Sound on Sound (June 2005)

In This Issue June 2005 In This Issue Click article title to open Reviews People Analogue Systems RS370 & RS375 Busi...

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In This Issue

June 2005 In This Issue Click article title to open Reviews

People

Analogue Systems RS370 & RS375

Business End

Polyphonic Harmonic Generator & Expander Having built their reputation on knob-heavy modular synths, British manufacturer Analogue Systems surprise everyone by bringing out a menu-driven additive synthesis module! But don't worry — the optional expander is covered with things to tweak and turn...

Reader Tracks Evaluated Listen online to the tracks whilst reading what music industry professionals think of the latest collection of SOS reader recordings.

ART Head Amp Headphone Amplifier

Arturia 2600V Virtual Semi-modular Synth [Mac/PC] Completing the quartet of vintage synth emulations they began with Moog Modular V, Arturia's latest plug-in aims to reproduce the sound of the greatest semi-modular of them all, ARP's 2600. We see how it fares up against the original...

Circular Logic InTime Real-time Tempo Tracking Software [Mac/PC] For those who want to put a human performance at the centre of their sequenced masterpieces, Circular Logic have developed software that can understand and follow tempo changes in real time.

DPA 4041T & HMA5000

DJ Format Sampling & Hip-hop Production DJ Format has been making funky, sample-led hip-hop since the old school was new — and it seems the world has finally caught up.

Recording Daúde, Neguinha Te Amo Will Mowat Will Mowat made his name first as a sequencer expert and then as a writer with Soul II Soul. As a world music producer, his recent projects provide inspirational examples of how to combine performance with programming and overcome the limitations of a bare-bones budget.

Rock Of Ages Paul White's Leader More thoughts on the state of the record industry and landing a job in a studio...

Sounding Off

Tom Flint Condenser Mic & High-voltage Preamp Is there a correlation between pottery and soft synths? DPA demonstrate the state of the art with their latest highStudio SOS voltage designs. Rod Brakes Dynaudio BM5A The owner of an unusually bijou studio setup provides the Active Monitors chocolate biscuits this month, as the SOS team get busy Dynaudio have concentrated on high-quality internal and helping to improve the performance of his gear and the sound external engineering to create a communicative new of his recordings. ported monitor which isn't afraid to go loud. Technique

Glaresoft iDrum

Virtual Drum Machine [Mac OSX] Espousing the simple, easy-to-use Apple-type philosophy seen in tools like GarageBand, Glaresoft's virtual drum machine has you piecing together beats in minutes. And it costs just £39! We check it out...

Korg KPE1 Kaoss Pad Entrancer Audio-visual Processor & X-Y Controller

Pro Tools updates & free EQ plug-in Pro Tools Notes This month sees a high-quality, free EQ plug-in gracing the Digirack suite, some intriguing options for collaboration via the Internet, and a host of updates.

Audio Input for Reason Reason Notes

file:///H|/SOS%2005-06/In%20This%20Issue.htm (1 of 4)9/28/2005 2:25:07 PM

In This Issue

The Kaoss Pad Entrancer builds on the success of the Kaoss Pad 2, offering the same versatile X-Y pad-driven audio processing capabilities, but adding video processing to give you real-time control over both sound and vision.

Line 6 Variax Bass 700 Modelling Bass Guitar Expanding on the modelling technology used in their Variax and Variax Acoustic, Line 6 have dropped an octave and turned their attention to the bass guitar. The Variax Bass features 24 models of 'classic' instruments: is it all the bass you'll ever need?

Magix Samplitude Professional v8 Digital Audio Workstation [Win] The tools available in Magix's highly regarded recording package cover every stage of the recording process, from MIDI sequencing to mastering and CD burning. Version 8 adds some neat extras including analogue-style processors, a virtual drum machine and an Acid-style beat-mapping tool.

With some clever programming, an Italian software house have created a program that claims to add an audio input to Reason. We take a sneak peek, as well as bringing you essential Reason news and quick tips.

Better Acoustic Guitar Recording In Logic Workshop We explain how you can mock up a live acoustic guitar sound from a DI'd recording using Logic's built-in plug-ins.

Building Combinator Patches in Reason 3 Workshop The most exciting addition to the new version of Reason is arguably the Combinator device, which greatly expands the flexibility and programming potential of everything in the Reason rack. We guide you through the creation of a Combinator patch to get you started with this great new device.

Cakewalk Z3TA+ Bandlimited Waveshaping Soft Synth Sonar Notes Cakewalk go global, and release a new soft synth.

Merging Technologies Pyramix

CLASSIC TRACKS: 10cc 'I'm Not In Love'

Digital Audio Workstation [Win PC] Pyramix might be the new kid on the block as far as audio recording and editing are concerned, but that hasn't stopped it quickly proving itself a very serious rival to Pro Tools and other established DAWs.

Producers: 10cc; Engineeer: Eric Stewart Disagreement can be destructive, but it can also drive a band on to new heights. So it was when 10cc's Kevin Godley turned up his nose at a love song penned by Eric Stewart and Graham Gouldman, insisting that it would have to be completely reinvented in the studio...

MOTU Traveler Firewire Audio Interface [Mac/PC] With two market-leading Firewire audio interfaces already part of the MOTU stable, where does the new mobile recording-oriented Traveler find its natural home?

Demo Doctor

Peavey PV8

Digital Performer Notes

Reader Recordings Analysed Listen to these tracks from SOS readers and see whether you agree with the good Doctor's prognosis...

Mixer 3rd party developments This cheery new unit brings clean eight-channel mixing to Now that the excitement surrounding the launch of DP 4.5 has the cash-strapped home studio. subsided somewhat, developments by MOTU-friendly companies get this month's column under way.

Presonus Blue Tube DP

Dual Mic/Line Preamp Valve warmth or solid-state transparency? You decide with this flexible new hybrid preamp.

Pro Tools M-Powered Recording Software [Win/Mac OSX] For the first time ever, Digidesign's Pro Tools recording software is available as a stand-alone product, which can be used in conjunction with any of five audio interfaces from M Audio.

Real Traps Mondo Trap Acoustic Panel Real Traps' biggest bass trap offers heavy-duty low-

DSD support in Cubase? Cubase Notes This month we look at the possibility of DSD higher-quality audio support in Cubase, a plug-in to help you write ringtones for Nokia phones, and a new patch for Windows users of Cubase SX/SL 3.02.

Exploring Sonar 4's TTS1 Synth Workshop The TTS1 soft synth bundled with Sonar 4 has many hidden talents — but you have to know where to find them. Read on...

From 32-bit to 64-bit PC Notes


In This Issue

frequency absorption for the serious home studio.

Sample Libraries: On Test Sample Shop Four new Sample Library collections get the aural review treatment from the SOS samplists: AMG Evolved Analog **** EXS24 Chemical Synths ***** REFILL+REX2 Power FX Cinematic Hip-Hop **** ACID Sample Lab Discography ***** MULTI-FORMAT

Sensorcom Soundcheck Sound Level Indicator

Sony Sound Forge v8 PC Stereo Editing Software Sony have opened their Sound Forge editing software up to new horizons with support for VST plug-ins and the ASIO driver protocol, and improved its usability with batch processing and a new scrubbing tool.

The magic number of 64 is gradually working its way into all aspects of our PC experience: processors from both AMD and Intel, the Windows XP OS, hardware drivers, and — coming soon — 64-bit music apps. But what should we be aware of as we consider the transition from 32-bit?

Hard Drive Defragmentation PC Musician Does defragmenting your hard drives, including the ones you use for recording audio, really result in better PC performance? Opinion is divided, so we take a considered look at the subject, as well as testing some of the most suitable 'defragger' utilities.

OS X Tiger, Apple Soundtrack Pro and more... Apple Notes We offer a brief preview of the features musicians can look forward to in Mac OS X Tiger, take a first look at a major new version of Soundtrack, and examine why the iPod Shuffle is hard to resist.

Understanding Audio Files In Cubase SX Workshop We take a look at the concepts of audio files, clips, events, parts and regions in Cubase, and explain how you can manage these objects in the Pool window.

Studio Projects B1

Using Pro Tools' Keyboard Commands Focus

Condenser Microphone The B1 may seem like just another cheap Chinese mic, but it punches well above its league.

Workshop Most people find that learning keyboard commands for their DAW enables them to get things done faster, and for the advanced user, Pro Tools has a special mode which turns the entire QWERTY keyboard into a bank of one-press shortcuts.

Wizoo Darbuka & Latigo Flexgroove Software Percussion Instruments [Mac/ PC] The humble sample CD-ROM is dying out, being replaced by sample collections with a virtual-instrument front-end. Wizoo's first forays into the field combine Latin and Arabic percussion loops with their own Flexgroove virtual-instrument engine...

Using Automation In Digital Performer 4.5 Workshop If you are a user of MOTU's Digital Performer sequencing software, you have access to an extensive and full-featured automated mixing environment. For hints and tips on how to use it, read on...

Yamaha Digital Mixer Upgrades

What does Logic v7.1 offer?

v2 OS Upgrades & Effects Not content with offering some of the most fully-featured digital mixers in the world, Yamaha have now upgraded them, and are offering extra effects packs to further expand their capabilities.

Logic Notes After months of speculation Apple have finally announced the release of the first major Logic 7 upgrade, but what can users expect from version 7.1?

Competition

WIN TB2SA Powered Monitors from PMC worth £1600

Yamaha AW4416 User Tips Masterclass: Part 2 Another selection of creative and time-saving tricks for Yamaha's fully featured hard disk multitracker.

Sound Advice

Q Can you explain how the 'Track


In This Issue

Expansion Method' works? Q Can you explain the track routing in my multitracker? Q Do I need to use a compressor before my soundcard? Q Is it worth recording at a higher sample rate? Q What kind of ear plugs should I get for wearing at gigs? Q What's wrong with my patchbay?



In this article:

Socket To Me Modes Arpeggiator On The Menu The RS375 Harmonic Generator Expander MIDI/CV Conclusions

Analogue Systems RS370/RS375 £595/£285 pros Allows you to integrate a DSP polysynth into your analogue modular. Unique real-time harmonic control via the RS375. Patch memories and extensive external control. RS370 serves as a versatile MIDI interface.

cons These aren't cheap modules, especially if you purchase enough VCFs, mixers, envelopes and LFOs to make a complete six-voice polysynth. Lots of menu-hopping to set up patches on the RS370 initially. Only 46 patch locations.

summary Incorporating a softwarebased synth into an analogue modular is an excellent idea. Whether used as an oscillator bank, for real-time harmonic changes or as an additive polyphonic synth, the RS370/375 modules offer something genuinely different. True, you have to be prepared to spend some time setting things up, but with the available control routings and ability to store user patches, even this becomes less of a

Analogue Systems RS370 & RS375 Polyphonic Harmonic Generator & Expander Published in SOS June 2005 Print article : Close window

Reviews : Modular Synth

Having built their reputation on knob-heavy modular synths, British manufacturer Analogue Systems surprise everyone by bringing out a menu-driven additive synthesis module! But don't worry — the optional expander is covered with things to tweak and turn... Paul Nagle

One of the greatest strengths of an analogue modular synthesizer is its scalability — it can grow not only with your budget but also according to personal taste. With over 40 modules currently available, nobody could accuse Analogue Systems of denying their users a wealth of expansion options. There's something for every synthesist, whether it's typical analogue building blocks or more exotic add-ons. Photos: Mark Ewing

In this review, I'll be looking at two new modules that slot firmly into the 'exotic' category. They are the RS370 Polyphonic Harmonic Generator and the RS375 Harmonic Generator Expander. Together they form a sixnote polyphonic DSP-based synth, a polyphonic MIDI-to-CV interface, or a monophonic synthesizer whose individual harmonics can be modulated in real time.

For this review, Analogue Systems supplied us with a small modular system containing a selection of modules, including the RS370 and RS375 under review (on the top row of the review system). The other modules, on the bottom row, were: (from left to right) the RS210 Fixed Filter bank, the RS360 Vocal/ Phase Filter, the RS400 Phase Shifter, the RS120 Comb Filter, the RS310 Reverb/ Chorus, and the RS390 Echo.

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chore after a while.

information RS370, £595; RS375, £285; RS376, £130. Prices include VAT. Analogue Systems +44 (0)1726 850103. +44 (0)1726 850103. Click here to email www.analogue systems.co.uk

Socket To Me The RS370 is 36HP wide (as with most modular synth modules, Analogue Systems modules are measured in Horizontal Pitch units) and features a single control knob, 25 neatly-arranged 3.5mm sockets, MIDI In and Thru (of which more later) plus a small backlit screen which is navigated using Edit and Cancel controls. The screen is the focus of all RS370 operations, and it soon became apparent that, unlike most SOS modular synth reviews, writing this one would involve a lot of menu-gazing. Of the mini-jack sockets, six provide individual outputs for each of the synth's voices, although you can opt to mix them internally and route the audio to all outputs simultaneously. CV and trigger points are present, giving individual access to the pitch and triggering of each voice. There is a single control knob (Ctrl 1) which can be freely assigned to perform a selection of roles, and three CV inputs (labelled Ctrl 2 to Ctrl 4) that can open up a wealth of external control if you attach suitable CV sequencers, envelopes, joysticks and so on. Some of the sockets are named rather vaguely. For example, the three control inputs are labelled 'In' but the outputs (Ctrl 1 to Ctrl 4) aren't given corresponding 'Out' labels. To add to the confusion, the CV and Trigger sockets may be either inputs or outputs depending on the RS370's selected mode.

Modes The RS370 offers four different modes of operation. They are: 'polyphonic MIDI', 'polyphonic analogue', 'real-time MIDI' and 'real-time analogue'. The two real-time modes transform the RS370 into a monophonic sound source but by way of payback, you gain real-time control (via the RS375 module) of the first 16 harmonics — something unavailable in polyphonic mode. We'll look at real-time operation when we discuss the RS375 later in this review. In polyphonic MIDI mode, the RS370 operates as a polyphonic synthesizer with up to six voices. As it uses additive synthesis techniques, there's a lot of raw sound potential even before you connect up an external filter or six. At the same time, the RS370 serves as a MIDI/CV converter, its CV and trigger sockets becoming outputs. In polyphonic analogue mode, these sockets switch to input duties, governing the pitch and triggering of all six individual voices. In this mode, the RS370 works as an oscillator bank, with common waveforms for each.



Arpeggiator The arpeggiator included on the RS370 has the customary up, down, alternate and random modes, and may be sync'ed to incoming MIDI Clock with the same set of clock divisions as the LFOs. A latch function can be enabled via an incoming MIDI controller message or by a voltage at one of the CV inputs (you could even allocate the Ctrl In 1 knob to the role).

On The Menu At the very top level, the RS370 features eight menu headings, as follows: Copy From Memory, Synthesizer, LFO, Arpeggiator, Memories, MIDI/CV conversion, MIDI Options, and Special Options. When presented with a list like this, pushing the Edit knob serves as an Enter key to access sub-menus. When you reach a value that requires editing, pushing the knob again selects it. Turning the same knob makes the adjustment, and once you are satisfied, a final push confirms the updated value. To annul an edit or to back out of a menu, you push the Cancel button. This method of navigation is fine for relatively simple menu-hopping, but when employed extensively, as here, I'd say the controls have been pruned below the minimum for pleasurable use. The first of the top-level menus, Copy From Memory, is used to load a previously stored patch. A number of these are present to get you going, including organs, polyphonic synths, leads and so on. Many serve as helpful illustrations of the complex modulation routings and MIDI control assignments available. There are just 46 memory locations, selected manually or via MIDI program changes. Next in the list is the Synthesizer menu. This contains the bulk of the voice-tweaking sub-menus, including voice control, envelope settings, the mapping of control inputs and MIDI control messages, the Expander module's function (if it's fitted), the current Mode and any overall transposition value. Drill down into With its lone Edit knob and menu-driven user voice control, and you discover further interface, the RS370 is a bit of a departure menus for selecting oscillator for Analogue Systems products, but it works waveform, harmonic content, tuning, well, and the optional RS375 expander (see page 86) is available if you miss the handsoutput mixing, plus 'vintage drift' on, real-time control. amount and speed. I mention all these to highlight the level of navigation required, but fortunately, many parameters can be programmed for external control. As an extra timesaver, superfluous menu items are removed automatically according to context. For example, you only see the options for pulse width when you select a square waveform, and you don't see the menu to set harmonic levels unless the 'waveform=synthesized' option is taken. file:///H|/SOS%2005-06/Analogue%20Systems%20RS370%20&%20RS375.htm (3 of 7)9/28/2005 2:33:12 PM


The RS370 is a novel concept: a software polysynth designed for close interaction with a hardware modular. Fortunately, much is familiar. Its oscillator waveforms include four synthesizer stalwarts: sine, triangle, sawtooth and variable square. A fifth waveform, 'synthesized' offers additive synthesis, and this is probably its most exciting aspect. Select this option and you can scroll through up to 32 harmonics and set their amplitudes in the range -127 to +127; negative amounts invert the phase. When generating waveforms in this way, the display shows the results graphically. It's rather wonderful to see the waveform changing as you adjust each harmonic level — and educational too. If you don't want to program all 32 levels whenever you make a patch, six rather conventional harmonic templates are provided, including a saw/square mixture, a low-pass sawtooth wave and a square wave. Any of these may be used as starting points, after which you perform further adjustments as required. Each voice consists of four oscillators. Although these all produce the same waveform, they can be detuned with respect to one another for some dense, warm textures. Each oscillator's level and pitch settings can be modulated by either MIDI controllers or incoming voltages. To further muddy things up, Vintage Drift acts on the oscillators' detune values; increasing its amount increases 'thickness' dramatically (very much like the effect of lager...). I've already mentioned that voices can be mixed internally, so that all six are present at every Voice output. By routing voices individually, you could (with sufficient external modules) pan, process or filter all of them in ways rarely possible on even the most sophisticated polyphonic synths. I should add a note of caution here: your modular could grow dangerously and expensively large if you attempt to explore the full potential of this particular module! In order to be as self-contained as possible, the RS370 includes software envelopes and LFOs. As the envelopes have 10 stages, each appearing on a separate screen, I must admit I often wished for a simple ADSR. The parameters available include attack time and level, three separate decay slopes, and release time and level. There's also an option to compress or expand the entire envelope, or compress it dynamically at higher pitches. Fortunately, there are a couple of ways to improve access to the envelopes, one of which requires the Expander module, the RS375. The other uses the modulation matrix. Generally speaking, modulation is well catered for, with the RS370's four analogue voltage sources routable to 30 destinations. Each source can be mapped to just one of these and should you select a destination that's already in use, it is instantly zapped in favour of the new one (there's no Undo facility, so it's a good idea to plan your modulation connections in advance...!). Destinations include overall pitch,

The RS370.



oscillator levels and detune, all the envelope stages, the speed of the two LFOs, oscillator pulse width and 'harmonic wave morphing', plus arpeggiator and portamento settings. Harmonic wave morphing is worth a quick mention — if modulated through its full range, it sweeps the waveform from a sine wave to a square wave via a sawtooth on the way. In addition to the analogue modulation inputs, up to 16 MIDI controllers (plus aftertouch and pitch-bend) can be added to the equation, with the same list of destinations available. So by cunning allocation of 10 of these, you could gain complete control over the envelopes and thus reduce the time spent in menus. Of the RS370's analogue modulation sources, one of these is in the form of a control knob (Ctrl In 1) whilst the remaining three are control inputs (Ctrl Ins 2 to 4). For maximum flexibility the voltage range may be set individually for each CV input. Still on the topic of modulation, the two LFOs feature the expected selection of waveforms: sine, triangle, saw, square and random. They may be free-running or can sync to MIDI Clock with an impressive set of divisions — from every sixth of a beat right up to one step every five beats. Usefully, you can control LFO amplitude via a mod wheel or aftertouch, but the available routings are pretty stingy in comparison with other modulation connections. There are just four possible LFO destinations: pulse width, harmonic wave morph, fine pitch and fine pitch 2 (so each LFO can control pitch at a different rate and depth). Once you've created a patch, programmed extensive modulations and tweaked harmonics, envelopes and LFOs, the next logical step is to secure your hard work. With no MIDI Out socket, it seems, at first glance, that there's no way to do this. However, within the Memories menu is the 'SysEx Dump Memory' option and here, a push of the Edit knob squirts the patch out in SysEx form via the MIDI Thru port, revealing this to be a 'soft Thru'. Patches are transmitted individually and when they are restored, they enter the edit buffer, so you can place them (manually) in the memory location of your choice.

The RS375 Harmonic Generator Expander The RS375 (shown below) is a 48HP wide expander module featuring 16 knobs, each with an associated CV input. This module could be considered a luxury addition if you mainly intend to use the RS370 polyphonically. It offers a quick and direct way to tweak the onboard envelopes, but then you can achieve the same thing using MIDI. It also throws in an extra four control inputs (labelled Ctrl In 5 to Ctrl In 8) and four control outputs (Ctrl 5 to Ctrl 8), supplementing those on the RS370. These inputs are also available on the smaller, cheaper RS376 module. However, if real-time harmonic synthesis is your bag, the RS375 is essential. In its real-time modes (MIDI or analogue), the RS370 becomes a monophonic synthesizer, and the knobs of the expander can be assigned to set the levels of the first 16 harmonics. On top of this, connections made at the CV control points (say, sequencers or LFOs) will transform the generated waveform dynamically. file:///H|/SOS%2005-06/Analogue%20Systems%20RS370%20&%20RS375.htm (5 of 7)9/28/2005 2:33:12 PM


Varying harmonic content with knobs (rather than one at a time via the menu) was great fun, and I stumbled across some particularly dramatic changes when I connected the internal envelopes to several of the harmonic CV inputs. Considered together as a monophonic synthesizer, the RS370 and RS375 don't come cheap. However, they offer a range of highly malleable sound sources that could transform the personality of your modular synth.

MIDI/CV As a MIDI/CV converter, the RS370 is worthy of note, because polyphony is rarely something incorporated into such devices. It receives on a single MIDI channel and can drive up to six voices of either 1V-per-octave synthesizers or those using the Hertz-per-Volt convention (eg. certain Korg and Yamaha monosynths). Even when you use the 'polyphonic analogue' mode of operation and trigger each voice independently via CV and Gate, incoming MIDI controllers are still available as modulation sources. Trigger outputs may be chosen from a selection including conventional gate signals, S-Trigs (the 'short trigger' used by Moog synths) and a trigger of userspecified length. Finally, 'mixed triggers' are provided — something you don't encounter every day. In order to understand these, cast your minds back to certain limited polysynths of old. Korg made quite a few, such as the the Poly 800 and the Delta, where the synth lacked separate filters/envelopes for each voice. Using a single filter saved money, and of course made for limitations, but it also gave rise to some unique playing styles. You could retrigger the filter for all held notes by playing a new note, for example. The RS370's mixed triggers recreate this effect by sending every trigger or gate to all six trigger outputs simultaneously. Another welcome addition, the Zipper Noise Filter, is a low-pass filter (often referred to as a lag processor or slew generator) whose job is to smooth the seven-bit MIDI control values and eliminate the artefacts known as zipper noise. Its settings range from 0 to 127, and progressively higher values reduce the definition and response of incoming The RS375 Expander module. controller curves. So you would tweak this value according to need; in most cases, I found a setting slightly below the mid point worked well. The RS370 has four control outputs (Ctrl 1 to Ctrl 4) and each of these can generate a voltage sourced from the MIDI controller of your choice (or aftertouch, or pitch-bend). In addition, you can choose whether to send voltages proportional to incoming MIDI velocity or from the internal envelopes. Sadly, the output of the two software LFOs is not available; these may only be routed internally.



Whether used polyphonically or monophonically, you need to decide what should happen when you play more notes than your synth can handle. The RS370 offers three choices whose functions are self-explanatory: high-note or low-note priority, or 'drop oldest note'. Although you would generally use the latter option for polyphonic playing, there are interesting results to be had by using high- or lownote priority instead. It might have been handy to include 'drop lowest-velocity note' too — perhaps in a future firmware release? I mention this because, happily, Analogue Systems have provided an operating system that can be upgraded via MIDI. This function appears under the Special Options menu, along with calibration routines, an option to display the voices used, display of incoming MIDI, and display of CV inputs (from the RS370 and the RS375, or the more affordable RS376 Expander if this is fitted).

Conclusions Incorporating a polyphonic DSP synth into an analogue modular is a bold step. The RS370 is innovative and occasionally mind-boggling. Even when used to create traditional fare such as strings, organs or brass, its modularity means you rarely approach it as you would a conventional polysynth. In conjunction with the RS375 Expander, you can switch to monophonic mode and gain unique real-time control of harmonics. Although representing considerable outlay for just two modules, it's undeniably powerful. On the other hand, the RS370 is a step away from the immediacy of analogue modulars, and if you wanted to build a complete polysynth, with an analogue filter per voice, you would need to add a whole bunch of duplicate modules to your system. But having said that, the RS370 can take your modular into realms you never thought it could explore. I'll be fascinated to see what users do with it. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895

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


ART Head Amp

pros Very affordable. Loud, clear outputs. Simple operation.

cons You have to make up a special lead to work from mono sources.

ART Head Amp Headphone Amplifier Published in SOS June 2005 Print article : Close window

Reviews : Accessory

summary If you need a dead simple headphone distribution amp with four independently adjustable outputs, the Head Amp is a good cost-effective choice.

information £49.99 including VAT. Sonic8 +44 (0)8701 657456. www.artproaudio.com

Paul White

The ART Head Amp provides simple headphone monitoring for several performers. It's built into a small foldedsteel box and powered by an included mains adaptor which is thin enough to fit alongside a normal 13A UK mains plug. There is a TRS stereo jack input alongside four TRS stereo jack headphone outputs, each output having its own volume control — everyone gets the same headphone mix. The output impedance is 51(omega), which means that headphones of most common impedances can be used, and the input is designed to accept the line-level output of a mixer send, but you can also feed it directly from another headphone socket. There's no mono switch, so you'll need to make up a special cable if you want to feed a mono source into the unit and hear it coming from both phones. The maximum input level is +14dBV, and the signal-to-noise ratio is a perfectly adequate 90dB, with distortion below 0.01 percent. I checked out the Head Amp using my own selection of Beyer, AKG, and AudioTechnica studio headphones, and found there to be plenty of volume in all cases. In fact, with my Beyer DT250s, the level was as loud as I could comfortably tolerate at around halfway up! The sound quality was subjectively crisp and clear — more than adequate for performance monitoring — and the outputs showed no sign of running out of steam when cranked up. While there's nothing at all fancy about this product, it does do exactly what is asked of it with no fuss, and it does so at a very modest UK price. Published in SOS June 2005

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ART Head Amp

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%2005-06/ART%20Head%20Amp.htm (2 of 2)9/28/2005 2:33:14 PM

Arturia 2600V

In this article:

Arturia 2600V

The Sound Sources Virtual Semi-modular Synth [Mac/PC] Pitch Control & FM Published in SOS June 2005 The Filter Contour Generators & Print article : Close window VCA Reviews : Software Output Section & Effects Other Modules The Tracking Generator Keyboard Functions ARP 2600 Revisions Completing the quartet of vintage synth emulations The 1601 Sequencer they began with Moog Modular V, Arturia's latest plugAble Cables in aims to reproduce the sound of the greatest semiIn Use modular of them all, ARP's 2600. We see how it fares up Conclusions

against the original...

Arturia 2600V £200 pros

Gordon Reid

A hugely flexible monosynth. A very interesting polysynth. The sequencer is an important bonus. Valuable additions to the original spec, such as oscillator sync, new filter profiles and more. Improved envelopes over those of the original. Improved voltage processors.

I 'm an ARP man; I always was. Sure, the Minimoog was a great synth... limited, but great. But apart from this and the original Taurus pedals, Moog's record was never better than patchy. Sure, you could compare the Minimoog and the Odyssey, but where were the equivalents to the wonderful Pro Soloist, the Axxe, and the Omni? The Satellite, the Micromoog and the Opus 3...? Give me a break. Then there was the ARP 2600. With no Moog equivalent, this combined prepatched synthesis with the flexibility of a modular synth, all housed in a neat suitcase that didn't cons need a pair of roadies and a transit van to move it The filter response is not from one gig to the next. true to that of the original synth, and nor is the triangle waveform. The ring modulator lacks punch. As on Arturia's other emulated instruments, the effects don't work properly. The manual is helpful, but contains errors.

summary Arturia's 2600V is the company's best software

Arturia's 2600V at its maximum

Nevertheless, the bias toward Moog survives to 1156-pixel height, with the 1601 this day, as demonstrated by the host of digital sequencer emulation and keyboard visible. imitations of the Minimoog. But perhaps this is about to change, with the almost simultaneous release of two ARP 2600 software synths. This month, we'll look at the first of these. It's the fourth software-based emulated synth to emerge from Arturia, and as with their previous emulations, it aims to produce a sound as close as possible to that of the hardware instrument, whilst also sympathetically extending the original's featureset with more modern facilities such as polyphony and MIDI capabilities. It's called 2600V, and as I've done with my previous reviews of Arturia's software emulations, I

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Arturia 2600V

synth to date. It's powerful, it's flexible, and the bugs are few and far between. While it does not sound precisely like an ARP 2600, it offers much more than the original. Given the cost of buying a vintage ARP 2600 and 1601 Sequencer, 2600V has to be worth serious consideration.

information £199.99 including VAT. Arbiter Music Technology +44 (0)208 207 7880. +44 (0)20 8953 4716. Click here to email www.arbitermt.co.uk www.arturia.com

put it up against the original hardware synth to see just how close the emulation was.

The Sound Sources I began by evaluating the primary modules in the signal path: the VCOs, VCF, VCA, the output section, and the envelope generators that control them. The first two graphs below show the sawtooth waveform generated by VCO2 on my grey-face ARP 2600, and on 2600V. The equivalence is clear, and, as one would hope, the sound from these is all but indistinguishable. The authenticity started to wobble when I inspected the sine waves (see the last two graphs on this page). When directed straight to its output, 2600V's sine is brighter than that of my ARP 2600 — and how can one sine wave be brighter than another? Patching the output to an oscilloscope revealed all — the sine wave on 2600V is not a sine wave at all. It contains significant overtones, although strangely, directing this wave through the pre-patched signal path eliminates the overtones, leaving the correct, pure tone (see the first two graphs on the next page). That's weird, but gratifying.

Test Spec MAC REVIEW SYSTEM 1GHz Apple G4 Powerbook G4 with 1GB of RAM running Mac OS v10.2.8. 1974 ARP 2600 (serial number 1618). Plogue Bidule VST host v0.8. Arturia 2600V version reviewed: v1.0.

The ARP 2600's sawtooth waveform.

2600V's sawtooth wave.

The ARP 2600's sine waveform.

2600V's sine wave.

Then I came to the triangle waves. As you can see from the last two graphs on the next page, 2600V generates a 'shark's tooth' waveform, while my ARP produces something thoroughly triangle-like. In listening tests, 2600V's sound is a little file:///H|/SOS%2005-06/Arturia%202600V.htm (2 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

brighter and less 'bottomy' than the ARP's, although the difference is subtle. However, I've seen the waveform before; it's very similar to the triangle wave produced by Arturia's Minimoog V (printed in my March 2005 review of that product), and even though it's not identical to 2600V's, I couldn't help but wonder if Arturia were recycling some of their technology, even though 2600V supposedly models a different synth. Happily, 2600V's pulse waves sound reasonably accurate whether static or modulated although, as on other software synths I've used, its square waves are too 'pure'; they're recreations of a near-perfect square wave rather than the imperfect, not quite 50-percent waves of a real analogue. The final audio source is a noise generator that approximates that of the ARP 2600, but with a wider range of colorations.

Pitch Control & FM The VCOs on 2600V offer numerous enhancements over those on the original, all of which are welcome (for more on the various types of hardware 2600 that were released in the 1970s, see the box later in this article). Firstly, each offers keyboard tracking from zero to 100 percent. Secondly, each has a five-position switch offering 4', 8' 16' and 32' settings as well as low-frequency oscillation. Next, the Initial Frequency control is quantised in semitones, which is a huge improvement over the original hardware. In addition, VCO3 has a PWM CV input and sine- and trianglewave outputs, all of which the ARP lacked. There are a further two additions which I've separated from the others, because the manual describes them incorrectly. Firstly, it tells you that you can modulate the 'widths' of the sawtooth, triangle and pulse waves of VCOs 2 and 3, but in fact the sawtooth waves are unaffected. Secondly, it says that you can sync VCO 1 to either of VCO 2, 3, or 2 and 3. In fact, only VCO 2 is available as a master. One idiosyncrasy of 2600V's oscillators is that they go sharp when you apply audiofrequency FM. This is not in accordance with FM theory so, when notified of this behaviour, Arturia stated on their web site, "FM modulation may not produce such a pitch deviation, neither with a DX7 nor with an ARP 2600" and undertook to correct the fault. But when I tested this on my original ARP 2600, I found that it did exhibit the same behaviour as the current version of the software synth; increase the modulation depth, and the pitch goes sharp!

The Filter Unlike the hardware original, 2600V offers no fewer than five filter profiles: 24dB-peroctave low-pass, 12dB-per-octave low-pass, 12dB-per-octave high-pass, 12dB-peroctave band-pass, and Notch types.


Arturia 2600V

Arturia's marketing information states that 2600V has two filters: one that emulates ARP's 4012 filter, and one that emulates the ARP 4072, but this is not the case. Furthermore, the manual makes much of the equivalence of 2600V's 12dB-peroctave low-pass filter and that of the original synth, but that can't be right, as both the 4012 and the 4072 were 24dB-per-octave low-pass devices. Fortunately, tests show that the 24dB-per-octave setting on 2600V does indeed roll off the signal at approaching 24dB-per-octave, so make of the marketing and manual what you will. To get a feel for 2600V's filter, I first tested my ARP 2600. Measurements showed that the front-panel sliders adjust the cutoff frequency from 9Hz to 19.4kHz. With CVs applied to the control inputs, the top end shoots off the top of the audible scale. Furthermore, the self-oscillation at maximum resonance behaves as you would expect. It produces a low-ish output (as set up) of 500mV at 9Hz, around 1000mV throughout the audio range, and it falls away to 140mV at 19.4kHz.

The spectrum of 2600V's 'sine' wave.

The ARP 2600's triangle waveform.

The same spectrum filtered by the signal path.

2600V's triangle wave.

I then tested 2600V's 24dB-per-octave filter; Arturia claim a frequency response of 10Hz to over 21kHz for this. Making the filter self-oscillate and patching the VCF output to the main output, I found the range to be 32Hz to 18.6kHz. This is almost two octaves short of the claimed specification at the bottom end. Furthermore, the upper frequency of 18.6kHz appears to be an absolute maximum, and no amount of controllers applied to the filter's CV inputs will cause it to exceed this. While taking these measurements, I noticed that the amplitude of 2600V's filter file:///H|/SOS%2005-06/Arturia%202600V.htm (4 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

oscillation is rather weird, with its highest output at the lowest and highest frequencies, and with a significant dip at mid-range frequencies. If, as I did with my ARP 2600, I set up 2600V's filter to generate an output of 500mV at its lowest frequency, the output was just 60mV at 1kHz, but 190mV at 18.6kHz. This is not the normal response of an analogue synthesizer's resonant filter. Once again, though, I realised that I'd seen this response before — and referring back to my review of Minimoog V confirmed my suspicions. Again, it's obvious to see what this implies. By a happy coincidence, I was able to discuss this matter with Arturia at the recent Frankfurt Musikmesse and it turned out that, as I had suspected, Minimoog V and 2600V share blocks of common code. Arturia justified the use of a common 24dBper-octave filter by suggesting that, as ARP had been sued in the 1970s for breaching Moog's filter patent, the response of the ARP 4012 would be identical to that of the Minimoog filter. When I suggested that this was not the case, they undertook to look into it and, if necessary, consider changing 2600V's filter. The same positive attitude prevailed when we discussed the incorrect triangle waves and other observations I'll make below regarding 2600V. I am very encouraged by this, especially since significant upgrades — such as the forthcoming v1.5 for Minimoog V — are supplied free of additional charge to legitimate users. Returning to the filter itself, Arturia claim that the other four filter profiles are emulations of the multi-mode filter in another ARP synth — the 2500. I have no way to test this, and I'm not going to try to make a judgement based on my quartercentury-old memories of playing a 2500. All I'll say is that these filters extend the range of sounds considerably.

Contour Generators & VCA The envelopes were perhaps the weakest aspect of the ARP 2600 because, without modifications, their maximum Attack, Decay and Release times were far from generous. Thankfully, Arturia have not been bound by these limitations, and, in recreating the envelopes and trigger/gate options, I'm pleased that they have extended the maximum times way beyond the original specification. More improvements lie in the minimum contour times. The first graph above shows the click generated by passing white noise through the ARP 2600's VCA when controlled by the ADSR with all values set to zero. The second trace shows the result obtained from 2600V. As you can see from the third trace, which superimposes the previous two, the durations are similar, but the software synth's amplifier begins to 'close' much more rapidly. The result is a more precise click than that obtained from the analogue synth. Another difference — although one that you may or may not view as an improvement — is the elimination of the famous ARP 2600 'thump' that can occur


Arturia 2600V

when you press and release notes when the envelopes are at their fastest settings. Most players viewed this as a fault, so at least one electronic 'fix' was developed for the original synth. By definition, therefore, the 2600V emulates a modified ARP 2600. There is another significant change in the VCA section. On the ARP 2600, the horizontal slider at the top applies a permanent 'Initial' gain to the VCA, famously prompting 2600 user Joe Zawinul to ask, "how do you switch it off?" This is replaced on the software synth by a Global Volume control, which is very useful, although not quite as sensible as it seems, especially if you want to play external signals though the filters and effects.

Output Section & Effects At the end of the pre-patched signal path lie the audio mixer, spring reverb, and stereo outputs. Apart from some minor graphical changes, these appear to be identical to the original's, but the sound is not. To be specific, the 'boinnggg' of a three-spring reverb is hard to emulate, and although Arturia have retained the spirit of a spring, the sound is very different.

Envelope clicks from (left) the ARP 2600 and (centre) 2600V. The right trace shows both superimposed.

Hidden away behind the left speaker grille lie two additional effects: chorus and delay. The chorus is a single modulated delay line whose output is directed equally to the left and right channels, no matter where the output from the main signal path is panned. It's no substitute for the lush textures of a triple-delay chorus, but it's acceptable if used subtly. The basic delay unit offers independent delay times (from 1ms to 3s) and feedback (from zero to 100 percent) for each of the left and right channels, without cross delays. The nice touch here (and nowadays a common one) is the ability to synchronise the delay to MIDI Clock. Ratios are available from oneeighth to nine times the currently set MIDI tempo. I have criticised Arturia in the past for attaching faulty effects to the outputs of their software synths and, unfortunately, this time is no different. This is because, if the delay is 'wet', merely the wet (ie. pitch-shifted) part of the chorus is passed to the delay. Only if the delay is completely 'dry' (ie. 'off') does the chorus wet/dry mix work file:///H|/SOS%2005-06/Arturia%202600V.htm (6 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

correctly.

Other Modules The Sample & Hold/Electronic Switch section was one of the additions that made the ARP 2600 special, and despite small graphical changes, this has been recreated precisely on 2600V. There's one extra: as elsewhere, you can synchronise the internal Clock to MIDI Clock. But if the S&H section is true to the original, the same cannot be said of the Voltage Processor section (shown left). It's better! Rather than the original's seven inputs, limited mixing, dual inverters, single lag processor and three outputs, 2600V offers eight inputs configured as four linkable input pairs with CV-controlled mixing, four inverters with on/off switches, four lag generators, and four outputs. The possibilities — such as dynamic mixing of multiple sources to create evil modulation curves — are considerable, and remarkable. If you wanted to get silly with an ARP 2600, you could present external signals to its inputs and see what happened. However, there was just one place where ARP intended that you should insert such signals. This was the Preamplifier, with its associated Envelope Follower. These are recreated on 2600V, and using them The Voltage Processors and Sample & Hold proves to be simple. In my case, it meant module. loading 2600V as a VST instrument under my chosen VST host (Plogue's Bidule), whereupon it appeared with two audio inputs as well as two outputs. I could then direct the output from other software synths to these inputs, patching the output from the Preamplifier and/or the Envelope Follower to the destination(s) of my choice. A Ring Modulator lies alongside the Envelope Follower. Strangely, while Arturia's panel has the Audio/DC legending found on the ARP 2600, the switch to which this pertains is missing. Testing showed that 2600V modulator is 'AC coupled' (the Audio setting) which means that the carrier and modulator frequencies are not present in the output. Given a single option, I would select this one, but it would be nice to have the choice.

The Tracking Generator Up to this point, Arturia's programmers have stuck to the spirit of the ARP 2600, but they couldn't resist adding something from the 21st century. It's the so-called 'Tracking Generator' that's tucked away behind the right-hand speaker grille. The manual describes this as having four audio inputs, four modulation inputs (the onfile:///H|/SOS%2005-06/Arturia%202600V.htm (7 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

screen jacks on the left of the module — although it's not clear what's being modulated) and four audio outputs (on the right). As you can see from the screenshot below, this is incorrect, as there are no audio inputs, just the modulation inputs on the left of the module, which affect the frequencies of the control waveforms. In essence, the Tracking Generator allows you to create four control waveforms using a variety of tools available in the Edit window provided for each. You can then smooth these if desired, and direct the outputs to the destination(s) of your choice. You determine the frequencies of the outputs using the Freq knobs and, as elsewhere, these can be synchronised to MIDI Clock.

The Tracking Generator.

In other words, the Tracking Generator is nothing more nor less than four programmable LFOs. While this may seem a bit advanced in an emulation of a 1970s monosynth, the philosophy is not outrageous. On a smaller scale, this is what ARP did when they added an LFO to the ARP 3620 keyboard, thus freeing up one of the 2600's oscillators and extending the modulation capabilities of the complete system. Which brings us to...

Keyboard Functions ARP developed three keyboards for the ARP 2600. The first, the 3601, was available only on the early models. Next, the monophonic, 49-note 3604P offered portamento and variable tuning, but this was later replaced by the duophonic 3620, a far superior unit that incorporated the dedicated LFO that I've just mentioned. The 3620 was quite a sophisticated piece of equipment. In addition to the LFO, it offered a two-octave up/down switch, a pitch-bend knob, portamento with a momentary on/off switch as well as a footswitch on/off input, single/multiple triggering, auto repeat, and dual pitch CV outputs. For aesthetic reasons, Arturia have sited An example of a Tracking Generator curve. many of these functions elsewhere on 2600V. Most noticeably, the LFO has migrated to the centre cabinet, where it has grown an extra output (a sawtooth wave) and MIDI-synchronisation capabilities. However, the output from the LFO is very strange. Notwithstanding the fact that it goes up to 100Hz (despite the manual saying that it has a maximum frequency of 20Hz), its waveforms exhibit 'stepped' file:///H|/SOS%2005-06/Arturia%202600V.htm (8 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

shapes and complex spectra. Counting the levels shows that the LFO is quantised with just three-bit resolution, but when you apply it to a destination, it does behave as it should, with smooth sines and correctly rendered triangles, squares and saws. Other keyboard controls now found in the sequencer section include the Global Tune knob, plus the Trigger Mode and Auto Repeat switches. These are joined by modern-day functions such as the Mono/Unison/Poly switch and the unison Detune amount. The space thus liberated to the left of the keyboard is re-used to house a number of performance CV outputs — velocity, aftertouch and so on, plus a pair of pitch CV outputs; one at 100-percent tracking, and the other at 400 percent.

ARP 2600 Revisions The original ARP 2600 appeared in four major guises, although many revisions were not honoured with a change of external design or a new paint job. The first few (built in 1970/71) incorporated the simple, monophonic 3601 keyboard and appeared in a bluepainted metal case with a wooden carry handle. Despite their cachet as the earliest ARP 2600s, these 'Blue Meanies' were hand-built, notoriously unreliable, and difficult to keep in tune. The next revision was the shortest lived, and only a handful of 'Grey The black and orange final revision of the Meanies' exist. These incorporated two ARP 2600. handles and were very stylish, but they were deemed less suitable for ARP's target markets — performing musicians and schools — than the design that was to follow. The most common ARP 2600s (1971 to 1978) are grey with white legending, and have separate synth and keyboard units. These came in two distinctive flavours. Early models, like the Meanies, used ARP's 24dB-per-octave 4012 filter, but this infringed a Moog patent and was replaced in 1976 by another 24dB-per-octave design, the 4072. Unfortunately, the 4072 had a fault (easily fixed, as it happens) that limited its bandwidth to just 12kHz. If you hear an ARP 2600 and wonder what all the fuss is about, you're playing one without the correction! The final revision saw the synth adopt the black-and-orange livery of latter-day ARPs. This offered numerous internal improvements, most significant of which was the demise of the epoxy-encased circuitry of earlier models, which made these 2600s relatively simple to repair.

The 1601 Sequencer 2600V's sequencer is perhaps the area in which Arturia have kept most closely to the form and function of the past... yet it was never originally a part of the ARP 2600! In fact, it's a recreation of ARP's separate 1601 Sequencer. As such, Arturia's file:///H|/SOS%2005-06/Arturia%202600V.htm (9 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

emulation offers 16 CVs, 16- and dual eight-step modes, sequential and random playback, three gate busses, the ability to set and modulate the clock pulse width, unquantised and quantised sequencer outputs, a novel Boolean AND function, and inputs to the quantisers (so that you can convert any two external CVs into semitone voltages). There's more that's the same as it was on the hardware so, rather than list everything, I'm going to mention the two features that have changed. Firstly, there is no multiple. This is no longer necessary because — as elsewhere on 2600V — you can direct a single output to multiple destinations. Secondly, the voltage ranges have been modified. The CV sliders on the ARP 1601 swept across 10 octaves (ie. 10 Volts) at the unquantised outputs, and two octaves at the quantised outputs. Arturia's version offers ranges of ±4 octaves at the unquantised outputs and approximately ±2 octaves at the quantised outputs. The benefits of the improved quantised range far outweigh any loss at the unquantised outputs.

Able Cables 2600V offers three colours of patch cable: red ones denote connections made from audio sources, orange ones denote signals from modulation and other control sources, and green ones denote modulation signals generated by the Tracking Module. Connecting could not be simpler. Drag from a source to a destination or hold down the Shift key and click on any socket to reveal the patching menu. A source can be directed to any number of destinations, but each input can only receive a single source at a time.

The Patching menu.

When you connect cables to one of the 10 sockets with a black surround, you can click on the 'nut', and it becomes a knob that allows you to set the input level between -100 and +100 percent. This is like having 10 additional VCAs at your disposal, and is extremely useful.

In Use First things first. 2600V runs on 1GHz Macs or PCs with 256MB of RAM or more, provided they're running Mac OS 10.2 or higher, or Windows 98SE, 2000, or XP respectively. Loading 2600V onto my test Powerbook G4 was glitch-free, and if you're using a Mac, the software installs stand-alone, VST, RTAS and HTDM versions (if you're using a PC, you also get a DXi version). Having done this, you'll need to set up the toolbar correctly, selecting such things as the MIDI input source, the keyboard range, and the audio channels for input and output. This is also where you save and recall patches, determine the number of voices for monophonic, file:///H|/SOS%2005-06/Arturia%202600V.htm (10 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

unison or polyphonic use, and select which part of 2600V you're going to view (the synth itself, the sequencer and keyboard, or everything). The last of these is important, because the full representation is big; 1156 pixels from top to bottom, which makes it too large for many monitors. You can also choose which 'skin' you wish to use: a Blue Meanie, a grey-face or a black/orange-face. However, don't think that the sonic characteristics of each are on offer; the choice is purely cosmetic. One final aspect of the toolbar deserves a mention before I move on: it's the Magnet tool, which causes the patch cables to move out of the way of the mouse pointer when you want to adjust a control that may otherwise be obscured. If only my modular synths had this facility!

The ARP 1601 sequencer was sold separately from the 2600, but 2600V incorporates a version of it.

Having set everything up, let's start by noting a couple of things that are not implemented on 2600V. Firstly, the duophonic mode of the 2600+3620 configuration is not recreated. Whether this is a problem or not is moot, especially when you remember that 2600V is polyphonic. More importantly, you can't use the Voltage Processors to create static voltages, or as manually-controlled voltage sources. Happily, the additions are more significant: oscillator sync, the multi-mode filter, the tracking generator and variable keyboard CV tracking are very welcome, even though they bear no relation to the performance of the original synth. Once I got used to the layout, everything fell to hand (mouse?) as I had hoped. Furthermore, you can assign a MIDI continuous controller to almost any control in 2600V, so I linked the knobs on my Korg Legacy MS20 (USB) controller to sensible equivalents in 2600V, and the software synth leapt into life. I wish Arturia produced a miniature ARP 2600 controller...! So, what about the sound? Is this truly a reincarnation of the ARP 2600? To test this, I set up a simple, monophonic, single-oscillator patch on my ARP 2600 and on 2600V. With careful tweaking (and no filter resonance) I could create sounds that were almost indistinguishable from one instrument to the other. I could also tweak 2600V with no discernable zipper noise. However, as soon as I added filter resonance, the differences began to show. Furthermore, when using the USB controller, the stepping of the filter cutoff frequency became audible. As my tests suggested, the 2600V's 24dB-per-octave filter may or may not be a good one, but it does not emulate the 4012 of my ARP 2600. As my patches became more sophisticated, the timbral differences became more significant, and it became apparent that, for some sounds, 2600V lacks the depth of a real ARP 2600. It's sometimes subtle, but there are other times when something has definitely been lost. For example, the Ring Modulator lacks the depth and 'grunt' of the device in the ARP 2600 itself. It's still a ring modulator, but — notwithstanding the loss of the DC-coupled option — this one just doesn't behave the same. I also discovered that it is important to keep an eye on the levels within 2600V, because file:///H|/SOS%2005-06/Arturia%202600V.htm (11 of 13)9/28/2005 2:33:18 PM

Arturia 2600V

too much signal will cause clipping and extreme distortion. If pushed, the ARP 2600 will clip by a smidgen, but nothing like 2600V, so you have to keep this under control. Similarly different, if that makes sense, are the factory sounds supplied with 2600V. These are supposed to be The Toolbar, incorporating the handy cablemoving Magnet tool on the far right. recreations of the sounds supplied on patch charts with the original 2600, but with the original ARP patch book to hand, an audio comparison revealed similarities, but no more than that. Overall, the presets are useable sounds, but they are not the ones from the original patch book. I experimented further with the sequencer and some of the more esoteric modules. The 1601 is far more flexible than its panel suggests, so Arturia should be commended for including this alongside the 2600 itself. I particularly like the way that the output jumps to the step you're adjusting whenever you move one of the pitch CV sliders. When patched properly, this makes setting up a doddle. Next, I got a bit daring, using 2600V as a signal processor for all manner of other software synths, virtual Mellotrons and electric pianos. The results could be stunning, and I had great fun sequencing parameter changes, especially when synchronised to MIDI. Furthermore, I was pleased to find that my 1GHz G4 was capable simultaneously of supporting 2600V (with moderate polyphony) and other packages without glitches, pops, or dropouts. My only disappointment was that I could not automate patch changes or the insertion/removal of patch cables. Finally, I created a range of polyphonic patches that included strings, brass, pads, electric pianos, percussion and effects. Initially, I found that 2600V excels at producing edgier and more percussive sounds such as electric pianos (at which it is superb) rather than strings and brass. Arturia's 'factory' patches seemed to bear this out; there's hardly a string patch among them. But wait... the filter and at least some of the oscillator code is the same as that in Minimoog V, so the same, fat timbres should be available. To test this, I loaded Minimoog V and 2600V simultaneously, and copied the parameters of one of my string ensembles from the former to the latter. Having done so, the sounds were almost indistinguishable. If nothing else, this shows that the user-interface is an important aspect of any synth, guiding the way in which you conceive and create sounds.

Conclusions Arturia's 2600V takes the form of the original ARP 2600, but adds many facilities not previously available, so we should judge it in two ways. Firstly, does it emulate — and sound like — an ARP 2600? Secondly, ignoring the original, is it a desirable synthesizer in its own right?


Arturia 2600V

The answer to the first question is no, unless your patches are quite simple. Nevertheless, the software retains much of the character of the ARP 2600, and I applaud that. The answer to the second question is more important. The 2600V is powerful, it's extremely flexible, and its sound makes it something that you might choose to use whether it looked like an ARP 2600 or not. Add the sequencer, polyphony, and all the other features and — despite a few niggles — it all adds up to a very attractive package. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Circular Logic InTime

In this article:

Installation User Interface Practice Makes Perfect Pitfalls System Requirements

Circular Logic InTime Real-time Tempo Tracking Software [Mac/PC] Published in SOS June 2005 Print article : Close window

Reviews : Software

Circular Logic InTime $159 pros Good real-time tempo tracking for keyboard and MIDI drum triggers. A quick way of getting tempo maps to overdub MIDI onto live takes.

For those who want to put a human performance at the centre of their sequenced masterpieces, Circular Logic have developed software that can understand and follow tempo changes in real time.

cons No audio in option. Needs a lot of practice to get the first results, although it gets easier to use.

Ingo Vauk

Attempts to infuse sequenced music with a natural feel go back a long way. summary In the days of tape machines and Atari The developers at Circular computers, engineers used to Logic should be congratulated painstakingly gate audio signals off for this innovative and stable tape to feed them into trigger units in piece of software. If you are their SRC SMPTE-to-MIDI boxes, or willing to invest enough time to overcome initial the audio in jacks of the trusted C-Lab frustrations, InTime turns out Unitor. In our times of weapons-grade to be a very clever application home computing we take these abilities that will reliably slave your computer's MIDI clock to your for granted, with all major DAWs featuring the ability to extrapolate live performances. tempo data from both MIDI and audio source material. information $159. www.circular-logic.com

Test Spec InTime v1.0.2. Apple G3 iBook 800MHz running Mac OS 10.3.8, with Emagic MT4 USB MIDI interface and Emagic EMI 2|6 USB audio interface. Apple desktop G4 933MHz running Mac OS 10.3.8 with Emagic Unitor 8 MIDI interface

However, the ultimate challenge now is to do it in real time: generate the clock while you are playing and have the computer follow you like Tony does George W. Circular Logic, a Florida-based company who proclaim as their mission statement that they "want to make computers that listen like musicians so musicians don't have to think like computers" are offering the technology. Their InTime software generates tempo information from any MIDI source and transmits this in the form of MIDI Clock data to the sequencer of your choice: this can be a drum machine, hardware sequencer or any of the current desktop systems, bearing in mind the limitations of these systems. For example, Logic 7 cannot cope with real-time tempo changes when using the Audio Engine and Apple Loops, although MIDI is no problem. Ableton Live, on the other hand, makes a perfect partner for InTime, and I imagine that Propellerhead's Reason

file:///H|/SOS%2005-06/Circular%20Logic%20InTime.htm (1 of 6)9/28/2005 2:33:22 PM


and Digidesign Pro Tools TDM hardware.

would suit the setup as well.

Tested with Ableton Live 4, NI Reaktor 4 and Apple Logic 7.

The basic idea, then, is to infuse the programming with some live feel, the computer slowing down and speeding up in sync with your playing. Another application, which ultimately could prove more useful, is InTime's Groove Tracking mode, but more about that later...

Installation After filling in a lengthy questionnaire you can download InTime, and the small file size (about 2.2MB for the OS X version) makes that quick and painless. Installation on my system was smooth, and the authorisation is of the challenge-and-response type used by most software houses these days. Once the program is installed it needs to be connected into your MIDI environment. Depending on your setup there are multiple ways of doing this; in OS X, which I use, InTime recognises all connected interfaces and allows you to access them from its MIDI Devices window. As you might imagine, it is here where you do your routing of MIDI signals, be it within the computer or to the external world. By enabling Beat Clock and choosing the port you want to address, you set up InTime to be the sync master. Now all you have to do is to connect a sequencer capable of handling tempo variations in real time to that MIDI port and you're up and running.

User Interface InTime is controlled from a sleek and intuitive user interface consisting of 10 separate windows that can be viewed simultaneously as well as individually. They are small enough to fit on to the screen at once, giving you all parameters at a glance. The main window, simply named 'InTime', also allows access to most other panels at the click of a button, which is very useful in the beginning when you are familiarising yourself with the software. InTime is designed around few parameters and is therefore straightforward to understand, and the few occasions I did consult the manual confirmed my intuitions about what a parameter was designed for. Apart from the MIDI Devices window, which is self-explanatory, the most important controls are located in the main window. Here you set the sensitivity of the tracking, which defines how responsively the clock reacts to any tempo



changes in your playing. The higher the sensitivity, the closer the tracking to the playing. Circular Logic recommend a low setting to start with, and it has to be said that high settings can make you seasick when you're not experienced. Generally speaking, values between 2 and 7 give a satisfactory result, depending on what instrument and sound you are playing, while the highest 11 (very Spinal Tap) makes you feel like you're trying to ride a horse that's not been broken in. Also located in the main window are the transport controls and various mode switches such as the toggle switch between groove and tempo tracking, clock out and tempo tap. In the MIDI Triggers window you can assign MIDI events and controllers to set certain parameters. This is a very useful feature in terms of playing live: for example, it allows you to use MIDI controllers to tap the tempo, stop, start, and adjust the tracking sensitivity. A MIDI learn function makes these assignments quick and easy to execute. The Tempo & Start Control window allows you to set the initial tempo as well as minimum and maximum tempo values, which can be overridden with the tap function should you want to do so. Also in this window are various count-in and start options: you can set the length of a count-in (irritatingly named 'countoff') or use the 'Wait Note' mode, which starts the clock on the first MIDI Note On. The next most important window is Advanced Tracking: it's here you really define how the tempo is going to behave, and it is here where you either get InTime to work for you or give up on it. A 'Rate Filter' eliminates MIDI clutter such as strumming or sliding up or down the neck of a guitar, drum rolls, piano glissandos and other sudden streams of rapid, out-of-time MIDI events that would otherwise confuse the tempo tracking. 'Sub Tracking', on the other hand, divides the bar into more precise subdivisions to allow certain kinds of syncopation that might otherwise be misinterpreted as speeding up or slowing down. For example, staying clear of the downbeat in a one-drop feel can confuse InTime if



'Sub Tracking' is disabled. In Groove Tracking mode, InTime stays around one average tempo but varies around that tempo in order to move material that is hard-quantised to fit the groove. Tracking Bias is one of the most important controls, since it is responsible for the overall 'feel' (I use that word with hesitation) of the tempo tracking. Generally speaking, a negative value will make it easier to slow down, while a positive one will make the tempo go up and up and up and up. Especially in the beginning, when you are getting used to InTime, it will seem like it is only ever going faster. A related control is Momentum, which in a sense reinforces the rate of tempo change. If, for instance, you are slowing down on a high momentum setting, InTime will anticipate you slowing down further. Setting up InTime so that it does what you want is a balancing act between the Sensitivity, Momentum and Bias settings. Once you have understood their relationship you are a good way into getting a result from the software, and finding the right settings depends very much on the feel of what you're playing. A laid-back feel can be hard to achieve with a negative Bias setting at a high Sensitivity because it will be interpreted as slowing down, while sudden tempo variations are near-impossible with high Momentum, yet low Momentum can be a pain as it will result in InTime following every little hesitation or push in the playing. Once you have achieved the right settings for the tune you are playing, you can record a take into InTime. This will not be quantised but will generate a tempo map. On playback of this track the software will keep the tempo variations intact, and once saved, the take can be loaded into any other software as a MIDI file. The tempo map is included in such a file, meaning that you can start building a track on your usual DAW starting from a feel-based take.

Practice Makes Perfect Anyone planning on using InTime on stage needs to do a lot of rehearsing, since the way your playing is interpreted very much depends on the settings. It is a little disconcerting how a slightly unsuitable setting can turn a mellow trip-hop track into a raving polka. Here, InTime's preset and set list functions come in handy, because they allows you to combine settings and MIDI files into presets, which in turn can be compiled into set lists. InTime will load the next preset and its associated MIDI files (should you so desire) after the end of the previous preset, either once a MIDI file has ended, or when you hit stop.

Pitfalls file:///H|/SOS%2005-06/Circular%20Logic%20InTime.htm (4 of 6)9/28/2005 2:33:22 PM


My main technical problem with InTime is that it extrapolates its tempo information from MIDI data. This brings with it a few inherent limitations, which I think restrict the usability of the software. Firstly, MIDI within a computer invariably suffers from latency, which in itself is a natural enemy of feel-based playing, although ever-increasing chip speeds make it less of a problem these days. For the purposes of this test I used a normal MIDI keyboard, a Korg Z3 MIDI guitar, and some bongos triggering a standard trigger-to-MIDI device. I found that the best result was achieved with the drums, since the guitar and the master keyboard both suffered from a compound effect of latency at every stage of the process. However, I suspect that many of those who are interested in InTime will be keyboard players, since most drummers and guitar players I know don't really want to get into digital equipment. They will either have to put up with latency, or invest in additional gear for the sole purpose of using InTime. However, Circular Logic have said that they would think about implementing an audio option at a later stage should the demand arise. Talking of audio, real-time timestretching is also asking a lot of any slave program: even Ableton Live uses a clock smoothing function when running in slave mode, which makes the overall response a little less sensitive. If you want to find out just how responsive InTime can be it is a good idea to load some MIDI files into its own file player and play along to those. In this case you aren't dealing with any audio time-stretching but only discrete MIDI triggers, and the tempo changes are very immediate. My final point is that there is a danger of losing exactly what you are trying to capture in the performance: the feel. Circular Logic state that you need to listen to InTime while you are playing, just as you would with another musician. I found that the difference between a computer and another musician is that you can trust your mate to play sympathetically with what you are doing, and that musicians feel and most importantly anticipate what is required musically. When listening to InTime you are very busy re-checking the effect your playing is having on the tempo (I can see where the name Circular Logic came from) and in this mental feedback loop you are in danger of losing the very feel you're after. I would imagine that jamming with a bigger bunch of people is very exhausting for the one connected to InTime, since he or she will have to do a lot of juggling to keep the groove going. I only used the software playing together with one other musician, and found that I was listening to either him or the software. But that might just be a question of practice: if you are into trying to marry live playing with responsive sequencing, InTime is certainly a very interesting option. Circular Logic say that they have been frustrated by people giving up on InTime before they realise its potential: it takes some practice to get the hang of it, and in file:///H|/SOS%2005-06/Circular%20Logic%20InTime.htm (5 of 6)9/28/2005 2:33:22 PM


the beginning you are likely to experience runaway situations when it just seems to go faster all the time. I have to say that it pays off to follow the manual in this respect and to play simple and rhythmically steady material in order to feel your way in. After a while, it becomes progressively easier, and after an hour or so the fun begins. As long as the playing is of reasonable quality and within sensible parameters as far as the tempo fluctuations are concerned, InTime is perfectly capable and in fact very clever at tracking tempo changes. Apart from fun, the main application I can see for the software is in productions where you want use a live performance as a basis for further sequencing. Unfortunately the examples given on the Circular Logic web site are not very inspiring, since they mainly demonstrate that it is possible to speed up and slow down but leave you wondering why you would want to do that in the first place. In the right hands, however, I think some amazing results could be achieved. The Groove Tracking facility also has potential: although it isn't a new idea, and can be achieved in DAWs such as Logic and Cubase by creating quantise templates that are generated from live takes, InTime provides a quick way of creating tempo maps for this purpose.

System Requirements Mac G3, G4 or G5 processor. Mac OS 10.2 or higher, or Mac OS 9.1 or 9.2.

Windows Windows 98, 2000, Me or XP. Microsoft Direct X version 8.0a or higher.

InTime will not work at all on Windows NT. To use InTime directly with MIDI software on the same computer, you can use a 'Virtual MIDI Device' such as MIDI Yoke or Maple MIDI Tools (especially on XP). Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



DPA 4041T & HMA5000

In this article:

DPA 4041T & HMA5000

Options & Pricing HMA5000 Preamp & Power Condenser Mic & High-voltage Published in SOS June 2005 Supply In The Studio Print article : Close window Microphone University

Preamp

Reviews : Microphone

DPA 4041T & HMA5000 pros Phenomenal sound quality, with superb dynamics and transient capability. Very low noise and extended bandwidth.

