overheard it somewhere, I would have thought this was the real thing. "I think it's good; I could imagine hearing it on a film or something. The voice isn't amazing or even very strong, but it works." NJ: "The singer reminds me of the guy from the Kings Of Convenience. I think the songs are quite accomplished, but I agree with Rennie that sometimes they do seem quite sugarysweet. I'm not a guitar player, but it seems quite predictable; the chord progressions seem uninspired." PG: "I do think this kind of singer-songwriter music works a lot better when it's got a bit more of an edge to it. file:///H|/SOS%2004-09/Business%20End.htm (2 of 5)9/26/2005 12:06:50 AM

Business End

The first track, which is a bit of a Coldplay/U2 soundalike, is maybe a bit transparent, and maybe a bit late! In terms of saleability, Coldplay did it, and Snow Patrol ripped it off — very well — but now it's got to move on." DM: "I think it sounds like this guy is hedging his bets. He's taken all of his influences, all of the bands everyone's mentioned and maybe thrown in a bit of country and folk as well, and hasn't committed to anything. It almost seems like he's thought, 'well I've got this demo, and I need to show that I can do a bit of everything'. I think the thing we're all picking up on is that there isn't enough of any one thing. He needs to concentrate on one style and perfect that." NJ: "The whole point of a singer-songwriter is that they've got this thing inside them, and they've just got to do it, and this doesn't feel like that." DM: "Initially, I found it quite annoying, but as it went on, I actually found it more endearing that this guy is trying to be everything to everyone. He does definitely have talent as a musician. "Some of the lyrics are a bit cringeworthy though — you really shouldn't be trying to rhyme 'crucible' with 'beautiful'."

Marsbleu JV: "This sounds like the only thing he's ever listened to is the first Prodigy album, but that said, I like it more than the latest Prodigy single! At least this isn't slavishly following fashion. You get the impression that this guy's doing this because it's what he wants to do. It's very honest, and that makes it stand out."

Track 1 1.7Mb

RP: "The first track starts really well and looks like it's going to be really good. It's very interesting, but I think he's trying to cram too many ideas into each song. Usually people have the opposite problem, where they're trying to stretch one idea out into a whole track. "The beats are very staccato and not doing a lot — well, they are doing a lot but they're all over the place. He needs to work more on that and try to make them more focused and direct." PG: "A lot of this sounds very amateurish, but I really feel that I like the guy from listening to the music. It's all over the place, the music over the top of the beats is completely stream-of-consciousness — there's no regard for what's right or what's wrong, or whether it's in the right key. However, there's something about it that's really appealing. "It sounds like it was done on pretty cheap gear, no pun intended, and I think it sounds like he hasn't quite got his head around the equipment yet. You can make pretty good-sounding tracks with cheap gear these days. He just needs to learn to take some of the ideas out of his tracks and concentrate on developing the good bits." NJ: "There are places in Hoxton that'd go absolutely bananas for this. It's got quite a kitsch feel to it; I hope that's intentional because it'd be a shame if that's not how he means it to be. "It does need some weight behind the beats. When you're using a lot of poor-quality samples you can get a sort of brittle sheen across everything, and I get the feeling a lot of his stuff is just coming straight off a CD file:///H|/SOS%2004-09/Business%20End.htm (3 of 5)9/26/2005 12:06:50 AM

Business End

and onto a mixing desk. It's a really raw sound he's got, and I like that. I think he just needs to beef up what he's doing. It sounds like he's mixed this on really quiet speakers and he hasn't got much idea what it sounds like loud. It sounds really natural, like he really means it, but I think it's going to take another person to see the beauty in it, to edit it down and develop it." DM: "I think this is painful. I can imagine this as the backing music to some really naff soap opera or Challenge Aneka-type TV program. I think it sounds like it was just knocked together in about five minutes. At least it has some personality, though." NJ: "I think there have been things like this all the way through dance music. Every so often someone comes up with a really nutty tune, and everyone adopts it and loves it, and has a good time when it comes on."

This Month's MPG Panel

Nel Johnson is a hybrid graphic designer, musician and DJ. As the latter, he has been involved in dance music since the mid-'80s, when he was an Electro DJ. Nel was a regular at the Hacienda, and later became involved in the Manchester music scene in various collaborations before starting a music-production company and recording studio with 808 State's Martin Price. Currently Nel is Creative Director at London's Metropolis Studios. Deirdre Melvin is Deputy Head of Music at the Student Broadcast Network, a service which provides student radio with specialist music programmes. Deirdre compiles the UK Student Radio Chart, which is keenly watched by the industry to see which acts are having an effect on the crucial youth market. Before joining SBN, Deirdre was a music promoter, and obtained a Masters in Radio Production from Bournemouth Media School. Rennie Pilgrem began his musical career by playing saxophone and keyboards in funk bands. Strongly influenced by Detroit house in the late '80s, Rennie went on to be a key part of the early-'90s hardcore scene as a member of Rhythm Section. In 1993, Rennie founded Thursday Club Recordings (TCR) which, more than 10 years later, is still going strong. His new album, Pilgremage, is out in September. Jamie Vaide has worked for the Atlantic Music Group for the last seven years. His role as Creative Production Manager has seen him working with acts like the Darkness, Jet, Ash and Muse. Before moving to Atlantic, Jamie worked as a press officer for Epitaph, Nuclear Assault and other independent labels. He has wanted to be on a review panel since he saw Sigue Sigue Sputnik doing one in the late Smash Hits magazine in the '80s. Shy and retiring keyboard player and vocalist Pete Gordeno preferred to be represented by one of his instruments of choice, but this is too modest. As a session musician, he's worked with Depeche Mode extensively (playing JP8000 on several of their world tours), as well as other household names such as George Michael and U2. He's now diversifying into production work, having recently produced a million-selling album for a French artist. Many thanks to Sam Stubbings and Metropolis Studios (www.metropolis-group.co.uk) for organising and hosting the session. Published in SOS September 2004


Business End




Crosstalk

In this article:

Permission To Rant The Need For Speed?

Crosstalk Readers Writes Published in SOS September 2004 Print article : Close window

People

Permission To Rant Since the August issue went on sale, we've been inundated with letters and emails responding to Paul White's Leader column, where he aired his frustrations with Mac OS X and its system of access privileges. Here's one such missive, which is broadly representative of the rest, and more polite than most! It sounds like Paul White has got his knickers in a bit of a twist over OS X! I had put off upgrading to OS X for as long as I could, having run Pro Tools on OS 7, then 8, and finally 9, and was reluctant to change — 9 always seemed fine to me. But now OS X has settled in, I love it. It looks like at least one of your problems — with Permissions and so on — can be easily solved. You should be running OS X as an 'Admin' user rather as a 'Standard' user, and as such, you then have permission to do what you want to all your files and folders, and full 'privileges' (within reason!). Once all permissions are restored, everything should be fine. I find my apps all open fast, and iTunes is great for playing CDs. OS X virtually never crashes and Pro Tools is now very solid too. Extension clashes, memory allocations, Appletalk problems, and the dreaded OMS are all things of the past. What more could you want? Roger Chatterton Editor In Chief Paul White replies: Well I've switched to OS X on all my Macs, and I like it very much too, but part of my duty in the Leader column is to rant about stuff that annoys me, and even though I am the one and only User and file:///H|/SOS%2004-09/Crosstalk.htm (1 of 4)9/26/2005 12:06:54 AM

Crosstalk

Administrator on my studio Mac, it still argues about permissions, especially if I've imported audio from an external source. For example, Logic may tell me that I don't have sufficient access privileges for it to create an overview waveform when I import audio from a CD-ROM or audio CD I've made. Rather than having to outwit the bugger, what I'd have preferred is a 'no paranoia please' mode where permissions became completely irrelevant, with maybe some polite warnings if you were going to do something that might trash the OS. After all, it's only a studio computer, not a strategic part of the national defence network. I know several other users who feel the same way. And yes, I know you can tell the system not to stick iPhoto or iTunes in your face at every opportunity, but I think it would be better if I should specifically have to ask for things to happen rather than the computer deciding what it thinks is good for me. I have the same issues with Setup Wizards that lull you into a false sense of security, then sod off to the pub when they come across something they can't deal with! So yes, there are things about OS X and virtually every other bit of technology on the planet that wind me up, but despite my whinges, I still think it is a great OS and wouldn't ever want to go back to OS 9.

The Need For Speed? Thanks for Dave Shapton's article on technical support and how to avoid it in the August issue of Sound On Sound. It provided some very useful information. However, I strongly disagree with his assertion that "the latest does not mean greatest" when it comes to buying your music computer. It is surely always far better to buy the fastest machine that you can afford. It will load faster, let you run more tracks and plug-ins, and you won't need to upgrade so soon. What's more, the most up-to-date machine will give you increased productivity over its lifetime and impress any clients who come to visit. One other very important point to mention on the subject of avoiding technical support is to ensure that your music computer remains a dedicated music computer — never connect to the Internet or email from it, and never install any non-music-related software. This minimises the likelihood of the incompatibility crashes spoken about in the article.

If you can wait, today's 'latest and greatest' in PC technology will cost you a good deal less in just a few months.

Finally, always virus-check anything that you plan to load on your music computer using an old obsolete or family PC. Also use that PC to connect to the file:///H|/SOS%2004-09/Crosstalk.htm (2 of 4)9/26/2005 12:06:54 AM

Crosstalk

Internet to get all your latest updates and samples. A fast PC will serve you well and give you the full value from all your software synths, samplers and plug-ins. A slower PC will just be a slow PC that you will want to upgrade sooner rather than later! Chris Furner SOS contributor Dave Shapton replies: Thanks for your comments, Chris, the majority of which make good sense. I don't even disagree with your thoughts about 'latest and greatest' computers, except to say that the system you choose to buy should take into account the circumstances in which you usually work. Like you, I'd normally want to buy the fastest, turbo-est, screaming-est computer on the planet. Just for the sake of it, I'd load up as many plug-ins and try to play as many tracks in real time as the beast could support. I'd actually take some satisfaction in the thought that I was doing with my computer what Formula One drivers do to their cars when they drive round bends on a racetrack. The trouble is that the computer equivalent of screeching tyres and barely controlled sideways movement is not necessarily what you want when you're approaching a deadline to deliver a project to a radio station or a record company. What you actually need in these circumstances is safety and stability. A case in point is the new Grantsdale chipset from Intel. These new chipsets are a pretty big jump forward in motherboard design. They've got the new Azalia audio sub-system, PCI Express and even onboard wireless networking. But would I use one on a serious project? Not yet. I haven't tested the boards with any of the esoteric hardware and software that I use. Most of it would probably work; but 'probably' isn't anywhere near good enough if you're trying to get a project safely delivered. So I'll stick with stuff I know works (or doesn't work but whose foibles are familiar to me) until this undoubtedly fine new Intel chipset is 'shaken out' by millions of users whose income doesn't depend on having a completely stable system. One other thing: I wasn't suggesting that you should necessarily buy computers using technology that's been available for a whole year. It's probably always going to be the case that if you wait just a few months, what was once cuttingedge in terms of both performance and high price is going to be a lot cheaper, and probably only very slightly slower, than the processor that came out yesterday. Published in SOS September 2004

file:///H|/SOS%2004-09/Crosstalk.htm (3 of 4)9/26/2005 12:06:54 AM

Crosstalk



file:///H|/SOS%2004-09/Crosstalk.htm (4 of 4)9/26/2005 12:06:54 AM

Dweezil Zappa: 64-bit Computing & The Frank Zappa Archive

In this article:

Rescue Missions Making 64-bit Compatible The 64-bit DAW Restoration Host Vs DSP Power A Question Of Preservation Sounding Right

Dweezil Zappa: 64-bit Computing & The Frank Zappa Archive DAWs Of Future Past Published in SOS September 2004 Print article : Close window

People : Artists/Engineers/Producers/Programmers

Frank Zappa produced a vast amount of recorded material during his career, often pushing both his musicians and the technology of the day to extremes. The task of editing, mixing and archiving these recordings is being undertaken by his son Dweezil, with the help of some cutting-edge computing hardware. Daniel James

As computers become ever more important in the music recording studio, manufacturers are at last taking this market seriously. AMD, for example, have a sponsorship programme which works with music industry figures including producer Phil Ramone and engineer Elliot Scheiner. This programme is now putting 64-bit dual-processor AMD Opteron Dweezil Zappa shows off the Verari dualmachines into studios to see how they Opteron PC in the UMRK machine room. perform in a real-world environment. And when Dweezil Zappa was looking for sponsorship to upgrade the family studio — known as the Utility Muffin Research Kitchen — his contact at AMD, Charlie Boswell, turned out to be a lifelong Frank Zappa fan, and was only too keen to work out a deal. Dweezil explains that the studio rebuild had to happen for a number of reasons, including work on the vast archive of his father's recordings. "There are tons of new projects that we're excited about getting involved with. Before, we weren't even able to listen to older recordings; the analogue machines were in disrepair,

file:///H|/SOS%2004-09/Dweezil%20Zappa%20%2064-b...omputing%20&%20The%20Frank%20Zappa%20Archive.htm (1 of 9)9/26/2005 12:06:57 AM


and we really didn't have the ability to work on any of my dad's music for almost a dozen years. So to put ourselves in a better position for the future, we had to make some serious choices about the technology that would help us do what was necessary. We were interested in doing projects in 5.1, doing restoration, and also having the ability to archive things better." The studio had accumulated not just an extensive vault of tapes, but a considerable collection of analogue hardware too. "Frank had several machines — I don't know if we have some of the earlier machines he recorded on. We have the Studer 24-track and we have three of the two-track machines. They all have different head stacks, and we even have a five-track head stack that was made by Paul Buff when Frank had a studio in Cucamonga — the Studio Z in the early '60s. We're curious to see if there's a way to get a machine to work with that head stack. It was before multitrack — Paul invented this thing and made it work, but I have never seen it in use. We found it in a box!" As interesting as the old machines are, Dweezil points out that they aren't always practical. "People say 'I like the sound', but a lot of these people have had no experience working with them. You may or may not know if it's working 100 percent unless you have an expert there. A lot of people these days are refurbishing the machines and putting different head stacks on them to specifically get one kind of sound, and people are going a little crazy. I got much more involved on the computer side of things, because I had to. I was tired of looking over someone's shoulder and saying 'No, no, no! I want the edit to go here,' and 'Do this with this.' That was a big learning curve for someone having zero experience with computers, learning to do everything from the ground up. "As I work, doing my own engineering and recording, and playing — I'm basically the only guy in the studio — I have to know how to work the stuff, otherwise someone's going to be sitting there babysitting, and being bored out of their mind. I didn't want to be stuck in the situation where if I have a deadline on something, and I need to work at two or three in the morning, I have to have someone in there just to do random things."

Rescue Missions Dweezil Zappa's own musical projects have run up against the limitations of technology in the past, and although the learning curve is steep, he's finding that his new DAW setup is enabling him to overcome problems that were insoluble with older equipment. "One project that I've been gearing up to finish is called 'What The Hell Was I Thinking?', a 75-minute continuous piece of music that is guitar, bass and drums. I describe it as an 'audio movie', because moment to moment the musical landscape changes, and it just morphs into different things. I started recording it on analogue tape almost 13 years ago. We were on the bigger reels of tape that could run longer per side, about 16 minutes. At the end, the machine was running a small amount faster; there was tape stretch, so when I tried to do some overdubs, nothing would stay in tune. You don't hear it as the file:///H|/SOS%2004-09/Dweezil%20Zappa%20%2064-b...omputing%20&%20The%20Frank%20Zappa%20Archive.htm (2 of 9)9/26/2005 12:06:57 AM


pitch drifting so much, except if you're trying to add something new to it — then it's uncomfortable. What I had to do was transfer everything to a digital format and replay some things. "The digital format we were using at the time was Frank's original Sony reelto-reel machines. By the time we were recording on them, they were at least 10 years old, if not older. We had done a passage where I did about 22 tracks of guitars, a ridiculous, overly patriotic section with 'Star-Spangled Banner', 'America The Beautiful' and 'Dixie' all at the same time, overlapping in this really uncomfortable way. It sounded The Utility Muffin Research Kitchen is based around a Euphonix System 5 console, and hilarious, but things just started now, an AMD Opteron-based 64-bit DAW dropping out, and we were hearing running Steinberg's Nuendo. some digital distortion. It just disintegrated, imploded — nothing went into record. Somehow, it decided it was never going to play back again! "Now, with my new setup I'm actually going to be able to compile the whole project. This requires a lot of editing, so now what I'm hoping to do is to put it together in sequence, as opposed to making stereo mixes and editing them together. I want to put the whole project together in multitrack form so it can be mixed in surround sound, and what's on tape will be all in real time. That's going to be the main challenge, and then I'll have a really good idea about what's really going to stay in the project. There are probably about 35 guest guitar players on it, everybody from Brian May to Edward van Halen, Eric Johnson, Angus and Malcolm Young — it's quite a crazy project. I'm still waiting and hoping to record Jeff Beck and Jimmy Page as some of my final guests on there." Running more tracks than most people would dare to, Dweezil could be accused of pushing the technology until it breaks, whether analogue or digital. Perhaps this attitude is what makes the studio a very useful test bed for AMD, together with the Utility Muffin Research Kitchen's reputation for technical advances. "I've had unfortunate experiences — you think 'this should be pretty simple' but then it doesn't end up working the way you planned. I've had that same experience with the computers as well. I don't feel as if I'm the kind of person who wants to use extreme amounts of processing to alter a sound — I pretty much try to get the sound the way I want it on tape to begin with, and I don't go crazy with processing."



Making 64-bit Compatible While there have been several 64-bit hardware platforms available for many years —such as the DEC Alpha and the Intel Itanium — the fact that you couldn't run a normal version of Windows on them made these systems largely irrelevant for the PC-based studio. The 'AMD64' architecture retains compatibility with 32-bit software by extending the established x86 instruction set, unlike the Alpha and Itanium processors which have their own, incompatible instructions. The AMD64 CPUs are branded as the Athlon 64 for the home user market, the 'prosumer' Athlon 64 FX and the 'enterprise' Opteron. There is also a Mobile Athlon 64 designed for laptops. The Opteron 200 series and above are the only chips intended for multiple-processor systems, with the 800 series (in theory) supporting up to eight CPUs in one machine. Apart from backwards compatibility with 32-bit software, the pricing of these chips makes them a realistic proposition for the studio market, with complete machines selling in the UK for under £1000. www.amd.com/amdopterondme

The 64-bit DAW "We met with Charlie Boswell, and AMD provided us with the opportunity to try some of their equipment. We have never had a specific alliance with any company, and Frank's music has always been misunderstood by so many people. Larger companies have never had any interest in helping in any way. So to have Charlie and AMD be interested in not only seeing what we could do with their technology, but continually supporting us, was a great advantage. The first workstation that I got was back in 2002, with 32-bit Athlon processors. I thought 'I'll never run out of processing power!', and then as I got better at what I was doing, things started to slow down, and I was having problems when I couldn't even get it to finish a certain mix because I had tried all of these processing plugins and virtual instruments. I didn't really know the inner workings of the computer and how bogged down it could get. Having not had the experience of many other users, I quickly found that if you have bunch of things all turned on at once, it won't work very well! "AMD said 'We have this other system that we'd like you to try,' and they delivered the 64-bit machine at the end of 2003. The problems that I was having with this one song in particular went away instantly — it was amazing to see the changes. It improved the sound of the song, which already (to me) sounded really good. There were effects that I had forgotten about because I wasn't hearing them as crisply as when I first put them on. They all came back, and you could hear the panning better because it wasn't being bogged down." Even if a 64-bit machine is running legacy 32-bit software, such as Dweezil's setup of Steinberg Nuendo on Windows XP, the newer architecture can still offer performance benefits. In particular, the Opteron has a DDR controller integrated with the CPU, which reduces the bottleneck to system RAM. Dweezil uses an RME audio card in his Verari dual-Opteron workstation, which is connected to a file:///H|/SOS%2004-09/Dweezil%20Zappa%20%2064-b...omputing%20&%20The%20Frank%20Zappa%20Archive.htm (4 of 9)9/26/2005 12:06:57 AM


Euphonix System 5 console. "I think the equipment that I have the ability to use now gives me so many options. I'm much more excited about being in the studio because I really only play guitar well, but with all these virtual instruments you can sit there and tweak, for hours, to do whatever you want. Not that I'm going to become some master programmer, but I can definitely start using other instruments that I can't physically play well. If I have the idea, I can incorporate it, which makes it much more interesting and challenging in the studio — but not challenging to the point where you can't do anything.

It's now possible to release many of Frank Zappa's previously unheard recordings.

"The real balance has to be struck between having the creative idea — knowing what you hear in your head — and finding the tools to make it sound like what you hear, or better. What's always amazing to me is that there's such a low noise floor that you can compress the hell out of something if you want, add all this gain and other effects, and you don't end up with hiss, or the maniacal, horrible sound that you would get if you were patching 10 devices together. Now, I will plug in 12 or 13 different things all in line just to see what it will do. I've been experimenting with multiple compressors on certain tracks just for fun; to triple and quadruple-compress something to see what kind of sounds it would make, but also to see how quiet it can remain. "I'm also fascinated with how people are making stuff sound so much louder. I hate it when it sounds distorted, but I am interested in when something actually sounds good, but extremely loud. There are certain people that have been able to achieve this — Dr. Dre records seem to be at least 40 or 50 percent louder in perceived volume than any other record and they don't sound distorted. It probably has to do with the fact that there are no rock guitars on those records; mid-range instruments take up so much space. I do hate the fact that people have taken it to the level where it's loud, but it's also distorted — and it actually hurts your ears. That I don't enjoy at all, because digital distortion is not pleasing."

Restoration With a massive amount of material in the Zappa family vaults — more than resides in those of many record labels — it must be difficult to know where to start the process of archiving and remastering for new formats. "People have no idea — it's mind-boggling. There's a film that's going to be coming out; it's a documentary about Frank as a composer, but it really gets into what his process of working was. When you see it, you realise how much he accomplished in 30 years — first of all, to make nearly 80 albums in that time, and have the potential for probably another 40 out of the vault. These are not like four-chord songs, this is serious music. He trained bands to play things that are incredibly complex and



difficult, but it stands the test of time. It's music that will be listened to, and should be listened to, for millennia. There's some incredible instrumental music, but he had a lot of great social commentary — I think he was widely misunderstood. "We have the oppportunity to listen to music from the very beginning — his first recordings in the early '60s — up until the digital recordings, and the things he was doing with the Synclavier. Because 5.1 is such a great way to listen to music — to actually get back into listening, as opposed to just watching it — I think many classic records are going to make a big difference to people in surround sound. Frank's music translates so well because there are so many intricate instrumental moments, and the textures in the recordings are so ahead of their time.

The projects Dweezil Zappa is undertaking include remixing and remastering many of Frank Zappa's classic albums.

"The challenge for us is to recreate and remix some of his most popular records in surround sound. I say 'recreate', because our biggest challenge is duplicating his working process. He would take things from alternate takes or live performances, and he would intercut them with what he was working on in a current session. He would edit them together and mix them down to two-track, and then create more edits. As the whole record was put together, there would be multiple edits from different takes. So to mix in surround, we would have to find the source of all those alternate edits — from there, make those multitrack edits, put the master together and then remix the whole thing in surround. It's like an archaeological dig! "We have a guy named Joe Travers, who we call the Vaultmeister — it's his job to go through these thousands of tapes, and listen. He's familiar enough with each record, because he's crazy, to start putting the pieces together. We're just now being able to find those pieces, because we weren't able to listen to stuff — plus, the labelling of the tapes is inconsistent at best. We found a lot of things that were quite interesting — there's a box that has a drawing on the back of it, and it's from 1970. There's a drawing of the room, and where the instruments are placed, and it's written on the box that it's a surround sound recording. So in 1970 Frank was thinking about surround sound, and he drew the picture so he could recreate the imaging later on. "It's fascinating that the recording sounds as good as it does, but also that he was thinking in those terms. I know there are people today who think they are so ahead of their time because they only write their music in 5.1. I think it's ridiculous if you're telling me that this music is only being heard correctly if you hear the guitar from behind you! And to the left! I've heard people talk like that, and it makes no sense to me."



Host Vs DSP Power Charlie Boswell is not only the point of contact for AMD's sponsored artists, but also travels around the world evangelising Opteron technology. "We have not only a 64-bit processor, but an integrated memory controller which provides lowlatency access, as well as Hypertransport [the intra-processor infrastructure]." Since programs like Nuendo and Cubase are multi-threaded, they should benefit from multiple processors, so the ability of the architecture to scale up is a critical factor. AMD has also talked about multi-core CPUs, which will effectively squeeze two or more chips into one package, adding further processing power to the host, and therefore diminishing the need for dedicated DSP cards. As general-purpose CPUs get ever more powerful, is a move away from custom DSPs an inevitable trend? Naturally, Charlie thinks so. "When your modem used to sit externally to your PC, that was a 150-dollar piece of hardware. That was one of the first things to become the victim of host-based processing. With software like Native Instruments, Propellerhead Reason, Ableton Live — all these programs replace traditional outboard gear using host-based processing. You need the processing power to keep this 'live' without intermediate rendering. There is a cost advantage; one would once have had special, boutique hardware that's expensive not only to purchase, but more expensive to maintain. You have fixed-point DSP hardware, and the audio application requires double-precision floating-point arithmetic, with full dynamic range. People are beginning to hear the difference between a front-end DSP that does some voodoo on the arithmetic to try to emulate floating-point, versus a normal PC architecture that actually supports double-precision floating point. When you have lots of effects and virtual instruments, each one of those has to have full dynamic range; otherwise sonic integrity suffers." In future, the much larger directly addressable memory of 64-bit systems may become important in the studio. "On a 32-bit system you have a wall at 4GB RAM, and when you hit that wall you begin to swap to disk — that means real-time operations suffer. Virtual instrument developers have many ways in which they would like to innovate, but on 32-bit systems they are constrained by the architecture."

A Question Of Preservation The Zappa family have now been able to complete projects based on Frank's material, both released and unreleased, using the 64-bit hardware. "We have several new releases coming out, including Apostrophe and Over-nite Sensation. Now that Joe Travers is up and running on his Nuendo system, he's archiving away and listening to all these things that absolutely no-one else gets to hear; that is, until we decide to release it. We could put out a series — put together by Joe — whenever we want, and still do everything else we've promised to do and need to do. In addition to all of that, this is our first totally in-house project, with the exception of mastering, done by Stephen Marcussen. We managed to do everything from start to finish within six weeks. "We have a lot of challenges ahead of us for the studio, to get ourselves to where



we're comfortable. With things moving so fast, we're running into problems of archiving, and everything that you can imagine that applies to a huge recording company applies to us because we have so many tapes. We have to bake the tapes, and make the transfer — but the question is, what is the new format for it to live on? Is it going to be on digital and analogue as well? Our situation is that we don't have enough room to duplicate everything in analogue again — we would have a building twice the size. So you need to be able to have a stable digital environment, but no-one has come to the forefront with anything that anyone's confident will last more than five or seven years without problems. If you don't start up a hard drive for a while, you have the potential of not retrieving that information, even though you made a clean transfer at the time. You're constantly in fear of losing information. "If an analogue tape has one play left in it, how are we going to have the music around for the future? We have thousands of examples of that — some tapes don't have the oxidisation problem, and then others clearly do. You can go through the vault, and see the history of tape. There's two or three kinds that stand the test of time, and others that when you open the box, it's dust, literally falling off the plastic."

Sounding Right There's also the question of which digital release format best does justice to the material in sonic terms. "Virtually no-one gets to hear what it sounds like in the studio when you're just hearing the purest version of what was recorded. In the best possible reproduction of it, you're hoping that it's going to be as pristine as possible, obviously, and I think DVD-A and SACD certainly give you a much better representation of what the actual master sounds like. When things get reproduced, sometimes they may not be using the best equipment to reproduce your stuff that gets out there — you may have some batches of CDs that don't sound as good as others. That was certainly the case with vinyl records; if they de-horned the master, you would lose the high end. "We're probably going to release a few records on SACD, and I've listened to a few different records that have come out in that format. Certainly when you hear a stereo version SACD and a regular CD there's a remarkable difference — but you're going to need the right player. Even though it may say Super Audio CD, there may still be a PCM translation problem on the low-end players. Is the average person going to notice the difference? They probably wouldn't unless they did the same comparison that you'd do in the studio. "I did some pretty extensive tests with some transfers recently, between Nuendo, the Euphonix R1 recorder, and the Sonoma SACD workstation. The machines

Photo: Ebet Roberts / Redferns Frank Zappa would often cut together



pieces from both themselves have their own playback characteristics, so studio and live we made sure that we weren't hearing one come back performances. louder than another — because louder always sounds better. The one that came out sounding the best was the Sonoma, but it wasn't by that much. In the studio the R1 and my Nuendo workstation sound pretty close, but by the time it gets put on a CD, it's not necessarily going to sound exactly that way.

"The Sonoma transfer is not going to change, so you're going to able to recreate that studio experience in your home. It's all good — especially when we start getting into the multi-channel stuff. That's when you're going to hear the biggest shift in quality, because there's less compression and there's more room for individual instruments to find their way into speakers. "I'm just excited to be in a position with the studio to work on not only my own music, but things that I've always wanted to do with my dad's music — to expose his music to a newer generation. There's never been a better time to show the difference between my dad's music and the musicians that played with him, versus what people think are the best musicians now. They should watch the Baby Snakes DVD and see the level that people were playing at 25 years ago." The official web site for all things Zappa can be found at www.zappa.com. Published in SOS September 2004 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



Jan Hammer

In this article:

True Expression Going Soft? The Feel Of Wheels Sound & Vision From Stereo To Surround Home Sweet Home Swapping Roles

Jan Hammer The Synthesizer Performer Published in SOS September 2004 Print article : Close window


As a member of the Mahavishnu Orchestra, a composer for film and TV and a performer in his own right, Jan Hammer helped to define the role of the synth player in rock and jazz. Paul White

Born in Prague, Czechoslovakia, Jan Hammer began playing piano at age four and by the age of 14 was performing and recording throughout Eastern Europe with his own jazz trio. A scholarship at the Berklee School of Music in Boston prompted his move to the US where he became a citizen, but today Jan's name is far more closely associated with the synthesizer than with the piano. Indeed, it could be argued that Jan was almost single-handedly responsible for establishing the electronic synthesizer as a virtuoso performance instrument in its own right rather than as a substitute for traditional instrument sounds. Since 'going electric' Jan has played with a Photo: Scott Weiner huge number of influential bands and Jan Hammer on stage in the 1970s. musicians and has evolved a distinctive and instantly recognisable synthesizer style. Who can forget those wailing synth lines in Miami Vice or the seminal albums recorded with the Mahavishnu Orchestra and British guitar legend Jeff Beck? Jan's project file reads like a Who's Who of music (Jeff Beck, Mick Jagger, Billy Cobham, Stanley Clarke, Tommy Bolin, Carlos Santana and Al Di Miola to name but a few), but a whole generation of TV viewers will remember him best as the composer and performer of the high-energy Miami Vice soundtrack with its thundering drums and searing, guitar-like synth lines. The theme hit number one

file:///H|/SOS%2004-09/Jan%20Hammer.htm (1 of 9)9/26/2005 12:06:59 AM

Jan Hammer

on the Billboard Hot 100 Singles chart and picked up two Grammy awards. 'Crockett's Theme', from the series, became a huge hit in its own right. Numerous film and television projects followed, and UK viewers will have heard Jan's work on the TV series Chancer and more recently Red Cap. He's also done music for Tales From The Crypt and Knight Rider 2000 as well as a few major film releases. In January 2001, reruns of Miami Vice on US television made clear that there was still an audience for an album based on music from the series that had never been made available on record. Miami Vice: The Complete Collection was thus released in 2002 as a two-CD set, and Jan's 1975 debut album The First Seven Days has also now been reissued in a digitally remastered form.

True Expression So how did Jan originally make his transition from piano to synthesizer? "It was an ongoing process where the piano was eventually not enough. The piano is a great instrument, like an orchestra at your fingertips, but I was hearing things with more fluidity and more movement. I wanted the tonal quality of bending notes and adding vibrato and all that. I heard things that were more expressive in that sense — and you just couldn't do it with a fixed pitched instrument. So I played electric piano, organ and eventually the synthesizer. I got hold of one of the first ring modulators that allowed me to bend the sound of the electric piano in such a way that it hinted at some sort of movement, but at the same time I was hearing synthesizers on albums such as Switched On Bach. Even then, the synthesizer was being used more as an orchestral instrument rather than as a solo instrument, and even where glide was being used, it wasn't being used expressively like playing a violin or even a voice. When I first laid my hands on a Minimoog with its pitch-bend wheel, it was truly a Eureka moment — I thought 'Wow, this is it! This is what I'm looking for.'" Most musicians associate the name Jan Hammer with the almost guitar-like synth tone that was used to great advantage on the Miami Vice theme music and underscores. Evidently much of this is down to controlling the pitch of the instrument using its pitch-bend wheel, but I wondered if there were any other special techniques or equipment involved. For example, was it necessary to play the synth through a guitar amplifier? "I did all kinds of things, including playing the synth through a distorted guitar amplifier, but now all that can be done in software using things like Line 6's Amp Farm in Pro Tools so I don't use a guitar amplifier any more. Whatever I use, I'm going for the same end result — an exciting rock & roll-like sound. It really started in the Mahavishnu Orchestra when I was looking for something to cut through that incredibly busy sound that we file:///H|/SOS%2004-09/Jan%20Hammer.htm (2 of 9)9/26/2005 12:06:59 AM

Jan Hammer

were creating. I had to have a sound that would project, so I used guitar amps and that's when the sound got really exciting." Just as many guitar players have traded in their huge stacks and racks of pedals for smaller, more technically advanced systems, Jan has also simplified his approach to recording and live performance. "For recording I use Amp Farm, though in the case of my Korg Triton Extreme, I use the onboard distortion effects. I've been using Korg synths for around 15 years now and the way their onboard effects are integrated is fantastic, to the point where I don't even use an amp on stage — I go straight out of the Triton Extreme into the PA and the sound is all there. On the current tour I'm doing with Jeff Beck, it's the only keyboard I'm using. I can set up my different sounds as keyboard splits or velocity switching and it does everything I need without the gymnastics of having to reach around multiple keyboards." It's clear, then, that Jan is not a musician to get hung up on specific vintage instruments or even the analogue/digital divide. "All the hum and noise associated with analogue has been banished. Obviously it took a while for the digital sound to develop, but with modelling and the onboard effects, I think the new digital synths prove that digital sounds can have as much warmth and presence as analogue sounds. Between the Korg Z1 and the Triton, the sounds are all there for me. I'll occasionally use samplers when working on film and TV music, especially if I need some crazy loops or grooves, but mostly I use my Korg synths. When it comes to samples, I like things like Distorted Reality from Spectrasonics. A new patch can set you off writing a new piece of music — it's like the grain of sand that starts the creation of the pearl."

Going Soft? When I met Jan he had a Mac G4 Powerbook setup in his hotel room, so I asked him whether he could envisage a day when he could tour with nothing but a master keyboard and a laptop. "It is definitely possible, but right now, for my peace of mind, I like hardware. I know that much of what goes on inside the hardware is virtual and runs on computers anyway, but there's something reassuring about a dedicated keyboard instrument. "There are some extremely good software instruments and I particularly like the Access Virus plug-in that's available for Pro Tools. I use that a lot and it has a wonderful, solid sound. I've heard lots of other things that I've liked but it's hard to pick anything out specifically because I get such great sounds from the hardware instruments I'm using so I tend not to use many software instruments. As software processing goes, I like the Waves processing plug-ins and I have used their Maxx Bass to reprocess an album I recorded that I felt was lacking at the bass end, probably due to my monitoring environment at the time. I'm from the old school of outboard gear so I particularly like their Renaissance-series plug-ins and their new convolution reverb is just incredible. I use virtual effects much more than virtual instruments." Given Jan's obvious fondness for Korg instruments, I wondered if he'd been tempted by their new Legacy Collection? "It's a bit too much like going back


Jan Hammer

whereas I'm more interested in taking synthesis forwards — how far can this instrument take me? As much as I love the Minimoog — I have two of them — I tend not to use them any more. I don't have nostalgia towards everything old."

The Feel Of Wheels Being a guitar player (or at least an owner who makes a noise with his possessions), I know how obsessed players are about the way their instrument feels and interacts with their fingers. By contrast, keyboards are basically levers with switches underneath, but performance controls such as wheels and joysticks allow the player to introduce a high degree of expression if they are used effectively. I've often wondered if keyboard players get as passionate about the feel of these performance controls as guitar players do about their guitars, and what happens when you switch between wheels and the joysticks. "I used wheels on the Moog and the DX7, and even had wheels installed on some of the early Korg instruments such as the T3 because I couldn't deal with the joystick. It was the same with some of the portable synths I used because they often made them with ribbon controllers, which are completely useless, so I had to add wheels to those. Now I'm using the new Korg instruments with the new joystick design and getting better tactile feedback than I got from the wheels. In the studio I still use wheels for certain parts, but on this tour with Jeff, I'm using only a Triton Extreme and I'm very happy with it." While Jan obviously feels very comfortable with Jan today, in the rehearsal studio his present instruments, I couldn't help with Jeff Beck. wondering what he'd ask for if he could influence the design of future keyboard synths. "I've always wanted a synthesizer that sounds as live as a string with all its convoluted cycles. A string is so dynamic, whereas a synth is so static, so you have to put flangers on it or some other processing to try to make it move and evolve like a natural sound. You try to make it respond like a living thing, whether you use velocity or aftertouch or whatever. I even tried one of those experimental keyboards where the keys moved sideways but that was too much — it was too easy to play out of tune if you put your fingers down and then they spread sideways a little. Keys were not meant to move sideways! I think that aside from mod and bend controls, velocity and aftertouch is about as much as you can do with the keyboard itself. The rest has to be done in software and it's getting better every year. I'm not in the business of thinking up new ideas, but when I hear it, I know what to do with it. I am the end user, not the designer, but I'm happy with the new instruments out there and I think modelling was the last big breakthrough."


Jan Hammer

These days, of course, synthesizers are predominantly associated with dance music, and virtuoso keyboard players are in a minority. Jan Hammer has no problem with dance music, but is disappointed at the lack of opportunity radio and TV provides for other styles of synth playing. "I have also produced dance music and I enjoy listening to some of it where there are stark sounds and imaginative programming, but with most of it there's a musician presence that's missing — where the musician has something to say," says Jan. "That used to matter, but now the concept of a solo is a no-no — they look at you as though you're from Mars — solos are distracting. They don't want anything with passion or anything that's burning, like the stuff we do with Jeff for instance. I think a market exists and I'm hoping that the Internet will prove to be a way of accessing that market by getting us back to the '70s FM radio paradigm, but it's still early days. The critical mass hasn't been achieved yet, but when everyone has broadband, I hope it will be different. At the moment, the music industry doesn't allow anyone who is musically inspired to be heard. There are only a couple of gatekeepers controlled by the big conglomerates and only three or four outlets for major, big deal music. Most people are barred from any access to it."

Sound & Vision If the pop world is eluding Jan, the film and TV world is a very different story, as a brief visit to his web site at www.janhammer.com confirms — if I listed everything he'd done, there'd be no room left for this interview. His crossover from virtuoso keyboard icon to film and TV music writer interested me, as the disciplines involved seem very different to those needed to play with a band like the Mahavishnu Orchestra. "Mahavishnu Orchestra was like the Olympic Games of music — higher, faster, stronger," says Jan. "It was all superlatives, so I wanted to back off and see what I could do with less. It was almost a visual approach, even though you don't see anything, but you can close your eyes when you are listening and imagine things. With my fist solo album, The First Seven Days, a lot of people heard it and thought about getting me to write music for visuals. At that time, all I had was a Scully analogue eight-track and I recorded everything the hard way, one track at a time. Now I use Pro Tools or file:///H|/SOS%2004-09/Jan%20Hammer.htm (5 of 9)9/26/2005 12:06:59 AM

Photo: Mark McCarty

Jan Hammer

Studio Vision depending on how much MIDI is involved.

This signature model Lync keyboard was created to meet Jan's requirements for a live performance controller.

"I've used Pro Tools from the very beginning, though I also use Opcode's Studio Vision, which was the first truly integrated MIDI and audio sequencing program, may it rest in peace! I'm very disappointed in what Gibson did to that product. I though Opcode were the most exciting software company working in that field and they are sorely missed by many people. I'm still using the last version of Studio Vision on a Mac G4 dualboot machine that I keep just for that purpose. But the Pro Tools MIDI side has developed enormously since the early days and as far as audio is concerned, they are it. In the old days, when you worked with multitrack tape, all the tracks were locked together in time, but now you can slip tracks, copy and paste — audio is arbitrarily movable. It means you can tighten up grooves and do incredible things. I don't agree with these people who say the sequencer is evil and only good for creating robotic music, because you can use it just like a tape recorder. It is nothing less, nothing more, and if you can play and have something musical to say, it will come out just the same whether it is on Pro Tools or on tape. Most of the people who make music today using copy and paste probably wouldn't have survived in the old days! "When I'm recording, I tend to record my keyboard parts as MIDI, because if I have a great take but there's one dumb note in the middle, I want to be able to fix it and keep the great take. I play a bit of guitar and that goes down as audio, and if there aren't too many MIDI tracks, I might record the MIDI parts as audio when I'm done so I don't have to fish for the sounds next time — it's there."

From Stereo To Surround Given that Jan is so immersed in film work, I thought he'd have some thoughts to offer on recording music in surround. "I have mixed some movie soundtracks where I monitored in surround via a Dolby matrixing system and then delivered a stereo master, but I'm not mixing in 5.1," he says. "I haven't being doing bigbudget motion pictures so I haven't been asked to produce discrete surround mixes and it's not something I'm really interested in. When I did the music for the BBC Red Cap series, that was all in straight stereo. I'm not sure that surround will take off as a music listening format because I'm not convinced that the impact is as great as when we moved from mono to stereo. It's interesting, but I still think it's a bit of a novelty. It's more impressive for home theatre where you hear the aircraft fly over. I think it is much more interesting for hardcore classical music than for rock or pop where you want the music to hit you from in front. I'm not sold on surround yet and I don't think many home theatre surround systems are good enough for serious music listening anyway."

Home Sweet Home With so much music to his credit, you might think that Jan spends most of his life file:///H|/SOS%2004-09/Jan%20Hammer.htm (6 of 9)9/26/2005 12:06:59 AM

Jan Hammer

in commercial studios, but nothing could be further from the truth. "I did all my records at home in my studio, which is a converted colonial New England house just north of New York City. I still live there, though the studio has since moved out into a purpose-built building in the grounds. Initially the studio was in the dining room, which we soundproofed and then we set up a control room next to it. This started with a Scully one-inch eight-track analogue tape machine, but by the time Miami Vice came round, I had a 16-track machine and later a 24-track. I was doing everything to tape, though there were some sequences set up on the Fairlight CMI which were also recorded to tape. I also had some early music programs running on an IBM PC — there was Roger Powell's Texture — but there was really no way to record all the MIDI parts in the way we do today, so all the keyboard parts were played in live. For the first year of Miami Vice, I didn't even have any sync setup so I just ran everything wild — I'd press Start on the video remote and then try to fly it in. If it didn't work, I'd have to start the take again. By the second year I got a proper SMPTE sync setup so it was all much easier after that. "Gated drums were the fashion then, I suppose because of Phil Collins, and a lot of the sounds came from my own studio drum kit that were sampled into the Fairlight and then processed using heavy compression and gated Lexicon reverb. I still do all my own engineering — I'm very much a one-man operation really. I monitor for 80 percent of the time on NS10s because I'm used to the way they sound, though I also have a huge pair of JBLs the same as they used to have in the Trident studio, just to check out the bottom end."

Photo: Justin Thomas A true supergroup: Jan Hammer (centre) with Jeff Beck, Ringo Starr and Dave Gilmour.

Most of the Miami Vice soundtrack themes were not structured as complete pieces, so for the newly issued Complete Collection Jan had to write new parts, rerecord the original parts using his old Memorymoog and antique Fairlight, and in some case, knit cues together to create viable album tracks.

Swapping Roles Currently, Jan Hammer is on tour with guitarist Jeff Beck, with whom he has played on numerous albums. It's an interesting partnership, because whilst Jan has often used synths to play guitar-like parts, Beck has developed some interesting playing techniques that make his guitar sound more like a synthesizer or Theremin, and the pair have now reached the point where, at times, Jeff is playing what appear to be synth lines and Jan is playing what sounds like a guitar part. "I think that is the secret to our compatibility," says Jan. "When we first met in 1973 we started talking and found we liked the same music, then he


Jan Hammer

came to my house a couple of years later and we worked on Wired. It was amazing because I found I was going more and more into the guitar realm and he was doing some of those unique tonal things that he does that aren't quite synth-like and it was amazing. We almost overlapped into each other's worlds. Some of the things he does are unbelievable and I've no idea how he does it. I've never heard anybody else do it and he is just mind-bending — wonderful." Did that mean their repertoire on the current tour was tightly orchestrated or was there plenty of room left for improvisation? "It's fairly organised — it's not a free-form fusion-fest. There are some tracks that Jeff did with Tony Hymas, a lot of my tunes and even one Mahavishnu Orchestra tune, so it's interesting. There are a couple of tunes where we can improvise rather more — places where we can let loose and Photo: Ebet Roberts the roof will come off — but overall the Jan in his studio, 1988. show is pretty structured and offers a lot of musical variety. There's no new material as such, but some of it is new for Jeff as we're playing arrangments of some of my album Beyond The Mind's Eye that he hasn't done before. On one of them, Jeff is actually playing a nylon-strung classical guitar, which is a first for him." I was fortunate enough to attend one of the band's concerts at the Warwick University Arts Centre and as expected, it was exceptionally good, with virtuoso playing from everyone in the band and great front-of-house sound. Jan was clearly having a great time trading licks with Jeff Beck and even though he only had the one synth on stage, he swapped effortlessly between his hallmark lead synth sounds, tonewheel organ emulations and some nifty electric piano. He even had the sound of one of Jeff's hot rods being revved with some enthusiasm sampled into his keyboard for the opening of one number. Perhaps after the lonely confines of his studio, getting out on the road and playing is just what Jan needs right now, but he's destined to be back in his studio before long as he's been commissioned to write and record the music for a new feature documentary series about the rise of the cocaine cartels in Miami. This apparently covers the time period from the '80s up until the present day, so I think it's fairly safe to say that before too long, we'll once again be seeing boat chases through the Everglades accompanied by pounding drums and searing synth lines! Published in SOS September 2004


Jan Hammer




Jim Sutherland: Producing Aberfeldy

In this article:

The Life Of Riley Making The Band Colour Coding Finding A Balance All Spill The Sutherland Sound Keeping The Atmosphere Plate Polish

Jim Sutherland: Producing Aberfeldy Mono Forever Published in SOS September 2004 Print article : Close window


Recent years have seen a revival in back-to-basics recording techniques, but few engineers or producers have taken things as far as Jim Sutherland did with Edinburgh folk-pop band Aberfeldy... David Greeves

The debut album from Scots indie popsters Aberfeldy, Young Forever, has all the ingredients of a classic Rough Trade release. It's full of charming, catchy melodies and deceptively simple songwriting with a laid-back pop sensibility, whilst the sensitive arrangements centre around acoustic guitar and loose-but-tight vocal performances, embellished with fiddle, organ, glockenspiel and a smattering of retro synth sounds. But for the average Sound On Sound reader, there's nothing really remarkable going on here: it's just a straightforward studio recording from a promising new band. Until, that is, you find out that it was recorded in mono, in one room and on one microphone. Photo: Ruth Barrie

The Life Of Riley

Jim Sutherland in his Edinburgh studio.

So when I spoke to Young Forever's producer, Jim Sutherland, my first question (immediately before 'how?') was 'why?'. "I'm always working on several projects at once," says Jim, "so there are always lots of different people popping in and out of the studio. Riley [Briggs, Aberfeldy's lead singer and songwriter] was one of them, and I wanted to get the ball rolling on a project based around him and his songs. I got some other friends to come in, just to see what was going on.

file:///H|/SOS%2004-09/Jim%20Sutherland%20%20Producing%20Aberfeldy.htm (1 of 10)9/26/2005 12:07:02 AM


"We started multitracking the songs, but it didn't really capture what was special about Riley. There's something special he has when he's sitting with his guitar and he sings. Maybe he kind of gets off on something to do with the resonance of the guitar, I don't know, but it certainly makes him sing better, and that was what I really wanted to capture. So I thought 'We'll just demo these songs for now,' and bunged up a mic. That was kind of the beginning of it all. "Those first one-mic recordings were just for ourselves, really, but I felt there was something really worth pursuing in them. They had real atmosphere, and I thought it would really great to be able to create the standard of production that people are more used to hearing these days and also retain the atmosphere that you get when a bunch of people play in a room together." Inspired by the sound and feel of their initial demos, the newly formed group were coming round to the idea that a whole album really could be made this way. But the decision was also tempered by Jim Sutherland's own views on how the multitrack recording process influences the way that much modern music is created. "On lots of records these days, somebody'll play some drums and somebody else'll come along and put down something else, then they'll send the tapes to New York... And it's cool that you can do that kind of thing, send stuff over the Internet and so on, but actually getting a bunch of people to sit down in a room and interact with each other is also pretty incredible. And that's an element of music-making that we can tend to forget. There's a feeling that musicians have when they're right in front of each other that you just don't have when you're sitting in a booth."

Making The Band This, of course, is how all records used to be made — the group would go into the studio, stand in front of the mic, cut the record and then leave. But the Aberfeldy project was not simply a return to the old way of doing things. For one thing, when Riley Briggs walked into the studio there was no record and no band: both were formed in the studio. What's more, the musicians brought together for the project, recruited Magnificent Seven-style from friends and acquaintances on the Edinburgh pub music scene, had little or no previous studio experience. Jim Sutherland's varied career, which has seen him being signed to Sony as a songwriter, to Columbia as an artist and writing music for film, theatre and TV, includes a background in Scottish folk music, but he confesses to little experience of traditional one-mic recording. For both band and producer, then, making the album in this way was a new experience. Getting a suitably polished finished product would be far from straightforward. One thing that made that goal seem feasible enough to make it worth proceeding was Jim's realisation that although he couldn't mix the recording or replace individual parts within it, he could nevertheless achieve an awful lot in postproduction. Dynamics processing, and multi-band compression in particular, allowed him a degree of control over the recording, and he's effusive in his praise file:///H|/SOS%2004-09/Jim%20Sutherland%20%20Producing%20Aberfeldy.htm (2 of 10)9/26/2005 12:07:02 AM


for the Waves Masters Bundle plug-in suite. "I'd just started using the Waves gear and I found it incredibly useful — I couldn't have done without it really. I used the phase-linear EQ, followed by the multi-band and then the L2 limiter. The multi-band compressor is amazingly transparent. It's also phaselinear so there's no messing about where the crossovers are, and I found I was able to adjust the balance of the mix without any of the usual compression side-effects. I could As well as a full drum kit, additional totally sit in the snare transients whilst percussion included the Korg Wavedrum. keeping the vocals at the front of the mix. Being able to focus in like that on five different frequency ranges of a one-mic recording almost lets you mix it!" Jim also found out that although everything would be recorded on one mic, he could still edit the takes. "I'd been playing around with the demos we'd done and realised that it was possible to combine bits of different takes as long as I was careful about where I made the edits. I couldn't retune the vocal and whatever else, but I don't want to be doing that anyway — that's not really what music is about for me."

Colour Coding Jim Sutherland edited all of the tracks on the album in Logic. "I don't know how many people use Logic's colour scheme the way that I do. Most people just go, 'Right, I'll colour the drum track red, the bass blue...' and so on, and I do that as well. But I also have a particular colour scheme that I use for editing. The version of Logic I've got lets you colour in individual parts of the track differently, so I can go through each take and mark up which Jim Sutherland used Logic's colour-coding sections are good or bad and so on. facility to identify the best parts of each take. Being able to do that is great because, instead of making copious notes, a year later I can go back to that track and I immediately know what's what and I can do a re-edit. "Generally, I was working around one master take and just nipping in out of other takes for the odd word or line. Most of the finished tracks probably include sections from three or four different takes. It was really important that the band had stuck to the arrangements we'd decided on and worked so hard on playing



consistently from take to take during the recording. If they hadn't, it would have made the editing even more difficult." Although each take was a pair of tracks — one for the mic and one for the DI'd bass — in general they were treated as if they were a single track. "The bass was kept together with the mic track for pretty much all of it. Occasionally I would have a couple of notes from the bass part in one take hanging on under the take it was going to, so the edit points on the mic track and bass track would be very slightly staggered. I think it's good not to edit everything in the same place. For example, I might edit the microphone recording on a good, hard edit — just before a cymbal or a kick drum, say — then I'd edit the bass a little further on, so you'd have some continuity in the bass to keep it smooth." With sometimes as many as 15 takes to go through, and as many as 30 edits in a single track, editing was a laborious process. "Yeah, the recording was the easy bit! Not for the band, of course — they end up doing lots of takes, but musicians often end up doing lots of takes. I don't know if it's possible to hear any edits, but I don't think so. It's hard for me to tell because, obviously, I know where they are!"

Finding A Balance For the recording itself, the signal chain couldn't have been simpler: a Microtech Gefell UM92IS mic, fitted with a vintage Telefunken EF86 valve, went straight into a preamp in Jim Sutherland's ageing Yamaha 02R mixer then straight into Logic, running on a PC, recording 16-bit audio. While Sutherland is well aware that this setup is hardly state-of-the-art, he's unrepentant: "Let's be honest, a straightforward 16-bit recording using a nice valve mic, straight into a computer — there's nothing wrong with that." Indeed, he maintains that the most important factor in a recording of this kind is not so much the equipment you use but the room you record in. Thanks to comprehensive acoustic treatment (see box), his small studio space proved up to the job. But while the room's well-balanced acoustics would clearly be an ally in the recording process, many other obstacles still stood in the way. Perhaps the biggest challenge was achieving the right balance between the instruments, which included a lead vocal, two backing vocalists, acoustic guitar, bass, a full drum kit, violin, glockenspiel, synths and, on occasion, a trombone too! Needless to say, arranging the instruments around the microphone was a lengthy process. "We spent hours figuring out the best positions for everything. This project has taught me so much about mic positioning, or, in this case, people positioning! I guess you can think of the microphone as being like a camera, and the mix comes together a bit like the way a photographer composes a wedding photo. "The mic was set up towards one end of the room. Riley sang between six inches and two and a half feet away, moving in and out depending on his dynamics. He was also playing the guitar at the same time and had to be careful not to move it too much. Sarah [Macfadyen] and Ruth [Barrie] were in a triangle around the mic with Riley. They played a variety of instruments — including violin, glockenspiel and keyboards — and sang backing vocals. They had to really work to get the balance, leaning right into the mic when they were singing."



Recording the bass through the same microphone proved more problematic: "We tried to record the bass acoustically through the mic but we just couldn't get it to balance with the other instruments and still sound good. When it was clear that it wasn't going Only battery-powered amplifiers were used to work we decided to just DI it. But for the electric instruments, and included this that wasn't really a compromise — the home-made 'biscuit tin' amp. project wasn't about doing a one-mic recording to make some kind of point. It was about trying to get something that sounded good." So, in the end, bass player Ken McIntosh's basses — sometimes an upright electric bass and sometimes a conventional electric — went straight into the desk, though he was standing close enough to the mic that some string slap and fret noise could still be picked up. As you might expect, the drum kit was placed some distance away. "With the drums, volume was always going to be a problem. The kit was set up around eight or nine feet away from the mic. Ian [Stoddart], the drummer, had a difficult job and did it very well. He had to control his dynamics whilst playing much more quietly than normal. There was a huge pressure on Ian as a drummer. A lot of the songs are quite delicate, and he was at a distance from everyone else. But I think he did a great job, especially in terms of being consistent." In addition to modifying his playing, Ian had to modify his kit with all manner of damping apparatus. Jim, a percussionist himself, pulled out just about every trick in the book. "We had to try and bring down the level of the drums without changing their sound too much. We covered the top of the snare in cardboard and then, to get some of the initial attack back, taped a bit of plastic — one of those plastic CD blanks you get at the top of a stack of CD-Rs — on top of that. Of course, every couple of takes that would break and have to be replaced. We went through hundreds of the things! I taped coins to the bottoms of the toms to shorten the ring a little and to stop them ringing on every snare and kick. It's bad enough having all that mid-range washing Photo: Alan Mcready about under normal circumstances, but with the With only a single mic used to single mic, it would have just destroyed the guitar capture all the instruments, sound. We stuck loops of gaffer tape to the distances between the undersides of the cymbals, too. If you just tap a performers and the microphone were critical. cymbal, it makes a noise, but we had to make it so you could get power and attack out of the cymbals and play quietly at the same time. As for the bass drum, it actually sounded pretty much OK. We just added a bit of damping around the edge to kill some of the higher, slightly honky frequencies and that was it." file:///H|/SOS%2004-09/Jim%20Sutherland%20%20Producing%20Aberfeldy.htm (5 of 10)9/26/2005 12:07:02 AM


The sound of the kit on the album is impressive, especially if you consider that some people use five mics on the drums alone and end up with a less satisfactory sound. Again, Jim feels the room acoustics played a big part. "I think the drums sound great — quite big and fat, even though they were so far from the mic. That's the great thing about having a fairly dry room that's so even through the frequency spectrum — you can be a little further away from the microphone and it still sounds close." Besides these acoustic instruments, the electronic keyboards used on the album — a Roland SH101 synth, a Travelpiano electric piano and a retro Charlie electric organ — are a key element of the Aberfeldy sound. "One of the problems we had was dealing with hum from the keyboards. I started off plugging them into proper amps [in this case Sutherland's beloved but aging Selmer Truvoice and Vox AC30 guitar amps], but when you add it all together you just end up with a racket. It sounds like a live band in a rehearsal room and that's no use!" The solution was to use small battery-powered amps, and adjust their volume and distance from the mic to blend in with the other instruments. "Ken brought in this fantastic battery amp he had built using a biscuit tin and an old Archer amp. We also used a little Dübreq battery amp which is pretty old. Ian was playing my Korg Wavedrum as part of his kit for some of the tracks, and that went through a battery amp too. There's no problem with hum with the battery amps and they really give the keyboards a certain character that they wouldn't have had otherwise. We did end up using the Selmer in the end, though. For the synth lead on 'Heliopolis At Night', we put the SH101 into it via an old Melos delay unit — I think that's about the only effect we used on the whole album!"

All Spill Once a workable setup had been established, the process of arranging the instruments and musicians became less time-consuming. And it would be fair to say that time spent before recording began was time saved in post-production. For one thing, when you record with one mic, there's no mixing to be done, and it would take a good deal of time and no small measure of skill to mix a multitrack recording to sound as cohesive as Young Forever. "I guess in some circumstances people see spill as a real problem, but in this case, it was all spill! The sounds mix together naturally in the room and everything seems so much more connected — the room's acoustics act as a kind of glue for the sound. But that only goes for what was being recorded. When you were just in the room it didn't sound right at all. There was a real leap of imagination required on the part of the players, because, for them, it just sounded like a pile of nonsense. The drummer's way over on the other side of the room, the bass is being DI'd, everyone's facing different directions — it all sounds different depending on where you are in the room. The only perspective where it actually sounds anything like a mix is the perspective of the microphone, and what I have in my headphones.



"Ken, the bass player, had to wear headphones too so that he could hear what he was doing — he'd maybe have one headphone on and one off — and sometimes Ian, the drummer, would wear headphones as well, but no one else did. They could have all worn headphones, but I think it would have broken up that atmosphere and connectedness that I was trying to capture. I also found that, with Riley, when he was singing with headphones he found it more difficult to pitch the notes. I think it's the same with a lot of people, though you can get used to singing with headphones. But the way we did it, the three vocalists were right in front of each other, with no barriers between them. "But there was an awful lot of faith required. They all worked really hard on being consistent — how hard Ian hit his snare, how close the singers were to the microphone, how they were holding their instruments — while not really being able to hear how it was all coming together. I did play back the tape from time to time, just to reassure everyone and so we could see how we were doing, but I didn't want to break everyone's concentration. We had to be confident in the arrangements too, because, obviously, they couldn't be changed later."

The Sutherland Sound Jim Sutherland's studio, a fairly small rectangular room in a block of artists' studios just off Edinburgh's Royal Mile, has received extensive acoustic treatment. "There's no point going out and buying a great microphone — which the Gefell is, by the way — if your recording room sounds shit! My friend and collaborator Nick Kinloch and I planned the studio together and we worked very hard on the acoustic treatment, particularly for the mid-range and bottom end. Neither of us had ever done anything like that before so it was all a bit Heath Robinson — entirely Heath Robinson in fact! Having said that, it worked out very well." The pair found the exact resonant frequencies of the room by playing a sine wave on a synth through a bass amp and noting down the notes which made the room 'ring'. Nick Kinloch then set about building some broadband absorbers and some Helmholtz mid-range traps, precisely tuned to these frequencies. The wedgeshaped traps, constructed from wood and half filled with rockwool, have holes or slits cut in them as dictated by the Helmholtz equations. The traps were hung from a dado rail which runs along the length of the walls, so that they could be moved and rearranged to find the right positions. A large bass trap, more broadband absorption and two large softboards, set at an angle to break up and absorb reflections, were attached to the ceiling and hidden by an old army parachute. The pair also constructed a small vocal booth along one wall, which didn't find a use on this project. "I don't really use the booth — we did a great job of insulating it and it's really dry — too dry almost, right the way through the frequency range. It's great for voiceovers and things like that, but not really for music, whereas the room sounds great."

Keeping The Atmosphere Building and maintaining this confidence and concentration was a big part of Jim



Sutherland's job. "When you've got five different people sitting down to record like this, you have to get everyone to focus together and leave everything else behind. Whenever you go to the studio it's a bit like that anyway, whatever kind of recording you do. But in this situation it was especially important — with one mic, everyone has to do a good take at the same time. If someone had an issue or a problem it had to get dealt with. There was nothing major — we're all good pals, which was a great help — but sometimes tensions were running very high. People would get frustrated, or someone might come late — I had to be right on top of all those kind of things. Somebody coming late could be a major issue, because you totally lose the atmosphere if you're all just waiting around for one person. "I tried not to put too much pressure on them in the early stages, but I'd have them in place, getting used to the slightly odd sitting positions they had to be in. Sometimes Riley would have to tilt his guitar up-the-way, for example, at the same time as trying to keep his distance from the microphone right for the vocal. So it wasn't easy on the band, who didn't have much experience of working in studios anyway. But I think the fact that people ended up in positions that weren't necessarily always comfortable for them had a psychological effect. I think it possibly makes you less nervous about playing when you've got to think about all these other things. And there's an incredible focus from having the one microphone — it's like having an audience. Everybody's united in their approach to that microphone. "Riley had an especially difficult job, because he was the lead singer. Sometimes we'd record all day, starting at 10 in the morning, and there are certain tracks where we did 20 or more takes, so by the time the band were coming together and really sounding great, Riley'd be knackered. The rest of the group would be just beginning to peak and Riley would be on the edge of going over the other side. That's a real feature of the one-mic recording — you've got people right on the edge of that a lot of the time and you've got to keep them there. You're trying to get everyone to hit it at the same time, and that's exciting! It makes for exciting music too." Although the aim was to keep the atmosphere Photo: Alan Mcready focused but relaxed, even the producer was feeling Aberfeldy in mid-take around the strain. "It was dead nervy! There was this Jim's Microtech Gefell valve mic. incredible tension in the room. On some tracks I was playing a few bits of percussion — things like small bells and crotales, just to brighten up the top end. I'd be sitting by the desk with my headphones on, trying to keep an eye on everything and feed information back to the band, but also joining in as well. So there was as much pressure on me as anyone. You could be close to tears sometimes — any one of us could've been. But then you get to the end of the song and it's absolutely bonkers! It's a brilliant feeling.



"We established kind of a rule that I was the only person who could stop a take, because most of the time a wrong note or a word that you couldn't quite hear wasn't really a problem because it could be edited out. Tempo wasn't too much of an issue either, but then, as I've said, we had a very good drummer." Although a take might start with a click or a count-in, playing to a click track throughout was out of the question. "You might think that you couldn't edit two takes together unless they've been played to a click, but, unless they're just miles apart, you really can. In the same way that I was saying singers can get used to using headphones, some drummers can play pretty well to a click track, but mostly you end up with these weird, rigid, careful performances. But when a drummer's just playing along with the rest of the band, and he plays a fill, he does speed up a little bit. In fact, the whole band will speed up and slow down going in and out of a chorus. There is some come and go in the tempo of natural playing. And, in a way, that's why some of the editing was made easy, because within the tracks there was a natural breathing of tempo, so you could find places, like on a fill, where it actually feels quite natural for the track to speed up slightly. "I guess you could say that not having headphones, or a click track, or the safety net of multitrack parts and retakes made it harder for the band, but the musical benefits were huge. The way I look at it, it made it easier for them to play like a group of musicians and get a real good performance down."

Plate Polish Once the finished tracks had been edited (see box), they were lightly treated with an old EMT gold foil plate reverb, a true mechanical plate reverb which used stretched gold foil plates instead of steel to achieve the same effect from a smaller plate. Jim Sutherland then took them to his friend Callum Malcolm's studio in North Berwick for a final polish. "We didn't have to do much, to be honest. It was more a question of getting the levels right from track to track and going somewhere else to check them on another set of speakers. I'd used the Waves L2 already and squashed things as much as I wanted to squash them. There were a couple of tracks that I took back a stage in terms of processing to change a few things — that's the beauty of working on a computer. "That's the thing: the way we made the record wasn't supposed to be the antithesis of modern recording — we incorporated lots of modern techniques and technology — but the emphasis was on the music and the musicians. I think it would be great if people thought a little more about what their options are and didn't just throw away the past. I mean, this was recorded on one microphone, in 16-bit, in mono!" Published in SOS September 2004







Let The Public Decide... Leader Published in SOS September 2004 Print article : Close window

People : Industry/Music Biz

We all want to make good music, but 'good' can be especially hard to define. Perhaps, then, we should replace 'good' with 'music that at least some section of the populace would like to hear'? On the other hand, it would be folly to say that the best music can be defined as being 'the music that the greatest number of people want to hear' — you only have to listen to the lowest-commondenominator tosh that spouts from radios and TVs to realise that. That said, I think it's fair to say that most composers want some of the public to like their music. The continually falling cost of technology has enabled almost all musicians to afford some kind of recording system of their own, but as most of us have discovered, working alone in the enclosed bubble of a studio isn't the ideal way to evaluate how your music will be received by the world at large. Asking friends and family for input isn't usually much help — either they are sick to death of hearing it, or they tell you what you want to hear. A friend of mine was offered very blunt advice by a record-company executive when he gave him a recording he had made saying that it warranted a release because everyone who'd heard it thought it was great, and should be made commercially available. The executive's response was to say that whenever somebody tells you how good your music is, you should ask them if they're prepared to invest in it. If they're not, their comments are of little worth. I don't know if I'd go so far as to subscribe to that view myself, but I can see where he was coming from — there's little risk for the musician, but a record label has to invest a huge amount of money in an artist. A more practical way of seeing if your music has public appeal is to take it on the road and play it to the public. This could either be in the form of a live performance, or by having sample records made to play at clubs, if that's your potential market. It's surprising what you can learn by doing this. For a start, playing your music in front of an audience heightens your own sense of quality control, so if you find certain sections of your music (such as over-long intros and bridges) make you feel anxious, then perhaps they need more work. The other constant surprise is that the music you have lovingly crafted over the past six months often meets with almost total indifference, while the three-chord bash you wrote last night goes down a storm. Alternatively, the Internet means that you can 'go live' without actually going anywhere or being live — it's simply a means to put your music before a greater number of people. However, the Internet is a big place, and is already awash with unsigned bands, so it's very hard to get anyone to even listen to your music, let alone buy it. Nevertheless, I feel that Internet radio technology combined with record/download sales offers a way forward. The technology already exists for Internet radio stations to set up multiple channels that can be sent to different listener groups based on information provided about their current listening habits. It follows then file:///H|/SOS%2004-09/Let%20The%20Public%20Decide....htm (1 of 2)9/26/2005 12:07:05 AM


that those radio stations that can best optimise their programme material to individual listeners will get the biggest following, and if the listener can press a button for artist info, album listings and samples of other songs by an artist they like, they're more likely to press the 'Buy' button at the end of it. Sounds great, doesn't it — but who's going to finance all this in the first place? This takes me back to the record-company executive who suggests asking people if they'd be happy to put money into your record, because if anybody should be prepared to make that commitment, it's you. Unsigned bands could pay a sensible amount of money to have their tracks added to a radio and record shop web site, so that the listening public really does get to decide whether their music is worth buying or not. Paul White Editor In Chief Published in SOS September 2004 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


file:///H|/SOS%2004-09/Let%20The%20Public%20Decide....htm (2 of 2)9/26/2005 12:07:05 AM

Sounding Off: Online Music

In this article:

About The Author

Sounding Off: Online Music Daniel James Published in SOS September 2004 Print article : Close window

People : Sounding Off

On-line music: are we missing an opportunity to make audio distribution a better deal for musicians? Daniel James

The legal download services prove one thing — music lovers really are willing to pay for songs that they could get elsewhere for free. The iTunes Music Store claims to have sold over 70 million tracks in its first year alone, with the reborn Napster and other clones hoping to emulate Apple's success. Assuming for a moment that this isn't just due to avoidance of the RIAA's civil lawsuits, we would hope that this means the people who appreciate music can also appreciate that it costs money to make good recordings, and this has to be recovered somehow. For many years, artists with an eye on their record label's accounts have dreamt up plans to replace the expensive and wasteful distribution of physical media. Frank Zappa once proposed to a firm of venture capitalists a service based around cable TV delivery and blank digital tapes, but perhaps it was just too far ahead of its time to get off the ground. Now the technology does exist, and portable hard disk players have proved a viable target platform, but the financial deal doesn't appear to have changed for many of the people who actually create the music. Just as in the early days of Compact Disc when artists were still being charged for shellac breakage and yet received a lower royalty rate for CDs than for vinyl (because CDs were 'new technology' and supposedly expensive to make), the sums could still be tilted against artists. So far, it looks like the benefits of network distribution aren't always being passed back down the chain. The cost savings involved are considerable — there's no packaging to print, there are no unsold returns

file:///H|/SOS%2004-09/Sounding%20Off%20%20Online%20Music.htm (1 of 3)9/26/2005 12:07:08 AM

About The Author Daniel James is a freelance technology writer specialising in Linux and audio. He runs a small studio on the edge of a cliff.


or shipping costs. There are no boxes of unlistened-to promo copies waiting in the attic of supposed 'reviewers' for the right time to be flogged on Ebay. Arguably, labels need a significant cut to allow them to underwrite recordings that may not be a commercial success. But as the real cost of recording equipment has fallen to its lowest level ever — with a lot of music created purely inside the computer — this role may not be as important as it once was. The primary functions of a label may become promotion and quality control, selecting the best artists and tracks in a particular genre. And is there any justification for an on-line retailer to keep a third or more of the sale price for the simple act of automatically processing a transaction on a web server? One straightforward solution would be to base royalties for paid downloads on licensing rates, not those for sales of physical CDs. These can be much more generous to artists, but naturally we would expect this step to be resisted by the major labels with large numbers of musicians under contract. If you don't have a major label record deal (yet), why should you care? Because if the industry doesn't take the opportunity to correct the perception that it rips off artists, then it risks undermining public support for paid downloads. According to the music activist site Downhill Battle (www.downhillbattle.org), if you divide the number of tracks sold via iTunes by the number of iPods sold, it works out at about 21 songs per machine. Given that iPods can store thousands of tracks, it seems that the argument about paying for on-line music hasn't been won yet. Even if you believe that CDs will still be around for a few years, it seems few people doubt that some form of network audio is the future of the industry. Labels may have to become leaner operations to survive; perhaps the multi-million advances, the marketing largesse and subsidised hotel-wrecking benders were just quirks of the way music was distributed and sold in the 20th century. It's hard to imagine that a young Led Zeppelin today would be able to fund a private jet from a small cut on 99 cent downloads. Warp Records' 'Bleep' site (www.warprecords.com/bleep/) provides an example of how legal downloads should be done. Firstly, the label is also the retailer, so there's more to pay the artist with from the start. The download files are high bitrate MP3s averaging over 200kbps, without annoying DRM features — the format that people actually want. Warp are also considering offering newer formats like Ogg Vorbis and the lossless FLAC; since they're not a hardware manufacturer they don't need to tie their customers to any particular technology. But best of all, they split the income from the downloads 50/50 with the artist, after bandwidth and running costs have been covered. Let's hope that Warp can get the support they need from the listening public. Published in SOS September 2004






Studio SOS

In this article:

Location, Location, Location... Acoustic Treatment Monitoring Adjustments David's Comments Success?