DPA demonstrate the state of the art with their latest high-voltage designs.

Interchangeable bodies for different sound characters and powering arrangements.

Hugh Robjohns

cons Very expensive. Requires a separate, dedicated power supply and preamp unit. The connecting cable between the mic and its power supply is a cost option!

summary A unique high-voltage omnidirectional microphone with interchangeable electronics and a stunning sound quality.

The Danish company DPA have a fine reputation for their range of studio microphones, several originally derived from Bruel & Kjaer measurement microphones. The 4041T is a large-diaphragm omnidirectional tube microphone intended for serious music-recording applications. It has an overall length of 170mm and a 'pencil' body measuring 19mm in diameter. The capsule is designed to be removable, but, unlike most modular mics, this is not to allow different polar patterns to be used. Instead, the omni capsule is retained and the body is exchanged for others with different impedance-conversion circuitry: high-voltage valve or solid-state designs, or a phantom-powered solidstate version.

information The complete assembly weighs around 190 grams and is Photos: Mike Cameron supplied with a clever all-metal stand adaptor which secures the microphone with a screw-down clutch action. The microphone body is simply slipped into the adaptor, and a knurled ring is tightened to impose a very firm grip on the microphone body. A 3/8-inch thread adaptor is supplied and a dedicated shockmount is available as an optional extra www.soundnetwork.co.uk — although, as a pressure-operated mic, the 4041 is not particularly prone to vibration. www.dpa

See 'Options & Pricing' box. Sound Network +44 (0) 20 7665 6463. +44 (0)20 7665 6465. Click here to email

microphones.com

The MMC4041 omnidirectional capsule employs a 24mm (one-inch) diaphragm made, unusually, from stainless steel. The capsule casing is also manufactured from stainless steel, and this combination is claimed to provide excellent immunity from ambient temperature (and thus humidity) variations. A quartz insulator isolates the condenser backplate from the housing, which is just as well

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DPA 4041T & HMA5000

given that it is polarised with 190V DC! DPA have long championed the use of high-voltage operation to provide extremely wide dynamic range and high sensitivity — in this case of 85mV/Pa, which is a good 12dB higher than for most large-diaphragm mics. The capsule specifications boast a very low self-noise of between 7dBA and 10dBA (depending on the preamp body in use), and a distortion figure of 0.5 percent at 120dBSPL, with clipping occurring at a huge 144dBSPL (peak). The microphone's frequency response extends from 10Hz to 20kHz (±2dB), although it appears to remain substantially flat well below that lower frequency limit, with a gentle on-axis boost of about 5dB. This peak is roughly two octaves wide and centred on 8kHz to provide a flat overall response in a diffuse sound field. The polar response is virtually a perfect omnidirectional pattern up to 5kHz or so, above which it narrows gradually towards a subcardioid response. This is due to the self-shadowing effect of the body to rearward sound sources. The review model was equipped with the MMP4000T high-voltage valve body. This uses a sub-miniature pentode valve configured as a unity-gain Class-A cathode-follower. This transformerless design deliberately introduces an element of 'musical' second-harmonic distortion, and endows the mic with the higher level of selfnoise mentioned in the specifications (9dBA in the case of the review model). The mic is connected to the preamp/power supply with a special six-pin XLR cable but, rather bizarrely, this is not supplied as standard with the mic. Where total sonic transparency and state-of-the-art noise performance are required, two solid-state alternatives are available. The first is another highvoltage design called the MMP4000S (achieving 7dBA self-noise), and the second is a phantom-powered version called the MMP4000SP (with 8dBA selfnoise). Both operate as transformerless unity-gain, Class-A impedance converters, and the only performance differences between the two designs are that the SP model clips 10dB earlier than its high-voltage sibling (maximum 134dBSPL), and has a slightly higher low-frequency limit (20Hz). The microphone bodies are exchanged by simply unscrewing the capsule, which is designed with a sleeve to ensure correct alignment and thus prevent accidental cross-threading.


DPA 4041T & HMA5000

Options & Pricing MMC4041 large-diaphragm omnidirectional capsule, £1862.38. MMP4000T high-voltage valve preamp body, £998.75. MMP4000S high-voltage solid-state preamp body, £998.75. MMP4000SP phantom-powered solid-state preamp body, £998.75. HMA5000 PSU and dual preamp, £998.75. DAO4110 six-pin 130V mic cable (10m), £176.25. 3541 kit, £3701.25.

Includes MMC4041 capsule, MMP4000S and MMP4000T preamp bodies, HMA5000 PSU/preamp, foam wind shield, pop screen, shockmount, six-pin 130V mic cable, and hard case. 3532S kit, £5581.25.

Includes two MMC4041 capsules, two MMP4000S preamp bodies, two foam wind shields, two six-pin cables, one HMA5000 PSU/preamp, one stereo bar, and two stand adaptors. 3532T kit, £5581.25.

The same as the 3532S kit, but with MMP4000T rather than MMP4000S preamp bodies. 3532SP kit, £4758.75.

The same as the 3532S kit, but with MMP4000SP rather than MMP4000S preamp bodies, and omitting the HMA5000. All prices include VAT.

HMA5000 Preamp & Power Supply The HMA5000 dual-channel preamp and power supply can be used with a range of other high-voltage DPA mics (4003, 4004, 4012, 4016) as well as with the 4041T and 4041S. It supplies 130V for the electronics, plus a 190V capsule polarising voltage. A separate balanced 6.3V supply is provided for the valve's heater. The mic's output signal is unbalanced, but its relatively high level, combined with the special construction of the six-pin XLR connecting cable, seems to provide excellent protection against interference. The HMA5000 measures 52 x 133 x 200mm (hwd) and weighs a substantial 1.85kg. The front panel is equipped with two seven-pin XLR sockets (although the central seventh pin is blind) to accept the outputs from a pair of microphones. Two rotary gain switches are scaled from -20dB to +30dB in 10dB steps. The rear panel carries a pair of three-pin output XLRs (sensibly marked as line-level outputs), each with an associated polarity-reversal switch. An IEC inlet accepts either 120V or 230V mains supplies with automatic switching. The electronically balanced line-level outputs are capable of driving up to 300 metres of cable, if


DPA 4041T & HMA5000

necessary, with a maximum peak signal voltage of a massive +32dBu! Strangely, when used with the 4041 microphone (either body type), the output polarity of the HMA5000 is inverted (relative to the 'pin two hot' standard), but it is correct when used with all the other high-voltage models. However, if absolute polarity is important to you, simply flipping the rear-panel toggle switches corrects the situation.

In The Studio I am quite a fan of pure omnidirectional mics — when employed in suitable situations — as they tend to provide an extremely natural and transparent sound quality which even the best cardioid mics simply can't approach. Furthermore, a good omni's frequency response will extend at least an octave lower than that of a cardioid, which is invaluable when recording wide-ranging instruments such as pipe organs or pianos. With the preamp gain set to 0dB, and after allowing ten minutes for the microphone to warm up, I initially tested it by walking and talking. It was immediately apparent that the frequency response extends smoothly to both extremes, and the polar response is a well-defined omni over most of the range, with a very smooth off-axis response falling gently towards The HMA5000 power supply and dualthe rear at extreme high frequencies. channel preamp. Noise levels were impressively low, and on spoken voice the claimed harmonic distortion of this valve body was extremely subtle. Next, I put the 4041T up against my favourite Sennheiser MKH20s (hooked up through a GML mic preamp), placed side by side in front of a baby-grand piano. The Sennheiser has phenomenally low distortion thanks to its unique symmetrical capsule design, and the slight richness associated with the 4041T's modest second-harmonic distortion became more apparent in comparison. Of particular note was the 4041T's ability to handle extreme dynamic ranges, drawing out superb ambient detail, and the way it could accommodate huge transients with ease — something made very obvious on the meters of a digital recorder. It's a good job I routinely record with plenty of headroom, especially in the light of the preamp's 10dB gain steps. Compared to the MKH20, the 4041T sounded fractionally more silky at the top end and a little less congested when (mis)placed close to the piano dampers to capture rather excessive hammer transients. On occasions I perceived a mild mid-range richness, presumably due to the harmonic distortion, but this characteristic remained remarkably subtle.


DPA 4041T & HMA5000

This is a very expensive microphone, especially when the required power supply is added into the equation — although the fact that the HMA5000 provides a pair of very high-quality preamps should be borne in mind when comparing prices. In extremis I think the DPA 4041T outperforms the Sennheiser MKH20, but the law of diminishing returns applies here. The sonic differences were extremely small, even at the limits, and only top-quality rooms, instruments, and performances would justify the expensive benefits of this flagship DPA microphone. However, it's good to know that the ultimate performance of microphones is so very high, and that it continues to improve with the increasing sophistication of new technology.

Microphone University The DPA web site is a mine of useful and instructive information concerning everything to do with microphones and microphone technique, much of which has been gathered under the tag of the Microphone University. Following the links through the site, the reader is presented with several options, including a Microphone Technology Guide, a Pro-audio Dictionary (with over 80 entries), and practical advice on both Stereo Techniques and Surround Techniques. In addition, there is an Application Guide which lists nearly thirty different instruments and ensembles, as well as offering some helpful information about the sound characteristics of each, with suggested miking techniques and recommended DPA mics. Each of the main subsections mentioned above is subdivided into numerous chapters. For example, the Stereo Techniques section has ten chapters detailing each variation of spaced and coincident techniques: A-B, baffled, binaural, Decca tree, DIN, NOS, ORTF, X-Y, Blumlein, and M&S. The Microphone Technology guide includes chapters on fundamental issues like balanced and unbalanced lines, as well as more complex subjects such as the technical differences between omnidirectional and directional mics, large and small diaphragms, transformercoupled and transformerless designs, and phantom or high-voltage mic-powering arrangements. All in all, there is some extremely good information to be found here, with practical examples and, in the case of the Application Guide, some excellent photographs to help illustrate the suggested mic positions. Published in SOS June 2005


DPA 4041T & HMA5000




Dynaudio BM5A

In this article:

Dynaudio BM5A

Driver Construction & Signal Active Monitors Conditioning Overload Protection Circuitry Published in SOS June 2005 Cabinet Design Print article : Close window

Dynaudio BM5A £881

Reviews : Monitors

pros Well built from high-quality parts. Useful EQ facilities. Accurate and revealing. Goes loud without complaint.

cons

Dynaudio have concentrated on high-quality internal and external engineering to create a communicative new ported monitor which isn't afraid to go loud.

Bass a little sluggish.

summary

Phil Ward

The BM5A is a high-quality professional product that is unlikely to disappoint. Worth the extra expenditure compared to apparently similar entry-level products.

You can't help but wonder just how many compact nearfield monitor products the market can support. Both new entrants and established manufacturers bring products to the market so fast that keeping up, for anybody, is little short of impossible. One information manufacturer that falls firmly into the 'established' £881.25 per pair camp is Dynaudio, and the company's BM5A, the including VAT. TC Electronic UK +44 (0) subject of this review, is in many respects typical of the rapidly expanding field. 800 917 8926. +44 (0)800 917 6510. Click here to email www.tcelectronic.com www.dynaudio acoustics.com

Driver Construction & Signal Conditioning Photos: Mark Ewing

The BM5A is a compact, two-way reflex-loaded system with a 170mm bass/mid-range driver and a 26mm fabric-dome tweeter. A rear-mounted panel carries two 50W amplifiers along with the necessary connection and control facilities. The BM5A's UK price is pitched towards the upper end of the range for the type of product, and it's clear from the speaker's components, engineering, and set of features that Dynaudio have aspirations for it that extend above the project studio to what they'd probably see as more professional applications. Dynaudio's drivers, for example, have always been engineered specifically for the high-power and low-compression demands of professional monitoring, rather than being inexpensive domestic hi-fi units adapted for the job, and the BM5A continues this tradition. The bass/mid-range driver features the company's external voice-coil construction, which results in a

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Dynaudio BM5A

much larger coil than would conventionally be found in a driver of this size — a feature likely to reduce thermal compression significantly. The tweeter is engineered to offer both high power handling and low compression, and it operates over a wider bandwidth than would normally be the case. The standard of engineering necessary to achieve these characteristics doesn't come cheap. While the basic set of connection and control facilities fitted to the BM5A is nothing out of the ordinary, the monitor seems subtly tailored towards the professional applications. Firstly, there is just one balanced XLR input socket provided (no unbalanced jack sockets here!), and, secondly, there is no gain control provided beyond an input-sensitivity switch with +4dB, 0dB, or -10dB settings. The lack of variable gain control might, on the face of things, seem remiss, but in any truly professional studio setup there would almost certainly be a convenient monitor gain control facility somewhere else in the signal chain. The lack of variable gain controls also brings the benefit that there's less likelihood of unbalanced left and right channels. Continuing the professional theme, the BM5A incorporates some thoughtfully considered equalisation presets selectable via small switches on the back panel. A high-pass filter, designed to aid the BM5A's integration with subwoofers, can be selected to operate at 60Hz or 80Hz. [During the review period news reached the SOS office that TC Electronic are about to release a new subwoofer designed to operate with the BM-series monitors, including the BM5A — Ed.] Also part of the design will be a low-level LFE input/output for daisy-chaining of subwoofers if necessary. A ±2dB low-frequency shelf can be switched in to provide some tailoring of the speaker's bass level to its immediate surroundings and to personal preferences. A -2dB or -4dB mid-range notch filter provides approximate compensation for BM5As located above large reflective surfaces — mixers or desks, for example. And finally, a ±1dB high-frequency shelf enables a bright in-room balance to be warmed or a dull one to be brightened. While broad-brush EQ facilities such as these can never recover an inappropriately positioned speaker or poorly performing up-stream electronics, they do provide a usefully comprehensive ability to tweak the balance of the BM5A to suit specific installations.


Dynaudio BM5A

Overload Protection Circuitry One of the many clear advantages of active speakers over passive is that the interface between the drivers and their driving amplifiers is firmly defined. Knowing exactly how powerful the amplifiers are and precisely the thermal and excursion-limited power handling of the drivers means that effective and worthwhile driver protection can be incorporated. With passive speakers, effective protection is almost impossible — it just can't accommodate the almost limitless variety of possible amplifier and driver combinations — but with active speakers it can work well. The BM5A takes advantage by incorporating a full complement of protection features for both drivers. In the case of the tweeter, a mute circuit kicks in if the current delivered exceeds a preset level. And in the case of the bass/midrange driver, a limiter circuit reduces the excursion if the extremities are approached. Along with the driver protection, the BM5A's amplifiers are also thermally protected and will shut down if they get too hot. Bearing all this protection in mind with the inherent high power handling of the drivers, I'd expect the BM5As to be all but unburstable. Should they approach bursting point, however, an orange front-panel LED indicates bass/mid-range driver protection and a red LED shows amplifier temperature protection.

Cabinet Design One result of the huge number of compact nearfield monitors available is that it becomes harder for manufacturers to differentiate products in terms of their appearance — and this, frustratingly for the manufacturers, at a time when product aesthetics is becoming a more significant factor in success or failure. With the BM5A, Dynaudio have gone for a simple, clean, engineered style that to my mind dovetails well with the market sector the product is aimed at. There's a no-nonsense, serious look to the BM5A that comes from it's heavily engineered and aluminium-look driver chassis, deep cabinet proportions, and semi-gloss lacquered front panel. If the BM5A were a car, I suspect it would wear a BMW badge! Its only concession away from rectilinear shapes is a subtle profiling of the front panel edge thickness that both provides relief from straight lines and perhaps helps a little to reduce acoustic edge diffraction. As far as it goes, I like the appearance of the BM5A. Internal construction of the BM5A continues the high-quality professional theme established by the external appearance and components. I was pleased to see, for example, a reflex port that's flared both internally and externally. And the BM5A is well screwed together, my only minor gripe being the insubstantial nature of the driver fixing screws — wood screws that would only need a little over-tightening before they stripped their MDF pilot holes. After all this positive news it would be a shame if the BM5A disappointed when switched on. But fear not, it didn't. It offers a pretty accurate and professional reproduction of the material it's fed, and despite gentle-slope crossover filters, which result in a wide band of driver overlap, it is relatively untroubled by vertical dispersion inconsistencies. In balance terms I felt happiest in my listening file:///H|/SOS%2005-06/Dynaudio%20BM5A.htm (3 of 4)9/28/2005 2:33:31 PM

Dynaudio BM5A

environment with the EQ presets set flat, but experimentation with them confirmed their usefulness. Along with being well balanced tonally, I found the BM5A to be revealing and detailed through the mid-range and high end, with an engaging coherence that helped make sense of the material. Hi-fi enthusiasts would call it 'communicative', which is actually not a bad term to explain what I mean. The BM5A also seemed, as expected, completely untroubled by being asked to play loud. Its balance remained consistent and signs of distress only became apparent at a volume level in my smallish room that I'd personally not be able to tolerate for long. If pushed, I'd make just one critical observation; on occasions I experienced the usual feeling when faced by a ported speaker that bass transient accuracy had suffered a little in the search for bass extension. So what if, rather than the market for compact monitors being full to overflowing, the Dynaudio BM5A were the only contender; would I hanker after some yet-to-be-designed alternative? Nope, I'd be perfectly happy with the BM5A.

The speaker's reflex port, flared both externally and internally, appears at the top of the rear panel. The only audio input is on balanced XLR, and level control is limited to a three-position switch. The three EQ switches allow you to tailor the tonality of the speaker within your room.

Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Glaresoft iDrum

In this article:

iFlyer Tweaker's Delight Basics Keeping Up Appearances Programming Sound Management Use As A Plug-in Conclusions

Glaresoft iDrum £39 pros Intuitive interface. Great sounds and presets. Multi-channel AIFF export options. Multi-channel operation in RTAS format. Jaw-dropping price for a Pro Tools-compatible plug-in.

cons VST version in still in the pipeline. No fancy time signatures, gentlemen please. Mac OS X-compatible only.

summary Quite simply, iDrum is brilliant value, and should be snapped up before someone comes to their senses and doubles the price.

Glaresoft iDrum Virtual Drum Machine [Mac OSX] Published in SOS June 2005 Print article : Close window

Reviews : Software

Espousing the simple, easy-to-use Apple-type philosophy seen in tools like Garage Band, Glaresoft's virtual drum machine has you piecing together beats in minutes. And it costs just £39! We check it out... Nicholas Rowland

Many more years ago than I care to remember, I came face to face with my first drum machine, a rather odd little number called the TR808 from a Japanese company with a name rather like my own. As a keen drummer myself, I was distinctly underwhelmed by the flabby bass drum, the splatty snare and the hi-hat that sounded like a wet snake sneezing. No, what fascinated me was how simple it was to work with, thanks entirely to its programming interface of 16 multi-coloured buttons, which assigned the currently selected drum sound to the steps of your choice within a 16-beat pattern. I'm not sure who first came up with this matrix system of drum programming — it certainly predated my encounter with the TR808 — but it's always struck me as brilliantly simple, requiring little or no knowledge of drumming, just a spare finger, a good ear and a sense of rhythmic taste.

information

Clearly, it's this type of old-school programming interface, that has inspired iDrum, a new sample-based, stereo virtual drum machine application for today's computerbased generation. Or to be more precise, today's Apple Mac-based generation as iDrum, subtitled 'the drum machine for OS X' by its makers, is designed specifically for OS 10.3. As you've no doubt spied, with its little 'i', the program is clearly attempting to align itself with Apple's iLife suite of digital-lifestyle products. But just to make it clear, www.maudio.co.uk it originates not from the big Apple itself, but from independent software house

£39 including VAT. M Audio UK +44 (0)1923 204010. +44 (0)1923 204039 Click here to email

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Glaresoft iDrum

www.glaresoft.com

Glaresoft, and is being marketed, sold and supported worldwide by M Audio.

Test Spec

iFlyer

MAC REVIEW SYSTEM 1.8GHz Apple iMac G5 with 1GB of RAM running Mac OS v10.3.8. Glaresoft iDrum v1.0.2 (stand-alone)/1.0.3 (RTAS). Digidesign Pro Tools LE v6.7.

As well as operating as a stand-alone program, iDrum will also work as an instrument plug-in with any other Audio Units host, which includes Apple's Garage Band, the (notso) 'entry-level' MIDI + Audio sequencer which is part of iLife, plus Logic Pro and Logic Express (also to be found under the Apple family tree) and MOTU's Digital Performer (v4.12 onwards). However, my obligatory trip to the Glaresoft web site to register the review package and check for the latest software version revealed two pieces of interesting news. Firstly, an RTAS version is already shipping, thus making iDrum relevant to anyone using Pro Tools v6.4 and above. Second, Glaresoft are looking for beta testers for a VST version. So users of Cubase and Tracktion (to name just two possible VST-capable hosts) should stay tuned, as everything I say is likely to come your way at some point in the future. As a physical product, iDrum consists of a box with a CDROM and a multi-lingual quick start guide that covers installation, getting started and a few basic principles of the software for those new to drum-machine-style programming and indeed, for those new to drum machines full stop, as so many people have now grown up working exclusively with drum loops. Installation itself is a two-stage process — you put the software on your machine, then connect to the Glaresoft web site to register with the serial number. After successful authorisation, you are sent log-in details for an iDrum account which gives you access to a download area containing the various software releases to date (including that RTAS version), plus fixes, extra sounds, more drum patterns and an extra 'skin'. You can also sign up for email notification of when new things are added and as Glaresoft's plan seems to be to develop something of a user community around iDrum, this is a recommended move. If you happen to have a machine that's not connected to the Internet, then registration is a bit more convoluted, with emails passing back and forth, but the general principle is the same. And in the meantime, you have a 30-day period to run the software in full. In terms of the software contents, what you get is the application itself, around 38MB of drum and percussion sounds (and a further 5MB on registering) plus some 130 preset patterns (and more of these when you register too). Although the printed manual is slim, there's an excellent set of help screens, available both from the web site and integrated into the application. These cover all the functionality in fine detail, and make you realise that there's more to iDrum than first meets the, er, 'i' (no more iPuns now, I promise).


Glaresoft iDrum

Tweaker's Delight Aside from the control over volume and pan from within the Channel 'strip', iDrum also offers a basic set of non-destructive sound-shaping tools to further tailor samples to your liking. A click of the 'i' button on the main screen button reveals a hitherto hidden sliding panel with controls for Pitch and Decay, plus low-pass and high-pass filters which offer some EQ capabilities. These controls are on sliders, however, and adjustments have to be done by ear, as there's no numerical calibration of the values. You can also grungify your samples by changing the bit depth, with a choice of 32-bit, 12-bit, eight-bit and four-bit settings. While iDrum's default sounds are all one-shot samples, of course the whole beauty of the program is that you can assign any WAV or AIFF file you like to any channel (up to a maximum of 2MB per file per channel). And that obviously that might include rhythm loops. iDrum's 'Fit' button instantly changes the the playback speed of the loop so its length matches the bar at the current tempo. At this price, it's too much to expect that iDrum might apply some clever beat-slicing or elastic audio manipulation, so the pitch of the loop does go up or down accordingly. But it works fine, so who's complaining? One final control on this panel, again borrowed from hardware drum machines, is the ability to assign sounds to one of two choke groups. This is normally used to make hihats sound more realistic by ensuring that the long open hi-hat sounds are cut off by a closed hi-hat sound. But in fact it can be used creatively for gated rhythmic effects, particularly if you use a sample of silence to cut off longer sounds.

Basics Manuals or no, it doesn't take long to get to grips with iDrum's basic principles, particularly if you've ever used a hardware drum machine. On first loading up, you are presented with a window containing eight drum Channels, which in the default skin look a bit like a rack of effects modules. The left-hand side of each module gives you the controls for assigning samples (of which more anon) plus the name of the currently assigned sample, and a software trigger button to audition it. Each channel has its own Volume slider, Mute and Solo buttons and a Pan control. In the right half of each module are 16 software 'buttons' or rather graphical blocks representing the software equivalent of the step-time programming buttons you would find on a typical olde-worlde beat box. At the bottom of the window are the transport controls (Play/ Stop, Record) the Tempo control and readout, controls for the Song Mode, plus some other things, which we'll address in a minute. While some functions such as loading and saving patterns are handled from the Apple menu bar, most of the housekeeping — and indeed all of it when iDrum is used as a plug-in — is done from a contextual menu opened by clicking an icon in the top left-hand corner of the application (see the screenshot on the second page of this article). On startup, iDrum preloads with a default Kit (Glaresoft's term for a pattern and its associated sounds). This Kit can be customised if you wish, so you can always be ready for action with your choice of sounds as soon as the software boots up. Checking out the 500 or so factory samples reveals them to be very good indeed — here I was pleasantly surprised, because the price led me to believe that somehow they would sound 'cheap'. And as there are loads of them, even simply as a source of file:///H|/SOS%2005-06/Glaresoft%20iDrum.htm (3 of 8)9/28/2005 2:33:35 PM

Glaresoft iDrum

drum samples, iDrum is worth its weight in coinage of the realm. Both in terms of the sounds and the preset patterns, the emphasis is very much on 'urban' styles — hiphop, trip-hop, techno, drum & bass and so on — which is obviously where the interests of the program's designer lie, and no doubt most of the target audience too. Of course, the whole point of iDrum is that if these are not to your taste, you simply load up sounds that are and get programming your own beats.

Keeping Up Appearances

Two of iDrum's alternative 'skins'.

As with a lot of MP3 players, you can apply different skins to iDrum to change its appearance. You get two extra skins with the program and a third can be downloaded once you've registered — and if you are handy with Photoshop, you could even create your own. This shouldn't just be seen as a gimmick for geeks with no social life; if you use multiple versions of the plug-in, using different skins can be a useful way of identifying the different instances.

Programming The principles of programming are simple — you just set iDrum playing and then click on a button to trigger the associated sound on the appropriate beat. As well as by clicking the buttons with the mouse, you can trigger sounds from the computer's keyboard and from an external MIDI device, such as a keyboard or drum pads. For this latter purpose iDrum gives you the useful ability to program the MIDI note numbers assigned to each iDrum channel. It even has a MIDI Learn function, so that a channel will be assigned to whatever incoming MIDI note you throw at it — a great time-saver when you're setting up iDrum with external devices. Press Record as you play and you also are able to tap patterns into iDrum in real time. Being very much of the 21st century, iDrum is velocity sensitive and offers a neat system of programming velocity on each beat. The velocity on an active button is displayed in the form of a bar-graph meter and you simply click on a button, hold down the mouse and then drag up or down to increase or decrease the value. What's more, if you wanted to (say) create a crescendo effect on a 16-beat snare drum roll, then you can hold the mouse down and drag it across the face of the several buttons file:///H|/SOS%2005-06/Glaresoft%20iDrum.htm (4 of 8)9/28/2005 2:33:35 PM

Glaresoft iDrum

to 'draw' virtual velocity curves over several beats at once. It's very slick and the interface gives you great visual feedback, although some advanced beat artists might bemoan the lack of a precise numerical indication of the velocity value. Although you only see 16 buttons at once, each pattern is made up of 64 steps as standard and in the default view the buttons represent the beats patterned on the 16th notes. But you can easily program and edit beats assigned to 32nd and 64th notes by using the Note Division View control at the top of the screen to view the pattern at the higher resolutions and then jump between groups of 16 steps. What's obviously missing is the ability to program in odd time signatures: iDrum is a straight four-on-thefloor kind of guy and proud of it. As you can count the number of musicians regularly working in 5/4, 7/8 or 15/32 on the fingers of both elbows, I guess this is OK for a lot of the audience most of the time. But plenty of mainstream music is written in 3/4, 6/8, and 12/8, particularly when it comes to Latin-style rhythms, so I really think this should be addressed. The normal approach on drum machines is through a control that allows you to change the number of steps per pattern from 16 to 12, and it strikes me that this could easily be adopted for iDrum's virtual interface. One feature which has been adopted from vintage hardware is the Swing control, an adaptation of what was often known as the Shuffle function on machines of yore. As either name might suggest, this is designed to give a more jazzy 'dotted-note' feel to drum patterns. On vintage drum machines, it could have mixed and quite extreme results, but with iDrum — where it takes the form of a percentage readout which can be adjusted via a stepper control between 0 and 26 — it works very nicely. Lower settings add a subtle humanising touch to patterns; higher ones introduce a tangible element of funky syncopation, which is great for jazz and trip-hop.

The main file-saving and housekeeping functions are accessed via the icon in the top left of the application window.

Another feature borrowed from the world of hardware is Song Mode. Each pattern can contain up to 99 variations and these can then be chained together in various combinations to create songs of up to 999 bars. As you would expect, the principle is very similar to that on a real drum machine: you define the number of bars you want in your song and then assign different pattern variations to the different bars. When I used to use real drum machines, I always remember this being a very tedious part of the creative process, but somehow with iDrum it's a lot easier and faster. I think it may be because with iDrum you can change the number of bars in a pattern and edit the order of patterns very much on the fly. iDrum also offers the ability to build up variations of your main pattern very quickly: not only can you copy and paste entire patterns, you can also copy and paste individual parts between patterns. This is all good stuff, but iDrum has yet more up its virtual sleeves. The first of these is the MIDI Drag button: not to be confused with the Musicians' Union annual Hi-tech file:///H|/SOS%2005-06/Glaresoft%20iDrum.htm (5 of 8)9/28/2005 2:33:35 PM

Glaresoft iDrum

Cross Dressers' Ball, this is actually a function which allows you to simply drag and drop the current pattern as a MIDI file straight out of the application on to your desktop. In fact, when using iDrum as a virtual instrument within Garage Band and Logic, you can cut out the middle man and drag the MIDI straight on to their timelines. With other applications (such as Pro Tools) you can still use this function, but you need to import the data as a file. Another surprise is to be found in the audio export department, where, at the click of a button, iDrum offers the ability to bounce either the current pattern or current song as a stereo AIFF file. As with many sequencers' audio bounce-down features, I expected iDrum to export only the pattern channels that were active, but it soon became clear that regardless of the status of the Mute or Solo buttons, the whole pattern is bounced down to the audio file. This isn't a problem, though, because iDrum also has the ability to export each channel as an individual audio file. This is incredibly useful if you want to then assign the audio drum parts to different tracks on your sequencer for, say, further processing by effects plug-ins, or for assigning to separate hardware outputs. You can of course re-import the various sound files back into iDrum where they can then be triggered and manipulated as loops, offering yet more creative potential.

Sound Management There are a number of ways of assigning a sound to a channel: you can step through them, or you can hold down the folder icon for a pop-up list of all the sounds available for that channel. Basically, iDrum organises sounds via their root folders: bass drums, snares, cymbals and percussion being the default folders that iDrum creates when you install it. So when you assign a snare sound to a channel, for example, bringing up the pop-up list will show you just the sounds in that 'snare' folder. Similarly, in the same example, if you used the Step Through control, the list you were stepping through would contain just the snare sounds. As there are lots of default sounds, this approach makes a lot of sense, particularly when you're at the stage of auditioning every single sample to see what they sound like. Obviously, using the Apple folder structure intelligently, you can then start to rearrange your sounds into collections that are more meaningful to you. Since iDrum can get quite upset if, on loading a pattern, the sounds it's looking for are not in the right folder, you can also choose to save the sounds as part of a pattern, so you always know that it's going to load up properly.

Assigning sounds to channels from the drop-down list in the RTAS plug-in version of iDrum.

There are several ways of creating new channels too. Dragging and dropping samples from the desktop (as mentioned in the main text) is one of them. But this functionality is also to be found in the contextual menu, and the '+' and '-' buttons at the bottom left of the main window offer another route to control this functionality. One thing the manual isn't clear about is just how many channels you can have. As there doesn't seem to be any software limit on the number of channels per pattern, I


Glaresoft iDrum

guess it would be more a question of where your hardware would max out. However, the real practical barrier to creating a mega-voiced beat box is that iDrum works in a fixed-size window that only has room to display eight channels at a time. Once you go beyond this, you have to start scrolling up and down within the window, so keeping tabs on all of your channels would start to get a bit unwieldy. To overcome this, you can change the order of channels simply by dragging the channel modules up and down.

Use As A Plug-in So far, my observations have been mainly made on the use of iDrum as a stand-alone instrument. When used as an instrument plug-in, the program's overall behaviour is largely the same, with the application appearing in whatever wrapper the host might provide for plug-ins. There are, however, some differences in functionality which are worth noting. The first of these comes in the form of the Slave To Host radio button which as its name suggests, hands over control of the transport buttons and the Tempo to the host application. Usefully though, this is all iDrum really surrenders. You still retain control over pattern and song programming, so you can construct a rhythm track as you listen to the rest of your sequence. And as an added bonus, when using iDrum as a plug-in, your song is always saved with the host's project file or with your iDrum file, even when you turn song mode off. Running under Pro Tools, a little more icing is applied to the operational cake as here you discover that iDrum can be used as a multi-channel plug-in within those versions of Pro Tools that support such a mode. This enables you to assign each Channel to either the main stereo out or one of seven auxiliary stereo outputs. These can in turn can be re-routed into Pro Tools via their own input channel with individual inserts and sends, allowing you to apply individual processing to iDrum's sounds. Incidentally, iDrum seems to be pretty kind to your CPU too. I found it possible to run a couple of instances on my 1.8GHz G5 iMac under Pro Tools without making much of a dent on the CPU meter.

Conclusions Given this program's name and its mass-market price, it would be understandable if some people dismissed iDrum as something of a techno plaything. But anyone who has used Apple's iLife products will know that Apple's genius has been to marry deceptively easy-to-understand visual interfaces with technology that is anything but basic. iDrum is very similar; here too, surface simplicity conceals hidden depths. If you think about it, what iDrum offers is a combination of intuitive drum sequencer and audio-file trigger, a sound editor and resampling tool, a MIDI file rhythm-pattern generator and an audio file-management program. On top of this, you get an excellent contemporary drum sample library, which, given the going rate of sample CDs, is probably worth at least half of the £40 UK asking price. file:///H|/SOS%2005-06/Glaresoft%20iDrum.htm (7 of 8)9/28/2005 2:33:35 PM

Glaresoft iDrum

Of course, it's not perfect. For example, if you're into advanced sound design then you'll find its sample-manipulation facilities rather basic. The lack of ability to program in odd time signatures has also already been mentioned, as has the absence of beat slicing. But the sub-£50, mass-market price really blows away even these (very few) shortcomings.

Using the RTAS version of iDrum in multichannel mode under Pro Tools (see page 44).

As a recent convert to Pro Tools, I found iDrum an answer to my prayers, giving me a easy-to-use step-time drum-sequencing program which doesn't load the CPU like Reason Adapted (now included in the Pro Tools bundle) and which makes up for the lack of the rhythm programming page that has always been one of the best features of Cubase, my previously preferred package to date. So for a Pro Tools user like me, used to paying 'professional' prices for plug-in packages (and suffering attacks of annoying alliteration as a result), the iDrum message is simple — don't think about it, just buy it. And for users of other sequencers, the message is equally simple; go and try it now, at the very least. With a fully-functioning 14-day demo on offer via either the M Audio or the Glaresoft web site, you'd be iMad not to. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Korg KPE1 Kaoss Pad Entrancer

In this article:


Mixing It Audio-visual Processor & X-Y Controller KPE1 Specification Published in SOS June 2005 In The Box VJ Web Links Print article : Close window KF4 Specification Reviews : Processor Video Sampling Korg Krossfour KF4 Vision Mixer Connection Points Right On Cue The Kaoss Pad Entrancer builds on the success of the Using The KPE1 & KF4 Live Kaoss Pad 2, offering the same versatile X-Y pad-

Korg KPE1 £599 pros Simple to use. Rugged construction is excellent for live use. In most cases, the KPE1 is a more suitable solution for live video processing than software-based alternatives.

driven audio processing capabilities, but adding video processing to give you real-time control over both sound and vision. Paul Gilby

There have been superstar DJs all over the world for years, but the cons concept of the VJ (video jockey) has Separate power supply. taken longer to get going, although it's been big in Japan for a while. Now that No 'scribble strip' below the Program Memory buttons. the idea is beginning to find favour in other cultures too (see the links near summary the end of this article), recent Don't think of the KPE1 as developments like Roland's V-Link being purely for DJs, or indeed VJs; as well as connection protocol (which permits the incorporating all the excellent sync'ing of video to other MIDI gear), audio and X-Y controller features of the Kaoss Pad 2, it and Korg's new affordable videoorientated processors and mixers can be used by anyone who wants to add live video effects begin to make sense. The new Kaoss to their performances. Pad Entrancer under review here adds Studio photos: Mark Ewing. Tour shots courtesy of Bill Nelson. video functions to those of the earlier information audio-only Kaoss Pad 2, which seems KPE1, £599; Krossfour, to have found a niche predominately in the DJ market. However, as Mike Senior £499. Prices include VAT. pointed out in his November 2002 SOS review of the Kaoss Pad 2, you shouldn't Korg UK Brochure Line +44 (0)1908 857150. dismiss this unit if you're not into DJ'ing. With a selection of 100 audio effects and +44 (0)1908 857199. the ability to manipulate them in real time via the X-Y touch pad, it's actually a Click here to email fantastic way to add exciting sonic textures to any live performance, regardless of musical style. www.korg.co.uk As the Entrancer includes the same audio functions from the KP2 Kaoss Pad, I'm

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not going to go into detail about those aspects of this unit — you can find Mike's KP2 review on the SOS web site at www.soundonsound.com/sos/nov02/articles/ korgkp2.asp for that information. In broad terms, the Entrancer includes 100 audio effects, 100 video effects, two sample memories, eight Program memories, a tempo-sync function and an X-Y touch-pad for real-time manipulation. Whereas the KP2 gives you only audio effects such as filters, delays, reverbs, a vocoder, a synth, and sample-manipulation features, the Entrancer's video-processing functions offer manipulation of the video sources by adding video noise, stretching, freezing, splitting, spinning, or colouring your images, generally allowing you to distort the picture in all sorts of interesting ways. Whilst there are dozens of software VJ programs out there, they tend to be far more complex, primarily because many of them integrate functions to generate raw material as well as the means to process it. The Entrancer is really all about real-time hands-on video processing, and in this role it excels and is truly creative.

Mixing It You can use the Entrancer purely for audio processing, as with the KP2, or solely for video processing, or both at once. Audio can be used as the source to modify the video imagery in real time either directly or by using the Tap/BPM The Headphone socket, associated Volume control, second main (video-only) input and function. You can call up any of the all-important Input 1/2 toggle selector switch 100 video effects which can be are all found on the front edge of the KPE1. allocated to eight different Program Memories for fast recall, making it possible to move through a variety of video effects as the music progresses. In addition, you can work with the X-Y pad to create a range of fabulous visual effects. All of these 'instant' The rear edge of the KPE1 is busier, with the effects make it very easy to add live connectors for the main audio and video output, the other audio and video input, MIDI visuals to any show and give you control over how images change in real and power. The small switch in the middle allows you to determine the format of the time. In addition, if you're into totally video the KPE1 will be using: NTSC is the programmed performances using US standard, while PAL is used in Europe sequencers and a bunch of MIDI gear, and elsewhere. you can use MIDI to control both the audio and video effects selection. For a one or two-person show, this is great for adding professional visuals to your performance. With just an LCD video projector, a DVD player and a Kaoss Pad Entrancer, you can produce a mighty visual experience to back you up on stage for very little effort — and it looks cool, too! To get something out of the audio side of the Entrancer, it's best to feed it



something. However, the video side is different, in that the unit can produce automatic video effects without having to process any video signal; you can just plug it in and watch the pretty pictures. These visuals are a collection of simple waveform, spectrum-analysis and oscilloscope-type imagery which the sound controls. The unit really comes into its own when you feed it a video signal, be it from a live video camera or moving imagery from a VHS or DVD player. Movement is the key here, as it's the basis for generating really exciting results. By manipulating raw source material, you can impose different video effects and modify the output in real-time using the X-Y pad. When you perform a pad movement you like, you can even record the pad motion and keep it repeating. The two Sample memories let you record up to six seconds of motion each and allow you to 're-effect' them as source material in your performance. For example, you can sample a still image (a logo, say) into one memory and mix it into the video stream as required throughout a show.

KPE1 Specification Effects: 100 video, 100 audio, 100 combination audio/video. Inputs: Two (Input 1, on the rear of the KPE1, accepts a composite video signal via a phono connector, and stereo audio via left and right phono connectors, while Input 2, on the front edge of the KPE1, is video-only, accepting an S-Video signal via a mini-DIN socket or a composite video signal via a phono. Inputs 1 and 2 are switchable). Main Outputs: One (on the rear of the KPE1. An S-Video signal is available via mini-DIN, composite video is available on a phono, and individual Left and Right phonos handle the stereo audio output). Headphone output: One (via quarter-inch stereo jack, on the front edge of the KPE1, with associated Volume control). Sampling frequency: 44.1kHz. A-D/D-A conversion: 20-bit linear. MIDI connectors: In & Out. Video format: Switchable between NTSC (the US standard) and PAL (the European standard). Power supply: 7V DC (via external AC adaptor).

In The Box The KPE1 is all about performance, and at the heart of this, as on the earlier KP2, lies the X-Y touch pad. Moving your finger around the Entrancer's pad horizontally or vertically creates smooth changes in the sound, the video or both, whereas tapping the pad in different positions gives you more dramatic changes. The pad also changes colour when you touch it, helping to create a very cool mini light show on stage. As far as the video side is concerned, the pad allows you to change various key parameters of the currently selected video-processing



effect; in total, the KPE1 offers 100 different effects grouped into themes. Some of the effects are self-explanatory and will be familiar if you've ever worked with video processing before, such as Emboss, Wipe, Mirror, Squash and Spin, while others are more cryptic, such as Dispersal, Sketch and Random Snap. Typically, moving your finger around will change the current image's scale, coloration, amount of stretch, spin speed, and forward or backward movement (this is great for 'scratching' a video sample), and the FX Balance knob tends to control the depth of the effect. When the unit is operating in 'Combi' mode, where audio and video manipulation is simultaneous, choosing a particular video effect also calls up a complementary audio effect (or what Korg deem to be complementary, at any rate — the choice is not left to the user). For example, auto-rotation of the video image is combined with a tape-echo treatment, while the Emboss video effect is combined with a phaser. The speed of movement is instant, and there's no discernible lag in the video display — if you wish, you can swirl your video around, spinning and flipping it until you make your audience ill. Oh, and on that point, Korg do include a warning about the very real danger of visually induced epileptic fits, so don't overdo it.

VJ Web Links If you'd like to find out more about VJ'ing, there are plenty of sites on the Internet with more information. Here are three starting points: CHANNEL 4 This VJ microsite was recently launched by UK TV company Channel 4. www.channel4.com/pixnmix

AUDIO VISUALISERS A good site which provides lots of info about the history of video synthesizers and VJ software. www.audiovisualizers.com

VJ CENTRAL A community site covering the VJ scene.

There are eight Program Memory buttons www.vjcentral.com where you can store any of the 100 effects for instant recall. This is particularly important in a live performance, as you don't really want to be dialling through 100 options looking for what you want. Unlike the two Sample Memory locations, these Program memories are stored when the unit is switched off, and will remain so until you overwrite a memory location with another choice. From a performance point of view, whilst the eight memory buttons are clearly backlit and well spaced, the lack of a 'write' or 'scribble' strip underneath them means you have to resort to sticking a length of masking tape on the unit and write any prompt notes on that instead. However, on the positive side, Korg have thought through the ergonomics of the KPE1 and must be congratulated for grouping all the buttons and switches in very handy positions and making them all backlit. This made the unit a joy to use live in the typical low lighting conditions on-stage during the recent Bill Nelson tour (see the box on the next page for more on this). Apart from the X-Y pad, Program Memory buttons and the Hold, Sample,and Rec/ Stop buttons along the front edge, the other main performance tool is a centre-



sprung toggle switch on the right labelled 'Pad Motion' in the upper position or 'Mute/Freeze' in the down position. Depending on your style of VJ'ing, this switch could well become the most frequently used control on the whole unit; it can really bring a performance to life. In the Audio mode, it allows you to mute the sound on and off so that you can superimpose your own rhythm onto the music. In Video mode, it freezes the motion of any image currently being played. With the switch pushed upwards, the unit functions in the Pad Motion mode and lets you record up to six seconds of your finger movement or tapping on the X-Y pad. Once captured you can play the captured motion back by holding the switch up. This is designed very much as a temporary performance 'macro' and it immediately disappears when you touch the pad again. This is great live, because it means you can grab bits of sound and video manipulation on-the-fly, build up 'phrases' and repeat them over the top of the music or video if you wish.

The KPE1's controls and main X-Y pad are sensibly backlit for ease of use in dark surroundings — and it looks cool, too!

The final performance feature to look at is the Tap/BPM function. Tempo values (in beats per minute) may be entered in one of three ways. You can dial in a specific tempo value using the Program/BPM knob, or switch into the auto-detect mode, where the KPE1 tries to lock onto a strong beat from the incoming music signal. Finally, you can tap along to the beat in manual mode and eventually the unit will sync up for you. There is actually a fourth way to sync, which is to use an external MIDI Clock signal (from a sequencer or drum machine, say). Several of the 100 effects in the audio and video selection are specifically designed for use with the BPM function, allowing the tempo to drive the visual effects and produce all sorts of dazzling colour shows, pulsating waveforms, spectrum analysis, lissajous figures and many more effects, all sync'ed to the beat.



KF4 Specification Inputs: Four (Input 1, on the rear of the KF4, accepts S-Video via a mini-DIN connector, or composite video via a phono connector. Inputs 2, 3, and 4 accept only composite video via phono, and input 4 is on the front of the unit). Main Output: One (on the rear of the KF4, in S-Video format via mini-DIN connector or composite video on a phono jack). Monitor Output: One (on the front edge of the KF4, on phono jack only as a composite video signal). Switches: Fader Curve (four crossfade types), Mode (offering instant black or white backgrounds, and Chroma-key and Luminance-key image-superimposition effects), plus a Fine control knob for adjusting the 'Keying' level. Video sampling: Eight-bit, 13.5MHz. Video format: Switchable between NTSC and PAL. Power supply: 7V DC (via external AC adaptor).

Video Sampling The sample modes let you sample visual material in a number of ways. As with the 'audio-only' KP2 Kaoss Pad, there are two Sample memories accessed via the two buttons on the front of the unit at the bottom edge. You can sample a still image into memory by feeding an image into the unit and then pushing the Mute/ Freeze toggle level and hitting the Sample memory button number 1 or 2; it's all very fast. Once sampled, you can treat the image by selecting an effect and then mess about with the X-Y pad to create new imagery. As mentioned earlier, you can use the same Sample memories to sample up to six seconds of motion video in both memories, and re-effect and manipulate them with the X-Y pad if desired. If you sample two different 'micro-movies' into these sample memory locations, you can use them to build interesting visual rhythms in an interactive way, and you can sample both audio and video, which is great for combined DJ and VJ work. Whether you use the Sample memories for stills or motion video, they are very 'temporary' and meant to be used as part of a live performance. Once you switch the unit off, the samples are lost.



Korg Krossfour KF4 Vision Mixer The Korg Krossfour is a very useful stand-alone four-channel vision mixer. Its solid aluminium construction, colour and size indicate that it's been designed to complement the Kaoss Pad Entrancer, but it is equally usable in its own right. Four video inputs are available for use with live cameras, VHS, DVD or computer sources. The idea is that you allocate each of the four sources to fader positions labelled A and B. For example, you could have a DVD player connected to Input 1 and routed to position A, and a live video camera plugged into Input 2 and routed to position B. By moving the KF4's main fader between A and B you can visually crossfade from one source to another, or position the Fader mid-way and blend the two images together. There are four different fader curves associated with the transition from fader position A to B (selected via a switch on the front of the Krossfour), and these range from the normal smooth crossfade to faster curves which result in a more instant fade from one image to another.

The front edge of the KF4. Different fader curve types are selectable via the switch on the left, and the important Mode switch allows you to experiment with Chroma-key type image-superimposition effects, or instantly switch the main output to all black or all white. The fourth video input and the output for the optional video input monitor are also located here.

The rear panel of the Krossfour houses the power and video-format selector switches, the inlet from the external power supply and the main video outputs and inputs — or most of them. As you can see from the picture on the left, video input 4 is on the front of the unit, presumably due to lack of space at the back. This was the right choice, however; the connections are better spaced as a result, and allow access more readily.

It's important to mention that all of the Input selection buttons have two functions. Pressing a button once selects the input to be assigned to A or B, but pressing the button twice causes that input to ignore the source and set itself to one of four Modes, either Black, White, Chroma-key or Luminance-key, as determined by a switch on the front edge of the unit.

The KF4 Krossfour video mixer. Any of the four possible video inputs can be assigned to fader

The Mode function is very useful in a live performance context, as you can instantly 'kill' a video image by making the screen black or white. Alternatively, you can assign any Input to the fader position which is not currently live and fade across to it (for example when fading to black at the end of a performance).



positions A and B, and then the fader is used to blend between the currently selected inputs. Here input 1 is selected for Fader position A, and input 2 for B. The person mixing can check what's on any of the four video inputs at any time without affecting the main output by attaching an optional video monitor and using the Monitor Select buttons at the top of the unit.

The Chroma and Luminance Key features are great for superimposing one image onto another, allowing you to mix one image into either the chroma or luminance part of the video signal of another image. There's even a small adjustment knob to allow you to finetune the extent of the superimposition. It's not really going to give you broadcast-quality back-projection type superimposition, but it's still very usable for effecting the overall image, and a worthy inclusion.

The last button to mention around the Input select buttons is marked Hold. As the name suggests, pressing this button will freeze the current image. Finally, a set of buttons along the top of the unit allows you to select any of the four input sources (or the currently live output) for viewing on an optional external monitor. This feature is very useful for cueing up a source before you bring it into the vision mix. Thankfully, all of the buttons and the A-B fader are backlit, which is exactly what you want when you're fumbling around in a live venue in the dark! My only reservation about the KF4 is the same one I have for the KPE1 — the power cable and connector are so flimsy for a unit otherwise so well designed for live use. In a nutshell, the Krossfour is a simple little unit that does its job perfectly well.

Connection Points The overall construction of the KPE1 is very solid and sports connectors on both the rear and front edges (sensibly, the included headphones socket and associated volume control are located at the front). As the unit is aimed at the serious hobbyist/semi-pro market, all of the connectors on the box are of a 'consumer' type ie. phono, DIN, and mini-DIN sockets rather than broadcaststandard BNC or XLR connectors. This is fair enough, though, as the chances are that you'll be plugging in the outputs from a VHS, DVD or a video camera as well as standard stereo audio. Video Input 1 provides a video and audio input whereas Input 2 only provides video. Only one of these inputs can be in use at any point in time, so a switch on the front edge of the KPE1 allows you to select between the two. If you want to choose between more than two sources, this is where the separate Krossfour vision mixer would be useful; there's more on this in the box on using the KPE1 live below. Finally, if I have a moan about any of the connectors on this unit it must be the power adaptor socket. In this day and age, we all seem to have to live with the 'wall-wart' adaptor; an understandable but often hated short cut which allows manufacturers to make the same basic unit for use worldwide and then make separate external power supplies with the correct voltage for different territories. However, in a live context, where a unit such as the KPE1 is being handled all the time, and is certainly subject to lots of vibration, the simple push-in power adaptor plug is a poor choice. Every night I used the KPE1 live, I had to gaffertape down the power cable to keep it in place. Whilst I realise it would be file:///H|/SOS%2005-06/Korg%20KPE1%20Kaoss%20Pad%20Entrancer.htm (8 of 11)9/28/2005 2:33:40 PM


impractical to manufacture the unit with a fixed power lead, the use of a better 'screw-on'-type chassis socket and line plug would have put me at ease.

Right On Cue You can think of the KPE1 as a kind of 'video synthesizer'. So, whether you're looking to get into the VJ club scene, add a dramatic visual backdrop to your live show or produce interesting video manipulation in a performance art environment, the Kaoss Pad Entrancer is a super little unit that's well built, simple to operate and is bursting with audio and visual possibilities. When you add in the Krossfour vision mixer, it makes for an excellent compact setup that'll serve you well in any live context.

Using The KPE1 & KF4 Live Back in October last year, Bill Nelson invited me to get involved in producing the visuals for his 30th Anniversary Be-Bop Deluxe & Beyond Tour, sponsored by Sound On Sound. Given the limited budget, there was no chance of getting a massive video production together, so I looked around for equipment that would contribute a lot to the visual experience without necessitating the use of broadcast-type gear. The Korg Kaoss Pad Entrancer and its sister unit, the Korg Krossfour mixer, seemed to be the perfect solution. With the addition of a couple of budget video cameras and some visual content fed from a domestic DVD player, this compact setup provided everything I needed to fill a two-hour live show every night, while keeping the visuals fresh for anyone who attended more than one gig. For the tour, I had the challenge of having to work in different venues around the UK, from small clubs to large theatres. The space available beside the mixing desk was always restricted, so the compact size of both these Korg units really paid off. Before the tour started, I created some motion video content for each song in Adobe Premiere and burnt this to DVD. On tour the DVD was played directly into video input 1 on the Entrancer, where I manipulated it in real time. In a simple setup, you could get away with just these two pieces of kit. However, as this was a two-hour concert and Bill Nelson would talk


Photo: Beck Studios, Wellingborough Bill Nelson live on tour in Autumn 2004, with the KPE1/ Krossfour combination providing video effects on the screen behind him.


The author on the Bill Nelson tour, with the video mixing setup next to the main front-ofhouse mixing desk.

between songs, I needed to fade out the video being projected on the stage screen between each song. As there is no way of doing that easily on the Entrancer, I needed a simple vision mixer, and this is where the Korg Krossfour came to the rescue.

A close-up of the live video mixing rig. The monitor at the top was for checking the four inputs to the Krossfour video mixer without affecting the main output to the on-stage screen. Below that is the DVD player used for the pre-recorded video material during the show, and at the bottom are the white control unit for the two cameras used on stage, plus the Entrancer and Krossfour. The Korg units both had to have home-made 'labelling' added due to the lack of built-in scribble strips!

The Krossfour also worked perfectly with the two live video cameras. The setup was simple. The Entrancer was plugged into Input 1, and the two cameras into Inputs 2 and 3. Input 4 was unused and set to Black, although you can make any channel on the Krossfour black at any time, regardless of the device attached to it, as explained on the previous page. Typically, I'd start off a concert with Input 4 assigned to fader position B, and the fader fully set to B, outputting a blank screen. Then, once the DVD started playing, I'd fade across to position A, which would have the DVD output assigned to it, thus bringing the movie on the DVD onto the main stage screen. The Krossfour's Monitor output was plugged into an LCD TV, so I could preview images before I committed myself to fading them up on the large live screen. This was particularly useful for checking the feeds from the live cameras for the numerous guitar and sax solos, allowing me to make sure that they were correctly zoomed in and focused before fading the picture up into the overall mix. When you're running a live show like this, reliability is absolutely paramount. Personally, I would not have trusted a computer-based software solution in this situation, and a system crash part-way through a live concert was not a risk I was prepared to take! Both the Entrancer and the Krossfour are solidly built units in metal boxes with rugged, well-sized buttons and switches, and control knobs designed for heavy use. There's nothing fragile about this kit, which is exactly what you need on the road. Published in SOS June 2005






Line 6 Variax Bass 700

In this article:

This Year's Model Instrument Table Setting The Controls Vicious Or Jamerson? Desert Island Basses?

Line 6 Variax Bass 700 Modelling Bass Guitar Published in SOS June 2005 Print article : Close window

Reviews : Modelling Guitar

Line 6 Variax Bass 700 £949 pros Great sounds. Simple and easily accessible controls. Well finished and set up.

cons Conservative design. Not the most comfortable bass to play. Short battery life.

summary Brilliant effort to model classic basses, although maybe not quite the selection of basses I would have chosen. Such a shame the look and feel is ordinary at best.

information £949 including VAT. Line 6 Europe +44(0) 1788 821600. +44(0)1788 821601. Click here to email www.line6.com

Expanding on the modelling technology used in the Variax and Variax Acoustic, Line 6 have dropped an octave and turned their attention to the bass guitar. The Variax Bass features 24 models of 'classic' instruments: is it all the bass you'll ever need? Phil Ward

How many bass players does it take to change a light bulb? Nobody knows. Nobody's ever dared trust us with something so high-tech, you see. Actually, I really don't buy the 'bass player as semi-educated Neanderthal' cliché, and not just because I am one (a bass player that is, not a Neanderthal). If willingness to embrace Photos: Mike Cameron new ideas is any guide, we bass players seem to be less conservative than our guitar colleagues an octave up (and yes, I can already imagine the flood of indignant SOS Forum posts). For example, it was no accident, when Ned Steinberger redesigned the guitar in the early '80s with bridge-end tuning and a minimalist one-piece carbon-fibre body and neck, that he launched his bass first. And you only have to spend a few mandays browsing the web to find that the population of weird and wonderful bass designs out there exceeds the number of similarly off-the-wall guitars by quite some margin. Not that I've ever done such a thing of course — I do have a life...

This Year's Model So at last bass players get to have a go with the Line 6 instrument modelling technology that electric and acoustic guitarists have had for a while. And judging

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by the posts I've read on various bass guitar discussion groups, bass players around the world are generally intrigued by the prospect. With the Variax Bass, Line 6 have produced an instrument that can, they claim, take on the sonic character of 24 classic models from the bass guitar hall-of-fame. It's a tall order. The characteristics of say, a Rickenbacker 4001 and a Kay M1 Upright, or a Hofner 500/1 and a Warwick Thumb, four of the basses modelled, are poles apart — they might as well be entirely different species of instrument. But before I cut to the chase and write about the sound of the bass there's some description of the instrument and analysis of its context to be done. The first thing that you can't miss about the Variax is that, despite its leading-edge electronic technology, its look is one of unadulterated tradition. The shape, body and headstock, are mild variations on a standard Fender-esque theme that could have come from any number of Far Eastern 'generic bass' manufacturers (the Variax is actually made in Korea). Perhaps the only element that stands out, or rather doesn't, is that the Variax has no obvious pickup. The job of turning the strings' vibration into electrical signals is done instead by piezo-electric transducers under the saddles of the bridge.

The included power supply/DI box 'phantom' powers the bass via a TRS jack lead.

From my point of view the conservatism of the Variax aesthetic and ergonomic design is a disappointment. For such a ground-breaking instrument it seems a shame not to have developed a similarly modern piece of design. And before you say it can't be done, that bass aesthetics are the result of decades of ergonomic refinement, I must cite Musicman with the Bongo, Cort with the Curbow and Parker with the Fly Bass — three examples among many I could have selected that have all shown that there are still viable and attractive new shapes for bass guitars out there if you just apply some imagination and industrial design to the problem. It's difficult to understand why Line 6 went for such a conservative design, it's not as if the company has a backward-looking philosophy in any other respect — far from it. Maybe the decision was marketing driven, or maybe it was cost driven, but in either case I fear they got it wrong. If the marketing bods decided, well my feeling is that the majority of bass players would be more attracted to something more forward looking; and if the bean counters decided, I think that a couple of hundred quid more for something that looks a bit special, rather than a bit ordinary, would have been worth the extra money, especially as the bass already retails in the UK at just under £1000 anyway. The Variax is also, thanks to its traditional 'slab' body, not a particularly comfortable bass to wear on a strap. It's a relatively heavy instrument, and



neither the elbow or waist profiling on the body are generous — in fact there is no waist profiling at all. But while we're on the subject of profiles, the neck feels to me to be somewhere between a Precision and Jazz Bass in basic proportions — a good comfortable compromise for most players. Build quality on the Variax is good, but as with the aesthetic, a choice of black or 'sunburst' doesn't break any new ground as far as finish is concerned and I'm afraid the bass doesn't, to my eyes, look as jaw-droppingly gorgeous as a £1000 instrument should. Set-up was good though; buzz free with intonation just about spot-on and strings nice and low. The fretwork is neat, with no sharp edges, lumps or bumps apparent.

Instrument Table Pair Name Green Bank

Red Bank

VINJ

1961 Fender Jazz

1960 Fender Jazz (flat-wound strings)

MODJ

2004 Fender Deluxe Jazz 1961 Fender Jazz Fretless

PREBASS 1963 Precision Bass

1958 Precision Bass (flat-wound strings)

MANTA

1977 Musicman Stingray

2003 Modulus Flea

CLANG

1971 Rickenbacker 4001

1963 Rickenbacker 4001 (flat-wound strings)

HOLLOW

1966 Danelectro Longhorn 1963 Hofner 500/1 (flat-wound strings)

THUMP

1963 Gibson Thunderbird 1966 Gibson EB2D (flat-wound strings)

MODERN

2002 MTD 535

2003 Warwick Thumb

ALCHEMY 1978 Alembic Long Scale 1984 Steinberger XL2 8 & 12

1968 Hagström H8

1994 Hamer B12A

ACOUSTIC Tacoma Thunderchief

Kay M1 Double Bass

SYNTH

Modern Bass Synth

Mini Moog

Setting The Controls Once the bass is strapped on and powered up, you're faced with four control knobs. From the top down there's volume, pickup blend, stacked bass/treble, and lastly the all-important instrument selection knob. The volume and tone are selfexplanatory but pickup blend and instrument selection have a little explaining to do. Let's deal with instrument selection first. The 24 Variax instrument models are held in two banks of 12 — the green bank and red bank. Each of the twelve pairs of instruments has a generic name and it's this name that's etched onto the side of the detented selection knob. Instruments are paired in loosely related couples. For example, the two vintage Jazz Basses are paired, as are the two Precisions. The adjacent table lists the instruments stored in each bank. The currently selected instrument pair is illuminated on the selection knob by an file:///H|/SOS%2005-06/Line%A06%20Variax%20Bass%20700.htm (3 of 6)9/28/2005 2:33:44 PM


adjacent green LED if the green bank is selected, and by a red one if the red bank is selected. Switching between banks is simply a case of depressing the volume knob. There's one final trick incorporated in the instrument selection knob. Once a preset bass model has been tweaked to personal taste using the tone and pickup blend controls, depressing the instrument selector saves the adjusted settings as a preset that is recalled as the default when that instrument is next selected. A useful and simple refinement.

The volume, pickup blend, stacked bass/ treble and instrument selection knobs provide a simple but effective interface for the Variax Bass.

In use, the Variax is pretty straightforward. Only a day or two after I'd first had sight of it, I was able to try it out on a few songs at a short gig without any trouble. (The sound guy's face was a picture as I dialled through a few different basses during the soundcheck!) The first thing to appreciate, however, is that the electronics within the Variax need a healthy supply of volts. Two options are available. Option one is batteries — either six AA or a single 9V battery within the bass provide an estimated 10 to 12 or one to two hours operation respectively. The 9V option is clearly for emergencies only. Option two is the included power supply/DI box that 'phantom' powers the bass via a TRS jack lead. Of course this box itself requires power, and that's supplied by a simple mains adaptor. What a shame the power supply/ DI box couldn't itself have been phantom powered from a 48V supply — presumably its current demand is too high. If you're at all familiar with any or all of the basses listed in the table you're probably doing a Mr Spock-style quizzical glance on reading the term 'pickup blend'. You see, quite a few of the basses modelled either have only one pick-up or, in the case of the two acoustic instruments, none at all. But it's quite logical, Jim, and works like this. On basses that do have multiple pickups, the blend control does exactly what it says and models adjustment of the blend between them. On single-pickup basses however, the blend control very cleverly models what would happen if the position of the pickup were moved either towards the neck or bridge. So if you've ever wondered what a Precision Bass would have sounded like if Leo Fender had stuck the pickup hard up against the bridge, the blend control is just the job. On acoustic instruments (the Kay and the Tacoma), the blend control models microphone distance by apparently adding some proximity affect and increasing the level as the microphone moves closer.



A second Mr Spock moment might be provoked by the two 'bass synths' modelled. These selections model a classic Mini Moog and an un-named contemporary bass synth — both with a layered two-voice patch and both with a filter sweep. The bass, treble and blend controls on the bass adjust filter speed, filter depth and voice balance respectively. I'll go on to describe a few of the conventional bass sounds in the next paragraph, so I'll start off with the synths — and get the bad news out of the way. The synth sounds really didn't work for me. Partly, I can't help thinking these sorts of sounds are best played on a keyboard anyway, and partly they seemed to me to have little depth and power — just not warm and rubbery enough, if you know what I mean. That's not to say they'll never find a niche, just that it'll be, to my ears, a small one.