Studio SOS TV Composer David Lowe Published in SOS September 2004 Print article : Close window

People : Studio SOS

This month, we help television composer Dave Lowe transform a cavernous-sounding spare bedroom into a usable home studio. Paul White

David Lowe is one of today's more successful writers of TV music. He is probably best known for the current BBC news theme, though he has also produced some excellent hard-hitting drama themes, such as Mersey Beat and, of course, the pop single 'Would You?' under the name Touch & Go. You may also recall that some months back he agreed to let me co-write the Royal Gardeners theme with him, so when he asked for help with his studio I could hardly turn him down! He needed help to turn a bedroom in his Spanish villa into a studio that could be used not only for writing but also for tracking and occasionally mixing. Although his main studio is near Malvern, David already has a duplicate setup in his London apartment and he wanted to do the same in Spain, so that he could continue working on his TV and album projects when away from home. Like most villas in the area, David's has hard tile floors, solid rendered walls, and (in the case of his studio room) flat terra cotta tiles on the sloping ceiling, below the main roof tiles. This gave it a sound rather like an '80s live drum room: very reverberant, and far too resonant to mix in.

file:///H|/SOS%2004-09/Studio%20SOS.htm (1 of 7)9/26/2005 12:07:11 AM

Studio SOS

The upstairs room he'd chosen had patio doors looking out onto a veranda, but also had a twometre opening looking out onto an adjacent room, which he felt might be useful for recording vocals. The studio room itself turned out to be fairly small, measuring around 3 x 2.5m, but the shape was also somewhat awkward, because of the large opening into the next room. To complicate things further, the adjoining room contained a door leading to a stairwell, and if this was left open the stairwell reverberation made its presence known in no uncertain terms!

Location, Location, Location...

The first step towards reducing the excess reverberation in Dave's room was to install several Auralex acoustic foam panels and corner traps using contact adhesive spray.

Finding space for the equipment was not a problem, as David wanted to keep this studio simple. The equipment comprised only a dualprocessor Mac G4 running Emagic Logic Pro, a Logic Control with two expanders, and a MOTU 2408 interface. An Emagic AMT8 was in place to handle the MIDI (though we had to make do with an MT4 and just the main Logic Control unit, as David had inadvertently left the AMT8's PSU at home), and an ageing Korg M1 was set up as a master keyboard. This didn't have the audio outs connected, because David intended to use only software instruments. As it turned out, we were just able to slide the M1 on its 'X'-frame stand under the rear of the desk so that only the keys protruded — a useful tip if your 'X'-frame stand won't go low enough for this is to dispense with the securing crossbar and instead use a strap or length of strong nylon cord to hold the top two supports the correct distance apart and therefore provide the necessary height. A Mackie 1202 mixer was used as a front end for recording, and the sole studio mic was a Neumann U87, mounted in a shockmount with a pop shield. Monitoring was courtesy of a pair of Dynaudio BM15s fed from a Samson 550 Servo power amplifier, where the monitors were set up on metre-tall stands and connected with adequately heavy speaker cable. Not knowing what supplies were available in the area, I studied photographs of the studio in advance of the trip and then suggested some materials for David to order and have shipped over. Happily these all arrived before I did! I planned to use simple Auralex foam to treat the walls and corners, augmented by a pair of Real Traps Mini Traps suspended from the ceiling to file:///H|/SOS%2004-09/Studio%20SOS.htm (2 of 7)9/26/2005 12:07:11 AM

A large area of foam treatment was created by sticking multiple foam panels to MDF, and this

Studio SOS

could then be hung on the wall deal with reflections from that area and also to like a picture. provide additional low-frequency trapping. I felt that, as far as bass trapping was concerned, the large window area in the room and the one adjoining it would help us, as would the wooden cupboards in the next room. Bass tends to pass through windows rather than getting reflected, so the more window area you have, the more lowfrequency energy is lost to the outside world. Cupboards with panelled wooden doors also help damp down bass modes, especially if they contain towels and bedding.

When I arrived, David had already set up his gear on a table in what he felt was the most appropriate arrangement. This had him looking into an alcove, with the dividing wall between the two rooms on his left. The patio door to the outside was on his right, and the wide opening to the next room was behind him and to his left, so there was no semblance of acoustic symmetry. Before looking at the acoustic problems, I suggested to David that he face his system the opposite way, so that he would be sitting in the only part of the room where he could have a wall on either side of him. This would provide a degree of symmetry, if only in the listening position, and it would avoid obstructing the patio doors. As a bonus, it would also afford David a better view of the outside world during those valued moments of reflection! We moved the table to see how this would look and David agreed it felt better, so we decided to continue work on this basis.

Acoustic Treatment The Auralex corner traps went in the two corners behind the mixing position, though these couldn't be exactly symmetrical, given that the roof was higher at one side of the studio than at the other. Also, the lower of the two corners housed some electrical fittings and switches that needed to be left exposed. The traps were glued directly to the painted wall using the included spray contact adhesive — you have to position the traps very carefully when using that stuff, though, as once the glue grabs there's little chance of it letting go again! To avoid doing anything too permanent to the After a trip to the Spanish equivalent of B&Q to purchase rest of the walls, the remaining four-inch-thick the necessary fixings and tools, Auralex panels were glued to MDF sheets and the Real Traps Mini Traps panel then hung on the wall like pictures, mostly using absorbers were rigged to the existing screw holes and wall plugs that had ceiling of the studio room using been fitted by the previous occupant to fix hooks and fine chain. hooks. Strong nylon cord was passed between holes drilled into the MDF before the foam was glued in place. Our trip to the local equivalent of B&Q turned out to be interesting, as we not only had to buy all


Studio SOS

the necessary materials, but also basic tools, such as a wood saw and a hacksaw — however, this was achieved fairly painlessly with the aid of much pointing and drawing of diagrams! Because the room comprised entirely reflective surfaces, I decided on a vertical panel at either side of the listening position with a further sheet suspended horizontally behind the chair at head height. One of the vertical panels was suspended from a curtain rail, as there was a narrow window in the wall to the left of our new mixing position. As the wall behind the monitors was flat and completely bare, I used the remaining three panels of foam to treat that, again fixed to MDF sheet and hung on hooks. The Real Traps Mini Traps work on a different principle, as their performance improves if they are positioned away from a surface rather than directly on it, but in this room it wasn't practical to place them in their optimum position for bass trapping, which is across a corner or wall/ceiling junction at around 45 degrees. Instead I suspended them from chains and hung them from hooks screwed into the ceiling joists, so that they would hang parallel to the floor above the listening position and the desk. This way they would intercept any ceiling reflections that tried to reach the listening position and would also provide additional low-frequency trapping, albeit not as effectively as if they'd been hung across corners. I was surprised at how hard it was to cut what was in effect plug chain, and in the end I had to use a hacksaw, as none of our pliers or wire cutters would touch it! We'd also ordered a pair of Auralex Max Walls, which are foam panels mounted on included stands to provide movable treatment. We placed these in front of the large window in the adjacent room, as David hadn't yet had the blinds made for this room, but the real reason for buying them was to create a localised area that could be used for recording vocals and instruments. David had felt that the adjoining room would make a suitable playing area, but because of the open-plan nature of the two rooms and the amount of noise kicked out by the G4, I suggested that the next bedroom along would be better for this purpose, as it had a double bed in it (which made it sound less lively) and there was space under the door for a mic cable.

Monitoring Adjustments Once the acoustic treatment was up and the equipment put in place, we found ourselves short of UK mains sockets. David had one spare four-way plugboard with a very short cable terminating in a European plug, so I attacked a UK IEC file:///H|/SOS%2004-09/Studio%20SOS.htm (4 of 7)9/26/2005 12:07:11 AM

Studio SOS

lead and used it to replace the existing short cable. A European plug adaptor was then used so that we could plug it in. Shortly after this, David said he was getting a tingle when he touched his Logic Control, which revealed an interesting fact about European mains connectors. Although these plugs look like two-pin devices (on account of them having two pins!), they also have a recessed earth connector on one edge that contacts a matching There was quite a lot of noise coming from connector in the wall socket. However, Dave's Apple G4 computer, so it was moved further away from the mixing position. you can insert these plugs either way Although this helped, a further improvement up, as the socket has grounding was achieved by draping a blanket over the contacts on both sides. In this case, top and sides of the computer, leaving the one of the contacts had lost its ventilation unhindered, with a piece of springiness and so wasn't connecting acoustic foam placed fairly close behind the machine to soak up a bit more sound. to the plug. By inverting the plug, the earth was restored and the tingling stopped, but replacing the faulty wall socket would be the only safe long-term option. Before doing any listening tests, we had to position the monitors correctly, and with the stands David was using the speakers were around 30cm too high. The answer was to use some foam offcuts as wedges to prop up the rear of the each speaker so as to angle it down and aim the tweeter correctly. Non-slip matting, from our friendly Spanish hardware shop, was used to stop the speakers sliding off their stands. The BM15s have different left-hand and right-hand speakers, so I set them up with their tweeters outermost to give the widest stereo image. The foam plugs fitted to the bass ports (these come with the speakers) were left in place, as it turned out that there was absolutely no shortage of low end.

David's Comments "I must admit to being a bit sceptical about what could be achieved in that room without spending loads of time and money filling in ceilings, knocking down walls, putting in new doors, and so forth. But it had all the makings of a fantastic room to work in, if we could get rid of the huge cave-like reverberations! Paul has worked wonders, achieving far more than I imagined could be done, and all with only a few simple but effective materials, all easily available at comparatively modest cost. When I sat in and listened to some of my tracks in the finished room, it was almost like having headphones on! An added bonus was Paul's idea to move the setup around the other way, something I'd never have thought of myself, but which was a tremendous improvement. I also loved the tip about lowering the keyboard stand — definitely worth the price of a Spanish packet of Jaffa Cakes!"

Success? file:///H|/SOS%2004-09/Studio%20SOS.htm (5 of 7)9/26/2005 12:07:11 AM

Studio SOS

Checking the audio playback confirmed that the original ringy, splashy space had indeed been tamed to a surprising degree. The bass end was adequately even and there was now much better stereo imaging. The flutter echoes were gone, and talking in the room felt more like talking in a studio rather than in a deserted warehouse. I wouldn't go so far as to say we now had a room you could mix and master in with complete confidence, but at least it was fine for tracking and perfectly adequate for mixing provided that you'd have a chance to master the material elsewhere. David agreed that the improvement was vast, and he also felt that the room had a better working ambience, with a more intimate vibe to it. A further acoustic improvement was realised when David hung some heavy made-to-measure curtains that his wife Helen had sent over for him. These went over the patio doors and the opening between the two rooms, but they improved the sound even when open. We tried to source some rugs for the floor, but rug shops (and rugs) seem very thin on the ground in Spain — although the room sounded perfectly usable without rugs, adding them to the studio and adjoining room would doubtless help dry up the sound a little more. I also suggested putting slatted wooden blinds over the very wide window in the adjoining room in order to scatter sound and let in light at the same time. Paul White, Paul White...He's a secret Jaffa Cake eater...

Of course all this new-found peace and tranquillity was disturbed by the racket of the G4 whirring away on the floor, so I decided to try moving it into the adjacent room where the dividing wall and new curtains would offer a degree of screening. A USB hub was pressed into service to extend the keyboard/mouse connections, and the monitor cable was just long enough. A simple sound-deadening trick I discovered when I had a G4 was to place a folded blanket or rug over the computer to form a tunnel, leaving the front and rear of the machine exposed to the air. The blanket needs to go all the way to the floor to be effective, but it is essential not to block the ventilation path to the front and rear of the machine. We tried this, adding the spare Auralex bass trap behind the G4 to try to soak up a little more sound. The drop in noise was significant, mainly due to the folded rug and the new computer position. Although still not completely abolished, the computer noise was now barely noticeable from the engineering position. Our last day was spent doing a spot of recording to check out the system, which happily confirmed everything was working fine and that there were no groundloop hums or other nasties. Even without rugs and blinds, the studio sounded perfectly workable, though it was important to keep the door to the stairwell closed. The only down side was not being able to have Hugh along on the trip, file:///H|/SOS%2004-09/Studio%20SOS.htm (6 of 7)9/26/2005 12:07:11 AM

Studio SOS

but out of a sense of tradition and loyalty I made sure to eat his share of the Jaffa Cakes! Thanks to Audio Agency and Sonic Distribution for arranging prompt shipment of the acoustic treatment products used in this article. Published in SOS September 2004 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



CLASSIC TRACKS: 'Anarchy In The UK'

In this article:


A Snotty Bunch Artist: The Sex Pistols; Producer: Either/Or Dustbins Kicked Downstairs Price Adulterated Sandwiches Published in SOS September 2004 Why You Need A Mix Print article : Close window Room... Technique : Recording/Mixing Perfectly In Time

Chris Thomas; Engineer: Bill

When punk rock broke in 1976, the Sex Pistols caused panic in establishment Britain — and more than a few raised eyebrows in Wessex Studios, where Chris Thomas and Bill Price recorded the band's milestone debut album. Richard Buskin

While the Sex Pistols made their mark on mid-'70s British culture with a torrent of spit, rage, abuse and up-yours antiheroics, their restorative impact on a music scene dominated by disco, glam, progressive and corporate rock was also undeniable. Over the course of little more than a year, under the guidance of manager Malcolm McLaren, by way of several record labels, and courtesy of topically controversial lyrics, Johnny Rotten's maniacal, sarcastically confrontational delivery, and the band's blisteringly abrasive performances on tape and stage, the Establishment was threatened, the music scene was reshaped and the three-minute single was re-established as rock's primary form of mass communication. After getting together in 1975, the quartet of Rotten, drummer Paul Cook, bass player Glen Matlock and guitarist Steve Jones quickly attracted a large and virulent fanbase on the live circuit, and by May of the following year they were recording demos produced by guitarist Chris Spedding. Two months later Dave Goodman produced some more demos, one of which, 'I Wanna Be Me', quickly ended up as the 'B' side of the Pistols' first and only EMI single, 'Anarchy In The UK'. Produced at Wessex Studios in October of that year by Chris Thomas and engineered by Bill Price, 'Anarchy', with its simple structure and defiant, in-yourface attitude, basically defined punk, and it caused an immediate furore following file:///H|/SOS%2004-09/CLASSIC%20TRACKS%20%20%27Anarchy%20In%20The%20UK%27.htm (1 of 9)9/26/2005 12:07:31 AM


its release on Friday, November 26, 1976. This, in addition to the firestorm generated by the group's expletive-filled live television interview with Bill Grundy seven days later, as well as reports of similar antics at Heathrow Airport en route to a tour of Holland, prompted EMI to pull the plug on January 6, 1977. A&M followed suit after a stormy seven-day relationship, and when CBS pulled out of negotiations it was Virgin who finally stepped in and released the next single, 'God Save The Queen'. By then, Sid Vicious had taken Glen Matlock's place in the line-up and the band had returned to Wessex to record their seminal album, Never Mind The Bollocks Here's The Sex Pistols.

A Snotty Bunch Primarily an engineer since beginning his apprenticeship at Decca Records in 1962, Bill Price had worked with Eric Clapton, John Mayall, the Moody Blues and Tom Jones before moving to AIR Studios in 1970, where he helped with the design and was involved with projects by Mott The Hoople, Pink Floyd, Stan Getz, Stevie Wonder and Paul McCartney. When Chrysalis Records purchased both AIR and Wessex in 1976, Price joined the latter where, as at AIR, he was appointed chief engineer. At the same time he was also made studio manager and befriended Chris Thomas, for whom he engineered productions of records by Elton John, Pete Townshend and the Pretenders, while Price himself would go on to produce anyone from the Clash and Jesus & Mary Chain to the Waterboys and Big Audio Dynamite. Having just produced an album by Ian Hunter, as well as another by a soft-rock outfit named Racing Cars who enjoyed chart success with the single 'They Shoot Horses, Don't They?', Bill Price experienced a little bit of culture shock when first encountering the Sex Pistols. "They were a bit of a snotty bunch to start with," is his diplomatic understatement. "They treated the studio like the BBC Home Service and looked at me as if I was wearing a white lab coat that had a pocket stuffed full of Biros. It was as if they had walked into the arms of the Establishment, and there was also a general reluctance to comply, although that didn't last long. Johnny was the main protagonist, acting like he was consorting with the enemy, so I had a little chat with him, telling him the

Photo: Barry Marsden Bill Price, who shared coproduction duties on the Sex

file:///H|/SOS%2004-09/CLASSIC%20TRACKS%20%20%27Anarchy%20In%20The%20UK%27.htm (2 of 9)9/26/2005 12:07:31 AM


Pistols' album with Chris artists I'd worked with, what the job was, and that Thomas. we were there to help, and after that the attitude completely changed. It was only an act, treating both Chris and I as if we were Bill Grundy, and it wasn't going to get them in the newspapers. All it was going to do was make the record worse, so they quickly became very pleasant, easy people to work with."

Having heard the demoed material, Price and Thomas were impressed by its raw energy as well as its commercial appeal. Melodic, no; catchy, yes. "By then there was already a lot of music biz buzz about the band," Price says, "and if you played any of the demos you would recognise they were by the Sex Pistols, not least because Johnny's voice is so distinctive. They might have had slightly different tempos and slightly different arrangements, but there was nothing too drastic... There weren't harps or violas! The energy was largely the same. I mean, it's pretty hard to make Steve Jones sound tame."

Either/Or Bill Price and Chris Thomas collaborated on numerous projects, but only the Never Mind The Bollocks album bears one of rock's most intriguing credits: 'Produced by Chris Thomas or Bill Price'. "That was something Chris and I agreed upon," Price explains. "When, following the success of 'Anarchy In The UK', Malcolm McLaren announced that he wanted the band to record an album, Chris himself wasn't available and so Malcolm hired me as the producer. Then, during the course of doing the album, Malcolm hired Chris to produce the singles 'Pretty Vacant' and 'God Save the Queen', and during the course of that we tried recording songs that ultimately ended up being album tracks, while during the course of me recording the album Malcolm heard songs that he wanted Chris to rework into singles. So, when it came to putting the album together, we had duplicate versions of some of the material produced either by Chris or myself, and I went to Malcolm at least three times with different running orders that utilised different versions. "Chris and I quickly came to the conclusion that Malcolm was playing silly buggers with the pair of us, trying to get us to agree to a total production on the album that didn't even amount to 100 percent. It was as if he was trying to convince me that all of the tracks had been produced by Chris while trying to convince Chris that everything had been produced by me. So, seeing as the sleeves were mastered long before the final mastered version, Chris and I agreed on that fairly unique credit. We never really got to the bottom of what Malcolm's ploy was, but we suspected that it was to avoid paying people, so we agreed to accept 100 percent of the payment for the job and take the responsibility for divvying it up ourselves. "All of the Sex Pistols' recordings were financed, for want of a better word — or not financed, to be more accurate — by two of Malcolm McLaren's companies, Matrixbest and Glitterbest. This meant the whole setup was completely independent of whoever the band was signed to. In fact, by the time we were about halfway through the album, it was financed by Malcolm with used notes in brown envelopes, and that was at Wessex Studios' insistence. We weren't going to record another song without him paying cash for the past week. Fortunately, he was quite happy to go along with that.



"Malcolm was often in the studio with his missus, wearing her clothes. He used to wear a pair of tartan trews and a sort of tartan nappy that went between his legs and hung from his shoulders. Very fetching. Still, at that time he didn't really interfere in the production process. That would only happen later on with lesser bands. And besides, Steve Jones would have smacked him in the mouth if he had tried to butt in. Malcolm was quite clever about not straying into territory where he couldn't command the band's respect. If he was talking about publicity or something of that nature they would listen to him, whereas if he'd talked too much about what songs he liked or how they should be played I don't they would have worn it."

Dustbins Kicked Downstairs When Chris Thomas attended a Pistols gig before the first 'Anarchy' session, he was so taken by the band's overall vibe that, once in the studio, he asked Bill Price to "make the drums sound like dustbins being kicked downstairs". Price's response was to employ a technique that, although he'd used it before, was certainly not the norm — rather than position the musicians in the main live area and try to isolate the drummer, he placed the drummer in the middle of the studio and attempted to isolate the rest of the band, thus surrounding Paul Cook with a much more reverberant sound. "In practice, the band members weren't totally isolated," Price explains. "Rather than placing a lot of screens around Paul Cook or putting him in a small booth, he had command of the whole studio while attempts were made to screen off the other instruments and stop them bleeding onto the drum mics. Various strategic ambience mics that suited the task were also placed around the room: a couple of old BBC ribbons were literally at floor level behind the drums to try to pick up the ambience of the bass drum and the bottom skins of the tom-toms, and a couple of Neumann KM84s were slung above the kit to pick up the ambience of the cymbals. "For the kit itself I had a standard setup: an AKG D12 on the bass drum, a Neumann KM86 on top of the snare, a Shure SM57 underneath the snare, an AKG 451 on the hi-hat, and a Neumann U67 for the tom-toms — this was before we had the Sennheiser 421s. The main effect inspired by Chris's 'dustbins being kicked downstairs' comment involved me keying different ambience mics off the Photo: Ian Dickson / Redferns drums as they were being hit, using the The original Sex Pistols line-up, with Glen old-fashioned Kepexes. These were Matlock (left) on bass. the earliest American gates available, and using them was pretty much an integral part of the sound. Chris's suggestion that we could shorten the ambience with gates, providing more without it sounding too distant, all made total sense to file:///H|/SOS%2004-09/CLASSIC%20TRACKS%20%20%27Anarchy%20In%20The%20UK%27.htm (4 of 9)9/26/2005 12:07:31 AM


me. "Paul Cook started off only just able to play drums, yet by the time we were halfway through the album he was really solid. It was definitely a good idea that he took up playing the drums, although it was also unfortunate that when we started recording the Sex Pistols he hadn't been doing this for very long. He'd only had a kit for a matter of months. He certainly hadn't been knocking around with other bands for several years. Still, within a couple of months he was 10 times the drummer he had been when we started the sessions. "On 'Anarchy in the UK' we're hearing him at the start of this whole process. He'd often have a problem keeping time when it came to a drum fill or changing from hi-hat to a cymbal; all of the things on which learner drummers tend to skip time. However, we managed to get around that. Chris Thomas would insist on getting pretty perfect rhythm tracks, so, particularly on those early sessions, there was a lot of multitrack editing of what was ultimately used as just the drum track — although the whole band would be playing for the vibe, the bulk of the day would be spent doing a lot of takes and these would be edited together purely for the drum track. Then the guitar, bass and vocals would be overdubbed onto the resulting patchwork." Price recalls that, in line with the rest of the band's equipment, Cook's drum kit was "pretty crappy and beaten-up. Steve Jones had a Fender Twin that had lost the benefit of its front grille, meaning that you could see the loudspeakers. People would say that made it sound better, but when I asked Steve what had happened to the grille he said 'Oh, I had to get rid of that. The name of the band I nicked it off was written on it.' And that was probably quite true!"

Adulterated Sandwiches Wessex Studios had once been part of a Victorian church hall, and at the time, was fitted with a 32-input Cadac console, 24-track 3M M79 tape machines and 15-inch dual-concentric Tannoy Red monitors. Looking out from the control room, the drums were placed in the centre of the live area, about two-thirds of the way back, while Jones's guitar amp was directly in front of the control-room glass and recorded with a KM84 and an SM57 placed about six inches in front of the speakers. "What one had to do was balance those mics equally, grab a pair of headphones out in the studio that were turned up nice and loud, and fractionally position one of the mics so that they were perfectly in phase at high frequencies," Price says. "Because if you had one mic five inches away and the other six inches away you'd obviously get really bad phase shift that would take the top off the guitar sound. So, the best way would be to get the guitarist to play, clamp a pair of headphones to your ears as loud as possible, and gently swivel one of the mics around until you got a perfect phase correlation between the two. "Unlike Paul Cook, Steve Jones came fully formed. I don't know how, but he was like a veteran from the first minute — so solid, so rhythmic and so tight. And Glen file:///H|/SOS%2004-09/CLASSIC%20TRACKS%20%20%27Anarchy%20In%20The%20UK%27.htm (5 of 9)9/26/2005 12:07:31 AM


Matlock was a perfectly adequate bass player, more than adequate for what was required in his gig with the Sex Pistols. I couldn't really fault him. He was very quiet, but he and Steve didn't appear to get on particularly well, and that probably contributed to his early departure. In fact, Steve had a particularly personal way of adulterating Glen's sandwiches... with added special sauce." For the 'Anarchy' sessions Matlock's bass amp was about halfway back on the right-hand side of the studio and miked with a Neumann U87 pointing at the speakers. Johnny Rotten, meanwhile, performed rough vocals into a Shure SM58 and stood facing the band with his back to the open doors of a booth that was in the near-right corner.

The layout of Wessex Studios for the recording of 'Anarchy In The UK'. The drums were placed near the centre of the 15 x 10 metre live room, with two pairs of ambient mics used to capture the room sound.

"Johnny had a great attitude on stage, and it projected so well that it didn't really call for much movement," Price remarks. "It was more about posing and conveying an attitude by way of his body, his microphone and his eyes, and he did adopt that in the studio, but he was never one for jumping or dancing around, and he was therefore easy to record. Nevertheless, although we had a beautiful pair of tube U47s at Wessex, when I put one in front of Johnny for his overdubs it sounded awful and died after about 30 seconds of being gobbed at due to him using his middle register as loud as he could. So, immediately realising I wasn't going to get him over the top of all these thrashing guitar parts with a beautiful tube mic that overloaded and didn't sound particularly pleasant when it came out at the other end, I put him back on the SM58 and thereafter always used that. It tended to bring out a lot more in the voice, and even if it remained on the stand it was quite OK when he wanted to grab it. "Within any one session Johnny would do no more than three to six passes and I would normally comp these. We certainly didn't do long vocal sessions, and if, after half a dozen passes, the results still weren't convincing, we'd probably return to that song another day."



Why You Need A Mix Room... Whereas 'Anarchy In The UK' was recorded and mixed over the course of about three days in the main studio at Wessex, some of the other mixes took place in the facility's Cadac-equipped mix room. One such example was 'God Save The Queen', which was actually remixed on March 9, 1977, the same day that the Sex Pistols commenced their very short-term relationship with A&M. "The ladies at EMI's pressing factory had refused to put the single into the brown paper bags that had a photo of the Queen with safety pins slapped on them," Bill Price recalls. "Then EMI decided to terminate the band's contract and stop production of the 'Anarchy' single before A&M demanded a different mix of 'God Save The Queen' for their release. "On the day the band signed with A&M — as depicted in cartoon form in The Great Rock & Roll Swindle — Steve Jones shagged the boss's secretary in his private bathroom and ripped the sink off the wall, and he and the other guys all got pissed. Unfortunately, they'd hired one of those Bentley State Limousines that the Queen normally uses and they had the misfortune of arriving at Wessex Studios just as the primary-school kids next door were coming out. Its playground was separated from the Wessex driveway by a 15-foot-high chain-link fence, and as soon as the kids spotted this car they all plastered themselves to the fence and saw the Sex Pistols pour out with literally a bottle of vodka in each pocket. They'd completely demolished the drinks cabinet in the A&M managing director's office, and the headmistress of the school was a little bit perturbed by all this, so she started screaming at the kids to climb down off the fence and go inside, and Johnny whipped around and hurled obscenities at her. You know, 'F**k off you fascist bitch!' and other words to that effect. "Luckily, we were working in the mix room situated at the back of the building, and since Studio One wasn't booked we just whisked the band off and put them in the mix room, where they promptly fell asleep all over the couches. Then, about two minutes later, the SPG turned up in a white transit van with black metal grilles on the windows, and about half a dozen burly coppers jumped out and said they'd had a complaint from the primary school about the Sex Pistols. I therefore took them into the control room of the main studio, which didn't have any Sex Pistols in it, and I have to say the police were actually very pleasant. One of them said 'Ooh, what's that?' and I said 'That's the mixing desk! It's got a lot of knobs, hasn't it? This one does this and that one does that...' 'I used to play guitar,' the cop said. 'It was a Fender Broadcaster.' 'Oh, that's a very good guitar,' I replied. This must have gone on for about 20 minutes before he said 'Ah well, we'd better be moving on, sir,' and I was able to continue my mixing session. And, lo and behold, the Pistols were still all fast asleep."

Perfectly In Time Used to collaborating on records that required plenty of carefully crafted overdubbing, Bill Price and Chris Thomas in this case had to contend with Steve Jones who, according to the producer/engineer, was involved in 99.9 percent of the overdubbing. "If you came up with a guitar part that you thought you'd like to hear on the record, it would be pretty hard to get Steve to play it. He was capable of playing something that was completely different but still perfectly in time with file:///H|/SOS%2004-09/CLASSIC%20TRACKS%20%20%27Anarchy%20In%20The%20UK%27.htm (7 of 9)9/26/2005 12:07:31 AM


everything else that he played. So, the technique that Chris used was to try to get Steve to play specific riffs or rhythmic variations of his riffs with an eye to using them for particular sections of the song, and without necessarily bothering Steve too much about where those sections were going to be. "We did quite a lot of guitar overdubs with Steve, several of which were intended to be brought to the fore at one particular point in the song before being completely switched out or just taken to the bottom of the heap, so to speak. And this was made very easy by Steve, because everything he did was almost like it was on MIDI — it was always so perfectly in time, you Photo: Jorgen Angel / Redferns could do what you liked with it. For After the release of 'Anarchy In The UK', example, if you said something like Glen Matlock was "theoretically" replaced by 'Oh, just play that on the bottom Sid Vicious (left) on bass, most of the tracks strings,' or 'Give us the same sort of riff in fact being completed by Steve Jones (right). but in double time,' Steve might play the whole song doing just that and we could then introduce it at a particular point to drive the number on. He was very good, and he was also quite good at playing the sort of root note eight-in-a-bar Sex Pistols bass parts in time with the guitar rhythm once Sid Vicious had joined. "Since Chris Thomas was originally contracted by Malcolm to produce a single, I don't think there was any talk of doing an album until 'Anarchy' had been released and the band's potential became more clear. However, at that point Sid was theoretically on bass, so he and Steve both played on some of the tracks and then it was a case of 'may the best man win'..." Indeed, for the album sessions the initial backing tracks usually comprised just Paul Cook and Steve Jones, with the latter acting as the metronome. "We tended to work on one song at a time rather than conform to the habit of recording all the drum tracks and then all the guitars and then all the vocals before mixing the album," Bill Price recalls. "I never liked that sausagemachine approach and neither did Chris, and it also wasn't terribly practical because, if you got to the week that had been set aside for doing all the guitars and the guitarist vanished or broke a finger, you'd be stuck. And if you ended up with an album's worth of finished backing tracks that didn't have any vocals,

Photo: Ray Stevenson / Retna UK The Sex Pistols' demos were recorded a few months before the album, also at Wessex. At the Cadac desk, from left, are Wessex engineer Tim Friese-Greene, Dave



Goodman, Johnny Rotten and Kim Chraves. everybody would be just sitting there, looking at the singer, and that could be a problem if he wasn't in good voice or good humour. So, normally we'd try to finish 90 percent of a song before moving on to something else."

Having already established the Sex Pistols' sound during recording, Price didn't tend to leave very much to the mix. Nevertheless, since these were the days before automation, there still had to be a lot of movement on the guitar and vocal levels throughout each song. "It was quite a well charted and rehearsed manual mix," he confirms. As for the mastering of 'Anarchy In The UK' and the band's other Never Mind The Bollocks releases, Bill Price was heavily involved in the sessions that took place under the auspices of top engineer Malcolm Davies at PRT, ensuring that the band's raw energy was preserved at the highest possible volume. "It was a case of telling Malcolm 'It goes without saying that we want this as loud as you can make it!'" Price recalls. "After all, with vinyl mastering you could always make the needle jump, but that needle couldn't belong to the people trying to play the record. So, Malcolm was given the task of pushing things as far as they could go, and I think he did very well in that respect." Finally, I ask, is there anything else that Bill Price would like to add with regard to the Sex Pistols recording sessions? "Er, nothing that's printable," comes the laughing reply. Published in SOS September 2004 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



Cubase SX: Using MIDI Device Manager

In this article:

Not Just A Number Adding Multiple Presets Folders & Presets

Cubase SX: Using MIDI Device Manager Cubase Notes Published in SOS September 2004 Print article : Close window

Technique : Cubase Notes

Cubase SX's MIDI Device Manager provides a way of choosing patches and banks on your external MIDI hardware by name rather than number. This month we look at how to create your own MIDI Patch Name Scripts to make this possible. Mark Wherry

One of the smallest features that impressed me the most in the original Cubase SX was the neat pop-up library, with inbuilt search facility, which appears when you choose a patch for a MIDI device that has published its list of patch names to the application. Most VST Instruments publish their patch names by default, but for this to work with regular MIDI devices you need to have assigned MIDI Devices to specific MIDI ports available to the system, in the MIDI Device Manager. A MIDI Device is a virtual representation of a hardware device in Cubase and is based on a Patch Name Script that describes to Cubase the patches and banks used by the hardware device. MIDI Devices are configured in the MIDI Device Manager, and you can add one by clicking the Install Device button and selecting the appropriate device from the Add MIDI Device window and clicking OK. The device you've just added should be selected in the Installed Devices list. You can assign it to a MIDI output port by clicking the Output pop-up menu (just beneath the Import Setup button) and selecting the MIDI output port to which you have connected the actual hardware device. Unfortunately, Cubase SX doesn't come with a Patch Name Script for every piece of MIDI gear in existence, but fortunately this Cubase Notes column is here to help when you need to create your own MIDI Devices and Patch Name Scripts from scratch.

file:///H|/SOS%2004-09/Cubase%20SX%20%20Using%20MIDI%20Device%20Manager.htm (1 of 5)9/26/2005 12:07:35 AM


Not Just A Number To create a new MIDI Device: Click 'Install Device' in the MIDI Device Manager. Select 'Define New' in the Add MIDI Device window and click OK. Next, type a name for the MIDI Device into the rather appropriately named 'Type in MIDI Device Name' window, and click OK.

The MIDI Device Manager in all its glory. Note the hierarchy of the Patch Banks list in the bottom-left of the window, where the elements labelled Patches and Performance are examples of Patch Banks, Group 1 is a Folder, and the icons with small 'P' symbols indicate Presets.

The next step is to add some Patch Banks to the device, but the first thing to point out about Patch Banks is that they shouldn't be confused with the regular MIDI banks on your instrument — Cubase provides another organisational layer for these, as we'll see in a moment. If your MIDI device has several modes that you might refer to as Patch, Performance and Drums (for playing individual sounds or programs, combinations of sounds, and drum sounds respectively), these are the elements we'd describe as Patch Banks when creating a new Patch Name Script. However, if your MIDI Device doesn't feature different modes that would be categorised as Patch Banks, don't worry. Since a MIDI Device has to contain at least one Patch Bank, just use the Patch Bank Cubase creates by default for a new MIDI Device. In the spirit of the British sit-com Allo, Allo, listen very carefully: I shall say this only once. When you want to edit Patch Banks — for example, for renaming them or creating new ones — you need to make sure the Enable Edit button is active (ticked) just above the Patch Banks list. If it isn't active, simply click it. Some early versions of Cubase had a problem wherein Enable Edit isn't always active, even when it appears to be, so if you're ever in any doubt simply disable Enable Edit and reactivate it, and everything should be fine. When Enable Edit is truly active, the Commands pop-up menu should appear in a lighter-coloured text. To create an additional Patch Bank, click the Commands pop-up menu above the Patch Banks list and choose 'Create Bank' from the pop-up menu. To rename a Patch Bank, such as the 'Empty Bank' one that Cubase creates for you, simply select the Bank in the Patch Banks list at the bottom-right of the MIDI Device Manager window. Click the name part of the Patch Bank entry, type a new name into the text-edit field and press Return. You can delete a Patch Bank by selecting the appropriate Patch Bank and pressing the Backspace key.



When there's more than one Patch Bank in a MIDI Device, an extra button labelled 'Bank Assignment' appears in the centre strip, providing access to another page of controls that allows you to set which Patch Bank is used by each MIDI Channel in the device. It's necessary to assign Patch Banks to MIDI Channels because only the Folders and Presets for one Patch Bank can be displayed when selecting presets for a MIDI Track on the Project window. To change the assignment of Patch Banks to MIDI channels: Click the Bank Assignment button in the centre strip of the MIDI Device Manager. Choose the appropriate Patch Bank for each MIDI channel by selecting it from that channel's pop-up menu. Click the Patch Banks button to return to the Patch Bank page once you've finished. Note that this assignment of Patch Banks to MIDI channels can be changed at any time.

Adding Multiple Presets Although adding presets one at a time is perfectly acceptable, this process can quickly become tedious, considering that most modern MIDI devices often have many hundreds of Presets, many at a time. So, to add multiple Presets in one go: Select the Patch Bank or Folder within which you want to create a Preset in the Patch Banks list Choose 'Add Multiple Presets' from the Commands pop-up menu.

The Add Multiple Presets window does exactly what it says on the tin, allowing you to create multiple presets, automatically incrementing the appropriate parameters, in one pass.

The Add Multiple Presets window appears, to let you configure how many Presets should be created and which MIDI events each Preset should contain. Click 'OK' when the settings in the Add Multiple Presets window have been correctly entered, and the Presets will be created.

By default, the Add Multiple Presets window provides a Program Change event; however, you'll notice that, unlike the columns provided in the Preset edit view, there's no Value column for the Program Change messages. Instead, there's a Range column. If Range is set to 0-127, for example, Cubase will automatically create 128 Presets where the first program change Value is 0 and the subsequent Values are incremented by 1 — so that a Preset is created for every value from 0 to 127. The Default Name field at the bottom of the window provides a root name for the Presets: in the above example, the Presets we'd create would be named Preset 0, Preset 1, and so on. The Presets created would then select consecutive sounds on your MIDI device.



If you need to send a Bank Change event before the Program Change in a Preset (or any other event, for that matter), you can edit or delete the Program Change event in exactly the same way as in the Preset edit view, and you can also select additional events below the first event if required. For example, if you wanted to create 128 Program Change events for bank 1 on your MIDI device, you'd simply create a Bank Change event first and set the range to the value of the Bank Change message you want for all the Presets — in this case '1'. Then create a Program Change event after the Bank Change event, as described in the previous step.

Folders & Presets After creating the new MIDI Device's Patch Banks, it's time to fill these with Folders and Presets. In the context of Patch Name Scripts, a Preset represents a single patch, program, sound, or whatever you want to call it, on your MIDI device. A Folder is a way of organising Presets within Patch Banks. You could use Folders to represent the banks of sounds on your MIDI device, or even to group Presets in terms of their category — for example, Folders for pianos, strings, basses, and so on. To begin with, let's consider a simple device where there might only be a handful of Presets required and no need for Folders. If you expand the default Patch Bank in the Patch Banks list, you'll notice that Cubase creates a default Preset labelled 'Off', which is effectively a blank Preset. To add a Preset to a MIDI Device: Select the Patch Bank or Folder within which you want to create a Preset (in the Patch Banks list). Click the Commands pop-up menu just above the Patch Banks list. Select 'New Preset' from the pop-up menu. Alternatively, assuming a Patch Bank or Folder is indeed selected, right-clicking anywhere in the MIDI Device Manager will display the same pop-up menu.

The Bank Assignment page allows you to assign specific banks to each MIDI channel of your device — useful, as shown, where you want the drum bank to be used only on channel 10, for example.

To configure a Preset, select the Preset you want to configure; in this case, select the 'Off' Preset and the bottom-right area of the MIDI Device Manager will open into a view where you can edit Presets. The Preset editor view is made up of a series of columns, and the idea is that each Preset contains one or more MIDI events that are sent to your MIDI device when the Preset is selected in Cubase, so that the right sound or program is selected on the synth or other MIDI device.



To add an event to the Preset, simply click at the top of the MIDI Message Name column in the Preset editor view and select the required MIDI event from the popup menu that appears. The basic event that changes programs or sounds on a MIDI device is known as a program change, so you'll probably want to select Program Change from the pop-up menu. This just happens to be the first entry in the list. Give the Program Change event a suitable value by clicking the default value and entering a new value within the range indicated in the Valid Range column. The value will depend on the Preset you're creating: for example, if you were creating a Preset for the Grand Piano sound on a General MIDI sound module, you'd want a Program Change event with a value of 0 (as this is the value defined for the General MIDI Grand Piano). You should be able to find the relevant Program Change values required by the sounds on your MIDI device in the device's manual. Additional events (for changing banks — see 'Adding Multiple Presets' box) can be added by clicking in the MIDI Message Name column below the previously created Preset. You can delete the event currently highlighted by the light-blue box by pressing the Backspace key, and select a different event by clicking (or using the cursor keys) in any column in another event. Using the up and down cursor keys is often the safest way, since you can accidentally change the settings of an event you want to keep by clicking with the mouse. Rename a Preset as you would a Patch Bank, by selecting it in the Patch Banks list, clicking the name part of the Preset, typing a new name in the text-edit field and pressing the Return key. And finally, to delete a Preset, simply select the Preset you want to delete in the Patch Banks list and press the Backspace key. You can also select multiple Presets and delete them in one pass, just as you'd select multiple files on your computer's desktop. In other words, select a Preset as normal and Shift-click another Preset to select all the Presets in between, or Control/Apple-click the individual Presets you want to select. Published in SOS September 2004 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



Demo Doctor

In this article:

Demo Doctor

The Man Who Was Thursday Doctor's Advice: Upgrading Analysis of Readers Recordings Published in SOS September 2004 Thorn QUICKIES Print article : Close window How To Submit Your Demo

Technique : Recording/Mixing

Resident specialist John Harris offers his demo diagnosis and prescribes an appropriate remedy.

The Man Who Was Thursday Venue: Home Equipment: Pentium 4 PC running Cakewalk Sonar XL 2 DAW, AKG C414 and Shure SM57 microphones. Track 1 This CD features an eclectic mix of styles and is 1.4Mb packed with excellent sounds and interesting Track 2 arrangements. It kicks off with a dirty fusion of 1.4mb progressive rock and dance, and composer Matt Track 3 Constantine is also not afraid to show his classical 1.4Mb leanings. After an 'in your face' introduction of thrashy rock drums with distorted synth, he leads us into a filter-modulated synth loop based around a simple pattern that wouldn't be out of place in a piece by Bach or Beethoven! I liked the way noises and other effects were used in addition to, and sometimes instead of, drum fills to lead us from one break to another. Yet it's the dynamic drop about half way through the track which most clearly demonstrates Matt's arrangement skills. Here he uses a resonant synth sound to carry the melody line. Underneath this, there's a variety of drum breaks with different amounts of bandwidth filtering and some excellent resonant sound effects. The difficult change back to a reprise of the thrashy intro is well handled, with a distorted loop acting as the link.

When I first heard this track, I confess I was expecting a sung vocal to come in at any moment and when one didn't arrive, I thought that this was going to be an instrumental CD. Surprisingly, then, the second mix turns out to be a song with vocals from Matt himself. They are well performed but have a hint of sibilance, emphasised by the high-frequency reverb treatment and probably a touch of highfrequency EQ boost. I think a warmer sound would have suited his voice, so perhaps the bright AKG C414 is not the best choice of microphone. Some lower-

file:///H|/SOS%2004-09/Demo%20Doctor.htm (1 of 5)9/26/2005 12:07:38 AM

Demo Doctor

mid EQ boost on the vocal could also improve it. On the other hand, a thin vocal sound may have been a deliberate production decision. I would argue that Matt's extensive use of vocal effects also means that he didn't really need to go for a very stylised sound from the vocal itself. The stereo echo and vocoder on the word 'insane' are very effective but a little too loud — perhaps Matt was a bit too pleased with this particular effect! The light distortion and delay on the word 'energy' is also good, but the front end of the word seems to be missing, as if the auxiliary send wasn't turned up quite fast enough to send the whole word to the delay. Sometimes it's better to trigger these kinds of effected individual words from a sampler, or to put them on a separate track to make setting up the effects treatments easier. When it comes, the drop to the middle section in the second track and the use of a music-box-style loop, played on an electric piano with slight distortion, is a nice touch. In fact a lot of the sounds are very good, especially the natural-sounding kick and snare. One of the things that doesn't work in this track is the ascending string line which needs a bigger sound and a faster attack to really lift into the heavy riff. With such a vast array of samples available there must be a more suitable one. For the third track, Matt moves more towards the classical but in an off-beat manner suggestive of movie soundtracks. His use of cello, harpsichord, a synthesized bassoon sound and pizzicato strings are framed by synthesized string sounds, while cracked tubular bells provide the title theme. Film and theatre work is surely worth investigating when you're armed with such an arsenal of interesting sounds! www.themanwhowasthursday.co.uk

Doctor's Advice: Upgrading Both the main reviews this month focus on demos recorded using setups heavily reliant on software, and, for a change, neither demoist complained about CPU limitations! Most of the demos in the Quickies section were recorded on a PC or Mac too and the quality was generally pretty good. Perhaps domestic computerbased recording systems have finally become stable enough to convert even the most stalwart hardware user. Well, until you decide to upgrade, that is. Upgrading has cost me more sleepless nights and extra work in the last six months than I care to think about. I recall SOS contributor Craig Anderton's words on the subject — how I wish I had heeded them myself — and advise you to take note: "never upgrade before an important project"!


Demo Doctor

Thorn Venue: Home Equipment: PC running Fruity Loops Producer 4 virtual studio, Linplug Albino, Novation Bass Station and NI Reaktor Session soft synths and Steinberg Wavelab 4 editor, Alesis M1A MkII monitors. Track 1 Thorn's first track features some excellent percussion 1.4Mb programming. Kicking off with an interesting drum Track 2 loop filtered into a narrow frequency band and, 1.4mb surprisingly, a running water sample, the track's techno leanings aren't made clear until a suitably metallic-sounding hi-hat makes an appearance. This is enhanced by a very digital-sounding comb filter (possibly supplied by Reaktor) which really seems to push the hi-hat to the front of the mix. Sadly, the kick drum is a let-down. It's got quite a long decay and isn't mixed high enough to have any impact. A harder, shorter, more punchy kick would really drive this mix along. There's a general lack of aggression in the production on this track — possibly a sign of low confidence — and I would urge Thorn to really go for it with some alternative, more extreme versions of these tracks.

Thin, ethereal strings are a feature of Thorn's mixes and these leave plenty of room for the drum loops which are their main strength. The second mix demonstrates this well, but the choice of bass sounds is poor. Given the plethora of excellent virtual synths at Thorn's beck and call (like the Novation Bass Station plug-in) this is rather disappointing. I know from experience that Bass Station can supply plenty of attack, yet although the bass sound driving this mix along is fat, it's also a bit limp. If this is definitely the bass sound of choice, it could be improved by running a kick drum underneath to give the sound some attacking edge. First, make a copy of the bass synth MIDI track on the sequencer then edit the MIDI note numbers to the right trigger value for the kick — the bass line is very simple so this shouldn't take long. Next, mix the triggered drum low under the bass synth so that you add to the attack without necessarily hearing the drum itself. In some ways, Thorn's mix already attempts to do this by accenting some of the bass synth notes with a kick drum but it's not working because the drum and the synth are operating in the same frequency range and neither has enough attack or punch. With a little extra work on the low-frequency area, both in the choice of sounds and the arrangement, these mixes would be improved immensely — they're so nearly there now. Finally, Thorn shouldn't be afraid of simplicity: if a track needs a strong four-four kick drum, put it in as the anchor point for the more clever and complex loops to swing around.


Demo Doctor

QUICKIES

Roger Barker I was impressed by Roger's songwriting as well as his singing style, which has a blues flavour with some elements of Crowded House's Neil Finn thrown in. Sometimes, however, the arrangements sound a bit too fussy. For example, in the second song, he sets up a really nice groove in waltz time, but then kills the mood by dropping into a 4/4 funk groove. The change is cleverly done, especially when the track goes back into 3/4, but it actually sounds like two songs forced together. I think it could work if the 4/4 section was used as a middle eight and the 3/4 groove was left to run for longer — say, two verses and choruses — before the change. Having said that, the play-out is particularly beautiful, with a backing vocal on the off-beat working almost in question-and-answer style with the main vocal.

Tart This is a pretty thin-sounding demo and I feel for the guys in the band because they've put so much effort into the recording. Everything is fighting for space in the 2kHz region in a classic case of over-EQ'ing, and there's no bass to speak of either. People often make this kind of mistake in their use of EQ when they listen to an individual instrument in isolation for too long and spend too much time working without a break. Often the real objective — the best sound possible — gets confused with clarity and brightness and this results in over-EQ'ing individual instruments in the upper-mid area. The trouble is that, in comparison, the other instruments then start to sound dull so you have to EQ them too, usually in the same frequency region, until all the mix elements end up with a similar EQ and the entire mix is upper-mid heavy. The mixes on this CD could perhaps be rescued with corrective EQ, but they really need to be remixed from the ground up. EQ can be a dangerous thing so remember to use it carefully, keeping the sound of the mix as whole in mind!

Portal There's a definite country-rock slant to this demo, with some nice acoustic guitar melody lines recorded using a Rode NT1 microphone and a touch of compression. However, there's a tendency to make the strummed acoustic guitars so bright that they obscure the hi-hat. This may have been done using postproduction EQ to improve the perceived presence of the mixes and compensate for the rather low level of the cymbals. If so, then a full remix would have been a better option, lifting the cymbals and taking some of that sizzle back out of the guitars to give them a more natural presence in the mix. Given that this demo was a means of testing some recording equipment out I can almost forgive the complete lack of taste in the application of effects like chorus and pitch-change. Of all the effects, reverb was, for the most part, well handled, but I wouldn't want to give Portal the confidence to apply any more! The songs are OK too, but they should ditch the modulation on the vocals next time.


Demo Doctor

Melonfinger A glance at this demo's well thought-out CD sleeve tells you a lot about the band's image and genre, and if you check out their rather stylish web site you'll see that this lot are pretty well organised (for a rock band!). Though they do rock, they have their mellower moments too, as the acoustic guitar-led second song demonstrates. Technically, I liked the way Melonfinger integrate the Yamaha MIDI drumkit seamlessly into their rock/pop sound, using NSG's Drumkit From Hell samples. Melonfinger have a high-energy approach to songwriting, performance and production, which is handled with great skill, on the first and third tracks in particular. The energetic vocal delivery is best placed lower in the mix — on the third song, for example, the voice almost has to fight against the backing and this gives the mix itself more energy. Musically, this band are rather hard to define, with their '70s pop/rock sensibility tempered by more contemporary influences. According to the web site they play a lot around the Hertfordshire area with occasional forays into London, and they seem well worth checking out. www.melonfinger.com

How To Submit Your Demo Demos should be sent on CD or cassette to: Demo Doctor, Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Please enclose a covering letter with details of your recording setup and a band/artist photograph and/or demo artwork (which we may use here and on our web site to illustrate your demo review). Samples from the two main demos reviewed will be placed on our web site. Including contact information, such as a telephone number, web site URL or email adress, will enable anyone who is interested in your material to contact you. Published in SOS September 2004 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



Instant Track Creation

In this article:

Reason Can, Too... Happy Birthday, Propellerhead! Into The Malström Quick Tips String-Driven Thing

Instant Track Creation Reason Notes Published in SOS September 2004 Print article : Close window

Technique : Reason Notes

We explore an advanced technique for creating 'instant' tracks. Derek Johnson

As a reviewer of hardware and software for this magazine, I often examine all-in-one workstation synths that try to be attractive to any style of musician at almost any level. Hence, some preset data is often dedicated to producing the 'one-finger track' effect: press a key and everything — drums, bass line, arpeggiations, chopped chords, the lot — happens. Very impressive in a demo situation, and a bit like cheating... but if I'm honest, this sort of thing can be rather a lot of fun. It's even more fun — and less like cheating — if you analyse the presets The rack you'll need to try out the 'instant track' technique. and work out how to create something unique to you, in your style. One-finger patches can also be useful in live situations, offering an instant way to vamp through a break while the next song loads or the band changes instruments, and so on.

Reason Can, Too... I wondered how I'd create a layered patch like this in Reason, without any extra bits of software. It's not actually too difficult, keeping a few issues in mind. Creating a layer of devices that can be played from your master keyboard is straightforward, but done in the obvious way, via the software's four incoming file:///H|/SOS%2004-09/Instant%20Track%20Creation.htm (1 of 5)9/26/2005 12:07:43 AM


MIDI busses accessed via the Hardware Interface, it requires you to copy and paste sequencer tracks to create a layered performance in the sequencer. This is not quite what we want. Of course, V2.5's Spider CV splitter/merger devices allow one Matrix step-sequencer to play up to three synth or sampler devices, but that's not 'real-time' any more. A further issue is that most modern workstations feature some sort of arpeggiator to add instant sparkly interest to a one-finger performance, and Reason currently lacks one. However, this software is nothing if not flexible, and we're going to ignore or solve these problems. The problem of not being able to layer multiple devices in a logical way without external help is solved by using one device: the NNXT Advanced Sampler. Though this might seem like a restriction, given the correct raw sample material the synthesis facilities on offer are powerful enough to let you get very creative. The only problem with making this example for readers of this column to recreate is that I'm restricted to the raw material contained in the factory Refills, which means selecting single samples from within multisamples. (I have a collection of looped raw synth-waveform samples that I would normally use for my own purposes.) First of all, make sure your Reason rack has a Remix mixer and whatever effects you favour loaded into it. I always start with an RV7000 advanced reverb and a couple of DDL1 delays (for tempo-sync'd delay effects). Now add an NNXT. Load the first sample, by clicking on Here's what the connections between NNXT the 'Load Sample' button. Navigate the and Malström look like, with all other cables removed for clarity. Reason Factory Sound Bank Refill, through NN19 Sampler Patches to the Basses folder and then the AccBass sample folder; choose '011,C1,Med,onG string.WAV', if you'd like to recreate my layer exactly. You may have to 'Set Root Note From Pitch Detection' and 'Automap Zones', from the Edit menu. Now go to LFO1: enable Tempo Sync, select the square or sawtooth waveform, and set the rate parameter to 1/8 (eighth note). Turn the Level knob fully right. Play a note, and the decay of the bass sample is chopped up into eighth notes. That's one layer. For the next layer, find the Fretless Bass folder in the NN19 patch collection and load 'FretlessC1,med.WAV'. We'll leave this for the moment, except to set its pitch and map it. We might also want to just individually highlight each of our first samples and choose 'Group Selected Zones' from the Edit menu. Each sample thus becomes its own 'Group', with dedicated sets of synth parameters. Now we're going to do something a bit different... Load the 'VibraA1.WAV' sample (NN19 Vibraphone patch folder, Mallet and Ethnic collection). The main thing we'll do with this sample, which we'll also make its own Group, is add a repeating octave effect. Set LFO1 to Tempo Sync, and give it a square file:///H|/SOS%2004-09/Instant%20Track%20Creation.htm (2 of 5)9/26/2005 12:07:43 AM


waveform. I set the rate to quarter-note, but it's your choice. The Pitch amount knob should be tweaked to around 1200; I prefer a negative value (left), but a similar effect will be produced with a positive value (right).

Happy Birthday, Propellerhead! This month, the producers of our favourite software electronic studio celebrate their 10th birthday. Yes, it was 1994 that saw the introduction of Recycle, initially for the Mac only, under the Steinberg badge. The company's still going strong, and even Recycle is still developing — see the review of v2.1 in the last issue. Who knows what Reason will be like 10 years from now...

Into The Malström We're going to make a drastic change to the texture produced by this sample, so first set its audio output to 3-4 (use the last knob on the right under the blue display). Next, create a Malström device, patch NNXT outs 3 and 4 to the Malström's two audio ins and enable both filters and Mod B. For Mod B, Select 'sync', choose a rate of 1/4 or 1/8, and select a stepped waveform — curve 23 or 24. Malström is now behaving as a signal processor, but the modulator (essentially a sophisticated LFO) is adding an arpeggiator-like stepped effect to each filter's cutoff frequency. You'll have to tweak frequency and cutoff to get the audible effect that's best for you, but start with frequency quite low (to the left) and resonance in the middle, with LP12 the filter type of choice to start. You can also turn the Spread control fully right, to create a pseudo-stereo effect. We're not done with Malström yet. Remember the second sample we loaded — the fretless bass — that we left alone? Let's go back to it, by clicking on the box in the Group column to the left of the sample. Before we go any further, I'll just note that I panned this sample (Group) hard left and the first sample we loaded hard right, and routed NNXT output 1/L to one Remix channel and output 2/R to another, obviously both in mono. Back to the Malström. We're going to be using its Mod A section, routing the CV output at the back to NNXT's Filter Cutoff CV input (on the rear of its global controls panel). Due to NNXT's design, this is is routed to the cutoff frequency of all filters currently in use, but in this case it's not a problem. Set Mod A to tempo sync, choose a rate of 1/8 and select a stepped waveform. We want to really hear the effect of this LFO on our fretless bass, so tweak its filter parameters thus: Filter type LP24; cutoff frequency 33Hz; resonance 79 percent. Now we have a chugging bassline, an impressionistic burbling texture created by the fretless sample, and high-frequency movement from the 'arpeggiated' vibe sample.

file:///H|/SOS%2004-09/Instant%20Track%20Creation.htm (3 of 5)9/26/2005 12:07:43 AM


Quick Tips When editing an NNXT patch that has more than just a couple of samples in it, remember that each sample can be isolated by means of the 'Solo Sample' button. Isolating samples in this way is ideal for really focusing on fine-tuning individual samples. Routing the left and right outputs of a Reason sound-making device to their own Remix channels, panned hard left and hard right, seems to produce a wider stereo image. Try it! Mac users, if you're still unsure about the move to Mac OS X, wait no longer: latency — the delay between playing a note on a MIDI keyboard and the sound coming out of your monitor speakers — in the new OS can be astonishingly minimal. I run an oldish G4, and steadfastly refuse to dump my Digidesign Digi 001 audio hardware, yet without trying my system offers 2-5ms of latency within Reason. That delay is as good as inaudible in virtually all circumstances.

String-Driven Thing The last thing we'll add to the mix is a straightforward two-note string chord. Here I chose a couple of samples from the Strings folder (that's in the Strings folder — really — in the NN19 Sampler Patches folder). Go for 'Bigsc4.aif' and 'Bigsd#4. aif'. Highlight both and 'Group Selected Zones' from the Edit menu; this way, you can apply synth and output settings globally to both samples, though you'll still be able to tweak each sample separately by clicking just on its name. In my 'patch', I routed these two samples to their own audio output pair — 5-6 — and on to their own Remix channel. Individually, I panned the samples slightly left and slightly right. Also, I retuned the D# sample to a root note of G#2, causing it to play a fifth higher than the other sample, producing an instant chord. I also EQ'd their Remix channel a bit, taking almost all the top off by moving the treble knob to the left. Add effects to taste Setting Reason to start playback by pressing note C3 — middle C — on an attached MIDI keyboard.

And that's it — almost. Right now, the texture we originally created is completed by an ethereal two-note top line. Adding drums is straightforward. You could create a little count-in within Redrum and automate how it plays back, perhaps giving you a one-bar click followed by one bar of silence before looping the desired drum pattern, or you could do the following.



First, make sure that the Remote Control input in Advanced MIDI preferences is set to your MIDI keyboard. Under the Options menu, select Edit MIDI Remote Mapping, and click the green arrow that appears over the transport bar's start button. Enable 'Learn from MIDI' in the pop-up window and play the first note of your one-finger performance. Close the window and disable Edit MIDI Remote Mapping, but ensure that Enable MIDI Remote Mapping is ticked. Now when you play the first note of your sequence, Reason's drum machine, and whatever else happens to be playing in the sequence, will kick in and all the tempo-sync'd LFOs and delays will play along nicely There are many ways of extending this idea. In fact, many a similar patch has formed the basis of an entire track for me — I write and automate a Redrum track, play one-fingered for the desired length of time and then add some more material to develop the original idea and texture. Have fun! Published in SOS September 2004 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



Layering Sounds In Logic's Virtual Instruments

In this article:


Layering Virtual Instruments Logic Notes Managing Note Off Messages When Splitting Notes Published in SOS September 2004 A More Convenient Method Print article : Close window Logic Tips Technique : Logic Notes Have Your Say!

Current Versions Mac OS X: Logic Pro v6.4.2 Mac OS 9: Logic Pro v6.4.2 PC: Logic Audio Platinum v5.5.1

Layering hardware sound modules is a simple matter, but things aren't nearly as easy when you're working with virtual instruments in Logic. Here are a few useful workarounds. Len Sasso

There are a number of situations in which you might like to play several different virtual-instrument sounds from the same live MIDI input. While Logic doesn't make this common task particularly easy, it can still be done. The problem is that Logic keeps only one Audio Instrument object in what is called 'live' mode. When MIDI is routed to multiple Audio Instrument objects, all but the one currently in live mode will exhibit some latency. Before proceeding, three things should be noted. The methods described here are not necessary for playing back MIDI sequences from Arrange window tracks — Logic then keeps all required Audio Instrument objects live. Logic's transport must also be running in order to layer virtual instruments, no matter which of the methods described here is used. Finally, you don't need to use these methods to layer different sounds from a single multichannel virtual instrument, and you can even apply separate effects processing to the sounds if the virtual instrument also supports multiple outputs.

Layering Virtual Instruments

Two Audio Instrument tracks are record-enabled, and each one has a bypassed I/O plug-in inserted to ensure the virtual instrument remains permanently 'live'.

file:///H|/SOS%2004-09/Layering%20Sounds%20In%20Logic%27s%20Virtual%20Instruments.htm (1 of 5)9/26/2005 12:07:47 AM


Logic comes with an effects plug-in called I/O, which you'll find in the Helper submenu of the effects plug-ins menu. The intended purpose of the I/O plug-in is to route audio out of and back into Logic for processing by external, hardware effects. To do that you choose an output and input of your audio interface from the I/O control panel. Output from the Audio object in which the I/O plug-in is inserted is routed to the chosen output, then received from the chosen input for further routing in Logic's signal path. However, the useful by-product of the I/O plug-in for our current purposes is that it keeps the Audio object into which it is inserted in live mode. It is probably now obvious that the trick to keeping a virtual instrument in live mode is to insert an I/O plug-in into its Instrument Audio object. What may be less obvious is that you don't need to somehow connect the output and input specified for the I/O plug-in — you can simply put it into bypass mode by clicking its Bypass button. However, you do need to specify an actual output and input in the I/O plug-in's control panel. Once you have 'enlivened' the virtual instruments you wish to layer, there's still the issue of getting the same MIDI input stream to each of them. The simplest method is to create and record-enable an Arrange track for each of the Instrument Audio objects. Selecting any of those tracks will then play each of the virtual instruments inserted in those Audio Instrument objects (when, as mentioned, Logic's transport is running).

Managing Note Off Messages When Splitting Notes Environment constructions that separate notes by their Velocity, or by some complex condition that does not route all notes of the same pitch to the same place can leave hanging notes during live play-through, because MIDI Note Off messages can wind up going down a different path than their associated Note Ons. A simple and effective solution is to separate all Note Offs before the processing and send them simultaneously to all destinations. That uses a little extra MIDI bandwidth (which is cheap), but ensures that the matching Note Off always gets to the right place. To separate the Note Offs, use a Transformer set to Condition Splitter (True->Top Cable) mode. Set its Status Condition to '= Notes' and set its Vel condition to '= 0'. All Note Offs (notes with velocity value of zero) will now go to the top cable, while all Note Ons will go to the second cable. Since you need to route the Note Offs to several places, cable the top Transformer cable into a new Monitor object, and cable the Monitor's outputs to the desired destinations. Cable the Transformer's second cable to the input of the process.

A More Convenient Method Although simple, that method for recording MIDI to be played by the group of virtual instruments is not the most convenient. MIDI gets recorded on the file:///H|/SOS%2004-09/Layering%20Sounds%20In%20Logic%27s%20Virtual%20Instruments.htm (2 of 5)9/26/2005 12:07:47 AM


selected track and Aliases of the recorded sequence get placed on each of the other record-enabled tracks. There's nothing wrong with that, but it becomes awkward when you want to start moving the MIDI sequence around in the Arrange window. An alternate method is to create a neutral Environment object, such as a Monitor, and cable one of its outputs to each of the targeted Audio Instrument objects. Assigning that object to an Arrange track and selecting that track will then get the MIDI routed to each of the virtual instruments. When you record, a single MIDI sequence will be placed on the selected track, though it will still play all the virtual instruments. Note that you do not need or want to record-enable the Audio Instrument objects in that case. The issue of keeping multiple Audio Instrument objects in live mode also arises when you want to apply some form of Environment processing to MIDI destined for several virtual instrument plug-ins. You might, for example, wish to cable several outputs from a Channel Splitter to different Audio Instrument objects. That would allow keyboards or other controllers on different MIDI channels to simultaneously play different virtual instruments. Here MIDI Note Off events are split by the NOF To Top Transformer object and sent to all Audio Instrument objects. Note Ons are routed to an Alternating Split Transformer, which splits them between the two connected Audio Instrument Objects.

Other Environment objects that send different MIDI messages to their outputs include the Mapped Instrument, the Delay Line, the Chord Memoriser, and the Transformer in some of its modes. With the Mapped Instrument, you can assign both cable and MIDI channel for individual pitches, which allows you to use different drum-synth plug-ins for different drum sounds played from the same MIDI drum part. Individual repeats from the Delay Line are routed cyclically through all the used cables. For example, if you connect a cable to two Audio Instrument objects and set the number of repeats to two, the incoming (original) note will go to the first cable, the first repeat to the second cable, and the second repeat to the first cable. You can generate all kinds of interesting echo effects using different cablings and numbers of repeats. The Chord Memoriser, when its Cable Split option is turned on, sends successively higher chord notes to lower cable outlets. If there are more chord notes than cables, the extras all go to the last cable. The Transformer modes that use different cables are Condition Splitter (True>Top Cable) and Alternating Split. With the former, you set Conditions that the incoming note must satisfy to go to the top cable, and all other notes go to the second cable. Conditions might include notes within a certain pitch, velocity, or file:///H|/SOS%2004-09/Layering%20Sounds%20In%20Logic%27s%20Virtual%20Instruments.htm (3 of 5)9/26/2005 12:07:47 AM


MIDI-channel range. In the case of a velocity range, special care has to be taken for notes with zero velocities, which are interpreted as Note Offs. Alternating Split mode simply sends incoming notes to alternating cables. You need to be careful with live MIDI input, because MIDI Note On and Note Off messages count as individual events. Cabling the top cable to one Audio Instrument object and the second cable to another will result in Note On messages going to one virtual instrument and Note Off messages to the other — a sure prescription for hanging notes. The trick is to precede the Alternating Split Transformer by a Condition Splitter Transformer that sends the Note Offs to both Audio Instrument objects and sends only the Note Ons to the Alternating Split Transformer. Of course, using single Environment objects to create splits and layers is just scratching the surface. You can use the Environment to separate and route notes by just about any complex criteria you can imagine. For example, you could route notes within a given scale to one virtual instrument and notes outside the scale to another. You could have individual notes create arpeggiated sequences routed to different virtual instruments. Whatever process you choose, be it simple layering or complex note-generating techniques, the methods outlined here will get all your virtual instruments singing in sync.

Logic Tips It's not unusual to wind up creating a lot of regions you don't use in an arrangement. Periodically using Select Unused from the Audio window's Edit menu to select and delete them will clean up your audio file list. The Edit menu's Select Files submenu offers other convenient selection options, but requires that the Project Manager's database be loaded to work. If you're trying to use Logic's Audio To MIDI Groove Template or Audio To Score functions and not getting what you want, try applying extreme EQ to the audio file to emphasize the groove or melody, then Bouncing the result to create a new file from which those processes might work better. The Matrix Edit window's Functions menu has an option to include non-note MIDI events when selecting a range of notes. If you're moving notes around in a sequence that contains other data that you want tied to those notes (such as pitchbend, modulation, aftertouch, or poly pressure) it's a good idea to turn that option on. On the other hand, if the sequence contains automation that is not note related, leave the option off. Volume, pan, and expression are examples of the latter kind of automation. If the sequence has both kinds of non-note data, open an Event editor, suppress all data types you don't want to move, and select the data between notes manually.



Have Your Say! If you want to suggest changes or improvements to Logic, then here's your chance! The Emagic 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. Published in SOS September 2004 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




In this article:

Hair Trigger Time & Space Stick To The Skin Monitoring & Pads Critical Timing Threshold Of Pain D-Day Bold As Love

Live Drum Triggering One Drummer's Search For Perfect On-stage Sample Triggering Published in SOS September 2004 Print article : Close window

Technique : Live Sound

When the epic, studio-created drum sounds of the '80s had to be translated to a stage setting, sample triggering from drum kits came to the fore — and our man Harrison began his personal quest for the ultimate setup... Gavin Harrison

During the early 1980s, drum sounds on records began to change. Snare drums mutated from a quick bash into something more like a nuclear explosion, accompanied by bass drums recorded in the Taj Mahal and tom-tom fills resembling rapid cannon fire. These new 'super-sounds' were the result of clever processing — one memorable example is the drum sound on Phil Collins' 'In The Air Tonight', created using a combination of heavily compressed room ambience and triggered noise gates. The sampling revolution transformed drum sounds even further. At first, samples were used to enhance 'real' sounds, and it became common to add a processed, ambient-sounding clap sample to a snare drum to 'big up' the backbeat. But before long, producers began to abandon acoustic drum sounds altogether, and within the space of a few years, sampled drums and percussion ruled the earth. Inevitably, gigging bands wanted to take advantage of this brave new sound world, but not all were prepared to replace their drummer with a drum machine (even though you only had to punch the rhythm into the latter once). At this stage, many musicians (myself included) began to look for an answer to the question 'how do you trigger samples from a drum kit on stage?'

file:///H|/SOS%2004-09/Live%20Drum%20Triggering.htm (1 of 8)9/26/2005 12:07:52 AM


Hair Trigger My personal quest for the Holy Grail of immaculate sample triggering began in 1986, when I went on tour with Iggy Pop to promote his Blah Blah Blah album. David Bowie (Dr Jekyll to Iggy's Mr Hyde) was the producer, and, typical of its time, the album featured sampled drum sounds — big, explosive snares and heavily-treated, massive bass drums. A real kit could not produce those kinds of noises on stage, and to make matters worse, the drum sounds on the album were significantly different from track to track. My mission was to play a real drum kit on stage with Iggy, while somehow also triggering those fabulous sounds from the record. The master tape was procured, and off I marched to a London studio for a sampling session, proudly clutching my cutting-edge, Hertfordshire-built Greengate DS3 sampler. (This idiosyncratic gadget, actually a card which fitted inside an Apple 2 computer, was cutting-edge technology back then, but seems laughable now — its total RAM memory was 1.5 seconds of glorious eight-bit mono sampling.) I duly sampled Bowie's monster drum sounds from the original multitrack tape into the Greengate and saved them on five-inch floppy disks, half afraid that the Thin White Duke would burst though the door and demand to know what the hell I was playing at.