Vicious Or Jamerson? So, finally, what does it sound like? Does it work? Well from one point of view, it's hard to say, as I own only one of the actual basses modelled (that original Steinberger) so could only do one direct comparison — a comparison that the Variax passed with quite extraordinary flying colours. It does a very good impression of the Steinberger, and not just in tonal character — the note shape (attack and decay) sounds and somehow 'feels' remarkably similar. From another point of view, without a near priceless collection of classic basses for direct comparison, which will be the position of the vast majority of players, I was left just to consider the sound of the Variax in isolation; and it sounds great, fabulous even. The ability very quickly and easily to try very different instrument voices within the context of a piece of music brings the kind of liberation to playing bass that keyboard players have taken for granted for years. It's a very seductive thing, and I found myself immediately hankering after real-life versions of the modelled basses. I especially loved the Precision, the Musicman, the Danelectro and the Tacoma, but all the basses (other than perhaps eight- and 12-strings) would no doubt get a regular run-out. The only issue that nags away at the back of my mind is wondering how the Variax performs once its strings have lost their brightness. I guess though that buying new strings more often than you're used to is very much less expensive and very much more convenient than a collection of classic basses. If you can accept, or even like, the conventional design of the bass, then it may well be 'The Only Bass You'll Ever Need'. However, I also have at the back of my mind a crazy scheme to wait for the five-string Variax, rip out the electronics and have them built them into a crazy, custom-designed, halffretted chimera — see, I said bass players are more radical than guitarists!



Desert Island Basses? Selecting the basses to model for the Variax must have been like choosing Desert Island Discs or an ultimate 'best of' collection — difficult and guaranteed to attract differing and passionately held opinions. So here, for what it's worth, are a few basses I'd have liked to see. Firstly, I don't have any problem with the first eighteen models and the two acoustics. Having said that I'm not sure I'd ever need four versions of the Fender Jazz — especially as the fretless doesn't work to my ears on a fretted bass (and I don't think it ever can, as the characteristic fretless sound comes as much from left hand technique as it does from the model of instrument). But I'm really not convinced about the value of the eight- and 12-string models or the bass synths. Replace those with a mid-80s Wal Custom, An Aria SB1000 from the same period, an original US Steinberger Electric Upright and the bass strings of a Chapman Stick and I'd be far more tempted by the Variax (once I'd de-fretted it, that is). If I'm lucky, Line 6 might read this and perhaps add those instruments later, as there's a presently unused RJ45 socket on the side of the bass that the manual says will 'connect to future Line 6 products with a Variax input adding additional sounds and abilities to your Variax Bass in the future'. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Magix Samplitude Professional v8

In this article:

What You Need Versions Paper Power VIP MIDI, Mixing & Effects Video Analogue Modelling Disc Authoring I, Robota Surround Sound Elastic Audio Remix Agent Summing Up

Magix Samplitude Professional 8 £695

Magix Samplitude Professional v8 Digital Audio Workstation [Win] Published in SOS June 2005 Print article : Close window

Reviews : Software

The tools available in Magix's highly regarded recording package cover every stage of the recording process, from MIDI sequencing to mastering and CD burning. Version 8 adds some neat extras including analogue-style processors, a virtual drum machine and an Acid-style beat-mapping tool.

pros Powerful, easy-to-use Paul Sellars multitrack audio and MIDI recording and editing package. Analogue Modelling Suite SOS last looked at Magix's Samplitude back in June 2003, when version 7 was and Robota Pro are fine released: you can read Mark Wherry's review at www.soundonsound.com/sos/ additions. jun03/articles/samplitude7.asp. Development has continued steadily in the Elastic Audio is a powerful intervening years, and the application now stands at version 8.0, offering a range tool. of minor refinements and some substantial new features. Remix Agent works well. Very nice convolution reverb.

For the benefit of the uninitiated, Samplitude is a host-based 'native' On-the-fly CD burning may Digital Audio Workstation (DAW) for be unreliable on all but the fastest machines. Windows, which combines powerful Bundled VST plug-ins multitrack audio recording and editing restricted to use within with MIDI sequencing and complete CD Samplitude only. mastering facilities. Its closest summary competitors are probably Steinberg's To sum up, Samplitude Nuendo and Digidesign's Pro Tools, Professional 8 is an although it arguably has a better claim impressive and very capable to providing a viable 'all-in-one' package, which justifies its package than either of these. not insubstantial asking price with a comprehensive suite of Samplitude is aimed squarely at the professional market, and boasts a suitably high-quality tools enabling the comprehensive feature set. It offers a total of 999 tracks, supports the Direct X user to perform just about and VST plug-in formats, with full delay compensation, and can work with any every conceivable pro audio hardware that uses the ASIO, WDM or MME driver standards. The ability to act task with ease. It's a product as a Rewire host has been added in version 8, offering new options for deserving of serious integrating soft synths and programs like Ableton Live. Samplitude also includes consideration by anyone in cons

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the market for a DAW.

information Samplitude Professional £695; Classic £350; Master £210. Prices include VAT. DACS +44 (0)191 438 2500. +44 (0)191 438 2511. Click here to email www.dacs-audio.com

Test Spec Samplitude 8.1. 1.8GHz Athlon PC with 512MB RAM and 2x 7200rpm Maxtor hard drives, Emagic Audiowerk 2 soundcard and VIA AC97 onboard sound, running Windows XP Service Pack 2.

high quality POWR dithering, a real-time convolution reverb processor, 5.1 surround mixing and support for a range of hardware control surfaces.

What You Need Samplitude will install and run on fairly modest hardware; a Pentium II with 128MB RAM is all that's required. For basic multitrack audio production, Magix recommend a 400MHz Pentium II or faster, and a 7200rpm hard disk. In reality, of course, the more processing power, memory and storage you can provide for any DAW, the better it will perform. Magix are at pains to point out that Samplitude supports older Windows versions, such as 98, NT and 2000, as well as XP. This is commendable, but even so, I'd hesitate to recommend any Windows version earlier than 2000 for serious production work. Samplitude 8 ships on a single CD, and is copy protected with a WIBU Systems 'Code Meter' USB dongle. Dongles tend to be unpopular, and not without good reason. They're an inconvenience, they take up ports that could otherwise have useful devices attached to them, and there's always the fear that a dongle will malfunction, leaving you without access to software you've already paid for while a replacement is sought. On the other hand, many modern PCs have half a dozen USB ports built-in, and USB hubs aren't too expensive if more are required. For what it's worth, Samplitude's dongle worked faultlessly for me during my few weeks of testing, and I managed not to lose or break it. Once the software and dongle have been installed, you need to register the dongle on-line with Magix within 90 days of first running the application. The arrangements are slightly different if you opt for one Magix's 'rental' schemes — see www.samplitude.com/de/sfr.htm for more details.

Versions Samplitude is available in three different versions: Professional (reviewed here), Classic and Master. Both Classic and Master are, in effect, limited versions of Professional, intended to provide cost-effective alternatives for users who are willing to do without some of the extra whistles and bells. See www.samplitude. com/de/versions8.htm for more details.

Paper Power Manuals are important. Whether you're a reviewer trying to get up to speed quickly, or a real person trying to understand the newest addition to your studio, a good manual is an invaluable resource. The Samplitude manual has been reworked for version 8, and it's very good: well written, nicely laid out, printed and bound into a proper handbook. file:///H|/SOS%2005-06/Magix%20Samplitude%20Professional%20v8.htm (2 of 10)9/28/2005 2:33:49 PM


It opens with a Beginner's Quick Start Guide which explains, step by step and in plain English, exactly how to accomplish a series of useful real-world tasks, beginning with multitrack recording, then mixing and using effects, before mixing down to a stereo file, and finally editing and mastering. This runs to perhaps a dozen pages in total — but those dozen pages really make a difference when you're first learning your way around. The remainder of the manual provides a much more detailed reference, and does a good job of explaining the finer points of the software in depth.

VIP Samplitude saves multitrack projects in its own '.VIP' file format, and the VIP window is where bulk of your work will be done. As you'd expect, a VIP project doesn't actually contain any audio, but instead links to whatever audio files are required by a project. A VIP project can contain one or more tracks, and a track can be either mono or stereo. Tracks are typically streamed to and from hard disk, but can optionally be recorded to and played back directly from RAM (to avoid the risk of interrupted playback, for example, when working with short, repeating loops). Each track in a VIP project will contain one or more Objects. An Object may point to an audio file, or part of an audio file. Each Object features five 'handles', which can be dragged with the mouse. The bottom left and bottom right handles are used to adjust the Numerous editing and processing tools can start point and length of an Object be applied on a per-Object basis with the respectively. The top left and top right Object Editor. handles can be used to set a fade-in and fade-out for an Object. By default, fades are linear, but right-clicking and accessing the Crossfade Editor allows you to set and fine-tune a variety of curved fades. The centre-top handle provides a quick and easy way to set the relative volume for an Object. Double-clicking an Object's name opens the Object Editor dialogue where a whole range of settings can be made, on a per-Object basis, including gain and aux send settings, plug-ins (up to six can be chained), four-band EQ, panning, volume, Object appearance, position, start and end times, and crossfade settings. Time-stretch and pitch-shifting effects can also be set, non-destructively and on a per-Object basis, with a choice of algorithms (Resample, Standard, Smooth, Beat Marker Slicing, Beat Marker Stretching and Monophonic Voice). Pitch-shifts can be specified in semitones or with a pitch factor, while stretches can be performed by supplying an absolute target length or a stretch factor. Stretches can also be performed by supplying current and target tempos in beats per minute. file:///H|/SOS%2005-06/Magix%20Samplitude%20Professional%20v8.htm (3 of 10)9/28/2005 2:33:49 PM


A 'Wave Editing' window can also be opened, where destructive edits can be applied to the underlying audio files, and various built-in effects processes can be destructively applied from a context menu accessed by right-clicking Objects. Right-clicking anywhere in the VIP window opens an extensive context menu from which almost all of Samplitude's editing functions can be accessed. These can also, of course, be found in the menu bar at the top of the main window, and via numerous buttons arranged along the top and bottom of the workspace. The VIP window is entirely customisable, and can be configured pretty much however you like. The Workspace pop-up menu allows you to choose from a variety of preset configurations specialised to suit certain tasks (such as Recording, Editing and CD Mastering), and you can create custom workspaces to suit your own preferred methods.

MIDI, Mixing & Effects Samplitude treats MIDI in much the same way it does audio. MIDI data can be recorded or imported into Objects in a track, and a track can contain a mixture of both audio and MIDI Objects. MIDI Objects have the same five 'handles' as audio objects which can be used to adjust start time, length, volume and fades. MIDI Objects also have their own Object Editor dialogue, accessible from the right-click context menu, and double-clicking a MIDI Object opens the MIDI Editor window, which provides the familiar piano-roll interface, with a strip for below for editing velocity Like most DAWs, Samplitude has a data and other controller messages. There's mixing window that mimics the layout also the option, new in Samplitude 8, to of a hardware console. choose a 'drum editor' view, and an event list editor can also be shown or hidden alongside the main window. Samplitude's other main window is the Mixer, which provides the standard graphical representation of a mixing desk. Each track in a VIP project has its own channel strip in the mixer, and each channel strip features a level fader, mute and solo buttons, a pan knob, four-band EQ, a Delay parameter allowing you to offset the playback of each track, six plug-in insert slots, and two aux sends, which can be set either pre- or post-fader. In addition to comprehensive third-party plug-in support, Samplitude provides a set of very respectable built-in effects, including a de-hisser, an FFT filter, which allows curves to be drawn freehand, multi-band and single-band dynamics file:///H|/SOS%2005-06/Magix%20Samplitude%20Professional%20v8.htm (4 of 10)9/28/2005 2:33:49 PM


processors, a multi-band enhancer, a very nice vocoder, distortion and the Room Simulator, an impressive real-time convolution reverb. Since I have plenty of new features still to cover, I'll simply echo Mark Wherry's comments in his review of Samplitude 7, and agree that these are among the best effects I've seen bundled with any application.

Video Samplitude allows video files to be loaded and sync'ed with projects via the builtin Link Media function. This is a useful feature, although somewhat basic. However, Samplitude Professional 8 ships with a bundled copy of Magix's Movie Edit Pro 10, a companion application offering a range of video-editing features, including VCD and DVD authoring. If you're planning to do a lot of work with video, this may provide an extra incentive to choose Samplitude Professional over the less expensive Classic or Master versions.

Analogue Modelling A new addition in Samplitude 8 is Magix's Analogue Modelling Suite. This is a pair of high-quality effects plug-ins that appear to be in the VST format, but can only be used from within Samplitude; attempting to access them in other hosts brings up a polite dialogue explaining that it's not going to happen. The two plugins are Amtrack and Ampulse; the former a flexible compressor and tapesimulation effect, the latter a 'transient modeller' similar in concept to SPL's Transient Designer, or Digitalfishphones' Dominion plug-in. While elsewhere Magix cite 'sound neutrality' as a selling point for Samplitude, these plug-ins are intended to have a bit of noticeable 'character', and to colour the sound in a manner reminiscent of various classic hardware devices. The Amtrack compressor and tape simulator is intended for processing individual tracks and small subgroups, rather than complete mixes. It operates in two different modes. In VCA mode, it mimics a modern VCA-based compressor; in Vintage mode it aims to emulate an older, FET-based device. Slightly different controls are available in each mode. In VCA mode, there are Ratio, Attack, Threshold, Knee and Release knobs on the compressor side, along with EQ and Bias controls, and level and output knobs on the tape-simulation side. In Vintage mode, the compressor section is simplified, featuring only Drive, Attack and Release controls. Hidden behind an 'expert' panel are still more controls, including a lookahead parameter calibrated in milliseconds, a filter which can be used to roll off low end in the side-chain signal to reduce pumping, 'adaptive release', which adjusts the release time semi-automatically in response to changes in the input signal, auto make-up gain, a wet/dry control, and 'tape low freq' and 'tape high freq' controls for fine-tuning the tape-simulation effect.



All these controls make for a plenty of flexibility, and Amtrack is capable of producing a wide range of compression effects. Twenty supplied presets provide useful starting points, which can be quickly tweaked as required. Despite my best efforts, I couldn't really get a bad sound out of it, and I can't think of many circumstances in which Amtrack wouldn't get the job done. I managed to get some quite nice effects with it plugged into the master buss, even that's not really what it's designed for. Analogue purists are likely to be sceptical about any software emulation of their favourite box of valves, but to my ears Amtrack is capable of some quite convincing 'old school' sounds. For the most part, its coloration of the sound is gentle and quite subtle, and the tape simulation doesn't ram itself down your throat as an 'effect'. While extreme settings do begin to drift into distortion, it's generally the right kind of distortion: edgy without being fuzzy, and quite musical. The Ampulse transient-modelling plugin is the less complicated of the two, and quite easy to use. The front panel is divided into two main sections, marked Attack and Sustain. Each of these features a linked stereo pair of level sliders, which can be unlinked and adjusted separately, and a Length slider. The level sliders are used to control the amount of amplification or gain reduction applied to the signal, while Length controls the amount of time each phase of a transient (ie. the attack or sustain) is 'held' for. This is much easier to understand in practice than it is to describe. Basically, Ampulse can be used to 'reshape' active, dynamic sounds such as drums, percussion or staccato guitar parts. If, for example, a drum track features excessive room ambience, you can use Ampulse to 'tune it out', creating a dryer, punchier, more 'up front' sound. Conversely, it's possible to soften the attack characteristics of a sound while emphasising the 'air', in order to push a track into the background a bit. In keeping with the analogue modelling theme, Ampulse also features a Saturation knob, which can be used to introduce a degree of tube-flavoured overdrive into the proceedings, and a pair of 'HF Details' controls labelled Tune and Level. These allow you to boost or emphasise higher frequencies (1kHz10kHz) in the sound, as a kind of enhancer effect. Ampulse can be used quite subtly, but can also produce some fairly dirty sounds, which I really enjoyed. Drum tracks in particular can be squashed and mangled into various interesting new shapes, with the HF Details controls adding some pleasing grit. file:///H|/SOS%2005-06/Magix%20Samplitude%20Professional%20v8.htm (6 of 10)9/28/2005 2:33:49 PM


Disc Authoring Samplitude provides full Red Book CD-authoring facilities without requiring any external software, and even allows CDs to burned 'on the fly', with all effects and mix automation calculated in real time. This has the advantage of not requiring you to devote extra disk space to rendering a stereo mixdown file. However, it does require a fast machine to work reliably. I burnt half a dozen CDs in the course of testing Samplitude, and while most were fine, one did suffer from some hiccoughs and glitches, requiring me to bounce down to a stereo file and burn in the conventional way. It's now also possible to burn DVD-Audio discs. These are so-called 'black discs' with no graphical menu, but basic track markers. Both 16- and 24-bit audio can be written to DVDs, at sample rates of 44.1 and 48 kHz. Up to six channels are supported for Dolby 5.1 surround, and several other multi-channel configurations (4.0, stereo) are also supported.

I, Robota Another VST plug-in supplied with Samplitude Professional 8 is Robota Pro, an impressive analogue drum machine emulation. Like the Analogue Modelling Suite, it's restricted to running within Samplitude, and cannot be accessed by other VST host applications. Robota Pro is eight-voice polyphonic, with a built-in eight-track, 64-step sequencer. Each of its eight 'instruments' is generated by an identical oscillator section, which can produce sine, triangle and sawtooth tones, and can load WAV-format samples. Each oscillator also features a noise generator, which can either be blended with the oscillator output, or used as a pitch modulation source. The Noise knob controls the frequency of the noise generator, with low settings producing 'sample and hold'-style effects. For each oscillator there's also a simple amp envelope with Attack and Decay controls, which can be used to modulate the oscillator pitch. This simple-but-effective arrangement of modules allows Robota Pro to The Robota Pro drum machine is new in version 8 of Samplitude. generate a wide variety of 'analogue' blips, pings and similar noises, along with some quite convincing TR808 and TR909 drum-machine emulations. Further sound-generation possibilities are provided, for each oscillator, by frequency and ring-modulation effects, a digital 'lo fi' effect, a multi-mode resonant filter offering low-pass, band-pass, high-pass and comb options with its file:///H|/SOS%2005-06/Magix%20Samplitude%20Professional%20v8.htm (7 of 10)9/28/2005 2:33:49 PM


own modulation envelope, and an output stage featuring a simple compressor and a 'tube' overdrive effect. Robota Pro presents itself as a kind of software 'groovebox', and as such it's impressive. It can be controlled either by its own internal step sequencer (the preferred method, to my mind), or triggered via MIDI. It offers various groove templates, affecting both phrasing and velocity. Its sounds are fat, solid and altogether convincing. It's very much a specialised drum and percussion synth, which doesn't try to do a lot else, but it works well, and is great fun to use. Loading your own custom samples into Robota Pro's oscillators isn't quite as easy as it might be: you have to prepare and edit your samples in advance, then save them in the Samples folder within Robota Pro's own plug-in folder. They will then be found next time the instrument is started. Robota Pro's effects and modulation functions allow your custom samples to be rapidly mangled beyond all recognition, and some great noisy, 'industrial' effects are possible.

Surround Sound Samplitude Professional 8 offers a range of facilities for mixing in various surround sound formats including 5.1, quad and stereo, among others, although if you want to mix for more than six channels you'll need to investigate Magix's more upmarket Sequoia instead. Surround settings can be made per mixer channel, or per Object. The Surround Panorama Module allows individual sounds to be positioned within the sound field via a graphical interface, in one of several modes. Surround panning can be recorded and automated in much the same way as ordinary stereo panning. Most of Samplitude's built-in effects, including the Room Simulator convolution reverb, can be used in Surround mode, and function more or less as they would in stereo, but supporting a maximum of 12 simultaneous channels.

Elastic Audio Another impressive new feature in Samplitude Professional 8 is what's called the Elastic Audio editor. In essence, this provides a fully automated real-time implementation of the same non-destructive pitch-shifting processes available via the Object Editor dialogue. Elastic Audio works in one of two modes. In Relative mode, complex control curves can be drawn, which dynamically affect pitchshifting throughout the duration of an Object. This is simple, but very effective. The large Elastic Audio editor window, with its freehand, quantised and curvedrawing tools, allows for extremely precise control over pitch, moment by moment. With careful choice of the right pitch-shift algorithm, it's possible to perform very natural-sounding pitch corrections quite easily. Equally, with less careful use, some mind-bending special effects can be created.



Direct mode is a bit different, and relies upon automatic pitch detection. In Direct mode, the left-hand edge of the Elastic Audio window is calibrated with a 'piano keyboard' semitone map. When the Detect button is clicked, the selected Objects are analysed and divided into a series of new Objects, which are distributed in the window according to their detected pitches.

The Elastic Audio editor offers real-time pitchshifting, with the option of Auto-Tune-style pitch detection.

A pitch-control curve linking these Objects is automatically generated, and can be edited directly with the mouse to create new pitch-shifts. It's also possible to select some or all of the slices, and automatically retune them to fit one of five different scale types (Major, Minor, Harmonic Minor, Pentatonic and Chromatic). Direct mode requires fairly simple monophonic input in order for its automatic pitch-detection to work reliably, but in my experiments the results were impressive. As with Relative mode, both subtle and extreme results are equally possible. Overall, Elastic Audio has a lot to offer, whether you want a tool for careful, surgical corrections or outlandish, creative sound design — or both.

Remix Agent Another useful new tool is the Remix Agent. This is very similar, in both concept and implementation, to the Beatmapper wizard in Sony's Acid loop-based audio sequencer. The Remix Agent can be started by right-clicking an Object and selecting it in the context menu. In order for the Remix Agent to work effectively, the source Object must be longer than 15 seconds, and must contain strongly rhythmic 'danceable' material. Once started, the Remix Agent presents a wizard interface, much like that in Acid, which allows you to guide it in detecting and marking bars and quarter-note beats in the audio. Provided the input material is sufficiently rhythmic, it usually guesses correctly, but it's easy to give it a gentle nudge in the right direction when required. Once tempo, beat and bar information has been ascertained, the next step is to apply it practically within the VIP project. This can be done either by changing the project's global tempo to

The Remix Agent provides beat-mapping features similar to those available in Sony's Acid.



match that of the source Object, or by slicing the original Object into 'Remix Objects', which can then be fine-tuned to match the desired tempo via a mixture of pitch-shifting and time-stretching. For anyone who has worked with Acid's Beatmapper this will all seem like second nature. The Remix Agent succeeds in providing a good, user-friendly solution to what could otherwise be quite a laborious and time-consuming problem. Obviously, there are limits to how natural the results of drastic tempo changes can be made to sound. I'd say, though, that the results produced by Samplitude are certainly of comparable quality to those produced by Acid.

Summing Up Overall, Samplitude is an extremely powerful and flexible application, which requires plenty of time to learn in depth. That said, one of its most impressive attributes is that Samplitude is not difficult to start working with. Its user interface, while complex, is quite logically laid out, and new users should have no difficulty performing basic recording editing tasks very quickly. The more time you spend working with the program, the better able you are to take advantage of its deeper features and its highly customisable interface. Published in SOS June 2005





In this article:

Overview Mykerinos Hardware Operation SP2 Update Top Down VCube Starting A Project Editing Basics Conclusion

Merging Technologies Pyramix pros Flexible and configurable hardware and software options. Intuitive operation and ease translating from other platforms. Open system architecture for interfacing and file exchange. Compatibility between Native and hardware-based versions. Pyramix versions available for all budget levels.

cons Native versions are limited in their functionality.

Merging Technologies Pyramix Digital Audio Workstation [Win PC] Published in SOS June 2005 Print article : Close window

Reviews : Computer Recording System

Pyramix might be the new kid on the block, as far as audio recording and editing are concerned, but that hasn't stopped it proving a very serious rival to Pro Tools and other established DAWs. Hugh Robjohns

Sophisticated multitrack audio editing using a computer-based system is part and parcel of today's post-production process, almost regardless of the end product. Whether we are talking about stereo or surround-sound music production, CD and SACD/DVD-A mastering, radio or TV post-production, film editing and dubbing, or games soundtrack production, all will use some form of Digital Audio Workstation (DAW) for the audio assembly, editing or mixing — either throughout the entire process, or to perform certain parts of it.

summary Pyramix is a very serious contender to replace the ubiquitous Pro Tools as an industry standard, extremely flexible PC-based DAW. It's configurable for every aspect of professional audio — CD and high-resolution format mastering, radio production, and film and TV dubbing — and available in two functionally limited Native versions, as well as the relatively cost-effective Virtual Studio configurable system employing bespoke hardware DSP and interface cards.

The range of DAWs available on the market is vast: some are rather simple but cost-effective, others are extremely complex and very expensive. Some employ bespoke DSP and interface hardware while others are far more generic, and many are optimised for specific types of application while some are flexible and configurable enough to be used in a very wide range of production environments. Probably the best known example of the latter is the ubiquitous Pro Tools, but there are plenty of other worthy systems in widespread use, such as the AMSNeve Audiofile SC, Fairlight Dream, Sequoia, and SADiE, to name just a few. One of the latest products to enter this highly competitive market comes from Switzerland: Merging Technologies' Pyramix. The Pyramix platform is a relative latecomer to the market, at least compared to many of its well known competitors, but that has given the manufacturer the opportunity to observe the way both the industry and the technology are

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information Pyramix Native £475.86; Pyramix Native Media Bundle £797.83; Pyramix LE from £2350; Pyramix Virtual Studio Core from £5875; VCube from £5170. Prices include VAT; contact Total Audio Solutions for details. Total Audio Solutions +44 (0)1527 880051. +44 (0)1527 880052 Click here to email www.totalaudio.co.uk www.merging.com

developing. The result is a product engineered afresh and from a clear viewpoint, rather than being the outcome of a series of evolutionary and often convoluted developments and diversions. To that end, Pyramix does some things very differently from its peers, while in others it presents the very best of all their design attributes, working practices and ergonomics.

Overview The first thing to explain is that, like most other high-end DAWs, there are several different versions of Pyramix. At the time of writing there are four basic options: Pyramix Native, Pyramix Native Media Bundle, Pyramix LE and Pyramix Virtual Studio Core, with prices ranging from under £500 for the simplest Native system through to about £12,000 for a top-of-the-range, fully loaded DSD-capable VS system. A typical well specified audio-for-video post-production Virtual Studio system would cost under £8,000 and a CD-mastering setup is about £5,500 — the exact prices depending on the specific software and hardware options installed. The versatility of the Pyramix DAW can be ably demonstrated by considering a small number of high-profile Pyramix installations in and around London, all of which I have seen in action. For example, the mastering suites at the Strongroom Studios use Pyramix systems in concert with DCS converters for CD and SACD mastering. Twickenham Film Studios have established an entirely digital signal path for the production of film soundtracks, combining high-end digital consoles with a total of 12 Pyramix systems, which are used for all the track laying and dubbing of feature films that pass through the facility. The BBC is another Pyramix user, now employing several Pyramix systems configured for multitrack recording duties in its London TV studios and on outside broadcasts. The Pyramix software is compiled to run only on PCs under Windows NT and XP operating systems — and there is no prospect of a Mac version, so Apple diehards should turn the page now to avoid further disappointment! The Pyramix LE and range-topping Virtual Studio (VS) systems run on a PC platform incorporating dedicated hardware DSP cards, and this review was carried out on a well specified VS system running the version 4.3 software with the latest SP2 update. As has been hinted at earlier, the core VS system can be expanded and customised with numerous hardware and software options to fine-tune the features of the complete system to suit various applications, such as audio mastering, audio-for-video post, and so on. The Pyramix Native version is a software-only configuration that uses the host computer's own microprocessor to perform the audio DSP, and as such it is limited in its functionality compared to the full VS version. However, the form and functionality of the program is identical for the Native and VS systems. The two versions look and work identically, and projects created with Pyramix Native are fully compatible with the Virtual Studio system, and vice versa. Machines running Pyramix Native and Pyramix VS can even be connected on the same standard



Ethernet network to enable the direct interchange of raw audio (and video) media, as well as entire projects. Indeed, many facilities houses employ the costeffective Pyramix Native version for pre-production and track-laying work, networked with full Virtual Studio machines to handle the final mixdowns and large project compilations. The Native software is protected and authorised with a USB dongle, whereas the VS system only runs if the bespoke hardware DSP cards are installed in the computer. The Native version has limited functionality compared to the VS system, providing only stereo audio in and out, with sample rates up to 48kHz. It is also restricted to four playback channels in the mixer, but all editing and audio manipulation is performed in real time (no rendering) with the same 32-bit floating-point maths as the full version. In addition to a small but functional range of built-in signal-processing tools such as equalisers and dynamics, Pyramix Native is fully compatible with Direct X and VST plug-ins, and the software supports PMF (Pyramix native file format), WAV, BWF, AIFF, SD2, OMF and CD Image audio file formats as standard. The Native Media Bundle option extends this feature set with double the number of mixer playback channels (eight), more elaborate signal-processing facilities, mastering-quality metering including a phase correlator and audio vectorscope, and sophisticated tools to support CD mastering, video playback and MIDI synchronisation, the last using Merging's own very clever Virtual Transport technology. The Pyramix Native and Media Bundle software can be used with a variety of soundcards — basically, any model will work that is fully compatible with the Windows Direct Sound and Direct Capture format, and supplied with WDM drivers. However, Merging recommend four specific interfaces: the RME Fireface 800, Echo Indigo, MOTU 828 and Yellowtec PUC. The Pyramix LE version is supplied as standard with the Mykineros hardware (see below), and provides much the same facilities as the Native Media Bundle but with double the number of mixing channels again (16). However, the real power of the hardware is only realised by moving up to the full Virtual Studio system which brings a wealth of additional facilities and flexibility — not the least of which are up to 64 inputs and outputs with corresponding mixer functionality, plus built-in timecode generation and external clock reference capability, allowing the system to be integrated into more sophisticated digital and video-based environments.

Mykerinos Hardware The processing heart of the Pyramix LE and Virtual Studio systems is the Mykerinos DSP hardware board, which plugs into a standard PCI card slot. This Mykerinos board uses a Philips Trimedia VLIW processor — a DSP chip originally designed for high-speed video signal processing — which is capable of file:///H|/SOS%2005-06/Merging%20Technologies%20Pyramix.htm (3 of 13)9/28/2005 2:33:54 PM


handling 128 simultaneous channels (64 inputs and 64 outputs) of 24-bit digital audio. Each card provides a stereo analogue monitoring output as standard, with an internal link buss which connects to various optional daughter cards that provide the standard analogue and digital I/O interfaces. As mentioned above, the DSP uses 32bit floating-point maths and can accommodate all sample rates up to 192kHz. Systems equipped with multiple Mykerinos cards (which must be installed in adjacent PCI slots) can also be configured to support the SACD (Super Audio CD) data format (DSD) with an additional software option. When suitably equipped, Pyramix can also carry out multitrack DSD recording and editing, currently providing up to eight tracks, although Although Pyramix systems can be configured there are plans to support 24 tracks in using any suitably specified PC, most buyers a future software update. Pyramix also opt for Merging's own rackmounting design. supports the DXD format which is growing in popularity as a highresolution stepping stone for DSD production, with linear PCM files running with 32-bit resolution and a 352.8kHz sample rate. Although it is hard to see why such a powerful system would ever be required, a single Pyramix system can include up to eight Mykerinos cards — assuming you can find a PC motherboard with sufficient PCI slots. The DSPs are linked together using a dedicated 128-channel internal data buss, hooked up with ribbon connectors on the top of each DSP card. As already mentioned, the signal I/O is accommodated with a range of daughter cards which plug directly to the Mykerinos boards. Currently, daughter cards are available to provide ADAT, AES, S/PDIF, TDIF, MADI and even good old-fashioned balanced analogue interfaces, and multiple cards can be connected to provide the required number of channels and types of interface. For post-production applications, both longitudinal (LTC) and vertical interval (VITC) timecode can be read and generated by the Mykerinos board, and a Sony 9-pin machine control interface is also included. Another key feature of the Mykerinos board is a very-low-jitter internal clock system, which can also be slaved to external video or digital clock references. Although this is perhaps less obvious, a very important element of the system is the graphics board. Most Pyramix systems are equipped with a high-quality dualhead graphics card as standard — as with most complex DAWs, spreading the various component windows over two screens makes it a lot easier to see what is going on and operate the system more efficiently. However, for working in the sound-for-picture domain a separate video capture/playback card is usually specified too, to allow the synchronised internal video playback to be routed to a



third screen. Ideally, where integral video playback is required, a dual-processor motherboard would be specified as well, simply to enhance the overall system performance when shifting large amounts of video data around. On the review model, the video clips related to various demonstration dubbing projects preloaded on the system were replayed over a sub-window sitting on the main playlist screen — and it frequently got in the way! Thanks to the standard PCI format of the various hardware DSP and interface cards, the Mykerinos system hardware can be fitted in any suitable PC chassis. However, Merging supply the vast majority of systems as fully warranted turnkey packages installed within a very nice custom-engineered rackmounting chassis. These rack cases contain a number of fans and are too noisy to be located in a quiet control room, but some attention has been paid to the noise level, and many systems are a lot worse!

Operation In the early days of magnetic tape recorders, there were many variations between manufacturers concerning spool directions, tape widths, capstan speeds and even the physical arrangement of transport controls. Yet after many years, a standardised set of formats and configurations emerged, and most machines shared a common arrangement of operating controls. Consequently, studio engineers could load and operate tape machines made by Tascam, Studer, Otari, ATR and other manufacturers with complete confidence. In the early days of the computer DAW, different manufacturers likewise often had radically different approaches, and no two machines looked remotely similar, let alone shared common tools or working practices. However, after a decade of evolution and convergence, the same kind of conformation that eventually applied to reel-to-reel recorders is finally becoming evident for computer The Pyramix LE and VS systems are based DAWs too. While there will always be around the Mykerinos DSP card. some detailed differences in the specific arrangement of tools, buttons, screens and menus, it is clear that the main facilities and features, operating practices and even the look of the displays of many different DAW products are converging into a form which makes it far easier for users to transfer their expectations, experience and craft skills from one manufacturer's machine to another. This is particularly important in the professional environment as the workforce is increasingly freelance, moving from facility to facility — and thus using different manufacturers' equipment — as the work requires.



Merging Technologies are obviously very aware of this and, like many other DAW systems, the keyboard shortcuts assigned to Pyramix's various operational functions can be fully customised to suit personal preferences and operating techniques. To make operating Pyramix even easier and faster for users more familiar with other DAW systems, three factory presets are also included, which will instantly reconfigure the keyboard shortcuts to exactly the same arrangements used on standard Pro Tools, Sonic Solutions and SADiE DAW systems. As a long-time user of SADiE, I found this feature invaluable: it enabled me to get to work straight away on Pyramix, using all my familiar keystrokes to perform the main editing functions. Furthermore, being able to translate my editing experience and skills so quickly and easily inspired confidence in, and respect for, the system. Having said that, most of Pyramix's operation is pretty intuitive anyway, and anyone with a basic understanding of audio editing and production will quickly be able to find their way around the system, using the default Pyramix shortcuts and tools. The various windows are all clearly laid out and most of the function buttons are marked with obvious legends or icons — and the pop-up help can always be used to remind the user of the function of a specific button if the mouse pointer is allowed to hover over it, of course. Another point in the system's favour is that the overall look of the Pyramix graphical interface is pleasantly understated and simple. The screen colours are generally fairly muted greys — apart from the audio clips, which can be highlighted with bright colours to help identify different kinds of clip — which makes it a lot more comfortable to stare at for hours at a time. A quick health and safety reminder might be apposite here, though: regular screen breaks are essential. Move about, relax those shoulders, refocus the eyes, and stretch those back muscles regularly!

SP2 Update The latest service pack software update for version 4.3 software has brought a range of useful new facilities. The headline addition is support for AAF File Interchange, an internationally agreed file format which describes the EDL information (clip names, comments, source tracks, timecodes, fade curves, pans and so on) in such a universal way that it provides genuine cross-platform compatibility with any other DAW or non-linear video editor that also supports the format (including Pro Tools, SADiE, Avid Media Composer and many more). In addition, the CMX file import and export facilities now support up to 64 tracks. In fact, Pyramix is able to import projects (either natively or via AES31, OMF, AAF or Open-TL formats) from Sonic Solutions, Nuendo, Pro Tools, Tascam, SADiE, Fairlight, Akai, Soundscape and AMS-Neve Audiofile. Another new feature is a 'virtual tape mode' in which destructive record punch-ins and punch-outs can be performed on a standard BWF file. It may sound like an odd thing to want to do, but this is actually a key requirement of many filmdubbing theatres. A new 'quick mount' feature speeds up the process of opening projects and mounting the relevant media files, and the 're-conform' functionality has been improved to provide automatic and completely seamless replacement of file:///H|/SOS%2005-06/Merging%20Technologies%20Pyramix.htm (6 of 13)9/28/2005 2:33:54 PM


the 16-bit mono or stereo audio files used in non-linear video systems with the original 24-bit multitrack source files when working with EDL, OMF or AAF project information. At an operational level, one of the most important new features is the inclusion as standard of the advanced crossfade editor window. This allows very precise realtime manipulation of edits by adjusting all the crossfade parameters, including gain, curve shape, duration, position and so on. Fades can also be linked, unlinked and mirrored as required. For complex music editing work, a new multi-point, multi-channel source-master editing facility is now standard in Pyramix (this DAW editing technique is known variously as three- or four-point editing, or source-destination editing). An unlimited number of source and master (destination) tracks can be created, with facilities to group and collapse tracks for easier editing of multi-take and/or multitrack recordings. Something that has been available in many 'simple' DAWs for a very long time, but which is missing from most high-end devices, is a 'pencil tool' editor. This latest Pyramix update has finally introduced a simple pencil tool that allows unwanted glitches in an audio waveform to be redrawn manually. For CD mastering applications, the PQ coding facilities have been enhanced to support hidden tracks and customisable ISRC parameters, and the Pyramix DSD option has been extended with facilities to create fully compliant surround and stereo SACD masters. A range of new CEDAR plug-ins has also been introduced, including the Retouch audio restoration tool, which allows a sound clip to be manipulated in the frequency domain. Selected areas can be interpolated over time or frequency (or both) to remove unwanted elements from a recording. There is also a new multichannel time-compression and expansion plug-in developed by Prosoniq called MPEX2, and all VST plug-ins can be used. Finally, there is a new 'cue sequencer' play-out function which can replay numerous audio cues via any available output connection — a system intended mainly for theatre applications. Cues can be triggered manually or from MIDI timecode via a dedicated operating page. Besides all these new and improved features, the update incorporates a number of important bug fixes.

Top Down Pyramix is a comprehensively equipped, state-of-the-art professional DAW system. Consequently, it is not practical to describe here every detail and nuance of its operation — that's what the 400-odd page handbook is for! However, I will try to give a flavour of what it is like to use, and highlight any unusual or particularly well designed aspects. As in most DAWs, there are two main window elements that I found I kept open most of the time in Pyramix. The first is obviously the 'timeline', the main playlist area comprising an unlimited number of tracks, each carrying the desired audio clips positioned in time and allowing them to be assembled and organised, and where most of the straightforward editing is accomplished. Arguably the mostused function in this window is zooming in and out, both horizontally in terms of file:///H|/SOS%2005-06/Merging%20Technologies%20Pyramix.htm (7 of 13)9/28/2005 2:33:54 PM


the time duration visible on screen and vertically into the various track groupings. On some machines these zooming processes can be frustratingly awkward, but not in Pyramix — the screen can be zoomed on both axes independently using dedicated buttons, keyboard shortcuts, click-and-drag mouse functions, or via any standard external controller panel. In fact, pretty much all of Pyramix's functions can be controlled in several different ways using the mouse, keyboard or external controllers — which means everyone can find an operating technique that suits them. The second main window which I kept up most of the time was the virtual mixer, although once a rough mix was established I found I rarely made alterations (other than to the monitoring level) until I came to the final mix stage. The mixer panel looks Pyramix's virtual mixer might appear simple, deceptively simple at first glance, but but it can be configured for complex routings everything you would expect to find on and processing setups. a comprehensive mixing system is available at the press of a button: stereo and surround panning, group and output routing, signal-processing tools, dedicated monitoring busses, and so on. Each mixer input strip can be configured for mono, stereo or surround, and can handle physical inputs or any number of timeline tracks. MS decoding facilities are incorporated, and for surround work, up to eight six-channel mix stems can be created. The physical input and output assignments are indicated with rather attractive XLR icons marked with the appropriate physical I/O connection identifier, and there are facilities to route effects sends and returns either to internal (plug-in) or external processors, complete with the appropriate automatic delay compensations. The next most frequently used windows are undoubtedly the crossfade editor — vital for fine-tuning difficult edits (see below) — and the timeline overview window, which displays a 'thumbnail' of the entire timeline in a long strip to help navigate around lengthy and complex projects. When working with external transports, then the Virtual Transport control panel would also be referred to regularly to operate remote transports and set up their relative timecode offsets and so forth. This Virtual Transport technology is interesting in its own right as it is a network-based, multi-transport synchronisation system linking a wide variety of real and virtual machines handling video and audio, including external MIDI and Sony 9-Pin devices. Virtual Transport is essentially a client-server architecture, allowing various applications to communicate together through a common interface and to be synchronised to the same timecode. Virtual machine applications can be running on the same computer or over the network on different computers, and once set up, the communication and synchronisation processes are entirely transparent for the user.



VCube The latest addition to the Pyramix family is the VCube, a hard-disk video player/ recorder system designed specifically for use in audio post-production environments. The VCube can operate as a stand-alone unit, or linked to other Pyramix systems via Gigabyte Ethernet, and can be synchronised to all the standard video, film and HDTV frame rates, with 9-pin machine control facilities. The Vcube is available as either a player only, or as a player/recorder, both being housed in a custom 4U rackmounting chassis with a 120GB media drive as standard. The player-only option is intended for use in large-scale installations where a central player/recorder can be used to digitise video material and distribute it to the players via Ethernet. A special OEM version of the Canopus ADCVX1000 video-capture PCI card is available, as well as an SDI, composite and component card. Basic video-editing functions are supported by both player and recorder models — while the VCube is not intended as an editing system, the ability to trim or modify an existing programme often enables post-production to continue without having to wait for a new master video file. The VCube is also able to import and stream multi-layer Avid Video OMF Compositions either by logging a suitable disk drive or via an Avid Unity Network connection, helping to streamline the workflow in post-production facilities houses.

Starting A Project When starting a new project, a simple 'project wizard' helps to configure everything in a fast and straightforward way. The first step is to choose the kind of project (standard recording, editing or mixing, batch digitising, DXD or DSD recording/editing/mixing) and, if relevant, the sample rate and word length. Next, the project is named and a suitable folder and file location on the hard drive used to store the audio and project data is nominated. This is followed by selecting or designing the required mixer configuration (another wizard can be run to help configure a new mixer, although there is a comprehensive catalogue of predefined stereo and surround assignments), and then assigning the physical I/ Os so that you can get signals in and out. If you always use the same connection setups you can save mixer templates for use later. Once these steps are completed, audio can be recorded into the machine, or existing audio files located on any connected drives can be imported. Each project has its own associated libraries in which to store related audio, but there are also global libraries in which generic material such as sound effects libraries, house stings or music beds, and so on can be stored and accessed from all projects and by all system users. Pyramix notices if you try to import audio with a different sample rate to the current project, and a built-in sample-rate conversion facility is available to convert files, either individually or in batches. To place audio clips in the timeline, files can either be copied and pasted from the media drives directory lists or the project libraries, or simply dragged across from the directory window into the timeline window. All very intuitive and simple.



Something I always seem to spend a lot of time doing, especially when working with radio or TV soundtracks, is entering timecode values to place clips in specific positions in the timeline. One particularly nice Pyramix feature that I found very useful is the Simple fades can be handled in the main ability to type in a relative timecode timeline window, but for more complex work value instead of a full timecode. For there's a dedicated crossfade editor. example, suppose I wanted to place a music sting 30 seconds from the timeline's present position of 10.00.05.00. Instead of having to type in 10.00.35.00, I could press the '*' key on the numeric keyboard to load the current timeline timecode, and Pyramix would then happily accept +30.00 and work out the new timecode value itself. OK, so this is a very simple example, but this feature really comes to the fore when you need to enter a relative value of 2.43.17, for example! My mental arithmetic is certainly not up to complicated timecode sums like that, so I really valued this simple but very effective facility. Another nice aspect of Pyramix is the way it can be set up easily as a conventional multitrack recorder, with some very efficient housekeeping functions built in. Each recording is logged automatically with sequential take numbers, and after each pass the recording can be flagged as a good or bad take. Both are saved to the hard drive but they are automatically highlighted in different, user-configurable colours on the timeline to make it obvious which are which. If a take really is no good it can be deleted to save disk space, of course, and any number of individual tracks can be over-recorded using normal punch-in and punch-out techniques. Related techniques are the batch digitising and autoconforming functions, in which audio can be recorded to the hard drives as a background process, or organised into a timeline according to a supplied edit decision list (such as from a video edit). I wasn't able to use these facilities myself, but I have seen the machine used in these ways in earnest, and it appears to work very slickly and reliably.

Editing Basics When a 'composition' is assembled in the 'timeline', each audio clip is shown as a coloured block containing the audio waveform. The block has six control 'handles' — small squares at each corner and at the centre of the left and right ends — and dragging these blocks adjusts the length of the clip, and the length and position of the fade-in and -out. The centreline blocks move the clip start and end points, while the corner blocks adjust the fade starts and ends — and if you hold the Ctrl key, a symmetrical crossfade is generated automatically with the appropriate adjacent clip. The other modifier keys (Shift and Alt) can be used in various combinations to allow, for example, a selected clip to be slipped in time, or for the clip's audio to remain static but its in and out points to be slipped in time. If required, the unused portion of a clip can also be displayed as a greyed-



out waveform, which can be helpful when trying to locate specific edit points within a longer recording. Text labels within the coloured blocks provide the name of each clip and its relative gain, and a vertical red line (with another control handle) indicates the clip's 'sync point'. By default, the sync point is aligned with the start of the clip and is the timing reference used to position the clip in the timeline. However, it can be moved to indicate a more appropriate reference point — the start of the vocals after an instrumental intro in a music track, for example, or the crash following a skid in a sound effect. The clip can then be located by instructing the system to place the sync point at the cursor position in the timeline, making it very easy to time music and effects precisely relative to other sound elements. Along with all the standard facilities such as deleting and inserting sections of audio with the rest of the track rippling along (or not), clips can be protected against accidental modification, or grouped together for easier manipulation. A nice feature Pyramix's transport panel. included as standard in Pyramix (but which fell out of SADiE a long time ago and has only recently been reintroduced) is the ability to remove silent sections of a clip automatically. The silence threshold and minimum lengths of sound and silent sections can be adjusted to achieve the desired effect, and this facility is great for defining rough edit points in interviews, removing unwanted background noise in vocal takes, and so on. If the tool is set up to remove even the shortest silent sections and to close up the gaps afterwards, impossibly fast 'ad-speak' messages can be created in seconds. Ideal for getting all those legal 'small print' disclaimers into the last 10 seconds of the advert! While most editing functions can be performed perfectly well by manipulating the clip handles within the timeline window, for really serious and complicated editing, the crossfade window is the tool for the job. This separate display window looks very reminiscent of a similar facility in Sonic Solutions, although most DAWs have an equivalent function somewhere — SADiE calls it the Trim Window, for example. The idea is to allow very precise control of fades and crossfades, either by using the mouse to drag out the required curve shapes and positions, or by using on-screen sliders and numeric displays. All the usual facilities are provided to audition the selected crossfade, as well as the incoming and outgoing sides of the edit, adjusting the gain of the clip on either side of the edit, and changing the fade shapes and durations. In days of old when editing quarter-inch tape, the choice of fade shapes was rather limited — always linear but with a couple of different fade durations courtesy of the cut angle — and shallow angles could easily result in 'flashing edits' where one channel was interrupted before the other. A big advantage of the DAW is that all channels are inherently edited and faded at the same time and rate, so no more flashing edits, and the fade shape can be varied in very



sophisticated ways to allow previously 'impossible' edits to be made seamlessly every time. All the usual fade shape suspects are provided here — linear, constant-power, cosine and so on — allowing even the most awkward of edits to be performed inaudibly given a little experimentation with the fade shapes, positions and level parameters.

Conclusion Pyramix is a very sophisticated, extremely powerful and remarkably flexible DAW, and it rightly deserves its position among the top-flight audio editing platforms. With its highly intuitive graphical interface and customisable control operations, the translation to Pyramix from other similarly specified DAWs is surprisingly quick and easy, and I found I was performing complex music edits, compilation and CD mastering efficiently and accurately literally within minutes of sitting down in front of the machine for the first time unaided. The more complicated processes of track laying to picture and building up surround stems required a little more familiarity, but I quickly felt very comfortable using the Pyramix Virtual Studio system, and was able to work quickly and with confidence. The remarkably comprehensive handbook is provided as an indexed PDF file, with over 350 pages of information — an indication of the flexibility and complexity of the Pyramix system — and it is written very well with clear descriptions and instructions, Video handling is, as you might expect, one and plenty of diagrams and screen of Pyramix's strong points. illustrations. There were no comedic translation errors either, despite its Swiss origins. An additional Quick Start guide was also supplied as another indexed PDF file, this time with a mere 84 pages, and between the two I quickly found the answers to any queries or uncertainties I had as I worked through various projects. When it comes to more obscure operations or problem solving, or just a need for the most up-to-date information, Merging Technologies run a couple of free-access Internet forums where users and support staff regularly exchange information, advice, hints and tips. In my previous professional occupations I have taught the operation of Sonic Solutions, Pro Tools and SADiE systems for music editing and CD production, radio production and TV dubbing applications, and have acquired reasonable knowledge of many other DAWs including the AMS-Neve Audiofile, Akai DD1500 and, of course, Cool Edit Pro (now Adobe Audition). All are very capable systems in their own way, and all have their own ways of doing some things, but translating that experience to the Pyramix was completely painless and I enjoyed using this new system very much. It is easy to see why it has become such a popular tool in the broadcast and postfile:///H|/SOS%2005-06/Merging%20Technologies%20Pyramix.htm (12 of 13)9/28/2005 2:33:54 PM


production industries in such a relatively short period of time. It is logical and easy to use, with every function well designed, as well as being easily accessible and configurable. Furthermore, the standardised interfacing options, which allow the simple connection of other digital equipment and third-party outboard converters, for example — as well as the support for a wide range of file types and data-interchange formats — make it easy to integrate Pyramix, both in terms of physical I/O connections and within a production workflow with audio, project and EDL file transfers. As far as audio quality goes, I couldn't fault Pyramix in any way. Its floating-point maths appeared to cope with internal buss levels greater than 0dBFS without any problems at all, and internal sample-rate conversion and word-length reduction with dithering all seemed flawless. While the mighty Pro Tools will remain one of the leading industry-standard DAWs for some time to come, Pyramix is certainly nipping at its heels in many parts of the audio industry, and after using it for a few days I can quite see why. If you are looking to install a state-of-the-art DAW — or a hard disk-based multitrack replacement for that matter — then this is a very serious contender indeed, at any level. Likewise, audio technology courses would be well advised to introduce their students to Pyramix, even at the Native level, as I can see this becoming a very widespread tool in years to come. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



MOTU Traveler

In this article:

Full Power! Specifications In Use Software & Compatibility Sound Quality The Bad News? Conclusions

MOTU Traveler Firewire Audio Interface [Mac/PC] Published in SOS June 2005 Print article : Close window

Reviews : Computer Recording System

MOTU Traveler £795 pros An excellent-sounding, wellequipped interface that's also extremely portable. Good-quality, low-noise mic preamps, with 73dB of gain and digital gain controls. Flexible powering options. ADAT, S/PDIF, TOSlink and AES-EBU digital connections. Decent control software for PC and Mac.

With two market-leading Firewire audio interfaces already part of the MOTU stable, where does the new mobile recording-oriented Traveler find its natural home? Robin Bigwood

cons Front-panel operation takes some getting used to.

summary The Traveler strikes a superb balance between audio performance, digital and analogue connectivity, and portability. A pleasure to use — thoroughly recommended.

information £795 including VAT. Musictrack +44 (0)1767 313447. +44 (0)1767 313557. Click here to email www.musictrack.co.uk www.motu.com

Test Spec MAC REVIEW SYSTEMS Dual 867MHz Apple Mac G4 with 1.25GB of RAM, running Mac OS v10.3.8.

MOTU have been one of the most enthusiastic supporters of the IEEE1394 Firewire protocol for audio use, and their original 828 (Mk I) interface is still a sought-after and useful tool for studio-based and mobile sound recordists alike. Firewire has proved itself to be a flexible and reliable platform for audio, capable of handling multiple channels of high sample-rate audio, and proving to be the perfect solution for laptop users who want or need to steer clear of USB. Time has moved on since the original 828 came out, though, and whilst MOTU's current 828 MkII and 896HD interfaces are both attractive, prospective Firewire interface buyers have plenty of other models to choose from. Where does the Traveler fit in to all this, and what does it have to offer? On the face of it, the Traveler is quite similar to the MOTU 828 MkII, reviewed in the July 2004 issue of SOS, in that it's a multi-channel interface with eight analogue inputs and outputs, ADAT and S/PDIF digital audio connections, plus MIDI In and Out and SMPTE sync facilities, but it differs in a few important ways. The Traveler can operate at sample rates up to 192kHz, has AES-EBU digital I/

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MOTU Traveler

1GHz Apple Mac G4 Powerbook with 768MB of RAM, running Mac OS v10.3.8.

O, and the four mic/guitar inputs (the 828MkII only has two) are mounted on the rear panel. Additionally, the gains for these inputs are digitally controlled — MOTU calls them Digital Precision Trims. The other big change is that the Traveler has no IEC mains power inlet — instead it can be powered by DC adaptor, battery pack (via a standard four-pin XLR connector), or via the Firewire connection itself. And this is not the only change that makes the Traveler more suited to mobile use...

Full Power! The Traveler can be powered solely by the Firewire connection to the host computer, but only if a six-pin Firewire connector is used at the computer end — miniature four-pin connectors won't work. Used this way with an Apple 1GHz G4 Powerbook, the Traveler gave me a very respectable two and a quarter hours of use before my Powerbook battery ran down, and this was with none of the laptop's energy-saving options enabled. Obviously the Traveler can run all day long if the host computer is plugged into the mains, but for more flexibility on the road two additional powering options are available. First, any mains power adaptor can be used so long as it provides between eight and 18V with sufficient current — 1.33A is needed for a 9V supply, but the Traveler isn't fussy about whether tip connections are wired positive or negative. Second, the battery packs commonly used in film and TV circles can be pressed into service via the four-pin XLR socket on the side of the Traveler. Any can be used as long as they supply between 10 and 18V, and 12W — names to look out for in the UK include Hawkwood and PAG, though US suppliers such as www.bhphotovideo.com often have deals on brands such as Bescor.

Specifications From just seeing the box the Traveler comes in, you know it's a lot more compact than other 19-inch rack-based interfaces. It's still a 1U height, and, at 25cm, reasonably deep, but the Traveler's width is only 37.5cm. Not coincidentally, this puts it at about the same size as an average 15-inch screen laptop. It's not pocketable by any means, but capable of fitting in a laptop or other similarly sized bag. At 1.8kg it's surprisingly light, and its case, with friendly rounded corners and edges, is manufactured not from plastic but a more confidence-inspiring aluminium alloy. I was relieved to see, too, that most of the rear-panel audio sockets are case-mounted, so 'enthusiastic' plugging in won't stress the Traveler's weaker internal components. The front panel is made from the same alloy, and the controls have a good, solid feel to them. I did notice that pressing the rightmost knobs caused a slight inward movement of the LCD unit, but further investigation revealed that this was nothing serious. As with any other audio equipment, users should probably be wary of allowing any real force to be applied to the front-panel knobs while the Traveler is being transported. If you did have a particularly tough life lined up for the Traveler you'd probably be advised, anyway, to use the 19-inch rack 'ears' that MOTU supply, and mount it in a heavyduty case. During the test period I was happy to be fairly 'robust' with the Traveler, and it handled everything I threw at it (not literally, I hasten to add!) with file:///H|/SOS%2005-06/MOTU%20Traveler.htm (2 of 9)9/28/2005 2:33:59 PM

MOTU Traveler

no trouble at all. For what is a compact unit, the Traveler packs in a lot of connectivity. To begin with the rear panel has four Neutrik combo XLR/quarter-inch sockets, catering for mic, guitar and line inputs. There are four additional line-level inputs on quarter-inch sockets, switchable in pairs (again via the front panel) between -10dBV and +4dBu operating levels. These have no gain stages other than an optional 6dB 'boost' which is achieved in software, not as genuine 'electrical' gain. Analogue outputs are on eight quarterinch sockets fixed at +4dBu. It's good to see that the spacing of all the quarter-inch sockets is wide enough to allow the use of chunky Neutrik jacks, something which causes problems on the 828 MkII.

The bundled Cuemix Console software controls the Traveler's zero-latency monitoring. Any four output pairs can be chosen to carry completely independent monitor mixes, and settings can then be saved as presets for later recall.

As for digital connections, the Traveler has eight-channel ADAT 'lightpipe' in and out, although the optical sockets used for these can be switched, independently, for optical S/PDIF (TOSlink) usage instead. S/PDIF signals are otherwise catered for on dedicated RCA sockets, but it's not possible to use these and optical S/ PDIF simultaneously. TOSLink gets priority if you try. The pair of male and female XLR sockets for connecting AES-EBU devices are unaffected by S/PDIF settings. All the digital connections can handle 24-bit signals at 44.1 or 48kHz sampling rates without restriction; in fact the AES-EBU and S/PDIF can handle 88.2 and 96kHz as well, but at these '2x' sample rates only four channels of ADAT input and output are possible. At the '4x' 176.4 and 192kHz sample rates, the Traveler's digital connections are disabled. Rounding off the Traveler's connections are word clock in and out on BNC connectors, a nine-pin ADAT sync socket, a pair of Firewire sockets, and on the right-hand panel, a pair of MIDI sockets (In and Out), a four-pin XLR power pack input, and the DC power socket. Moving round to the front panel, there's a slim rocker-type power switch, four 48V phantom switches and endless encoder-type gain knobs for inputs 1 to 4, and six additional endless knobs for working with the Traveler's Cuemix Plus monitoring and stand-alone functions. The headphone socket can be set up to 'mirror' any output pair but can just as easily be configured as an additional, independent pair of outputs. The accompanying Volume knob allows you to independently adjust volume for headphones and the Traveler's main output pair — a great feature. file:///H|/SOS%2005-06/MOTU%20Traveler.htm (3 of 9)9/28/2005 2:33:59 PM

MOTU Traveler

Display duties are taken care of by a bright 2x16-character backlit LCD and, to its right, there's a panel of LEDs showing input levels and output activity for the analogue, S/PDIF and AES connections, ADAT and MIDI activity, and SMPTE and clock status. All in all, the Traveler is an exceptionally well-equipped interface, and with the exception of the shared ADAT/TOSlink optical connectors, there are no real catches with input- or output-channel provision — it looks like you get 20 inputs and 22 outputs, and that's just what you do get.

There are so many connections on the Traveler's back panel that MOTU had to site the MIDI I/O and power connections around the side.

Just as impressive, if not more so, are some of the Traveler's 'under the hood' features. The Cuemix Plus system allows four independent monitor mixes of all 20 inputs to be set up on any four pairs of outputs, at the same time as recording if necessary. The obvious application for this is indeed setting up headphone or foldback mixes whilst tracking, but it also effectively turns the Traveler into a digital mixer (20:6:2, I suppose), albeit one with an unconventional set of controls, and no EQ, inserts or auxes. Never the less, for classical and live sound recordists the Traveler can simultaneously fulfil conventional audio interface duties for an attached laptop as well as supplying a 'safety' mix to a two-track recorder, which would remain unaffected in the case of a computer crash. What's more, Cuemix Plus is configurable from software (more on this later) and from the Traveler's front panel, so genuine 'stand-alone' operation is certainly possible. For users in the audio-visual field who work with timecode, the Traveler can both sync to and generate SMPTE using its analogue connections. This feature is configured with the bundled Firewire SMPTE Console software (shown opposite), and is easy to work with whilst providing a sophisticated implementation including freewheel and regeneration options.

In Use The Traveler inspires confidence from the moment you take it out of the box. The construction quality is excellent, and all sockets and controls have a nice solid feel. Setting up and software installation is very straightforward and it's quite a liberating moment when you power up a laptop-based system for the first time and realise there's not a mains lead in sight. Switching on the Traveler causes the LCD to display a brief 'splash' screen reminding you that, yes, you're using a MOTU Traveler, and also showing the firmware version in use (mine was v1.04). After that, the display switches to a 'Mix Bus' screen that gives a simple visual representation of the current Cuemix Plus settings for a single pair of audio outputs. This is the way the Traveler appears for the vast majority of the time during use, so if you buy one you'd better learn to like it!


MOTU Traveler

Like MOTU's other respected interfaces, the Traveler packs a lot into its 1U rack casing, including eight analogue ins and outs, word clock, ADAT optical, S/PDIF and AES-EBU connections.

Using the supplied software, nearly every feature of the Traveler can be controlled from a connected computer — only phantom, main power, and main/ phones output level can't be — but as I mentioned earlier, there is extensive 'stand-alone' provision too. This includes basic setup functions, such as sample rate, optical mode and display options, but also encompasses all aspects of the Cuemix Plus internal mixing environment. It would be much too tedious to go into detail about this, but despite good visual feedback from the LCD, I did find the front-panel operation of the Traveler a little non-intuitive sometimes. For example, there's one knob marked 'Mix Bus' and another marked 'Mix', and they do quite different things under different conditions. Similarly, 'Select' often changes the value of a parameter, and yet there's a 'Value' knob as well! What can really get you, though, is that every knob can also be pushed like a switch. This is confusing because sometimes doing this accesses a crucial function, sometimes it's a trivial shortcut, and sometimes it does nothing at all. I don't want to give the impression that the Traveler is hard to use or that its software is confusing — far from it — but new owners should probably set aside a little time to get used to things before using the Traveler at an important gig. Something I have no complaints about is the front-panel volume knob. As mentioned already, this performs two roles (it's switchable by pushing the knob) in adjusting headphone volume and the level of the output sent to analogue outs 1 and 2, to which you'll probably have your monitors connected. In one fell swoop, this knob might do away with the need for a dedicated monitor controller, especially when you take into account Cuemix Plus's talkback and listenback facilities. I couldn't help wondering whether MOTU might be able to provide an option for the volume knob to control the output level of Analogue outs 1-6, thereby allowing the Traveler to act as a volume control for a surround setup. That really would be a money-saver for people working in that field.


MOTU Traveler

Software & Compatibility The Traveler ships with Core Audio drivers for Mac OS X (10.2 or later) and WDM, ASIO2 and GSIF2 drivers for Windows ME, 2000, and XP. Users of both platforms also get appropriate MIDI drivers as well as MOTU's Firewire Audio Console, Firewire Cuemix Console and Firewire SMPTE Console applications. Together these take care of all aspects of the Traveler's basic configuration, zero-latency monitoring and timecode synchronisation abilities respectively, and are all robust and easy to use.

The MOTU Firewire Console application handles basic setup of the Traveler, although all these settings can also be made from the frontpanel controls.

Mac users additionally get a freebie audio recording application, Audiodesk, which is a very capable cut-down version of MOTU's flagship Digital Performer software. Audiodesk lacks any MIDI support, but it does come bundled with a healthy selection of MAS (MOTU Audio System) plug-ins and is a useful, solid application that may well satisfy all the needs of some users. If you already own one of the main sequencing packages, the printed Traveler manual includes information on how to set it up with your new audio interface. For Mac, there are sections on Digital Performer and Logic, and for PC Cubase, Nuendo and Sonar.