Gavin's original 'hi-tech' drum triggering setup used contact mics on snare and kick to send an audio trigger to a Roland Octapad, which in turn sent MIDI notes to a Greengate DS3 sampler. The early '90s saw him moving to a more refined version using an Octapad II (left) with drum sounds from an Akai S3000XL sampler, plus commercial drum triggers.

To trigger the samples from my drum kit, I used miniature piezo contact mics (I can't recall the make, but I believe they were the same ones Simmons used inside their octagonal drum pads). I attached these gizmos to my snare and bass drums and plugged their flimsy cables into the audio inputs of another miracle of modern science, the Roland Octapad. Each time the snare or kick was played, the contact mics emitted an audio 'spike' which the Octapad duly transformed into a MIDI note-on command (MIDI, of course, being another new wonder of the day). The MIDI notes were relayed into the Greengate, which pondered what sample should be played, and about six weeks later, the selected noise would come flying out of its audio output. Hurrah!



Time & Space Whether triggered samples are involved or not, no drum kit can sound and feel right if it is mixed incorrectly. Sound engineers tend to fixate on kick drums, and I've often stood out front at a concert in which the bass drum is very loud in the PA, but the snare drum (which in real life is much louder) is being left out of the mix. This can happen because the sound engineer feels that the snare drum is acoustically loud enough to be heard without amplification. However, the sonic separation of kick and snare, depending on the size of the venue and the distance of the drum kit from the PA speakers, can cause terrible imbalances of timing and effectively ruin the precious groove. The reason for this is that sound takes roughly three milliseconds to travel a metre, so if a drum kit is set ten metres back from the front of the stage, its acoustic sound will arrive in the auditorium 30 milliseconds later than the sounds coming through the PA. Even 30ms is enough to really upset the integrity of a groove; if the bass drum is heard through the PA and the snare isn't, the two will lose their all-important timing relationship and the groove will suffer. The same phenomenon affects all loud on-stage sound sources, such as guitar, bass and keyboards. A smart front-of-house engineer could compensate by delaying the PA speakers by the appropriate number of milliseconds, to ensure that the stage and PA sounds arrive at the listeners' ears at the same time.

Stick To The Skin The contact mics had their own self-adhesive backing, but during 90 minutes of non-stop Iggy Pop thrashing they were liable to jump off the drums and land on the floor. Attaching them to the drum shells didn't work, as that failed to generate enough level for a reliable trigger. The only solution was to fix the little blighters directly onto the drum skins with silicone glue — definitely not ideal from an audiophile point of view, but with the samples belting out of the PA at high volume, who cared about the tone of the real drums? Unwanted triggering was a big problem — sometimes just the sound of me clicking my drum sticks together during a song count-in would be enough to activate the snare trigger, causing an embarrassingly loud snare sample to come blasting through the PA. Vibration from my feet on the drum riser, loud bass notes (depending on the proximity of the bass cab), or Iggy himself jumping on the drumset from a great height smeared in peanut butter while howling 'I Wanna Be Your Dog', were all events which could set a trigger off. Loading the samples between songs was a nightmare; the crowd would be going mad, Iggy would be screaming 'come on you muthaf***ers, kick it off!' at the band, and I'd be down on my hands and knees staring at a blank green screen, trying to remember which 'key word' to type into the Greengate so it could load the right sample. Still, when the triggering system worked (which was about 80 percent of the time), everybody was pleased and I was hailed as a conquering hero. Our frontof-house engineer was particularly happy with the enhanced sound of the bass



drum (I used a big, crunchy kick from the album, and also a tight, dry Linn LM1 bass drum sample, a useful An Alesis DM5 drum sound and trigger-toweapon in overcoming the boomy bass MIDI module marked one step in Gavin's response of some large halls). The journey towards a setup that would really trigger samples instantaneously — but the fly audience joined in the celebration too, in the ointment was still MIDI's intrinsic delay. showing their appreciation by throwing things at me: during the course of the tour I was hit by various items including rolled-up tee-shirts, a woman's blonde wig, a metal ashtray, and, strangest of all, a bible, clearly one satisfied customer's way of indicating his faith in the Godfather Of Punk's music. Amidst all this unconfined joy, one person was less than impressed... me. The samples always sounded late (when I could hear them — the stage monitoring wasn't exactly reliable), and if the acoustic drum source and the triggered sound were mixed into the PA at equal volume, everyone would hear a pretty obvious flam. What I really wanted was an instant, delay-free triggering system where the sound of the real drum and the triggered sample happened at precisely the same time and became one new super-sound. It seems simple enough, but this divine synchronicity proved to be highly elusive.

Monitoring & Pads When I first played electronic drums in a rehearsal room, I realised that it's a very strange feeling to hit a pad and hear a sample come out of a monitor some distance away. We drummers are used to hearing acoustic drums 'explode' in our faces, and after many years of that experience it's hard to play any other way. For that reason, it's crucial to have a good live monitoring system. I recommend 'inear' monitors or headphones as the best way of hearing your samples directly.

Fast forward to 2004 and Roland's V-Drum systems are at the cutting edge of electronic percussion, using real mesh drum heads for a comfortable playing feel.

Drum-pad technology has improved greatly over the last twenty years. The original Simmons SDSV electronic drum pads were made out of an incredibly hard, shiny plastic substance also used in police riot shields. Hitting the pad hard with a drum stick sent a nasty shockwave up the arm, causing carpal-tunnel syndrome, loose teeth and various minor personality disorders. The pads' surface also made a loud 'clack' when struck, which would cause problems when the noise spilled onto drum mics. It also pointed up the latency problem, as the 'clack' was liable to flam with the attack of the sample! The Roland Octapad's plastic pad zones had more 'give', generated less impact noise, and were much nicer to play



Nowadays, units such as the Pintech drum pad use mesh drum heads. These are incredibly quiet, and a joy to play from a feel point of view. As you can't really hear the sound the pad makes when it's struck with a drumstick, any obvious flamming with the triggered sample is eliminated, which helps create the illusion that triggering is instant. Clavia also make drum pads with a choice of mesh heads or real drum skins, the latter padded out with foam to kill all resonance. As an alternative to drum-shaped pads, the KAT MIDI controller is a good professional device for percussionists. If you want to get away from the traditional drum kit, there are now several electronic drum kits out there with agreeable playing surfaces. The Roland Vdrum kit (which also uses mesh heads) is very nice to play and reasonably responsive, but the big problem with such units is that you're stuck with the factory sounds, and however great and versatile these seem at first, you'll always want to add something new at some point. (Yamaha's new DTXtreme IIs has on-board sampling). But personally, though I'm impressed by the technology and came close to buying a V-drum kit, I feel most at home playing acoustic drums; their nuances can't really be captured by a set of samples, no matter how comprehensive.

Critical Timing Fast forward ten years to 1997. A leaner, wiser, more experienced drummer takes the stage with Iggy Pop, but he wasn't me — I was touring with soul diva Lisa Stansfield, and still trying to get really tight sample triggering from my acoustic drums. By this time I was using a Trigger Perfect trigger on the bass drum, running its audio output into a Roland Octapad II and the Octapad's MIDI Out into an Akai S3000XL sampler. The Akai contained a sampled bass drum triggered by my kick, and a variety of TR808 samples that I played directly from the Octapad. The results were still (to my ears at least) pretty unsatisfactory — everything sounded late, and the bass drum sample and the real kick still produced their usual horrible flam when mixed together. In an attempt to get rid of the flams, I tried other triggering interfaces, such as the Akai ME35T. According to the sales pitch, this audio-to-MIDI box was capable of instant triggering, but I discovered that it had a 13ms delay from input to output. When I mentioned this to an Akai salesman, I was told that I was splitting hairs, and that no one could hear that kind of delay. That made me even more obsessed. Determined to hunt down the fastest system in the west, I bought an Alesis DM5 drum module, which has a set of internal sounds and built-in audio trigger inputs, but this also suffered from late timing. I began to suspect that the problem was caused by the very thing that was sent to this planet to save our souls... MIDI. Every time MIDI was involved, I got horribly late triggering and, more worryingly,


Impossible as it might seem, the mechanism used in this Axis bass-drum pedal makes it


inconsistent timing. Finally, I decided to use Logic possible to set it up to trigger a sample fractionally before the Audio to test the timing response of the Alesis bass-drum's beater hits the DM5, by recording a live bass drum playing a head. See main text for a series of single hits into Logic; feeding the output description of how it works. of the recorded bass drum track straight into the DM5's audio trigger input; playing back the recorded bass drum track while recording the DM5's analogue output onto a new Logic audio track; and then finally repeating the previous step, but this time recording the DM5's MIDI output onto a Logic MIDI track. Recording the analogue outputs enabled me to look at the DM5's audio response time, while recording the MIDI notes it generated in response to incoming triggers gave me an idea of its MIDI processing speed. Having recorded these tests, I compared the timing of the DM5 recordings to those of the original bass drum hits, and found significant delays in both the audio and MIDI events. I also found that the delays were slightly different each time, even though the recording conditions were identical. I also tried the same test on my Akai sampler with similar results, eventually concluding that although MIDI wasn't entirely to blame for triggering delays, it was a major contributory factor. Using a triggering device to fire off custom sounds in an external sampler, I could expect unpredictable MIDI processing delays from both units — clearly a recipe for timing disaster. In the end, I found the only fix was to trim the front off the bass drum sample to reduce its attack and minimise the flam effect. Far from ideal, but it got me through the tour!

Threshold Of Pain Whatever the triggering system, accurate sample triggering relies on the manual setting of an audio threshold for each trigger source. The threshold value determines the minimum level at which the incoming audio generates a trigger impulse — all signals below this level are ignored. The level therefore has to be set low enough to pick up quieter hits (not that there were many of those in Iggy's set, for example), but high enough to rule out cross-triggering from adjacent triggers or background noises. With multiple drums triggering samples, crosstalk becomes an even bigger issue, as a loud hit on one tom-tom can easily cause its neighbour to trigger. Onstage background noise is often extremely loud, so getting the threshold settings right in a live situation can be very tricky. Another problem is that trigger boxes sometimes over-react and generate two or three impulses in response to one loud hit; some units cleverly overcome this by supplying a 'hold' function which can be set to prevent multiple hits occurring within an excessively short period. I've learnt that setting up live drum triggers is a lot like making love to a beautiful woman — it requires a lot of patience, determination and fiddling, and you seem to have to tweak the damn things almost on a daily basis.

D-Day file:///H|/SOS%2004-09/Live%20Drum%20Triggering.htm (6 of 8)9/26/2005 12:07:52 AM


By the time a second tour with Lisa Stansfield came around in 2001, I had bought a Clavia ddrum4, a professional drum module from the company who make the Nord synthesizers. The ddrum4 has a large menu of high-quality mono sounds, 8MB of Flash RAM for user samples and its own set of built-in audio trigger inputs, enabling internal samples to be triggered directly, without MIDI being involved (having identified MIDI as a source of timing slop, I was happy to see it eliminated). But even with direct audio triggering, the ddrum still outputs its samples slightly late — 4.6ms, to be precise. The good news is that this delay is at least consistent, so once you find a sample that blends well with the acoustic drum source, it will sound the same each time — a major breakthrough! Although 4.6ms sounds insignificant, it's enough to cause samples with really sharp attacks to flam with the acoustic source, but by choosing samples with a less pronounced attack, you can get satisfactory results. Thanks to the ddrum, I was able to use a triggered bass drum sample all the way through the tour with no problems. Clavia have shown their commitment to sample triggering by manufacturing their own, very good drum triggers — these clamp onto the rim of the drum and have built-in jack or XLR sockets so that you can attach your own cables. Being able to use your own samples in the ddrum4 is a great selling point, but loading them in is pretty painful; the unit doesn't have front-end sampling, so you have to use an external sampler to transfer your samples into the ddrum4 one at a time. This process Clavia's own high-quality drum triggers. relies on that slow, tedious transfer method known as MIDI sample dump, which, along with hexadecimal code and haemorrhoids, is one of the greatest pains in the rectal zone known to mankind. Using MIDI sample dump, it takes a long time to transfer even a very short bass drum sample — having got to the point where I thought I could dispense with an external sampler, it was ironic to discover that I still needed my Akai S3000XL to perform the transfers!

Bold As Love Once my drum samples were resident in the ddrum4, things started to look good. They got even better in 2003 when I discovered a really fantastic bass drum pedal made by US company Axis, who have invented their own ingenious system of mechanical triggering. Axis trigger devices are quick off the mark — so much so that they can actually be set up to trigger a sample before the bass-drum beater hits the skin!



Here's how it works: that funny-looking miniature hammer's thin metal arm (right) is attached to the central crossbar. As the pedal is pressed, the crossbar rotates towards the head and the beater (which is just out of view) is thrown forward to hit the drum. At the same time, the tiny hammer moves down and strikes the surface of a black steel disc, visible on the left of the picture. The trigger sensor lies just underneath the surface of this disc. Using an Allen key, you can adjust the angle of the hammer's arm so that it hits the trigger disc fractionally before the pedal strikes the bass drum. The triggering mechanism has a female jack socket on its underside, from which a cable can be run to one of the ddrum4's trigger inputs. After some experimentation with the hammer angle, I was able to effectively 'advance' the trigger point and eliminate the ddrum4's 4.6ms delay, thereby achieving instantaneous sample triggering in real time. Eureka! The Axis mechanism is very precise, so accurate triggering does depend on you having a nice clean bass drum technique. Sadly, no-one has yet manufactured viable 'advance triggers' for snare drum and tom-toms — one solution would be for a drum stick to break a beam of light as it descended, but the potential for crosstriggering would be enormous. However, over in nutty California, a friend reports seeing a small group led by US synth designer Don Buchla using light-beam technology to trigger percussion samples, the joke being that samples of cymbal, snare drum and cowbell were sounded when the player's stick hovered a foot in the air above the instrument in question! Published in SOS September 2004 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




In this article:

Specialist Benefits Selecting Components Customer Problems Noise, Heat & Cost AMD Vs Intel Reliability & Testing Meet The Panel Technical Support Final Thoughts

Music PCs: Manufacturers' Round Table PC Musician Published in SOS September 2004 Print article : Close window

Technique : PC Musician

Buying a music computer from a specialist builder is an increasingly popular option for the PC musician. We rounded up some of the leading names in music PCs to find out exactly what they offer. Martin Walker

There are people posting on PC music forums who insist that anyone can knock up a good music PC in a couple of hours from a list of parts recommended by other forum users, and who can't see what benefits there are in paying a little more for a machine readymade from a specialist music retailer. It's true to say that if they know what they're doing, are provided with a good parts list, and are prepared to put in a fair amount of detailed research, some people can find the DIY approach works for them. However, it won't ensure that they get a reliable system from day one, with a guarantee and ongoing technical support if anything does go wrong in the future. And sometimes the DIY approach can also go disastrously wrong. Tony Cox (Digital Village):

One SOS forum poster was recently trying to track "The most difficult hardware down a problem that prevented his new DIY PC from problems are intermittent ones that can only be booting up properly. He was obviously resolved by a long-term knowledgeable, systematically tried a variety of soak test. Fortunately, most solutions, appealed to other forum members for hardware faults are further suggestions, and eventually sent back his new detected and resolved at motherboard for a replacement, but the replacement either the build stage or install/testing stage." gave exactly the same problems. The cause eventually turned out to be due to being sold the wrong sort of RAM, and the poster openly admitted that he'd wasted a total of two months before his new PC was properly up and running.

file:///H|/SOS%2004-09/Music%20PCs%20%20Manufacturers%27%20Round%20Table.htm (1 of 16)9/26/2005 12:07:59 AM


Such things happen all too often, and with this in mind I thought it would be fascinating to find out how the specialists choose their components, test out incompatibilities, and ensure their systems are reliable, as well as asking what technical problems they run into. I contacted a cross-section of retailers who advertise in SOS, all of whom were very interested in taking part in this 'Round Table' discussion. You can read more about them in the box on page 222.

Specialist Benefits Which benefits do you feel your customers obtain from buying one of your systems, as opposed to building their own PC from a recommended list of parts? Phil Elliott (PE): "I could write a book on this subject! However, I can summarise with two words: service and support. Red Submarine supply fully supported solutions, not just boxes with components in them. If you build your own PC, you are your own technical support and warranty service. If you buy from us, you are forming a partnership with a group of people that understand the purpose of your purchase, have access to a wide range of resources such as spare parts, and maintain direct links with manufacturers." Robin Vincent (RV): "Peace of mind, I think, is the biggest benefit among many. The knowledge that if it does go wrong you have a professional company that will sort it out for you, and that you'll have a working system in the first place. It's not that difficult to build a computer from parts, assuming, of course, that all your parts work, and that they can work together, and that you're not in any particular hurry. With Carillon you're not taking any risks — you've got a computer with a warranty that will be completely compatible with your soundcard and music software, and we'll install all your stuff for you if you like. If you can move a mouse around, you make music on a Carillon computer — you don't have to be, and you shouldn't have to be, a computer technician." John Oxley & Joe East (JO): "It's no secret that PCs are easy to build. The hard part is finding out what is wrong when things are not working as they should. We've probably all been there at some point, pulling our hair out, thinking what could possibly be wrong, having tried everything imaginable to make things



happen. By purchasing from a company such as ourselves it takes away these stressful moments from the customer. However, if you build a PC yourself you become your own first and last line of support, and if you have a problem with a component then the onus is on you to sort out a replacement with the manufacturer or supplier, sometimes only to find out that the component in question is working perfectly, and that something else is causing the problem." Rick Holliday (RH): "When a customer buys a system from Millennium, any component Phil Elliott (Red Submarine): "There failures or potential problems with the setup are many off-the-shelf systems that are sorted out before the customer receives make superb office machines, but the machine. Many people who build their own are unsuitable for musicians machines experience problems they do not because they might have a Firewire chipset or drive controller that have the resources to solve. It can be very wasn't intended for high-bandwidth difficult to work out which component of a PC audio applications." that will not boot is faulty without spare components to test. We also spend time thinking from the customer's perspective, to make our machines as easy to use as possible, by setting up templates and example projects." Phil Rees (PR): "Our customers place a single order with a single supplier, and then we take responsibility for getting all the parts together and assembling them into a properly working machine. We know and avoid potential pitfalls, and our experience helps us to spot when things aren't behaving normally. Our customers also don't have to spend hours installing hardware and software and getting it working, or frustrating hours on the phone or on the Internet, trying to get information — they can just get on with making music." Dave Elliott (DE): "We have five years experience in building music-based PCs, we use recommended components which have been fully tried and tested, we run an eight-hour burn-in test to ensure components work correctly, and we set up soundcards and DSP cards correctly to avoid IRQ clashing. So the customer receives a PC that is ready to go straight from the box, and no setup or configuring is required on their part. We also have direct access to component manufacturers, to ensure that any faults or incompatibility issues are ironed out before the system is ready for manufacture." Tony Cox (TC): "Choosing your own system at Digital Village allows you to spec it exactly as you see best, so you get a totally personal and custom system, but without any of the problems of building it yourself."

Selecting Components file:///H|/SOS%2004-09/Music%20PCs%20%20Manufacturers%27%20Round%20Table.htm (3 of 16)9/26/2005 12:07:59 AM


How do you decide which selection of components to use in each of your PCs, and how often do you change this list? TC: "Our main criterion is stability in use and supply. To maintain quality control we only change the critical components when there is a major shift, for example when a new chipset needs evaluating, or a new CPU model, or type of RAM." PR: "We read articles, and we pay attention to news of new components in the trade press and from suppliers. We regularly get evaluation samples of new hardware and software into the workshop, where we can put them through their paces. We put significant effort into testing, benchmarking and checking for compatibility. However, we intentionally limit the range of choices that we offer, as we prefer to use quality parts we know we can trust." DE: "Our decision is based primarily on the quality of components. Each system we design undergoes a four-week test period to ensure chipsets and components prove to be reliable and stable. In the interest of compatibility, we stick to brands such as Asus and Intel, as these have proved to work best with a large range of soundcards and music software. Each month we review all our systems to ensure our customers receive the latest components and equipment as technology progresses." RH: "Our decisions about which components to use are based partly on exhaustive research but also on firsthand experience and testing. We try to use components that will make a fast, reliable, easy-to-use PC, but which also represent value for money. We also try to consider the upgrade potential of the PC and the long-term reliability of the components and the system as a whole, probably to a much greater extent than most mainstream PC retailers, as we realise that musicians who use PCs tend to rely on them heavily for their work. When we have identified a candidate for a new component to use, we test it thoroughly with the most popular music software and hardware to be as sure as possible

John Oxley

John Oxley & Joe East (Nusystems): "We offer Intel- and AMD-based systems as an option, but we seem to have fewer problems with Intel-based systems. This may be due to the fact that components and various hardware and software products are produced with Intel processors in mind."



that it is fully compatible. We will consider a new component for use as soon as it is released, so our list of preferred components changes with the market." JO: "There are many issues for us to consider when deciding which components to use in our systems: reliability, longevity, availability, manufacturer's warranty and returns procedure, to name but a few. We have even had to change components due to the way courier companies handle packages in the past. How do we decide? We test them. We will buy in different components that we come across or hear about, put them together and carry out several benchmark tests to push our systems to the limit. We're always looking to improve and if we see a development that we consider to be a benefit to our customers, we will implement it in our systems. If we can find other components that will do just as good a job but cost less, we will use them instead. It's our aim to bring the cost of a serious digital audio workstation down to that of a standard off-the-shelf PC, or as near to it as possible without compromising on quality." PE: "Our choice of components is based on compatibility, performance and reliability, and our list is updated as soon as we have verified this information. There are many off-the-shelf systems that make superb office machines, but are unsuitable for musicians because they might have a Firewire chipset or drive controller that wasn't intended for high-bandwidth audio applications. Many companies put too much emphasis on CPU speed, but it's essential in an audio PC that every component is able to deliver sustainable bandwidth." RV: "Although we like to keep our ear to the ground for new and interesting products, one thing we've learned is that 'reviews' are no more than an indication that something might be worth looking at. How well something actually works comes down to using it for ourselves. We also seem to be in a slightly different world from the rest of the IT community, and things that they consider 'quiet' are quite a long way from what we would consider quiet. Our criteria for components are firstly stability, then performance and noise, then ease of use/install, then lastly price and reputation get a look in. You also have to look past the specs and the numbers and get down to the nitty-gritty of whether any gains are to be found in real life plug-in or track counts — the components we use change with technology once we're satisfied the technology is working."



Customer Problems Even specialists occasionally have technical problems to solve, and sometimes very obscure ones — although the fault is not always theirs... Robin Vincent: "At times you have to assume that the customer knows nothing about anything. We had one customer who called us because she wasn't getting any sound out of the computer. After checking all the settings with her, then dialling in to check the settings too, everything seemed fine and as it should be — we could even see an output indication in the soundcard mixer. After half an hour of fiddling around and scratching our heads we finally asked if the speakers were turned on. This was met by a prolonged silence which ended with 'Speakers?'. Perhaps she was hoping for audio via osmosis." John Oxley: "We just had a customer wipe out his BIOS trying to flash it. There was no reason to upgrade it — everything was in perfect working order but it was just something he wanted to do. It failed, wiped his BIOS clean out and he now has to fork out for a new motherboard, as this has voided the warranty. We contacted the manufacturer — Asus, in this instance — but they couldn't do anything about it. Often you can replace the BIOS chip but it was not possible for this particular motherboard. We have offered him a replacement at trade cost, as we do feel sorry for him, but it's a good example of 'if it ain't broke, don't fix it'." Rick Holliday: "Probably the strangest genuine problem was a PC with a PCI soundcard which would start to stutter and glitch after about four minutes of playback. As it turned out, this only happened if Power Management was disabled on the USB ports of a combination USB/Firewire PCI card. If Power Management was enabled, there were no problems. There was nothing plugged into any of the USB ports. We stopped using those combination USB/Firewire PCI cards immediately." Dave Elliott: "We had a situation where a new system failed on delivery to a customer. When the machine was set up it began to bleep and not POST (Power On Self Test). The bleep code was checked and this indicated that the graphics card was not present. We had the machine back to base for repair to find it working perfectly. As a safety measure we replaced the VGA card to ensure there was not an intermittent fault with it. When the machine was sent back to the customer it failed with exactly the same fault. After going through a step-by-step diagnosis with the customer (involving unplugging various cables from the machine) it turned out that a bent pin in the PS2 keyboard had caused the machine to suffer an unusual fault with the display card." Robin Vincent: "The Intel D875PBZ motherboard came with an inbuilt Serial ATA RAID controller allowing us to combine two drives into a high performance RAID0 drive. Unfortunately the earlier versions of the BIOS had a habit of resetting themselves to defaults for no apparent reason. The default setting disables the RAID controller, which results in the entire RAID drive vanishing from Windows. Not a big problem, you'd think, unless you've spent the entire weekend recording a very important session, only to find the whole recording drive missing. We were greeted with death threats from one psychotic producer, but thankfully it took two seconds to fix, once we'd talked him out of the idea of murder!" John Oxley: "One common problem we come across time and again is due to the customer changing settings, specifically on their audio interface. People tend to set the buffer size to achieve the lowest possible latency figures at maximum sampling frequency, and then they start to have problems. We feel that some file:///H|/SOS%2004-09/Music%20PCs%20%20Manufacturers%27%20Round%20Table.htm (6 of 16)9/26/2005 12:07:59 AM


people are a bit too obsessed with latency figures and how low they can go. We encourage customers to use their ears. If there's no audible delay, why push the hardware any further?"

Noise, Heat & Cost Musicians like low-noise PCs, but these are more expensive to build. How do you come up with the best compromise between noise, heat and cost? RH: "There are two ways to reduce noise: dampen the noise which is produced, or ensure that as little noise as possible is generated in the first place. We choose hard drives and optical drives that are as quiet as possible while still being reliable, and cases with internal layouts that allow efficient airflow, to make sure that even our 'standard' systems run cool and quiet. Some of our customers want inexpensive but reliable systems, and some are willing to pay a little extra for quieter machines, so we give them a choice of components, and we offer specialist cooling products (such as quiet power supplies and CPU fans) that we feel give good performance for the money. We also subject all our new PCs to stress tests while monitoring temperature and we always allow a little margin for error, as we understand that our test conditions can never represent the extremes of possible use." JO: "Low-noise PCs do not necessarily have to be that much more expensive to build — it depends on the individual customer's requirements. If the PC has to be deadly silent then, yes, it is going to cost a bit more, as acoustic case insulation and hard-drive enclosures are required, as well as a good silent CPU fan and power supply. However, at Nusystems we fit silent power supplies and graphics cards as standard, so if the customer is not recording in the same room as the PC or not particularly bothered about the PC being truly silent, we would just recommend a silent CPU fan and then you have a quiet PC that won't break the bank."

Philip Rees: "We try to deal with support queries in the same way we would if personal friends sought technical help. We will reply by email to technical support queries received by email or via our Online Support Service, and we aim to respond within one working day."

PR: "I have made a personal speciality of designing quiet systems. I apply a little theory and a lot of practical experimentation to this on-going project. I'm pleased with the results, and many customers have expressed their satisfaction. All the systems we supply are tailor-made custom solutions, and we can offer an optimised trade-off between low noise and price to suit each customer's requirements. My exclusive central airflow control system makes sure that fans only run as fast as needed, to ensure safe operating temperatures. Also, the fan file:///H|/SOS%2004-09/Music%20PCs%20%20Manufacturers%27%20Round%20Table.htm (7 of 16)9/26/2005 12:07:59 AM


speeds are co-ordinated to maintain a balanced and efficient stream of air. Its operation is independent of the computer motherboard, so it imposes no overhead on the computer hardware or software." TC: "The best option for us is to use quality components. They are more expensive than standard items, but by using quality cases, PSUs and fans we can mod these ourselves to achieve the same results as using specialised items, but still keep the build cost down." RV: "Ideally, you don't want to compromise at all — if a musician wants a quiet PC, that means paying a bit extra. At Carillon we provide the AC1 LE system which is identical to our flagship AC1 system but without the noise-reduction technology. It's a cheaper alternative for people who don't consider noise a huge problem, although there are 'levels' of quietness, and we find that our LE systems are still much quieter than most off-the-shelf PCs. However, with Intel's new Prescott CPUs the heat problem will get worse, and being able to move enough air through the system to keep it cool while keeping the noise under control is certainly a challenge." DE: "We work closely with our suppliers, Chieftec, to source a high-quality chassis with excellent air flow and heat dissipation, and also standardise on Chieftec power supplies that offer a high output with a low noise level. This helps keep costs down, as we bulk-buy power supplies and cases, and this saving is passed onto the consumer. For noise dampening we use products such as the Zalman 7000 ALCU ultra-quiet cooler and the AcoustiPack foam kit, which helps to reduce high-pitched noises from peripherals such as hard disks and cooling fans. Overall we feel this provides a good balance between noise, heat and cost." PE: "A number of different components make up a quiet PC. We allow customers to select one, several or all of these, depending on their budget and requirements. In most circumstances we would suggest, in order of noisereducing effectiveness: PSU; CPU fan; VGA heatsink (if required); acoustic insulation or drive enclosures; case. It's a good idea to get the case right at the start, as this is the most inconvenient component to change later. If a customer wants the quietest system possible, we don't compromise at all."

AMD Vs Intel If you only sell Intel-based systems, why is this, and what, if anything, would change your mind? If you do sell PCs based on AMD as well as Intel processors, what prompted this split approach? TC: "We only sell Intel-based systems using Intel chipsets, simply because this is the only way you can guarantee full compatibility. All hardware and software is tested on Intel systems, and although it is also tested with AMD systems the main problem with these is the variety of chipset permutations that can cause



issues — whereas with Intel, as long as you stick with an Intel CPU and Intel chipset you are almost certain to have no compatibility issues. The only change to this will be with the introduction of AMD's 64-bit processor and the eventual arrival of Microsoft's 64-bit Operating System. Once this leap in technology has settled down and is fully embraced by the hardware and software companies in our industry, the gain in performance would be worth making the switch for. On the other hand, Intel are already about to introduce 64-bit instructions to their new CPU range, so only time will tell." RV: "Intel has worked really well for us over the years. The combination of performance and stability has been unbeatable. AMD has always been of interest to us and we have spent time testing out AMD systems to see if they offer anything worth pursuing. Up until recently the issues of noise, heat and compatibility have made AMD a troublesome choice next to the cool, quiet and compatible running of Intel systems. However, we are currently testing out Opteron and Athlon 64 systems to see for ourselves what steps AMD have taken to address these issues, and testing out the potential performance gains with music applications. Carillon aim to offer the best systems available and if AMD can provide them, we are happy to supply them — but we'll be our own judge of that." DE: "We sell both Intel- and AMD-based systems. As stockists of all AMD processors and compatible motherboards we are in a strong position to design a workstation that is stable and reliable for music use. Stocking both processors means that the customer has a wider range to choose from." JO: "We offer both Intel- and AMD64-based Dave Elliott (Inta Audio): "The systems as an option for desktops and laptops, customer receives a PC ready to to give customers the choice. Within our team go straight from the box, and no we have more or less an equal amount of setup or configuring is required on experience in building and supporting both AMD their part. We also have direct and Intel systems. Intel are far more popular but access to component manufacturers, to ensure any it often comes down to an individual's past faults or incompatibility issues are experiences. We seem to have fewer problems ironed out before a system is with Intel-based systems, and this may be due ready for manufacture." to the fact that components and various hardware and software products are produced with Intel processors in mind. At the end of the day, though, it comes down to what each individual customer wants, and as long as we still have customers



who prefer AMD over Intel, we will continue to build AMD-based systems." PR: "We intentionally restrict the range of options we offer, so that we sell parts we are sure will work well. Our preference for Intel derives from their chipsets, rather than the processors themselves. This has been especially important for glitch-free streaming audio over USB. Intel are closely involved in the development of important interconnection standards such as PCI and USB. As a result, their implementations tend to be regarded as definitive. We have built AMD systems, including Athlon 64, for evaluation, but found that the benchmarking results did not make a clear case for adopting the platform at this time. If there was a significant price/performance advantage, and we found no compatibility problems, we would reconsider our policy." RH: "At present we only sell Intel-based systems. Many manufacturers still advise using their products with Intel systems (for example, Digidesign), and many more will unofficially suggest that Intel would be better, if you speak to their technical departments. We want our computers to be compatible with as wide a range of music hardware and software as possible, and we do not want customers in a position where they're using equipment in a way 'not recommended' by the manufacturer. If something goes wrong, they could find themselves stranded. If more manufacturers were willing to endorse AMD systems, we would certainly reconsider. The situation is much less 'black and white' than it used to be." PE: "We supply both Intel and AMD systems. This approach is driven by customer demand, as AMD has gained a reputation for being very fast, yet costeffective. Initially, there were problems relating to heat dissipation, reliability and compatibility, but we have solved the majority of these issues now, aside from a few compatibility questions with certain interfaces. However, we no longer supply dual-processor AMD systems, as reliability, compatibility and availability of spare parts presented problems we could not easily solve."

Reliability & Testing What steps do you take to ensure that your systems work first time when delivered, and remain working reliably? JO: "Every system we build undergoes an intensive 48-hour burn-in test, to verify that it is stable and reliable, before being shipped. This is carried out to stress the components of the system, asking them to execute tasks that will highlight any potential problems. Issues with heat, stability and compatibility become evident during this period, and if there is a weak link we have the opportunity to swap out the offending component, which may otherwise have failed in the hands of the customer — a possible disaster for the recording musician. Although all our components are rigorously tested for compatibility before they are chosen, there are occasions when components, specifically RAM, can exhibit glitches or faults which cause intermittent errors such as blue screens or system crashes. The file:///H|/SOS%2004-09/Music%20PCs%20%20Manufacturers%27%20Round%20Table.htm (10 of 16)9/26/2005 12:07:59 AM


tests we carry out ensure that this will not affect our customers once they have received their system." DE: "Each of our systems is rigorously tested using our burn-in test software. These tests put the hard disks, memory, CPU and graphics under stress for a period of eight hours, to ensure that all components are working efficiently. Once the workstation has passed the tests, we then provide a report for the customer, outlining the test results. At this point, the chosen soundcard and software is installed and configured. We then take steps to ensure that all updates and drivers are complete. Finally, we follow a 22-point checklist that applies all the necessary tweaks to the system. It is then packaged in a specially designed box, to guarantee that it arrives in full working order." PE: "Firstly, when building the actual chassis we run burn-in tests for several Rick Holliday (Millennium Music Software): "We offer telephone support for any initial days to ensure stability of every problems, although our customers rarely component. A system designed to be need to use this, partly because we very quiet creates additional encourage them to send in any software and complications, as we prefer to utilise hardware they intend to install on their fewer fans running more slowly. So machine so we can configure and test it for them." temperature levels are monitored carefully, as a CPU running a few degrees too hot is likely to result in a shorter life span. During the configuration stage we have a strict test procedure. Many companies clone the Windows installation from a master copy, but we don't do this, as this process also acts as a test for each component, and can highlight potential problems. As all our systems are bespoke, we create a 'ghost' backup image of the specific installation we create. This is useful just in case a user accidentally formats their hard drive. Yes, it does happen!" PR: "We start with reliable compatible parts, such as Asus motherboards and Corsair memory, which we know are capable of stable performance. We also build the systems as solidly as we can, making sure that parts are fitted securely, and install the latest software and firmware updates. Then we run extensive functional tests, diagnostic tests and soak tests. We test the actual installed software with the actual hardware. Finally, we run Passmark's BurnInTest for at least 12 hours before we ship systems. We believe that this is an excellent way of catching potential instability."