Sound Quality The Traveler's 64x oversampling A-D and 128x oversampling D-A converters support sample rates up to 192kHz, at 24-bit resolution. Aside from that, MOTU don't publish any specifications or measurements for either the converters or the analogue I/O. Whilst that may make it harder for prospective purchasers to make a comparison between the Traveler and other similar interfaces, the on-paper specs for virtually all modern audio interfaces and digital mixers are so good, and so broadly similar, as to make comparison almost meaningless. In any case, it's always the subjective 'feel' factor that is much more important to end users, and it's that that I'll focus on here. The Traveler's four mic preamps are equipped with MOTU's new Digital Precision Trims, which allow gain to be set precisely, in 1dB increments, and even to be stored as part of a Cuemix Console preset for later recall. To get an idea of their quality, I compared them to two other mic preamps I own: the SPL Goldmike MkI valve/transistor hybrid, and an M Audio DMP2. The DMP2 is of similar quality to the VLZ Pro preamps used in older Mackie mixers but offers file:///H|/SOS%2005-06/MOTU%20Traveler.htm (6 of 9)9/28/2005 2:33:59 PM

MOTU Traveler

more gain, whilst the Goldmike is often considered to be one of the best preamps on the market short of really big-money models. To cut a long story short, the Traveler mic pres are excellent. They're more fluid-sounding than the DMP2, especially at higher frequencies, but leaner too, with more transparent lowmids. Putting them up against the Goldmike, I was a little shocked to hear how similar the two units sounded, and it was only with quite careful and detailed listening that a little congestion For those working with video and timecode in the Traveler's upper mids and a audio sources, the Firewire SMPTE Console slightly 'colder' overall balance became software configures the Traveler for receiving and generating SMPTE via its analogue apparent. I'd have no hesitation connections. whatsoever in using the Traveler mic preamps for almost any application — they return a colourful, involving sound with bags of detail and a convincing musicality. I was also surprised to discover that a whopping 73dB of gain is on offer — this is just what's needed for low-output dynamic and ribbon mics, and compares favourably with the 60dB on offer from a Mackie Onyx preamp, for example. The gain comes without a noise penalty, too. As someone who works with classical musicians a lot, using small diaphragm mics in fairly distant locations, I was delighted to discover that even with maximum gain dialled in, the sound remained transparent and fluid, and with an impressively low noise floor. Compared to the Goldmike's maximum 72dB of gain there was nothing to choose between the two in this respect. I also checked out DI'd electric bass into one of the Traveler's mic pres with its 20dB pad engaged. As expected, the sound was lively, articulate and reassuringly 'ballsy' without being obviously coloured. An external DI box may increase your tone options, but the Traveler does a great basic job by itself. As for A-D conversion quality, this again comes over as being excellent. Listening critically to very low-level recordings made on a Tascam DA20 MkII DAT machine using its own converters, and then the Traveler's (routed using Cuemix Plus to the S/PDIF outputs) the difference was pronounced, with the Traveler being apparently quieter, smoother and more detailed, and the Tascam rather 'fuzzy' by comparison. I also briefly fired up a dbx 386, a valve mic preamp which has a rather nice A-D stage, and other than a hint of difference in highfrequency detail, which I couldn't say was better or worse, the two sounded exactly the same. It's possible that very slight gains in detail may be possible using outboard A-D conversion, like that offered by the likes of Apogee's Rosetta, but there's nothing shabby about the sound of the Traveler. Comparing the effect of the Traveler's different sample rates, there is some difference between the 1x, 2x and 4x rates, but not much! I made recordings of an authentic copy of a Flemish harpsichord (an acid test for treble resolution and


MOTU Traveler

clarity) at all the available sample rates, and then played them back to a group of five listeners who were told that something had changed, but not what. Everyone noticed the difference between the 44.1 and 192kHz rates, but only one between 44.1 and 96kHz. The 192kHz recording was described as being more 'stable' and authoritative, but interestingly, detail and smoothness were not mentioned. This corresponds broadly with my own observations, that high sample rates can make a difference, but not always in ways you'd expect.

The Bad News? In the time I had with the Traveler whilst writing this review, I tried to build a balanced picture of its strengths and weaknesses, and I can honestly say that while there's lots to like, I could find almost nothing to criticise! I've mentioned the rather quirky front-panel operation, and I suppose I'd prefer it if the input-level meters had more resolution than is offered by their four LEDs, but as for genuine drawbacks I could only find one. It's this, and it's pretty inconsequential: at the 4x sample rates of 176 and 192kHz, certain restrictions come into play over and above the disabling of all the digital I/O. The headphone output level control still works, but the headphone outputs themselves cease to be available as a separate entity in any software other than the Firewire Cuemix Console. On my review unit, too, the main output level control was bypassed, so that any 4x sample rate audio was played back over the monitor speakers at full volume. No sooner had I mentioned this to MOTU, though, they announced a new firmware revision, v1.05, which promised to fix the problem.

Conclusions Having owned several MOTU Firewire and PCI interfaces over the years, I wasn't expecting the Traveler to offer very much more than I'd grown used to. Nothing could be further from the truth, though — MOTU have equipped the Traveler with such an extensive and well-balanced feature set, implemented for the most part with great elegance, that it is a genuine pleasure to work with and to have around in the studio. The Cuemix Plus monitoring makes mixerless setups a reality, not just a possibility, and has benefits in the studio and on the road for users working in a variety of different areas. The Digital Precision Trims are wonderful, especially together with the four great-sounding mic preamps and bags of available gain. And I very much appreciate the multitude of digital file:///H|/SOS%2005-06/MOTU%20Traveler.htm (8 of 9)9/28/2005 2:33:59 PM

The Cuemix Console software also supports talkback and listenback facilities, allowing individual inputs and output pairs to handle communications between the control room and musicians.

MOTU Traveler

connections on offer, negating the need for any external format converters. It's quite some package, and the portability and flexible powering options are the icing on the cake. Of course, the Firewire audio interface market is somewhat more crowded nowadays than it once was. The Traveler shares a broadly similar feature set with RME's Fireface, the Presonus Firepod, Tascam's FW1804, and perhaps its closest competitor, Metric Halo's Mobile I/O. MOTU appear to have been careful, though, to give the Traveler some distinctive features that not all the others can match — genuine stand-alone operation, the excellent mic preamps and digital gain controls, flexible powering options, the extensive Cuemix Plus monitoring facilities, and of course genuine portability. The Traveler also plays well with additional MOTU Firewire and PCI interfaces should you ever need to up the total number of inputs and outputs available on your setup. Its software is also not particularly biased towards Mac or PC — both are equally well supported, with the exception that Mac users additionally get access to Audiodesk. If all these are features that are important to you, then the Traveler presents itself as the stand-out choice. By any standards, it's a hugely capable, great-sounding and brilliantly conceived product which will not disappoint. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Peavey PV8

In this article:

Verdict

Peavey PV8 £89 pros

Peavey PV8 Mixer Published in SOS June 2005 Print article : Close window

Inexpensive in the UK. Visually distinctive. Good audio performance.

Reviews : Mixer

cons The inability to monitor the two-track input on its own may be restrictive in computer studios. The feel of the EQ-pot detents varied a lot on the review model.

This cheery new unit brings clean eight-channel mixing to the cash-strapped home studio. Paul White

summary As a problem-solver for your toolbox, a simple mixer to go with your recording setup, or a live mixer for duos and discos, the Peavey PV8 has much to commend it.

information £89 including VAT. Peavey Europe +44 (0) 1536 461234. +44 (0)1536 747222 . Click here to email www.peavey-eu.com

Peavey still tend to be pigeonholed by some as a guitar company, but for many years they've also been in the business of building mixers and PA components. They even had a decent range of synths on their books at one time. Based in Meridian, in the south of the US, Peavey still manufacture a significant part of their inventory in America, but like many other MI companies they've been pushed into moving to 'off-shore' manufacturing for Photos: Mike Cameron some of their more cost-effective products. The PV mixer range is designed in the US, but built in China, which means the end user gets a good design at an affordable price. The little PV8 is well suited to some desktop recording applications, as it features four very respectable mic preamp channels with global phantom power, plus two further stereo line-only channels. Cosmetically, it looks a little different to the competition, with its curved, overhanging top plate, brightly coloured knobs, and distinctive screening. Straightaway I felt that it looked attractive and clearly laid out, with all the connectors (other than the two-track input and output phonos) on the top panel for easy access. Although there's nothing radically unusual about this mixer, it is simple, quiet, and has a lot of headroom (+22dB maximum output) for a product of its type. The manual is concise and clear, though it omits to tell us at what frequencies the EQ controls operate, and it claims the mid-band EQ has a shelving characteristic, which I find most unlikely. It does however tell us that the EQ has a ±15dB gain range.

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Peavey PV8

Powered via an included AC adaptor, the Peavey PV8 mixer offers four mono mic/line channels and two stereo line-only channels. Both the mono and stereo channels have three-band EQ, and the stereo channels have both jack and phono inputs with a button to select between them. All four mic channels have insert points on TRS jacks, as well as line inputs on balanced jacks. While you wouldn't expect phase inversion, pad, or low-cut filter on a mixer of this type, there is a global 80Hz low-cut filter that could be very useful, especially in live situations. There's also a global contour switch to add a bit of a 'smile' curve (somewhat like a hi-fi loudness button) to the sound, and while this might not be much use for recording, it can sometimes be useful when using the mixer live, especially to make recorded background music sound more punchy. All channels have input gain trims and a pair of sends, one pre-fade for monitoring and one post-fade for effects — unlike some compact mixers, the sends have corresponding output level controls in the master section. Rotary pan/balance and level controls complete the complement of channel knobs, though there is also an individual overload LED for each channel. It's worth mentioning that while the knobs may look like chewy sweets, they're actually made from hard plastic and have a clearly defined pointer line on top. There's no EQ bypass switch, but all the EQ controls have a centre detent. However, on the review sample — apparently a pre-production model — the detents varied in feel so much that some were very obvious while others felt almost non-existent. Hopefully this will be cleared up for the main production run, and it was the only mechanical problem I encountered. The master section is simple enough, with global phantom-power switching, master level controls for both the effects return and monitor output (a headphone jack is located above the phones control), and a single slider to control the main stereo output level. There are physical outputs for the main stereo out, the control-room output, and the two aux sends — all on unbalanced, groundcompensated jacks, which help to avoid ground-loop hum — plus there are the stereo aux return jacks. Six-section LED meters monitor the stereo output level. The two aux send master knobs are colour matched to the channel send controls, and there are switches to send the tape to the control room output and/ or to the mix output. Unfortunately, there seems to be no way to select the twotrack input as the only monitor source, which means that you can't do the old computer studio trick where you monitor the soundcard output via the two-track return while recording into the computer from the main stereo outs.

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Peavey PV8

This little mixer checks out well in all respects, with low noise, generous headroom, and a clear layout. The manual quotes an EIN figure of -129dBu with a 150(omega) input load and a frequency response of 14Hz-25kHz (+0dB/-1dB). It's also worth mentioning that the mid-band EQ sounds as though it's set at around 500Hz rather than the usual 2.5kHz, which I feel is a worthwhile departure from tradition. A lot of the unpleasant boxiness in sounds happens at around 500Hz, so it's useful to have a tool that allows you to cut that. I find that recessing the phantom-power button is a bit of a mixed blessing, as it's almost impossible to turn it on or off unless you're armed with a matchstick or a ballpoint pen. It certainly means that you won't switch it on by accident, but come on guys — it's only phantom power, not the president's big red button! I like this little mixer a lot and it has some nice practical touches, such as the use of a fader for the master level control and a sensible choice of EQ mid-band frequency. Whether it is the one for you or not depends largely on whether you need the ability to select the two-track input as the sole control-room monitor source. As a cheap source of decent mic preamps (which you can take directly from the insert sends) or as a means of combining a small number of mics, linelevel signals, and DI-box outputs, the Peavey PV8 offers a great deal of performance, not to mention a certain amount of visual flair, for the price. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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In this article:

Studio Tests

Presonus Blue Tube DP £165 pros Good basic sound quality. Nice styling. Inexpensive.

cons Tube sound is less subtle than from true high-voltage tube circuitry.

Presonus Blue Tube DP Dual Mic/Line Preamp Published in SOS June 2005 Print article : Close window

Reviews : Preamp

Valve warmth or solid-state transparency? You decide with this flexible new hybrid preamp.

summary For the desktop studio user needing a decent pair of mic preamps that also double as instrument DI boxes, the Blue Tube DP fits the bill nicely without costing the Earth. You have to use the Tube Drive sparingly on vocals, but it is a useful effect nonetheless.

information £165 including VAT. Hand In Hand +44 (0) 1579 326155. +44 (0)1579 326157. Click here to email www.handinhand.uk.net www.presonus.com

Paul White

The Presonus Blue Tube DP supersedes their existing Blue Tube and features what Presonus call Dual Path technology. Essentially this teams a solid-state preamplifier stage with a Photos: Mark Ewing tube stage, where each of the channels functions as a microphone or instrument preamp. A 12AX7 dual triode is shared between the two channels so there is one stage of tube gain per channel. This tube is also run at well under the normal death-dealing voltages used in traditional tube products, so there's no safety risk. However, the authenticity of the added 'warmth' can vary in low-voltage designs, so you really need to judge them subjectively rather than assuming that they will sound exactly right simply because they have valves inside. Presonus are by no means the first company to take this hybrid design approach, as solid-state input sections tend to be cheaper to build and less noisy than their all-tube counterparts, but they have managed to bring in the Blue Tube DP at a surprisingly attractive UK price and claim to offer the user the ability to create 'a fat warm tube tone or invisible solid-state transparency'. Predictably, the degree of warmth is determined by how hard you drive the tube, as warmth in this context is really just another way of describing subtle harmonic distortion. The visual design has been brought up to date with brushed aluminium, blue switch LEDs, and red LED backlighting to the tube, though the 1U half-rack shape and external transformer powering have been retained. As the Blue Tube DP is simply a preamp without EQ or dynamics, the

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manufacturers have been able to provide all the essential features that you'd expect in a mic pre, specifically a -20dB pad, phase-reverse button, 80Hz low-cut filter, and individually switchable 48V phantom power. Each channel has two rotary gain controls, one of which works more or less conventionally to adjust gain while the other, labelled Tube Drive, sets the amount of valve gain. The Tube Drive control has an integral push switch that bypasses the tube stage completely — as you'd expect, this setting produces the least distortion and the best noise figures. The Tube Drive control increases the input to the tube stage, but reduces its output to compensate, so there's little overall level change while this is being adjusted. For those who like to check the figures, the combined THD + Noise (unweighted) in solid-state mode is less than 0.005 percent, but can be anything from 0.01 to 30 percent when the tube stage is in circuit, depending on how hard you drive it. Metering is courtesy of two small, circular moving-coil meters, and there are also fast-acting overload LEDs located between the channel gain controls. These come on at +22dB, so there's a decent amount of headroom available. All the connections are on the rear panel, including the PSU connection point. A hook on the rear panel provides somewhere to anchor the cable so that it won't fall out in the middle of the session. For the inputs, the designers have opted for Neutrik combi jack/XLR connectors, as these save on space and can accept both balanced XLR and quarter-inch, unbalanced jacks. The jack inputs have a high input impedance (1M(omega)) designed to accept instruments with magnetic pickups, but they can also accommodate line levels. By contrast, the line-level outputs are on separate balanced XLR and unbalanced jack connectors which are active simultaneously, providing a practical way of splitting the signal for zerolatency monitoring or other applications.

Studio Tests My first check was with a large-diaphragm capacitor mic using the Blue Tube DP with its tube stage bypassed. Although the maximum gain is limited to 57dB, that's plenty for capacitor mics, though it may be a hint on the low side for some dynamic models or for recording quiet sources. By contrast, most mixers offer 60dB of mic preamp gain with a further 10dB or so of gain available by pushing the faders up past their 0dB positions. To my ears, the sound is quiet and clean and, to be honest, fairly indistinguishable from any number of well-designed, midprice mic amps I've tried in the past. With the tube stage switched on, there seemed to be a fine line between not hearing much difference at all and hearing obvious distortion, so for normal work I'd use this below its midway setting and only venture into heavier distortion for special effects. There's quite a lot of distortion available at more extreme file:///H|/SOS%2005-06/Presonus%20Blue%20Tube%20DP.htm (2 of 3)9/28/2005 2:34:13 PM


settings, no doubt to give flexibility to those using the unit as an instrument DI. The type of distortion it produces should be useful for taking the edge off hyper-clean rhythm or bass guitar and it could also suit many keyboard sounds, including tonewheel organ. It's less successful as an electric guitar treatment, because as soon as you get into the obvious distortion range, you need to follow up the preamp with a speaker simulator to attenuate the 'fizzy' high-end harmonics that would normally be removed by a typical guitar speaker. Having said that, used with care it can add a bit of tube character to an already 'produced' guitar sound from a Line 6 Pod or similar device, provided that you don't add so much drive that the sound becomes raggy. In designing the Blue Tube DP, Presonus have managed to produce a costeffective, sweet-sounding mic/instrument preamp that has the added benefit of tube coloration only when you need it — and even then, you're in full control over what you add. I don't think it sounds as subtle as a well-designed, high-voltage tube circuit, but used sparingly it adds to your creative palette and also allows you to add more warmth to DI'd electric instruments. As a desktop unit, the Blue Tube DP looks great, it's very easy to use, and it has no obvious vices. Even the gain control operates progressively without everything being bunched up at one end, which is a common failing with other low-cost devices. I would have liked just a little more available gain, but in the majority of studio situations there is more than enough available. Given the low UK price and very decent level of performance, you can't really go wrong. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Pro Tools M-Powered

In this article:

Lok Down Bundled Software Outline I/O Silver Lining In The Thick Of It MIDI Recording Gripes & Grumbles A Force For Good?

Digidesign Pro Tools MPowered £239 pros Increases the range of hardware options available to Pro Tools users. Provides users of M Audio hardware with the option to use one of the best recording packages around. Fully Session-compatible with other Pro Tools systems.

cons No crossgrade deals for existing Pro Tools LE users. Doesn't support Direct Monitoring on M Audio hardware. Not compatible with the DV Toolkit, and doesn't include the new features in Pro Tools 6.9. There seems to be a bug in the I/O Setup window that fails to create mono Input Paths. There's still no PCMCIA interface that can be used with Pro Tools.

summary If you own a compatible M Audio interface, you can now add the industry's standard DAW to the equation when choosing your recording software; and if you're an existing Pro Tools user, you can now choose from a much wider range of recording

Pro Tools M-Powered Recording Software [Win/Mac OSX] Published in SOS June 2005 Print article : Close window

Reviews : Software

For the first time ever, Digidesign's Pro Tools recording software is available as a stand-alone product, which can be used in conjunction with any of five audio interfaces from M Audio. Sam Inglis

From humble beginnings as Midiman, makers of handy gadgets such as format converters and MIDI interfaces, M Audio's growth is the stuff of business legend. By the time the company was bought by Avid Technology last year, sales of M Audio's core products — affordable soundcards, USB and Firewire interfaces, and controller keyboards — were bringing in tens of millions of dollars. The motives behind Avid's acquisition seemed pretty clear: Digidesign, also owned by Avid, had reached a dominant position in the pro and semi-pro markets, whilst M Audio were one of the leading players in the home-studio computing arena, so the takeover gave Avid a substantial slice of the latter market to complement Digidesign's dominance at the higher end. There was, however, a certain amount of overlap between the new siblings' product ranges. In recent years, Digidesign had devoted increasing attention to their Pro Tools LE systems, which package a slightly cut-down version of their Pro Tools recording software with small audio interfaces such as the M Box and 002, and some of these have very similar counterparts in M Audio's range. There was also something of a clash of corporate philosophy, in terms of the two companies' approaches to open standards. M Audio have always tried to support all of the common driver standards for both Mac OS and Windows, ensuring that

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Pro Tools M-Powered

hardware — if you're prepared to pay for another copy of Pro Tools to go with it.

information £239 including VAT. M Audio +44 (0)1923 204010. +44 (0)1923 204039. Click here to email www.maudio.co.uk

Test Spec Pro Tools M-Powered version 6.8. M Audio Firewire driver version 5.10.0.5034. Inta Audio Centrino laptop with 2.0GHz Pentium-M processor and 2GB RAM, running Windows XP Home Edition SP2.

their interfaces would work with as many different recording packages as possible. Digidesign, on the other hand, have never made a version of Pro Tools that would support other companies' hardware via protocols such as ASIO or WDM. Pro Tools has always used the proprietary Digidesign Audio Engine instead, and the same is true of Digidesign hardware. (In recent years, Digi have written ASIO drivers allowing their hardware interfaces to be used by third-party programs, but they rarely exploit all of the features of a particular interface.) The takeover thus led to plenty of speculation about how the two companies would work together. Would Digidesign make an ASIO-compatible version of Pro Tools? Would M Audio continue to bundle other companies' software with their interfaces? Would we see the end of open driver support for M Audio hardware? Well, the first fruits of the marriage were announced at the Frankfurt Musikmesse, in the shape of a version of Pro Tools designed to run on M Audio hardware. Initially, five M Audio interfaces are supported: the Audiophile 2496 and 192 PCI soundcards, the Ozonic combined controller keyboard and Firewire interface, and the multi-channel Firewire 410 and Firewire 1814 interfaces. With the Audiomedia III and Digi 001 interfaces both long discontinued, this means you can now run a 'lite' version of Pro Tools with a PCI interface for the first time in ages, but it's interesting that none of M Audio's USB audio interfaces is supported. Perhaps Digi still have a warehouse full of M Boxes somewhere... Those who were hoping for an 'open' version of Pro Tools will also be disappointed. PT M-Powered uses the existing M Audio device drivers, provided you have the latest version, but is unable to 'see' any audio devices other than the five listed above.

Lok Down Previous versions of Pro Tools have, in effect, used Digidesign's hardware as a dongle, with PACE's iLok system employed as an adjunct for authorising thirdparty plug-ins. In Pro Tools M-Powered, the iLok USB dongle is used for the program too. It's hard to love any dongle-based copy-protection scheme, but the iLok system is at least tried and tested, and it makes it possible to gather all your plug-in authorisations in one place; it also allows you to install a single copy of Pro Tools M-Powered on as many different machines as you like, as long as you only intend to use one at a time. However, the sheer physical size of the thing can be a nuisance — the two USB ports on my laptop are positioned in such a way that it's impossible to insert a USB pen drive at the same time as the iLok. As well as the dongle itself, which comes in a fetching shade of red, the box includes Windows and Mac installation CDs, a brief Getting Started manual and an even briefer Basics Guide. These are reproduced in PDF form on the discs, along with the full manual and various other bits of documentation. I've always found Digidesign's product manuals, especially the Pro Tools Reference Guide, to be among the best around, but it would be handy to have this in paper form file:///H|/SOS%2005-06/Pro%20Tools%20M-Powered.htm (2 of 9)9/28/2005 2:34:26 PM

Pro Tools M-Powered

too. It would also be nice if the Reference Guide had been updated to cover Pro Tools M-Powered, which has been given the version number 6.8 — the Reference Guide is for 'version 6.7 for HD and LE systems', and doesn't include any info specific to the MPowered version. The LE and TDM versions of Pro Tools The Mix window in Pro Tools provides a have just been updated to 6.9, but the standard 'virtual mixer', although nearly all of new features in that version don't the same functions can be accessed from the Edit window. appear in Pro Tools M-Powered. The DV Toolkit, which allows Pro Tools LE users to import project files from Avid's Xpress DV video-editing package, is not supported here. Pro Tools M-Powered does, however, include the major features that were added to the LE version in the 6.4 and 6.7 updates, including the Beat Detective LE drum-fixing tool, and for those who might want to bring their projects into a larger studio at a later date, it is fully Session-compatible with LE and TDM systems. The idea that you can take your hard drive into almost any studio in the world and have your recordings ready to mix or overdub will be a major selling point. I tested Pro Tools M-Powered on a Centrino laptop running Windows XP, with an M Audio Firewire 1814 interface. Installation is fairly straightforward, and although it does demand a couple of tweaks to System Properties settings, the manual does a fairly good job of holding your hand through the process. Getting your M Audio interface up and running with Pro Tools requires that you install the WDM driver, which is equally straightforward. Plug the iLok key in, navigate the Found New Hardware wizard yet again, and you're ready to go.

Bundled Software These days, it seems that no selfrespecting MIDI + Audio recording package can be competitive without the sweetener of free effects and processors, and Pro Tools MPowered is no exception. The standard Digirack suite of basic plugins has now been swelled to 35 items, to which those with an Internet connection can also add the new EQ III (see this month's Pro Tools Notes). It's probably best not to get too excited about such thrillers as DC Offset Removal and Normalize, but most of the Digirack tools will come in handy sooner or later. The compressors and equalisers are usable, if not exactly characterful, and D-Verb is more versatile than most bundled reverbs. All the dynamics plug-ins can be keyed from any of the internal busses in the Pro Tools


Pro Tools M-Powered

mixer — an invaluable feature which is sorely lacking in most MIDI + Audio sequencers. What's more, all the real-time plug-ins are also available in off-line Audiosuite versions, and if you run out of processing grunt, it's always been easy to copy your settings from the one to the other and apply them permanently. Perhaps Digidesign should give this function a snowflake logo and a catchy name — something like 'Freeze' should do the trick. As with all current versions of Pro Tools, you also get a selection of plug-ins from the Bomb Factory line, which was bought up by Digidesign a whille back. Most of these fulfil useful but unglamorous functions such as clip removal and metering; the highlight for most users will be BF76, a recreation of the classic Universal Audio 1176 compressor. I think I prefer Universal Audio's own software recreation of this unit, but since you'll need to fork out for a UAD1 card or a TDM system in order to use that, BF76 is not to be sneezed at. Also included on the CD is Ableton's Live Digidesign Edition, but the other 'lite' programs bundled with LE versions of Pro Tools are absent here. Propellerhead's Reason Adapted is bundled with most M Audio hardware in any case, but it's a shame that you don't get IK Multimedia's Sampletank SE, Amplitube LE or TRacks EQ.

Outline The basic features of Pro Tools have been covered in SOS many times before, so I won't go into detail here except on points that are specific to the M-Powered version. As ever, almost all recording and editing functions are carried out in one of two windows: the Edit window shows each track arranged along a horizontal timeline, while the Mix window displays the same tracks as channels in a virtual mixer, with faders, pan controls, aux sends, inserts and so on. Seasoned Pro Tools users will know that you can actually accomplish almost everything within the Edit window, which makes a welcome change from those applications that open a new window if you so much as sneeze at them. Even MIDI note editing is carried out directly within MIDI tracks in the Edit window, with no need to open up a separate piano-roll screen. Audio can be recorded at 16- or 24-bit and at sample rates up to 96kHz. Unlike most DAWs, Pro Tools doesn't include any template setups for typical tasks like multitrack recording or stereo editing, so the first time you start it up and open a new Session, the Edit and Mix windows will be blank until you create tracks to fill them. You can have up to 128 mono or stereo audio tracks, but like current LE versions, PTMP provides only 32 'voices' — playing back a mono track requires a single voice, whilst a stereo track will use two — so 96 of these must be considered 'virtual' tracks. Aux inputs and MIDI tracks also appear in both the Edit and Mix windows, and a Session can include up to 256 of the latter. As well as audio and Aux tracks, there are also 16 mono audio busses, which can be paired to make up to eight stereo busses as appropriate. The output of any track can be routed either to an output on your audio interface or to a buss, and each track also features five Aux sends which can likewise be routed to any output or buss. Aux input tracks can be fed from any audio input or any buss, and


Pro Tools M-Powered

if you want to have three aux sends feeding one half of a stereo Aux track and two feeding the other, before sending from that Aux track to the key input of a plug-in on another track and routing all of the resulting mess to another buss feeding another Aux track, no problem. To my mind, the flexibility of this bussing system is one of the strongest features of Pro Tools, and this alone would be enough to make me choose it over most other MIDI + Audio applications for mixing.

I/O Silver Lining Once you've created some tracks, they should be available for recording straight away, with input and output routings automatically assigned by Pro Tools and visible in its mixer. As far as the M-Powered version is concerned, however, there seems to be a tiny fly in the ointment. When I booted it for the first time, I was able to select inputs only on stereo tracks — whenever I created a mono track, all the possible inputs were greyed out. After a bit of head-scratching I traced the problem to the I/O Setup window. Channel inputs and outputs within the Mix window are selected from a list of Input Paths and Sub-paths, and the I/O Setup window is where you create these Paths and assign them to physical inputs and outputs on your M Audio hardware. For some reason, the Pro Tools mixer won't allow you to select one side of a stereo Path as an input source or output destination for a mono channel; instead, you need to manually create a mono Input Path or Input Sub-path within the I/O Setup window. This means that the natural way to set things up is to use a stereo Input Path for every pair of inputs, with mono Sub-paths for each half of each stereo Path. This has always been the default in my TDM system, and according to the Reference Guide, the same should be true of Pro Tools MPowered — there's even a dedicated Default button which is supposed to return you to that state. In my system, though, the default setting most often created a stereo Path for each input and output, but no Sub-paths (I say 'most often' because sometimes it did work properly on the inputs, though never with the busses). The upshot of file:///H|/SOS%2005-06/Pro%20Tools%20M-Powered.htm (5 of 9)9/28/2005 2:34:26 PM

The default I/O Setup is supposed to create mono Sub-paths for every stereo Path, as shown left, but more often than not it fails to do so, as in the screen right.

Pro Tools M-Powered

this is that no mono inputs are available for selection within the Pro Tools mixer; and likewise, no mono destinations are available on Aux sends. This problem is easily resolved by visiting the I/O Setup page and manually creating some mono Sub-paths, but if it's reproduced on other machines, I think Digidesign's tech support people can expect a few calls from frustrated first-time users. Routing and mixing are handled entirely in Pro Tools M-Powered rather than the M Audio Control Panel — fader settings in the Control Panel mixer have no effect in PTMP— but you do need to visit the Control Panel in order to carry out some tasks, such as switching the format of an interface's digital I/O. Annoyingly, anything that does have to be done in the Control Panel requires that you quit out of Pro Tools in order to do it, though the same is true of any other audio application running with M Audio hardware. What will be more annoying for many users is that the Direct Monitoring options in interfaces such as the Firewire 1814 are not supported in Pro Tools MPowered. In other applications such as Cubase SX, these allow you to route input signals directly to the interface's outputs with negligible latency, so that you can hear what you're playing without the aid of unwanted slapback delay. Pro Tools LE provides a similar function with the 002 interface, but unless you have a hardware mixer or similar that you can use to set up an all-analogue monitor path, you'll will be stuck with the Elvis effect when recording in Pro Tools MPowered. At the default 512-sample buffer size, the latency will certainly be noticeable, and the smallest buffer size available is a rather conservative 128 samples, which equates to a latency of around 6ms at 44.1kHz — the 64-sample setting available to other applications with M Audio hardware is absent from Pro Tools.

In The Thick Of It Pro Tools is a versatile piece of software, but its central application is still recording, editing and mixing multitrack audio, and I took the opportunity to test the M-Powered version in a real session. For a while now, I've had my eye on an attic room at my dad's house, which I had always thought would make a beautiful acoustic environment for tracking a band live. (The difficulty had always been finding musicians foolish enough to let me do this, but I solved this problem by starting a band of my own.) As luck would have it, the review copy of Pro Tools M-Powered turned up the day before our session, and — throwing caution to the winds — I decided to press it into action. We encountered more than our fair share of problems on the day, the low point being a power glitch which took out an eight-channel analogue-to-digital converter and a ring main. Forced to rely only on the Firewire 1814's unbalanced analogue inputs, we then ran into horrendous ground-loop interference, and the only way we could record anything at all was to disconnect the laptop from the mains and run it on battery power. file:///H|/SOS%2005-06/Pro%20Tools%20M-Powered.htm (6 of 9)9/28/2005 2:34:26 PM

Pro Tools M-Powered

Despite all of these setbacks, Pro Tools M-Powered held up pretty well under pressure. The input routing anomaly described above caused a certain amount of cursing, and I also became frustrated with how long it took to open even a fairly simple Session with few tracks and no recorded audio. When it really mattered, however, Pro Tools M-Powered delivered the goods, recording eight tracks of audio and a MIDI keyboard without any fuss. Even when my laptop was on the verge of giving out, PTMP was faithfully capturing the 'low battery' beep, and although the interface became a little sluggish, every single take was recorded perfectly. Pro Tools M-Powered has not crashed once since I've installed it, and has behaved as predictably and reliably as you'd hope. Internet folklore has it that the Windows version is inferior to the Mac original, but I experienced nothing to suggest that this is the case.

MIDI Recording Whereas programs such as Logic began life as MIDI sequencers and added audio recording at a later date, Pro Tools has gone the other way. It was originally an audio-only package, but its MIDI capabilities have come on in leaps and bounds in recent years. MIDI recording in Pro Tools M-Powered is a piece of cake. When you create a MIDI track, it defaults to accepting input from any connected device — simply arm the track and play a few notes, and you should see the incoming MIDI data light up the track's meter. Pro Tools M-Powered detected both the MIDI ports on the Firewire 1814 interface and a connected Edirol PCR1 USB keyboard without any intervention at all on my part; and if you need to create a more complex MIDI environment, the MIDI Studio Setup window is only a mouse-click away. Under Mac OS X, the same functions are accomplished using the standard Audio MIDI Setup utility. To use virtual instruments in Pro Tools, you simply create an audio track or Aux track and add the instrument of your choice as an insert plug-in. That plug-in should then appear as a possible destination in the Output pop-up on MIDI tracks. I say 'should' because I experienced a few anomalies in this regard. Arturia's CS80V was happy to load in to an audio or Aux track, and could be 'played' via its on-screen keyboard, but didn't show up as a destination in MIDI tracks, while IK's Sampletank LE unaccountably refused to accept input on MIDI channel 1. Whether these problems were down to Pro Tools itself or poor implementation of the RTAS standard on behalf of the plug-in developers, I don't know.

Gripes & Grumbles I've been an unapologetic Pro Tools fan for a long time now. Where other software fills the screen with flashy graphics and cryptic icons, Pro Tools presents only the information you need to know, in a way that I find perfectly intuitive. Its two-window design is supremely elegant, its editing tools are powerful and quick to use, its mixer is one of the most flexible around, and you have to kick it pretty damned hard to get it to fall over. I'll take those basic qualities over fancy bundled synths or loop-mangling tools any day of the week, and now that Pro Tools M-Powered has made them available on a much broader file:///H|/SOS%2005-06/Pro%20Tools%20M-Powered.htm (7 of 9)9/28/2005 2:34:26 PM

Pro Tools M-Powered

range of hardware, what's not to like? Well, it is slightly disappointing that the integration between Pro Tools and M Audio hardware is, in some ways, less complete than you get with an ASIObased application such as Cubase SX. I doubt that too many people will be bothered by PTMP's inability to record at 192kHz, but its lack of support for Direct Monitoring is unfortunate, given that Pro Tools is perfectly capable of making use of the same feature within Digi's 002 and 002 Rack hardware. It might be that the port to M Audio Pro Tools M-Powered displays its 'LE' hardware has been a little rushed — heritage... some of the windows within Pro Tools M-Powered still say 'Pro Tools LE' at the top — but at least the features that are included work, and work well. Whilst Pro Tools M-Powered does open up welcome new hardware options for Pro Tools devotees, though, I can think of one group of users who will remain frustrated. M Audio interfaces such as the Firewire 410 and Audiophile PCI cards fill some of the gaps in Digidesign's hardware range, but I can't be the only Pro Tools user who dreams of editing and mixing on my laptop using a slimline PCMCIA audio interface, rather than having to cart around an M Box or a Firewire interface and all its associated cables and power supplies. The likes of Echo's Indigo range and Digigram's VXpocket cards make this possible for those using ASIO-based software, and it's about time Digidesign or M Audio produced something comparable. Existing Pro Tools users who want to move to M Audio hardware might also feel justifiably aggrieved at having to pay full price for the M-Powered version of a program they already own. The cheapest of Digidesign's LE systems is the M Box, which still retails at over £300 in the UK, when other USB interfaces with almost identical hardware features, including M Audio's own Mobile Pre USB, are available for little over £100. M Box owners have, in other words, already paid £200-plus for their copy of Pro Tools LE — so is it really fair to ask them to spend the same amount again, just in order to use the same program on a Firewire 410 or Ozonic? After all, if you're a Cubase user and you decide to upgrade your audio hardware, you don't need to hand over any extra cash to Steinberg. Personally, I would hope that existing users' loyalty to Pro Tools would earn them at least a substantial discount, but Digidesign evidently don't agree.

A Force For Good? Ten years ago, the idea that one might actually want to pay money for a copy of file:///H|/SOS%2005-06/Pro%20Tools%20M-Powered.htm (8 of 9)9/28/2005 2:34:26 PM

Pro Tools M-Powered

Digidesign's editing and recording software would have seemed crazy to a lot of people. Sales of TDM systems were driven by the hardware, with its unique lowlatency, DSP-assisted mixing capabilities, and most of those using Digidesign hardware in a music-recording context left the software in the box and ran Logic or Digital Performer as a front end. The fact that Digidesign are now able to charge £239 just for a copy of Pro Tools shows how much things have changed since then; and the fact that all those other programs have spent the intervening decade growing steadily more similar to Pro Tools tells you even more. It's not the most visually exciting piece of software, and it still lacks some of the advanced MIDI features available in rival products, but in terms of simplicity, intuitiveness and flexibility — not to mention stability — it's hard to beat. Pro Tools M-Powered is simply the latest iteration of a mature, stable and comfortable recording program, and the fact that it exists is a powerful reason to consider M Audio hardware over the alternatives. The feature set is almost exactly the same as in the LE version, and at £239, it's in the same price range as Steinberg's Cubase SL, Apple's Logic Express and Cakewalk's Sonar Studio 4. Whether you're already a Digidesign diehard or someone who's in the market for recording software for the first time, you simply can't afford to ignore Pro Tools M-Powered. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Real Traps Mondo Trap

In this article:

Physical Characteristics Real-world Testing

Real Traps Mondo Trap Acoustic Panel Published in SOS June 2005

Real Traps Mondo Trap £399

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Reviews : Accessory

pros Serious bass trapping. Smart cosmetics. Comparatively compact given the level of performance.

cons None, as long as the results justify the cost in your particular studio setup.

summary

Real Traps' biggest bass trap offers heavy-duty lowfrequency absorption for the serious home studio. Mike Senior

A very effective low-bass absorber panel which can transform a problematic monitoring environment quickly and easily. However, you'll want to try out a pair in your room in order to decide whether they justify the substantial cost.

I recently moved house, and found upon reassembling my studio setup that the bass levels I was hearing from my Blue Sky Pro Desk system were all over the place. In my previous flat the bass response had caused no real problems, partly because of the design of the particular room, and partly because of the way my wife and I had packed every conceivable space with furniture, CDs, books, and assorted clutter! In the new larger room, however, the levels of bass information varied wildly with even small changes in listening position, and Mondo Trap, £399; there was a severe 80Hz dip in the response at any sensible optional floor stand, £129.25. Prices include VAT. monitoring location. Sonic Distribution +44 (0) 1525 840400. +44 (0)1582 843901. Click here to email www.sonicdistribution.com www.realtraps.com

Having seen Martin Walker's review of the Real Traps bass traps back in SOS September 2004, I headed over to the Real Traps site and had a look through some of proprietor Ethan The Mondo Trap Winer's writing about the principles behind his acousticfitted with its treatment products. My interest piqued, I was on the point of optional floor arranging a loan of some Mini Traps from UK distributor Sonic stand. Distribution when news came through that Real Traps had just released their new Mondo Trap, a larger version of the Mini Trap with much more effective low-bass absorption, so I decided to try this new model out to see how it would deal with my real-world problem.

Physical Characteristics The Mondo Trap is a 10cm-thick panel of high-density fibreglass covered in a felt-

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like fabric and mounted in a smart painted-metal frame. The stiffness of the metal frame allows the panel to be moved around with comparative ease, notwithstanding the substantial 60 x 145cm dimensions. The metal frame also makes it possible to mount the trap in your room using strong picture wire and wall hooks, much as Martin did when reviewing the Real Traps, but there is also an optional metal floor stand available if wall or ceiling mounting is inappropriate. This stand is attached to the trap by hand using knurled bolts, and it allows for vertical orientation at a range of different heights. Traps and stands are available in a variety of colours, including white, black, and 'wheat', the latter being pretty much like magnolia.

Real-world Testing After unpacking the Mondo Traps from their individual shipping cartons, it was the work of only a few minutes to attach the optional stands. These made it a piece of cake to shuffle the panels around the room while listening to the effects. As I'd expected from Ethan Winer's discussions and Martin's experiences, placement close to walls and room corners offered the most dramatic bass absorption, so after some more listening tests, I detached the floor stands so that I could place the traps into the corners behind my studio desk. With the traps in position, the change in the overall system sound was dramatic. Finally I could usefully adjust the positioning and output level of my subwoofer, and the bass response remained much more uniform as I moved around the studio setup. Crucially, though, the ferocious 80Hz dip in the frequency response was dramatically reduced, much to my delight, and I have now come to be able to rely on the overall sound even more than in my previous setup. Real Traps claim that the Mondo Trap has twice the absorption of the Mini Trap below 100Hz, and I can well believe it given these kinds of results from just a pair of traps. The performance of these traps is certainly impressive, as it should be given that they cost £800 a pair in the UK. Whether they justify their price in your setup will depend on the severity of any acoustic problems, the quality of your monitors, and the kind of work you're doing. If you've already spent more than a grand on monitors, but you have bass problems which are still making it impossible to mix properly, then I'd guess you'll get your money's worth. However, given the UK loan scheme operated by Sonic Distribution, you don't have to guess — try them out in your own room, and you should know within half an hour whether they'll be worth the outlay. Published in SOS June 2005






Sample Libraries: On Test

In this article:

Chemical Synths ***** Evolved Analog **** Cinematic Hip-hop **** Discography *****

Star Production Clichés

Sample Libraries: On Test Sample Shop Published in SOS June 2005 Print article : Close window

Reviews : Sound/Song Library

***** Bell Tree Glissandos **** Auto-Tune Turned Up To 11 *** R&B Triangles

Chemical Synths ***** REFILL+REX 2

** Telephone Vocals * Orchestral Hits

This Reason Refill is aimed squarely at producers of dance music. In fact, the samples, which run to approximately 1GB of data, are split across two Refills; the first contains a large number of Subtractor, Malström, NN19, and NNXT patches, while the second contains a collection of REX 2 files. Usefully, the 400 or so REX 2 files are provided within a separate folder and so can be used in other music applications that can read the format. A number of Scream and RV7000 patches are also contained within the first Refill. As a whole, the collection concentrates upon bass and lead sounds, and even the REX files focus on synth arpeggios and riffs. There are no drum loops (which makes a nice change!). The 50 Subtractor patches focus upon bass sounds and a range of 'soft' and 'hard' leads. The latter dominate, and it is clear that creator Jonathan Heslop has put his hours in getting to grips with Reason's various instruments — the sounds make good use of Subtractor's programming options. As with the patches for the other instruments, don't expect anything too timid here — in the main the sounds are aggressive and create a dark mood. The 200 Malström patches pick up where the Subtractor ones leave off. As well as groups of both soft and hard lead and bass sounds, there are also a collection of special effects. There is some really good stuff amongst this lot, and patch names like Agrogate, Devilinside, and (at the more subtle end) Junotrance give a clear idea of the content. Some 150 NN19 and 200 NNXT sampler patches are provided. After the Malström collection, the NN19 sounds rein things in a little, although there are some very usable 'bread and butter' sounds that would suit a range of dance and electronica styles. However, things start to get out of hand again with the NNXT patches. More aggressive basses and wild lead sounds abound — Sithsweep, Destroyer, and Widemutator suggest the territory covered. These sorts of sounds

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could be used subtly (if that's not a contradiction in itself) to add an aggressive edge to a mainstream dance, R&B, or hip-hop track, while they would be at their best when used with a little less restraint for full-on, in your face, rave music. Some of the sounds are, however, very big, so a little would go a long way in a mix. While the instrument patches dominate, the REX loops should not be overlooked. As mentioned above, these include 200 arpeggio loops and a further 200 riffbased loops. Along with the Malström patches, these are a highlight of the Refill. I could easily imagine building a track around a couple of these combined with a suitable bass sound taken from elsewhere in the collection — just add in your choice of drum loops and a pad or two from elsewhere. In summary, the sounds within this collection have 'attitude' written all the way through them. If you are a Reason user and have a fondness for the more aggressive end of dance, then Chemical Synths is well worth adding to your Refill collection. John Walden Reason Refill and REX 2 DVD-ROM, £59.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.zero-g.co.uk

Evolved Analog **** EXS24 This EXS24 library from AMG is dedicated entirely to analogue synth sounds recorded without processing. Although this makes them seem somewhat dry on first listening, it does allow the user the flexibility of adding effects later on — it's always easier to add effects than to take them away. The library is based on waveforms recorded from classic analogue instruments, such as the ARP 2600 and the Oberheim Matrix 12, but much use has also been made of the EXS24 filters, allowing the sounds to respond properly to playing dynamics. The samples are divided up into Bass, Hit & Effects, Mix, Pads, Rez, Synth, and Vox, where synths that were originally monophonic or which had no velocity sensitivity can now be played with the benefits of both. All the examples are multisampled to a very high standard and tend to favour dance-music production insomuch as there are more aggressive sounds, hits, and stabs than might be expected in a general analogue library. The fact that these are presented dry is a file:///H|/SOS%2005-06/Sample%20Libraries%20%20On%20Test.htm (2 of 6)9/28/2005 2:35:16 PM


brave move, as you really need to add delay or reverb to bring them to life, so what may not initially sound all that impressive will probably sit more happily in a mix than something that's been overproduced. I don't think that AMG have broken any new ground here, but then when you're looking back in time to sample older instruments, there probably isn't that much that hasn't been covered reasonably well anyway. That doesn't detract from that fact that this is a useful and varied collection, and even though existing users of Apple's own Extreme Analog library are unlikely to find their horizons significantly widened, there's a lot of good stuff within Evolved Analog that is instantly usable. Although there are pads and basses amongst the more angsty sounds, I think my earlier comments about this collection being more suitable for dance-music production are valid, yet because the EXS24 filters and level envelopes are used as part of the sound creation process, the sounds can be edited quite extensively. When you think about it, the EXS24 is really a digital emulation of an analogue synth, using samples as oscillators, so having so many classic raw waveforms available (as you have here) gives the more adventurous synthesist plenty of scope for creating new sounds or re-shaping existing ones. Evolved Analog isn't unduly expensive, so even if you already have other similar titles, it provides a fast way to inject some variety into your songs without resorting to editing. Based on the quality of the content and the modest cost, I think four stars is in order. Paul White EXS24 CD-ROM, £60 including VAT. AMG +44 (0)1252 717333. +44 (0)1252 695680. Click here to email www.amguk.co.uk www.samples4.com

Cinematic Hip-hop **** ACID This release from Sweden-based Power FX is, like other titles in their product line, competitively priced. It contains 695MB of material spread across about 500 Acidised WAV loops and one-shot samples. While the bulk of the library comprises 34 construction kits, a further folder titled Elements & Extras makes up the final third of the collection, and this contains additional drum loops, bass lines, keyboard/synth loops, and various special effects and/or hits. As suggested by the title, most of the original tempos are in the 110-80bpm range. Each of the construction kits contains the usual collection of drum loops alongside a small number of bass, keyboard, and other loops — these can be file:///H|/SOS%2005-06/Sample%20Libraries%20%20On%20Test.htm (3 of 6)9/28/2005 2:35:16 PM


combined to form a complete musical bed. The keyboard loops include pads, chord-based progressions, and a selection of melody lines. There is also a smattering of string, percussion, and vocal-effects loops. I was expecting a few more of the latter given the 'Parental Advisory: Explicit Content' label on the front of the packaging — and I couldn't find anything that might be considered particularly offensive in the content. The number of loops within each kit varies from four or five up to about 20 — although in testing within Sony Acid Pro 5, it was generally very simple to mix and match loops between the various construction kits. While there are some exceptions, the overall style of the kits is actually quite mellow, and I can see what Power FX are getting at with the word 'cinematic' in the title; this material would work really well in a TV/film context given the right sort of 'urban night-life' footage. The titles of some of the kits reflect this vibe (Summer, Sweet, and Moody, for example), and these typically include some nice, lazy synth-melody loops that top off the laid-back mood. There are some darker moments (as in the Bad Ass and Evil kits), but on the whole this will suit those that like their hip-hop mellow rather than gritty. That said, with suitable application of a little lo-fi processing, many of the drum and bass loops can easily be given more of an edge — and at lower tempos, might even work well in a trip-hop context. Without exception, these loops appear to have been well recorded and edited. I'd have no problem using them in a commercial setting, and, as with all the Power FX sample material, the license allows the loops to be used in almost any musical context. That said, the sample world is not short of hip-hop or R&B loop collections, and I'm not sure there is anything dramatically new on offer here. At this price Cinematic Hip-hop might be considered a casual purchase — as such, it undoubtedly offers very good value for money and contains some very usable laid-back beats and loops. John Walden Acidised WAV CD-ROM, £29.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.powerfx.com

Discography ***** MULTI-FORMAT This Sample Lab loop library is aimed specifically at the disco-funk-house hybrids which have enjoyed so much success on the dance floors of Europe in recent years — where the 'disco' is concentrated in the breaks, snares, and bass/guitar licks, while the 'house' is most apparent in the kicks, hats, and ethnic percussion.



The layout of Discography has a commendable amount in common with that of the company's Luscious Grooves, which enjoyed a five-star review back in SOS October 2004, so the 600 loops are grouped into sample-accurate tempo groups of 120, 122, 125, 128, and 130bpm, allowing immediate and accurate layering of elements. The key of each loop is documented in the booklet where relevant, and every one of the main drum loops also has a short written description, which makes it a hell of a lot easier to find the one you want. Each tempo group has sections for drums, percussion (bongo, conga, tabla, shaker, and tambourine), guitar, and bass. The drums put in a strong performance in exactly the areas that you'd expect, stalking along effortlessly and supplying power where it matters in the kick and snare departments — many of the disco-influenced libraries I've heard have seemed rather anaemic, in my opinion, but there's no danger of limpness here. Although you're never in any doubt about what you've come to this library for, it's nice that Sample Lab include some more stylistically wayward breaks which provide variety without stepping too far outside the overall genre. The percussion tracks are all solid and functional, including some fine conga playing and a few notably weighty shaker parts. The guitar loops pull all the funk rhythm shapes you're likely to need, with clean, wah, and distortion sounds supplemented by the odd rhythmic delay. All slot well into the stylistic niche, and are certainly comparable in quality to the contents of some of the more specialised rhythm-guitar titles. However, the highlight of Discography for me is the selection of bass loops, which are effortlessly nimble and so deep in the pocket that they might as well be hiding underneath a layer of loose change! On the downside, you only get the one bass guitar sound, but I'm not complaining because it's a lovely one, with weight and 'wire' in pleasing proportions. Upon discovering that only the first three tempo groups included bass loops, I almost chewed a star petulantly from the SOS rating, but on closer examination bass loops turned out still to make up 10 percent of the total file count, so the toys were soon back in the pram. Overall, the taste, craftsmanship, and musicality of Discography fully deserve five stars. Mike Senior Audio CD and EXS24, Halion, Reason Refill, and WAV CD-ROM 3-CD set, £59.95 including VAT. Time + Space +44 (0)1837 55200. +44 (0)1837 55400. Click here to email www.timespace.com www.samplelab.com Published in SOS June 2005






Sensorcom Soundcheck

pros


Informative tricolour LED Sound Level Indicator metering. Published in SOS June 2005 Indicates when noise levels Print article : Close window become potentially dangerous. Small and inexpensive. Reviews : Accessory

cons Accurate only to within 3dB, but this should come as no surprise given the price.

summary A very useful little gadget which could save you from hearing damage.

information £10 including VAT. Sensorcom +44 (0)870 901 6070. www.sensorcom.com

Hugh Robjohns

Tinnitus and other permanent hearing damage are apparently reaching nearepidemic levels. To help warn people when their hearing is at risk, the hearing-protection specialists Sensorcom have produced the Soundcheck key fob. This is a simple audio meter with an LED noise-level display. The first flickerings of the green LED indicate a noise level of 55dB, becoming steady at 60dB. The amber LED starts flickering from about 75dB and becomes steady at 80dB. The official First Action Level, as defined by UK Noise At Work legislation, is a noise level of 85dB (although this will reduce to 83dB from April 2006), which obliges employers to advise staff of potential danger and to offer some protection. The red LED has three indications. A slow pulsing represents a noise level of 100dB, which is above the Second Action Level (currently 90dB but reducing to 87dB from April 2006) at which employers are obliged not only to provide hearing protection but also to enforce its use. Rapid pulsing of the red LED represents levels above 100dB, and the light becomes steady over 105dB, where noise levels become very dangerous. Clearly, the Soundcheck is not intended to replace a properly calibrated sound level meter for accurate assessments of sound levels — although its calibration does appear reasonably accurate when checked against an NTI Minilyser with its calibrated mic. But it is certainly a very handy tool to have to hand, and will apparently perform around 1000 level measurements per set of button cells. It is available on-line at a very affordable UK price. A highly recommended accessory for anyone interested in protecting their hearing. Published in SOS June 2005

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file:///H|/SOS%2005-06/Sensorcom%20Soundcheck.htm (2 of 2)9/28/2005 2:35:48 PM

Sony Sound Forge v8

In this article:

Three Of The Best Previous Sound Forge Reviews In SOS Scrubbing & Scripting CD Architect 5.2 Mirror, Mirror, In The Hall Sony Sound Forge Studio Hammer & Tongs In Use Minimum System Requirements Summing Up

Sony Sound Forge 8 £299 pros

Sony Sound Forge v8 PC Stereo Editing Software Published in SOS June 2005 Print article : Close window

Reviews : Software

Sony have opened their Sound Forge editing software up to new horizons with support for VST plug-ins and the ASIO driver protocol, and improved its usability with batch processing and a new scrubbing tool. Alan Tubbs

VST support. ASIO support. Batch Converter and Scripting relieve the tedium of repetitive jobs.

Everyone needs a stereo audio editor. It's not that you can't accomplish many of the same tasks using your Digital Audio Workstation (DAW), but it is a lot CD Architect is back, with more elegant to use a hammer to drive CD-Text capabilities. in a nail than the flat side of a wrench. Audio scrub tool. Just because it can be done, doesn't Looks and works more like mean it should be. For many sound Vegas and Acid. jobs, a stereo editor is that hammer, cons making the given task easier and Won't import AC3 files. faster. Sound Forge was one of the Doesn't support surround A single sound open in a window, ready to first professional audio editors formats. be edited. Common procedures are available developed for the PC, and was last from the drop-down menus under Process No Dim button. and Effects. VST and DX effects can be reviewed by Sound On Sound when summary called up from the Plug-in Chainer tool version 6 appeared back in 2002. In Sony have added functionality highlighted by the cursor. the intervening years, Sony have and flexibility to their old bought Sonic Foundry's entire line of audio-editing warhorse while audio and video software, including Vegas and Acid, updated them all, and have retaining the ease of use of the program. Sound Forge 8 just released version 8 of Sound Forge. It is high time to take another look at this adds much of what was PC staple. missing from the early versions, other than support for surround sound formats, and for stereo editing and CD creation, it is hard to beat.

Three Of The Best

information £299 including VAT.

There is, of course, the usual laundry list of improvements, but three updates

file:///H|/SOS%2005-06/Sony%20Sound%20Forge%20v8.htm (1 of 10)9/28/2005 2:35:58 PM

Sony Sound Forge v8

SCV London +44 (0)20 8418 0778. +44 (0)20 8418 0624. Click here to email www.scvlondon.co.uk http://mediasoftware. sonypictures.com

Test Spec Sound Forge v8.0 with CD Architect v5.2. AMD Athlon PC with 1.7GHz CPU and 1GB RAM, running Windows XP. Tested with Sony Vegas, Waves and Prosoniq Timeworx effects.

really stand out. Back in 2002, Martin Walker's review of SF v6 (www. soundonsound.com/sos/sep02/articles/soundforge.asp) lamented the disappearance of CD Architect, which had been included with previous versions of SF. Well, it's back. Granted, CD burning is no longer the mysterious alchemy it was right after the millennium rolled around, when one was as likely to add to the toaster collection as get a playable CD. Still, CD Architect is a professional-grade tool and a step up from the software that comes bundled with every computer or CD burner these days; see the CD Architect box for more details. There is even an Export to CD Architect command in SF8, which automatically opens CD Architect and puts the sound files into the media browser. The second major update in SF8 is Previous Sound Forge Reviews In SOS ASIO driver Sound Forge 6 support, www.soundonsound.com/sos/sep02/articles/soundforge.asp which means no more Sound Forge 5 switching www.soundonsound.com/sos/nov01/articles/soundforge5.asp preferences in your Sound Forge 4 soundcard www.soundonsound.com/sos/1997_articles/mar97/soundforge4.html drivers to Sound Forge 3 WDM, or having to www.soundonsound.com/sos/1996_articles/may96/soundforgev3.html listen to a sound file over computer speakers. If your system uses ASIO, SF8 can use the same soundcard and monitor chain as the rest of your computer music software. Sound Forge should have had this capability all along, so I'm not sure whether this is as much a plus as simply dispensing with an anachronism, but better late than never. The third major improvement is VST plug-in support. Again, this should have been done before the dust settled in the battle between Microsoft's Direct X plugin standard and the universally accepted VST standard, but it is still a very welcome addition. Certainly SF8's included effects are not to be sneezed at, but most musicians and engineers have VST effects on their computer, ranging from the free sort found on the Web up to the ones that linger on your credit card. If you are anything like me, these effects include several favourite mastering tools. As one of the main reasons for having a stereo editor is mastering, finally being able to use these VST processors is a major plus for SF8. Before, I was forced to switch back to my DAW to apply my favourite VST finalising, which is not the quickest way to polish a track.

Scrubbing & Scripting


Sony Sound Forge v8

Other updates to SF8 concern the ergonomics of using the program. Sony have added a scrubbing tool just like the one in Vegas, which can be controlled by a mouse or assigned to a jog-wheel controller. It scrubs sound and video forward or back, slow or fast, and in any combination thereof. Double-clicking with the mouse allows you to write in the rate of playback, and for those who prefer to keep their hands on the keyboard, the letters 'J', 'K', 'L' and the Ctrl key can replace mouse control. The scrubbing tool will be a major upgrade for many users: while space-bar play/stop and the virtual Play, Record and Stop buttons are more often used in the typical audio job, the scrub tool is great for finding precise edit points. It might not be as tactile as 'rocking' the reels of an analogue recorder, but it gets the job done just the same, and it is a lot faster than an analogue deck when finding the next edit point. The different sections of a song might be easy to discern in a multitrack DAW by the start of a backing vocal or a discreet hit, but masters of today's overcompressed mixes often look like a straight-edge ruler plopped down in the middle of a stereo editor. Fast-forwarding while listening is often the quickest way to find, say, the second chorus, or any other edit point. Sony have also added customisable keyboard mapping for those who prefer their fingers never to leave the computer keyboard. Under Preferences, a whole page is devoted to setting up keyboard commands, as well as importing and exporting them. If you have no excess cranium capacity left for yet another series of keyboard commands, you can simply match Sound Forge's keystrokes to those of your DAW.

The Batch Converter in action: Metadata can be written to any or all of your files.

New efficiencies also come from Sony's Batch Converter and Scripting. The Batch Converter is as simple as it sounds. Of course, one can do more than convert batches of sounds to a different format with it, but other processes are well laid-out and easy to understand. From Tools, choose Batch Convert and a separate window pops up. There are five tabs. 'Files to convert' is the first tab. Choices include single files or entire folders, which is simple enough. The Process tab inserts any effects to be applied during Batch Convert, making it an easy way to dither mixed files down to 16-bit for CD, turn WAV files into MP3s, or master a CD's worth of songs all at once with the same chain of effects. The Metadata tab embeds your choice of information with the file, so that record exec can't lose your name and phone number as long as he has the CD. Once a batch job is ready to go, use the Save tab. If a deadline is looming, you can keep an eye on the process in the Status tab. The whole is very simple, very neat and very functional. Scripting provides similar functions for more esoteric tasks. There is a list of common jobs, such as normalising and rendering to formats, cropping and fading, and modifying summary info (for when the lead singer finally quits the


Sony Sound Forge v8

band for good). Sony have also thoughtfully included an editor for writing and modifying scripts. At the Sony site (http://mediasoftware.sonypictures.com/), SF users can exchange scripts or tips. Along with the disclaimer about downloading someone else's scripts — after all, scripts can change and erase files on your hard drive! — the forum moderator had already supplied a couple of bug fixes and some requested scripts. I wouldn't depend upon Sony's moderator for writing a script if I had to have it today, but it is reassuring to know that a week after SF8's release there was already some good stuff in the forum. The above features may turn out to be most useful for major studios or big post-production houses, but I, for one, never complain about being able to play on the same field with the big boys when at home.

CD Architect 5.2 Today, burning CDs comes as standard on computers, but this wasn't always the case, and CD Architect was one of the first readily available CD-burning programs. There are still a couple of things that make CD Architect a more professional tool than most bundled software. You can apply any effect to all tracks to be burned, or just to a single song. New in 5.2 is support for CD Text, allowing the latest CD players to display song title and band name as your music plays. Song lists can also be exported or printed from CD Architect. Most of these functions can be done in SF8 itself, so they might seem redundant, but if you've ever wanted to get the fade between songs just right, CD Architet allows you to open up two tracks and apply different fade shapes to each song. Say bye-bye to the frustration of automatic crossfades.

Mirror, Mirror, In The Hall How do you get to play Carnegie Hall? Practice using convolution reverb. But seriously, convolution reverb is the latest and greatest method of adding natural reverberation to a dry sound. Paul White's leader in SOS January 2005 related how he had replaced his hardware effects boxes with computer-based convolution reverb as part of simplifying his studio, and Martin Walker covered the ins and outs extensively in the April issue (www.soundonsound.com/sos/ apr05/articles/impulse.htm). It doesn't hurt, of course, that convolution reverb sounds good, too. Convolution reverb can be thought of as 'sampling' an acoustic space, within which a sound can be virtually reverberated. When a sound comes out of the other side of a convolution algorithm, even an older PC can put your lead vocal into Carnegie Hall or another nice, natural-sounding space using offline processing. Sound Forge was one of the first programs to use convolution reverb, in the shape of its Acoustic Mirror plug-in, and this has aged gracefully. You can not only hear the reverb of the real space, but see it, too, as the included reverbs have small pictures of the spaces captured. The plug-in has a generous amount of control over the reverb itself, including delay and width, low and highfrequency shelving EQ and the application of a breakpoint envelope to the file:///H|/SOS%2005-06/Sony%20Sound%20Forge%20v8.htm (4 of 10)9/28/2005 2:35:58 PM

Sony Sound Forge v8

natural decay. A Preview button auditions the chosen reverb, and if none of Sony's own impulse responses fit the sonic bill, it is possible to create one that does. Included on the CD are two test tones for this purpose. To create an impulse response you simply amplify them in the chosen environment and record the result, a process which is clearly explained in the manual.

"Holy smoke, Robin! To the Batcave!" The Acoustic Mirror convolution reverb; other pages offer controls that can shape the natural reverb.

This is an invaluable tool for dialogue replacement and foley work, because if you have an impulse response for a space, you can match the characteristic of the original recording. Even if you don't plan on a career in A/V post-production, having an impulse for an acoustic recording room can save a song. Years ago, I replaced a lead vocal chorus, and the new version had to be recorded in a different room from the original. The replacement lines not only had a totally different reverb, but even a naturally different EQ. To fit them together I had to slather them in artificial reverb. The lead vocal required reverb anyway, so it didn't sound bad, but today I would use Acoustic Mirror to match the characteristics of the original room, requiring less reverb on mixdown. It is even possible to capture the sound of a favourite compressor or other outboard device in Acoustic Mirror and use that to process a recording. And by using a plain old WAV file instead of an impulse file, you can create all kinds of special effects. It takes time to come up with something that works, but such an experiment can produce some interesting results.

Sony Sound Forge Studio Everyone may need an audio editor, but not everyone can afford it. Along with most other manufacturers, Sony have their product line covered for this unfortunate eventuality. They retail for a fraction of the price of the full programs — SF Studio costs £69 including VAT. What goes missing along with your pounds? First, there's no support for sample rates above 48kHz. Second, there is no VST support. Third, the effects package is truncated, although most of the offline processes remain, and there isn't even support for third-party Direct X plugins. Finally, there's no Acoustic Mirror or CD Architect, although SF Studio will still burn TAO CDs. These might or might not be crippling problems; don't forget that up until version 8, the full Sound Forge wouldn't do VST either, even if other DX plug-ins worked. And lack of 24/96 support might not be too disastrous: after all, 16/48 was state of the art just a few years ago, and unless you already own an expensive recording studio you are not likely to hear much difference. Otherwise, Sound Forge 8 and the Studio version look and work much the same. Nobody needs to know except you and your wallet. file:///H|/SOS%2005-06/Sony%20Sound%20Forge%20v8.htm (5 of 10)9/28/2005 2:35:58 PM

Sony Sound Forge v8

Hammer & Tongs Sony include a comprehensive round-up of processes and effects under dropdown menus. All are compiled off-line, but include a Preview button so you can hear the effect before processing. Besides the normal track and buss effects one expects to see in any audio program, there are some audio editor-specific processes such as Auto Trim/Crop, Insert Silence and Reverse. Sony's Wave Hammer compressor/maximiser is very nice, and a step above Sony's other dynamic effects, with a slightly more 'vintage' sound. It doesn't have a fancy user interface, but all the usual controls are there. I originally thought there was something wrong with my copy of Sound Forge, since Wave Hammer came up in demo mode. It was only after perusing the Readme file that came with Sony's SF Audio Studio program (see box overleaf) that I realised installation of both on the same partition was verboten. The Readme file warned about installing SF Studio after SF, but it turns out there are problems installing SF8 after Studio, too — which is, I imagine, more likely to happen. Uninstalling SF Studio and perhaps reinstalling SF8 would doubtless solve the problem, but in the meantime I used Wave Hammer Surround from Sony's Vegas multitrack software, which defaulted to stereo processing within Sound Forge — effects from one Sony program work with the other(s). Acoustic Mirror is available in Vegas, just as Wave Hammer Surround pops up for SF. For Sony users this is another plus, and each subsequent roll-out of the different programs tends to make the whole line look and act more alike, with enhancements to one program also added to others. However, Sound Forge is still waiting for the surround sound support that is now available in Acid and Vegas, not to mention some competing 'stereo' editors like Wavelab. Not everyone who needs an audio editor wants or can afford Vegas too! Another negative aspect to the Sony range is that their Direct X effects can't be used within other companies' software, which is a shame — Cakewalk's, for instance, get along just fine within the Sony DX environment.

For the audio engineer on the go, Sound Forge's built-in synthesis functions allow you to create test-tone sweeps and burn the results to CD.

Those who prefer to roll their own loops and samples will find a stereo editor such as Sound Forge invaluable. SF, of course, is designed to work hand in glove with Sony's Acid software. You can turn a hit into a one-shot sample, or Acidise a longer file to allow its tempo and pitch to be manipulated in Acid or Acid-compatible software. One of the original uses for Sound Forge was exporting trimmed and polished samples to hardware


Sony Sound Forge v8

units, and it still supports specific models from the likes of Akai, Emu and Kurzweil, as well as generic SMDI and SDS modes. SF will also produce synthetic sounds for sampling. Most samplers today, whether soft or hard, come with extensive libraries, but if you need a specific wave shape or FM sound, it can be produced inside SF and exported. The FM synthesis uses only four operators, so it is more like the Yamaha FB01 than the DX7, but it is still capable of making the distinctive sound of frequency modulation (or four-partial additive synthesis). The wave synthesizer can also produce frequency sweeps, which are perfect for checking out recording room or venue anomalies. There is even an arcane process for producing TDMF/MF tones, just like on a punch-button phone. I'm sure some designer had a good reason to include this function long ago, but I can't imagine what it was! One or two minor things are still missing from Sound Forge 8. Sony introduced a Dim button into Vegas, where I seldom use it, and I was hoping it would make it into SF8, where I would. Sometimes, when you over-process a sound, the file redlines the meters and output. My audio interface, mixer and amplifier are all within reach, of course, but it would be quicker to limit the damage with the mouse.

In Use As mentioned above, a DAW can accomplish many of the same functions as any stereo audio editor. So why lay out good money for another program? Well, I said many tricks, not all. Otherwise, why would your DAW have an option for exporting a sound to an audio editor? The first job that comes to mind is mastering. SF8 does non-destructive editing, at least until the file is saved, Minimum System Requirements contains a host of Sony effects, and can Windows 2000 or XP. now use VST effects too. Once the 500MHz processor or faster. song is mastered to you or your client's taste, it can be dithered to 16-bit, if 150MB hard disk space. necessary, and burned to CD using the 128MB RAM. Track at Once (TAO) CD writer built into SF8 or by exporting to CD Architect. The dithering in SF8 is first-rate, with several different presets available. Finally, there is the psychological advantage of mastering in a different program from your DAW. Close the DAW and open SF8 and you've switched hats. No more going back again to adjust the volume of that shaker sample that spices up the third verse — at least not without stopping and reopening both DAW and song. About the only function that would be quicker in a DAW is laying different files on a timeline and fading between them — it would be nice to be able to fade sounds over one another in a window directly. Even if a DAW suffices for mastering, a garden-variety musician/engineer will file:///H|/SOS%2005-06/Sony%20Sound%20Forge%20v8.htm (7 of 10)9/28/2005 2:35:58 PM

Sony Sound Forge v8

find that many tasks go easier and quicker using a stereo editor. A band I had worked with wanted to swap out one song on a demo CD for another from a second. A simple enough task: SF8 can extract songs off a CD — or two CDs in this case — and automatically opens windows for the data; in this case, the threesong demo in one window and the single song in the other. Highlight the song to be eliminated, cut it and delete. Highlight and cut one of remaining two songs, save and paste to new, with SF8 conveniently opening a new window. The three songs were now ready to export, except that the new song had an unnatural ending because its reverb tail was chopped off. Either someone had trimmed it back too far or had forgotten to tick the Play Effects Tail command. Never mind — it was easy to place the cursor on the last beat, pick an appropriate, unobtrusive reverb and add it to only the end of the song. The song now faded into silence, rather than a jarring cut. Save, then burn the demo. The whole job took only a little longer than to write the description of it. Another chore for which I used SF8 was stripping out the audio of a song from a video project. The singer had changed her performance, was going into the studio and wanted a copy of it so she could practise in her car. This was harder than it should have been. I had a copy of the band's promo DVD, which included a live version of the song, but none of the Sony software will load the AC3 or VOB audio portion from a DVD. This was a problem in Vegas, too, where to make even a minor change to a DVD one has to go back to the original source files. I hoped SF8 would cure this problem, but no such luck. The video loaded in fine, but no audio. (Video in SF8 is displayed as a film strip, which can be animated. In the default film strip size, the motion within the small frames looks like stop-motion animation or Gumby claymation, which is cool, but didn't do anything about the audio problem!) Fortunately, I got a hold of the DAT tape they had used, and I was able to retrieve the song from there. I simply recorded the song straight into SF, cleaned up the in and out points, resampled the 48kHz file down to 44.1kHz and burned it straight from the TAO function. For sound design and manipulation work, moreover, SF8 is perfect. I had collected some wicked-sounding metal gates at a farm. I want sound effects for my library to add a little ominous spice to almost-finished tracks. Inside SF8, the metal screeches became ominous and more. After capturing the five-plus minutes of DAT tape in SF8, I deleted all the unwanted noises — I had also tried to record my footsteps as I tromped across a wet cow lot, but I Four tracks are better than two: choose the was torn between getting a good signal right fade for each song or butt-splice two songs together. and getting the mic too close to the sucking muck, and the borrowed stereo mic won out. Once the useless parts were excised, I saved the good parts and then began my Frankenstein experiments. Likely bits were pasted out for mangling. Along with the more commonplace effects, SF8 includes some esoteric sonic software. Time-stretch and pitch-shift are catered for, both at the file:///H|/SOS%2005-06/Sony%20Sound%20Forge%20v8.htm (8 of 10)9/28/2005 2:35:58 PM

Sony Sound Forge v8

same time if needed, and the latter includes a bend function, allowing the amount of pitch-shift to change over time — one of the presets is called Whammy Bar, which can be customised to less (or more) extreme settings. The stretch/pitch algorithms are good: although I did notice some artifacts using the Whammy Bar preset, at more sane settings they did a great job. In no time at all I had a collection of evolving sounds that were brasher and more interesting than most pads. They all reeked of metal that was twisted and distorted. SF8 includes an Undo function, of course, so when a sound went bad, I could just back up a step and try something else. I also tried using some of the same files I was working on as impulses in Acoustic Mirror. This was the only time SF8 crashed on me, but that was because I was changing impulses while previewing the effect — Acoustic Mirror uses a lot of computer overhead, and I overtaxed my system. When I restarted SF8 after the crash, the file I had been working on was recovered and popped up in a window, ready for action.

Summing Up Sound Forge 8 is a mature program. There are few things it won't do, given its intended purpose, and the more you use it, the more tricks you realise you can do with it. Not only does it make many tasks quicker, but its speed and ease of use practically beg you to try out new ideas. That in itself is worth the price of admission, in my book. SF8 is still no impulse buy, unless you're in a different tax bracket from most of us, but it is a better deal than ever. Even if you don't really need an audio editor and can live with your computer's basic CD burner program, Acoustic Mirror might seal the deal, if you don't already have a good reverb — you can simply export your track(s) to SF8 and add convolution reverb to taste. You could even export multiple tracks and use the Batch Converter to apply different reverbs to them. It's not as elegant as using a VST effect within your DAW, but then a VST effect won't edit your stereo masters, either. Like its sister package Vegas, Sound Forge 8 is a very easy program to learn and use. Not having to deal with MIDI (except for sync functions) helps, of course. But if you come from the analogue recording world, or are at least familiar with cassette decks, everything is well laid-out and logical. The ergonomics of analogue recording developed over half a century or so, and there is no reason why an audio-only recording software package should try to reinvent the wheel. Sound Forge doesn't reinvent the wheel, but the additions and upgrades in version 8 make it a more perfect circle. Published in SOS June 2005


Sony Sound Forge v8




Studio Projects B1

In this article:

Fixed-cardioid Capsule Subjective Impressions Specifications

Studio Projects B1 £67

Studio Projects B1 Condenser Microphone Published in SOS June 2005 Print article : Close window

Reviews : Microphone

pros Open but smooth sound. Effective shockmount included. Very inexpensive.

cons

The B1 may seem like just another cheap Chinese mic, but it punches well above its league.

None at this UK price.

summary The B1 is one of the cheaper large-diaphragm capacitor mics available at the moment, but there's nothing about the sound quality that gives away the price.

information £67 including VAT. PMI Audio UK +44 (0) 1803 215111. +44 (0)1803 215111. Click here to email www.pmiaudio.com

Paul White

Studio Projects are part of the PMI Group, which now owns the Joemeek brand and distributes a number of other products including Toft Audio Designs. They're often seen at trade shows with their range of Studio Projects mics set up alongside industrystandard high-end models, where the visitor is invited to compare them, and they openly admit that their aim is to capture the essence of the German microphone sound at an entry-level price. To achieve this, their mics are designed in the USA by Brent Casey, and then built by 797 Audio in Beijing using high-quality components, including Wima capacitors. 797 was the original government number for the facility in the days when it was under government ownership, but today 797 Audio build mics and capsules for a small number of different companies around the world. Rather than trying to disguise the fact that these mics are made in China, Studio Projects put the 797 Audio logo on all their mics, in part I'm told to ensure better quality control from the factory as a result of their name being on the mic. Each microphone is tested at the factory and again by Studio Projects in California prior to packing and dispatch.

Fixed-cardioid Capsule The B1 is the least expensive mic in the range and offers a fixed-cardioid pattern file:///H|/SOS%2005-06/Studio%20Projects%20B1.htm (1 of 4)9/28/2005 2:36:07 PM

Studio Projects B1

with no pad or roll-off switches. The centre-terminated, large-diaphragm capacitor capsule (approximately one inch in diameter) requires a standard 48V phantom power supply and feeds a FET cascade input with a direct-coupled transformerless output. The diaphragm itself is made from Dupont Teijin three-micron, gold-evaporated mylar. Housed in a satin-nickel plate-metal enclosure with gold-plated XLR connector pins, the B1 has a fairly wide cardioid response that remains tonally true beyond 45 degrees off axis. As with all cardioid mics, it also exhibits a proximity bass boost when used up close, and this can be used creatively to add weight and warmth to close-miked vocals. A dual layer metal basket protects and screens the capsule, which is itself shockmounted, and the Studio Projects logo denotes the live side of the mic. The B1 comes complete with shockmount, foam windshield, and soft plastic storage pouch, but to keep the cost down it is supplied in a cardboard box rather than a fancy camera case or wooden box. I'm unsure what the foam windshield is supposed to be for, as these things are usually pretty ineffective against popping and also tend to compromise the high end, though they can be useful in reducing wind noise when used outdoors. I feel it could mislead the inexperienced user into thinking they can work without a separate pop shield, which when recording vocals is most definitely not the case with any microphone of this type. Unusually, the shockmount features a custom plastic moulding that clips to the lower end of the mic suspended within a metal ring using standard fabric-covered elastic belts. The mic is a tight fit in the cradle, but once it's in, it's not going anywhere!

Subjective Impressions My practical tests confirmed that the B1 is more than adequately quiet, and similar in sensitivity to the other large-diaphragm models I used for comparison. It has an open, subjectively uncoloured sound, but with a welcome density at what I tend to think of as the 'chest' frequency of male vocals. If there is a presence peak, it is suitably subtle. Tube mics and some compressors also create this impression of density, and it sounds particularly flattering when you're closemiking vocals. However, one very pleasant surprise was the way this mic interpreted acoustic guitar. Most capacitor mics will render a fairly clean and natural-sounding recording of an acoustic guitar if correctly positioned, but this model added some flattering weight to the sound and seemed less critical of positioning. It also seemed to smooth out the rough edges without losing any detail — the top end comes over as open and detailed, but without being harsh or scratchy. Although

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Studio Projects B1

the overall effect may not be quite so refined as that of a £1000 mic, it's really not as far behind as you might imagine. Given the level of performance of which this microphone is capable, I'd say that it would suit the vast majority of home-studio vocal or instrument applications (including guitar amplifiers), and if you can't get a good sound out of it, then the chances are that the room acoustics or the mic position are at fault, not the mic itself. Clearly it isn't the best mic in the world, and you wouldn't expect it to be for only around £70 in the UK, but I'd be very happy to use it as a main vocal mic for serious recording projects. There's nothing not to like — the tonal balance is nice, the off-axis response is more even than most mics manage, and the build quality is solid enough. The budget Chinese mic market is hugely competitive, but the B1 is one of the few mics that has made me sit up and take notice. In fact, I'm seriously thinking of adding one to my collection, if just for what it does for guitars!

Specifications No response curves are provided with this microphone, though I managed to get some from the designers that show the mic to have a very subtle presence lift centred at around 10kHz combined with a gentle bass roll-off below 100Hz. The published frequency response specification simply says '20Hz to 20kHz'. The noise level (EIN) is 12dBA (giving a signal-to-noise ratio of 82dB), which is slightly quieter than is typical, and the sensitivity is 34dB (reference 1Pa), which is The frequency response and polar pattern pretty standard for this type of mic. diagrams showing the technical performance The maximum SPL is an impressive of the Studio Projects B1. 137dB, which is higher than normal and certainly enough for any conventional vocal, acoustic instrument, or ampmiking work, which is what the mic is really designed for. Published in SOS June 2005


Studio Projects B1




Wizoo Darbuka & Latigo

In this article:

Focus On Darbuka Play Today MIDI Control Drum Lore Focus On Latigo Mix It Up Installation & Hosting Close To The Edit Conclusions

Wizoo Darbuka/Latigo £179/£179 pros Easy to use. Love that icon-based mixer! Varied collections, wellrecorded. Patterns can be varied and manipulated to a decent extent.

cons No access to individual hits. Restricted to just the patterns supplied.

summary Great recordings of great percussionists, shoehorned into an easy-to-use plug-in: it's like inviting these musicians to your bedroom studio for an indefinite sleepover. What's more, there's enough editability to keep things fresh, and to let you put your stamp on the software's output.

Wizoo Darbuka & Latigo Flexgroove Software Percussion Instruments [Mac/PC] Published in SOS June 2005 Print article : Close window

Reviews : Software

The humble sample CD-ROM is dying out, being replaced by sample collections with a virtualinstrument front end. Wizoo's first forays into the field combine Latin and Arabic percussion loops with their own Flexgroove virtual-instrument engine... Derek Johnson

If you're a regular reader of SOS's Sample Shop pages, you can't have failed to notice the increase in the number of sample libraries being sold with virtual-instrument 'wrappers' or front ends, to the point where traditional sample CDs or CD-ROMs are beginning to look as viable as the dodo. And it's not hard to see why — the sheer convenience of accessing sounds via the familiar interface of your favourite sequencer is hard to beat, compared to dragging in multisamples from a CD-ROM, or looping and multisampling material from audio CDs.

This pair of plug-in instruments from long-standing sound designers and samplelibrary creators Wizoo (the team behind Steinberg's Hypersonic virtual information instrument) take the concept a little further. If instrumental sample collections can £179 each including VAT. be packaged with plug-in front-ends, why not loops of rhythmic material? Such is M Audio +44 (0)1923 the rationale behind Darbuka and Latigo, two 'virtual percussionist' instruments 204010. that offer users the sounds and performances of the Middle East and Latin and +44 (0)1923 204039. Central America respectively, in surround sound if you wish. Click here to email

www.maudio.co.uk

Test Spec

Neither Darbuka nor Latigo allow direct access to individual percussion hits, although of course you could resample them if you wanted to. At their collective hearts are large collections (1GB for Latigo and 2GB for Darbuka) of themed

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Wizoo Darbuka & Latigo versions reviewed: v1.0. Cakewalk Sonar v4.1. Steinberg Cubase SX v2.2. MAC REVIEW SYSTEM 450MHz Apple Mac G4 with 896MB of RAM, running Mac OS v10.3.8. PC REVIEW SYSTEM 3.06GHz Pentium 4 PC with 512MB of RAM, running Windows XP.

percussion performances featuring instruments typical of their genre, played by experts in their field. It's rather as if a group of the most patient (and cheap!) session percussionists were living on your hard drive. A healthy selection of typical styles is available, with individual instrument patterns that can be triggered solo or en masse, complete with variations, intros, fills and other extra material. And of course it all integrates simply into your host sequencer. Providing access to the sampled performances in both of these plug-ins, and allowing users to manipulate them and inject something of their own personality is Wizoo's new Flexgroove virtual-instrument engine, a technological development which is apparently planned for use in more Wizoo libraries. As you might expect, both libraries allow you to play back their looped performances at any tempo while sync'ed to a host sequencer (triggering via an attached MIDI keyboard or on-screen with a mouse), but Flexgroove goes further than that. You can customise the final arrangement, loosen or tighten a performance with timing effects, quantise performances, adjust Complexity (which makes a sampled performance more or less busy) and freely set level and pan for all instruments. And all this is done via an intuitive graphical front end. Despite the different performance loops in these instruments and the different overall musical styles, these two plug-ins are conceptually and operationally identical, so be aware that my comments on usage apply to both Darbuka and Latigo unless otherwise stated.

Focus On Darbuka This 2GB Middle Eastern collection features two percussionists — Suat Borazan and Mohamed Zaki — playing no less than eight instruments in various rhythmic styles from Egypt, Algeria, Lebanon, Turkey, Sudan, Tunisia, Morocco, and Libya. All in all, these are very groovy performances, recorded with a very 'live' feel. The collection's name is also the name of one of the drums being played: it's a single-headed, Darbuka's Play page, complete with typically waisted, hollow hand drum played geometric arabesque look. with both hands that's found all over the Middle East. Also in the set is the 'douhola', a kind of bass darbuka. 'Bendir' and 'riqq' are both circular handheld frame drums, the latter being rather like a small tambourine. Finger cymbals are an effective part of the mix: the small ones are called 'sagat', the larger 'tura'. The collection also features, rather less exotically, bongos and shaker. They need no introduction to SOS readers, but fit well with the rest of the ensemble. A reversed darbuka or douhola hit also appears in some styles.



Play Today Before having a look at the software, let's get some naming issues out of the way. Wizoo have made their nomenclature a little confusing, though in practice it's fairly straightforward. At the highest level, both programs contain lists of Styles. A Style in Flexgroove-speak is a thematically related collection of percussion performances (known here as Patterns). The confusion arises because each Style can accommodate up to 14 percussion instruments, arranged as Tracks. These track/instrument combinations play back individual rhythms, dubbed Grooves or Parts by Wizoo. Multiple Grooves/Parts set to trigger simultaneously from one MIDI event form a Pattern, and each Style can contain Patterns triggered by each MIDI note in a five-octave range. Keeping it simple, a Pattern could contain just one Part, say a kick-drum performance. Equally, it could contain several Parts: a combination of kick, snare and hi-hat Grooves played from one key as a Pattern. The Mix page (here from Darbuka) illustrates

When you fire up Darbuka or Latigo, the excellent Stage graphic, here set to offer the first screen you see is the Play draggable control over pan and level. The window. There are three main windows shaded drum (a bendir) is muted in the track list below. Note also the individual track in total, although each one has four parameters; the ones here are those for the common elements, namely a Style/ selected douhola. Pattern name display (of which more in a moment), a five-octave pattern trigger mini-keyboard, the master volume control, and a switch labelled 'XXL'. All samples within the plug-ins are provided in two formats, either RAM-intensive fullbandwidth samples or more economic examples treated with lossless data compression, and the XXL switch makes the choice between the two. In practice, I could discern no audible difference between them, and the software defaults to loading the compressed samples. The vertical Style/Pattern list on the right is also visible in all three operating windows. Initially, the Patterns will all be related, producing rhythmic elements that belong to the selected Style, but you're free to collect Patterns from different Styles and assemble them into new Styles, to edit their playback settings and save them for future exploitation. Styles are loaded and saved via the display immediately below this part of the window. When you access the Play window, a row of eight percussion 'tracks' appears at the bottom, which fill up with percussion icons when a Style is loaded. Each icon relates to a particular drum, and you gradually become familiar with them as you



work with the plug-ins. In this window, each track has a Mute and Solo button plus a level meter, but clicking on the name brings up the Mix page, of which more later in this review. If a Style has more than eight 'tracks' (there can be up to 14, remember), the row can be scrolled left or right as required. The remainder of the Play page offers control over some global parameters. The master controls for the plug-ins' built-in 'Ambience' processor, for example, are found here — that's reverb to most of us. It's simple, yet rather pleasing: there are three varieties each of hall, chamber, room, ambience and stage algorithms (here called 'characters'), plus just two controls, one for the wet/dry mix and one for the Decay time. The reverb sounds smooth, though, and suits percussion well — not an easy job for digital reverb.

MIDI Control MIDI control within Latigo and Darbuka doesn't begin and end with note triggering and MIDI Clock. In fact, most on-screen controls can be easily and quickly assigned to respond to incoming MIDI controller data. You can choose from a pop-up list or have the parameter learn from the incoming data stream.

Other master processors accessed on the Play page include 'dynamics' and a two-band EQ. The former, a kind of preset adaptive compressor, is even simpler than the Ambience processor, offering just fast, tight or slow responses, and a single control, confusingly labelled 'Density'. Master pattern-playback parameters are also located at the top of the Play page (although individual patterns can be altered in the Edit window to have their own offsets to the values entered here). These playback parameters are where Flexgroove really comes into its own. First of all, there's a Speed control. Both plug-ins can sync to themselves, running at the nominal tempos of the selected Style or Patterns, or have everything sync to the host application. Leaving the plug-ins to run together at their own tempo can lead to some interesting polyrhythmic results! But whether sync'ed to the host or not, the Speed control lets Darbuka or Latigo play at double or half the overall tempo, which is great for those moments when plug-in tempo and host sequencer tempos are so far apart that the feel is suffering. The next playback parameter is Variance, which, as its name suggests, adds variations to patterns within a Style by borrowing hits from other Patterns. The resulting extra complexity can be subtle, but completely in keeping with the current Style. If Variance is an odd parameter for what is essentially loop-playing software, then Complexity, accessed via a three-position slider, goes even further. The three levels (Low, Mid and Max) determine how 'busy' the Patterns in a Style will be. Imagine being able to ask a session percussionist to just lay back a bit and take it easy: that's what this parameter does. Even if a pattern is incredibly busy, the simplified version resulting from a reduction of the Complexity parameter still makes musical sense, and preserves the 'feel' of the original performance.



Next door to the Complexity parameter is another, dubbed 'Instruments'. This is the one parameter that works slightly differently depending on whether you're using Latigo or Darbuka. The parameter simply divides the plug-in's percussion voices into broad groups — five for Latigo (Drum, Skin, Metal, High and Miscellaneous) and three for Darbuka (Bass, Mid and High) — and lets you mute or enable them on the fly. These divisions allow users to subdivide their percussion section either for creative purposes, or for fine-tuning portions of a mix while having instant access to all the voices playing in a Pattern.

Drum Lore If you're new to the percussion instruments offered by Wizoo's software, keep yourself informed by right-clicking on the percussion icons in the Play page's track list: the main display area then brings up a full-sized picture of the chosen instrument and offers a paragraph of description (see the screenshot on the last page of this review).

The remaining controls govern more recognisable sequencing parameters. Timing lets you tighten up or loosen the performances: at one extreme, they become 'robotic', and at the other they can become exaggeratedly loose. Quantise (with Off, 16th-, eighth- or quarter-note options) also does what it says on the tin, and Swing adds a fixed triplet feel that's most effective when you've tightened up the performance with the timing control. Put all these features together — along with the Pattern-specific parameters we'll encounter shortly — and the basic library becomes almost infinitely variable.

Focus On Latigo With a name that's slightly less explicable than that of Dabuka, if perhaps more immediately reminiscent of its content, this 1GB collection features the input of three members of Gloria Estefan's backing band, Miami Sound Machine: producer Clay Ostwald and percussionists Edwin Bonilla and Olbin Burgos. The set is as wide-ranging, geographically, as Darbuka; it takes in rhythms from all over Latin America and the Latigo's Play page — note the jungly Caribbean, as well as recent North background! American mélanges such as salsa. It manages to pack in examples of more instruments into roughly half the space of Darbuka, too. Instruments used that you might be familiar with include agogos, bongos, cabasa, claves, congas (including, from largest to smallest, tumba, conga and quinta), cowbell, cuica, maracas, guiro, samba whistle, shaker, timbales and triangle. On a more exotic note, you'll also find patterns played on bombo (a large Brazilian twoheaded drum), caxixi (a woven basket shaker from Brazil), djembe (a West



African single-headed drum), ganza (a Brazilian metal tube shaker), pandeiros (a large tambourine), reco reco (rather like a guiro, this is played by rubbing a metal bar across a pair of springs), repenique (a Brazilian tenor drum), shekere (a gourd covered with woven beaded netting), surdo (the samba bass drum), tambora (a double-headed barrel drum) and finally guataca, a Brazilian cowbell. And that's not quite all: a more or less standard drum kit is also part of the sonic signature of this music. A kick drum, snare, sidestick, sticks, hi-hat, toms, and crash cymbal usually play together in some combination or other, but can indivudally be solo'd or muted out of a mix.

Mix It Up The Mix window, shown on the previous page, offers one of the most graphically interesting approaches to mixing I've seen recently. An X-Y grid (the Stage), dominates the screen, and is populated by percussion icons. To alter level and pan, you simply grab an icon and move it up and down (level) or and left and right (pan) respectively. It's really simple, and yet provides you with instantly comprehensible visual feedback about your percussive mix. If I liked nothing else about these plug-ins, I'd have to praise the Mix page. Nor is the vertical axis hard-assigned to level: it can adjust no parameter at all, be set to alter the send level to the Ambience processor, or alter front/rear balance (when using the applications' surround options). Another option, Room Mode, uses both X and Y coordinates to control level, pan, rear balance and the Ambience send simultaneously to more realistically place individual instruments in a space; again, though, this works best in surround mixing environments. The instruments then move around an imaginary listener placed at the centre of the stage.

In the Edit window, Grooves are assigned to MIDI notes to form triggerable patterns; individual track offsets are also accessed below.

Another nice trick of this page is that clicking on a drum icon in the Stage immediately highlights the appropriate drum track in the lower window, even if it's not in the currently visible row of eight tracks at the bottom of the window. Incidentally, the Track Mute buttons have an unexpected graphical side-effect. Any drums that are muted are greyed out on the Stage — more handy visual feedback of the software's current state. The Track selectors echo those visible in the Play window (eight are visible, as in the other page), but the main part of this lower display is taken up with Trackspecific parameters. First in line is a three-band EQ, edited directly via a displayed EQ curve. Clicking and dragging three 'handles' alters frequency and file:///H|/SOS%2005-06/Wizoo%20Darbuka%20&%20Latigo.htm (6 of 9)9/28/2005 2:36:28 PM


gain; the mid band even has a controllable Q parameter. In all, the EQ has a satisfyingly wide range of 50Hz to 20kHz. Each Track also has a 'Punch' parameter, a preset compressor that's roughly equivalent to the global dynamics section. Simplicity is again the name of the game: Power, Snap, Hard and Soft are the basic, informatively named presets, you choose the 'Drive' value with the continuously variable knob, and a neat meter that encircles the Drive control keeps you informed of the gain reduction in operation. Obviously, this effect suits percussive material especially well, allowing you to sculpt a sound's volume envelope as well as giving it some 'oomph'.

The Setup display is as one might expect, providing control over the number of individual outs, sync mode and other global options.

Level, Pan, Ambience send and Rear balance controls are also located here (the last of which is operative in surround mode only), plus an individual output routing control. Reflecting the maximum number of Patterns in a Style, the applications can each have up to 14 mono outs, in addition to the main stereo out. How well these can be accessed depends on the host application. All 14 were easily available from within Cakewalk's Sonar.

Installation & Hosting I mentioned near the start of this review that Darbuka and Latigo are operationally identical, but what this doesn't make clear is that despite the different exterior packaging, even the DVD case containing the installation disc is identical. Just like those strange two-in-one sci-fi paperbacks from the 1960s, one plug-in is featured on one side of the case label, and the other side features the other plugin, though upside-down. If you've bought Darbuka, then the Latigo side will have a '30-day fully functional demo version' label stuck to it, and vice versa. With both plug-ins in front of you, opening their boxes reveals the same installer DVD with exactly the same contents. The certification card inside the case (and the overall package and manual, of course) makes things clear, but having both discs on your desk can be confusing. This method of distribution obviously makes economic sense in terms of Wizoo's tooling up for production, and the advantage to you, of course, is that you can try the software you didn't buy before you purchase it! You still have to buy two packages to get both plug-ins, though. The installation process is fairly straightforward: installers are provided for each of the different supported platforms on the Mac (VST, Audio Units and RTAS), but one 'certification', as the authorisation process is called, works for all platforms. In fact, I successfully 'certified' both applications installed on my Mac and PC. There is only one install for the PC — VST — though most serious software will support that format directly or indirectly. For example, I evaluated both plug-ins from within



Cakewalk's Sonar 4 on the PC, which doesn't natively support VSTis. There is no stand-alone mode, so a host sequencer is a necessity. Note that the data folders which contain all the looped performances have to be dragged manually to a Mac's hard drive, whereas they're installed automatically as part of the process on a PC. Also, be aware that an update is available on-line for Mac users who plan to access The installation even includes this the Audio Units versions via Apple's percussion encyclopedia: right-click on a Logic v6.x and higher. Sadly, this track's graphical icons and up pop handy had no effect on a strange anomaly definitions such as this. involving the RTAS version: the plug-in seems to load into Pro Tools LE, but doesn't do so fully. There is no way to access the main window, as a result of which no patterns can be loaded. This issue was still unresolved as I finished my review. Steinberg's Cubase SX was a well-mannered host on both platforms, but be warned now that Wizoo's percussionists behave much more politely on newer computers with lots of RAM. I can report, happily, that Latigo and Darbuka were able to run on my ageing Apple G4. The overheads were manageable, and I was able to do a fair bit of other music alongside. Adding both to a session at once, though, threw Cubase SX over the edge — all sound ceased, and the Performance meter completely maxed out.

Close To The Edit The final window is the Edit page, and it's a busy affair, again split into two sections. The upper section provides you with a Pattern Arranger, whereby individual instruments playing loops can be stacked to create a finished Pattern. You do this by selecting instrument Grooves and assigning them to the horizontal slots next to a MIDI note. This is done via little pop-up menus, and the options available depend on the Style or Pattern chosen. Sometimes there's just one Groove option, sometimes several. It's also possible to assign track Mutes to MIDI keys, for instantly muting a Pattern or Groove at any point. Patterns don't have to loop, but they can be latched, which amounts to the same thing; being able to mute a latched Track gives more control over the final perfomance. More Wizoo Flexgroove technology comes into play in the Edit page, since each drum track can have its own parameter offsets. The parameters here echo those offered globally in the Play page: Timing, Quantise, Complexity, Speed, Level and Variance. In addition, tuning, bend range, dynamics and decay parameters let you fine-tune the end result even further, and an offset option lets you alter the feel completely



by delaying or advancing a track by up to 100ms. Note that the 'dynamics' parameter here isn't a DSP effect, but a way to control the volume difference between loud and soft hits. And bend range governs the effect of incoming MIDI pitch-bend data: you can't trigger individual drum sounds, but this offers one more way of controlling the performance.

Conclusions Though I tend not to gravitate towards software that is essentially 'preset', Wizoo's tools are quite flexible, given that you can't program your own Patterns. There are a lot of elegant touches in the user interface, not least of which is that X-Y mixing stage. The quality of the raw material can't be faulted, either — the included performances and recordings are top notch. And the variations and new patterns that can be coaxed from the basic looped samples with the Flexgroove engine do hold out the prospect of making your own contribution to the final output, too. Of course, you're still not really creating the patterns to start with. But how many of us have the cultural experience and rhythmic discretion (not to mention a sufficiently extensive sample library) to be able to create, within our standard MIDI sequencing environment, such stylistically correct, vibrant, musical patterns as the material presented here? Well, those of you who can will still probably save time with Darbuka or Latigo! And while I don't know exactly how much session percussionists charge (not ones this groovy, anyway), I'd guess that they wouldn't spend much time in your personal studio for £179. So give these instruments a try: your mixes may well thank you. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

DM1000 v2 Add-on Effects AE011 Channel Strip AE021 Master Strip Studio Manager v2 Improvements AE031 Reverb Conclusions

Yamaha 0-/DM-series Mixer OS Upgrades & Add-on Effects pros Easy installation. The upgrades add significant features to the 01V96/02R96 and DM-series desks. Realistic, 'analogue'sounding EQ plug-in with the add-on AE011 Channel Strip package. Open Deck plug-ins from the AE021 Master Strip package are extremely versatile and controllable. Very high-quality REVX reverbs in the AE031 Reverb package.

cons Limited number of plug-ins can be run at once.

summary The v2 OS updates for the DM/0-series mixers make a great set of consoles even better. In addition to a host of important operational and functional updates, v2 also brings support for the Add-on Effects packs, all of which perform admirably and extend the sonic quality and versatility of the consoles considerably.

information 02R96, DM1000 &

Yamaha Digital Mixer Upgrades v2 OS Upgrades & Effects Published in SOS June 2005 Print article : Close window

Reviews : Mixer

Not content with offering some of the most fully featured digital mixers in the world, Yamaha have now upgraded them, and are offering extra effects packs to further expand their capabilities. Hugh Robjohns

The march of progress blurs many a once-distinct boundary, and it's no different in recording studios. Until about a decade ago, audio recorders and mixing control surfaces traditionally had separate roles in professional studios, but the digitisation of audio has meant that the The REVX 'Hall' reverb in the AE031 Reverb two functions have become package (above), as seen in version 2 of increasingly intertwined in the form of Studio Manager, and (right) the detailed the DAW, or digital audio workstation. mixer channel view from the same application. Computer-based DAWs with graphical mixing facilities, and hardware mixers with built-in hard disk-based recorders, are now the norm. Indeed, even these two forms of audio production system are gradually converging, as computerbased systems incorporate sophisticated hardware controllers based on the proven ergonomics of traditional consoles, and hardware platforms adopt the computer practice of software upgrades and elaborate third-party plug-in signalprocessing options. Which is exactly what this review is about: Yamaha, who are famed for their affordable and fully-featured digital mixers, have now introduced a range of optional software plug-in effects processors for their current (third) generation of digital consoles. The mixers, which comprise the 01V96, 02R96, DM1000 and DM2000 desks, are powered by Yamaha's own highly evolved dedicated DSP chips, and the current DSP7 version bestows on these consoles sufficient

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DM2000 v2 upgrades, £219 each (01V96 upgrade is free); AE011, AE021, and AE031 Add-on effects, £349 each. Prices include VAT. Yamaha-Kemble Brochure Line +44 (0)1908 369269. +44 (0)1908 368872. www.yamaha proaudio.com

computing power to retain the full suite of facilities and channels, even when operating at 96kHz sample rates. Equally bespoke and highly optimised DSP6 processors are used to power the built-in digital effects, such as reverb, and once again there is a significant capability here, easily able to operate more sophisticated algorithms than those shipped as standard. At the beginning of this year Yamaha released version 2 operating software for all of their recent 0- and DM-series consoles, and all of the v1 console OSs can be upgraded as a cost option. You pay your Yamaha dealer the fee (£219 in the UK) and receive in exchange a CD-ROM with the upgrade. You then need to hook up your desk to a USB-equipped computer, insert the CD, and follow the instructions. Once you've carried out the main chargeable upgrade from OS v1 to v2, all further updates can be downloaded free from Yamaha's Pro Audio web site (see www.yamahaproaudio.com/download/index.htm). The exception to this process is the 01V96, for which even the basic v2 upgrade is free; it can be downloaded from the same web site at no cost. For the record, the current v2 software versions at the time of writing were v2.13 for the DM2000 and 02R96, and v2.04 for the DM1000 and 01V96. Yamaha's Studio Manager software has also been updated to v2.11, and there is an upgraded USB MIDI driver for Windows XP platforms (v2.13). Incidentally, for those desk owners who don't wish to pay to upgrade, it's worth mentioning that Yamaha have not abandoned their v1 users in the rush to develop the v2 software. Recent bug-fixes were issued for the DM1000 as v1.07 software, and as v1.23 for the DM2000 and 02R96 desks. The latest v1 software for the 01V96 remains v1.03. All new 0- and DM-series consoles are being shipped with the current v2 software, and for most v1 users, upgrading would be a pragmatic and costeffective decision. Aside from various minor bug-fixes, the new v2 software adds significant new features and facilities to these consoles while streamlining the operation of several existing features — in particular expanding the console's versatility and suitability for applications such as theatre and broadcast in addition to the recording and post-production mainstays. More importantly, though, the v2 software opens the door to Yamaha's new range of optional Addon Effects packs. This review is based around a DM1000 console running the latest software in all respects, linked to a Windows XP laptop which handled the upgrades and ran Studio Manager.



DM1000 v2 The DM1000 console was the last desk in the 0-/DM-series to be launched, and as a result the original v1 incarnation benefited from the inclusion of several features that only appeared with v2 software in the two larger consoles — for example, the mix-minus facility generated using the Aux busses. However, the v2 upgrade still brings around 25 new features to the DM1000, in addition to the support for the Plug-in Effects packs. Similarly, the v2 Studio Manager update has several key new features (see separate box on page 184). Among the new console facilities is the ability to assign the channel Encoder knobs to operate the fader level of the alternate fader layer (in addition to all of the existing assignment options), and the pre-fade source for the aux sends can now be switched before or after the channel mute button. The routing of paired channels to the stereo buss can also be linked to make the operation more logical and quicker. Version 2 also enhances the Solo functionality. The Aux Buss Select buttons can be configured to operate the Aux solo mode (instead of having to switch to the Master fader layer), pre-fade channel listens can operate in stereo to reflect the channel panning, and raising a fader from the end stop can automatically cancel the channel PFL solo mode.

Photo: Mike Cameron This review was carried out using the upgraded version 2 OS on the DM1000.

The monitoring section has been upgraded to allow simultaneous monitoring of stereo sources: the two external digital stereo inputs and the desk's stereo output can all be mixed together, if required. Likewise, the Buss and Slot sources can be monitored simultaneously when working in surround. The console's comprehensive bass-management facilities now incorporate THX-approved presets, giving standard settings for DVD, film and music production. There is also a new facility to reset the monitoring level to a predefined volume — nominally the cinema-standard 85dB SPL, but in practice any required house standard level. A new function has been introduced to simulate conventional VCA-style fader operation, called Fader Group Master. With this mode turned on, the level of all channels assigned to a fader group can be offset from their actual fader levels (but while maintaining their relative levels) by using the virtual group master fader. A mute group master function has been added as well, and the user-defined keys can be used to assign selected channels to fader or mute groups. It is also possible to assign the user-defined keys to switch between the various windows of the Studio Manager software remotely. The Scene Memory functions have been extended, with the ability to copy entire channel or selected parameter settings from the current scene and paste them into other scene memories. The list of parameters that can be protected with the Recall Safe function has been supplemented with the channel delay and channel routing settings, and the monitoring section has been added to the list of facilities



protected by the Operation Lock Safe function. The Automix system now has the ability to insert static mix parameters between specified In and Out points of the Automix data — for example, to insert a new EQ setting over a specified time span. A new Overwrite mode also allows all selected channel parameters to be punched in and out simply by touching the relevant faders. Finally, the v2 software can now remotely control Yamaha's AD8HR A-D converters and Steinberg's Cubase SX (as a Remote Layer target), and the frontpanel joystick can be assigned to the surround-panning functions in Pro Tools. The desk can also be controlled remotely from a video editor using the industrystandard ESAM protocol — a function that many video houses have apparently been waiting for. All in all, a lot of useful new and improved functionality has been introduced, all of it detailed clearly in a new handbook supplied in the upgrade pack. This new edition boasts nearly a dozen more pages than the original and an entire new chapter covering the new ESAM functions. Similar but even more extensive enhancements have been made to the other DM- and 0-series consoles, but rather than list them all here I will refer you to the PDF documents on one of Yamaha's many web sites which detail the specific information relating to each console — check out: www.yamahaproaudio.com/download/catalog_lib/d_mixers/ down.htm.

Add-on Effects While the v2 upgrade brings many useful benefits of its own (see the box on the previous page for examples of what installing the v2 OS did for the review DM1000, for example), its real raison d'être is the embedded support for the Addon Effects packs, of which three were available at the time of writing, with two more anticipated shortly. The three packs reviewed here are the AE011 Channel Strip package (providing two mono and two stereo compressors, plus a six-band equaliser), the AE021 Master Strip package (containing four tape-machine emulations that can be mixed and matched), and the AE031 Reverb package (which brings Yamaha's most sophisticated REVX technology to the consoles with hall, room and plate algorithms). Two additional packs which weren't yet available for review when this was written were the AE041 Surround Post package and the AE051 Vintage Stomp Box, which recreates three classic Phaser effects pedals. The Surround Post pack is intended for sophisticated film and TV post-production applications, and includes an early-reflections generator with source-position tracking, a Doppler shift system to modulate the pitch of a source moving around the surround-sound stage, and a program which allows the entire spatial field to be rotated. This optional pack cannot be used on the 01V96 console, because it has no surroundcapable effects facilities. The new Add-on effects packs are based upon three innovative Yamaha technologies. 'Virtual Circuit Modelling' (or VCM) is used for the Channel Strip, Master Strip and forthcoming Vintage Stomp Box plug-ins, and the idea is that by file:///H|/SOS%2005-06/Yamaha%20Digital%20Mixer%20Upgrades.htm (4 of 11)9/28/2005 2:36:37 PM


modelling every aspect of an analogue electronic circuit, its sonic nuances can be emulated precisely. This origins of this VCM technology are to be found in Yamaha's first physical modelling synths, the VL1 and VP1, released in 1994. The early modelling techniques were developed by a team of Yamaha engineers led by Toshifumi Kunimoto in what became known within Yamaha as 'K's Lab.' The R&D work continued, and for the last few years it has been focused on the modelling of analogue circuits (instead of acoustic instruments) with a view to emulating classic analogue signal processing. 'Compressor 276' and (below) 'Compressor 260' from the AE011 Channel Strip.

Virtual Circuit Modelling technology has now reached the stage where it can emulate every key parameter and nuance, not only of electronic components such as transistors, resistors, capacitors and so on, but also of complex inductive devices like tape heads, transformers and even magnetic tape. In fact, a lot of the R&D effort has been expended on modelling the subtle magnetic saturation effects that are such an integral element of a lot of analogue audio systems. The Surround Post pack will apparently employ another set of modelling techniques: Yamaha's innovative Interactive Spatial Sound Processing (iSSP) technology. The idea here is to produce precise simulations of real acoustic spaces by using modelling to accurately predict reflections and decays, taking into account a specified room shape, surface materials, and the directivity of the sound sources. This is supplemented with further processing which uses source position data to generate distance-related decay and pitch characteristics, providing precise imaging information and even doppler shifts as a mono source is panned around, for example. The Reverb pack uses the sophisticated REVX algorithms which were first introduced in Yamaha's top-of-the-line SPX2000 multi-effects unit. This system represents the current state-of-the-art in reverb from the Yamaha stable, and broadly equates to the top-flight systems from Lexicon and TC Electronic. Each of these plug-in effects packs is supplied in the form of a CD-ROM, with an Installation Guide. The effects algorithms themselves are already embedded in the console's v2 software, and the CD-ROM is essentially a means of authorising and enabling the relevant software components. The authorisation process requires a computer with Internet access linked to the console via USB, and each Add-on effects pack is authorised for use on that specific console via Yamaha's web site. If you want to load a pack onto a different console, you must disable it on the first console, cancelling the authorisation via the web site, before attempting to load and re-authorise it on a second desk.



I found this whole enabling and authorising process painless and fairly quick, and was able to activate and license all three current Add-on effects packs easily within 15 minutes. Once authorised, the relevant effects appear in the console's effects library (starting at position 53), and the desired effects can then be routed and applied within the console as required. Most of the effects reviewed here are probably best used as an insert into a channel, group or master, but the reverbs will obviously be more appropriately used within an effects send and return loop. Of course, it is the console's dedicated effects processors that are used for these plug-in effects, so the number of plug-ins that can be used at any one time will depend on the quantity of effects processors available in the console — four in the case of the DM1000 and 02R96, and eight in the case of the DM2000 — although all can be used at any sampling rate up to 96kHz without limitation. The 01V96 is more limited, however, offering four effects at standard sample rates but only two at double rates. Nor does it have any provision for dedicated surround effects, whereas both the DM1000 and 02R96 can run a single surround effect in addition to three mono/stereo effects. The top-of-the-range DM2000, meanwhile, can run two surround effects and six mono/stereo effects. All of the selected plug-in effects parameters can be controlled from the console's LCD screen in the usual way, just as for any other internal effect. Alternatively, they can be controlled remotely from the Studio Manager software, using either the generic Effects Editor window, or the bespoke and very attractive graphical interfaces provided for each of the new effects within the version 2 Studio Manager software, screenshots of which you can see throughout this article.

AE011 Channel Strip The Channel Strip package provides three effects — two compressors and a stereo six-band EQ — all modelled from classic analogue devices dating back to the 1970s, with the effect's name and graphical interface giving fairly generous clues as what has been modelled... but without risking copyright litigation! Each of the compressors is available in dual-channel, mono and stereo forms, giving a total of five effects in all. These appear in the Effects Library in positions 53 to 57. 'Compressor 276' and 'Compressor 276S' (dual-mono and stereo forms respectively) are modelled on an FET-based compressor with a fast, peak-acting response reminiscent of a Urei 1176 (2-76 — geddit?). The controls are typically simple, with input- and output-level attenuators, Attack and Release controls, and a ratio control that can be switched between 2:1, 4:1, 8:1, 12:1 and 20:1 settings. There is also an automatic gain make-up option, and a high-pass filter for the side-chain to restrict the amount of compression applied to powerful lowfrequency signals. The stereo version uses a single set of controls to configure both channels, whereas the dual-mono version has two complete sets of file:///H|/SOS%2005-06/Yamaha%20Digital%20Mixer%20Upgrades.htm (6 of 11)9/28/2005 2:36:37 PM


independent parameters — and the desk's own internal routing allows the dual-mono compressor to be applied to any required channels. If the input level is raised, the sound thickens up nicely in a distinctly analogue way, with a subtle but detectable saturation effect. The compressor offers plenty of punchy dynamic control, and is ideal for smoothing out bass guitar lines, 'Equaliser 601', also from AE011. helping to fatten up drum parts, and tightening vocals — much the same purposes that suit a classic 1176, in fact. Compared to the standard desk channel compressor, the 276 plug-in sounds less clinical, transparent or precise. It introduces a distinct character and thickness to the sound which worked well in appropriate situations and certainly reminded me of the classic 1176 effect. It may sound odd, but I found this compressor easier and more precise to control from the console's LCD screen. Having said that, the Studio Manager graphic is very attractive and presents the control information clearly, complete with useful analogue-style metering offering signal-level or gain-reduction displays. Adjusting the controls with a mouse on the computer screen seemed cumbersome compared to the LCD and data wheel of the console, but I dare say practice — and a larger screen than the 13-inch one on my laptop — would help. The 'Compressor 260' (dual mono) and 'Compressor 260S' (stereo) plug-ins are undoubtedly based on the old Dbx 260 VCA-based compressor. Again, the stereo model provides a single set of user controls whereas the dual-mono version has two independent sets. This compressor features RMS level detection and has an adjustable knee characteristic with soft, medium and hard options. The Ratio control offers extremely precise settings, starting at a very gentle 1.05:1 and increasing with astonishing resolution all the way to a genuine infinity:1. Other controls include Threshold, Output Level, and the usual Attack and Release times. The RMS detection and adjustable knee used here gives a very different kind of response to that of 'Compressor 276', providing for some very smooth compression effects if required. I found '260' worked well as a stereo compressor on complete mixes, and I also used it to bring out the room character of distant drum miking. It was also very controllable on individual sources, especially guitars and keyboards, maintaining control without becoming distracted by occasional transients. The final offering in this pack is the stereo 'Equaliser 601', the graphical interface for which looks remarkably like an old Neve rackmount EQ unit. This effect can be switched between Clean and Drive modes, the latter providing a much higher internal signal level which results in significant but musical saturation artefacts. file:///H|/SOS%2005-06/Yamaha%20Digital%20Mixer%20Upgrades.htm (7 of 11)9/28/2005 2:36:37 PM


The EQ provides high and low shelf bands (with switchable slopes, and alternatively configurable as high- and low-pass filters), plus four fully parametric mid bands — all with separate Bypass buttons. The mid band sections all offer up to 18dB of cut or boost, with variable Q from 0.5 to 16, and centre frequencies spanning 16Hz to 20kHz (and up to 40kHz in 96kHz sampling mode). An inherent part of most analogue filter designs is the interaction between bands, and Yamaha have ensured that this modelling emulation behaves in the same way — which you can see from the very clear graphical interface within Studio Manager (see page 183).

Two of the modelled open-reel recorders from the AE021 Mastering Strip package.

In fact, this graphical display allows the frequency response to be adjusted either by clicking on the control knobs, or by dragging nodes on the response chart itself. There is also a very useful 'Flat' button to cancel previous EQ settings. I have become rather attached to this EQ, and in many ways, it's a shame that I can only access four stereo instances of it on the DM1000 — it would be fantastic if it was available on every channel as standard! Not only is it superb for gentle musical tweaking and shaping, it also serves well for most surgical duties, and in the Drive mode it can add a welcome analogue-like richness and body to suitable sounds if the input knob is cranked up a bit. Unlike so many digital EQs, I found when running at 96kHz sample rates that this one was able to add the kind of 'air' or sparkle that is the hallmark of a good analogue EQ.

AE021 Master Strip I thought the modelling was impressive in the Channel Strip package, but I was completely blown away when I first saw and heard this one! The Master Strip pack provides 'Open Deck' simulations (effects program 58) which recreate the typical analogue circuitry and magnetic tape characteristics of four different openreel tape recorders, two kinds of tape, and two tape speeds — all of which influence the sound in important ways, of course. The recorders that have been emulated so carefully are three Studer machines, the A80 MkI, A80 MkIV and A820 (identified as 'Swiss 70', 'Swiss 78' and 'Swiss 85' respectively), plus the American Ampex ATR100 (called 'American 70' here). The two types of tape are modelled on new BASF and older Ampex formulations.



The effect is a stereo-in, stereo-out patch, and in setting it up you can choose which recorder model to use for the record electronics, and which for the replay, in addition to selecting new or old tape and 15 or 30ips 'tape speeds'. On the record side, controls are provided to adjust the virtual high-frequency equalisation, the bias level and the overall record drive, while on the replay side there are high- and low-frequency equalisers, plus replay gain. For convenience, the record and replay gains can be linked with an Auto Make-up mode to maintain consistent output signal levels regardless of the amount of drive applied to the tape. I found the differences between the various vintage recorder electronics very subtle, but changing the tape type and speed, the record drive level and especially the bias control really did alter the sound in precisely the way I would expect a real recorder to behave. There are lots of digital emulations of analogue tape recorders around, with varying degrees of success, but I found this one to be particularly accurate, controllable and easy to set up. The transient crushing effect is delicate but perfectly judged, as is the gentle response-rounding at both frequency extremes, and the finesse with which the sound can be tailored is remarkable. And if the sound quality is not enough on its own, the graphical interface in the Studio Manager is pure joy, complete with whirling tape reels, waggling meters and dented face plates! This is not an effect that should be overdone, of course, although Yamaha's are more subtle than many similar systems I have heard. When used across the main stereo output as a mastering process, where some gentle analogue rounding and warming is required, I found the Open Deck modelling to be virtually as creative and effective as the real thing, but a lot easier and more flexible to set up and maintain; all the sonic benefit without the chore of cleaning heads and rollers! Analogue die-hards will have to try hard to find fault, but digital converts will enjoy using this plug-in very much indeed.

Studio Manager v2 Improvements The version 2 edition of Studio Manager boasts several new features, as well as some worthy updates to the existing functionality. For example, it can now support multiple different hardware products, with multiple control windows open at the same time. Control windows can also be opened and closed from the console via the user-defined keys. The Patch Editor window can now be resized, physical port names can be listed and edited, and patches can be set up by using the mouse to click on the appropriate crosspoints. There is now an Effect Patch window to show the ins and outs corresponding to each effects processor. The Fader Layer window can be customised to show or hide various channel parameters (see screenshot, right), and the numeric value of fader levels can be shown. The Master fader section may be shown in a separate window, and there is now an option to view User Assignable layers and Fader Group Masters. A set



of more detailed meter options has also been added (a new Meter window shows all the meter displays at once), and you can jump to a selected channel window by double-clicking on a Channel's ID label. Other key new features include a new Automix Library window, an upgraded Surround Editor window which now supports 6.1 working, and an upgraded Effects Editor window which incorporates the bespoke interfaces for the Add-on effects packs.

The Fader Layer window in Studio Manager v2.

AE031 Reverb The final review package ports Yamaha's REVX algorithms into the 0- and DMseries consoles. There are three stereo-in, stereo-out algorithms — Hall, Room, and Plate — which occupy effects library positions 59 to 61. Each program can be comprehensively tweaked via either conventional control parameter knobs on the desk LCD or Studio Manager's generic Effects Editor window, or via the graphical display screen within Studio Manager, (shown at the head of this review). The REVX 'Plate' setting provides a brighter and relatively straightforward reverb character, while the 'Room' and 'Hall' settings recreate spacious but detailed acoustic spaces, with well-defined early reflections which create believable environments. Within a mix, these reverbs sit very nicely indeed, adding space and perspective to individual sounds without clogging up the natural spaces between instruments in the way that so many less accomplished reverb algorithms do.

Conclusions The three new Add-on effects all introduce useful and impressive features to the 0- and DM-series consoles. Clearly, given the nature of the effects and processing provided, and that the number of effects useable at any one time is limited, these plug-in effects are intended to be used mainly for final mix sweetening rather than on a track-by-track basis. The compressors and EQ all work superbly well and offer a useful range of facilities and characters, with a distinctly analogue sonic quality. The open-reel file:///H|/SOS%2005-06/Yamaha%20Digital%20Mixer%20Upgrades.htm (10 of 11)9/28/2005 2:36:37 PM


plug-in is intriguing and surprisingly flexible, and will be an instant hit with anyone wanting to add an analogue flavour to their digital productions. Equally, the REVX algorithms provide a very useful step up in quality from the standard-issue reverb effects. Useable though these older effects are, they aren't in the same league as the best, which is why I have a Lexicon PCM90 permanently hooked up to my DM1000. It's seen much less use since I installed the REVX plug-ins! The original DM1000 was a very impressive, versatile piece of equipment, and I have never found it wanting in any way. However, the v2 upgrade has tidied up several operational quirks and limitations, and the new Add-on effects further extend the versatility and, more importantly, the sound quality in new and creative ways. All three packages reviewed here are thoroughly recommended.




Q Can you explain how the 'Track Expansion Method' works?

Q Can you explain how the 'Track Expansion Method' works? Published in SOS June 2005 Print article : Close window

Sound Advice

Here's something that's been bothering me for nearly 20 years. Back in about 1986-7, when we were all taking out second mortgages to buy our Tascam 38s and Seck 18:8:2 desks, I used to subscribe to Home & Studio Recording. In the smaller adverts tucked away towards the back of the magazine, there was an ad for an amazing and inexpensive technique which, it was claimed, added many more audio tracks to your eight-track reel-to-reel tape machine. I don't remember it giving any details. My first thought was that it was some sort of sync'ing device slaved to a timecode track to give you extra MIDI capability, but there was something in the text to the effect that the extra tracks were audio not MIDI. Engineers I worked with at the time were as baffled as I was about these claims — it couldn't work! If it did, it would have been the Holy Grail of multitrack recording! So can anyone at SOS remember these ads and, even better, does anyone have any idea what it was all about? I've lost hair and marbles thinking about this, so any help would be most welcome! SOS Forum Post Reviews Editor Mike Senior replies: Several of us here at the office had recollections of these ads, so we headed for the capacious SOS archives and discovered (just on the left past the Lost Ark) a set of old back issues of Home & Studio Recording. After An original advert for the Track Expansion a little hunting, we found a series of adverts placed by Mr Warwick Method which appeared in the May 1988 Kemp of Spectrum Studios in Essex. He claimed to be able to get 13 issue of Home & Studio Recording. tracks from a four-track machine and 57 tracks from an eight-track machine, all 'without track bouncing, sequencing, or code-track synchronising'. Furthermore, all tracks were apparently retained for remixing if required. The pricing in these adverts was £500 including VAT, and for this fee Mr Kemp provided one-on-one tuition in the mechanics of the process at your studio, as well as a hardware device to assist with the process. After a fair bit of head-scratching in the SOS office, we figured that the scheme probably used some kind of submixing and synchronisation process, but we were unable to get to the bottom of how the system worked, so

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I set about tracking down Mr Kemp himself, and was surprised to find him at the same address, where Spectrum Studios is still a going concern. He explained that he had marketed his system of track expansion successfully for a number of years, and had users all over the world. The initial scheme with the hardware component (as shown in the adverts we found) wasn't that successful, and it also turned out to be a nuisance for Warwick to have to travel to studios around the country. He later refined the process into an instruction booklet which sold for a rather more affordable £49 — the hardware element, a type of monitoring amplifier not absolutely essential to the process, was abandoned. The booklet apparently did well for a number of years until the nonlinear editing and virtual tracks of digital recorders rendered the process obsolete for most musicians. So how did the track expansion method work? The main trick behind it was a method that Warwick discovered for synchronising a multitrack with a two-track master recorder — in his booklet he referred to this as the Image Shift Monitoring (ISM) approach. For it to work you need exactly the same audio to be recorded to one of the tracks on each machine. You then pan the audio playback from the two machines to opposite sides of the stereo image, and make sure that both tracks are playing back at identical levels. With the machines playing perfectly in sync this setup produces a phantom mono image of the audio at the centre of the stereo soundstage. Any slip in synchronisation between the two machines is heard as a shift in the stereo positioning of this phantom image. This stereo phenomenon occurs because of something called the Haas Effect. When you hear sound from a source off to your left, it reaches your left ear fractionally before it does your right ear, because your right ear is further away. Haas showed that the brain uses the interaural difference in the sound's arrival time as one of its tactics for perceiving the direction of the sound source. As a survival skill, detecting the source direction of sounds is very useful to us humans, so the brain is very sensitive to the very small timing differences involved. You can clearly hear sub-millisecond interaural delays as changes in stereo positioning on headphones, and this is the secret to the accuracy of Kemp's ISM. The correlation between the playback timing offset and the position of the phantom image also made maintaining the synchronisation more straightforward. As long as the two audio tracks were rigged up the right way around (it was a simple matter of experimentation to find the correct way), the varispeed knob on the multitrack recorder could be used to 'steer' the phantom image's position as if with a steering wheel, which made it fairly intuitive to keep the synchronisation tight by guiding the phantom image back into the centre of the soundstage. It required a little practice, but the knack could be quickly acquired. Interestingly, Warwick has discovered since marketing his idea that similar synchronisation methods were used at the BBC and at Deutsche Grammophon, but that they were never considered as a means of track expansion.

Warwick Kemp's instruction booklet, MultiTrack Expansion By Image Shift Monitoring, explains how to synchronise multitrack and master tape recorders to increase available tracks.

'But the advert said that there was no synchronisation!' I hear you cry. Well, to be fair, 'code-track synchronising' was what was mentioned in the advert, and there is no timecode involved in the ISM process. However, the track used for synchronisation purposes has to be selected quite carefully. For a start, it needs to play pretty much constantly throughout the track in order to keep the two machines consistently sync'ed together — the drums therefore usually make a good choice of sync track. Warwick notes in his booklet that it helps if the sync track has a fair amount of transient information and a good degree of mid-frequency content, as this makes the stereo effects easier to track. It's also necessary to have a good long count-in (at least 30 seconds) to allow you to get the



two machines exactly into sync before the music starts — at one point he even supplied a general-purpose prerecorded count-in on a tape included with the booklet. Once you can synchronise your multitrack and two-track machines, you can add extra tracks to a project whenever you like. For example, lets say you're recording a song to four-track and you've recorded tracks of drums, bass, guitar, and piano on a given piece of tape — we'll call it Tape A. Now you want to add backing and lead vocals, but you have no spare tracks. What you do is record the drums to the left track of your twotrack machine while simultaneously submixing the bass, guitar, and piano to the right track. You then dub from the two-track recorder to a pair of tracks on a new piece of four-track tape, which we'll call Tape B, leaving two tracks spare for the vocals. One might protest that this process is tantamount to track bouncing, which was specifically ruled out in Warwick's advertisement, but the difference is that you retain the original tracks for remixing at any time. To remix, you return to Tape A and record an updated submix to your two-track recorder as before. You then synchronise this two-track recording with the multitrack Tape B (using the identical drum tracks on the two machines to achieve synchronisation) and replace your original submix track with the new one from the twotrack recorder. Here's how you can extend the principle to get the 13 tracks out of a four-track machine which the advert claimed. First submix the whole of Tape A to one track of your two-track machine, and record it from there to a single track on each of three new pieces of tape: Tape B, Tape C, and Tape D. Once the resultant nine extra tracks have been recorded, Tapes B, C, and D can each be submixed to the stereo recorder using the submix of Tape A as the sync track. Finally, on a new piece of multitrack tape you can use the ISM technique to combine the four-track submix of Tape A with the threetrack submixes of Tapes B, C, and D. You then have a mix of 13 tracks coming from your four-track machine. Or, by the same method, 57 tracks coming from your eight-track machine. And of course, that's not the end of the story. If your recorder has the fidelity, you could submix your submixes to add even more tracks, the only limits being your recording technique and the noise performance of your studio — using a digital master recorder such as a DAT as the two-track machine helps with this considerably. Or you could combine this method with traditional track bouncing.