TC: "Once again, using branded components means that we reduce the failure rate to a minimum, and every system is fully set up and tested prior to delivery. Also, to ensure that they remain working, all our systems have a full customer restore option. The most difficult hardware problems to resolve are intermittent ones that can only be resolved by a long-term soak test. These may be caused by a BIOS setting/revision change or a memory issue. Fortunately, most hardware faults are detected at either the build stage or the installation/testing stage, and they are resolved then." RV: "Testing combined with the use of known components is the best defence we have against unreliability. If you know that your component choices are good in the first place and your testing has borne this out, you've done the hard work already. All our systems are installed by an engineer who then spends time testing and checking the system. That system is then tested and checked by a second engineer as a safeguard. Every input and output, MIDI and audio, is tested, and the music software is run and used during the testing process. When the system's backup recovery disk is created we know that system works, and if something subsequently gets installed that messes up the system, or there's a virus attack — or the more common attack of the six-year-old daughter — you can return the system to a working state in a few minutes by running that recovery disk. Unfortunately, after all our hard work we're left in the hands of the courier who delivers the system to the customer. The small number of DOA (Dead On Arrival) systems we experience are down to extreme bad handling by couriers. For example, one delivery was dropped off the tailgate of a truck onto concrete." RH: "We subject each new system to a battery of tests. These include hardware stress tests using programs designed for this purpose; individual component tests, such as burning test CDs; performance tests to ensure that, for example, a new hard drive transfers data as quickly as it should; and 'real use' tests — for example, recording audio test tones to ensure that the system records and plays back without glitches. We make sure that any potential problems resulting from the choice of hardware and software are solved before our customer receives their PC."

Meet The Panel Tony Cox, PC Sales & Marketing Manager, Digital Village www.dv247.com

Tony first started out working for Francis Day & Hunter, an MI Store in Charing Cross Road in 1983, later moving to Turnkey as Sales Manager. His introduction to using computers for music was via a Commodore 64 using Pro 16 (later to become Cubase) software, followed by an Atari, then PCs in the early '80s. He joined Digital Village in 1999 as Sales Manager of their Croydon store and became DVPC Sales & Marketing Manager in early 2001. Dave Elliott, Technical Director, Inta Audio Computer Systems www.inta-audio.com

Dave has been interested in making music from an early age. He had his first experience with computers and music back in 1999, and went on to build his first



music computer. It was at this point that the idea for Inta Audio began to evolve. With help from other technically minded people, Inta Audio has become a welestablished company known for its outstanding music-based computers. Phil Elliott, PC Music Specialist, Red Submarine www.sub.co.uk

Having worked for nine years as a recording engineer, Phil joined Red Submarine in 2000 as a sales consultant, covering PCs for music. As a self-confessed 'frustrated musician with perfect pitch' Phil spent many years lending an ear to budding musicians in the studio, and now spends his time lending an ear to people who want their own studio! Rick Holliday, Managing Director, Millennium Music Software www.millennium-music.co. uk

After releasing a string of albums and singles as part of the '80s band B-Movie, Rick started building PCs for Millennium customers over 10 years ago. His first PC was supplied to a Radio 4 reporter who wanted to edit his own reports in the garden rather than book time in one of the BBC studios. After everything went 'pear-shaped' on the reporter's home PC, Rick was instructed to build him a "PC that would just work!" and did just that. John Oxley and Joe East, MD & Executive Director, Nusystems www.nusystems.co.uk

John and Joe form the nucleus of Nusystems, with a combined 15 years experience in building computer systems for music production. Nusystems was born with the aim of making dedicated audio PCs more affordable, bringing the cost down as close to a standard PC as possible, while maintaining the superior quality that a dedicated DAW requires. Philip Rees, Owner, Philip Rees Music Technology www.philrees.co.uk

Philip Rees is an experienced, qualified electronic and computer engineer who established his own business in 1986. He's designed and manufactured a range of MIDI accessories marketed under his own name, and has retained a personal interest in hi-tech music-making. In recent years, he and his team moved into building turnkey music PCs, which is now the main focus of their activities. Robin Vincent, Technical Director, Carillon Audio Systems www.carillondirect.com

Robin is responsible for ensuring the quality of Carillon PC systems, which includes overseeing production, research and development of new products and continuing development of existing products. He's responsible for the technical support department, dealing first-hand with customers and the occasional on-site installation. He also writes all Carillon's tutorials and documentation, maintains their web sites, and trains all the staff at Carillon and their international distributors. He's also managed to find time to write two computer music books (PC Music — The Easy Guide and The Guitarist Guide to Computer Music, both available from PC Publishing).

Technical Support One obvious benefit to customers over a DIY PC is after-sales technical support. What do you offer? DE: "We offer three years technical support on all hardware and software



purchased with a workstation. Our tech support can be reached via telephone, email or fax; tech support hours are Monday-Friday 10.30am to 3.30pm and Saturday 10am to 1pm." TC: "Digital Village Tech support comes in two tiers. Our customers have the option of either talking to a PC specialist in one of our stores or receiving tech support direct from Digital Village head office, where all our PCs are built and prepared by one of our technicians." RV: "We provide 'Carillon How' tutorials, Carillon Fix remote dial-in diagnostic software, and a 'Carillon Recovery' backup image free with every system. Systems that cannot be fixed with the above procedures or have gone down due to hardware failure will be collected and fixed/repaired/ replaced at Carillon and returned to the customer free of charge. We also have a dedicated support web site containing all the latest driver and software version information, issues, tutorials and resources, plus office hours phone and email tech support. We have experienced support staff who have come across every imaginable configuration and can use our Carillon Fix utility to dial into a customer's machine and see and operate it remotely. This not only allows us to fix problems but also lets us see how the customer uses the system, which can often show where the problem originates." JO: "As a new company we are still improving our support, and have a few new features to add to the package. We hope they will be in place by the end of Summer 2004. These will include three years technical support, by phone and email, as Robin Vincent (Carillon): "Although we standard with all our systems. We already like to keep our ear to the ground for new and interesting products, one thing provide support on all aspects of our we've learned is that 'reviews' are no computers, as well as music software and more than an indication that something soundcard configuration. One good thing might be worth looking at. How well about dealing with an up-and-coming something actually works comes down company such as ourselves is that each to using it for ourselves." system we build is unique and individual and we remember all of our customers. Generally, the person that actually built the PC is always on hand to answer any query the customer may have." PR: "We try to deal with support queries in the same way we would if personal file:///H|/SOS%2004-09/Music%20PCs%20%20Manufacturers%27%20Round%20Table.htm (14 of 16)9/26/2005 12:07:59 AM


friends sought technical help. We will reply by email to technical support queries received by email or via our Online Support Service, and aim to respond within one working day. We also respond to technical support queries received by telephone, which is what most customers prefer. The support contract is free for the first year and renewable at charge for subsequent years. The annual charge for renewal of the support contract is currently £25." RH: "We offer technical support by email for the lifetime of the machine. All our PCs are supplied with the Millennium Help Files, containing comprehensive help, advice and solutions to common problems. We also offer telephone support for any initial problems, although we are proud to note that our customers rarely need to use this, partly because we encourage them to send in any software and hardware they intend to install on their machine so we have the opportunity to configure and test it for them. Finally, our systems are covered by a 'whole system' warranty for the first year, meaning that if, for example, the hard drive fails, we will collect the PC, replace the hard drive, reconfigure and test the machine and return it to the customer ready to use once more." PE: "Red Submarine have sold audio computer systems for seven years now, and our 10 full-time members of staff are able to draw on this experience to help customers avoid many of the usual pitfalls associated with computer music production. We provide technical support by telephone and email weekdays from 9am to 5.30pm, and customers often speak to the engineer who actually built their machine. If we are unable to solve a problem by email or telephone, the technician will arrange free collection of the machine within the warranty period. We aim to turn around most repairs within five working days."

Final Thoughts The answers provided by the retailers who've participated in this article provide an insight into the amount of effort that goes into building bespoke PC systems that should work first time they are switched on and carry on functioning reliably for the lifetime of the machine. You can build a PC from a list of parts, but it's obvious that a considerable amount of research goes into the systems being discussed here, not only in choosing a balanced and reliable set of components, but also sourcing the most reliable suppliers and soak-testing the assembled PCs. Published in SOS September 2004







In this article:

More Chips Please PC Snippets What Does It All Mean? Changing Drive Letters

New Intel Chipsets: Good For Musicians? PC Notes Published in SOS September 2004 Print article : Close window

Technique : PC Notes

Intel's latest chipsets are finding their way into new PCs — but do they offer any advantages to the computer musician? Martin Walker

As I write this at the end of June, the first motherboards featuring Intel's new 915P (Grantsdale) and 925X (Alderwood) chipsets are appearing from manufacturers including Abit, Asus, Gigabyte, MSI and Intel themselves. They herald the start of a new era for Intel-based PCs, with loads of new features. Let's start with the new LGA775 (or With many new features, Intel's latest 925/915 chipsets could take the world by Socket T) CPU socket. This is a storm — once their performance improves. completely new way of interfacing Pentium 4 processors to the system buss, and in a radical departure from previous designs it places the pins on the socket rather than the processor. Motherboard manufacturers have apparently been complaining that this puts up the cost of their half of the assembly tenfold, although initial concerns about the socket's fragility seem largely unfounded. The extra 297 pins of LGA775 are mostly to supply power, and Intel have also improved thermal performance so that heat can be removed more efficiently. For musicians this is a possible plus, since it enables the use of a new heatsink and large-bladed fan that can apparently spin at lower speeds, and even stop if temperatures drop sufficiently. Intel have already released a new 3.6GHz Pentium 4 model in the LGA775 package, along with repackaged versions of various other slower models. All of them use the Prescott core that Intel first introduced in their Pentium 4E processors to manage clock speeds higher than

file:///H|/SOS%2004-09/New%20Intel%20Chipsets%20%20Good%20For%20Musicians.htm (1 of 5)9/26/2005 12:08:04 AM


3.6GHz, but apart from the new packaging nothing else has changed internally. As with the Dothan processors discussed in last month's column, the new Pentium 4 models adopt a new and confusing numbering scheme, with the 2.8GHz version becoming the Pentium 4 520, while the the 3GHz, 3.2GHz, 3.4GHz and 3.6GHz models follow on as the 530, 540, 550 and 560 respectively.

More Chips Please The 925X chipset roughly replaces the existing 875P Canterwood, while the 915P replaces the 865P Springdale range. Both support the new 800MHz LGA775 processors, dual-channel DDR2 memory, up to four PCI Express x1 slots, and one PCI Express x16 slot for the graphics card. DDR2 memory is more flexible than its predecessor, allowing (for instance) a pair of 256MB DIMMs to be paired with a single 512MB DIMM to run in interleaved dual-channel mode, and the memory controller supports a bandwidth of up to 8.5GB/second. Only the 925X supports ECC (Error Correction Code) RAM for extra security, but the 915 chipset supports a slower 533MHz in addition to the 800MHz System buss, and slower (and cheaper) DDR400/333 memory as well as the faster DDR2 533/400 memory. In practice, most motherboard manufacturers will only provide DDR2 support with the 925X chipset and partner the 915P with DDR, for a significantly cheaper product. The PCI Express x1 expansion slots use a high-speed serial buss and are only about an inch long, but offer double the bandwidth of a standard PCI socket. For graphics work, multiple 'lanes' are used to increase this bandwidth still further. The 915/925 both support PCI Express x16, using 16 lanes for up to 4GB/second bandwidth to and from the graphics card (ideal for Microsoft's forthcoming Longhorn operating system). Both chips still support up to six PCI slots, but AGP support has been abandoned. Four SATA-150 ports and one PATA/100 IDE controller are available for attaching drives, and there's Intel's Matrix Storage Technology (MST), which supports two new features. One is NCQ (Native Command Queuing), which executes hard-drive commands out of order if this results in fewer disk rotations or seeks to find particular data, but it's only currently supported by Seagate's Barracuda 7200.7 series. MST also lets you use RAID 0 and RAID 1 on the same set of drives — so, for example, in a twin 200GB-drive setup you could create a 100GB RAID 0 (Stripe) array to achieve more simultaneous 24bit/96kHz audio tracks, and devote the remaining space to a RAID 1 (Mirror) volume for extra security of other data. Intel's Wireless Technology is also built into both chipsets, but sadly no-one has yet been able to test this aspect of them, because Intel don't yet have the feature working. Finally, Intel's new HDA (High Definition Audio) replaces the nowarchaic AC97 standard with support for up to eight channels of audio in formats of up to 24-bit/192kHz on the motherboard. However, before you get too excited file:///H|/SOS%2004-09/New%20Intel%20Chipsets%20%20Good%20For%20Musicians.htm (2 of 5)9/26/2005 12:08:04 AM


about not ever having to buy a soundcard again, it's best to bear in mind that a typical HDA motherboard codec will only offer A-D (recording) conversion with an 85dB signal-to-noise ratio. Most soundcards designed for musicians manage signal-to-noise ratios of at least 100dB, giving less background noise on recordings.

PC Snippets Since I last looked at version 4.0 of the versatile universal sample converter CDxtract, in SOS October 2002, it's had quite a few incremental updates, improving the various conversions and adding new destination formats on the PC version — such as SampleCell, the new EXS24, Kontakt, Kurzweil, Reason NNXT, NI Battery, and VSampler 2 import. Now version 4.2 adds several significant new options, including NI Kontakt and Recycle import, export of all supported formats to Muon's DS404, a new DropSound utility for playing WAV and AIFF files, and further enhancements to the existing translations.

CDxtract 5 is already in development and promises a completely new interface and several new features when it appears at the end of this year as a free update to registered users. I don't normally advertise products until I've seen them with my own eyes, but this case is an exception, since the developers are actively seeking suggestions for improvements or new features. Check out www.cdxtract. com. I reviewed some Tracker software in my PC Music Freeware Roundup in SOS July 2004, and now I've discovered yet another contender. Skale Tracker is based on the Fast Tracker 2 environment, runs under Windows 98, 2000, XP and Linux, and goes beyond many other trackers in supporting MIDI, DirectSound and ASIO soundcard drivers, plus VST Instruments and effects. It also includes a sample editor and an attractive graphic mixing console. If you like the traditional tracker environment but want to take your music further, Skale Tracker looks ideal, especially as it's free! Many thanks to SOS reader Laurence Gillian for the tip-off. More details at www.metamacro.com/awezoom/skale/ I reviewed Iomatic's Registry Medic in last month's column, and since then I've found another related utility that I can recommend. Regseeker not only cleans the registry, but also lets you examine such things as startup entries, colour schemes and application histories, as well as providing a set of OS tweaks like those of Microsoft's TweakUI. However, for me its most useful function is 'Find in registry'. This lets you enter a word or phrase and then displays every reference to it in a single list. This is far easier to use than Regedit's simple Find function (which displays one reference at a time), and is ideal if you're trying to remove references to an old soundcard.

After I had removed Audiotrack's Maya 44 and its drivers from my review partition, Regseeker still found 62 'Maya' matches, most of which had not been added by the soundcard itself, but were references to it in Cubase and Pro-53's lists of ASIO, DirectX and MIDI driver entries. Overall, I didn't find Regseeker quite as thorough as Registry Medic at cleaning the registry, but every cleaner I've tried seems to find a slightly different list of problem entries, so it complements the latter very well. The best news is that it's freeware. Find it at www.hoverdesk.net.

What Does It All Mean? So what are the implications of all this for the PC musician? Well, as I said way back in PC Notes November 2002 (when first discussing PCI Express), we won't file:///H|/SOS%2004-09/New%20Intel%20Chipsets%20%20Good%20For%20Musicians.htm (3 of 5)9/26/2005 12:08:04 AM


have to abandon our PCI soundcards just yet. Most motherboards featuring the two new chipsets generally have four PCI slots and two PCI Express ones, or just two PCI slots and one PCI Express slot, plus the new PCI Express 16x graphics slot in both cases. This is understandable, since there are so few PCI Express cards yet available. The main fly in the ointment is that initial test results with the first 915/925 motherboards are disappointing. Their performance is equalled in many cases by an Intel 875 and P4 Northwood, and they're soundly trounced by AMD's new Athlon 64 3500+, 3700+, and 3800+ models. The 925X does out-perform the 915P by some one to three percent because of its DDR2 RAM, but this option is considerably more expensive considering the small increase in performance. A problem with the Grantsdale chipset also resulted in Intel recalling it just a week after its launch. Moreover, the Prescott core also runs much hotter than the Pentium 4 Northwood series, so for the time being, at least, if you want a quiet PC for audio work, a 915/925 model won't be suitable. As always, it pays to stay on the sidelines until the teething troubles are thrashed out. I suggest at least six months.

Changing Drive Letters Do you find that your various drives and partitions seem to be assigned drive letters in an almost random order, so that you're often left searching up and down the list for a particular drive in the left pane of Explorer? This can become particularly annoying with a multi-boot PC, since you may find that particular partitions get a different drive letter in each boot, so they change order in Explorer's list. And have you ever thought how convenient it would be to always have your CD drive appear in Explorer below your various hard drive partitions, rather than wherever Windows thinks fit to place it? Well, you can do all this, courtesy of the Disk Management extension of Windows XP's Computer Management tool. Launch the Computer Management tool. Do this from Control Panel, by clicking on the Administrative Tools applet and then on Computer Management, or by launching Windows XP's Run dialogue, using the Windows-R shortcut, and typing 'Compmgmt.msc' (but without the quotes) into its text box and pressing Return. Next, click on Disk Management in

Finding your data could be even easier once you abandon Windows' drive letter allocation and choose your own, as shown here.



the Storage section, and by default all your drives will appear in the right-hand pane as both a list of volumes and a graphical view beneath. To change the volume letter (and hence its position in Explorer's list) simply right-click on it in either view and choose 'Change Drive Letter and Paths...', click on the Change button in the window that appears, and then choose another currently unassigned drive letter from the drop-down box. If you want to swap the letter with another existing volume, just assign it another temporary unused letter, change the other drive to the now released letter, and then change the letter of the first to your desired letter. You can't change the drive letter for your system or boot volume, as this must always remain as 'C:'. In addition, 'A:' and 'B:' are reserved for floppy drives, but otherwise the sky's the limit. I find themed letters are best, such as 'P' for Projects, 'S' for Samples, and 'U' for Updates. My CD drive is 'Z:', so it always appears at the bottom of the list, and if you have any removable drives, such as external Firewire or USB models or memory cards, these are easier to spot near the bottom too. You won't be able to change the drive letter on a partition if it houses a page file, although of course you could change the location of the page file first. Also, watch out for any applications that rely on specific drive letters and paths to find associated files — while it's normally fairly easy to relocate audio files belonging to a particular song if they've moved, some apps — such as the Spectrasonics soft synths — lose track of their huge data files if you change the drive letter of the partition involved, and show an error message when you next try to use them. However, Spectrasonics have now provided the SpectraMove utility on their web site (www.spectrasonics.net) to take care of problems like this for PC users. It lets you move the data safely while preserving Registry entries, or repair them if you've changed the drive or path name. Published in SOS September 2004 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



OS X Tiger, Cinema Displays and new iPods

In this article:

ADC — RIP The Fourth Generation The Eye Of The Tiger More Core

OS X Tiger, Cinema Displays and new iPods Apple Notes Published in SOS September 2004 Print article : Close window

Technique : Apple Notes

New Cinema displays, a new operating system upgrade, and a new iPod revision — all in this month's all-new Apple Notes column. Mark Wherry

They say that size isn't everything. They probably weren't talking about computer displays, though, as demonstrated by the unveiling of Apple's new 30-inch Cinema display at the recent Worldwide Developer's Conference (WWDC) in San Francisco. Apple's Cinema display line-up had long been both rumoured for and due a revision. The new models sport a Photos courtesy of Apple. redesigned enclosure with a slim, Apple's new line-up of Cinema Displays anodised-aluminium casing that has feature a new anodised aluminium casing the advantage of not only making the and are available in three sizes: 20-inch, 23inch, and a new 30-inch model (above) displays look rather good — the old offering a resolution of 2560 x 1600 pixels. Cinema designs always looked a little aesthetically challenged standing next to the Power Mac G5 — but also makes it more practical to stand multiple displays alongside each other without huge bevels creating a large, annoying break between LCD panels. The 30-inch Cinema display offers an impressive resolution of 2560 x 1600 pixels, which, according to Apple, is useful for displaying 126 Logic mixer channels at once. I wonder how many narrow-view Pro Tools channels that equates to? And, in addition to the 30-inch model, the new Cinema display range includes a 1920 x 1200 23-inch model, and a 1680 x 1050 20-inch model.

file:///H|/SOS%2004-09/OS%20X%20Tiger,%20Cinema%20Displays%20and%20new%20iPods.htm (1 of 5)9/26/2005 12:08:09 AM


Getting back to the design of these new displays, the monitor itself is mounted on a stand that allows the screen to be tilted forwards and backwards, and this stand can be detached to enable a VESA-compatible (and matching aluminium) mounting panel to be attached, so that you can fix the Cinema display to a variety of other surfaces. In keeping with the USB connections found on previous models, the new enclosure offers two USB 2.0 ports and two FireWire 400 ports.

ADC — RIP One of the biggest changes in the new range of Cinema displays is Apple's switch back to implementing a standard DVI interface instead of the proprietary ADC (Apple Display Connector) interface used on the company's displays for the last few years. One of the advantages of ADC was that the interface carried video, power and USB to the monitor in one cable, and although this isn't possible with DVI, the one-cable advantage isn't completely lost. There is still one cable that runs from the monitor, but it's effectively a snake that contains separate video, power, USB and now FireWire 400 connections. The new Apple Cinema displays can be used with any computer with a DVI connection, which means that the market for DVI-to-ADC conversion is going to be drying up shortly... The majority of DVI implementations on most graphics hardware available today, and every Mac sold up to this point, are based around the Single-Link standard, which officially supports a maximum bandwidth of 165MHz to give 1920x1080 at 60Hz (or 1280x1024 at 85Hz), even though most displays and graphics cards can usually push this to 1920x1200. Clearly, this wouldn't be enough to support the 2560x1600 resolution of the new 30-inch Cinema display, which, instead, requires Dual-Link capable DVI graphics hardware. Dual-Link DVI uses the same type of connector as Single-Link DVI (utilising all 24 pins instead of the 12 actually used by a Single-Link DVI signal), but, as the name suggests, carries two 165MHz signals for resolutions of 2048x1536 at 60Hz (or 1920x1080 at 85Hz). Obviously, Apple's 30-inch resolution is still in excess of these, which suggests the refresh rate must be more in the range of 50Hz. As mentioned earlier, because all the DVI ports on current Macs are of the SingleLink DVI variety, anyone wishing to use a 30-inch Cinema display with a G4 or G5 Power Mac (Powerbook users are unfortunately left behind here) will need to purchase a new graphics card. Although there are many Dual-Link DVI-capable graphics cards available, Apple have worked with Nvidia to provide a version of the GeForce 6800 card for £449, which is actually the first time such a card has been available at this price point. However, even more interesting is that this card offers two Dual-Line DVI connections, which means you can actually hook up two 30-inch Cinema displays for a total resolution of 5120 x 1600, or, as Apple like to point out, eight million pixels!



The new Cinema displays are expected to be available in August and will retail at £999, £1549 and £2549 for the 20-, 23- and 30-inch models respectively. The Nvidia GeForce 6800 Ultra DDL graphics card costs an additional £449 for those purchasing a 30-inch display, and the VESA mounting panel costs £19.

The Fourth Generation With a suitably modest tagline — "the best just got better" — Apple introduced a fourth-generation (4G) iPod this month. The new iPod is a successor to 'the big white iPod' as opposed to the recently introduced Mini iPod. This fourthgeneration device draws on many of the ideas first seen in the Mini iPod, such as the Click Wheel, which incorporates the iPod's navigation buttons into the touchsensitive wheel previously used only for scrolling. The 4G iPod is also marginally lighter and thinner, but perhaps the most welcome change will be Apple's attention to addressing the issue of battery life. The fourth-generation iPod Although the 4G iPods still don't offer as incorporates many of the ideas from many hours of listening as some the Mini iPod, in addition to offering competitors, the increase from eight to 12 improved battery life. hours will certainly be welcome. Another interesting point about the new iPod is that it's now completely cross-platform, with both Firewire 400 and USB 2.0 connectivity straight out of the box. You can now also charge it from the USB 2.0 port, as well as the Firewire port. The new 4G iPods retail at £219 and £299 for the 20 and 40GB models respectively. And while iPods are still more costly than some of the competitors, these new models do represent good value when compared to the previous generation, and really make you wonder whether this will reduce the previously high demand for the Mini iPods. For around £40 more the 4G model offers 16GB extra storage and four more hours of theoretical battery life in an enclosure that's hardly any bigger. Still, I think I might stick with my 3G model for a little longer...

The Eye Of The Tiger As promised before the event, the WWDC also saw the first public demonstration of the next major release of Mac OS X, 10.4, or — to use the codename — Tiger, which isn't expected to be publicly available until sometime early next year. Although this year's keynote featured a brief demonstration of Native Instruments' Guitar Rig amp simulator, compared to last year's Logic/Mac vs. Cubase/Windows shoot-out, relatively little focus was given to music or audio. This also extends to the features demonstrated for Tiger, but although there were



no direct music or audio-related announcements at this time, there are still operating system issues that will offer improvements to musicians and audio engineers when using Mac OS 10.4. Under the bonnet, Tiger features increased support for 64-bit memory addressing, allowing 64-bit applications running on the G5 to address vast amounts of virtual memory — 16 exabytes, apparently, or one million, million, million bytes. This will be great for high-performance computing, but hopefully it will also mean that audio applications such as Logic will now be able to address more of the total available physical memory for storing sample data, as an example, past the limit in Panther.

Hear it roar: Mac OS 10.4 Tiger offers new technologies for 64-bit memory addressing, image and video processing and searching. Here you can see a forthcoming version of Safari with RSS news-feed support, a new version of iChat supporting videoconferencing with up to three extra parties, and the Spotlight searching technology.

Tiger also introduces support for socalled Fat Binaries, which can simplify software distribution by allowing for an application to contain both 32-bit and 64-bit binaries, with the operating system executing the appropriate code. Longtime Mac users should remember a similar technique being adopted during the transition from 68k-based code to Power PC back in the mid-'90s. And, to assist developers in porting 64-bit code from other UNIX-based platforms, Tiger supports the LP64 programming model, which offers good interoperability between 32-bit and 64-bit environments. Apple's preliminary information suggests that many low-level performance improvements can be expected in Tiger, but one specific improvement concerns optimised kernel resource locking. Such locking basically means that multiple processors can now access different parts of the kernel (the heart of the operating system) simultaneously, which will offer increased SMP (Symmetric Multiprocessing) performance, hopefully equating to improvements for applications that run audio-processing algorithms across multiple processors simultaneously, either on G4 or G5-based systems.

More Core Apple's Core Audio technology, which handles all of the low-level issues of getting audio into and out of the computer, plus some processing issues, has been a success, as has the implementation of Audio Units, providing a way for developers to add algorithms that can bring extra functionality to any Audio Unitscompatible host application. At the WWDC, Steve Jobs announced that Tiger will include similar technologies for digital image and video-orientated developers file:///H|/SOS%2004-09/OS%20X%20Tiger,%20Cinema%20Displays%20and%20new%20iPods.htm (4 of 5)9/26/2005 12:08:09 AM


and applications. Core Image is a technology designed to make it easy for developers to work with image processing that's fully optimised to take advantage of the processing abilities of modern graphics hardware, without requiring specialist knowledge. Core Image offers a plug-in architecture known as Image Units, whereby third parties can develop image-processing algorithms to work with Core Image Applications, in a similar vein to Photoshop plug-ins, or, of course, Audio Units. In addition to Core Image, Tiger also features Core Video, which does for moving pictures what Core Image does for still pictures: namely, provides hardwareaccelerated video processing, to improve the performance of QuickTime video output by taking the strain off the main computer's processor). And, staying with the video theme, the keynote also included a presentation of the H.264/AVC (Advanced Video Coding) video codec (aka MPEG-4 Part 10), which offers higher-quality images at lower bit rates. Because H.264 is part of QuickTime, any QuickTime-based application can benefit from it — such as iChat AV. Jobs demonstrated a new version of this (which now supports video conferencing with up to three other parties simultaneously) at the WWDC. The other new technology demonstrated for Tiger was Spotlight, a new way of searching for files and data on your hard drive and within apps. In the preview shown at the WWDC, a new magnifying-glass icon appears at the top-right of the menu bar. When clicked on, this displays a search field that uses metadata to trawl through data upon receipt of a search string. Conceptually, this works like searching for songs in iTunes — a comparison made by Jobs — but it doesn't only work for file names. Spotlight builds its metadata by context and can integrate with compatible applications, so if you're looking for all the information on your computer regarding a particular person, for example, Spotlight can pull together related files, emails stored in Mail, contacts stored in Address Book, and so on. It's a simple idea, but it seemed effective, and it will be interesting to see how third parties build Spotlight capabilities into their applications — especially if musically-useful metadata is incorporated. Published in SOS September 2004 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



Recording Violin & Piano

In this article:


Two Different Recording Pro gear vs. Project Studio gear Systems Published in SOS September 2004 Hugh's Recording Setup Placing Performers & Setting Print article : Close window Up Mics Technique : Recording/Mixing ORTF Or M&S Miking? Getting Down To Recording Rough Mixes & Editing Decisions Mixing & Editing Recording chamber music in Birmingham's Symphony Comparing Approaches hall is not something you get to do every day, so we

decided to make the most of the occasion by using two completely different setups, one based on professional equipment and one on project-studio gear. Paul White

A few months ago, I was approached to see if I would be interested in recording the young violinist John Garner, and I was asked if I could suggest a good location. As John was to be accompanied by piano, I said that the prerequisites were a room with acoustics suitable for the violin and also a good-quality, well-maintained piano. John's father Frank clearly thought that John's playing was something special, but I really had no idea what to expect from a lad who'd only recently turned 14, so I was taken rather by surprise when I heard that they'd hired Birmingham's prestigious Symphony Hall for the day-long recording session! To give you some idea of how serious this is, the normal hire rate for a day in Symphony Hall is around £3000, though they had managed to negotiate a discount because the day in question fell between two concerts, so the hall probably wouldn't have been used on that day anyway. I think the management also liked the idea of what John (a local musician) was doing, and wanted to offer what encouragement they could. Given that this was starting to look interesting, I asked Technical Editor Hugh Robjohns if he'd like to get involved. Also, because of the expense of hiring the Hall, the accompanist, and a standby piano tuner, I felt it would be safer if we file:///H|/SOS%2004-09/Recording%20Violin%20&%20Piano.htm (1 of 10)9/26/2005 12:08:14 AM


took along two completely independent recording systems so that if there was a technical problem with one system, we'd still have the recording made on the other. Unsurprisingly, Hugh was very enthusiastic, because there's nothing he likes better than pointing expensive microphones at even more expensive instruments in a prime acoustic location!

Two Different Recording Systems Hugh has a Genex hard disk recorder, which, despite its limited SCSI internal drive size, is ideal for location recording where you don't need a huge number of tracks — see the 'Hugh's Recording Setup' box for more details. For my own system, I used an Alesis HD24 running at 44.1kHz/24-bit resolution and teamed this with an Alesis 3204 mixer, which is surprisingly quiet and transparentsounding, as well as having full in-line monitoring. At Hugh's suggestion, I also packed a pair of Mackie HR624 active monitors so that the performers could listen to the playbacks. Although it is perfectly possible to mix and record into stereo on location, there's no opportunity to redress the balance between instruments later if the client requests it, and there's also no way to experiment with time delays to compensate for the different mic locations and distances unless you take a digital console (with individual channel delays) along and then hole up with a pair of reference monitors in an isolated room that you can use as a recording control room. Coming from a multitrack background, I wanted to keep my options open while at the same time using fairly standard miking techniques for this kind of project.

Paul and Hugh set up their separate recording rigs just off the stage in the wings. Paul recorded through an Alesis analogue mixer to an Alesis HD24 multitrack recorder, while Hugh recorded to his Genex highresolution multitrack recorder via Focusrite and GML preamps.

When we arrived at Symphony Hall just before eight o'clock in the morning, my car packed to the roof with our combined recording systems, the designated parking place had indeed been reserved as promised and the security staff couldn't have been more helpful in showing us around, helping plan the easiest route to move the gear and finding trolleys for us. Having dealt with so many disinterested 'jobsworth' types in the past, this came as a refreshing change! Symphony Hall is a relatively new concert hall which was designed to have a very low noise floor. This was achieved by sitting the entire hall (around 35000 tons of it) on 2000 rubber pads (50 feet thick) as a completely separate room within the International Conference Centre complex. (Rumour has it that cats are used to track down rats that might try to eat the rubber!) It also features file:///H|/SOS%2004-09/Recording%20Violin%20&%20Piano.htm (2 of 10)9/26/2005 12:08:14 AM


mechanically variable acoustics, with vast echo chambers behind huge doors, so that it can be used for both classical and contemporary music performances. It had been set up in classical mode for our session. We needed a place to set up the recording systems where we wouldn't get in the way of the performance, but where we could still see what was going on. After wandering around backstage looking for areas that didn't sound too live, we eventually settled on using the wings to the right of the stage, where there was also clean audio power and tables we could use for the gear. This meant we looked out directly onto the right-hand side of the stage and could keep the microphone cable runs to a sensible length. We finished rigging everything just after nine o'clock — about an hour after we had unloaded the car, which was about the same time that the piano tuner finished as well.