This diagram illustrates how the track expansion method would be used to record 13 tracks on a four-track recorder.

While maximising the raw track count is pretty vital when working on four-track, it isn't as important a consideration if you are using an eight-track recorder. In practice, Warwick found that he usually recorded his main arrangement on maybe five or six tracks, and then used the ISM technique to add massed backing vocals and synth overdubs on the final tracks. This approach allows you to mix down from the original reel, retaining individual control over the most important separate tracks. An alternative use for the extra tracks mimics the role of virtual tracks on a digital system, where they give you the freedom to record multiple versions of a solo for comping, to store alternate takes, or to save unprocessed versions of processed audio tracks in case you change your mind at mixdown. These are all facilities that we rather take for granted with modern digital equipment, but which were very powerful features when analogue multitrack machines were still the norm in the home studio.



The only real disadvantage of the ISM scheme for many musicians of the time would have been that a twotrack master recorder only allowed you to transport mono submixes between different bits of multitrack tape. However, this problem could be circumvented by anyone with more than one multitrack machine — just a three-track machine would do the trick. The ISM approach also has the downside that you end up using more tape, but that would probably have seemed a small price to pay for those who found themselves needing the extra track count. Spectrum Studios +44 (0)1702 354045. Click here to email Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q Can you explain the track routing in my multitracker?

Q Can you explain the track routing in my multitracker? Published in SOS June 2005 Print article : Close window

Sound Advice

I have bought an Akai DPS24, which is great, but I am having trouble getting signals to disk tracks. I previously owned a DPS12 and that was fairly simple, but the DPS24 manual talks about using 'groups' to route inputs to tracks. What is a group and how are they used? Rupert, Bath SOS contributor Steve Howell replies: This is a common cause of confusion when working with digital multitrack recorders. The mixer section of a digital multitracker typically has a number of inputs and a number of 'group' outputs to connect to the multitrack recorder. Each input channel has a collection of switches that allow you to route any one of these inputs (or, indeed, combinations of inputs) to any given group output (and hence track). So, for example, to route input channel 5 to track 3 of the multitrack recorder (MTR), you would assign channel 5 to group 3 — the signal comes in through input channel 5 and gets sent to group 3's output, which is connected to track 3 of the MTR.

The Akai DPS24's use of mixer groups is grounded in traditional analogue recording and offers great flexibility to the user.

This way, any input can be routed to any track without re-patching. However, to make grouping more versatile, many desks send signals out to group output pairs — 1+2, 3+4, 5 +6 and so on. The principle is the same except that the input channel's pan control is used to send the signal to odd- or even-numbered group outputs. Citing the above example, to get input 5 to track 3, you'd route it to group 3+4 and pan the channel hard left; to overdub the same input onto track 4, you'd pan it hard right. Paired output groups are also useful in that you can record multiple inputs in stereo, especially when recording multiple input sources. For example, you might have multiple mic inputs for a drum kit (or the line outs from a drum machine or sampler). The kick drum can go to track 1 (route the input channel to groups 1+2 and pan the input channel hard left); the snare can go to track 2 (route that input to groups 1+2 and pan the channel hard right); the hi-hat can go to track 3 (route that input to groups 3+4 and pan hard left). However, you wouldn't usually use separate tracks for every tom and cymbal. So, route the high tom input to, say, groups 5+6 and pan that some way to the left. Now route the mid tom's input to 5+6 as well but pan that central. Now route the low (floor) tom to 5+6 and and pan that some way to the right. The overheads can be similarly grouped and recorded to, say, tracks 7+8 in much the same way as the toms. If you are recording a live band with a bass

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player and other musos in tow, their instruments could be similarly routed to other tracks (again, via the groups). This is pretty much exactly how the Akai DPS24 works. To get input 5 to track 3, press the 3/4 Assign button, press input channel 5's Select button and pan it hard left; to route it to track 4, simply pan it hard right. To get input 1 to track 8, press the 7/8 Assign button, press input channel 1's Select button and pan it hard right. In a similar fashion you can route any channel to any track. Now, where it gets a bit tricky is that the DPS24's mixer only has eight (four stereo) group outputs to service 24 tracks. This is common on many mixers, and what happens in this case is that group outputs are 'doubled up', that is, each one is sent to more than one track. Let's rewind again to a more conventional studio for a moment... It was common practice (mostly in the interests of economy) to limit the number of group outputs. Thus, it was not unusual to use an eightgroup mixer with a 16- or 24-track MTR. What you would do in this case would be to connect group output 1 to tracks 1 and 9 of the MTR (and also track 17 in the case of a 24-track). Similarly, group output 2 would be connected to tracks 2, 10 (and 18), group output 3 to tracks 3, 11 (and 19) and so on. So, to get input 5 to track 11, you'd route it to groups 3+4 and pan it hard left, before putting track 11 into record. This is the principle used on the DPS24. It can take a bit of headscratching at first but the panel is labelled accordingly to assist you (plus you just kind of get used to it after a while). You might still be thinking that this is unnecessarily complicated for simple track laying but it has many benefits. For example, we have already seen how we can combine several inputs to a pair of tracks. This simplified block diagram shows a Well, the same principles can be applied to 'bouncing down' tracks, typical input channel in an eight-group that is, mixing down several tracks onto a stereo pair of tracks. For mixer such as the one found in the Akai DPS24. example, you might have 18 tracks of backing vocals (perhaps for that Queen number you're covering!). To free up those tracks, you can bounce those down to a stereo pair using the groups in exactly the same way simply by routing the BV tracks to a group pair (panning each track accordingly to create a stereo image) and enabling the appropriate MTR tracks to record. The fact that the process is consistent for inputs and tracks means that once you've grasped the basic concept, it's pretty straightforward. Groups can also be very handy during mixdown. By sub-grouping certain instruments and then routing those groups to the main L/R output you can be controlling the master level of a whole section of instruments with just one fader. All your keyboards could be submixed to groups 1+2, your guitars to groups 3+4, drums and percussion to groups 5+6, backing vocals to 7+8 with other primary instruments having their own individual channels. Thus you can control the overall level of these different instrument groups far more simply. Take the time to digest the possibilities and flexibility offered by the use of groups. It's a time-honoured practice that has been the cornerstone of 'traditional' recording studios pretty much since the inception of multitrack recording. Fortunately, in the case of the DPS24, it's all very consistent and there are no special modes or pages you have to enter — it's all available from the front panel (almost exactly like a traditional console) and once you've got the hang of it, you'll find groups your flexible friend in the track-laying and mixdown processes!



If it's any consolation, you can route any given input to any given track directly using the DPS24's flexible internal patchbay (there's even a patchbay template for this) but this requires delving around in the Mixer mode's pages. The group facility, however, allows you to route any signal (or combination of signals) to any destination in a very 'hands-on' way from just the front panel. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Q Do I need to use a compressor before my soundcard?

Q Do I need to use a compressor before my soundcard? Published in SOS June 2005 Print article : Close window

Sound Advice

If I limit using a Alesis Nanocompressor when recording into my Emu 1820 soundcard, can I still limit the final mix with, say, Waves' L1 Ultramaximizer without any problems? I thought that you should only limit once, and then you have to unlimit before limiting again. Is this true or will the Nanocompressor cause no problems? I've set a threshold of just under 0dBFS and a ratio of 100:1. The attack is fast, and the release 200ms. Soft-knee and peak-sensing modes are active. I am using the compressor to drive the A-D converters on the Emu 1820. Will this cause me any problems when limiting later to record to CD-R? Or should I just compress slightly at 2:1 with a -9dB threshold and fast attack and release times when recording at the input stage? Thomas Jay Features Editor Sam Inglis replies: I'm afraid there is no such thing as 'unlimiting'. Once you have applied dynamic processing such as limiting or compression to a signal, there isn't an awful lot you can do to reverse it. However, it's not uncommon to apply multiple stages of compression. Compressing an entire mix is also very different to compressing an individual signal within the mix, both in terms of what you're trying to do and the settings you would use. Excessive dust, smoke and moisture can lead to unreliable patchbay performance.

Some people like to use a hardware compressor before the A-D converter, if it has a specific 'sound' that they can't get later using plug-ins. However, in normal use, there is absolutely no need to use either a compressor or limiter to 'drive' the A-D converters. You only need to use a limiter on the way into the soundcard if you are dealing with wildly unpredictable levels and there is a possibility of a rogue peak exceeding the headroom you've left. In general, it would be much better simply to leave more headroom at the input of the A-D converter. The dynamic range of a modern A-D converter is huge, and there is no need to push your input signals anywhere near clipping in order to get good sound quality. So, in short, unless your input signal levels are hugely and widely variable, or you particularly like the sound of the Nanocompressor, I'd simply remove it from the signal chain altogether, and turn the preamp gain down.

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Q Do I need to use a compressor before my soundcard?



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Q Is it worth recording at a higher sample rate?

Q Is it worth recording at a higher sample rate? Published in SOS June 2005 Print article : Close window

Sound Advice

I've recorded several songs in 24-bit/48kHz. When I went to burn the CD, the application I'm using for burning (Nero) did not recognise the files. I had to convert them to 16-bit/44.1kHz first. So does it make a difference in the audio on the final CD tracks to record at the higher rates when its then converted back to 16-bit/44.1kHz? Maybe it's not worth getting a fancy audio interface after all? SOS Forum Post Technical Editor Hugh Robjohns replies: If your end format is destined for a 44.1kHz sample rate — the standard audio CD format (the 'Red Book' standard) is 16-bit/44.1kHz — there is no point in recording at 48kHz in the first place. There is no significant quality gain involved in using a fractionally higher sample rate (48kHz is only eight percent higher than 44.1kHz), and the technical losses and time involved in sample-rate conversion aren't very constructive either.

The Alesis Nanocompressor can be used to compress the signal prior to A-D conversion, but is this necessary?

I would recommend recording your material at 24-bit/44.1kHz and then truncating and re-dithering the finished tracks to 16-bit as the final stage before burning the CD. There is a useful advantage in recording your original material with 24-bit resolution, as this increases the dynamic range available to you. This translates into greater headroom and a reduced risk of overloads and transient clipping. If you want to start with higher resolution source recordings, possibly with an eye to releasing the material on high-resolution formats in the future, then you have a choice of sample-rate options. Obviously, sample-rate conversion will be needed for a CD release, and many argue in favour of recording at 88.2kHz as this is double 44.1, making the down-sampling relatively simple. In the early days of sample-rate conversion this made a significant difference to the resulting sound quality, but modern sample-rate converters appear to handle noninteger conversions with no loss of quality at all, and a 96kHz sample rate is more widely used in highresolution formats. In my opinion, there is very little to be gained in going to higher sample rates, so I would use 24-bit/44.1kHz for a CD-only release (reducing to 16-bit/44.1kHz at the last possible stage), and 24bit/96kHz for everything else. Published in SOS June 2005

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Q Is it worth recording at a higher sample rate?



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Q What kind of ear plugs should I get for wearing at gigs?

Q What kind of ear plugs should I get for wearing at gigs? Published in SOS June 2005 Print article : Close window

Sound Advice

I've been coming home from gigs recently with my ears ringing and I'm worried about damaging my hearing. I think it's definitely time to invest in some kind of (preferably unobtrusive) ear protection, but what kind of ear plugs should I be looking at? I still want to be able to hear what's going on but keep my ears out of danger at the same time. I guess I can't wear earplugs when I'm actually performing, but at least I can reduce the chances of permanent damage when I'm watching the other bands. What's your advice? Patrick Bailey Technical Editor Hugh Robjohns replies: You are very wise to be concerned and to want to do something about it. A set of ear plugs needn't cost a lot and, if you play or go to loud gigs regularly, should be considered a necessity. Hearing damage is directly related to both sound level and length of exposure. So, even if you don't want to wear ear plugs when you're performing, consider wearing them when you're rehearsing, as well as at gigs — it has been suggested that musicians often do more damage to their ears during the many hours of rehearsal than in the comparatively short time they spend on stage. I would recommend investigating the options for good-quality ear plugs that reduce the overall level of sound but maintain an even spectral balance so that you can still hear everything clearly, Some generic attenuating ear plugs (top) although the overall level is reduced. Disposable solid-foam ear and some custom-moulded ear plugs (above), both manufactured by Sensorcom. plugs won't give you this even balance and will adversely affect your enjoyment of the music. You can often find suitable generic ear plugs in the good musical instrument and equipment retailers, sold as 'musicians' earplugs', and available in different strengths (amounts of attenuation). Obviously, the greater the number of dBs of attenuation, the better overall protection they offer. However, for a really comfortable and long-lasting solution, I would recommend making an appointment with a good audiologist who will be able to take ear moulds and make earplugs to your precise specifications that will file:///H|/SOS%2005-06/Q%20What%20kind%20of%20ear...0should%20I%20get%20for%20wearing%20at%20gigs.htm (1 of 2)9/28/2005 2:38:20 PM

Q What kind of ear plugs should I get for wearing at gigs?

be comfortable to wear for long periods and easy to clean and look after. Custom-made earplugs will cost more, but considering that hearing damage is irreversible, if you value your ears the cost should be irrelevant! More information and advice is available from the RNID (www.rnid.org.uk). The web site of their ongoing 'Don't Lose The Music' campaign (www.dontlosethemusic.com) is aimed specifically at musicians, DJs, clubbers and concert-goers and is linked with two hearing protection specialists — Advanced Communication Solutions, or ACS for short (www.hearingprotection.co.uk), and Sensorcom (www.sensorcom.com) — who can produce custom-fitted ear plugs. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Q What's wrong with my patchbay?

Q What's wrong with my patchbay? Published in SOS June 2005 Print article : Close window

Sound Advice

I have a Neutrik quarter-inch jack patchbay that I'm having problems with. The unit is only a few months old but already I seem to be suffering from poor connections, with signals being quiet or not coming through properly and becoming distorted. I have isolated my outboard and tried different cables to check if the problem lies with these, and all roads lead back to the patchbay. Are there any methods for cleaning the contacts in patchbays or fixing them? The patchbays I had before never had any of these problems and I was using them for over four years. SOS Forum Post Technical Editor Hugh Robjohns replies: It is very unusual for a patchbay to become unreliable in such a short time. I presume there is no obvious environmental problem such as excessive dust, damp or smoke? Dust, damp and smoke tend to work together, gathering on the socket contacts to form a sticky residue which acts as a high-resistance layer, giving the kind of problems you seem to be experiencing. It helps if you can make sure the faceplate of the patchbay is vertical in the rack, rather than horizontal or sloping, as this minimises the risk of dust falling into the sockets, and that the room is kept dehumidified and well ventilated. Another related cause is dirty (tarnished) plugs. This used to be a real problem in professional studios using PO316 or bantam patch cords which employed brass plugs, but tends not to be an issue with the plated domestic quarter-inch plugs used in most home-studio rigs. Professional studios using brass patch plugs often use a mechanical burnisher to clean and polish the plugs, along with an aggressive cleaner for the sockets, but the equipment is designed to be cleaned in this way. The plated domestic plugs and sockets are often quite soft in comparison and will wear out very quickly if treated this way, so gentle hand cleaning with a mild metal polish or contact cleaner — Deoxit or Servisol, for example — might help. Don't get too enthusiastic though: excessive rubbing with an abrasive cleaner will quickly damage or even remove the plating, making your problems a whole lot worse! A quick wipe over with one of the gentle cleaners mentioned above every month or two should keep everything in good order, if surface contamination is the problem. Another likely problem, probably the most likely, in fact, is that your patch cables are of a non-standard size. Some of the cheaper Chinese-made moulded patch cables are fitted with locally made plugs that are slightly undersized and don't conform to the correct quarter-inch specifications. Consequently, they sometimes don't make reliable contact with some types of socket. The solution here is obvious: try patching using good-quality leads (ideally with Neutrik jacks on the end), and see if that works any more reliably.

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Q What's wrong with my patchbay?



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Business End

In this article:

The Secret Hairdresser Birdpen This Month's Panel

Business End Reader Tracks Evaluated Published in SOS June 2005 Print article : Close window

People : Miscellaneous

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

The Secret Hairdresser GN: "There's quite a mix of styles; they remind me of the 5678s at times but then there's this bit in the second track that's very like Neil Hannon."

Track 1 1.9Mb

CW: "The harmonies are actually quite clever." GN: "It's like they haven't found their identity yet. It's a real mish-mash of influences." CW: "I think it's quite good — I think it's fun, but this is a hard one for me, because it's personally not the sort of music I would listen to. My advice to them would be make the intros a lot shorter so the songs come in lot quicker — I mean I thought the first song was an instrumental! If you're specifically aiming for this genre you need to be doing something that's going to get radio play somewhere, and for that it needs to go into the song in under 20 seconds at most, 10 if you can." GN: "I think there's a lot of mileage to be had and really good music to be made from two human voices of different gender singing together — I like that aspect of it, but it seems that this band haven't nailed down where they want to go. It comes back to what we were saying earlier about writing from the heart rather than writing stuff that's derivative of other things." MN: "I find it hard to get a complete picture. It sounds like they're slightly tongue-

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Business End

in-cheek and it's a bit hard to gauge what they're going for. Are they presenting a visual thing? Would seeing them live or seeing a video complete the impression or something?" CW: "It would be good to see what they look like and see what they're like live. They've got that slightly mad sort of B52s-ish thing going on." MN: "Yeah, but with that you've had such strong melody and lyrics. I think the B52s are a very strong thing to compare it to; I mean you've got the quirky side but you've also got no-messing-around pop songs. Maybe they're on their way to something good like that. "I think with some things you get a very strong impression, like with the last CD by David Kenny [featured in last month's Business End] you get a very clear image just from listening, and with this I feel like I'm getting 60 or 70 percent of a picture. Possibly it's just about developing songs — maybe as they progress and get more eloquent they'll be able to project what they're doing and what they intend to do more strongly." CW: "Maybe they should try putting something out as a one-off on a small label like Transgressive and see how it does." MN: "They imply that they've got their own label — it's got 'Lap Records' on the sleeve of the CD. But how do you interpret that? On the sheet here it says that there's a record label already there. What does it mean?" CW: "I think someone's told them that it looks better." MN: "So on the one hand you get the impression that they're on the case and things are organised but on the other hand it doesn't feel fully formed musically yet. "I guess with this one they're trying to present it as a finished thing and saying 'will you release it please', and, if I received this, I would have to say 'No, but keep going, because it sounds like it could be possible.' I can feel that they have an idea of what they're doing — it's just not fully formed yet." GN: "If you're going to present it like this and say you're from such-and-such records you're obviously trying to pitch it as a finished thing and it's just not. It's just not focused enough. I think it could do with some more hooks; this is a very common thing you find with demos — you get bands who write really good, interesting verse melodies but if you write a good verse melody you've got to write a chorus that's going to top it. You've got to step it up because otherwise it's just not going to work dynamically."


Business End

MN: "I was reading Sound On Sound magazine this morning and there was an interview with Nile Rodgers in it, where he was saying that Chic always came up with the chorus first, and once they had that, the rest of the song would pretty much write itself."

Birdpen GN: "He's got a great voice — it's one of those voices that just stands out."

Track 1 3.3Mb

CW: "Yeah — it's not that pure but it's very distinctive and that's what I prefer. Even when he's singing on the edge of his range it sounds good." GN: "There are some really strong production ideas in the third track, it's got a much stronger sense of identity." MN: "Yeah, I think this third track is easily the most accomplished so far, both mixing-wise and production-wise." CW: "I think it's absolutely brilliant." GN: "You get the feeling that they know what they want to sound like." MN: "I think this third one is nearly great, I just think he's maybe pedalling the same feeling for slightly longer than maybe he should." GN: "You can just see that they've got the beginnings of something really good here. You can see that he's just feeding his own need to write but it also needs to go somewhere. There's a bit of Talk Talk here — those kind of bands." MN: "Yeah, Tears For Fears, that sort of thing." CW: "There's a real Depeche Mode, Bauhaus influence." MN: "They've got their own sound though. You don't have to completely reinvent everything but you do need to bring something new to the party. "The first song is quite striking and the second song has a very similar composition but the third one is a bit different becasue the melody started straight away, but then it stays there. I think it could have progressed a bit more file:///H|/SOS%2005-06/Business%20End.htm (3 of 5)9/28/2005 2:40:54 PM

Business End

and maybe done something different." GN: "This is a perfect example of what a demo should be. There's only three tracks, which is as many as you need at this stage, and each track shows you a different side of the band." MN: "It's about imagination, if they came up with a second melodic place that this could go to where he sang out and it moved then this third one could be a really engaging track. "They definitely have a direction, they just need to concentrate on the writing." CW: "If I got this in my office I'd certainly look into it more and go to see them play." GN: "With most people, they're either fully fledged and ready to roll, or you simply think 'They'll come back to me when they are ready.' With something like this, I'd bang off an email saying 'Look, I've had a listen to it, you've got something here but take a step or two back and listen to your influences again and try to work out what they were doing right — how they keep your interest.' It is just the interest factor: all the raw materials are there and there's a super voice as well." MN: "It's really well produced. There's a real sense of space but it still has a very rich sound." CW: "I really like the vocals. I always like singers who do really long, sustained notes." MN: "In many ways it sounds so accomplished. You feel like they're only a step away from being a really great band."

This Month's Panel


Business End

Gavin Nugent is the Label Manager for Double Dragon Music (doubledragonmusic.com), an independent label which has released music by Ash, The Crimea, Fi-Lo Radio and Charlotte Hatherley. Gavin's music career has been interesting and varied; in and out of bands since 1987, he went on to specialise in studio design and construction and subsequently became involved in designing bespoke music computers. He ran his own studio in Dublin for four years before moving to the UK to work for Double Dragon. Michael Nielsen joined Strongroom in the early '90s where he trained as an assistant engineer. He soon went on to become a recording and mix engineer and began producing records in 1993. His production credits to date include two Jamiroquai albums and he has been responsible for the mixing on four albums by Underworld. Lately he has become interested in 5.1 mixing and has worked on several albums (including Underworld's groundbreaking Everything Everything LP) and two movies using this format. Coral Worman is a Director of a management company for artists, producers and composers based at Strongroom Studios, and entitled (perhaps unsurprisingly) Strongroom Management. Coral has been involved in the music business for more than 20 years. In that time she has worked in A&R at both RCA and Polydor. Her long experience in artist and producer management led her to work for Orinoco Management in the early '90s and latterly to her current position at Strongroom. Many thanks to Nina Mistry and Strongroom Studios (www.strongroom.com) for organising and hosting the session. Published in SOS June 2005




DJ Format

In this article:

DJ Format

Back To The Sampling & Hip-hop Production Future Burger King Published in SOS June 2005 Playing Print article : Close window Patience People : Artists/Engineers/Producers/Programmers Going Live New Old Ways

DJ Format has been making funky, sample-led hip-hop since the old school was new — and it seems the world has finally caught up. Sam Inglis

"UK hip-hop being cool is a very new thing," says Matt Ford. "I've been doing hip-hop since the early '90s, and I've worked with mainly English people. That was all great, but none of us were getting anywhere, and no-one was really interested." They're certainly interested now: released under the name DJ Format, Ford's debut album Music For The Mature B-Boy has been one of the most successful British hip-hop records of recent years, and his second long-player If You Can't Join Them... Beat Them looks set for even bigger things. Yet as Ford acknowledges, his career only took off when he looked across the Atlantic for vocal talent. Both albums feature Canadian MC Abdominal on most tracks, with guest appearances from Jurassic 5's Charli 2NA and Akil. With Abdominal in particular, he feels he's found the perfect foil for his funky, old-school production style. "I'm not a fan of much modern music, especially hip-hop," admits Ford. "You just hear someone, and you feel that their style would complement yours and vice versa, or you can see potential, or you can pretty much say outright that it's not going to work, and you can decide from there whether to pursue it. When I heard Abdominal's first record I was thinking 'I love the way this guy raps, he's got a lot of Big Daddy Kane's qualities that I think are missing in hiphop at the moment. This really is going to suit the type of hip-hop I'm making at the moment.'"

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DJ Format

It's not only the sound of DJ Format's albums that is old-school: his approach to creating them would be equally familiar to any hip-hop producer of the late '80s or early '90s. Apart from the rapping, almost all the musical elements are spun from vinyl into two Akai S950 samplers, and triggered from Cubase running on an Atari ST. "Basically there's two responses when people find out that I use this equipment," laughs Matt. "They either think it's really fucking cool, or they think it's fucking hilarious — 'Why are you using that crap?'" Unlike many hip-hop producers, Ford's motivation for using antique samplers has little to do with their 'special' sound quality. "I do like the sound that I'm getting, but it's mainly laziness and fear of new technology. With an MPC or an [Emu] SP1200, it's nice the way they really crunch up the beats. The SP12 will take drums and does something that almost nothing else will do, it crunches them down and makes them sound great. But as much as I appreciate how it crunches them up, for me personally, the way that hip-hop has gone over the last few years is that people have taken that technology and used the shit out of it. It just seems to be all unnatural beats that are very programmed, whereas I like to make my stuff funkier and have more of a live feel to it." DJ Format beats are rarely programmed as such: Ford will usually sample a one- or two-bar drum break in its entirety, and though these are invariably chopped into individual hits in the sampler, the point is not to change the rhythm in any radical way. "I generally like to reassemble the drums pretty much as they're played. If a beat's got a nice pattern to it and a nice feel to it, I pretty much want to replicate it, but tight. To the ear, no-one's going to think 'Oh, those drums are so regimented, it's too much,' like modern-day jiggy hip-hop — they keep their live feel, if you like, but they are completely chopped up, every kick drum, snare and hihat. I'll assign different hits onto different keys and sequence the rhythm from Cubase. It's not a case of 'Yeah, OK, just take a loop,' because that just doesn't work." Much the same goes for bass lines and any other musical phrases. "The way I make music, people think that you just stick a loop on top of a loop. But of course if things have got bits of percussion and drum hits behind them, you've got to have them exactly hitting on point, or they're going to clash and sound awful. What I'll do is take the bass line, or whatever, and put it into the S950. Then I'll chop it up into as many pieces as I need to, and line it up with the metronome, and make sure that it's all in time. I'll chop it up into as many pieces as necessary, and then piece it back together again within the sampler and arrange it in the Atari."

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DJ Format

Burger King "The most valuable piece of equipment, for me, is a portable record player — and that's not an exaggeration," says Matt Ford. "When I go out record shopping, I take a portable record player. The Sound Burger is the really collectable one that everyone wants, because it's so small and the sound quality is good. You put the record in and you can actually shut it down like a burger, but to be honest, as much as sound-quality-wise they are the best, the one that I personally use is a Fisher-Price kids' one. I got my mate to fit a headphone socket where there was an electrical adaptor, and I don't want to tempt fate, but I've never broken a needle and I've never scratched a record with it. I used to waste a fortune, when I didn't have a fortune to waste, going out and buying a pile of records that looked good. Records might look good, but half the time they sound awful. I go out to shops and skip through loads of stuff until I find what I'm looking for.

A selection from Matt Ford's surprisingly large collection of portable record players, including models from Philips and Panasonic, the sought-after Sound Burger, and the modified FisherPrice model that gets the most use.

"You go to most other countries in the world, and they'll have a record player in a second-hand record shop, and they'll allow people to use it within reason. I don't know what it says about our country that that's literally unheard-of, and if you do take a record player in, sometimes people will let you use it, but a lot of the time not — especially in London." And what of that time-honoured source of sample fodder, the charity shop? "Charity shops aren't as fruitful as they used to be. I think it's something that everyone's on to, which makes it very difficult to find anything really good. I do occasionally do charity shops, but it's more odd record shops around the world."


DJ Format

Playing Patience The skill in this style of production is not so much to do with technical wizardry as it is with judging what works with what — and, of course, in shopping for records in the first place. "There's an element of skill, but it's a lot about perseverance and patience, and a lot about luck," admits Ford. "You've got to have ideas in your head, and you've got to have the vision to think at least 'I should be looking for a guitar, or a bass, or some drums,' but ultimately it's more about patience. You've got to have the patience and perseverance to dig out all these interesting records in the first place, and you've got to have luck to find something good, because you can look for hours and get nothing. And then when it comes to putting stuff together, it's exactly the same. You can have the best intentions and the best ideas, but unless you actually have the perseverance, patience and luck to try the right things and make them work... Obviously it can vary, but I might play 100 records or more, just looking for one sample. "Some records are utter shit but with one amazing track on, and with those it takes me two seconds. All I do is go straight to the good track — 'How does it sound? Does it work? No' — and I move on. With other records, there's so much good stuff on them, you might be trying stuff, and spinning it back and pitching it up and down, and you can spend hours adjusting and trying things out. Do I sit down and listen to an entire record from beginning to end? Never. "You have a certain mentality when you're looking for certain things. You're listening out for some drums or a guitar or a bass that really jumps out at you. You just have to skip through and do what you can. The classic place to find a drum break is the start of a song, but you can always miss stuff if you don't skip through properly. It's the same with all the instruments. You'd never look through a record without checking the start of every song."

The mainstay of DJ Format's studio: two Akai S950 samplers and an Atari, running Cubase.

A typical DJ Format track will usually begin with a single sample that happens to catch the ear. Ford takes as an example the semi-instrumental 'Black Cloud', from his new album. "I found this bass line on my travels, and I thought 'There's some potential in that. That's a song right there, I've just got to find the other bits to go with it.' So then I'm digging around, same as usual, and I find a guitar twang and some little bits of percussion on another record. I'm listening to it thinking 'Hey, that might work with that bass line' — it's literally as simple as that. Obviously, you try stuff, and nine times out of 10 they're in the wrong key, but sometimes they're either right or you can make them right. And that was how that song started out. The drums are always easy, I've got millions of drums. It was a case of 'I've got the bass line — Oh my God, this guitar fits perfectly with it.' So I started by putting those two together and making them fit. "From there, I'm not some great musician, I don't think 'What the song needs now is a flute!' I just play around with stuff until I find something that fits. What I'll do is I'll have the track playing and then I'll be going through some horns and stuff, and if it's not in the right key, I'll try pitching file:///H|/SOS%2005-06/DJ%20Format.htm (4 of 6)9/28/2005 2:40:58 PM

DJ Format

it up or down on the deck. Sometimes it works, sometimes it doesn't. I'll sample a longer piece than I use. The way that I do my music, I want it to sound natural, like it was played that way, I don't want to play it like it's an obvious sample." The original, central sample will usually dictate the tempo for the track. "If you've got a bass line or a guitar, you don't want to start fucking around with it too much, or it's going to lose how it sounds. I've probably selected that particular bass line or guitar because I love how it sounds the way it is, and I don't want to change it too much, I just want to merge it with other things. You can mess around with drums a bit more.

Ford's records are organised into loose categories, depending on their sampling potential: "These are all records with really nice drum breaks on," he says, indicating one of the many racks. "I always keep drums separate. These are drum breaks and bass lines for up-tempo, B-boy stuff, and this pile here is similar, but more '80s old-school sounding drums — still up-tempo, but 110115 rather than 120-130 bpm."

"The good thing about sampling is that sometimes you do things that musically aren't correct, but to the ear, they work. A musician would never play, for example, some of the things over 'Black Cloud', but to the ear they sound all right. For instance, I'm sure that the guitar that comes in halfway through is not quite in key. That's one that I spent so long fine-tuning in the sampler, just changing the pitch. I chopped it into different pieces and had to tune each one slightly. You might have to tune the second half of the sample lower than the first half for it to work in the track. You can do little time-stretches with the Akais, but it's very primitive. If you've got something really slow and you want to speed it up, you can probably get away with it, but if you've got something fast and you need to slow it down, forget it."

Going Live At the time of writing, DJ Format and Abdominal are about to embark on a lengthy live tour with labelmates Little Barrie. The live show is, it seems, as old-school as the records: "Just two turntables and a mixer, and if I've got Abdominal or D-Sisive, they're on microphones, it's as simple as that. I'll cut dub plates on vinyl — I don't want to sound snobby, but it's the hip-hop way. I'll do whatever's best for the show. I'm the DJ, but I'm featuring these rappers, and if I go out and present a live show, a lot of the show is them at the front and me at the back. The least I can do is be a good hip-hop DJ and play it off vinyl. That's just how I was brought up — hip-hop, vinyl, decks. We might do a basic instrumental, or sometimes we'll do a special mix where I'll add some bits in, or on a dub plate I'll add a few things at the end of the vinyl so you can scratch it at the appropriate places."

New Old Ways These days, the S950s and Atari have been joined by an Apple Mac G4 running Logic, but it's still the hardware samplers that do most of the work. The Mac is fired up where long samples are required, or where something — like the string sample from 'Black Cloud' — needs to be subjected to a more extreme EQ treatment than Ford's Mackie desk can provide. "It's still really file:///H|/SOS%2005-06/DJ%20Format.htm (5 of 6)9/28/2005 2:40:58 PM

DJ Format

important to me to use a desk when I'm doing the initial songs," admits Ford. "I've never quite been brave enough to go over to getting a laptop and doing it all internally. I do like being able to stand there and move controls, and I do still use my old ART effects unit." The Mac also comes into its own when it comes to marrying Ford's backing tracks with each rapper's vocals. "I'll have given them the song in the post on a CD, whether it's a loop or a pretty much finished track," explains Ford. "They write to it, we discuss it on the 'phone, I get to the studio, they lay it down. Obviously I'll have a bit of input, but ultimately I like to let people pretty much do their thing, within reason. Then I'll take the Pro Tools session disk and either take it to another studio, or mix the vocals at the studio and bring back an a cappella for me to work with. I could have done the entire song on the Each of the Akai samplers has eight outputs, Atari, but when I get back, I need to put the a routed into this Mackie mixer. cappella into the Mac, put the entire song in and start rearranging it, deciding where the cuts are needed. That's where the Apple Mac comes in, especially with doing scratches — unless you want to do your scratches live, first time, no mistakes, you've got to record on a loop until you get it right." Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Recording Daúde, Neguinha Te Amo

In this article:

Forced To Mix Taking On The World A Moving Experience Back To Bass-ics Not So Hi-Fi Cheaper Than Chips? Covering Costs

Recording Daúde, Neguinha Te Amo Will Mowat Published in SOS June 2005 Print article : Close window

People : Artists/Engineers/Producers/Programmers

Will Mowat made his name first as a sequencer expert and then as a writer with Soul II Soul. As a world music producer, his recent projects provide inspirational examples of how to combine performance and programming and overcome the limitations of a bare-bones budget. Paul White

I first met Will Mowat over 15 years ago when he was helping Sound Technology to demonstrate and support a new piece of Atari sequencing software called C-Lab Creator. What I didn't appreciate at the time was just how busy he was as a session musician. One of his session jobs was for Soul II Soul, after which began a long period of involvement Photos: Richard Ecclestone. with the collective, culminating in Jazzie B asking him to join them fulltime. Will ended up doing a lot of the writing for the band, though he also got involved in what might more broadly be termed production, working alongside Jazzie. For the last few years, however, he's focused his energies on world music, recording and producing a string of acclaimed albums by artists from South America, Africa and Europe. "People tend to assume that I produced for Soul II Soul, but that's not the case," explains Will. "I was a kind of Music Director for the time I was there, which was from 1990 to 1993, which meant I worked as Jazzie's co-writer, which meant writing for James Brown as well. On Soul II Soul's third album Just Right, I composed most of the music on it. Of course a lot of what I did strayed into production territory — the demarcation was necessarily very fuzzy — but Jazzie file:///H|/SOS%2005-06/Recording%20Da%FAde,%20Neguinha%20Te%20Amo.htm (1 of 11)9/28/2005 2:41:02 PM


B being the main artist and the man with the clout and also with the vision of what he wanted, there was of course only room for one producer. It didn't occur to me to be producer, though I was absorbing the whole ethos of production all the time. When my first production job came along, I just continued doing what I was doing, but this time I was responsible for the whole project! An aspect of production is the mental shift you go through, from 'only' being engineer or musician, to producer. Most of the things that have happened in my life seem to have happened organically, against my better judgement!" Will's Creator and Logic expertise earned him the reputation of being a technology guru. As his confidence as a producer has grown, however, he finds himself making fewer demands of his equipment. "Intrinsically, I really don't like technology. I'd rather be walking in the hills, but technology just happens to stick to me like a burr, and I seem to be able to get my head around it even though I'd rather not! Will Mowat's studio room, with Rhodes MK80 and Oberheim OB8 keyboards. When the C-Lab opportunity came up, I took the attitude: in for a penny, in for a pound — and I actually grew to like working with it. Increasingly though, you realise that what's important is not the technology but the 'blood and guts' relationships between people — the quality of the musical vision and ideas. If the fundamental structure is there, everything else falls into place. As I've grown, I've got better at what I do and used less and less technology to do it. "When I started playing in bands I only had a Korg 700S synthesizer, and went on to get paid for selling and programming Oberheim and ARP synths, which was great fun — I got to work with U2 and New Music and Gary Numan. You knew where you were since the technology was limited. The parameters were narrow and well defined, but you pushed the envelope all the time and extracted maximum energy out of what little you had. Jaz Coleman from Killing Joke was brilliant at that, turning his Oberheim OBX into a screaming machine! As the Atari, then the Mac, took over from the mid-'80s on, those parameters have widened more and more. You now have so much flexibility, you can do so much, that there is a danger of losing focus and of using technology because it's available, rather than because it is necessary for the job in hand. You use plugins because you can, not because you need to. That's where technology and me part company. I can do most of the things that I need to do just using an old beige Mac G3 tower running Logic 6. If I need any other bits of technology, I'll hire them in — I don't actually have to own them any more. If anything, I'm using my experience to get the desired results rather than relying on plug-ins. "When I started producing under my own name in 1992 with artists such as Angélique Kidjo [the album Ayé with the hit 'Agolo'], I had upwards of 60 tracks because I felt it highly important that little shaker should be doing what it's doing, but as you progress, you realise that people are only listening with half an ear

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anyway and if you take out half the stuff, the mix actually sounds much better! You have more space, less is being crammed into those two little speakers left and right, and the fun and games now is to see how little you can get away with to produce your album. The people involved in the production of Norah Jones have got it absolutely right. You have brushed drums, double bass, her voice, some electric guitar lazily noodling around and piano, and that's all it needs."

Forced To Mix "One of the problems with today's shrinking budgets is that we're all being forced into roles that we don't want to do," muses Will Mowat. "I don't want to be a studio engineer — I loathe most engineering with a passion. I love recording vocals because once everything is set up, it's a blood-and-guts thing between you and the vocalist, but I'd really much rather have an engineer do all the other stuff. But I just can't afford it now. The labels are wise to this — they know you have a studio and the right software so they pressure you into recording it yourself. I'm only glad that there are mixing and mastering engineers around who can take the stuff that you've done and work with it, because that is something I won't touch. When mixing I always work with an engineer, and I cannot wrap my head around mastering so I always go to people who I know are good at it and can extract the very best from your mixes, but then you still have to give them good stuff to start with — they can't make anything out of a sow's ear! "Fortunately, because we had plenty of time if not money on the Daúde project, we were able to do a mix in Pro Tools and then leave it for a while before listening to it. There's nothing like time for telling you what a track needs. Sure it's important to do things in the heat of the moment, but if you can do a lot of work in a short space of time while working under pressure, then leave it for some time before coming back to it, the track will almost give you a shopping list of what it needs. And one of my tips is to sit down when you first play a track after a period of not hearing it and write down any problems as they occur to you. Doing it on the second play-through is too late as you'll already have got used to the sound — your psychoacoustic perception has already made allowances. You make your list and then stick to it. Then, if you get the opportunity, fix the problems and then listen again a few days later. You can do this when you're working on an album as you can cycle around the various tracks at different times."

Taking On The World As a producer in his own right, Will Mowat has specialised in world music. "We whites have stolen enough black musical ideas to last quite a few lifetimes," he explains. "In Soul II Soul there was a feeling that whenever the black community came up with a feel or an idea, the white-dominated industry was very quick to take it and put their marketing muscle behind the white equivalent. It was the same with blues where Eric Clapton and his contemporaries made the megabucks and all the original black players were sidelined. Perhaps that's why I've spent my entire production career 'bigging up' the Third World, or Southern Hemisphere. I've worked in Africa and South America — and Glasgow. Well, that's the third world as well..."



Among Will's recent projects is an album called Daúde, Neguinha Te Amo, recorded with Brazilian singing artist Daúde (pronounced dah-oo-jee), which is a good example of his general production philosophy. "My approach is based on deconstruction — not accepting things as they are. If you are given a song to do, you don't accept it at face value but rather pare it right down — skin it right down to the bones and the sinews, then you pull the skeleton apart before reassembling it. You roll up your sleeves, get dirt under your fingernails and really get to work on it by trying to understand what the song is saying and what the artist has to say. "Most of the tracks on the Daúde album are either extremely old Brazilian songs or songs that haven't been played for some time. Most people probably wouldn't recognise them until maybe halfway through because I've completely changed them in the process of finding out what I could say about these songs that hadn't been said before. They had such strong skeletal structures that I was able to be quite invasive with them and still come up with something meaningful. "On this particular project, the artist came to me with old vinyl and motheaten cassettes of these old songs so my first job was to spend time listening and understanding what the harmonic structure was. Then I thought 'What can I extract from the original that will be useful — is there anything I can sample or use?' It might just be an obscure timbale roll or a weird noise that acted as a catalyst to get me interested in the track. Sometimes I'll replace these sounds later but occasionally I'll keep the sample if it's not an important part of the original recording. "At the time I started this project in 2000, I was also into UK garage, so I chatted to people in my local specialist outlet and bought a load of vinyl albums to see what they were doing. I deconstructed some of these in Recycle and analysed the rhythms before using my own sounds to Mowat has been at the forefront of construct a rhythm that was maybe computer-based recording since the 1980s, based on that Recycle template. I when he was a product specialist for Cwanted the album to have an 'up' Lab's Creator on the Atari. garage feel, but it had to be feminine, cool, dancy, warm, Brazilian, African, urban, sophisticated, accessible, highly produced yet organic, and so on. So I wrote this mission statement and pasted it to my wall and stuck to it. I've worked on albums lasting a year, recording in Johannesburg, Durban, London and Paris, file:///H|/SOS%2005-06/Recording%20Da%FAde,%20Neguinha%20Te%20Amo.htm (4 of 11)9/28/2005 2:41:02 PM


and if you haven't got a mission statement to work to, you're going to wobble. You need those self-imposed guidelines, where you say 'That's what I'm going to do with the album.' "So, I may have my Recycle template going in Logic with some samples running on EXS24. And what Soul II Soul taught me was not to be afraid of using a onebar loop through a whole song — don't be too clever. A lot of my stuff is one constant undertow throughout, but you probably wouldn't know from listening to it as on top of that is live percussion and perhaps a live drummer. Give me a rolling groove and I'm happy! "It's also a good idea not to add the music too quickly — and don't bring the musicians in too early because you don't want them to hijack your track and take it away from your direction. I also like using the unexpected, for example, analogue gear like the Oberheim Matrix 1000. When you first hear a lot of the sounds in that box, you might think they're unusable, but when you play them into the mix, somehow they fit. I even use my old Korg M1R where the sounds are complete rubbish for modern productions, but there are certain things on there that work. I've also used a fair amount of the World card from the JV1080. Even if the part only acts as a placeholder that you take out when the percussion player comes along, you know what that part is going to be doing. "A few chords are usually enough for the singer to pitch to, and I always approach the vocal recording as if it's going to be the master, never as a demo, because nine times out of 10, they won't better that first take. In this case we used an AKG C12VR through the very wonderful Avalon 737 preamp. Once they've sung, you take all the music away and deconstruct the chords to see what you can add to the song harmonically that wasn't in the original version. Sometimes you can juxtapose a melody with what appears to be a jarring harmonic structure underneath it and come up with something weird but that makes sense. I did a lot of that on this album and a lot of the Brazilian standards on the album would be unrecognisable to Brazilian purists. "Then you start to manipulate the sounds or add retro things, like the M-Tron vibraphone and some of the M-Tron string phrases. Am I the first to have found a use for the entire phrase in the library? I didn't build the song around the phrase — the phrase happened to fit the song. I don't hunt for sounds — if I find myself trying more than 15 patches, it just isn't the right day! If it doesn't come easy, it probably isn't right."

A Moving Experience Like many of Will's projects, Daúre's album needed to be easily transferable between studios. "I didn't want to go above 24 tracks simply because I didn't know where I'd be recording next, whether it would be a studio in Rio de Janeiro or São Paulo or my own facility in London, and so I wanted to make sure that my project could fit into any Pro Tools system I was likely to come across. My file:///H|/SOS%2005-06/Recording%20Da%FAde,%20Neguinha%20Te%20Amo.htm (5 of 11)9/28/2005 2:41:02 PM


original pre-production recordings were made in Logic, but once all that was recorded, I pared it down to a manageable number of audio tracks. As it happened I did go over 24 audio tracks but I needn't have done. "I started the album in London with Daúde back in the summer of 2000 and everything was done in Logic up to the mix. I always start new projects in Logic; I feel Pro Tools is unfriendly as a creative tool, compared to Logic. It's a wonderful tool for boffins and engineers and people who are into lots of processing but that's never been my bag. I'm more concerned with the philosophical and musical side of things, though I very much appreciate Pro Tools for certain things it does for me as a user, and its graphics are second to none. So, I tend to start work in Logic and then transfer to Pro Tools. I work as long as possible with real-time MIDI until the time comes when you have to nail your colours to the mast and say 'OK, I'm happy with this performance, this Moog bass sound is working well so I'm going to record it as an audio file and work with that from now on.'

Rackmount synths in Mowat's studio include a Studio Electronics MIDImoog and an Oberheim Matrix 1000.

"The album took a long time because it was a speculative venture and I had to fit it in between other paying projects. I treated Daúde's project as my baby as I really wanted to do it and she was absolutely convinced about her vision. She chose the repertoire, so it was just a matter of shoehorning it in between other projects. A label might call me up from Africa or Brazil and ask me to produce something for one of their artists, so working around that, we did a month in 2000, a month in 2001 and a month in 2002. I was financing the album and so couldn't risk that much of my own money. Two of the artists I produced during that time were Brazilian stars Daniela Mercury, and then latterly the brilliant Chico César. "Even in 2002 I had a list of things that I thought I wanted to improve on Daúde's album, but by that time the idiosyncrasies in the performances were so ingrained in the project that fixing them wouldn't have made it sound any better or have helped to sell any more records. It wouldn't have made me feel any better about the project either, so in the end I ignored just about everything that had been on my original list to repair." The album has been released on Peter Gabriel's Real World Records to extremely positive reviews in the press. It combines obviously studio-manipulated sounds with loops, acoustic percussion and some great vocal and instrumental performances to create a very sophisticated and yet easy-listening, almost lounge, feel. "My approach is to try to create a marriage of organic and technical," explains Will. "I call it 'technorganic' and if you listen to Soul II Soul and the stuff I was file:///H|/SOS%2005-06/Recording%20Da%FAde,%20Neguinha%20Te%20Amo.htm (6 of 11)9/28/2005 2:41:02 PM


writing with Jazzie, it was all done on the Atari with Notator, using samples from Akai 1100s, but we also had live musicians. Of course we didn't get into the situation of cutting audio recordings up into segments and manipulating them as we do now, but there was definitely technology in play. Nevertheless, it was very much a feel thing and for me the important elements were the message, the song and the groove. I learned a huge amount through that, and my aim has always been to end up with something where you can't tell where the joins between real and manipulated are."

Back To Bass-ics "On this album, budgets being what they are, we had to cut our cloth to suit our pockets. I had to close my parameters right down to achieve maximum bang for minimum bucks. I wanted the highest possible quality of artistic, musical and sonic quality for the minimum outlay. I managed to do that by spending as little time as possible in paying studios, and doing as much work as I could in my own studio. I also minimised the time working with musicians, who naturally had to be paid. So rather than get somebody in the studio all day trying out things until they got it exactly as I wanted it, I'd record less and then spend a lot of time cutting and rearranging what they'd played. So, there was Logic with all its plug-ins, some MIDI instruments, the bass line from my 10-year-old MIDImoog... Originally the Moog bass was just a placeholder to give me something to work to until I could replace it with a really good bass part played by a top funk player from São Paulo or somewhere, but all the while Daúde was saying that the Moog part sounded fantastic and that I ought to keep it as it was. I argued against the idea, saying that it was electronic and wasn't appropriate, but in the end she was right. I just had to play it to a few people to confirm what she was saying. Then I had to find a mastering engineer who could deal with that kind of sound! "If I'm doing a Moog bass part, I always use just one oscillator, never two, because I don't want any of that beating stuff that can play havoc with acoustics and frequencies. It would typically be based on a square wave or a pulse just off square, very careful filter and envelope settings, but the most important thing always is getting an appropriate part. My bass lines are never just root notes — they're melodies in their own right.

Although Mowat now works mainly 'in the box', some of his outboard gear still hangs on in there: from top, Swissonic AD24 A-D converter, Lexicon Alex and Alesis Microverb reverbs, and Focusrite Platinum Mixmaster compressor.

"There's also quite a lot of sampled stuff on the album, but these were all samples that were played for me for this project or ones that I already had from previous, relevant, projects. These would be cut up as audio files or used in Logic's EXS24 sampler, and before the



EXS24, played via an Akai sampler. I've never bought a sample CD. It's become a bit of a mantra now, but the truth is that I didn't need to because I was always working with artists who had their own scratchy records or whatever. I've always been involved in productions where there's been a budget for musicians to come in, and being environmentally minded, I like to recycle ideas, sounds and samples, albeit in different ways each time. I do have my trademark motifs — there's a kind of slur I do on my Moog bass that crops up in several of my productions — but the way I see it is that it's an organic progression from production to production. There may be different countries and different artists, but if they're involved with me, then they're also involved with my musical history. "I also set out to extract the maximum energy out of something. For example, what's the point of having a really nice bass line if it's only going to be used on one track and is only going to be listened to once? If you have a particularly nice bass line or groove, the chances are that most people in the world won't have heard it, so why not use it again in another context? Sometimes I'll do this consciously, sometimes unconsciously, so that my productions have a certain joined-up continuity to them. I'm not imposing anything on the artist, but rather just bringing in my personal instinct."

Not So Hi-Fi Will Mowat is scornful about the potential benefits that new formats such as SACD and DVD-Audio might bring for musicians and producers. "More and more artists are engineering themselves — and the technical expectations have become lower, even though we've never had access to such high-quality audio technology! You can get away with recording a number one hit in your bedroom on a Minidisc, so I find it hard to come to terms with this split between SACD and all the other esoteric formats at one end of the scale and the MP3 culture at the other. Most people listen on crap systems, maybe on MP3s, maybe on a hi-fi where one speaker is out of phase and facing the wrong way on a shelf — the other speaker is on the floor and has got a biro stuck through it! Or they're listening in the car where the first 200Hz are dominated by the engine noise, or on cheap headphones. "It just doesn't make sense. There seems to be a schizophrenic approach by the recording technology industry — it's not driven by the needs of the marketplace. On one hand the industry encourages us to think we need high-end tools to achieve the best quality. On the other, the audience doesn't want to pay for the music, and anyway is willing to sacrifice quality in the interests of getting it cheap or free. The market needs fairly traded, accessible music that people will want to pay for, and we're not going to achieve that by being distracted by the likes of 24bit, 96kHz or surround sound. SACD? What SACD? Classical music sales have never been so low. Yes I appreciate if you A/B a 16-bit CD against one of these new high-end formats, there is a difference, but it really won't make your life any better and doesn't address the real problems. I feel very strongly that the whole music technology industry is not coming to grips with its real structural problems and the labels are, at their peril, chasing the fast buck and neglecting the longterm health of the market. I think surround sound is a red herring, a complete irrelevance to our end of the market where music is made on minimal budgets. Our concern is whether we can sell the CD, let alone a DVD!"



Cheaper Than Chips? What is particularly impressive about the album is that it was completed on a budget of under £20,000. "That figure includes lawyers' fees, flights and everything," asserts Will. "The budget dictated what we could do, so as I was paying for this album and there was no guarantee that it would be picked up by a label, I had to ensure I used my money wisely. That meant I had to do a lot of the stuff myself. When I hired a musician, I hired them for as little time as possible in a studio that cost me as little as possible, and I had to do deals everywhere. I had to play God with everyone's performances simply because the budgets forced me to. "Musicians are savvy now — they know you are working digitally so their expectations of what they have to do are lower. They know you can chop and paste fairly quickly, and I get the impression that some musicians have become very lazy. They expect you to manipulate what they've done after they've gone home. In this case, however, I simply didn't have the money to spend in the studio for a long time perfecting their parts, so I might record three or four tracks of a trumpet solo and then say 'That's fantastic, thank you very much!' The more honourable ones might question whether they'd given me enough for their money, but I knew that I could edit the part and come up with exactly what I wanted: it is still their performance, and I always make sure the musician is properly credited for it.

A Soundcraft analogue mixer connects the different components of Mowat's studio.

"Even if you've recorded something where the timing or pitch is cruddy, you might still just find a piece that has that magic 'X' factor and works. In fact, on Daúde's album I was working on one session where the trumpet player played two notes, one after the other, which to him sounded abominable, and right away he stopped playing and asked me to erase them. But I was jumping up and down because knew those two notes had captured exactly what I wanted for the track. Now when we do the live show, the poor trumpet player is going to have to learn all these new parts that I've created by editing his studio performances. But he is still credited with the original performance, and I like to think that what you hear after my editing is what the player would have done himself if I'd had the budget to have him in the studio for long enough. You could argue that slashed budgets do sometimes bring out your creative side! "In a pop production, I think it's important that the musicians should be proud of what they've done, but at the same time they're not here to show how well they can play; their role is subservient to the main thrust of the album — the product. file:///H|/SOS%2005-06/Recording%20Da%FAde,%20Neguinha%20Te%20Amo.htm (9 of 11)9/28/2005 2:41:02 PM


The artist and the production come first. So yes, we'd spend maybe an hour each recording the trumpet, clarinet or percussion parts for a couple of songs, then I'd spend literally days manipulating those performances, after which I'd leave them for a few days before listening to them again. "To make an album of this quality for what amounted to around £12,000 excluding air fares, expenses, lawyers' fees and so on is quite an achievement, and with hindsight it might have been done for even less than that. Even so, I didn't want to compromise on quality and I wanted to have good mastering, but even then I cut a deal with this brilliant mastering engineer in São Paulo and got a fantastic result. It was really down to getting a good balance between the various demands — it would have made no sense to cut corners recording the album, then have to pay a fortune for a mastering engineer to try to put it right."

Covering Costs When margins are so tight, cost is a constant worry. "You just have to hope that you cover your costs during the making of the album and then hope that it's going to be taken up," explains Will. "I try to structure a deal where I have a share of the synchronisation fees, so if Adidas or somebody else comes along and wants to use the track, we've made some money. It's a gamble, but if you're not a gambler in this business, you shouldn't be doing it. "There isn't a huge amount of money in this for me, but the artist comes away with an amazing-sounding CD that the label is pleased with. What's also important for this section of the market is that they can go away and play live. Though live performance may seem a quaint notion in the UK, there's a very healthy live music market outside the British Isles, where you can earn a hundred times more from playing live A truly vintage drum machine: Roland's CR78. than you ever get from your CDs. Very few world artists make much money from their CDs — they make it from live performance. In Brazil, the price of an album is four pounds or less, so nobody makes any money out of it unless you sell a million. Piracy is also an issue of course, and if you have any kind of name, as soon as the album is released, it's cloned and sold on the street at a quarter of the price. Live music can't be pirated so live music rules out there, and the CD is more of a calling card." This applies to Will as producer as much as to the artist: "Unless I'm doing something that is specifically speculative, I know that people high up in the business are going to hear what I do. It's effectively part of my showreel, so it's worth waiting for the right collaborations to come along. file:///H|/SOS%2005-06/Recording%20Da%FAde,%20Neguinha%20Te%20Amo.htm (10 of 11)9/28/2005 2:41:02 PM


"As a producer, the first test you have to pass is the one with the record label — what the fickle buying public is going to do after that, you don't know. I'm much more involved in this Daúde project because I'm also the production company, so as well as having the label on board, I also want the reviews on board. So far, that side of things is panning out OK, after which you have to be sure the artist can cut it live — and Daúde is very good live so we're OK there. After that, you can just hope the record-buying public are going to go for it..." www.realworldrecords.com/daude Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Rock Of Ages

Rock Of Ages Paul White's Leader Published in SOS June 2005 Print article : Close window

People : Industry/Music Biz

Despite the UK record industry reporting a particularly good year, the general consensus is that the worldwide record business is in something of a decline, due in part to piracy in all its forms, but also I think due to it still largely ignoring the middle-aged market where most of the cash and the serious musical interest is. It's OK for rock stars over 30 (or even over 60) to go on selling records and make a fantastically good living, yet the record industry doesn't seem prepared to take the risk of signing anyone new who's old enough to go out without their mother! Most musicians don't reach the peak of their performance capabilities until they're in their 30s or 40s, so what we are really being subjected to is a market dominated by talented beginners. By the time they've matured, the record companies will almost certainly have scrapped them in favour of the next teen style icon. But is part of the problem that musicians and their agents, like film actors, somehow expect to earn a lot more money than 'normal' mortals? If great musicians were hard to find, perhaps there would be some justification for this way of thinking, but in reality, there are probably thousands of very good musicians and songwriters for every act that gets into the charts (and hundreds of thousands of not very good ones!), yet we rarely get to hear them. And don't even get me started on TV reality pop shows — how many of the mega-huge pop icons of our age would have even made it through the first round? "I'm sorry Mr Dylan but that rendition of 'Bridge Over Troubled Water' sounded just a bit nasal, your pitching was a touch off and we can't allow you to play your guitar as that's not the format we have in mind for the programme!" Even the Beatles wouldn't have got through the first round playing by these silly rules. In most industries where there is gross oversupply of product, prices come down, yet in the music business, a small minority earn silly money while the rest get virtually nothing. For the cost of developing one totally anonymous-sounding gel-haired crooner, it should be possible to pay sensible wages to 10 equally talented people or, alternatively, to drop the price of recorded music to the point that piracy is no longer a desirable option. While the supermarkets often discount chart music to around the £10 mark or less, 'serious' music often still retails at £15 to £18, even when some of it is extremely old back-catalogue material. Depressing? Maybe, but if the route to rock stardom looks difficult, what about those thousands of students on music technology courses who want to be studio engineers? Every year, many thousands of people take these courses, and a large percentage of them want to work in professional studios, but there really aren't that many big-league studios left, mainly because people like you and I have decided to go it alone and record most of file:///H|/SOS%2005-06/Rock%20Of%20Ages.htm (1 of 2)9/28/2005 2:41:04 PM

Rock Of Ages

our work at home. On top of that, many successful musicians also have their own private studios, and so spend much less time in commercial facilities. Of the big studios, each one probably deals with dozens of successful bands and artists each year, so it stands to reason that the total number of studios must be well below the number of successful artists currently recording. What this boils down to is that while becoming a houseshold name in music is incredibly difficult, it is probably still some 10 to 20 times easier than landing an engineering job in a first-division studio. What's the answer? We followed the home recording revolution so now we have to carry that DIY ethos through to creating good records and marketing them as best we can. Some are already doing it successfully, although as with any artistic outlet, some projects will be more commercially viable than others. Either way, it's no good sitting at home and waiting for the record companies to come and find you, because that's no longer the way it works — if it ever did. If the world won't come to your music, then you have to find a way to take your music to the world. Paul White Editor In Chief Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Sounding Off

Sounding Off Tom Flint Published in SOS June 2005 Print article : Close window

People : Sounding Off

The correlation between pottery and soft synths. Seriously. Tom Flint

Although the music technology industry creates a lot of innovative new products each year, surprising numbers of them are designed to look like well-known designs from the past, even borrowing their names from the models they imitate. Those that have a retro look also tend to be marketed as having 'vintage' qualities. The motivation for manufacturers to make their products look old is to attract us, the consumers. We have a fundamental mistrust of new things — even though we are fascinated by novelty and progress! Anything displaying the characteristics of a triedand-tested product reassures us that it is likely to perform in a reliable and accepted manner, because deep down, we believe that the past (from the late '50s on into the '70s) was a time when people were more honest, and when things were made with more integrity and care. Something that is entirely new, on the other hand, gives us a feeling of uncertainty and distrust. But as it turns out, far from being a mere symptom of our cynical modern society, the 'rose-tinted spectacles' view of the past is itself as old as the hills! Let's rewind to the 1750s, when the industrial revolution in Britain was finding its feet, and various entrepreneurs were experimenting with new progressive methods of mass manufacture. At that time, the middle and upper classes were becoming fascinated by the antiques excavated from places like Pompeii, which they believed were the produce of a near-perfect civilisation. The rich frequently took study tours of Greece and Italy, shipping artefacts back to their aristocratic homes to furnish them in classical style. For those that couldn't afford a house full of plundered antiques, there was another way to own a slice of the golden age.

About The Author Tom Flint is a freelance writer who spends as much time as he can recording and producing his own music in his home studio. He's also a painter and sculptor, but has no plans to fabricate any Etruscan pottery.

In the 1760s, merchant Thomas Bentley, and entrepreneur Josiah Wedgewood, who became one of the most successful Staffordshire pottery manufacturers, teamed up to begin producing pottery to fit the market. Having acquired a collection of classical archaeology books, Wedgewood directed his craftsmen to make reinterpretations of various pots and ornaments. He even named one of his factories 'Etruria' to evoke thoughts of the legendary

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Sounding Off

Etruscan culture, even though the workings of an 18th-century Stoke-on-Trent factory must have been totally different from the potteries of Tuscany 2000 years earlier! Through careful marketing, Wedgewood made his products appear as though they were fundamentally of the same breed as the objects they superficially resembled. Of course, Wedgewood's progressive manufacturing processes and his innovations didn't sit comfortably with the image of Etruscan craftsmanship, so he was careful not to draw too much attention to them. He even marketed some designs as the rediscovery of a lost craft! Wedgewood realised that antiquity was the saleable quality, not novelty. And that brings us right back to the audio technology industry today — a time when even cutting-edge software is designed to visually and sonically emulate classic analogue audio hardware. You only have to flick through the reviews and ads in SOS to see any number of new companies busy making products that are the spitting image of various classic bits of gear. The 'vintage' imagery makes us think of white-coated engineers sporting wire spectacles and comb-over hair, scribbling test results onto clipboards as they diligently study rows of glowing valves and large illuminated dials. In the back of our minds grows the idea that the product can give us the sound that made the Beatles a success. Just as the 18th-century upper classes were being sold a dream of a better time, the modern music industry exploits our collective suspicion that there was something fundamentally special about all that vintage gear, and that it was somehow responsible for the success of the great records of the day. The truths are that the Etruscan civilisation had just as much conflict as that of 18th-century Europe, and that the engineers and producers working on all those great records, using fantastic analogue gear, were dreaming of a future when they wouldn't have to put up with the shortcomings of the equipment they had at their disposal. I'm sure that in 30 years, we'll be looking back at today and thinking, 'that was a golden age for music technology. Those early software synths and processors were really reliable, and gave us an earthy sound you just can't get with today's gear!' It says a lot about how conservative we are as consumers, and how much value we place on established forms of music and technology. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Studio SOS

In this article:

Studio SOS

Hear Here: Monitoring Rod Brakes Rod's System & Methods Published in SOS June 2005 Tweaking The Drums Getting To Grips With Guitar Print article : Close window Rod's Comments On The People : Studio SOS Session Improving The Electric Which Sample Rate? Optimising The Processors

The owner of an unusually bijou studio setup provides the chocolate biscuits this month, as the SOS team get busy helping to improve the performance of his gear and the sound of his recordings. Paul White

Rod Brakes' studio is probably the smallest the Studio SOS team has ever seen, but he still has everything necessary to produce good-quality recordings. His main instrument is the guitar and his home studio setup includes several shelves full of pedals and processors dedicated to creating new and unusual guitar sounds. Space is clearly a limitation, and the studio is set up as a booth in one corner of his living room, where a set of pine shelves forms a partition to his left and provides space for his computer, as well as for some storage (see photo on page 102). The space between the wall on the right and the shelves on the left is little over a metre, so this clearly isn't the ideal place to set up accurate monitoring. Rod called us in mainly to concentrate on his acoustic and electric guitar recording technique but, as always, we tried to improve the monitoring first.

The lovely Martin acoustic wasn't sounding as good as Rod Brakes had hoped when recorded, so Paul threw on some headphones and set to locating a miking position for the Rode NT1000 that would truly reflect the sound of the guitar as Rod experienced it while playing.

Hear Here: Monitoring Working in such a confined space, Rod had wisely chosen some fairly small file:///H|/SOS%2005-06/Studio%20SOS.htm (1 of 9)9/28/2005 2:41:11 PM

Studio SOS

monitors, in this case the surprisingly effective and inexpensive Egosys Near 05 Active 129 models he'd picked up from Turnkey. When we arrived, these were mounted directly on his glass-topped desk — along with his flat-screen computer monitor, outboard rack and QWERTY keyboard — but they were facing backwards! However, this wasn't some weird new monitoring technique: Rod has a very young daughter with a propensity for sticking her hands into exposed tweeters, so he turns the monitors around to face the rear wall when they're not in use, to protect them. As Rod had them set up on his desk, the monitors were much too low, so we raised them slightly on a set of Auralex foam speaker pads, configured back-tofront to allow the speakers to be angled upwards (rather than down). The extra foam wedges provided with the pads were also used to increase the upward angle, which resulted in the tweeters firing more or less directly towards the listener's head — exactly as it should be. UK Auralex distributor, Audio Agency, kindly donated the speaker pads to the cause, along with a few panels of acoustic foam. Playing our test audio disc showed that the monitoring actually sounded a lot better than we'd expected in such a tight location, but the stereo image was somewhat messed up by reflections from the nearby wall on the right and the computer on the left. We improved this situation by fixing just two Auralex 24inch-square foam tiles to the left and right of the mixing position at head height, one glued to the wall (using a spray carpet adhesive from my local DIY store costing £3.99 a can) and the other wedged temporarily next to the computer on the pine shelves. Rod planned to use map pins to fix this more securely to the side of the shelving unit. The result was a slightly tighter sound with noticeably better stereo imaging. A further benefit was that the foam on the left also reduced noise from the computer. Our verdict was that, with care, Rod could achieve decent mixes with this simple setup, but that if he intended to release any of it he'd ideally need to get it mastered somewhere that had accurate monitoring.

Rod's System & Methods The rest of Rod's system is based around Steinberg Cubase VST software running on a fairly brisk PC with an Emu 1820M audio interface. Level control for the monitoring is done in the Emu software that essentially channels audio into and out of Cubase VST, so one of our first recommendations was that Rod buy a simple monitor controller, such as the Samson C-Control, to allow him hands-on control over monitoring level and also to let him play his CDs or keyboards without first having to fire up the computer. This same box would also provide a headphone output and talkback. Talkback might not seem such a big deal in such a small studio, but Rod tends to use different rooms for recording different instruments, the bathroom being a favourite for acoustic guitar. Having talkback would enable him to communicate with the player in the bathroom via the headphones, although most of Rod's current projects seem to involve him doing all the playing, singing and engineering.

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Studio SOS

Most people recording an electric guitar in a small domestic studio would use something like a POD XT or other direct-recording device for convenience and flexibility, but Rod is a bit of a traditionalist and has a 100W Marshall JMP Mk II Lead amplifier (dating from 1979) feeding a Marshall 4x12 cabinet built in the '60s — one of the ones fitted with Sutton Surrey Celestion 'green back' G12M speakers. Rod's monitors were far too low on the desk. Raising them and angling them upwards This isn't quite as rash as it might slightly on foam pads made a big seem, though, as Rod houses the improvement to the monitoring sound. cabinet in a kitchen closet facing a wall covered with acoustic foam and runs cables under the door. There's just about room between the speaker and the wall to get a microphone into position and that's how he records the cabinet sound. His traditionalist approach also extends to synths and drum machines. He has just one soft-synth bundle but is most happy when he's putting the separate outputs from his cherished Roland TR909 drum machine through his guitar pedals to wring something new out of it. His only hardware synth is a threeoctave Roland Sound Canvas keyboard. Rod has been DI'ing his Ibanez bass guitar via an Electro-Harmonix Black Finger optical tube compressor and a little Dbx Vacuum Tube Preamp, then feeding the output from this into the mic input of his Emu interface. This particular Dbx preamp has a high input impedance on the line jack, and so is ideal for guitar and bass DI purposes. However, we suggested he used the line input of the Emu interface in future, as the levels would be better matched and the result would be less noise in his recordings.

Tweaking The Drums Before trying anything else, we sat down to listen to some of Rod's own recordings while he furnished us with cappuccino and Hobnobs. (If anyone else is thinking of asking us over, might we mention that there is now a new Extra Chocolate version of Hobnobs that we haven't tried yet?) As it turned out, Rod's recordings were all pretty clean and tight-sounding, and his traditional approach even extended to recording his own drum samples from an acoustic kit, with drums recorded individually using a Shure Beta 57 dynamic mic and a Rode NT1000 capacitor mic. The only serious flaw was the sound of the kick-drum sample, which was quite short and 'boxy'. Kick drums really need a dedicated mic with lots of low end to record them convincingly, and although I knew that processing was never going to make this one sound great I decided to have a go anyway, using the EQ and compressor in Cubase VST.


Studio SOS

To improve the overall sound, I set up an EQ with a fairly aggressive boost at 90Hz, a dip at around 200Hz and another peak in the 4.5kHz region, the idea being to lose some of the boxiness, beef up the low end and emphasise the click of the beater. There was a very noticeable improvement in tone, and after I compressed the kick sound with auto release and an attack of 10ms, then switched on the soft-clip limiter, the kick sat reasonably well in the mix — although it still wasn't going to win prizes. Rod's best bet is to become slightly less purist and substitute a nice bass-drum sample from elsewhere!

Getting To Grips With Guitar Turning our attention to the rest of the mix, it seemed that Rod had tried to optimise each sound in isolation, the result being that when they were playing together they all fought to be at the front of the mix. This wasn't so bad in the intro, where only a couple of instruments were playing, but it got fairly messy when the electric guitars came in. The acoustic guitar, though clear, had a somewhat hard quality to it and the DI'd bass also sounded quite assertive. I tried some EQ on the acoustic guitar to cut the region between 500Hz and 1kHz, which took some of the hardness out of it, but the real solution to these problems was to examine Rod's recording technique and see what could be done to improve the sounds at source. Acoustic guitar figures quite highly in Rod's music and he'd recently treated himself to a mahogany Martin Auditorium guitar, which sounded rather nice played acoustically. Rod also uses alternate tunings, such as DADGAD, which suit this guitar well. Normally he records the guitar using his Rode NT1000 (his only capacitor mic) or his Beta 57, with the common technique of pointing the mic towards The trusty foam came out again to improve the sound of Rod's listening position. Here the position where the neck meets the Paul's holding two foam tiles in place so that body. He generally uses the bathroom Rod can hear the difference they make. to add a bit of life to the sound, but we decided to experiment in the living room, where there was more space, then move to the bathroom after we'd established a working method. Given the choice, I'd never record an acoustic guitar using a dynamic mic, other than perhaps a Sennheiser MD441, as capacitor mics generally handle the top end more effectively and they're also more sensitive, which helps keep noise down. So we rigged the Rode NT1000, initially with the mic in Rod's usual position (aiming at the neck/body joint from 12 to 18 inches in front of the guitar). We also put a couple of boards on the floor in front of Rod, to reflect some sound back up into the mic. The board furthest from him was angled up, originally


Studio SOS

because a chair was in the way, but it sounded good so we left it! We used boards because the living room was carpeted and acoustic guitars often respond well to reflective wooden or tiled floors (see photo on the first page of this feature). With Hugh Robjohns driving Cubase we made some test recordings and, sure enough, the result was the same clear but hard sound that Rod had been getting. Apparently, Rod had also tried the higher mic position he'd read about in one of my books, where the mic, in effect, looks over the player's right shoulder, but it still sounded too hard. This just goes to confirm that the sweet spot varies enormously, depending on the characteristics of both the mic and the guitar, plus the tonal preferences of the player or engineer. The task was now to try to find a mic location that would give a sound close to what we felt we were hearing acoustically from the instrument, and I did this by monitoring via headphones while moving the mic around. Rod played as I moved the mic, and eventually I found a great-sounding spot around a foot from the floor and two feet from the face of the guitar, with the mic aimed up at the 12th fret. Here the sound was really sweet and airy, without any of the harshness we'd heard before. After making more test recordings, we played the original and new mic-placement versions back to Rod and he agreed that the new placement had really captured the qualities of the guitar that he heard while playing it. Transferring this setup to the bathroom gave us a similar tone, overlaid with a short, bright room ambience, but we Having worked out how to mike the guitar had to move Rod around so that he for a good sound in the lounge, Paul and wasn't playing parallel to the wall, in Rod adjourned to the bathroom, where Rod order to make the sound suitably likes to make use of the ambience, to settle on the best seating and miking positions subtle. At this point, Rod asked if it was worth miking the guitar in stereo. This is there. Rod also wanted to try miking in stereo (lower photo). largely a matter of taste. My own view is that mono miking works well where the guitar is part of a busy mix, but stereo can add dimension to the sound where the guitar is a featured instrument. I'd brought along my own SE Electronics SE1a small-diaphragm cardioid mic, in case extra mics were needed, but you can try this technique with just about any decent capacitor model.


Studio SOS

Rather than using a traditional spaced-pair or coincident-pair approach, we left the NT1000 where it was, as the main mic, and set up the SE1a on a separate stand around six inches from the headstock of the guitar, aiming it more or less at the nut (see photos on page 103). More test recordings confirmed that this produced a very different and much brighter tone than the main NT1000. With equal contributions from both mics the sound was probably too bright for most applications, but by reducing the contribution from the mic on the headstock and panning the two sounds apart, we were able to achieve a very nice, spacious sound with a good tone. In this type of situation, the room ambience can be regulated quite effectively by hanging towels over the bath and radiators, and for this session we found that a towel over the radiator definitely helped. The final sound had just the right amount of ambience to allow it to work in a mix with little or no added reverb.

Rod's Comments On The Session "It's not every day you get the chance for two of the world's most respected experts on music technology to visit your home and spend the afternoon giving you a one-to-one home-studio consultation! I was and still am blown away. I will never forget it. Thanks guys. "Now everything sounds infinitely better. I certainly don't take the Auralex acoustic foam gifts lightly and I really appreciate them. They make a massive difference to the way things sound. "As well as being great company, Paul and Hugh were both amazingly helpful and insightful; I feel as though I've learned a huge amount following the visit, and have been immediately putting all the hints and tips into practice. For example, I've just recorded a superb take of an instrumental acoustic number that sounds absolutely first-rate thanks to Paul and Hugh finding the 'sweet spot' for my Martin, angling the mic correctly and making use of some reflective surfaces and ambience. It sounds so good from source now; this one's gonna need pretty much sweet FA in terms of further processing. Nothing short of a triumph! "Overall, Paul and Hugh's comments were very encouraging, and all I can say is that I must be paying attention when I read Paul's books and SOS! Now all that remains is for me to record a great album..."

Improving The Electric Our next port of call, after the ginger cookies had been consumed, was the electric guitar. I brought along my POD XT for comparison and Rod was impressed both with its versatility and its ability to emulate his miked Marshall fairly authentically. We also coaxed a very nice DI'd bass sound out of it with little


Studio SOS

effort. However, as he was already getting the sounds he wanted from his Marshall, he felt that sticking with that would be more organic, especially as he'd built up an impressive collection of Electro-Harmonix pedals and other processors to use with it. We decided to work through the guitar-cab miking method to see if we could improve on it, and I achieved what I felt was an accurate interpretation of the amp sound by using the NT1000 about two inches away from one of the upper speakers. Rod had been miking one of the lower speakers, but we thought it was a good idea to move the mic up, to minimise any reflection effects from the floor. The mic was positioned roughly midway between the centre and edge of the cone, which worked very well, with surprisingly little sound leakage back through to the studio. However, I felt that the midrange might benefit if the thickness of foam on the wall behind the mic was doubled, and as we had one piece of foam remaining we left it with Rod to install in the cupboard himself.

Rod prefers the traditional approach to electric guitar sounds, miking up a Marshall cab that's hidden away in a cupboard.

Rod then set about demonstrating some of the sounds he likes, using his processors in front of the Marshall, which he generally leaves set to a fairly clean sound. He has an Electro-Harmonix Hot Tubes overdrive pedal, a Marshall Guv'nor distortion pedal, Boss DS2 Turbo Distortion and OS2 Overdrive/ Distortion pedals for producing overdrive, and he's customised his Mexican Fender Strat by fitting a Deluxe-style roller nut and locking Sperzel machineheads. He's also changed two of the pickups. The bridge pickup is now a Seymour Duncan Hot Rail and the neck pickup a Quarter Pounder from the same company. A non-locking mute button has also been installed for creating effects. For special effects, Rod has an E-bow, which he was still coming to grips with at the time of our visit, but his piéce de resistance is using wah, overdrive and echo while playing the guitar with what might be most safely described as a batteryoperated, variable-speed personal massaging device! The golden vibrating tip is effective in exciting the strings, while the electromagnetic interaction between the varispeed motor and the guitar pickups produces a high-energy sound not unlike a light sabre burning its way through a Morris Minor engine block! Apparently, he got some very funny looks while in the shop trying to select the most appropriate model... Our slightly tweaked mic setup was delivering the goods with the guitar sound, but one thing I noticed was that the Marshall amp had a slightly 'barky' quality when its own overdrive was invoked, which suggests that the output valves may need to be checked, along with their biasing.


Studio SOS

Which Sample Rate? At the end of the session, Rod asked us whether he should be working at 44.1kHz or 48kHz, as he felt 48kHz would give a better result. Technically, 48kHz can sound a little better than 44.1kHz, but in a system like this one there is unlikely to be any detectable difference — and by the time a 48kHz recording has been reduced to 44.1kHz, by means of the Cubase sample-rate converter, it may actually sound less good than a straight 44.1kHz recording. Both Hugh and I felt that working at 44.1kHz and 24-bit would be best, leaving dithering to 16-bit as the last step of mastering.

Optimising The Processors The final part of our assignment was to advise on the best way for Rod to use his Aphex 204 Big Bottom Exciter, his Focusrite Compounder and his new Lexicon MPX550 reverb unit. These sit in a rack on the desk beneath his computer monitor, and previously he'd passed the entire mix through all three in series. This is, of course, quite limiting, as the same amount of processing is applied to everything, but there is a limit as to how flexibly you can use hardware when your audio interface has only eight inputs and outputs. Our final recommendation was to set up an aux send and return in Cubase VST (using a spare output and two inputs on the Emu interface) and use this to feed the MPX550, set to 100 percent wet. In this way, differing amounts of reverb can be added to each of the VST tracks and the quality will be much better than the rather indifferent software reverbs that come with Cubase VST. Furthermore, Er... actually, we said show us your vibrato... because the MPX550 has digital I/O, it can be connected via the S/PDIF socket of any suitable audio interface that has S/PDIF I/O, to avoid an extra stage of analogue-to-digital conversion. The Compounder and Exciter can be left at the output of the chain to process the stereo mix, which can then be recorded back into the system as a new stereo track. Used with care, these two processors are very good for overall mix sweetening, while the EQ and dynamics in Cubase are perfectly adequate for most track-tweaking needs. Our final advice to Rod was to burn test CDs and play them on as many systems as possible, to allow him to get used to and compensate for the inaccuracies of his monitoring system. Though it's not a bad monitoring setup, the bass end will inevitably be misleading to a greater or lesser degree with such small speakers in such a compact location.


Studio SOS




Pro Tools updates & free EQ plug-in

In this article:


HD Upgrade Offer Extension Pro Tools Notes Collaborative Software Published in SOS June 2005 Software Upgrades & Bundles Print article : Close window Pro Tools Surround Sound Technique : Pro Tools Notes Mixing

Current Versions Mac OS (10.3.2, 10.3.4, 10.3.5, 10.3.7 and 10.3.8: note that Mac OS X 10.3.3 and 10.3.6 have not been fully tested and so are not recommended): Pro Tools HD and HD Accel: v6.9. Pro Tools Mix and Mix Plus: v6.4.1cs3. Pro Tools LE: v6.9. Windows (XP Professional or Home Edition with Service Pack 1 or 2): Pro Tools HD and HD Accel: v6.9. Pro Tools Mix and Mix Plus: v6.4.1cs3. Pro Tools LE: v6.9.

This month sees a high-quality, free EQ plug-in gracing the Digirack suite, some intriguing options for collaboration via the Internet, and a host of updates. Mike Thornton

There's plenty of news from Digidesign this month, including major launches such as Pro Tools M-Powered and Pro Tools 6.9, which was released just as we went to press. There'll be more on v6.9 next month, while M-Powered is covered in depth elsewhere in this issue, but there's no shortage of smaller items to tell you about in Pro Tools Notes. First up, Digi have released a new free plug-in: EQ III is a 48-bit EQ plug-in available for Pro Tools and Avid systems. The latest addition to the Digirack suite of plug-ins, EQ III supports TDM, RTAS, and Audiosuite formats, and is available in one-, fourand seven-band configurations. As well as conventional parametric bands, EQ III includes selectable shelving filters and settings plus separate high-pass, low-pass, and variable-Q notch filters. However, be aware that EQ III operates as a mono or multi-mono plug-in only, with stereo and multi-channel tracks supported through multi-mono operation. EQ III works on all Pro Tools HD and Pro Tools LE systems running Pro Tools 6.7 and higher on Mac OS X and Windows XP, as well as Avid Xpress, Avid Xpress DV and Avid DNA systems. Although it hasn't been tested by Digidesign on Mix systems running v6.4, users have reported it works successfully provided you have at least v6.4.1cs3 of Pro Tools installed. However, on Mix hardware it will only be available as an RTAS and Audiosuite plug-in, as the TDM version

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has been written specifically for HD/Accel cards. Two further words of warning regarding EQ III: first, do not attempt to install it on a system running Pro Tools v6.2.x or lower. If you do, you will have to delete some files by hand and then re-install Pro Tools from scratch! Second, there is no compatibility between EQ II and EQ III. EQ III is 'in addition to' EQ II not 'instead of', so you will still need EQ II to open any of your previous sessions that use EQ II. As will be apparent from the screenshot, the user interface is a vast improvement over EQ II, with nice knobs and a graphical display similar to that of the Waves EQ plug-ins. The reports on EQ III are already very complimentary, with lots of people preferring it to the Waves Q series and some even rating it more highly than the Waves Renaissance EQ; others are comparing it to the Sony and McDSP EQs, which for a free plug-in is praise indeed. Digidesign are already talking about an update with some user interface improvements, and there is unconfirmed speculation that they might bring out a new compressor, which in my opinion would be a long-overdue improvement. www.digidesign.com/download/eq3/

HD Upgrade Offer Extension If you've been thinking of upgrading to Pro Tools HD, it might be worth doing so before 20th June. First, Digidesign have permanently lowered their hardware exchange prices for Pro Tools 24 and Pro Tools Mix to Pro Tools HD upgrades, and for the first time, are allowing Pro Tools III owners to take advantage of these offers. By exchanging your Pro Tools III, Pro Tools 24 or Mix system, you can save up to £5000 off the exchange list price for Pro Tools HD2 Accel and Pro Tools HD3 Accel systems, and up to £3400 off the list price of Pro Tools HD1 systems. Second, if you do it before June 20, 2005, you'll receive free TDM plug-ins. Those exchanging up to Pro Tools HD1 will get Digidesign's Smack! compressor, which is valued at £375; Pro Tools HD2 Accel buyers will get both Smack! and the Bomb Factory Pultec EQ bundle, together worth £750; and if you're going to Pro Tools HD3 Accel, you'll get Smack!, the Pultec EQ bundle, and Digidesign's Revibe modelling reverb plug-in, with a total value of £1375. All hardware exchanges up to Pro Tools HD and Pro Tools HD Accel systems also include up to £5985 in free TDM plug-ins with the current HD Pack promotion. www.digidesign.com/exchange/index.cfm

Collaborative Software This month also sees two announcements that will be of interest to users seeking to collaborate and work across the Internet. Source-Connect is a Pro Tools plugin that uses the 'voice over IP' technology that makes it possible to make phone calls via the Internet, enabling you to make audio connections between Pro Tools



systems anywhere in the world. Source-Connect enables direct-to-timeline recording of real-time, broadcast-quality audio with an AAC codec and requires a T1, cable or DSL Internet connection of at least 300kbps bandwidth. It is crossplatform, so Macs can connect with PCs and vice versa. Source-Connect works with Pro Tools TDM or LE version 6.4 or later; you will also need an iLok and a minimum 2GHz PC running Windows XP or minimum 1GHz G4 Apple Mac running OS X. www.source-elements.com

Digidesign have updated their Digidelivery server software to version 2.0. This update enables Server-to-Server Relay, adds the sender's user name to the deliveries list, and fixes a bug with work order entries, along with other minor fixes and improvements, and all Digidelivery administrators should update to this version. Digidelivery is the first module of Digidesign's collaborative system, which is based around two Ethernet network units and allows you to exchange any kind of file of virtually any size with anyone in the world. Pro Tools media files can be automatically included with the Session file, and the system has 128bit encryption security and lossless compression for speed and reliability. www.digidesign.com/products/digidelivery/

Software Upgrades & Bundles The plug-in formally known as Finalis (see last month's Notes) has changed its name to Finis and has been updated to v1.0.1, with a few small bug fixes. It's available from Elemental Audio (www. elementalaudio.com). Digidesign have updated their Revibe TDM plug-in to v1.1cs1, fixing a problem with certain Revibe reverb types where Ls and Rs channels of a surround mix were not properly fed to the reverb algorithm. Meanwhile, Audio Ease (www.audioease.com) have updated their Altiverb convolution reverb plug-in to version 5. You can now choose either iLok or challenge-and-response copy protection, while there's a new positioning control and four-band EQ, internal level meters, input and output controls for front, rear, centre bleed and LFE channels, separate gain and delay controls for the Direct, Early Ref and Tail sections, a CPU load control and a Reverse reverb mode. Internal test samples can now be triggered from the Mac's keyboard, preset switching is automatable, and there's three-band damping. Also included in this new release are presets ranging from the Amsterdam Concertgebouw to the back of a Ford Transit van!



Antares have announced the release of their Auto-Tune TDM Production Suite for Mac OS X. It includes the TDM versions of Auto-Tune 4, which corrects intonation problems in vocals or solo instruments in real time, and Microphone Modeler, which is designed to impose the characteristics of one microphone on material recorded with another. Auto-Tune TDM Production Suite also includes RTAS versions of three more plug-ins. The first is the Kantos audio-controlled synth, which analyses incoming audio and extracts pitch, dynamics, harmonic content and formant characteristics and uses them to control the Kantos sound engine in different ways from a conventional MIDI synth. The second is Antares' Filter, which features four stereo multi-mode filters, a complement of control sources, modulation capabilities, and features optimised for rhythmic loop-based applications. The third RTAS plug-in is Tube, a valve simulation offering a range of effects warmth to serious distortion. www.antarestech.com/products/atsuite.html

Pro Tools Surround Sound Mixing Finally, it's not often we mention a book in this column, but here is one that has come to my attention. The book is called Pro Tools Surround Sound Mixing, and is written by Rich Tozzoli, who describes himself as a 'surround sound guru'. In it he covers preproduction, recording, setup, mixing and delivery of surround music, and discusses formats such as DVD-Video, DVD-Audio, and SACD, showing how Pro Tools can be used to deliver mixes for these various formats. The book also covers encoding mixes for Dolby Digital and DTS, mixing to picture, and other information for working with Pro Tools. The bonus DVD-Video includes over a dozen 5.1 examples of surround sound production with individual explanations for each example. www.backbeatbooks.com Published in SOS June 2005






Audio Input for Reason

In this article:

Hammer It Home Patch It Up Quick Tips

Audio Input for Reason Reason Notes Published in SOS June 2005 Print article : Close window

Technique : Reason Notes

With some clever programming, an Italian software house have created a program that claims to add an audio input to Reason. We take a sneak peek, as well as bringing you essential Reason news and quick tips. Derek Johnson

Showing a good mix of ingenuity and lateral thinking, Italian developers Petertools have struck again. They've already made an impression with their Live Set, which provides a performance biased front end, consisting of various nifty MIDI manipulation tools, aimed at Rewireequipped applications — although Reason was the main platform for Live Set's development.

A couple of examples of how users are customising the look of the Combinator, with the 'Select backdrop' option.

Now Petertools have taken a close look at the Rebirth Input Module (RIM) device and think they may have half an answer to many a Reason user's major wish. How does the prospect of an audio input sound? That will be the aim of the software currently known as Hammer.

Hammer It Home Hammer is still in beta testing, but, once finished, should allow audio to be routed to Reason via full-duplex, multi-client, ASIO-compatible audio cards. Don't get too excited: Reason still lacks any kind of target for recording audio, so that remains a dream, but it certainly offers plenty of devices which can be used to treat audio. file:///H|/SOS%2005-06/Audio%20Input%20for%20Reason.htm (1 of 3)9/28/2005 2:41:32 PM


With the help of Hammer, external audio can be routed via the RIM to the Malström audio inputs, BV512 vocoder, Scream 4 distortion or any other effect (or effect-laden Combinator). Such treatment will always be a live experience, since neither of Reason's bounce-to-disk audio options works in real time. It's also not possible to Rewire Reason to another application while Hammer is in use (a side-effect of how Rewire works). But there should be a way to record the finished output, to an external digital recorder or direct to your hard disk via other software you might own. It seems that latency is a bit of an issue when integrating the Hammer input with Reason — perhaps unsurprisingly, given that the audio is passing through a number of stages that add increasing delays. Petertools thus recommend the use of high-end audio cards that are capable of very low latency. Once the software is released, the combination of Hammer and such a low-latency audio card will bring Reason more definitely into an interactive performance environment. Keep track of developments at www.petertools.com.

Patch It Up I'm sure we all appreciate the efforts of third-party commercial entities, and other Reason users, to provide novel sonic material — and it's always interesting to see how other people, whether commercial or enthusiast, push the platform. For example, the new Combinator device introduced with v3 is the main focus for any sound designer working with Reason, and commercial developers and users are debuting their creations as I write. A visit to the popular Reasonstation (www. reasonstation.net) reveals an expanding range of Combis, some of which just sound good and some of which showcase clever programming ideas, from creative use of velocity splitting to unusual soundscapes and rhythmic experimentation. The new examples also show off the graphic possibilities offered by the fact that the Combinator's 'skin' can be customised with the 'Select backdrop' command. Users are often as creative in this department as they are in the sonic field.

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Quick Tips Let's say you've loaded a Combinator with several Subtractors (or other devices), which you'd like to velocity switch and play as one super-instrument. In some circumstances, you might like to save a Matrix sequencer as part of the Combinator setup, and in order to play the multiple devices from the one Matrix you might think it necessary to split the gates and CVs with Spider CV devices. Not at all: simply route the Gate and CV outs of the Matrix in the Combinator to that Combinator's Gate and CV ins. The routing will be saved when you save the 'Combi', allowing you to include a range of patterns in the patch. Don't forget that Reason v3's new patch browser provides auditioning for all patch types. That includes effects devices, such as RV7000 and Scream, that offer patch saving. Simply ensure that the device you're loading the patch into is highlighted in the sequencer track list, and that the MIDI In icon is enabled. Another v3 feature worth remembering is the Combinator's FX Bypass switch. This is especially useful when working with the MClass Combinator type, or a custom Combi made up of effects. All effects can be muted with one click, allowing the dry or unprocessed mix to be checked in its raw state. Be creative when assigning Combinator rotary knobs: one knob doesn't have to be assigned to the same parameter on every related device in a Combinator. For example, if you've included a DDL1 delay, you could set its feedback parameter to change with the filter resonance parameter on a couple of Subtractor synths, to make delay time increase dynamically with changes to the filter. Or set a knob to increase reverb time on an RV7 or RV7000 and delay feedback on a DDL1, for instant dubby effects. If you find that high feedback or reverb decay time values get out of control in these specific examples, remember that it's possible to cap the parameter range of knob assignments. In the case of DDL1 feedback, for example, set the 'max' value to around 100 instead of 127, and even when the rotary knob is tweaked full right, feedback will not reach its full level. The same technique could be used in other potentially unstable circumstances, such as when using the rotary controllers to increase LFO depth or speed. (See the workshop feature starting on page 244 for more on Combinator programming.) Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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Better Acoustic Guitar Recording In Logic

In this article:

Matching EQs Conclusions Quick Tips: Miking Acoustic Guitars

Better Acoustic Guitar Recording In Logic Workshop Published in SOS June 2005 Print article : Close window

Technique : Logic Notes

This month we look at how you can mock up a live acoustic guitar sound from a DI'd recording using Logic's built-in plug-ins. Paul White

Recording the acoustic guitar is much the same regardless of the recording hardware or software being used, but Logic has a number of plug-ins that are well suited to creating both contemporary and more traditional acoustic guitar sounds at the mixing stage. There's also a new plug-in in Logic Pro 7 called Match EQ, and, as I'll show later, this may sometimes be effective in improving the sound of those guitars fitted with piezo electric bridge pickups. Don't forget that if you have one of those rare passive piezo pickup systems, you need to DI it using an active DI box or other device with a very high input impedance. Guitars will in-built preamps can be DI'd directly via a line input of your audio interface. Life is easier from a technical standpoint if you DI the guitar via an inbuilt pickup system insomuch as you don't have to worry too much about noise pickup from the computer or from other instruments, but in my experience, the output of a piezo bridge pickup takes a lot of processing to get it to sound anywhere near right. Systems that combine pickups and mics, such as the Fishman Rare Earth system, fare rather better, but if you want to capture the real acoustic sound of a great instrument, there's no better option than to mic it using a decent capacitor microphone. However, if DI'ing is your only practical option, then what can Logic do to help you get somewhere close?

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Matching EQs This brings me back to Match EQ, which provides a more or less automatic way to EQ one sound to make it sound more like another. Match EQ uses a very large number of EQ bands and starts by analysing a reference sound (the sound you wish to emulate), which in my experiment was a short section of the acoustic guitar recorded using a microphone. It could also be an excerpt of solo acoustic guitar imported from CD if you have something you like the sound of. In Logic parlance, this reference audio is referred to as the template material. Match EQ is simply dropped into an insert point, and in this example, I found it convenient to record my section of reference guitar on the rhythm guitar track before the actual start of the song. For the main part of the guitar track, I set the EQ on my acoustic guitar pickup system to flat and recorded a couple of minutes of playing. Pressing the Analyse Template button on Match EQ while the reference section was playing back stored an EQ curve, after which I moved the cursor to the DI'd guitar part, resumed playback and pressed the Analyse Current button. This stores a second curve, then when Match EQ showing the DI'd signal combined with the reference EQ. you hit the Match button a correction curve is computed to make the Current sound spectrally similar to the Template. Note that you may not hear the audio play back while a file is being analysed but you'll see the frequency curve build up in the plug-in window. There's a fader that allows you to apply more or less of the correction curve, even going beyond 100 percent, and there's also a slider to smooth the EQ curve being applied — something best judged by ear. A further button provides a choice of linear or minimum phase EQ and again the sound is slightly different so you need to listen and choose the nicest-sounding one. If you feel unsure of these controls, simply stick with the defaults, as they usually do a pretty good job. In my test, the general sound of the guitar after processing via Match EQ was very much closer to that of my referenced miked recording, even though the unprocessed sounds were hugely different. However, you can tweak the final sound further by creating EQ curves by dragging up or down in the Match EQ window, and if this doesn't work out, you can restore the original curve by following the prompt at the bottom of the plug-in window. Match EQ takes a fair amount of processing power and also introduces a little processing delay, so you may wish to use the track Freeze button once you have got the sound you want. Using this method, you may get an even better match if you use the guitar's own EQ system to get the pickup sounding as close to the miked sound as possible,



but an exact replication of the miked sound is unlikely with a piezo pickup system, as the attack dynamics of the notes seem a little different and can give the sound a slightly 'scratchy' character. Interestingly, I repeated this experiment using the piezo bridge pickup on my Parker Fly electric guitar, and though I've never been a great fan of the sound, doing the Match EQ trick probably brought it as close to the original acoustic sound as I got by processing my DI'd acoustic in the same way. As an afterthought, I tried the same process using the single-coil pickup position and got a surprisingly musical result — and without the scratchiness of a piezo pickup. The rest of the processing, again The EQ curves for the original DI'd recording and the reference recording. applied using only Logic's plug-ins, is relevant to both DI'd and miked guitars. For this project, gentle compression was used to even up the sound (you can see the settings I used in the screen shot) and I also used Logic's Exciter to add in a little high-end zing to the sound. It's best to set the Exciter frequency control to around 3kHz or above in this application, which prevents harmonics being added to the upper mid-range where harshness can be a problem. By restricting the added harmonics to the top of the audio range, you get a nice open, airy sound that isn't too aggressive. The final touch was to add a hint of reverb, and though there are some greatsounding rooms in Space Designer, I found that a short, bright reverb cooked up in Platinumverb added the right amount of space without getting in the way. Again you can see the settings in the screen shot. Where the guitar is required to sit in a pop mix rather than being the featured instrument, it may also help to pull out some low end using Logic's high-pass or low-cut filters, though you can also reduce the bottom end in Match EQ by dragging the curve downwards to around 200-300Hz.

Conclusions Although you can use more sophisticated plug-ins or external modelling preamps, Logic provides a great basic set of tools for recording the acoustic guitar either with a microphone or by DI'ing. It even

The rest of the processing chain — using only built-in Logic plug-ins.



includes a very accurate guitar tuner plug-in! Match EQ is a great help in trying to get the DI'd sound somewhere close to the miked sound, and Logic's compressor is smooth enough to use with acoustic guitar as long as you don't apply too much of it, in which case level pumping may be noticed. You can put the compressor before or after the equaliser, but in this case I felt the subjective result was best by compressing after the EQ, as this helped tame any harsh peaks occurring at specific frequencies. Acoustic guitar doesn't normally need a lot of reverb to make it sit well in the mix, and in most cases Platinumverb does the trick nicely, though Space Designer is a better option for more featured parts. The fact that these processing tricks also worked well on my electric guitar with a piezo bridge pickup is a bonus.

Quick Tips: Miking Acoustic Guitars If you're recording using a microphone, I can't stress too highly that stock mic positions don't often cut it unless you fine-tune them. Use them as a starting point by all means, but monitor the sound through headphones while moving the mic (or if you're also playing, while shifting the position of the guitar) until you pinpoint that all-important sweet spot. Sometimes the standard '18 inches from the neck/body joint' will work out, but you're equally likely to find the best spot with mic above or below the guitar body — every instrument and every room is different. It's also well known that acoustic guitars sound better played above a reflective floor rather than a carpeted one, so put a piece of board on the floor directly in front of the player's chair if the sound needs a bit more life. Even a few tea trays will make a difference. The only other special consideration is that of computer noise — whether you record in the same room or not depends on how loud your computer is, and in any event it's best to have the mic pointing away from the computer rather than towards it (assuming you're using a cardioid-pattern mic). Sometimes you can throw a couple of towels over the computer as you do the take, but don't forget to remove them again as soon as you're done, as there's always a risk of the computer overheating. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Building Combinator Patches in Reason 3

In this article:


12-step Program Workshop Skin Deep Published in SOS June 2005 Patch Portability Sequenced Synths & Effects Print article : Close window

Technique : Reason Notes

The most exciting addition to the new version of Reason is arguably the Combinator device, which greatly expands the flexibility and programming potential of everything in the Reason rack. We guide you through the creation of a Combinator patch to get you started with this great new device. Simon Price

In the build-up to the launch of Reason version 3.0 (see review in the last issue of SOS), Propellerhead Software released pictures and basic information about the update's largest single addition: the device known as the Combinator. The weeks following saw the Props' web site host one of the most heated debates ever seen on a music software web forum. Everyone understood the Combinator's functionality: it allows collections of rack devices to be compacted into a single device. The problem appears to have been a failure of imagination on the part of some forum participants who couldn't see the huge new power and potential this single device brings to Reason.

Just some of the many faces of the Combinator, courtesy of its 'skins' feature.

Here's the basic idea: 'combined' devices are all moved inside the new Combinator 'shell'. Audio connections are provided to and from the internal devices, and to and from the 'outside' main rack environment. The Combinator appears as a single MIDI destination and features a programmer for setting how the internal modules respond to MIDI, similar to the key-mapping within NNXT but for devices, not samples. Finally, there are four knobs and four buttons on the Combinator's front panel that can be assigned to file:///H|/SOS%2005-06/Building%20Combinator%20Patches%20in%20Reason%A03.htm (1 of 8)9/28/2005 2:41:42 PM


any of the controls on the combined devices. This offers some previously difficult or impossible options, for example: Producing complicated instrument patches like those found in 'workstation' synths, with sound sources, filters, modulation and effects all present, and easy control over the key parameters via assignable knobs. Creating layers and splits combining any instruments and samples. Loading complex arrangements and live setups (synth combos, or even entire tracks) that are collapsed into 'Combis' for easy recall, without disturbing clock by changing songs. Creating entirely new instruments and sequenced synths, Reaktor-style. Making new effects devices using the Combinator inputs.

12-step Program If you're still not seeing why this might be useful, let's go through an example of how it works and build a simple layered pad patch. Devices can be added to an existing Combinator, or selected objects can be collapsed into a new Combinator using menu option Edit / Combine. In this tutorial you'll do the former, starting with a fresh Combinator: 1. Create a new song and add a Remix 14:2 mixer. Select the mixer and choose menu option Create / Combinator. A new Combinator will be added, and if you hit Tab to flip the rack you'll see that the Combi Outputs are jacked automatically into channel one of the mixer. If you have the sequencer window visible, you'll also see that a new track has been created for 'Combinator 1', and your master keyboard will be ready to send MIDI to it. (See screen above.)

Start with a Remix and a Combinator.

2. Now click within the narrow black area at the bottom of the Combinator. file:///H|/SOS%2005-06/Building%20Combinator%20Patches%20in%20Reason%A03.htm (2 of 8)9/28/2005 2:41:42 PM


A red bar will appear in this box, indicating that any object you add from the Create menu will be put inside the Combinator. Add a Malström synth. Reason will automatically cable the Malström's audio outputs to the Combinator's 'From Devices' ports, and you will be able to play it from the keyboard. From the Malström's patch selector window, open up the Reason Factory Soundbank and load the Malström Pad patch called 'PHAD'. 3. Now we'll add another layer using another synth. Again, click in the empty space within the Combinator, just below the Malström. Add a Subtractor, this time from the Create menu. 4. There's a problem now: the audio output is not connected to the Combinator, so you can't hear the Subtractor. This is where Reason 3's new Line Mixer device comes in. Click on the Subtractor and choose Create / Line Mixer 6:2. 5. Now you can probably see what needs to happen. As shown in the middle screen on the right, manually recable the devices so that both synths are plugged into the Line Mixer and the Line Mixer's Master Out is connected to the Combinator's 'From Devices' ports. You will now be able to play both synths from the keyboard, something very difficult in previous versions of Reason. The last part of this step is to load the Subtractor pad patch 'Cloud Chamber' from the Factory Sound Bank.

Add a Malström for the first layer in the sound.

After adding a Subtractor for the second part of the layer, create a Line Mixer and re-cable both the Subtractor and the Malström so that their audio outs are plugged into the Line Mixer. Then connect the Line Mixer's Master Out to the Combinator.

Add a UN16 Unison device.

6. Congratulations! You've built your first Combinator patch — but let's go further. Select the Subtractor and choose menu option Create / UN16 Unison. The Unison effect will automatically be cabled in line between the synth and the Line Mixer, fattening file:///H|/SOS%2005-06/Building%20Combinator%20Patches%20in%20Reason%A03.htm (3 of 8)9/28/2005 2:41:42 PM


the sound nicely to match the width of the Malström layer. (See screen below right.) 7. The final building block for your patch will be a global filter. Click in the space below the Line Mixer and choose Create / ECF42 Envelope Controlled Filter. Reason will try to be helpful by connecting the filter to the aux send and return ports of your Line Mixer, but you should re-cable it so that it sits between the Line Mixer's Master Out and the Combinator's 'From Devices' inputs, as shown in the first screen overleaf. 8. The next stage in building your patch is to add some hands-on control to the Combinator's front panel. In this way you can eventually hide the internal devices and treat the Combinator as a single instrument. In the second screen overleaf I've hidden the devices and brought up the Programmer using the Show Programmer/Show Devices buttons on the panel. On the left-hand side is a list of all the devices contained within the patch. The small keyboard display works in the same way as the keymapping window on the NNXT sampler. Key ranges and velocity zones can be set here for each soundgenerating device. In the case of this patch you can ignore it, as we want both synths to respond to all keys and velocities. 9. To the right of the Programmer is the Modulation Routing section, where all front-panel knob and button assignments are set. The most obvious thing to do with this patch is program in some control over its master filter. Select Filter 1 in the device list, then click in the routing grid to the right of Rotary 1 (see the third screen on the left). Choose 'Frequency' from the pop-

Add an ECF42 Envelope Controlled Filter and re-cable it between the Line Mixer's Master Out and the Combinator's 'From Devices' inputs. The correct connections are shown above.

Bringing up the Combinator's programmer, to add some front-panel controllability to the sound.

Setting the Combinator's Rotary 1 to control filter cutoff frequency.

Setting Rotary 3 to control both Oscillator A and Oscillator B attack for the Malström.

Editing the value range for Malström Oscillator Attack on the Combinator programmer.

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up list of available parameters. Now try turning Rotary 1 on the Combinator's panel and you should have direct control of the filter cutoff frequency. Assign Rotary 2 to Resonance and rename the panel control labels by double-clicking on them and typing in new names. 10. A typical layered patch on a synth workstation would feature additional controls to alter the sounds and effects. A good idea for this patch would be to add control over how quickly the first pad layer comes in, because by default it has quite a hard attack. This is a slightly more complex process, as the Malström synth uses two oscillators with individual volume envelopes. Select the Malström in the device list and give Rotary 3 a target of 'Oscillator A Attack' in the routing grid. Now notice that the grid has two spare slots at the bottom. These provide for situations where you want to assign the same controller to more than one parameter on the same device. In the first spare slot, select Rotary 3 in the Source column (see the fourth screen on the left). Now you can add Oscillator B Attack to this knob. 11. You can add a further level of sophistication by editing the control ranges in the routing grid. At the moment, Rotary 3 moves the Attack sliders on the Malström across their whole ranges. However, you can tweak the Min and Max columns in the routing grid so that the knob only operates within the most useful range of values. Try setting the Max fields to about 80, by clicking them and dragging down with the mouse.

The finished patch. Two of the Combinator's buttons have been assigned to activate mute on two channels of the Line Mixer, so that the patch layers can be switched in and out individually.

A set of devices I use live and for songwriting. Pre-Combinator I could re-use them in other songs but the process was a bit laborious and time-consuming.

12. A good final step would be to use two of the front-panel buttons to switch the two sound layers in and out. The easiest way to do this is to map



buttons 1 and 2 to the mute buttons on channels one and two of the Line Mixer. When you map controls to buttons, the Min and Max ranges reflect the number of states the parameter you choose can be in. So, for example, the mixer's mute buttons can be either '0' or '1' (off or on). Notice, in the first screen overleaf, that I've reversed the default Min/Max values so that the Combinator's buttons act like 'enable' buttons rather than mute switches.

When the Combinator arrived I was able to transform my live chain of devices into a neat 'Combi' patch that can easily be opened in different songs.

Skin Deep One of the things that earned the Propellerhead brand a certain credibility was the fact that their Rebirth TB303-emulator software could be hacked and modified by users to change the graphics and samples. Clearly with those days in mind, the Props have included a 'skins' feature in the Combinator. In other words, you can import your own graphics to replace the plain light-grey front panel. Simply rightclick on the Combinator (or select it and go to the Edit Menu) and choose 'Select Backdrop'. If you fancy finishing off your unique Combinator patch with its own frontpanel design, you'll need to use a graphics program to make a backdrop 754x138 pixels in size, and export it as a JPEG. Once the backdrop has been selected, it will be saved into the Combinator patch file, so the original JPEG won't need to be referenced next time you load the patch.

Because the Combinator lets you combine all device types, you can even work Matrix pattern sequencing neatly into patches. The 'Graintable Sequencer Light' patch from the Reason Factory Sound Bank, designed to emulate how certain Reaktor Library patches work, is an example.

Patch Portability The example above shows how it's possible to create new and previously impossible sounds from smaller building blocks. This was one of the main intentions that Propellerheads had for the Combinator device, their minds being on modern hardware synth workstation patches. However, this is by no means file:///H|/SOS%2005-06/Building%20Combinator%20Patches%20in%20Reason%A03.htm (6 of 8)9/28/2005 2:41:42 PM


the only use for the new device. For a start, it's much more flexible, as you can combine synths, drum machines, samplers and matrix sequencers to create entirely new instrument concepts. Something else you can do is take complicated rack arrangements that have grown in a Song and unify them into manageable chunks. Many Reason users will have played around with patching CV and audio between different devices to get certain, often unexpected results. It's quite common to copy these linked devices from one song to another, and the Combinator provides a much tidier way to achieve this. It's particularly useful when you're trying to use Reason in a live setting. The middle screen on the left shows a collection of devices, based around a Redrum drum machine, that I've used in several songs and in live situations. There's quite a lot of patching, with the drum machine triggering a filter envelope, a Subtractor used as an LFO for the filter, and some effects connected in-line. Previously, whenever I've wanted to use this ensemble I've had to open a song containing it, select the objects and use the Copy Devices command. Then I'd paste it into the new song, get it connected back into the mixer, and create a sequencer track for it. Now I can just turn the whole thing into a Combinator patch for recall in any song. To do this I just need to select the objects, choose Edit / Combine, and the whole thing is collapsed into one box and can be saved as a patch. The bottom left screen shows the result. I've opened up the patch in a new song, and even added a few control mappings to make a really useful live instrument.

Sequenced Synths & Effects If you're like me, you've probably spent a lot of time sitting in front of Native Instruments' Reaktor, and wishing you could take a year off to design and build something big and wonderful. Particularly appealing is Reaktor's ability to make sequenced sound generators and innovative effects units. The Combinator opens up some cool possibilities in these areas. Obviously, it doesn't approach Reaktor's scope and flexibility, but the fact that the building blocks are much higher-level components than the nuts and bolts of a Reaktor ensemble means that it's much easier to quickly build things that work. You can learn a lot by studying some of the Combinator patches in the Factory Sound Library. The 'Pattern Based' folder contains patches that all feature some kind of sequenced element — drum machine, matrix or tempo-synced LFO. You can set these running using the 'Run Pattern Devices' button on the Combinator front panel, and they'll also kick in when you hit Play. Check out the patch 'Graintable Sequencer Light', which was designed to emulate how certain Reaktor Library patches work. See if you can unpick how it functions. Notice that it uses Matrix Pattern Sequencers to automate some elements, and an LFO connected to the Combinator's own CV inputs to control others. Published in SOS June 2005






Cakewalk Z3TA+ Bandlimited Waveshaping Soft Synth

In this article:

Musikmesse 2005 Z3TA+ Synthesizer Now Available Squashing A MIDI 'Bug'

Cakewalk Z3TA+ Bandlimited Waveshaping Soft Synth Sonar Notes Published in SOS June 2005 Print article : Close window

Technique : Sonar Notes

Cakewalk go global, and release a new synth. Craig Anderton

Historically, computer-based music in the US was the province of the Macintosh, due to the early appearance of MOTU's Performer, Passport's Master Tracks Pro and Digidesign's Sound Tools (later Pro Tools). The Atari also scored big for a while, with Cubase and Logic (orginally Notator). Initially, the PC was not taken seriously by the pro music community — with the exception of Cakewalk, who committed to the PC right from the days of DOS. Cakewalk stayed on the sidelines for years; 'serious' musicians refused to give their software any credit, mostly because of the Windows platform. Yet, slowly, a group of dedicated Cakewalk users started to grow. They enjoyed the low cost of the PC and found that Cakewalk's products did the job quite well. It's not a very informative message, but at

Eventually, Apple faltered, providing a least you know something's wrong. window of opportunity for Cakewalk's Pro Audio line. Apple have come back strongly of late — but before they renewed their commitment to music, many who had just discovered computer recording cut their teeth on Windows, while some musicians crossed to the PC after frustrations with the Mac. Sonar's introduction marked a turning point for the company, and PC users. The tide in the US started ebbing away from Logic and Cubase: Logic became Mac only, while Cubase was acquired by Pinnacle, who, it seemed, couldn't quite figure out what to do with it (happily, Steinberg have now found a home with file:///H|/SOS%2005-06/Cakewalk%20Z3TA+%20Bandlimited%20Waveshaping%20Soft%20Synth.htm (1 of 3)9/28/2005 2:41:53 PM


Yamaha). Sonar continued to grow in popularity, and a partnership with Roland provided extra muscle with respect to distribution and product development.

Musikmesse 2005 Cakewalk then set their sights beyond the boundaries of the US — and it seems that 2005's Frankfurt Musikmesse marked a turning point for those efforts. The push for internationalisation started in late 2004, when Sonar 4 (both Producer and Studio Edition) started shipping in a combined French/German/English version. At the Messe, Cakewalk added Spanish to the list. (And credit where it's due: Sound On Sound were ahead of the crowd in adding a Sonar Notes column to their line-up of sequencer-related editorial offerings back in January 2002.) Significantly, Cakewalk also took first place for recording software in the prestigious Music Industry Press Awards (MIPA) at Messe 2005, beating both Apple Logic Pro 7 and Ableton Live 4. This was quite an achievement, given that Sonar wasn't even nominated last year. For those not familiar with MIPA (now in its sixth year), the winners are nominated and chosen by the editors of 58 music magazines from all over the world. Many consider it to be the 'purest' award scheme, as neither reader votes nor advertiser pressure are elements in the voting process. So, Sonar fans, don't let it bother you if some isolated gear snob sneers at you for using "that Windows program." The rest of the world is starting to catch on.

Z3TA+ Synthesizer Now Available Soft-synth connoisseurs have long lauded rgc:audio's Z3TA+ soft synth, which combines synthesis, processing, arpeggiation and more in a package with a distinctive, clean, almost metallic sound. Now that Cakewalk have acquired rgc: audio, the Z3TA+ is being marketed as a separate product. The Z3TA+ is not a virtual analogue synth, although its layout and rgc:audio's original version of the Z3TA+, a parameter selection will not be popular download among those seeking new sounds. Cakewalk's version retains the unfamiliar to those versed in features of the original. conventional synthesis. Cakewalk call the synthesis engine 'Bandlimited Waveshaping technology' — basically, a variation on 32-bit wavetable synthesis, but with extensive waveshaping options that add a 'signal processing' feel to the basic oscillators. The bottom line is that the sound quality cuts well, and the synth can produce FM-type sounds as well as fairly lush, evolving pads. Another nice feature is 19 stereo amp cabinet simulations — useful file:///H|/SOS%2005-06/Cakewalk%20Z3TA+%20Bandlimited%20Waveshaping%20Soft%20Synth.htm (2 of 3)9/28/2005 2:41:53 PM


even if you're not going after a guitar sound.

Squashing A MIDI 'Bug' Occasionally I see comments in web forums about Sonar's MIDI 'bugs'. But dig a little further and it's almost invariably pilot error. One of the most common complaints occurs because of an incorrect choice under Drag & Drop options (something we'll cover in a later issue). However, another of these so-called bugs occurs when you drag a note to extend it and strange things happen — sometimes additional notes show up and sometimes they don't, but when you release your mouse a box appears that says "This editing operation is not allowed." (See screen, left.) What it should say is "You're editing a MIDI groove clip and trying to extend a note past the loop boundaries" — because that's what's happening. For example, say you have a MIDI groove clip that lasts four bars, you have it 'rolled out' to cover 16 measures, and you're looking at this in the Piano Roll view. There's a note at the third beat of the eighth measure and you want to extend it past the beginning of the ninth measure. But you can't, because there's a MIDI groove clip loop boundary at the start of measures five, nine and 13. If you try to extend that note, you're actually trying to wrap it around to the beginning of the loop, and that's not possible. The solution is simple: turn off MIDI groove clip looping, copy and paste the four-bar pattern to fill up the appropriate number of measures, then do your edits. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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In this article:

Neanderthal Man Bouncing Around Maximum Isolation Blunt Instruments Share And Share Alike The Icing On The Cake Persistence Rewarded

CLASSIC TRACKS: 10cc 'I'm Not In Love' Producers: 10cc; Engineeer: Eric Stewart Published in SOS June 2005 Print article : Close window

Technique : Recording/Mixing

Disagreement can be destructive, but it can also drive a band on to new heights. So it was when 10cc's Kevin Godley turned up his nose at a love song penned by Eric Stewart and Graham Gouldman, insisting that it would have to be completely reinvented in the studio... Richard Buskin

With the recording of their third album, 1975's Original Soundtrack, 10cc truly hit their stride. The Mancunian quartet had formed five years earlier as a session outfit named Hotlegs, comprising singer/guitarist Eric Stewart, a former member of the Mindbenders, both with and without Wayne Fontana; singer and guitar/bass player Graham Gouldman, ex-Mockingbird and composer of songs for the Yardbirds, the Hollies, Jeff Beck and Herman's Hermits; and former art students and multi-instrumental sessioneers/singers Kevin Godley and Lol Creme. It was in 1968, after the Mindbenders had disbanded, that Eric Stewart bought into a little demo studio in downtown Stockport. This he did by generously purchasing the facility a Neumann valve U67 as his part of the deal, and thereafter, always fascinated with the control room when he'd recorded as part of a group, he learned the ropes of engineering by way of numerous demos tracked to a pair of TEAC stereo machines. "I loved the early Presley stuff, the Sun records," he says. "There's a distortion that's really human and very, very nice and warm; like a good valve mic with a vocalist. It has this gorgeous edge on it which, if you stuck it through an file:///H|/SOS%2005-06/CLASSIC%20TRACKS%20%2010cc%20%27I%27m%20Not%20In%20Love%27.htm (1 of 13)9/28/2005 2:42:00 PM


oscilloscope, you'd say 'No, no, that looks bad. It's ripping at the edges.' However, it sounds really great to me, and I wanted that sound. "During the early to mid-'60s, the studio was where the musicians were and the control room was always hallowed ground. You were never allowed in there. 'No, no, no, boys. We'll let you come and hear the mix when it's finished.' I'd go in and thrill to the sound — and I thought to myself 'I've got to get my hands on one of these.' It was a dream, but I knew I'd get a place eventually."

Neanderthal Man Now in that place, located above a hi-fi shop, the new studio owner soon learned his establishment was being given the boot to make way for a works canteen, so he had to find new premises. Next stop: an ex-armaments factory, also in Stockport, housing a crude custom-built control desk and brand-new four-track Ampex recorder. "We got the local bank to loan us some money to rent the premises and buy the gear," Stewart recalls. "The Ampex was installed and the first thing we recorded will be on my bloody gravestone. It was an instrumental called 'Neanderthal Man'. Basically, I was experimenting to see how many drum tracks we could squeeze onto the Photos: Eric Stewart Ampex's four tracks, and I was in there 10cc in Strawberry Studios, 1975. From left: doing it with Kevin Godley and Lol Lol Creme, Kevin Godley (at rear), Eric Creme. Not a group really, nothing at Stewart, Graham Gouldman. that point in time; just a bunch of musicians messing around. Well, to keep Kevin in time on the drums, Lol just sat near him on a stool, off mic, and sang 'I'm a Neanderthal man, you're a Neanderthal girl, let's make Neanderthal love in this Neanderthal world.' "When we got four tracks of this thing down, Lol's little vocal in the background of each drum track began to sound like a distant chant, and there was something very hypnotic about it. Dick Leahy came into our studio just after our experiment to do a demo with Mary Hopkin. He used to be one of the A&R guys at Phonogram when I was with the Mindbenders and I knew him quite well, so when he asked 'What have you guys been doing recently?' I said 'No songs yet, but have a listen to this. It's an instrumental.' And I played him 'Neanderthal Man'. Well, he nearly fell off his chair and said 'Jesus Christ, that is a smash! I'll buy it now. I will do you a deal right this minute.' So, he offered us a very good deal and we got a number two record out of it. We added a Moog solo and a few more percussive things, but it was still just a studio experiment that ended up giving us some more, much-needed money with which we could buy our first 'professional' file:///H|/SOS%2005-06/CLASSIC%20TRACKS%20%2010cc%20%27I%27m%20Not%20In%20Love%27.htm (2 of 13)9/28/2005 2:42:00 PM


piano-styled Helios desk, designed by Dick Swettenham, as well as upgrading to an eight-track, one-inch Scully tape machine." It wasn't long before pop impresario Jonathan King spotted the commercial potential. After Graham 'GiGi' Gouldman joined the trio on bass, King signed Hotlegs to his own UK label and rechristened the group 10cc (purportedly the metric amount of semen ejaculated by the average male). A hit UK single followed in 1972 with '50s doo-wop satire 'Donna', featuring Lol Creme's piercing lead falsetto, and further domestic success ensued with the singles 'Rubber Bullets', 'Wall Street Shuffle', 'Silly Love' and 'Life Is A Minestrone', as well as a pair of long-players, 1973's 10cc and 1974's Sheet Music. However, it wasn't until the aforementioned third album and the release of 'I'm Not In Love' that the band at last achieved their American breakthrough. In light of this, it's interesting to learn that this landmark track, with its ethereal feel, lush production, multilayered vocals and slyly affectionate lyrics, started life as a Brasil '66-style bossa nova number. Talk about what might — or might not — have been...

Bouncing Around "At that time my wife and I had been married about eight years," Eric Stewart recalls, "and she asked me 'Why don't you say "I love you" more often?' I had this crazy idea in my mind that repeating those words would somehow degrade the meaning, so I told her 'Well, if I say every day "I love you, darling, I love you, blah, blah, blah," it's not gonna mean anything eventually.' That statement led me to try to figure out another way of saying it, and the result was that I chose to say 'I'm not in love with you,' while subtly giving all the reasons throughout the song why I could never let go of this relationship." Evidently, the reverse psychology worked, because the Stewarts recently celebrated their 39th wedding anniversary. "I had the guitar hook first —a little arpeggio on an open 'A' chord — and the melody kept going through my head, so when I got the idea to write the words 'I'm not in love' it just sort of slotted together," he continues. "Once I'd clicked on the idea to approach it that way, it was actually very easy to write the rest. I made things fit phonetically, and it just sort of rolled out very smoothly in a bossa nova shuffle. You know, Stan Getz, Astrud Gilberto. "So I had the first six chords or so of the verse figured and I had the melody

The Helios desk installed in Strawberry Studios for the recording of 10cc's third album was a custom wraparound design featuring 24 input channels (centre and right) plus monitoring and processing on the lefthand side.

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already figured in my head, as well as the first verse lyrics 'I'm not in love, so don't forget it, it's just a silly phase I'm going through...' so I took this to the studio, played it to the other guys and asked 'Would anyone like to finish it with me?' GiGi, the bass guitarist, said he would. We usually wrote in pairs, and while the major hits came out of Godley and Creme or myself and Graham Gouldman, we were a very incestuous bunch — we used to swap partners all the time. I wrote 'Life Is A Minestrone' and 'Silly Love' with Lol, and we did swap around a lot just to keep the writing freshness going. It worked beautifully for us. Anyway, at that time Godley and Creme were writing the mini musical 'Une Nuit à Paris' [which would open the album], so they went into one room to finish that, and GiGi and I went into another to work on the 'I'm Not In Love' idea with two guitars. We developed the song pretty quickly. "We started with the chords that I already had and we began bouncing ideas off each other. We were both very good at steering something away from the norm, looking for another way to make the chords move so that it didn't become a 'regular' pop song... or a regular bossa nova, for that matter. We would say 'What about this?' 'Nah, I don't like that.' 'What about this?' 'Nah.' 'What about...' 'Ah! Now that's got something.' It's just tangential thinking, bouncing off each other — it's very productive if you've got two competent musicians working in this way. I usually wrote on the keyboard, but 'I'm Not In Love' was written on two guitars, and the ironic thing is that there is no featured guitar on the finished product, just a little DI'd Gibson 335 playing a light rhythm pattern. In the end, we must have spent about two or three days writing before completing it. "There's no middle eight or chorus in 'I'm Not In Love'. GiGi came up with this lovely little fill in between verses: an open 'E' string with the chords moving from 'E' to 'A' to 'G', with this 'E' bass string ringing through it — very, very tasty. I eventually played that on the recording with a Fender Rhodes. A beautiful progression with a beautiful sound. He also came up with the opening chords of the song, an 'A' chord with a 'B' bass, moving to a full 'B' chord, all very 'expectant' of what is to follow. Then we wrote a second verse and, because we thought this was going to be something different, we also wrote what could be termed as a middle eight quite early on in the song. We got the melody for that very, very quickly, but the words just sounded naff: 'Don't feel let down, don't get hung up, we do what we can, we do what we must.' We looked at each other and went 'Oh Christ, that sounds crap, doesn't it?'" Crap or not, those words remained 'in' for now. Otherwise, since Stewart had a lot to say to his wife, he was responsible for about 90 percent of the lyrics, and these included those for the third verse which he and Gouldman composed before coming up with a bridge: 'Ooh, you'll wait a long time for me...' This refrain was repeated four times, before leading back into the last verse, a repeat of the first.



"GiGi contributed quite a lot on some of the chord changes to take them away from what I'd originally figured," Stewart remarks. "He'd take them in a different direction. There were, as I said, these little fills in between the verses, and he also pulled that chord progression for the 'Ooh, you'll wait a long time' bridge out of the bag; a nice arpeggio run-down. When you've been writing with somebody for quite a long time these things just sort of happen naturally, and you instinctively know whether or not someone's getting off on it. They don't necessarily have to come up with something new, just that look that says 'Hmmmm... no!' and you find yourselves searching for something that will turn you both on. That's the chemistry. Godley and Creme, however, wrote in a very different way." And they could also think differently. For, when Stewart and Gouldman first played them their sub-four-minute bossa nova classic, Kev and Lol were a little coy: "Yeah, yeah, it sounds nice, it sounds a bit cute. But hey, let's try it!"