Hugh's Recording Setup My recording background is rather different to Paul's, and my normal approach for a project like this would have been to use a backstage dressing room as a temporary control room, making critical mixing decisions using known loudspeakers and recording straight to stereo. Of course, good communications between control room and stage are essential when working this way, and so I have built up facilities for loudspeaker talkback, a telephone, and cue lights with cables to enable the control room to be located up to 50 metres from the stage. A simple black-andwhite TV camera and monitor can also be used to see what's happening on the stage if needed, but I often find the pictures can become a distraction. However, as this was primarily Paul's project, and a suitable room hadn't been booked in advance, I went along with his way of doing things, which was to set up at the side of the stage, multitrack everything, and sort it out later on. I don't like making mixing decisions on headphones and, although Paul decided to bring his little Mackie monitors along, it wasn't possible to mix while recording, because we were located at the side of the stage and so had to turn down the speakers while recording. Therefore I used my Sony MDR7509 headphones to try to judge the quality and perspective of each mic (or mic array), recording to multitrack to increase the probability of getting something suitable. For location work I use a Genex 8500 recorder which can record up to eight tracks at 16- or 24-bit resolution at sample rates up to 192kHz. The recording medium is either an internal SCSI hard drive or internal MO drive, and external SCSI drives can also be connected if required. Had I been recording a stereo mix I'd have worked at 24-bit/88.2kHz, but as I needed to record a full eight tracks for the entire day's recording, the limited disk capacity restricted me to 24-bit/44.1kHz. I actually recorded over three hours of material across eight tracks, and had to resort to two sides of MO disc once the hard drive was full!

file:///H|/SOS%2004-09/Recording%20Violin%20&%20Piano.htm (3 of 10)9/26/2005 12:08:14 AM


I have GML 8300 and Focusrite ISA428 four-channel mic preamps. The Focusrite is fitted with the optional A-D card, and the GML outputs are hooked in to use channels five to eight. The eight-channel AES output was then hooked directly into the Genex's AES inputs. Recording levels were set so that the highest peak was about -4dBFS, with most tracks peaking around -8dBFS. Hugh Robjohns

Placing Performers & Setting Up Mics For my rig, I opted to use a Soundfield SPS422B mic (set up to behave as a coincident cardioid pair) with its controller box plugged directly into the recorder as the main stereo mic. Initially I set this up around three rows of seats back from the stage, with Sennheiser cardioid (MKH40) and omni (MKH20) smalldiaphragm mics spot-miking the violin around three feet above and just in front of the player. I used both because I wasn't sure which would sound best in such a lively acoustic environment, and I had plenty of tracks so I could leave the decision as to which to use until mixing. For the large Steinway concert grand piano I used a pair of very inexpensive SE1 small-diaphragm cardioid mics (I wanted to see how these would rise to the challenge) spaced a little over a metre apart and aimed into the piano from a little under a metre away. All the mics were mounted in shockmounts. We decided on a 'half stick' position for the piano lid after hearing how loud the piano was with the lid fully open! Because of the volume of the piano, we tried to get a reasonable distance between the piano and the violin without compromising their line of sight, so as to give us adequate separation. When we arrived, the piano was to the left of the stage and John was standing to the right looking at the pianist, Jo Sealey, which meant that his violin was pointing towards the back of the stage. This was clearly less than ideal from the point of view of the front-of-house stereo mic, so we moved the piano to the right and angled it slightly so that John could stand on the left of the stage facing outwards, while still maintaining good Both Paul and Hugh chose Sennheiser eye contact with the pianist. John MKH20 and MKH40 mics for close-miking the violin. However, Paul used a widely ended up standing just under five spaced pair of SE Electronics SE1 mics for metres to the left of the piano, which homing in on the piano, in contrast to the was about four metres back from the closely spaced Sennheiser MKH20s which front of the stage. John was further Hugh opted for. forward, and about 1.5m from the stage front — which helped to balance his relatively quiet violin against the powerful piano. John played two violins, file:///H|/SOS%2004-09/Recording%20Violin%20&%20Piano.htm (4 of 10)9/26/2005 12:08:14 AM


mainly his own J B Collin-Mezin dating from 1895, while the other was by Kudenowski, a local Birmingham maker. The Kudenowski was used for only one piece to avoid a 'wolf' note present on John's own instrument. Classical miking often involves the use of larger mic stands than usual, as normal stage stands tend to become very unstable when used at their maximum extension. Professional mic stands of this type are quite expensive, but I found a cheap solution by buying a pair of lighting stands from Adam Hall Supplies in the UK and then slotting the inner sections of regular boom stands into the tops of them. This gave me all the height and stability I needed for around £20 per lighting stand. Once we'd both set up our microphones and tested them (see the 'ORTF Or M&S Miking?' box for more details of Hugh's miking choices), we noticed an audible hum in the building which was traced to some overhead work lighting. A technician turned up within minutes and killed the offending lights, after which we did a short test recording to set the levels and to confirm that everything sounded OK. Hugh then dutifully measured and recorded all the distances between mics and instruments, just in case we wanted to delay the close mics to timealign them with the distant stereo pairs when it came to the mixing. After recording the first two or three pieces (with the Although Paul's Soundfield levels peaking at around -10dB in my case), we mic was placed only three took a short break and I listened to my playback rows from the front of the stage, it was found to be more critically. I concentrated particularly on the picking up too reverberant a Soundfield mic tracks, as I was planning for these sound, so it was moved to be the foundation of the recording, with the spot closer to remedy this. mics used to add focus and to help balance the violin and piano. On the whole, I was very pleased with what I heard, though I felt that the Soundfield was picking up an excessively reverberant sound. To address this, we moved the Soundfield mic a couple of rows closer to the front of the stage, and this produced a more satisfactory wet/ dry balance, necessitating less use of the spot mics in the mix. It really surprised me how reverberant the sound became only a little distance from the stage. Hugh had set up an M&S pair at about the same distance as my original Soundfield position and was finding the same thing, but he'd also set up a spaced stereo pair closer to the front, which is what he mainly ended up using in his final mix.



ORTF Or M&S Miking? Because I didn't want to have to make critical decisions about mic placement over headphones, I wanted to keep as many options open as I could, so I miked accordingly. My basic plan was to use an ORTF pair of closely-spaced largediaphragm cardioid mics, in this case a pair of Microtech Gefell M930s, to provide the main stereo image. Wandering around the stage and hall as the musicians warmed up led me to believe that a position about two rows back would give a good balance of direct sound and reverberation. The Symphony Hall's acoustic is quite superb, with a lovely smooth decay. I placed the mics on a stereo bar to provide the required 17cm spacing, and relied on the engraved markings on the mic's own stand clips to get the 55-degree offset angle correct. I then fixed these to one of my K&M tripod stands, and raised the mics to a height of about 3.5m. The stage itself was a little over one metre high, so the ORTF pair ended up about a metre above the level of the violin, and roughly three metres away. As a flexible fall-back option I also rigged an M&S pair of Sennheiser MKH30 figure-of-eight and MKH40 cardioid mics in a bespoke Rycote suspension mount. I recorded the raw output of these mics on the Genex and decoded them to normal left-right stereo back at base using my Yamaha DM1000 mixer. I placed this array on another tall K&M tripod stand about 1.5m further back, and about an extra 0.5m higher. Listening to the output of the Middle signal, it sounded a little too 'wet' on the headphones, but often things can sound drier over loudspeakers, so I decided to go with it. The Sides output provided a glorious reverb and ambience, which I was very pleased with. In addition to these two stereo arrays, I decided to add some closer mics, in case I required some additional definition in the mix. I covered the very fine piano with a pair of Sennheiser MKH20 omnidirectional mics spaced on a bar about 40cm apart. I find omni mics are invariable the best choice with a nice grand piano, and give a natural rich sound that cardioid mics just can't match, which is especially noticeable at the bass end. These were supported in a rather odd way from a standard K&M boom stand, to place the mics just below the half-open lid, looking in towards the strings from almost exactly the centre of the piano's waist. The relative distances between piano and violin meant that there was plenty of separation, even though I was using omnidirectional mics. For the violin I put up two more Sennheisers: an MKH40 cardioid and an MKH20 omni — the same as Paul. However, unlike Paul (who used separate stands), I mounted them adjacent to one another on the same stand, then moved them around until I found a position that gave what I thought was the most balanced and representative sound. That ended up being almost directly in front of the violin, about two metres away and at a height of about 2.5m. As Paul also discovered with his setup, the difference in sound quality between omni and cardioid was fairly subtle on this occasion, but listening back afterwards I ended up favouring the omni by a small margin. The cardioid rejected more of the piano spill, but the omni just sounded a touch more natural and the transients from pizzicato sections seemed slightly faster and with a better-defined edge. If you want to hear the differences between the different mic placements for yourself, there are some MP3 files available for download at www.soundonsound.com/ soundbank. I ran the two stereo pairs through the Focusrite mic preamps and the close mics



through the GMLs. As it turned out, all eight preamps ran with exactly the same amount of gain of just 30dB, and in the case of the ISA428 I used the highimpedance setting and bypassed the high-pass filters — all the mics (except the M930s) were in decent shockmounts. As I mentioned earlier, the highest peaks of the day hit -4dBFS, and these were on the close piano mics and the ORTF pair. The two close violin mics peaked at -8dBFS. At the other end of the scale, I was amazed at how low the background noise was in the Symphony Hall. In most public venues I find the ambient noise floor is typically about 50-55dB below the peak level of a modest orchestra, organ, or choral group. In the empty Symphony Hall it was easily 70dB below, so the audible reverb tail really did run on for a considerable time! Hugh Robjohns

Getting Down To Recording Over the course of the day, John, and his accompanist Jo, played for more than six hours, recording between two and five takes of each piece, as well as rerecording some short sections that could be edited in if necessary. We took meticulous notes documenting all the takes and part takes, including locate times, and jotted down the times of any passages that we thought might need replacing. At one point during the session we both heard a loud clunk in our headphones, which we duly noted, and it turned out that somebody had allowed a couple of Japanese tourists in to see the hall (despite this being a closed session), and they'd allowed a door to bang shut! We also noted any unusually loud piano pedal noises, which can often become audible in exposed sections — where necessary, these sections were redone. It turned out that Mr Garner's opinion of his son's talents was completely justified, as he played some very challenging material with a great deal of passion and maturity. He's definitely a 'big name' in the making, so it was a privilege to be involved with his first ever serious recording. John's not one to blow his own trumpet (to use a rather inappropriate metaphor!), but he only started playing the violin at the age of seven, and by eleven he had achieved grade eight — with distinctions at every grade along the way!

Rough Mixes & Editing Decisions Once I got home, I ran off rough mixes


Hugh kept detailed notes of all the takes recorded, complete with time locations and suggestions for editing. He also carefully measured the positioning of all the mics to facilitate the calculation of delays for the spot


mics during the mixing process. of the whole day's recording in Logic. All I did was set levels and match the spot mic panning to the stereo mic image. As it turned out, both the omni and cardioid violin mics sounded good, though I felt the omni's more open sound was more artistically well-suited. Instead of delaying all the spot mics, I added a negative delay of roughly 3ms per metre to the main Soundfield mic signal, then fine-tuned the result by ear. I sent these preliminary mixes to John so that he could pick the best takes and decide if any sections needed to be replaced. I received a very complementary email back regarding the sound quality, but I felt the final edit should be done using Hugh's recorded version, as he hadn't needed to move a mic part-way through the session, and we'd also be able to work on Hugh's SADiE system, which is better suited to final editing than any of the software on my system. See the 'Mixing & Editing' box for Hugh's account of what we did.

I was surprised how similar in sound our two recorded versions were, especially when you consider that I was using a small Alesis mixer while Hugh used his Focusrite and GML preamps and had better mics available for the piano. Neither of us experienced any noise problems, and the natural ambience of the room meant that no artificial reverb was needed.

Mixing & Editing When it came to post-production, I decided to do as I would for a normal recording session, and simply use the recorded Genex tracks as if they were live microphone sources. So I hooked up the AES outputs to my Yamaha DM1000 console and mixed accordingly, recording the result straight into the SADiE as a stereo track for subsequent editing. I monitored on my PMC IB1 three-way speakers with a DK Audio MSD600 meter to check levels, panning, and stereo widths, as well as to keep an eye out for LF anomalies. The Yamaha has facilities for M&S decoding built in as part of the stereo pairing options, so I ended up with the decoded M&S pair, the stereo ORTF pair, the stereo piano close mics, and a pair of violin mics. I adjusted the panning of the two violin mics to match the position of the violin in the ORTF pair (slightly left), and I also did the same with the piano, which occupied a narrow spread from slightly right of centre to three-quarters right. I also adjusted the balance of the M&S pair to match the image with that of the ORTF pair, and dialled in appropriate delays for the close mics and ORTF pair to time-align everything to the M&S pair — the latter was the most distant mic array, and was therefore the reference point. The Yamaha console makes life easy by allowing delays to be set either in milliseconds or metres, so I was able to translate my measured distances directly to delay values. In fact, I chose to increase the close mic delays by a few additional milliseconds to ensure that the main sound source was perceived as the ORTF pair — a file:///H|/SOS%2004-09/Recording%20Violin%20&%20Piano.htm (8 of 10)9/26/2005 12:08:14 AM


technique used by a lot of classic engineers. All equalisation and dynamics processing was bypassed, and the desk output was re-dithered to 24 bits to feed the SADiE. Switching between the close violin mics revealed that the omni was the preferred option, and switching between M&S and ORTF pairs showed that the M&S was a tad too wet and ambient, while the ORTF pair was, if anything, slightly too dry. So I ditched the Middle component of the M&S pair completely, and retained a little of the Sides component to add a little reverberation and spaciousness to the ORTF pair. The close piano mics brought in a little more low-end weight and clarity, which was useful, and the close omni violin mic added a touch of extra definition as well — although these close mics were a good 12dB below the level of the ORTF pair, so their contributions were subtle. Having established a mix I was happy with, I recorded the entire three hours of material into SADiE in real time, with peaks around -4dBFS again, logging each individual take in SADiE with the appropriate music title and take number to make the editing process as simple as possible. Most of the editing was very straightforward. John had generally recorded a couple of complete takes of each work until we had a master take we were all happy with. In addition, he had often recorded additional short sections to capture a perfect rendition of any particularly tricky parts — and there were a fair few of those! So the editing mainly consisted of trimming the start and end of the main master take for each work, dropping in any section re-takes as necessary. In addition, I also replaced the odd note or short section here and there where there was a better version on an alternative take — John chose some of these as well during the editing session. Of the twelve tracks recorded, few tracks required more than three of four edits in total, which is a testament to John's abilities. At a practical level, I copied all of the takes belonging to a particular work from the initial SADiE recording playlist into separate playlists for each work, and aligned each take on separate tracks so that it was easy to switch between them when looking for the best take at any given time. Once each work had been edited to everyone's satisfaction, I copied the complete piece to a new playlist to form the final CD compilation, and at this point I also raised the level slightly to optimise the final CD volume. Everything came up exactly 3dB, which maintained the relative levels and dynamics between each track, and gave a peak level of -0.6dBFS. With everything in the right order, we were able to optimise the gaps between tracks and the fade-outs (which essentially just chased the reverberation down). SADiE's mixer incorporates provision for a word-length-reduction and dithering plug-in, which I set up to provide 16-bit resolution with simple triangular dither for the final CD, and when I ran the entire CD playlist through this, the peak level rose slightly to -0.4dBFS. I then burned this version to CD-R as the finished product. Hugh Robjohns

Comparing Approaches This project was both enlightening and artistically rewarding. It was good to work with Hugh on his familiar territory to see how a BBC-trained engineer goes about the job — Hugh was clearly very much at home with this kind of recording. I also found it interesting to compare results, given that I was using a lot of very much 'project studio' kit while Hugh has an enviable collection of 'serious' mics and preamps. Critical listening revealed that Hugh's Sennheiser MKH20 Omni mics file:///H|/SOS%2004-09/Recording%20Violin%20&%20Piano.htm (9 of 10)9/26/2005 12:08:14 AM


did sound cleaner as close mics on the piano, but once the spot mics were blended with the main stereo pair the differences became less apparent and I was astonished at how well the budget SE1s held up in this difficult application. Clearly the more professional kit does sound better, but it's amazing what you can do with well-chosen budget equipment — where you put the mics is still the most important part of the process. I was also interested to see how Hugh's editing process differed from mine. When it comes down to it, the basics are pretty much the same but SADiE lends itself very well to this kind of task. However, had I done the editing, I would still have mixed all the takes to stereo first, then edited the stereo files, rather than try to edit the multitrack recording. In all this was a thoroughly enjoyable project, and I'd like to extend a special thanks to Hugh's wife Caroline for providing us with seriously high-grade chocolate biscuits throughout the evening we spent editing!




Stereo Mixdown of Projects

In this article:

Quick Tips Bounce To Disk Quick & Dirty Bounces The Final Stage Further Bouncing Considerations

Stereo Mixdown of Projects Digital Performer Notes Published in SOS September 2004 Print article : Close window

Technique : Digital Performer Notes

This month we're feeling bouncy, as we explain how to use DP's facilities to mix down Projects into stereo files for CD burning, MP3 conversion or mastering processes. Robin Bigwood

In last month's Performer Notes, I discussed the first stage of getting a MIDI and audio project out of DP and into the real world. This involves 'printing' or 'rendering' any external audio sources, such as synths and external effects processors, as audio tracks within DP so that they're all 'under one roof', as it were. It also allows any final EQ, dynamics or effects to be applied to these tracks prior to the final two-track 'mastering' stage, the subject of this month's article.

Quick Tips When you 'record in' your external sound sources and software synths as separate tracks in DP, it's possible that some won't play back at all, for no apparent reason. The prime suspect for this is nearly always the Configure Studio Size settings, in the Setup/Configure Audio System menu. Make sure you've assigned enough mono and stereo voices for your project there. Although Bounce To Disk has become a stable and reliable operation under DP4, it's still worth making a quick save of your Project before you do a bounce, as some third-party plug-ins can misbehave when asked to perform 'offline' fasterthan-real-time processing.

Bounce To Disk Assuming that you've recorded all your software and software synths, as well as external effects, to audio tracks in DP, and you're happy with your mix, the obvious way to produce a stereo version of it is to use DP's own dedicated file:///H|/SOS%2004-09/Stereo%20Mixdown%20of%20Projects.htm (1 of 6)9/26/2005 12:08:20 AM


'mixdown' feature — Bounce To Disk. This, for the most part, generates a split stereo file from any tracks in your mix being directed to a single hardware output pair or buss pair. If you have your studio's monitoring speakers being driven from a hardware output pair in DP called 'Analogue 1-2' (for example), this same output pair will be available in Bounce To Disk's dialogue box, which should then create a stereo mix that's exactly the same as what you hear through your monitors. Bounce To Disk is a process that runs faster than real time, so DP can often produce a stereo file of quite long multitrack mixes in a matter of seconds, providing they don't use too many processor-intensive effects. While this is often a boon when you're up against a deadline, it does mean that Bounce To Disk can't (at least for the time being) gather the output of software synths running in DP — you'll still need to record them as audio tracks first. And it certainly can't deal with any external audio coming in from outboard effects processors, hence the need to record the signals from these first of all, too. Before I run through how to do a Bounce To Disk, note that DP will only bounce those parts of your project that you've selected, allowing you to mix down anything from your entire project to just a couple of bars of it using a few tracks. If you're bouncing down the whole thing: Go to the Tracks Window and hit Apple-A to Select All. Then trim the selection, if necessary, by using the selection number fields at the upper left of the window, or the dedicated fold-out Selection pane in the Control Panel, and shift-clicking track names to deselect any you don't need.

DP's Bounce to Disk dialogue box. This bounce uses the hardware output pair 'Analogue 1-2' from a MOTU 896 interface as its source.

Now choose 'Bounce To Disk...' from the Audio menu. In the dialogue box that appears, make sure the Channels pop-up menu is set to 'Split Stereo (two mono files)', and choose an appropriate Source to match the outputs you've been using for monitoring (often 'Analogue 1-2' or similar). Under 'Resolution', you'd normally choose 16-bit if you were producing a mixdown ready for burning to CD or for producing an MP3 or other file for web distribution. You might choose 24-bit if your project was recorded in 24-bit and you were intending to do some more work on your bounce, or were sending it off to a mastering engineer. You'd probably only choose 8-bit if you were bouncing for very low-quality web-distribution formats, such as low-bandwidth Real Audio, for example.

file:///H|/SOS%2004-09/Stereo%20Mixdown%20of%20Projects.htm (2 of 6)9/26/2005 12:08:20 AM


The Import setting determines what DP does with the Split Stereo format file it creates. 'Add to Sequence' causes a new stereo audio track to be created in your project and the bounced file to be placed on it. 'Do not Import' is a roundabout way of saying 'Export', in the sense that DP just creates the bounced file and doesn't attempt to reference it from any of its editing windows (though you could always re-import it). The final option, 'Add to Soundbites Window', causes the bounced files to be listed in the Project's Soundbites Window, which is more often than not the most useful option, as we'll see. In all cases, after you've made the appropriate settings, and optionally defined a new destination folder for the files that Bounce To Disk produces, hitting the OK button causes the bounce to take place, and a Split Stereo file in Sound Designer II format to be produced. Split Stereo files are actually two mono files with the same filename but different 'L' and 'R' file extensions, and while they're the native format for stereo in DP, they're not widely When the Split Stereo format files that a Bounce To Disk produces aren't much use to compatible with other software, you, use the Soundbite Window's more especially more consumer-level stuff. flexible export feature instead. You'd need something like Roxio's Jam to burn a CD directly from a Split Stereo file, and if you've only got Toast or iTunes, you're a bit stuck. This is where Bounce to Disk's 'Add to Soundbite Window' option comes in handy. First, make sure your stereo bounce shows up in the Soundbites window, then select it there and use the extended exporting capabilities the window offers to produce a more widely-compatible stereo file. In the mini-menu, choose 'Export Selected Bites...', specify an export location for the file, and select from an altogether more helpful selection of formats, including Interleaved Sound Designer II, Windows WAV, or — probably most compatible of all on the Mac — AIFF. At this stage I should mention a caveat concerning the use of Bounce To Disk. Much has been written on various Internet chat rooms and email distribution lists about the audio quality of stereo files produced in this way, calling into question Bounce To Disk's ability to produce a mix that sounds exactly the same as the project played in real time. Personally, I've never had any problems with this, but if you suspect all is not well, or you're experiencing problems with third-party plugins that are not playing ball with the Bounce To Disk process, you can always do a 'real-time' bounce inside DP, though it's a lot more labour intensive than a Bounce To Disk operation. Basically, what you need to do is the following: Set up a new stereo audio track and choose as its input a buss pair that hasn't file:///H|/SOS%2004-09/Stereo%20Mixdown%20of%20Projects.htm (3 of 6)9/26/2005 12:08:20 AM


been used for anything else in your Project. Now re-route every track whose output is your normal 'monitoring' pair (eg. 'Analogue 1-2') to use the new buss pair as an output. Record-enable the new stereo track and hit Record. Although fiddly, and not as fast, there's much to commend this approach to bouncing, as you can hear the bounce as it happens, and of course you don't need to have 'recorded in' external audio sources or soft synths beforehand, as they'll play back as normal during the bounce. You'll also end up with the new 'mixdown' soundbite listed in the Soundbites window, and you can export it as an AIFF just as if you'd done an 'Add to Soundbites Window' Bounce To Disk.

Quick & Dirty Bounces Although there's a lot to be said for 'recording in' all your hardware synths and external audio sources, then using Bounce To Disk to produce a mixdown of your Project, an entirely different method stands out above all others in terms of simplicity and speed, but might seem like sacrilege to anyone who's spent the last few years perfecting an all-digital workflow. It relies on using an external mixer to co-ordinate the outputs of your audio interface(s), synths and other sound sources, as many people do (but you can still make it work if you've committed to the 'mixerless' studio concept). When you're ready to make your master recording, use your mixer to route your interface's output, along with any external synths and effects, back to a single pair of its inputs, and hence back into your Mac. In DP, create a new stereo track with the same inputs, record-enable it, and hit record. Your sequence should play back whilst simultaneously making a master recording of exactly what you can hear through your monitors.

The clear advantage of this method is predominantly its speed and ease of setting up. The disadvantages, however, are obvious, especially to Rogue Amoeba's excellent Audio Hijack Pro audio purists. By taking the trip is a quick and easy way to produce MP3 through your external mixer, any bounces from DP 4 projects. signals originating within DP pass through an additional D-A conversion stage, probably a few metres of cable, the gain stage, EQ and mix buss of your mixer, before finally re-entering through another set of A-D converters. Only you can decide whether this introduces an unacceptable amount of signal degradation, although it's obviously not much different from when everyone was using DAT machines to make their two-track masters.



If you're not using any external sound sources at all, or you always route them into DP via aux tracks, there's yet another mixdown option: Audio Hijack Pro. This OS X audio-recording application can make AIFF or MP3 recordings of the outputs of other applications, including DP. You simply set it up to 'listen' to the output of DP and start it recording, and whilst it feels a little haphazard it's a tremendously useful tool for making quick 'working' bounces as you go along, without generating more soundbites within DP.

The Final Stage If you were only ever planning to burn a CD of your project, you may already have what you need, as both Toast and iTunes will happily work with AIFFs, as will virtually any other Mac Blacktree's iTunes-LAME encoder allows audio application. But if you want to iTunes to access the LAME MP3-encoding produce an MP3 you still have work to algorithm. do, as DP can't help you in this area. MP3s can be produced in iTunes, of course, but amongst MP3 aficionados iTunes' MP3 encoding is not considered one of the best — for any given bit-rate there are different algorithms available that could give better results. One of these is LAME, and (using the donationware script from http://blacktree.com/apps/iTunes-LAME/) you can use it in iTunes. There's also Lamebrain, a pretty serious front end for the LAME algorithm that allows all sorts of encoding settings to be made and batch encoding to take place. This is also donationware and can be found at www.funkatron.com Amongst the many other options for web distribution, one of the most useful is Real Audio. Perhaps the easiest and cheapest way to produce files in this format nowadays is to use the free Real Export plug-in for Quicktime Pro, downloadable from www.realnetworks.com/products/realexport

Further Bouncing Considerations If you're using Bounce To Disk to bounce a 24-bit project down to a 16-bit file, you'll ideally want to use some dither to mask any low-level distortion introduced by the digital quantisation process, especially on classical or acoustic material. Despite what you may have heard elsewhere, DP's Audio menu Dither option does not apply to Bounces To Disk, so the easiest way to apply it is to create a new Master Fader track (if you don't have one already) with the same output pair as the main pair you're using to derive the bounce, leave its fader to 0dB, and add a bit-depth reduction plug-in. Use the simple Quan Jr if you're not ready to 'master' just yet, but consider one of the final limiter plug-ins I discussed last month (MasterWorks Limiter, Waves L2, etc) if you also want to maximise loudness. There's more on dither in DP in April 2002's Performer Notes. Published in SOS September 2004






Using SampleCell libraries with Pro Tools

In this article:

Sampletank Kontakt Mach V CDXtract Translator Pro Further Information

Using SampleCell libraries with Pro Tools Pro Tools Notes Published in SOS September 2004 Print article : Close window

Technique : Pro Tools Notes

Current Versions 6.4: HD, Accel & LE systems on Windows XP and Mac OS X. 6.1: Mix systems on Windows XP and Mac OS X. The most recent bug-fix version is v6.4cs5, which should only be installed if you are experiencing the problems this release addresses. More info at www.digidesign.com/ download/cs.

Digidesign's Samplecell products once had a huge army of users, many of whom still have well-loved sample libraries stranded in this format. We look at the options available to continue using these sample patches in Pro Tools. Simon Price

If you think of computer-based samplers today, you're probably picturing EXS24, Kontakt and Mach V, to name just a few, and the name of Digidesign is unlikely to come to mind. However, the company's Samplecell was the first mainstream product to bring professional sampling to the desktop, and was a huge success. Samplecell started out as a standalone card-based sample-playback device released in 1991. Over the years the cards moved from Nubus on the Mac and ISA on PCs to PCI, then finally to the Samplecell II Plus, which featured onboard RAM instead of empty slots. The cards featured four stereo outputs, and were controlled by a stand-alone software editor. Like modern samplers, the creation, saving, and loading of samples, instrument patches and multitimbral banks was handled by software.

These two screens show a Samplecell Bank (multi-instrument patch), a keygroup map for one of the instruments, and a modulation window.

The development of Samplecell TDM, moreover, was a stroke of genius, and

file:///H|/SOS%2004-09/Using%20SampleCell%20libraries%20with%20Pro%20Tools.htm (1 of 5)9/26/2005 12:08:25 AM


probably the first example of a synth or sampler fully integrated into a DAW. TDM hardware interfaces were added to the Samplecell cards, allowing you to ignore the cards' built-in analogue outs and route your sampler instruments straight through Pro Tools' mixer. In more recent years, the rise of virtual instruments has led to a move away from hardware, and in 2001 Digi released Soft Samplecell. This used the same editor software, and was compatible with existing patch libraries, but was host-based, and used connected audio interfaces for its outputs. Integration with Pro Tools was by Direct Connect (Digi's inter-application audio routing system before they adopted Rewire), rather than running internally as a plug-in. Unfortunately, Samplecell has pretty much had its day. Despite the advantage of being able to use existing Samplecell libraries, SSC was never quite up to the abilities of rival soft-sampler products, and has not made it to OS X. The hardware version is no better off, as it has no TDM support in Digi's current products (HD/Accel), and is physically incompatible with G5 Macs. All the signs are that Digidesign's priorities are firmly fixed on Pro Tools and control surface development, and have no plans to do anything further with Samplecell. The good news is that there are now many far more advanced samplers available, with various ways to continue using your existing Samplecell library. There are two approaches: either use a software sampler that directly supports Samplecell files, or use a translation program that can convert Samplecell files into your sampler of choice. Here's the lowdown...

Sampletank IK Multimedia's RTAS-format soft sample player appears to be Digidesign's recommended route for Samplecell users, as the two companies cooperate on a crossgrade offer. All Pro Tools systems currently ship with a copy of Sampletank SE bundled, but this doesn't read Samplecell libraries itself. Instead, you can see if you get on with it, then have the option to upgrade to the full version CDXtract can strip out all the mapping and of Sampletank XL for £232.50 at IK's modulation information from a Samplecell on-line store; if you don't have the Bank, which can then be saved in any format bundled Sampletank SE but do have you like, such as NN-XT. Samplecell, you can also crossgrade to Sampletank XL for the same price, which is £100 less than list. The smart money, however, would probably go on Sampletank L, which is the same as XL but comes with a reduced factory library. If your main goal is to use your Samplecell library, this will be of little consequence, and means you only pay £199.17 on-line. The main point of all this is that Sampletank L and XL can open and use Samplecell Banks and Patches directly, solving the problem in one easy step. file:///H|/SOS%2004-09/Using%20SampleCell%20libraries%20with%20Pro%20Tools.htm (2 of 5)9/26/2005 12:08:25 AM


Kontakt Native Instruments' sampler is so good that it's been well worth converting Samplecell patches to its NKI format using CDXtract (see below). However, the even better news is that the most recent version (1.5) can open Samplecell files without any need for pre-processing. Kontakt runs as an RTAS plug-in directly within Pro Tools, or can run stand-alone. Update to version 1.5 is 49 Euros, or just 19 Euros if you bought an earlier version after August last year. The full version is listed at £299.99 including VAT. The screen shots show an example of opening a Samplecell 2 bank in Kontakt, and the high level of compatibility with most parameters of the original patch.

Mach V Like Native Instruments, MOTU have always provided their products with a high level of Pro Tools integration. Kontakt has the advantage of being able to Mach V, the latest member of the read Samplecell II files directly. If you heavyweight soft-sampler club, is no navigate to the Bank file, and simply drag it exception and works as a RTAS pluginto the main area of Kontakt's window, a Multi will be created with the individual in in Mac OS 9, OS X and Windows instruments intact along with all their level Pro Tools systems. What's more, part and pan settings. The three screens show of the concept for Mach V was total file the keygroup, sample looping and and system compatibility, so the modulation parameters retained as closely Samplecell file format has been as possible within Kontakt. supported from the outset. This is probably your best bet if you're on OS 9, and costs £295.

CDXtract With the above samplers bringing on board such wide file compatibility, the need file:///H|/SOS%2004-09/Using%20SampleCell%20libraries%20with%20Pro%20Tools.htm (3 of 5)9/26/2005 12:08:25 AM


for conversion may be dwindling, but there are still a number of extremely useful aspects of CDXtract. This clever and well-thought-out utility runs on Mac OS 9, OS X and Windows, and can handle file interchange between just about any sample patch format you can think of. It remains useful if you use samplers that are not directly supported within Pro Tools, such as Emagic's EXS24 in Logic, or Halion in Cubase for example. In particular, you may find that CDXtract's ability to save patches in Reason's NN-XT sampler format is the killer trick. If you've got Reason (and most people have!) then this could save you from buying another dedicated sampler plug-in like those listed above. The full list that CDXtract can convert Samplecell files to is Soundfont, MESA, Akai S5000/6000, Reaktor map, Pulsar STS, Gigastudio, EXS24, Halion, Vsampler, Kontakt, Reason NN-XT, Battery and DS404. On-line purchase is 139 Euros and entitles you to free updates indefinitely.

Translator Pro This utility from Chicken Systems performs much the same conversions as CDXtract, although the Mac version is currently a bit limited, only being able to convert Samplecell to EXS24 and Halion formats. One nice feature of TP is that you can order a Special Edition that just supports a limited translation option, so if you just want to convert from Samplecell to one format that you'll be using, you only pay $79.95 instead of the usual $149.95.

Further Information Sampletank www.sampletank.com

SOS review February 2004 Kontakt www.nativeinstruments.com

SOS review August 2002 Mach V www.motu.com

SOS Performer Notes feature March 2004 CDXtract www.cdxtract.com/home.php

SOS review October 2002 Translator Pro http://chickensys.com/



SOS review January 2002 Published in SOS September 2004 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



Using Storm with Sonar

In this article:

Quick Tips The Track Menu Tweaking The Dream Machine A Perfect Storm

Using Storm with Sonar Sonar Notes Published in SOS September 2004 Print article : Close window

Technique : Sonar Notes

We bring you yet more Sonar stratagems, including further adventures in envelope copying, some handy track tricks, and advice on using Arturia's Storm with Sonar. Craig Anderton

After I described in detail, two issues ago, how to copy envelopes in Sonar, a couple of readers asked for some examples of why one would use a 'copy envelope' function. Fair enough: one example that springs to mind is a time when I recorded a lead vocal and used a level envelope to get rid of noise between phrases. I later decided to double the vocal (not just clone it, in order to get a more interesting sound). I copied the old level envelope to the new vocal, and it did a great job of cleaning up the new track — I needed to make only a few small tweaks, which was better than starting again from scratch. To give another example, I had recorded two rhythm guitar parts playing power chords in opposite channels. I wanted to add some rhythmic 'chopping' effects, so I did that for four measures on one of the tracks, then copied the envelope to other parts in the track where it would sound good. Then I copied the same It's possible to assign an existing envelope to pattern, but offset by an eighth note, a different parameter. into the other channel, so that there was an interesting panning/chopping effect. So yes, being able to copy envelopes has its uses. However, Sonar can also assign an envelope to a different parameter. Right-click on the envelope or node from an envelope, select 'Assign Envelope', then click on the new parameter you want the envelope to control. file:///H|/SOS%2004-09/Using%20Storm%20with%20Sonar.htm (1 of 5)9/26/2005 12:08:31 AM


Perhaps Cakewalk's engineers have too much time on their hands! Actually, the ability to assign an envelope to something else can be useful. I once used the Project 5 Tempo Delay plug-in for a track, and automated the feedback amount so that certain notes would have lots of echoes, but others wouldn't. During mixdown, I decided that that more delay lines were in order, so that I could have separate delay times in the left and right channels. So I deleted the Tempo Delay and inserted the Cakewalk FxDelay, which includes four delay lines. I wanted to use two of them, and apply the same feedback envelope to each. First I reassigned the existing envelope to FxDelay Voice 1 Feedback, then copied just the envelope from the track (to do this, use the Copy function but untick everything except 'Track/Bus Automation'). If you attempt to paste this to create a second envelope, though, that won't happen because you're pasting the same type of data, so it just replaces the existing data. Before pasting, I therefore reassigned the envelope once again, this time to FxDelay Voice 2 Feedback. Now, when pasted, the envelope assigned to FxDelay Voice 1 Feedback appeared in the track along with the one for Voice 2 Feedback.

The Track menu has several useful options; this shows the various ways you can sort tracks.

Perhaps a more obvious application is using something like a footpedal or modwheel controller for parameters that otherwise can only be envelope-controlled. Record the pedal or mod-wheel motion; you may want to assign it to some parameter that responds to this type of controller, just so you have some kind of auditory feedback on what the pedal or wheel is doing. After you've recorded the automation envelope, right-click on the envelope, choose 'Assign Envelope' and assign it to the parameter you want to control. On playback, the parameter will respond to the pedal or wheel messages. As you were 'flying blind' you may need to make a few tweaks, but at least you could create the changes in real time rather than having to slave over drawing envelope nodes.

Quick Tips To open a Template file, go File / Open, select the drop-down menu in the 'Go to Folder' line, and choose 'Template Files' (.CWT, .TPL). If you want to insert a track in the Console view, right-click on an empty space in the console and select either 'Insert Audio Track' or 'Insert MIDI Track'.

file:///H|/SOS%2004-09/Using%20Storm%20with%20Sonar.htm (2 of 5)9/26/2005 12:08:31 AM


The Track Menu Because you can easily access most Sonar track functions at the track itself, by right-clicking on it, the Track item in the menu bar may not get a lot of exercise. Yet it has a couple of useful options you won't find elsewhere. If you select one or more tracks in the Tracks Pane and go Track / Wipe, the data in the track(s) will be deleted, but any effects, EQ settings, I/O and so on will remain exactly as they are. Another useful option is Sort. Go Track / Sort, and you can arrange tracks based on a variety of criteria. For example, if you sort by 'Muted' and 'Ascending', all muted tracks will jump to the end of the track list. Or choose 'Selected' and 'Descending' and all selected tracks will jump to the top of the list. Note that this option is also available in the Console View.

Tweaking The Dream Machine How many of you think the Dreamstation DXi2 is a great soft synth? Hmmm...I thought so. In today's world of exceptional soft synths, the Dreamstation seems a bit dated, and the presets don't exactly show its best qualities. Yet it's possible to get some pretty groovacious sounds if you know some of the magic parameters to tweak (and did you know there's a formant filter option?). As a bonus, it takes up almost no CPU power, so you can pile on the instances. Insert the Dreamstation by going Insert / DXi Synth / Dreamstation DXi2. Call up preset 'Bass: 08-FM Synth Bass'. It sounds kind of thin, so let's add sub-bass. Click Oscillator 3's Off button so it turns On. Turn up the Osc 3 volume to 90 percent or so, Osc 3 Tune One way to get some really tough, growling to -50 percent (one octave down), and bass sounds out of the Dreamstation is to the Amplifier envelope Gain control to follow it with the FX2 Tape Sim processor. 100 percent. Gain is actually a velocity control, so turning it up means that the amp is triggered at full velocity, regardless of the actual velocity it receives. Thus, the bass sound will be at a constant level. If you want some more dynamics in the bass, turn Gain down. Also turn up the Oscillator 2 Vol control to maximum. Next, adjust the Decay parameter in the Amplifier and User Envelopes (the latter controls the amount of FM), as well as the User Envelope gain, to taste. I prefer Amplifier Decay at 80 percent, User Envelope Decay at 60 percent and User Envelope Gain at 25 percent. Let's add some filtering, too. Turn on the Filter and set Cutoff and Reso to 25 percent. Click the LP2 button for the filter mode. Set the A, S, and R parameters to zero percent, D to 40 percent and Gain to 25 percent. Now let's rough up the sound. Insert the Cakewalk FX2 Tape Sim in the Dreamstation's audio track. Set Input Gain and Rec Level to 4.95, Warmth to 10.0, Hiss to 0.00, Tape file:///H|/SOS%2004-09/Using%20Storm%20with%20Sonar.htm (3 of 5)9/26/2005 12:08:31 AM


Speed to Flat (EQ Curve doesn't matter), and tick LF Boost. Adjust Output Gain to avoid distortion — you'll have to pull it back quite a bit, like -11.0 or so. For the final touch, add a little channel EQ; I like 1.5dB of boost at 1.3kHz, with a Q of 4.8. This adds a little extra mid-range to make the bass 'speak' better in a track. You might also want to add a bit of a low-shelf boost, and re-tweak the Dreamstation's User Envelope Decay and Gain parameters.

Sounds a lot better, eh?

A Perfect Storm Back in Sonar Notes April 2003, we covered using Arturia's Storm 2 as a costeffective VST instrument rack for Sonar. Now Storm 3 is out, and although it costs a little more it now makes an even better Sonar accessory and also requires less CPU power than its predecessor. Storm 3 is Rewire 2 compatible, so there's no longer any need to use it as a VST instrument, because it has full MIDI functionality under Rewire. Insert it as a Rewire client, and Storm 3 can become an instrument expansion rack, a 'groove' pattern sequencer that plays along with Sonar, or a pattern generator that you can convert into audio tracks within Sonar. For now, let's look at the expansion rack possibilities. Upon going Insert / Rewire Device / Storm 3, if you tick 'All Synth Outputs' in the dialogue box that appears, 32 stereo channels will be added to Sonar's mixer. Within Storm 3, the Settings / Audio Connections box lets you assign outputs from individual Storm instruments to the desired Sonar channels. (If you don't need individual outs, you can, of course, tick 'First Synth Output', which adds only the two channels from Storm's master outs.)

In this song, five Storm synths have been loaded into Sonar, each driven by a different MIDI channel and having its own audio output in Sonar 's mixer. One of the instrument knobs is about to be tied to a MIDI controller via the 'MIDI Control' box.

When you play MIDI data into Sonar, Rewire directs that data to Storm. In Storm's sequencer, you can enable MIDI for each instrument individually, as well as set channels. Typically, I set each instrument to a different channel and reassign the output channel of the Sonar MIDI track driving Storm to trigger a particular instrument. If a lot of instruments are involved, it's less confusing to create a separate MIDI track for each instrument, especially if you're adding automation.

Speaking of automation, an outstanding MIDI feature is Storm's 'Learn' function, which lets you tie parameters to controllers. Just Ctrl-click on a knob in Storm, file:///H|/SOS%2004-09/Using%20Storm%20with%20Sonar.htm (4 of 5)9/26/2005 12:08:31 AM


click on 'Learn', and tweak the controller you want to use. If desired, you can record this as MIDI automation, which will drive the parameter on playback. One caution, if you're using Storm solely as an expansion rack of sound generators, is that you should clear the current pattern in whatever instrument(s) you're using. Storm is a very groove-orientated device, so as soon as you press play in either Sonar or Storm, Storm will start playing the preset patterns stored in whatever instruments have been loaded into the Storm 'rack'. However, this can also be a handy feature if you want to have some patterns going in the background as a reference. Published in SOS September 2004 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


file:///H|/SOS%2004-09/Using%20Storm%20with%20Sonar.htm (5 of 5)9/26/2005 12:08:31 AM


In this article:


Delivering The Goods Paul Sanders of Consolidated The 51st State Paul Sanders' Five Steps To Published in SOS September 2004 On-line Music Store Perfection Print article : Close window Open Standards Music Business : Online Music Other state51 Projects Consolidated Independence The New Supply Chain

Independent

In the age of iTunes, if you want to sell your music, you have to be able to get it on-line. But to do this, you need to understand a ton of Internet technology. Does this man have the answer? Simon Trask

With the advent of Apple's iTunes Music Store last year, the music industry finally got a compelling commercial alternative to the free filesharing of the peer-to-peer networks. The commercial success of the iTunes store — it recently passed its hundred millionth track sale —and the emergence in its wake of other on-line music stores has finally established digital download sales as a viable market in the minds of the majors. And the recent spread of iTunes, Napster MkII and Sony Connect from the US to Europe — or at least to part of it — has raised the profile of legal download services over here. What all this means for record labels large and Photos: Piers Allardyce small — and musicians everywhere, from major- Consolidated Independent label artists down to those of us recording in our director and co-founder Paul Sanders outside the CI offices in bedrooms — is that getting CDs pressed and East London. onto the shelves of record stores is no longer the only viable sales opportunity. And thanks to indie-label trade bodies like the Association of Independent Music (AIM) in the UK, which has negotiated a number of digital distribution deals for its label members, and the likes of CD Baby and The Orchard which have done the same for unsigned independent artists, the opportunity to earn money from sales at these stores is not only confined to major-owned labels.

file:///H|/SOS%2004-09/Getting%20Indie%20Music%20Online.htm (1 of 7)9/26/2005 12:08:52 AM


But having access to the deals is only part of the story. There's the issue of how the music gets onto the on-line stores, and this in itself can be a barrier to entry. It's not just a matter of the record label, or even the artist themselves, sending CDs to the on-line stores and having them do all the ripping and encoding work — just as you wouldn't expect Tower Records to press CDs. What's more, getting music onto the stores isn't only about ripping and encoding audio files. There's also the issue of metadata, or the data which describes the music. This is what enables you to correctly see the track names, lengths, artist names, year of release, and so on, when you're browsing stores, looking at tracks and deciding whether to download them. And if the merest mention of the 'M' word brings on a headache, take heart, there's help available.

Delivering The Goods Formed at the beginning of 2003, Consolidated Independent is a 50/50 joint venture between veteran Internet music specialists state51 and indie music news and reviews site Playlouder. London-based CI are essentially acting as the digital download supply chain equivalent of the pressing plant and the CD distributor rolled into one. Put simply, they encode the raw audio data that makes up songs, putting it into the formats required by the on-line music stores and attaching the required metadata, then delivering the results onto each on-line service with which the label has a digital distribution contract. CI themselves have contracts with the labels, not with the on-line services, but as a result of these contracts, they deliver data to a range of services, including iTunes, Napster, Sony Connect, Japanese mobile content provider Crosswarp, Entertainment UK, OD2, UK mobile provider 02, and Musicmatch. And they do all of this with their own custom-built software platform. "We have definitions within our platform of all these different services, and their file and metadata requirements, their packaging format and their delivery method," explains Paul Sanders, co-founder and director of state51 and Consolidated Independent. "We've done the engineering and built all the intelligence into our platform, so we can simplify the process hugely from a management point of view. We haven't yet encountered two on-line services that have the same metadata requirements — the number or arrangement of parameters, or even what they're called. They all have different ways of naming things, and they've all designed a slightly different format to submit everything in." The service being offered by CI to independent labels permits the labels to hand over their digital masters to the company and not worry about all the digital spadework that follows. There's still that pesky metadata to get together, but for people who don't know their metadata from their elbow, Consolidated can advise on exactly what data they need to gather and supply in order to get their music onto the on-line stores. "The digital side of the music business is a licensing business," Sanders says. "Each point of sale is an exercise of rights, and there's nothing you can do to get file:///H|/SOS%2004-09/Getting%20Indie%20Music%20Online.htm (2 of 7)9/26/2005 12:08:52 AM


around that. And just to make it a bit more complicated, each set of rules is potentially different in every country."

The 51st State Consolidated's predecessor and associate company state51 began working with music and the Internet in 1994, and developed the pioneering Raft music web site for Virgin Records in the mid-'90s. The company, who operate from a factory in Brick Lane in the East End of London, are providing the technical know-how and the digital platform for Consolidated Independent, while Playlouder are providing the marketing know-how and their connections in the indie community. Sanders says that CI hasn't approached the majors with its service: "The majors have the resources to do it for themselves. If they don't have the brainpower, they can hire it in. So we don't see much of a need there. And it's a lot less interesting to us, because if you're working for one major it's very unlikely that you'll be working for another. It's a position that state51's been in before, of being effectively an external department [to Virgin, back in the '90s]. It's more of a challenge for us now to put together an infrastructure that services an industry, rather than a single company." The idea for Consolidated Independent came about when state51 had a bit of a revamp. "We were managing a little network of record-label web sites with shopping, news and mailing lists, that sort of thing," recalls Sanders. "We had quite a good system, but it was limited, so we built a new system from the ground up, with a knowledge of what we needed to build to make it useful for record companies, artists and people who wanted to run businesses around music."

Paul Sanders' Five Steps To On-line Music Store Perfection "Make sure you know who contributed what to the music, and that all the copyrights are cleared. Supplying music into on-line stores is a licensing business, not wholesale. All your deals will make it absolutely clear that if there's a dispute, it's you that's in the firing line." "Register songs and recordings with everyone you can think of. In particular, join PPL and use CatCo, and get an ISRC for each sound recording and an MCPS licence. Most on-line services won't let you in without them." "Become a database head — but take advice before you set it all up, because there's nothing worse than being told your metadata is useless and you have to do it all again. If you're putting out CDs, remember that the on-line world also sells each track individually, and create your database



accordingly. And use proper capital letters at the beginning of titles and artist names — all caps will get chucked out, and all lowercase looks silly in a normal list." "Make a great digital master, taking into account the effect of radical compression if you can. It might well be someone else that encodes it, but you can make sure that the end result will retain its sparkle and depth if you use some clever ears at the mastering stage." "Don't forget all the other stuff, like high-quality pack shots, artist photos, biographies, and anything else that might convince an on-line service to give you a featured spot. And make sure they're in easy-to-use formats that can be simply copied onto a web site. An online store might look like an impersonal database, but its staff and customers are just as human as the guys in your local indie record store, and will respond well if you treat them that way."

Open Standards Next, state51 built a web services-based platform using open standards like SOAP (Simple Object Access Protocol) and XML (the Extensible Markup Language used to describe data, and therefore drive databases worldwide). "We need to be able to communicate using open standards" Sanders explains. "The world is moving towards communicating systems using Web Application Programming Interfaces, or APIs. Amazon do it, eBay do it, Google do it — and now we do it. Of course if we went to a small indie label and said 'We can deliver your data and music applications through web APIs over SOAP' they'd look at us as if we were aliens and say 'I just want to sell some music, alright?' But it's going to be very relevant to them in a few years' time." In fact, he says, using the system that state51 developed, a label can already do something like plug its own mail-order shop into Amazon using the web API. "If a label's using our system to manage their on-line shop, they can list and sell their products automatically on Amazon. No-one wants to do it yet, because it's too much to manage, but they'll grow into it. If labels could sell their own music directly through the Amazon Marketplace, that would be quite a strong offer. But doing it manually is very time-consuming, and very easy to get wrong. So what we did was build a system that just works, so when labels put their new release in their own on-line shop they also have the option of putting it onto Amazon and whatever other systems there are that open up. "Another example is supplying music digitally to iTunes, Napster and the like. Again it's an example of where you do need a very smart platform, because all these services have got their own particular way of wrapping up their music files, and they all demand XML."



Other state51 Projects Consolidated Independent coowners state51 also run the state51 conspiracy web site (http:// thestate51conspiracy.com), which the company describe as "a collection of like-minded businesses and partners hooked into the state51 network". Members of the state51 conspiracy site include labels like Tummy Touch, Paper Recordings, Memphis Industries and Boy George's label More Protein, and the band Lamb. The conspiracy provides web site hosting, mailing lists, forums, eThe web site set up by state51 for their commerce, text messaging, and soundnoisemusic network, or sonomu. digital distribution. If a label or artist's on-line shop meets a low minimum sales target then the web service is provided free, as state51 takes a 21percent commission from on-line and digital sales. Here the company do exclusive worldwide deals for digital distribution rights. Another state51 venture is the membership-based community site sonomu.net (http://sonomu.net/), short for soundnoisemusic network. Created and managed by state51, sonomu is "all about nurturing the sector of the community who are dedicated to underground experimental music," as the web site puts it. Designed to enable small independent labels to communicate with their fans and start selling music on the Net, sonomu includes 'microsites' with e-commerce capabilities for label members. "One of the next developments on sonomu will be to open up individual artist memberships," says Sanders. "At a certain level, it's very difficult to distinguish between an artist and a record label. We've recognised that, and also that we can accommodate artists within the sonomu framework."

Consolidated Independence Sanders says the applications for the new system were obviously going to be far broader, and state51 didn't feel that by itself it was in a position to get the most out of it. Hence the decision to go the joint-venture route. And state51 already knew Playlouder, as they had provided the technology for Playlouder's web site. "They're great at marketing." says Sanders. "They've got really strong connections with the indie community. We're small, and we have ideas, do interesting things and make great software. But they have a very different focus; to build a big indie music business. Consolidated Independent really fits into what they're trying to do as the back-end services part. And frankly, for a lot of the independent sector, things aren't possible until you've got CI. You can have any file:///H|/SOS%2004-09/Getting%20Indie%20Music%20Online.htm (5 of 7)9/26/2005 12:08:52 AM


number of great ideas about how maybe you'd like to do indie music communities, indie music radio, indie music rights aggregation, indie portals, and so on. But until you've actually got the music and the information managed in some way, you can't even start." Sanders says the new company were called Consolidated Independent to highlight the way that the indies could work together on a common platform and use it as their coordinated access point to the digital marketplaces. "I think we're going through a phase when it really does help the indies enormously to pull together. You can't really imagine it working nearly so well if they all tried to do their own thing — particularly on software, because there are very big barriers to entry there. "The way we've stuctured CI is that the labels that use the platform maintain their independence completely. They own all their data, and CI does nothing with it unless it's instructed to by the label. But also if the circumstances arise, and they agree amongst themselves, they can collaborate, knowing that they're compatible with each other. There's a de facto consolidation on the technology side, in that it's the same van that arrives at the shop with all the music in it, so to speak. Particularly when we're delivering to the digital music services, they want a very simple delivery route. So that aspect is consolidated, even though each label maintains its independence."

The New Supply Chain The new digital supply chain is essentially about enabling a CD-free distribution path for music. Ironically, then, the current starting point in the chain is typically for labels to post or bike over CDs! "Usually, we use a CD as a source for the master and take a WAV of each of the sound files, then treat the WAV as the digital master from which all the encodings are made," Sanders reveals. "We're dealing with a huge legacy of labels' existing catalogues, and it's just impractical to go back to the studio masters." However, as new-release material becomes available to the on-line services, so a new challenge arises: how to get it through the digital supply chain and into the on-line services in a timely fashion. "The speed of the supply chain is an issue, so the quicker we can get access to the digital files, the better," observes Sanders. "Then the metadata can be built up round a track as it's developed through the record labels' production and release processes, so the track can be ready to go on the Web for sale as soon as it goes to radio, or even before. That's what the digital services want. They see themselves as competing with the illegal peer-to-peer filesharing services, so they want the music in the on-line shops by the time it gets out elsewhere on the Internet."



Sanders sees new opportunities and requirements for studios as the digital supply chain becomes more integrated into the labels' thinking. "There's a speed issue, and studios can obviously be involved in that process by delivering directly into the digital warehouse," he observes. "If labels are happy with the studios delivering the master straight to us, then that's a lot of time and money saved. It's between the three parties — the label, the studio, and the digital warehouse service like us — to get together and work out how to give everyone access to what they need as soon as it's there, rather than go through this manual process and involve couriers. And studios who are thinking about the future should be thinking about how they're going to do that, and how they're going to manage it with their clients. It will be very much to a studio's advantage, I think, if they can offer different ways of delivering the master. The other aspect is that there's a big opportunity here for studios to start thinking about how to do a mix that's tailored for the digital services. I don't think the economics of mixing for each encoding format would stack up, but what you can do is try to create a middle-ground digital master that stands up a bit better to the encoding process than a CD master. And I think it's something that the forward-looking studios should be thinking about." Published in SOS September 2004 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




In this article:


A Winning Getting Your Music Heard Attitude The Letter Published in SOS September 2004 Cover Art Print article : Close window Out With Music Business The Intro Back In The Box Doing Your Research Find out what you need Cutting It your demo to stand out The Song Count Tom Flint Demo Doctorate

to do — and avoid doing — if you want from the crowd.

Ever since the dawn of rock & roll, and probably before that too, musicians have been shuffling into budget studios or borrowing recording equipment to produce demos, in the hope that they can capture on tape their playing skills, vocal talents, or songwriting abilities. Although a demo is not the same as a commercial recording, it still requires a great deal of thought and care, because it is often the one chance a person has to sell themselves and their work to its recipient. The archetypal demo is a short recording made specifically for record companies with the aim of securing a recording contract; however, there are several other main reasons for creating a demo. Songwriter demos are often just sent to publishers with the focus more on the songs than the presentation of a band's or artist's performance. Other demos are delivered to producers and managers by bands and artists who are hoping to involve a commercial production team in their project. At the very least, a demo is a good thing to take along to gigs in case someone shows interest, or to have available for the day you get talking to an important contact in the pub. While I worked in the SOS office, we ran a number of big competitions offering readers the chance to win recording time with a top producer in a professional studio. From wading through the hundreds of submissions we received then, and from subsequently putting together the Business End column, I've not only seen how the best demos stand out, but I've also seen many of the same mistakes being made again and again. In fact, most of the many hundred demos I've listened to over the years were clearly compromised by some aspect of their presentation — while the different Business End panelists often disagreed about what was good or bad, there were certain things which always managed to wind everyone up! In this feature, I'll be looking at the most common areas where demos are let down by poor presentation. Plus I'll suggest ways to improve the demo package so that it delivers the music

file:///H|/SOS%2004-09/Presenting%20Your%20Demos.htm (1 of 8)9/26/2005 12:08:58 AM


and all the important information about its creator as successfully and directly as possible.

A Winning Attitude It has been known for people to become very exasperated when they realise that their demo has not received a printed review after it's been sent to SOS, and they then begin showering the magazine with phone calls and emails rudely demanding an explanation. However, far from having a positive effect, abusive emails or phone calls tend only to reduce the chances of getting a review. Even those famously 'difficult to work with' pop stars probably spent years being nice and subservient before they got to the position where they could afford to behave like a rotter. By the same token, you might not like the fact that the A&R person has been sent one or two of your demos and hasn't called you back, but bombarding them with angry mail telling then how aggrieved you feel is not the best way forward.

The Letter All demos should ship with a covering letter, preferably addressed personally to each recipient. In general, all the necessary information can be conveyed on a single page of A4, which should include some basic information about the artists, writers, and/or musicians, their immediate aspirations, and, most importantly of all, their contact details. The letter is the vehicle through which the demo's creator can directly speak to the listener and, as such, it is an important first point of contact. Nevertheless, many people use their covering letter as some sort of advert through which they hype their demo and themselves as much as possible. Texts of this kind often go on to explain how the artist's brilliance has been confirmed by the crowd reaction at gigs, or by some other respected critic. Unfortunately (or fortunately, depending on your view), this kind of self-publicising is likely to have a negative effect on the seasoned pro, who will be reading the letter to get some basic useful information and not to hear the reported opinions of a stranger. They will also be intending to use their own ears to assess the music as they find it. In my experience of running demo review panels, brash statements usually leave the people reading them thinking one of several things: 'This had really better blow me away!' 'Full of pop hits?! Demos that say that sort of thing are invariably crap!' (Which they do tend to be, for some reason...) 'This person sounds very arrogant. I don't think I like them already.' 'So this upstart thinks they can tell me what I should


Tactics like this are almost certain to backfire — the SOS Business End panel certainly


gave this jiffy bag's contents very short shrift!

like? How dare they!'

We've all seen television and newspaper adverts for new releases which include value statements telling us how everyone who is anyone thinks the record is wonderful, so it's not really surprising that people try to emulate advertising strategies in their demo notes. But adverts use the opinions of famous people because they are targeting the general consumer market, in which music, at least partly, functions as a signifier of taste and social acceptance. Most adverts are merely trying to convince someone it's worth buying a CD, whereas the covering letter of a demo will be read by people who are looking for a project genuinely worthy of the investment of huge sums of money, so the stakes are much larger. A brief, straightforward, polite, and informative letter is infinitely more attractive than an arrogantly boastful one, and it also shows the listener that you are respectful of their position and their opinion.

Cover Art Artwork is something which is often a cause for debate. Some people believe that it is best to leave a demo's design as simple as possible, so that the artwork does not affect the perception of the music. Others stress that its aesthetic appearance can help a demo stand out from the pile of other demos which inevitably litter the listener's desk. It is certainly true that demos with no artwork are easily overlooked in favour of more interesting designs. However, sometimes great demos are left to one side just because they look terrible! Thanks to domestic CD printers, software programs like Adobe Photoshop, and digital photography, it's now pretty easy and quick to produce artwork at home. However, such technology has encouraged people with absolutely no artistic aptitude or graphic-design skill to have a go, and the results are sometimes pretty bad. Particular bugbears of mine are badly pixelated images that have been shot with a low-resolution digital camera before being enlarged, and the common use of Photoshop-style preset effects which require no effort to apply. If you even suspect that your artistic talents lag behind your musical ones, then it's probably very wise to keep things simple, or find a friend who is artistically inclined. Including a photo with the demo is also something which can work both ways. Some people like to see a picture of the artist, especially if the music is song based, or if there is a whole band involved. Unfortunately many pictures end up looking like a bit of a cliché. Common 'classics' are the 'working-class band hanging out in an alley' image, the 'sole songwriter clutching the neck of a guitar' studio shot, and the 'attractive girl, gazing into the camera from a silk-covered bed' photo. Any attempt to ape a hackneyed shot like one of these is bound to look like a cheap imitation, and you can guarantee that anyone in the industry will have seen it a hundred times before. Whatever visual package you come up with, it is essential that you make sure your contact details are included on everything. Covering letters are easily separated from CD cases, and CD cases can get separated from CDs, so full contact details must go on all three. Also, some demos sit in the system for months or years before finding their way to the right person, so



picking phone numbers and emails which have a long shelf life is vital. For example, make sure you don't rely on a web-based email contact which you end up closing down after a few months. I've often found that numbers and addresses are often so hurriedly written that they are rendered unreadable, or have digits missing. Taking a little time to type text out accurately, and check it afterwards, may make all the difference between making an important contact or not. Using arty fonts is also a really bad idea, especially for contact information, as many are unreadable or have ambiguous digits and letters.

Out With The Intro If you're having trouble paring down your intro, perhaps it would be worthwhile looking at some former hits which make their mark in just a few seconds. Most are classic three-minute hits, recorded in an era when a composer had to deliver all the elements of the song as directly and succinctly as possible. 'Wild Thing' by The Troggs starts with a single bending guitar note which instantly identifies the track. The song's main chords begin afterwards and the first lyrics are reached a mere eight seconds after the very start of the track. Despite being quite a long track, the Beatles' 'Hey Jude' actually begins with the vocal. Even so, as soon as McCartney starts singing, the vocal is distinctive enough to be recognised by anyone who's heard it before. 'You Really Got Me' by The Kinks has a very simple but effective introduction which just repeats the same chord motif several times and then continues it into the verse. Despite the repetition, the song still gets to the vocals in nine seconds. 'All Day And All Of The Night' has a similar guitar-riff introduction, but takes just seven seconds to get to the lyric. Take a look at some of the Kink's other uniquely recognisable hits: 'Dedicated Follower Of Fashion' gets to the lyrics in seven seconds, 'Autumn Almanac' takes eight seconds and 'Days' manages it in six seconds. Pick a few Elvis records and the same brevity can be observed. 'Hound Dog' begins with the vocal straight away, 'Jailhouse Rock', with its unmistakable guitar and piano chords, takes six seconds, and the epic 'Suspicious Minds' gets going in five seconds. 'Return To Sender' takes nine seconds for the lead vocals to start, but by then the song has already played a section of the chorus backingvocal part!

If you're having trouble keeping the introductions to your songs concise, try checking out classic singles like these for inspiration.



Back In The Box Exactly how the CD, its box, and your letter are packaged is not particularly important, as long as they have adequate protection from the knocks they are likely to suffer in the post. Some CDs we've received at SOS arrive without a jewel case, probably because they were sold that way. However, it's worth remembering that shops like Maplin and catalogues like Studiospares (www. studiospares.com) do sell cases separately. Many demos are only protected by a thin plastic sleeve which is only just enough to stop scratches, but not enough to guard against a hard knocks. A jewel case is the safest option, and it is also a good holdall for artwork and information. Even the covering letter can be folded and stored inside the case. Furthermore, I've found that when a CD is one of many sat on my desk waiting to be heard, a hard case makes it stackable, and harder to lose than a sleeveless one. Padded jiffy bags or cardboard CD wrappers are the best and most readily available packages for sending CDs, even though, ironically, it is often the CDs without cases which are sent in a plain envelope! Some people decide to decorate their Jiffy bags to help them stand out from the rest, and it's true that a little colour can certainly CDs and their cases are easily separated, so make sure you include your help a package get opened full contact details everywhere — on the booklet, the CD, and the case, as well as on any letters and photos. sooner rather than later. However, if time is short, it's much better spent on improving the package content, because the wrapping is the one thing which is sure to be thrown away straight after it has been opened. Amazingly, it is not at all uncommon to receive CDs with no audio on them, or ones which don't seem to play properly. To avoid mixing recorded and unrecorded CD-Rs, it's essential to have a marker pen handy to label them as soon as they have been recorded, especially if you have a desk covered in disks! It's also important to check that each CD-R plays properly, and the only way to do that is to spend time listening to each one. Unfortunately, some CD players are more particular than others, and will refuse to play CD-Rs which play perfectly well in other machines. One way to lessen the chances of this problem occurring is to buy CD-Rs with recognised brand names, rather than their cheap alternatives. Even then, it's worth testing your recordings on a range of CD players.

Doing Your Research Record-company public-relations agencies seem to be very good at sending out promotional CDs and press releases without doing any research to find out the requirements of the companies they are sending them to. SOS, for example, doesn't review commercial CDs, yet the magazine is constantly being sent commercial CDs for review! However, just because those file:///H|/SOS%2004-09/Presenting%20Your%20Demos.htm (5 of 8)9/26/2005 12:08:58 AM


people, who should know better, don't do their research, it doesn't mean that you can afford to be so careless when sending out your demo. If you are sending a demo to a magazine, find out what sort of thing they typically feature by getting the latest copy and reading the reviews that already exist. Adhering accurately to any printed entry instructions will further your cause greatly too. If you are contacting a record company A&R department, publisher, or producer, then it probably isn't going to be quite so obvious what kind of package they'd like to see. However, with a bit of research you can find out if they have any specifications, and then you can tailor your submission accordingly. The simplest method is to ring the record company or publisher and ask for the address of the correct department and the name of the person to whom you should send your work. And at this point you can ask if they offer any submission guidelines. These days, many companies place all the necessary contact information and A&R details on their web sites, so those with a phone phobia don't miss out.

Cutting It So far, we've talked about the non-musical elements of a demo package, as these are the things which are most commonly neglected or dealt with hastily after the music has been finished. However, changes can be made to the presentation of the tracks themselves, without overtly tampering with their musical integrity. Keeping your introductions as short as possible is something worth doing, regardless of the type of music you are creating. It is easy to feel that your song's dramatic intro is an essential precursor to the rest of the song, especially if you are in the habit of listening to it with the stereo pumping, atmospheric lighting overhead, and a half-drunk beer in the hand. Unfortunately, it's very likely that the majesty of your A picture of your band loitering in a alleyway is a bit of a cliché, so intro, as experienced by you at it's unlikely to reflect well on you. The SOS Business End panel certainly felt that this picture let down what was otherwise a home, will be totally lost on your genuinely interesting demo. listener who will, no doubt, be playing it either in their car on the way to work, in a busy and noisy office, or on a small stereo in the corner of their kitchen. Many people try to get round this problem by writing in their covering letter things like 'Please play loud' or 'Please listen with the lights down low', but it's wise to assume that people are going to be far too busy to do anything other than fit their listening sessions around their daily routine. Long intros also take up too much time. No matter how well orchestrated the intro, it is often expendable in terms of the song's overall composition, especially if it is repeating the main chords of the song a number of times. You can safely assume that the listener will have lots of other demos to get through and other work to do, so they will really want the song to start file:///H|/SOS%2004-09/Presenting%20Your%20Demos.htm (6 of 8)9/26/2005 12:08:58 AM


quickly, especially as the previous 100 demos they listened to almost certainly suffered from long intros too! It's often said that a typical A&R guy will listen to the first 30 seconds of a demo, and if the song hasn't started happening in that time it will be consigned to the reject pile. Some people refuse to believe that a responsible industry professional would spend so little time listening to a track, but, as you will find out if you ever try listening to a large number of demos one after another, even the most enthusiastic and caring music fan can comfortably settle into such a ruthless state of mind in a very short period of time! All styles of music follow rules and use common compositional cues to speak to the listener, so it's unsurprising that people in the business develop an uncanny ability to predict exactly what a track is going to do, just from hearing its first few notes. Long intros just tend to prolong the inevitable, and often end up getting skipped over with a press of the fast-forward button. A good way to test your demo tracks is to edit them into chunks to see exactly how much of your composition happens during any one block. If there is a minute-long section which contains all the best elements of the song then it may be wise to start the song at the beginning of that section and indicate that you have edited it as such on the sleeve notes. Bear in mind that a demo does not have to provide complete tracks — it is primarily a taster of your work.

The Song Count It's always difficult to choose which songs to include on your demo and which to leave off, especially when you have a personal affection for them all. For this reason it is often best to seek a few different opinions. Asking family and friends is not always a good idea however. Indeed, TV shows like Pop Idol and Fame Academy have proved that, no matter how bland or how out of tune the contestant appears to be, their nearest and dearest usually think that they are wonderful, and are prepared to boo and hiss anyone who says otherwise. It's much better, then, to find people who have no vested interest in your music, and ask them which tracks they find most compelling. Some people try to avoid the issue altogether by putting all their songs on the demo, usually together with a written disclaimer saying something like 'we think all our songs are great, so we couldn't leave any out.' Taking the easy way out like this is actually a very bad idea. If the recipient of your demo has only a few spare minutes to listen to each artist's work, then they are not going to welcome a CD full of music. The chances are that they will listen to the first part of the first song, and maybe a little of song two, and if these early tracks do not cut the mustard, they probably won't listen any further. What's more, presenting an album's worth of material as a demo demonstrates a lack of focus and awareness, and may annoy some A&R personnel. Three tracks is almost certainly the maximum number file:///H|/SOS%2004-09/Presenting%20Your%20Demos.htm (7 of 8)9/26/2005 12:08:58 AM

Hastily written information is not only difficult to read, but also reflects badly on


the music before the CD's even been you should include. Some people even restrict played. themselves to a single track, although it is likely that an interested party will want to be able to check out a second track if there is one available, so two is probably a more sensible minimum. Of course, if your demo does garner some interest there will be ample opportunity to showcase more of your work later on.

Demo Doctorate Many of the mistakes mentioned in this article are very easily avoided, but, because a demo is a very personal thing, most people find it hard to stay objective and business-like when presenting their work. There's certainly no way to guarantee that your music will be liked or accepted, but at least by showing some discipline and care you can ensure it stands the best possible chance of meeting its mark. Published in SOS September 2004 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



Sound on Sound (September 2004)

Sound on Sound (September 2005)

Sound on Sound (November 2004)

Sound on Sound (March 2004)

Sound on Sound (October 2004)

Sound on Sound (January 2004)

Sound on Sound (June 2004)

Sound on Sound (February 2004)

Sound on Sound (July 2004)

Sound on Sound (May 2004)

Sound on Sound (December 2005)

Sound on Sound (June 2005)

Sound on Sound (January 2005)

Sound on Sound (May 2005)

Sound on Sound (November 2005)

Sound on Sound (January 2006)

Sound on Sound (February 2005)

Sound on Sound (July 2005)

Sound on Sound (March 2005)

Sound

Sound

Sound

Sound Advice on Mixing

Sound Symbolism

Sound Insulation

Sound Doctrine

meyer sound design reference (for sound reinforcement)

Sound Advice on Microphone Techniques

Carly's Sound

Secondary Sound

Sound Reading

Sound on Sound (September 2004)

Sound on Sound (September 2005)

Sound on Sound (November 2004)

Sound on Sound (March 2004)

Sound on Sound (October 2004)

Sound on Sound (January 2004)

Sound on Sound (June 2004)

Sound on Sound (February 2004)

Sound on Sound (July 2004)

Sound on Sound (May 2004)

Sound on Sound (December 2005)

Sound on Sound (June 2005)

Sound on Sound (January 2005)

Sound on Sound (May 2005)

Sound on Sound (November 2005)

Sound on Sound (January 2006)

Sound on Sound (February 2005)

Sound on Sound (July 2005)

Sound on Sound (March 2005)

Sound

Sound

Sound

Sound Advice on Mixing

Sound Symbolism

Sound Insulation

Sound Doctrine

meyer sound design reference (for sound reinforcement)

Sound Advice on Microphone Techniques

Carly's Sound

Secondary Sound

Sound Reading

Recommend Documents