Maximum Isolation 10cc's second album, Sheet Music, had been recorded with the eight-track Scully tape machine and Dick Swettenham-designed Helios desk that had been financed by the success of 'Neanderthal Man'. For the recording of Original Soundtrack, a 16-track 3M machine replaced the Scully, and Stewart had his sights set on a wraparound desk, so he designed one in conjunction with Dick Swettenham. This featured 16 channels in the centre, eight more on the right-hand side, and monitoring and outboard gear on the left. "The outboard gear wasn't all 19-inch rack stuff," Stewart explains. "There were these little compression and EQ units made by Audio & Design and API, just useful 'extra' bits from everywhere; graphic equalisers, you name it. Even a couple of Neve compressor modules eventually! Still, there were things about the first Dick Swettenham desk that I preferred. Sure, in retrospect it was a bit rough and ready, but while the mic amps were a little crude, they were also a bit tasty if you messed around with them. I used to use the line-amp side of them for all our guitar solos. You pushed them into distortion and that desk had the most beautiful, fine distortion that you couldn't get on an overdriven Marshall or with a fuzzbox. It was gorgeous, fabulous, unique. "When I ordered the new wraparound desk, Dick Swettenham said 'I've improved the mic amps, Eric, I think you'll be pleased with them.' Well, I plugged my guitar in, pulled down the fader, wound up the line amp, and the thing sounded so brittle. It just cracked and ripped and it was bad, unusable. We'd sold the original black desk to somebody local and I desperately tried to get at least one or two of the mic/line modules back, but the bastards wouldn't sell them to me. That meant we had to develop other ways of creating this lovely, smooth 10cc sustained guitar sound." Strawberry's 18 x 18-foot control room, housing JBL monitors, had a seating area at the back and three Studer stereo machines in front of the desk, below a window that looked out towards the spacious 60 x 30-foot live area. Having been converted from an armaments factory, the place boasted iron pillars supported by iron posts, creating the effect of what Stewart describes as "a giant Meccano set... We had to clad all the pillars in carpet. The room itself hadn't been acoustically file:///H|/SOS%2005-06/CLASSIC%20TRACKS%20%2010cc%20%27I%27m%20Not%20In%20Love%27.htm (5 of 13)9/28/2005 2:42:00 PM


designed by anybody. We just insulated it and did what we thought was right, and it seemed to sound OK. The recordings translated quite well when we took them away to cut. "The drum booth was mostly created with portable screens on wheels, which we made ourselves. Heavily influenced by Steely Dan's 'dry studio sound' at the time, we were very much into close-miking and very, very tight, close drum sounds. In fact, there was very little room on any of our sounds. I remember Glyn Johns walking in and saying 'Fuckin' hell! What is this? I'm not hearing anything. I want to get some air around my sound. This is terrible. It's too dead.' We were going in totally the opposite direction. We were also using Dbx noise reduction instead of Dolby, again because I read that the Steely Dan team were using it. I loved it, it added a nice little frisson of compression that enhanced the overall sound as well." So it was that Kevin Godley's oyster-shell Ludwig kit was miked with Neumann U87s overhead, a D12 in the bass drum, a Shure SM57 under the snare, a Neumann KM84 on the hi-hat also picking up some brightness off the snare to complement the thud of the 57, and all five tom-toms very closely miked with Beyer M88s to achieve maximum isolation and dryness of sound. "I'd previously used a five-mic setup on the drums, but never close on the toms like that," Stewart says. "It was all about the Steely Dan sound. I adored it. I adored that close miking. It sounded like everything was right next to you. There was no 'room' around it. You were almost inside the kit itself, and there was a phenomenal sound on the vocals, too. It was so precise, although I think they eventually became a little too obsessive with timing via oscilloscopes and so on. It became almost clinical. But I did like that close-miked sound on the earlier stuff."

Blunt Instruments All four began recording the song immediately, with Eric Stewart behind Strawberry's new Helios desk (see box, left). While Stewart engineered, the other three recorded as a band, with Gouldman's Rickenbacker bass DI'd and Creme's Les Paul going through a Marshall 50, played at a low level and miked with a valve U67. "We were always very blunt with each other," says Stewart. "We recorded everything we came up with, but we were very brutal at the end of it, saying things like 'Is this working?' or 'Do we like this? Is this gonna fit? Yes or no?' Out of four people we needed a majority of three votes to say 'Yeah, we carry on,' or 'Yeah, it's going on the album' or 'No, it's out.' Well, we recorded 'I'm Not In Love' as a bossa nova and Godley and Creme didn't really like it! Kevin was especially blunt. He said 'It's crap,' and I said 'Oh right, OK, have you got anything constructive to add to that? Can you suggest anything?' He said 'No. It's not working, man. It's just crap, right? Chuck it.' And we did. We threw it away and we even erased it, so there's no tape of that bossa nova version. It pissed me off no end at the time, but it was also very democratic, as I've said, and so we turned our attention to the recording of 'One Night In Paris'. "At the studio we had various staff, including a secretary, Kathy Redfern, and another engineer, Pete Tattersall, who used to spell for me when I was singing or



playing in the studio itself. He was my partner when we built the first four-track studio, and the poor bugger used to have to do all the Granada TV sessions — you know, Coronation Street and Muriel Young's Five O'Clock Club; all these weird things that used to come in with proper MU orchestra musicians. I didn't get along with all that, so Peter got lumbered with it while I got to do the interesting tasty bits. Anyway, walking around Strawberry each day, I kept hearing people singing the melody: 'I'm not in love...' And I kept going back to the band and saying 'There's something more with this song. We've not got it yet, but I don't want to lose this song, because it's got people hooked.'

Lol Creme at the grand piano, 1975.

"Then the secretary Kathy said 'Why didn't you finish that song? I really love it. It's the nicest thing you've ever done.' This didn't really impress Kevin, of course, but we discussed it again, and believe me, it was Kevin who suddenly came up with the brainwave. He said 'I tell you what, the only way that song is gonna work is if we totally fuck it up and we do it like nobody has ever recorded a thing before. Let's not use instruments. Let's try to do it all with voices.' I said, 'Yeah. OK. That sounds... different.' A cappella, vocal instrumentation is what he was talking about. "I said 'Well, we're gonna need something instrumental in there to sing the whole backing track to,' and he said 'Yeah, we'll keep a rhythm going with something simple; a bass drum, whatever. We can have a guitar just giving us chords, but otherwise it could be all voices.' I said 'Right, there's just four of us to do the whole thing with voices. How are we going to do it?' And it was Lol who then said 'What about loops? Tape loops. Endless voice loops. We can make endless loops of a chromatic scale.' I said 'Right. OK, Jesus, this is really off the wall.' "I think they'd been at the wacky baccy at this time, and it took me a couple of hours to get my head around the idea. But then I figured how we could physically make the loops and set up the studio to do that. I rigged up a rotary capstan on a mic stand, and the tape loop had to be quite long because the splice edit point on the loop would go through the heads and there'd be a little blip each time it did. So, I had to make the loop as long as I could for it to take a long, long time to get around to the splice again. That way you wouldn't



Strawberry Studios secretary Kathy really hear the splice/blip. We're talking Redfern, whose enthusiasm for 'I'm Not In about a loop of about 12 feet in length going around the tape heads, around the Love' helped to persuade the band not to scrap it, and who eventually supplied the tape-machine capstans, coming out whispered female vocal. away from the Studer stereo recorder to a little capstan on a mic stand that had to be dead in line vertically with the heads of the Studer. It was like one of those continuous belts that you see in old factories, running loads of machines, and we had to keep it rigid by putting some blocks on the mic stand legs to keep it dead, dead steady.

"It worked, but the loop itself — and this is where it gets interesting — had to be made up from multiple voices we'd done on the 16-track machine. Each note of a chromatic scale was sung 16 times, so we got 16 tracks of three people singing for each note. That was Kevin, Lol and GiGi standing around a valve Neumann U67 in the studio, singing 'Aahhh' for around three weeks. I'm telling you; three bloody weeks. We eventually had 48 voices for each note of the chromatic scale, and since there are 13 notes in the chromatic scale, this made a total of 624 voices. My next problem was how to get all that into the track. "I mixed down 48 voices of each note of the chromatic scale from the 16-track to the Studer stereo machine to make a loop of each separate note, and then I bounced back these loops one at a time to a new piece of 16-track tape, and just kept them running for about seven minutes. Because we had people singing 'Aahhh' for a long time, there were slight tuning discrepancies that added a lovely flavour, like you get with a whole string section, with a lot of people playing. Some are not quite in time, some have slightly different tuning, but musically a lovely thing happens to that. It's a gorgeous sound. A very human sound, very warm and moving all the time. Anyway, after putting the 13 chromatic scale notes back onto the 16-track, it meant there were only three open tracks left! "On one mono track we put a bass drum and me playing the Fender Rhodes piano as well as bumbling a guide vocal very, very crudely, just to keep the song's timing. Kevin actually did the bass drum using a Moog bass note; a funky sound with a little edge on it, a little click almost. The timing had to be perfect, with no metronome! Then, all four of us manned the control For the a cappella backing of 'I'm Not In desk, and each of us had three or four Love', the group multitracked themselves faders to work with. We moved the singing each note of the chromatic scale to a 12-foot tape loop. These were then recorded faders up and down and changed the onto 13 tracks of a 16-track reel, and the chords of the 13 chromatic scale notes band then pushed faders to 'play' chords. as the chords of the song changed — 13 tracks on a 16-track tape, fed through the control desk faders, back out of the master fader and onto that stereo pair of open tracks that was left free on the 16-track machine. It took a long time



before we thought we'd got something really interesting, but blow me down, if we hadn't got it right we would have been buggered, because at the point in time when we had that stereo pair of the whole backing track mixed down, I would have to erase those 13 continuous voice notes in order to give me 'clean' tracks to start doing the real vocal, the answer backing vocals, the bass solo, the grand piano solo and the rhythm guitar, which was just a DI'd Gibson 335. "Luckily we got it. We got it just right. We very, very quickly got the lead vocal down and then we sat there, I tell you seriously, for about three days, just listening to this thing. I was looking at Kevin and the other two guys saying 'What the fuck have we created? This is brilliant.' We knew we had something very, very special, very different. I'd never heard anything like it in my life. I mean, the Beach Boys were seriously good at harmonies, but they hadn't, as far as I knew, done anything this way. It was a very, very unusual sound. And sound degradation caused by all the bouncing didn't matter at all because, when each of us were using control desk faders to mix the voices, there was a piece of gaffer tape across the bottom of the fader paths to stop them ever going to the bottom. That meant we had a chromatic scale sizzling underneath the track all the time, a hiss just like the hum you sometimes hear at a football match when nobody's shouting. If you listen to the opening of the song, where the bass drum beats us in, you will hear a sizzling hum there that continues all the way through the track. We actually created 'hiss' on the track, when we would normally have been fighting to get rid of hiss! "Some of the low voices on there sound like 'cellos. If you slow the loop down to 7.5ips, a human voice sounds amazingly just like a 'cello. It's got the rasp from the throat that sounds like the rosin on a bow swiping across the strings. Unbelievable. We used the sound on that album quite a lot. There's a track called 'Blackmail' where you hear 'cellos chugging very, very fast in rhythm all the way through, and that's the voice loops slowed down to 7.5 fed through two faders, which were pushed up and down rhythmically. It was wonderful the way it worked."

Share And Share Alike "An interesting 'deal' we had with 10cc, until Godley and Creme left us in 1977, is that we agreed that we would share the record royalties four ways, no matter who wrote the songs," says Eric Stewart. "This was a great way of only using the best songs, and everybody in the band would benefit. I had seen and heard many albums where, for instance, 'the drummer', bless him, had to have as many songs on the albums as the 'real writers' — and my God, they were usually crap, and pulled the album down."

The Icing On The Cake Not surprisingly, the song would only take half a day to mix. In the meantime,



however, Eric Stewart had to track his lead vocal, and this he did with his usual valve U67, recorded dry with no reverb or EQ, and certainly with no comping. "There's a very different vibe to somebody singing from start to finish," he says. "You get the whole feeling of the song together. I can spot comping a mile off. So we didn't comp it. It's not a difficult song to sing. I got it down in one and then dropped in to correct a few mistakes. The little high answers at the end of the verses where it goes 'It's because...' — Lol and Kevin could do that. They had great high voices, those two, so I multitracked them about four times on those lines. "At this point there was no bass on the track whatsoever. The left-hand side of my Fender Rhodes was providing the bass notes — I played them in octaves with my left hand, which is how I normally play keyboards, and that was enough. It didn't need a bass guitar. But again, there was another unusual idea suggested: why don't we try a bass solo? A bass solo in a ballad? Bloody stupid you'd think. However, it did fit beautifully. It's all about searching for something that hasn't been done before, and believe me, we sometimes spent days, sometimes weeks searching for sounds that we thought were different. The original master tape of 'I'm

"Kevin and Lol were pretty much the ones who Not In Love'. would always say 'I want to do something different here!' They would, for instance, look for a bass drum sound and we'd go around the studio, banging everything from the floor to the wall, and even if we later returned to a straightforward bass drum, at least we'd tried. And more often than not we'd get something that was quite unusual. Now, if you've got the luck to convince the public that it's unusual and good as well, you've cracked it, and 'I'm Not In Love' did just that. "For the bass solo, GiGi came into the control room, I DI'd his Rickenbacker through one of those lovely Dbx 160 compressors to keep its gorgeous, round, thumping sound tight and smooth, and he played the solo. We sat there and he played bits, and we said 'Like that,' 'Don't like that,' 'Do that again,' and it developed. When we got that down, the song was, to all intents and purposes, finished, but again we sat there listening to it, wondering what else we could do to 'screw' this song up. That's the way it was beginning to look to me. "I'd been reading a book about the philosophy of getting your point across in an argument. So, we each had a sign that we crayoned on a piece of cardboard, and if we didn't like something happening in the studio we'd hold the sign up to the window, saying 'Stop' or 'Next'. Then, when the person came back in, he'd hold up the sign that said 'How dare you!'"



This would be the title of the band's next album. "The theory was 'How dare you put me down before you've even spoken to me about why I'm doing this. You've got to let me try my thing before dismissing it,'" Stewart continues. "We cracked up employing this idea, it lasted on the one album. Kevin kept doing that, though. He was off on a tangent somewhere, but he had some great ideas because of it, so you couldn't stop him. He might just come up with that little bit of gold dust that you sometimes needed on a track if it wasn't going anywhere. Well, when we listened to 'I'm Not In Love', he kept saying 'It's not finished, it's not finished,' and I remember saying 'What do you want to try next? A fucking tambourine solo in the middle of it? What do you want?' We kept thinking and kept thinking, and Lol remembered he had said something into the grand piano mics when he was laying down the solos. He'd said 'Be quiet, big boys don't cry' — heaven knows why, but I soloed it and we all agreed that the idea sounded very interesting if we could just find the right voice to speak the words. Just at that point the door to the control room opened and our secretary Kathy looked in and whispered 'Eric, sorry to bother you. There's a telephone call for you.' Lol jumped up and said 'That's the voice, her voice is perfect!' "We got Kathy in the studio just to whisper those words, and there it was, slotted in just before that bass guitar solo. And it fitted beautifully. Again, another little twist of fate, an accident that wasn't on anybody else's songs. We'd never heard that before. It just clinched it and made the song even more original. Poor Kathy was bemused. She didn't want to go in the studio, we had to drag her in, but she was very, very sweet and we Kevin Godley at the drum kit during the eventually persuaded her: 'You've just recording of the Original Soundtrack album, got to whisper. Just whisper, don't where close miking and portable screens worry. You're not singing, just talking. were used to create a very dead sound. Use your best telephone voice.' She had a gorgeous voice, and there it is; it's on the record... and she got a gold record for it, too." There was one further addition. "The last thing recorded on 'I'm Not In Love' was a child's music box over the fade out. We sent the secretary out to buy a simple plastic one, attached it to a piece of string, and Lol sat at the drum kit and whirled it slowly over his head while I recorded it on the overhead drum mics." Given Kevin Godley's initial reaction to 'I'm Not In Love', it may seem a little surprising that he was subsequently willing to not only give it another chance, but also to give it sufficient consideration to come up with such innovative ideas. However, it wasn't the song that he hated as much as its initial sugary samba arrangement, and despite having been turned off it, he was eventually swayed by file:///H|/SOS%2005-06/CLASSIC%20TRACKS%20%2010cc%20%27I%27m%20Not%20In%20Love%27.htm (11 of 13)9/28/2005 2:42:01 PM


its popularity among the studio staff. As Eric Stewart surmises, "He must have just sat there thinking 'How can we do it and make it different? How can I not make it schmaltzy?' And he figured it with the a cappella idea. It was great. A lovely piece of chemistry coming from his head."

Persistence Rewarded Regardless, when the record company execs first heard 'I'm Not In Love', they gushed over its sound but balked at any suggestion that it might be a single. To their way of thinking, it was a ballad and, at just over six minutes in length, it was too long for radio airplay. A case of "Yeah, bloody hell, fantastic track. Let's go with 'Life Is A Minestrone'..." Which they did. And 'Minestrone' was indeed a UK hit. However, Eric Stewart then received many telegrams, including one from that king of overkill, Roy Wood, singing the praises of 'I'm Not In Love' and urging him to get it released. Others joined the chorus, even employees of other record companies, but Mercury still demurred until changes were made. "The BBC asked me to edit it," Stewart recalls, "and I said 'No, I can't. What am I going to take out?' As far as I was concerned, it was like taking half of a masterpiece portrait painting out. What bit do you want to cut off? The head? Eventually, however, they persuaded us to take the piano and bass guitar solo out of the middle and cut the song down from 6:06 to 4:10. The solo must have been about one and a half minutes, and then we cut the fade-out, which is quite long as well, by 30 Graham Gouldman and Lol Creme, seconds or more. Well, the record Strawberry Studios, 1975. charted, and by the time it got to number 28 and after pressure from the public and the media, the Beeb started to play the whole thing. And then it went to number one. Justice. It's a good song. I'm very proud of it, as well as the way we all got it together." It was, in effect, the song that wouldn't die: the survivor of Godley's disapproval, record company indifference and radio station pickiness. While the Original Soundtrack album spent 25 weeks on the Billboard charts after its release in the spring of 1975, 'I'm Not In Love' became a worldwide smash, not only topping the UK charts but also reaching number two in the US. Nevertheless, following 1976's How Dare You!, Godley and Creme departed 10cc to focus on video production and their development of the Gizmo guitar-sustain device. Stewart and Gouldman kept the band alive with the assistance of other musicians and enjoyed further success until calling it a day in 1983. Eric Stewart now markets file:///H|/SOS%2005-06/CLASSIC%20TRACKS%20%2010cc%20%27I%27m%20Not%20In%20Love%27.htm (12 of 13)9/28/2005 2:42:01 PM


his music, including new album Do Not Bend, via the www.ericstewart.uk.com and www.strawberrysoundtracks.com web sites. Following the demise of 10cc he also produced Sad Café and ABBA's Agnetha, and worked with Paul McCartney for several years, before helming and participating in two new 10cc albums during the early '90s: ...Meanwhile and Mirror Mirror. The first of these reunited all four original members. "But it was like trying to reheat a soufflé," says Stewart, who has recently been working in France's Dordogne region, producing Spanish-born Belgian artist Pascal Escoyez in a stunning subterranean studio that is partly built into a hillside. "And to be honest with you, ...Meanwhile was more of a contractual thing. It was the deal to get Godley and Creme out of their contract with Polygram. They said 'If you work with Eric and Graham again we'll let you go.' So, that's what we did and, sadly, it was all a bit of a compromise, in spite of having Jeff Porcaro on drums throughout, and Dr John's piano and vocals on one track. "Years ago I remember Kevin being asked 'Why did you split? Why did you leave 10cc?' and he said 'Because I didn't want to do any more crap like "I'm Not In Love".' Blow me down. I was stunned. However, I heard recently — because we were awarded all these lovely Ivor Novello Awards last year for the block success of our work in the '70s — somebody asked him again and he said 'Yeah, yeah, I didn't like the song at the time, but I wished I'd written the fucking thing.' Because, you know, it's been the most successful track we've ever had released." Indeed. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Demo Doctor

In this article:

Malaya Doctor's Advice: Running Orders Monday 6am QUICKIES

Demo Doctor Reader Recordings Analysed Published in SOS June 2005 Print article : Close window

Technique : Recording/Mixing

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

Malaya Venue: Home Equipment: PC running Steinberg Cubase sequencer, Halion sampler and Halion String Edition orchestral instrument, AKG C3000 mic, Dbx Mini Pre valve preamp, Klark Teknik Jade monitors.

Track 1 1.4Mb Track 2 1.4mb Track 3 1.4Mb

This material leans towards R&B in the vein of Jennifer Lopez and Destiny's Child and draws on a wide range of influences, from soul to jazz-rock. Indeed, the inclusion of an excellent jazz track at the end of the CD and the standard of musicianship throughout suggest that the duo of Graham Pettican and Malaya may work in that area of music for a living. The first few tracks use overdriven electric guitar samples, skilfully played from the keyboard as melodic hooks between the sung lines. Reminiscent of Santana, this sound is clearly a favourite of Graham's but would have more impact if the mixes featuring it were spaced further apart on the CD. A nylon-stringed acoustic could have occasionally been used instead, and would have been in keeping with the R&B genre. Having said that, Graham does use some steel-stringed acoustic guitar samples, and he might also consider taking the same approach with clean electric guitar samples, as heard in recent Ashanti tracks. The songwriting, arranging and performances are of a high standard throughout file:///H|/SOS%2005-06/Demo%20Doctor.htm (1 of 5)9/28/2005 2:42:11 PM

Demo Doctor

and Malaya's voice sounds superb. This is a testament to the quality of the vocalist on a microphone (the AKG C3000) many would regard as an entry-level condenser. For example, on the second track, 'Reach Out', the C3000 in combination with a Dbx valve preamp provides a clear vocal sound, but with enough edge to cut through the full-sounding keyboard textures during the choruses. I was pleased to hear that the quality of the material holds up well throughout most of the album and the few vocal effects used, all of them standard for the genre, like filtering, modulation and MIDI-triggered Auto-Tune, are handled with confidence. They're also used quite subtly, which seems more in line with the female R&B production style. I should also mention the wonderful backing vocals, both single lines and tracked-up harmonies, which were performed by Malaya herself. Where more conventional instrumentation is used — the Hammond organ, piano and lead guitar on track four, for example — the string arrangements and hip-hop beats keep the sound firmly in the present. Add to this the effective use of looped 16th-note patterns with a variety of effects like filter and wah, and you have an accomplished piece of work by a duo working in a highly competitive but immensely popular area of the market. www.malaya.me.uk

Doctor's Advice: Running Orders The work of the artist recording at home doesn't stop with mixing and mastering. After post-production tweaking, there's also the running order of the tracks on the CD to consider. Some impact is lost when you put too many tracks with similar instrumentation together, or if the tempo and key of adjacent tracks are the same. You'll obviously want to include your best compositions and it's always worth testing the water by playing a selection of tracks to a few people first and gauging the response. Your best live track may often turn out to be a bit lifeless in the studio, just as a song you never really had much hope for can turn out to be a little gem.

Monday 6am Venue: Home Equipment: PC running Emagic Logic 5 sequencer and Sony Sound Forge editor, Focusrite Trakmaster voice channel, AKG C414 mic, Tascam M1600 mixer, Spirit Absolute 4 active monitors.

Track 1 1.4Mb Track 2 1.4mb Track 3 1.4Mb

This is a good listen and perfect for chilling out at home. From the opening filtered drum beat to the simple electric piano chords file:///H|/SOS%2005-06/Demo%20Doctor.htm (2 of 5)9/28/2005 2:42:11 PM

Demo Doctor

with vibrato moving from A minor seventh to D, it oozes relaxation. And then there's Kevin Jones's voice, a lesson in how to sing in a relaxed and mellow fashion. It's perfectly recorded using the AKG C414, a classic condenser microphone which seems to be a bit out of fashion these days, but which captures the presence of Kevin's voice very well. The opening song also features a sympathetic choice of instrumentation. In addition to the electric piano, there's a double-tracked acoustic guitar strumming with an electric guitar picking along in the background. Later, when the backing vocals and electric guitar solo kick in, the drum programming and synthesized loops give the track a contemporary edge. I like the fact that the production has been geared to suit the voice, and the general understatement adds to the laidback feel. The second track picks up the pace and proves that Kevin can layer up his voice effectively for multitracked harmonies. Again, a double-tracked acoustic guitar fills out the sound, and the rhythmic inconsistencies between the two takes, which are panned hard left and right, add interest for the listener. This simple rhythmic interaction of instruments shows up again on the third mix. On the playout (from about three minutes) you can clearly hear the way that the acoustic guitar chords emphasise the first beat of the bar and the electric guitar chords emphasise the third beat, which is an effective way to add a rhythmic swing to the track. Listening closely to the drum track there's also a slow brush stroke on the snare (or a programmed sound that's very much like one) which helps the rhythm by playing initially on the fourth beat, then playing on all four with an emphasis on the fourth beat. This kind of attention to rhythmic detail, although superficially simple, can be very effective and, in this instance, is very well thought out indeed. Although Kevin's songs are built around fairly ordinary chord sequences he manages to inject enough of his own character into them to give a new lease of life. The mixes sound very professional too.

file:///H|/SOS%2005-06/Demo%20Doctor.htm (3 of 5)9/28/2005 2:42:11 PM

Demo Doctor

QUICKIES

Dead Heroes Club This Irish progressive rock band set out to record as much of their demo as possible live in the studio. I checked out their web site and they have some good live reviews, which suggests a reason for this decision. However, though the band's playing is commendably tight, the bass end of the mix is a little indistinct. It lacks punch because both the bass and kick drum are in the same frequency range and there's no real attack to either sound. In this context, it is the kick which should be sharpened up a bit. The kick drum is one instrument that you can rarely leave with a flat EQ. Try using an EQ cut at 200Hz or so, allowing the bass and upper midfrequencies to cut through more. Elsewhere there is some excellent mixing and engineering. I particularly liked the start of the third mix, 'Sunrise On The Trenches', where the low notes on a five-string bass extend the frequency range of the mixes rather nicely. Post-production done at another studio adds a professional sheen to the finished CD, but I still think that a demo recorded live at a gig is more likely to capture the energy they're looking for. www.deadheroesclub.tripod.com

Hansom Pilot Hansom Pilot go for the trashy pop sound and succeed on the songs that use basic keyboard chord pads. Elsewhere, the mixes leave a big gap in the frequency range between the scratchy punk guitar sounds and the bass. This is rectified on the choruses of songs where mellow-sounding block backing vocals are used, because they tend to perform the same function as a low-level keyboard pad. The arrangements sound like live ones, and this demo will work as a promo to get more gigs, especially with the fine vocal performances of Bianca Denham. But a proper studio recording will require more keyboard ideas to enhance interest in the arrangements and add some light and shade to the dynamics. Some of the earlier albums by Elvis Costello, Blondie and the Stranglers would be good places to draw inspiration. I was surprised that the most commercial-sounding tracks were left to last. I'd suggest putting tracks five and six closer to the start of the CD. www.hansompilot.com

Kris Parker Kris likes us so much he sent two CD's separately! One concentrates on film soundtrack material while the other features his songwriting. The instrumental CD mixes are accomplished, and sound good both at a background listening level and right up loud. Kris also has a knack for sound composition, as the rhythmic industrial clanging and steam effects of the third mix demonstrate. What's also impressive about this track is the well-controlled mid-range. It can be so easy to over-equalise these industrial sounds in the mistaken belief that it will add clarity when a good sound balance is all that's required. As a singer Kris is more Depeche Mode than David Sylvian (two of his stated influences) and uses the same kind of cold reverb on his voice as the former quite effectively. However, a file:///H|/SOS%2005-06/Demo%20Doctor.htm (4 of 5)9/28/2005 2:42:11 PM

Demo Doctor

few lessons in the use of vocal range and pitching wouldn't go amiss because producing a good vocal tone isn't always enough.

Lisa Faye Lisa has also sent two CDs separately, which feature two different versions of a song called 'My Favourite Place', one in the style of power ballad and the other a dance track. Lisa's excellent, well-trained voice has been well recorded using a Rode NT1A mic and a Focusrite Voicemaster. In contrast to the vocals, the backing is as bland as those you purchase for 'live' solo work on the club circuit, and the key change up a tone at the end of the ballad stinks like the cheesy old thing it is! The anthemic house mix of the same song is better fare, although the arpeggios on the chorus should have been better defined and a bit more in time. Both mixes would be improved with backing vocals, and the dance mix needs a less dry vocal sound. Nevertheless, these CDs definitely show that Lisa is a versatile singer who can move between genres with ease. www.lisa-faye.co.uk

Monsieur Landslide The songs, arrangements and vocals are good even though this indie demo suffers from rather stiff drum programming. The same sampled snare hit placed at the front of the mix is a bit wearing, so try layering a different one on alternate beats for some dynamics and rhythmic variation. In contrast, the sloppy groove of the second song, 'Pebbles Fall Down On Me', works better because it has the rhythmic feel the others lack. A bit of an anthem with some well thought out lyrics, it was my favourite song on the demo. The guitars are well played but the acoustic sounds a bit thin and scratchy. While the vocal performances were good, too much compression has been added on mixdown, leading to some sibilance and a rather squashed sound. Try backing off the compressor a bit until the vocal sounds natural, but still dynamically under control. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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In this article:

Reason To Be Cheerful Plug-in Spotlight Introducing iZotope

Digital Performer Notes 3rd party developments Published in SOS June 2005 Print article : Close window

Technique : Digital Performer Notes

Now that the excitement surrounding the launch of DP 4.5 has subsided somewhat, developments by MOTU-friendly companies get this month's column under way. Robin Bigwood

The most important news of recent weeks from MOTU themselves has been the availability of updated OS X USB MIDI drivers, from www.motu.com. The version 1.3 drivers provide improved MIDI timing and efficiency, and an update of the Clockworks application that is now compatible with the Digital Timepiece synchroniser. The update is recommended for any users of MOTU MIDI interfaces, including the Fastlane. Obviously, if you're using another manufacturer's Core MIDI-compatible interface, or the MIDI sockets on a MOTU audio interface (such as the 828 MkII) you're not going to need these drivers.

Reason To Be Cheerful If you use Reason with Digital Performer, you're no doubt aware of the new version, Reason 3, which was released a few weeks ago. To some extent it's business as usual if you use Reason Rewired to DP, but the new Combinator device really makes a difference. It's much easier to drive multiple Reason devices simultaneously from DP, as you can select a Combinator as an output destination for a MIDI track in DP. The only thing you have to watch is that your Combinators are named appropriately, as all the devices they contain also show up in DP's output popups, and it's all too easy to end up driving something within a Combinator, rather than the Combinator itself. Sometimes you may want to do this, but it could lead to major confusion when working with big track counts file:///H|/SOS%2005-06/Digital%20Performer%20Notes.htm (1 of 3)9/28/2005 2:42:16 PM

Musolomo: strange but


and an even bigger Reason rack!

good.

Another thing to be aware of is the slightly altered Preferences window in Reason. Previously, when working with Reason rewired to DP you'd go to the 'MIDI' page within Preferences and select 'No MIDI Input', to make sure that it was only DP that drove Reason and that it was never being 'played' directly by your master keyboard. In v3, though, you can leave your 'Control Surfaces and Keyboards' set up, and instead deselect the 'Use with Reason' option for them before starting on a Rewired session. This is a much more elegant solution. Ultimately, the DP/Reason combo remains as useful and robust as ever, and it was a great pleasure, after I first installed Reason 3, to discover that joint DP and Reason projects opened and ran just the way they'd always done, with no hoopjumping whatsoever.

Plug-in Spotlight This month's prize for strangest plug-in goes to Musolomo, from www.plasq.com. Plasq are a collective of musicians and programmers including some familiar names: Robert Grant of Granted Software (Rax, Audio Unit Manager) and Adrian Pflugshaupt (Apulsoft Wormhole) to name just two. Their first creation, Musolomo, is a specialised sample-playback instrument. Although it can be applied to a track like any other processing plug-in, it won't do much until you 'play' it via a MIDI keyboard, at which point all sorts of mayhem ensues, including scratching emulation and that sought-after 'tape stop' effect. Be prepared to put in some work, though — the manual is essential, as are the instruction videos at the Plasq website. You might still find that it takes a few minutes to get your head around the basic idea, but Musolomo seems like a lot of fun and can do things I can't imagine achieving in any other way. Watch out for a tutorial in a future issue.

Introducing iZotope DP's bundled plug-ins are great, but it's not until you use the best plug-ins thirdparty developers have to offer that you realise how good plug-ins really can be. The big-money Waves collections are something of an industry standard, but many DP users also rely on much more affordable plug-ins by the likes of Wave Arts, Audio Ease and PSP Audioware for decent quality and distinctive audio processing. Having recently had the pleasure of checking out their entire product range for the first time, I'd definitely recommend adding iZotope (www.izotope. com) to your list of 'must audition' third-party plug-in developers. Their products are genuine MAS plug-ins, and they're both extraordinarily powerful and lovely to work with. First up is Vinyl, a freebie vinyl simulator — nice, but inevitably somewhat limited. Ozone is a different kettle of fish, offering six simultaneous mastering-type processes that could be equally applicable to individual tracks on file:///H|/SOS%2005-06/Digital%20Performer%20Notes.htm (2 of 3)9/28/2005 2:42:16 PM


occasion. The loudness maximizer and harmonic exciter elements, in particular, are stunning, with a genuine 'high-end' feel. Then there's Spectron, another multi-talented plug-in that performs all sorts of mind-bending and genuinely novel frequency domain tricks. Finally, there's Trash, a flexible and greatsounding distortion and cabinetmodelling processor treading a finely judged line between guitar friendliness and ultimate flexibility for other tasks.

iZotope's Ozone plug-in.

All the iZotope plug-ins represent the cutting-edge of signal processing, interface design and ease of use. Wonderful stuff, and highly recommended, as long as you've a nice fast Mac. My dual G4 just about coped... Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895


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DSD support in Cubase?

In this article:

A Patch For Cubase SX/SL 3.02 Connecting People

DSD support in Cubase? Cubase Notes Published in SOS June 2005 Print article : Close window

Technique : Cubase Notes

This month we look at the possibility of DSD higherquality audio support in Cubase, a plug-in to help you write ringtones for Nokia phones, and a new patch for Windows users of Cubase SX/SL 3.02. Mark Wherry

While there was no major news for Cubase users at this year's Frankfurt Musikmesse, Steinberg showed a couple of new products, such as The Grand 2 (featuring a second, brand-new acoustic piano), and some interesting new technology. ASIO (Audio Streaming Input Output) has been the technology for getting audio into and out of Cubase for nine years now and it was originally invented as a way of dealing with multi-channel PCM (Pulse Code Modulation) audio at relatively low latencies. At the Musikmesse this year, Steinberg introduced ASIO 2.1, a small, yet significant update to the ASIO 2 technology introduced at the Winter NAMM show in 1999 that adds support for DSD (Direct Stream Digital) audio, of which the best commercial example of usage is currently in Super Audio CDs (SACDs).

With Nokia's SP-MIDI Creator VST Instrument, you can use Cubase to make better ringtones for Nokia mobile phones. Imagine the creative possibilities...

Very briefly, DSD is a different way of representing digital audio, compared to PCM (Pulse Code Modulation). With PCM, the exact level of the audio is sampled and represented with the available bit resolution (24 bits, for example) a specified number of times per second, according to the sampling rate (44100 times per second at a rate of 44.1kHz). DSD is similar in that the signal is sampled a specified number of times per second, but the samples store whether the signal rose or fell since the last sample. The sampling

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rate used for DSD is significantly higher than the highest we might consider working with in PCM: 2.8224Mhz, 64 times higher than 44.1kHz. At Musikmesse, Sony showed a Windows workstation featuring a prototype, ASIO 2.1-based audio device developed by Sony that supported the recording and playback of DSD and PCM digital audio. While the debate continues as to which is the 'better' high-resolution audio format between 192kHz/24-bit PCM (for applications such as DVD-Audio) and DSD (for SACD), DSD has the advantage of having both a wide dynamic range and a high bandwidth, which is easier to balance than with PCM. Although DSD requires a fundamentally different approach to working with digital audio data than PCM, the fact that there will be a low-cost, ASIO-based platform for working with high-quality DSD audio in the near future is great news, since it will open up DSD development and could eventually lead to a DSD-capable version of Cubase or Nuendo.

A Patch For Cubase SX/SL 3.02 In last month's Cubase Notes we mentioned the 3.02 (build 622) update to Cubase SX/SL. Since this was released, a performance-related problem has emerged that has affected some users (including users of Nuendo versions with the same build number). As has been noted by various people on the web forums, working with certain user-interface elements, performing actions such as opening and closing of windows or zooming, has caused disk dropouts during recording and playback of audio. The good news is that not all users are affected by this problem: it doesn't seem to be an issue for Mac users, or for Windows users with reasonably high-end machines, especially dual-processor models. The better news is that the fix is only a small patch, which updates Cubase to v3.02 build 623. Windows users of Cubase SX/SL 3.02 can download it from ftp://ftp.steinberg.net/Download/.

Connecting People At the other end of the fidelity spectrum, no-one could argue that mobile phone ringtones do anything but enrich our musical culture — could they? To aid in their composition, Nokia have released an update to the Nokia Audio Suite of tools, which Steinberg were including in their 'World of VST' discussions at Frankfurt. The main part of the suite is the combination of SP-MIDI Creator, a VST Instrument that simulates the sound and behaviour of the built-in sound engines and capabilities of most Nokia phones, and Auralisation Tool, a VST effect that simulates their acoustics and DSP. SP-MIDI Creator (SP stands for Scalable Polyphony) is an extension to a standard MIDI file, brought forward by the MIDI Manufacturers Association, that specifies how a given file should be played on devices with differing polyphony capabilities, and was conceived as a solution to creating a single MIDI file that could be played back on different 3G and similar mobile devices. Nokia's SP-



MIDI Creator allows composers to use Cubase to preview how a piece of music will sound on different Nokia phones, and add the appropriate commands to that MIDI file. (You must export the file from Cubase before adding the commands from within the plug-in.) Aside from the obvious application of creating ringtones, you could also use the plug-ins creatively — or even in a practical way for post-production people who need to create sound effects for phones and so on. Nokia Audio Suite is a free download, although you have to be a member of Nokia's Developer's Resources Forum first (also free to join). Visit www.forum.nokia.com for more information.




Exploring Sonar 4's TTS1 Synth

In this article:

Multiple Outs Other System Settings Editing A Part Saving Performances Saving Edited Sounds MIDI Control Controller Maps & Automation Other System Options Closing Words Just For Effects

Exploring Sonar 4's TTS1 Synth Workshop Published in SOS June 2005 Print article : Close window

Technique : Sonar Notes

The TTS1 soft synth bundled with Sonar 4 has many hidden talents — but you have to know where to find them. Craig Anderton

From the early days of Sonar, Cakewalk bundled the program with the Edirol VSC (Virtual Sound Canvas). However, the advent of Sonar 4 brought a change: goodbye VSC, hello Roland-powered TTS1. The concept is the same — a General MIDI synth that doesn't use a lot of CPU power — but the new synth is surprisingly capable, especially once you know how to take full advantage of it.

The visuals are those of a 16-channel mixer, but it's really an interface for up to 16 instruments. Additional windows allow fairly extensive editing.

In terms of basic architecture, the TTS1 is 16-part multitimbral and is both General MIDI and GM2 compatible. Each part is 'hardwired', so that MIDI channels 1-16 are mapped to the equivalent TTS1 parts. The parts are arranged visually as 16 channels of a virtual mixer, complete with level fader, pan control, two effects sends and a master output section. Additional pop-up windows allow you to dig beneath the surface.

Multiple Outs Let's start our tour with the four individual outs. To take advantage of these, file:///H|/SOS%2005-06/Exploring%20Sonar%A04%27s%20TTS1%20Synth.htm (1 of 8)9/28/2005 2:42:32 PM


when you insert the TTS1 into Sonar 4 tick 'All Synth Outputs (Audio)', instead of 'First Synth Output (Audio)' when the 'Insert DXi Synth Options' window appears (unless, of course, you want only a stereo output instead of multiple outs). I'd also suggest creating additional MIDI tracks for each instrument you want to use, so that each instrument has its own MIDI track. I find this more convenient than cramming several MIDI channels' worth of data into one track. On the TTS1 itself, to assign parts to outputs: Click on the System button toward the upper right. A bright-red System Settings box appears: click on its Option button and a window appears. Under the Output Assign tab, assign parts to outputs by clicking the radio button to the right of each part (see the screen on the left). Note that you can assign multiple Parts to the same output, but you can't assign one Part to multiple outputs.

It takes a few mouse-clicks to get there, but you can assign each TTS1 part to one of four outputs.

Clicking the Reset button reassigns all the Parts to output one, which may be important if you're using the built-in reverb and/or chorus effects (I'll be returning to this subject later).

Other System Settings The red System Settings window also provides three other useful adjustments: Master Tune (variable in 0.1Hz increments, from 415.3Hz to 466.2Hz), Master Key Shift (transposes from -24 to +24 semitones), and Polyphony Limit, from 10 to 128 voices. To change a parameter value you can click on a knob and drag it, but also note the numerical field below each control. Clicking on the arrows to the side provides 'fine tuning' — for example, if you click on the Master Tune parameter's right arrow, the value increments by 0.1Hz. Furthermore, double-clicking on the numerical field allows you to type in a number; clicking OK then enters it. All numericals work in the same way, including the ones for the level, pan, and send controls on the main front panel.

Editing A Part

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At the top of each part's 'channel strip' you'll see an Edit button. Click on it, and a window opens up that offers a variety of editable parameters — something the VSC never had (see screen below). You can still tweak any visible 'mixer' parameters, even when the Edit window is on top. There are two time-saving features to be aware of. Firstly, you can cycle through the parts via the part left/right arrows towards the upper left, so it's not necessary to close the Edit window, open one for the next channel, close it, open another one, and so on. Second, the little preview button with a note symbol plays a pre-recorded note or riff in the style typically used by the selected sound (you can choose between note and riff, as described later). Sadly, you can't latch the button so that it plays while you tweak parameters.

The TTS1 offers far more sound-shaping options for synth sounds than the earlier VSC.

The editable synth parameters are pretty standard fare, but there are a few fine points to bear in mind. The On/Off button in the top row is needed only to enable/ disable the tone controls (bass, mid and treble). The Filter and Character controls are always 'live'. In the middle row, Envelope and Vibrato are, again, self-explanatory. However, note that even if the Vibrato depth control is at zero, you can still add vibrato with your synth's mod wheel, but this requires that the Mod Depth parameter in the lower row be set to a non-zero value (the main Vibrato depth control is designed to add a constant amount for each note you play). Unfortunately, the delay parameter doesn't fade in the vibrato; instead, it switches in after the elapsed delay time. The lower row has controls for Tuning, a Mono/Poly switch, a Portamento control with on/off switch, the aforementioned Mod Depth parameter, and a Bend Range parameter (up to a maximum of plus or minus two octaves). The drum sets have their own editable parameters. The middle part of the window lets you step through the various drum sounds, each of which has adjustable Level, Pan, Coarse Tune, Fine Tune, Reverb Level (send), and Chorus Level. The MIDI Edit

The drum programs have some parameters that are individually editable for each drum sound (middle section), as well as parameters that alter all drum sounds in the



program. button, when enabled, lets you choose the drum sound to be edited simply by playing its corresponding MIDI note. Note that the master channel Pan control weights the various drum sounds more to one side or the other, but the pan relationship between drums is maintained within those constraints.

The lower strip of parameters (Filter, Resonance and Tone controls, which must be enabled if required) affects all drum sounds in the program: these parameters are not individually adjustable for each sound.

Saving Performances The TTS1 has a 'combi' mode, for creating 'performances'. If you come up with a particular performance you like — specific instruments loaded as specific parts, which may also have been edited — there are three ways to save this. The simplest is just to save the project containing the performance. When you call up the project, the TTS1 will appear as you last left it, pretty much like any other virtual instrument. However, if you want to be able to call up the performance independently for different projects, you need to save it as a file. One way to do this is to type a name for the performance in the Presets field at the top of the instrument, then click on the floppy-disk button, as shown in the screen below. (For younger SOS readers, a 'floppy disk' was a prehistoric storage medium used by the ancient Etruscans. Its legacy lives on to indicate a 'save' function, but it's otherwise pretty much forgotten.)

Here, a preset is being saved. The dropdown menu will let you choose from any of the presets you've stored. If you select a preset and then click on the 'X' button, the preset will be deleted.

The other way to save a performance is to click the System button, then the System Settings Option button, then the Options tab. The top line has buttons for loading or saving a performance file. There's no pre-ordained folder for this. I simply created a TTS1 Performances folder inside the Cakewalk folder on the root drive. (TTS1 performances have a .GMF suffix.) If you call up a performance, it's a good idea to set the patch parameter for any MIDI channels feeding the TTS1 to 'none' before you hit playback. Otherwise, if a patch is specified, it will be called up in the TTS1 and you'll need to reload the performance.

Saving Edited Sounds



This process isn't exactly obvious, but then that's why Sound On Sound publishes this type of article! The confusion arises because the Part Edit window has a Write button. So you click on that, choose a program location in a User Normal bank (1-4, or the User Rhythm bank if you're saving a drum set), and you're asked if you want to overwrite the existing patch. Having clicked on OK, you might think the job was done, right? Well, not You can create a library of TTS1 really, unfortunately. The patch will performance setups for use in any project. show up in a MIDI channel track if you select the correct bank and patch, but if you exit from Sonar at this point, then launch the program again and call up the same bank, the patch will be gone and all the patches in the user bank will have reverted to a Piano sound. This occurs because although writing the edited part stores it in the bank, the bank is in RAM. So you need to take the additional step of saving the bank. Once you have the edited patches written into the bank, click on a patch name, either in the Part Edit window or in the strip to the right of the channel fader in the main mixer view. From there, select Save Bank. Choose the bank you want to save, give it the desired name and save it in an appropriate folder. The next time you want to access your edited sounds, click on a patch name, select Load, and choose the bank containing the sounds you want to use.

MIDI Control Just about all controls of the TTS1 — knobs, sliders, and buttons — can be controlled by MIDI controllers (not notes, however). Just right-click on the parameter you want to control and the Control Change Assign window appears. You can enter a controller and channel number here or, more conveniently, tick the Learn box and simply tweak the physical controller you want to use. Having said the above, there are a couple of points to bear in mind. Firstly, if you assign a controller that was previously assigned to a different parameter, the previous assignment will be shown in the window's lower-left corner. If you decide to go ahead and reassign it, the previous assignment will be cancelled. As a result, you can't control multiple parameters with the same controller. Secondly, you might think that checking the 'Apply to All Parts' checkbox and, for example, assigning a controller to filter cutoff makes the control affect filter cutoff for all parts. But that's not the way it works. Instead, checking this box assigns the parameter in question to respond to the same continuous controller number for all parts. For example, if you assign Controller 15 to Filter Cutoff for part six and the Apply to All Parts box is ticked, Controller 15 will also affect Filter Cutoff for



any other selected part. Furthermore, ticking Apply All Parts forces each part to respond only on the MIDI channel whose value is the same as the part number. In the example above, if you wanted to control Filter Cutoff in, say, part 11, Controller 15 would have to appear over channel 11. If the Apply to All Parts box is not ticked, a part can be assigned to respond to a controller coming in over any MIDI channel.

The 'Normal' controller map has been loaded. This assigns the most commonly automated TTS1 parameters to various MIDI controllers.

Controller Maps & Automation If you click on the System button, select 'Option' from the System Settings window, then choose the Options tab, you'll find the option to Load and Save Control Change maps. Under 'Load', you have three options: Minimum Map, Normal Map, Logic Map (which uses Logic mappings for controllers — probably for those who switched over to Sonar from Logic for PC) and File, where you can load maps you've created and saved. So what are these maps? Well, when you go to create a MIDI envelope in a MIDI track, TTS1 parameters are pre-mapped to particular controllers and labelled in the Envelope drop-down menu, and the maps govern exactly how this is done. If you load the Minimum map, only a few parameters show up. The Normal map is what you'd, well, normally use. You can also create custom maps for working with external hardware controllers. Speaking of real-time control, you'll also notice another option just below the Control Change Map buttons: Record Panel Operations. When The TTS1 offers a Learn function for this is checked, if the MIDI track driving the MIDI controllers, which can control Part is in 'record', any TTS1 front-panel virtually any TTS1 parameter. control movements will be recorded as MIDI data. On playback, the knobs will move to reflect those controller changes. So there are three ways to record automation with the TTS1: via envelopes; by recording knob motions from the TTS1 panel; or by recording MIDI data from an external control surface, and passing it through to the TTS1.



Other System Options Before signing off, I'll just bring to your attention and explain the significance of the three other checkboxes under the System Options tab: Tone Remain allows notes to sustain while you change from one program to another. This eats more polyphony, but will not be an issue in most cases, as the TTS1 is pretty efficient. Just make sure you've set the polyphony control high enough to accommodate any transitions. Enable Phrase Preview determines what happens when you click on the little note symbol on the preview button. If Enable Phrase Preview is unchecked, you'll hear a single note played on the currently selected sound when you click. If it's checked, clicking on the note symbol instead plays a phrase in a style representative of how the currently selected instrument is played. Light Load Mode reduces CPU consumption even further by using lessintensive internal algorithms. Unless your CPU is seriously performancechallenged, the odds are that you won't need to use this option

Closing Words Some Sonar users look down on this humble instrument because it's bundled — so how can it be any good? The surprise is that the TTS1 can be a very effective sound module, and its editability makes it that much more useful. When you throw in the ability to add expressiveness through the use of external controllers, you can make some pretty good noises. Give it a go and you'll see what I mean.

Just For Effects The TTS1 provides two effects: chorus and reverb. Each part's 'mixer channel' has a chorus and reverb send control, which is duplicated in the edit window. However, note that the effect sends are taken only from instruments whose outputs feed the main output (output one in a multi-output setup). Choose the chorus and reverb algorithms, Furthermore, the effects return only then adjust the parameters as desired. If to the main output. As a result, if you're not using the processors, turn them off to save CPU power. you're using multiple outs and you turn up the reverb and/or chorus send control for any parts that aren't assigned to output one, you won't hear the effect, as those parts are not being processed.



The effects can be edited in a fairly basic way. Click on the Effect button (above the Master Volume slider) and the Effect window appears. Here you can choose one of six chorus/flanger algorithms and one of six reverb types. The only reverb parameter you can edit is reverb Time, but in the case of the chorus you can alter Rate, Depth, Feedback and Rev (reverb) Send. The last sends some of the chorused sound to the reverb. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



From 32-bit to 64-bit

In this article:


Gigastudio 3 Effects Presets PC Notes Caveat Emptor Published in SOS June 2005 PC Notes In Brief More Fabulous Freebies Print article : Close window

Technique : PC Notes

The magic number of 64 is gradually working its way into all aspects of our PC experience: processors from both AMD and Intel, the Windows XP OS, hardware drivers, and — coming soon — 64-bit music apps. But what should we be aware of as we consider the transition from 32-bit? Martin Walker

Intel's new Pentium 4 600-series 64-bit processors were launched in February, giving anyone about to purchase a new budget PC a rival to AMD's increasingly popular Athlon 64 range. Initial tests with Intel's 600 series indicate that its 3.6GHz P4 model offers almost identical performance to an AMD Athlon 64 3800 + model with the current 32-bit Windows, although as I write this it seems too early to form an opinion about their relative 64-bit performance, as so few 64-bit applications are available. However, as with many past comparisons, an Intelbased solution still seems to be slightly more expensive than an AMD one of equivalent performance. Microsoft's Windows XP Professional x64 operating system is due to ship in April (so should be available by the time you read this) and will finally enable many existing users of PCs with processors including AMD's Athlon 64 and Opteron, and Intel's Xeon and P4 600 series, to benefit from 64-bit computing. However, as I mentioned in last month's PC Notes, only applications that have been optimised and re-compiled to take advantage of its new capabilities will provide better performance. file:///H|/SOS%2005-06/From%2032-bit%20to%2064-bit.htm (1 of 5)9/28/2005 2:42:36 PM

Apart from a new design of wallpaper, there's little on the surface to distinguish the new


Windows XP Professional x64 Edition from the 32-bit XP versions, although the underlying code is completely different.

Most mainstream users are unlikely to get worked up about 64-bit computing, unless the marketing people pull off something really special, as it won't benefit the vast majority of mainstream or even office applications. However, a fully 64-bit PC will theoretically be able to utilise far more memory (64GB in the case of Intel's new range, and 1024GB for AMD's 64-bit processors), which means that some musicians may be able to abandon sample streaming from their hard drives in favour of 'instant' access from system RAM, with correspondingly vast polyphony as a result. However, the actual RAM ceiling of any 64-bit PC is more likely to depend on the motherboard limits than anything else. The second benefit of 64-bit architecture for musicians is its improved floatingpoint performance and the greater number of internal registers, each of which is 64 rather than 32 bits wide. This will result in far more efficient audio algorithms: early indications from Cakewalk's Sonar x64 Technology Preview indicate that we can hope for performance gains of up to 30 percent from a 64-bit-capable processor running under Windows XP Pro x64 in 64-bit mode (compared to using a processor of the same clock speed with the Windows XP 32-bit operating system and 32-bit Sonar application). Fortunately, all the signs are that Windows XP x64 is already extremely compatible with most existing 32-bit apps (subject to a few caveats that I'll come to shortly), which is a relief, considering how much most of us have invested in them. Existing 32-bit apps will also be able to access up to 4GB of memory if the Large Address Aware switch that I discussed in last month's PC Notes has been used.

Gigastudio 3 Effects Presets In Gigastudio 3, Tascam abandoned their previous regime of saving NFX1 plug-in user presets into a single file in the C:\Windows folder and replaced it with a collection of individual FXP files inside the C: \ Program Files \ Tascam \GStudio \ Presets folder. Tascam also took the trouble to convert the contents of any existing NFX files to the new format, so if, like me, you upgraded from version 2.5, your old NFX1 presets will be ready and waiting for use in their new format. This also means that presets can be swapped more easily with other GS3 users. During my exploration of the NFX1 reverb plug-in as part of a Gigastudio Power Tips feature in SOS September 2002, I created a bank of 21 presets ranging from halls, gymnasiums and cathedrals to special effects such as 'Ricochetverb', 'Boomverb', and 'Splatterverb'. NFX1 has always been more capable of smooth tails at low CPU overhead than many users realised, and of course GS3 Solo and Ensemble purchasers don't get the Gigapulse Pro convolution reverb, so if anyone would like a copy of these presets, just drop me an email ([email protected]).

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Caveat Emptor Perhaps understandably, Microsoft have finally abandoned support for 16-bit legacy and DOS applications in x64, so we may have to leave behind some of our 'old faithful' utilities. Unfortunately, many 32-bit applications also use 16-bit installers, and these will need updates from their developers before they can be installed under x64, so yet more may fall by the wayside. There's no doubt about it: the change-over could be messy for many people, especially since every hardware item will require 64-bit drivers to be compatible.

If you like experimenting with surround, you'll love these freeware offerings from Acousmodules, which offer sophisticated control over the movements of multiple samples in a 3D world. They also allow you to generate three sets of MIDI controller information, based on user-created paths, for controlling further spatialisation plug-ins.

While some interface manufacturers, including Creative, Edirol,and M Audio, have already developed suitable drivers for their audio interface ranges, many others haven't yet, and some never will. For instance, my Yamaha SW1000XG soundcard would be an early casualty, as Yamaha have long since abandoned further development on a product that was originally launched in 1998. Its MIDI synth may not be cutting-edge, but I'm not looking forward to having to discard it altogether. Other casualties are likely to be scanners, printers and the like, which are often so quickly superseded by newer models. Even the various devices on the motherboard will require 64-bit drivers in order to function, and at the moment some early adopters of Windows x64 are finding themselves (for instance) unable to use some onboard SATA RAID controllers and sound chips. Nevertheless, for anyone thinking about buying a new PC at the moment, it makes sense to strongly consider a 64-bit one, even if, for the time being, you buy it with the 32-bit Windows XP installed and continue to run it in 32-bit mode until you replace any non-64-bit hardware. Microsoft have announced their intention to offer a 'Technology Exchange Program', so that customers who buy a 64-bit capable PC with Windows XP Pro pre-installed can swap their operating system for XP Pro x64 later on. This won't be an update — it will require a clean install and re-installation of all your applications, which is usually a major undertaking. Nevertheless, it does sugar the pill for those who simply can't wait to buy their next PC, yet don't want to make an expensive mistake.



Meanwhile, further to my discussions in PC Notes January 2005, dual-core desktop processors (two processor cores on a single piece of silicon, shipped in an identical package to a single-core processor) are now expected from both AMD and Intel during the second half of this year. However, while AMD's dualcore Toledo Athlon 64 processor range should require only a BIOS update to be fitted onto most existing Athlon 64 motherboards, Intel's new Pentium D will require a new motherboard, with an as-yet-unreleased 945 or 955X chipset. So, for the moment, it looks as if buying a PC based on the Athlon 64 is safer for anyone who wants their investment to last as long as possible.

PC Notes In Brief Soundblaster Audigy and Extigy users will be interested in the result of a lawsuit that may allow them a $62 discount on their next purchase from the Creative web site. Creative claimed on their packaging that these products supported 24-bit/96kHz audio, but as I pointed out in my review of both the Audigy platinum eX in SOS November 2001 and the Audigy 2 Platinum eX in SOS April 2003, most of the audio passing through these cards would still be processed through the fixed 48kHz audio engine. Although no liability has been accepted, Creative have graciously agreed to provide the aforementioned discount to anyone who bought an Audigy ES, Audigy Platinum, Audigy Platinum eX, Audigy Gamer, Audigy MP3+ and also the original Extigy external USB sound module before the end of 2004. There's a deadline of September 22nd 2005 to pursue your claim, but if you feel you qualify, you should read the claim form at www.audiocardsettlement. com. Fancy installing Windows XP, 2000 or 2003 in a 300MB partition? nLite (a free download from http://nuhi.msfn.org) is an 'Installation Deployment Tool' that claims to do exactly this, by letting you specify exactly what parts of XP you install, rather than accepting Microsoft's choices. For some musicians, being able to abandon Internet Explorer and Media Player in favour of other alternatives will be the carrot, while for others being able to strip things down sufficiently to let them run XP effectively on an elderly PC will entice. You just copy Microsoft's entire Windows XP CD-ROM into a folder or partition on your drive, then start nLite and let it guide you through the process of customising and preparing the Windows installation. You can also incorporate Service Packs, drivers, and hot-fixes into the installation files. I haven't tried it myself, but nLite already has lots of enthusiastic followers, and there are English and German support forums in case you run into difficulties. For those running lacklustre motherboard sound chips, or those who have ambitions to run multiple ASIO audio interfaces inside Cubase, Michael Tippach's famous ASIO4ALL WDM wrapper is now up to version 2.5, and better than ever. I had rather more success with it than 2.0, getting both my Echo Mia and Emu 1820M running together within Cubase SX with fairly low latency. The 319Kb download is still free from http://tippach.business.tonline.de/asio4all/index.html, and there's even a support forum available.

More Fabulous Freebies Like many other enthusiastic designers using the PC-only Synthedit design environment, Jean-Marc Duchenne's web site offers both plug-ins and soft synths. However, he's also a musician with a passion for spatialisation and 'acoustic space composition', so once you enter the world of Acousmodules



(http://acousmodules.free.fr) the idea is not to get your head around the sounds, but rather the sounds around your head. On offer are 'Playing Surfaces' such as Animasampler 3D, which animates various parameters of eight samples using a 3D gaming joystick (one with a rotating handle) into eight outputs, designed to be fed into eight speakers set up at the corners of a cube. Other hardware controllers, such as a mouse, standard joystick or graphic tablet, can be used with his Joysynth, Joysampler, and Joymachine. For host applications that can accept MIDI output data from plug-ins (such as Cubase, Nuendo, Sonar, Energy XT, Synthedit, Console, and Bidule) there's a range of MIDI controller plug-ins that can generate up to 24 sets of controller data derived from just one gesture on a 3D joystick, or animate them from mouse movements. There are also 'trajectory' plug-ins to capture performance movements, or generate 3D paths that are output on three MIDI controllers, for spatial or other multiple-parameter control, MIDI processors that generate, transform or refine MIDI note or controller data, various utilities and test generators, and even plug-ins to visualise and control the P5 virtual reality glove controller (www.essentialreality. com). Various spatial processors are also available, such as Spatgrains 8, which provides octophonic granular processing so that each grain can be separately positioned within eight outputs. You can also try out multi-channel feedback delays, resonant filters and frequency band-splitting. There are dozens of free downloads here, and many of them look visually stunning with their spatial animation. Some are still experimental and are very CPU-hungry. Nevertheless, this is a truly inspiring collection for anyone who wants to explore new control methods and spatial possibilities. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Hard Drive Defragmentation

In this article:


What's Fragmentation? PC Musician Internal Disk Geometry Published in SOS June 2005 Defragging Guidelines When To Defrag An Audio Print article : Close window Drive Technique : PC Musician Microsoft's Bundled Defragger Defragging Tips Executive Software Diskeeper 9 Pro Does defragmenting your hard drives, including the Raxco Software Perfect Disk ones you use for recording audio, really result in Workstation better PC performance? Opinion is divided, so we take O&O Defrag 6.5 a considered look at the subject, as well as testing HDD Health some of the most suitable 'defragger' utilities. Final Thoughts Martin Walker

Defragmentation is essentially the art of rearranging files on your hard drives to enhance performance, and there are regular queries on the SOS Forums from people asking what is the best 'defragger' utility. Noticing these queries, I thought that I'd investigate a few such utilities and report back with my findings, as part of a more general roundup of software that proves particularly handy for the PC musician. However, during the course of my Fragmented hard disk data as represented in the Perfect Disk defragmentation utility. research I discovered so many conflicting opinions on the actual merits of hard drive defragmentation — ranging from those who recommend 'defragging' after every recording session to those who never do it at all, claiming either that it's unnecessary or that it can even degrade performance — that I decided to explore the whole subject in more detail, with the needs of the musician in mind.

What's Fragmentation? Because Windows saves data wherever it can on your hard drives, often in

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unused gaps between other saved files, some files may end up in several scattered fragments. Reading such a file takes longer than reading one stored in a single piece, as the read/write heads have the additional travel time of jumping from the end of one fragment to the beginning of the next before continuing to read its data. As you carry on deleting and saving files, and particularly as your drive fills up beyond 70 percent or so of its total capacity, fragmentation may get worse. Defragmenting the contents of your hard drive involves locating all the parts of each fragmented file and bringing them back together, by saving the now contiguous data to another more suitable location on the drive. This means that Windows only has to look in one place for each file, which can help its performance, by avoiding unnecessary read/write head activity, and can result in both Windows and its applications loading more quickly. General file access can also be smoother. If you adopt the outer 'Current Project' partition and inner 'Project Backups' partition arrangement that I suggested in last month's PC Musician, before you start on any new project you'll be able to delete the entire contents of the outer partition, so that it starts with a clean slate and no fragmentation, for maximum performance. However, if you're maintaining large unpartitioned hard drives holding vast amounts of data, fragmentation can become an increasingly important issue. While your Windows partition may benefit from regular attention, drives containing audio and particularly video files may benefit even more, because they are not only much larger but also more likely to be regularly edited during a project, resulting in further fragmentation.

Internal Disk Geometry Many large, modern drives contain multiple platters and four or more read/write heads, so we can't always visualise our files as being best laid-out neatly in one area of one platter — indeed, it may sometimes be preferable to have a single file spread over several platters, so that it's easily accessible to several read heads almost simultaneously. A drive may also feature cache memory of 8MB or more, which can also affect the issue of optimum file placement, because some of the required data may already be present in the cache (although reading and writing audio files nearly always results in large files that will soon swamp any cache). Some commentators claim that multiple platters and large caches mean that defragger utilities that gather together all the fragments of long files into one neat area will automatically result in worse performance. They also imply that the utility developers are conning the public, because this approach undermines attempts by both drive and operating system to place the data according to their own internal algorithms. However, this view doesn't take into account the fact that defragger utilities are also written by file system experts, and their own algorithms are obviously honed and polished by practical tests with real-world systems. It doesn't matter how much theoretical discussion there is: if you defrag your drive and can measure an improvement in performance, such as an increased number of audio tracks before your audio application conks out, that utility works for you! Unfortunately, in my discussions with defrag utility developers it became clear that



their algorithms don't take into account the unique access pattern of multitrack audio applications with large simultaneous numbers of huge audio files — so, as they say, your mileage may vary!

Defragging Guidelines Apple state that their users probably won't need to defragment at all if they run the Mac OS X operating system, because the Extended formatting (HFS Plus) of Mac OS X avoids re-using space from deleted files as much as possible, to avoid prematurely filling small, recently freed areas. Windows doesn't yet seem to be quite as clever, so defragging is still useful for the PC Musician. It also seems generally accepted that keeping plenty of free space on Windows drives or partitions (30 percent or more) will help Windows save new files more sensibly, rather than letting it work around gaps between existing files. The general advice given by most PC experts is that anyone who still creates their hard drives in FAT32 format should periodically analyse them to check fragmentation levels. Those who have adopted the more recent (and more secure) NTFS format are less likely to experience fragmentation, but they should also still occasionally check on fragmentation levels. If you click on the Analyse button of Window's own bundled Disk Defragmenter tool, it will suggest you defragment a partition or complete drive once the fragmentation level reaches a certain threshold (implying a noticeable downgrading of performance) — although you can, of course, ignore its advice and defragment as often as you like, for smaller performance benefits. It certainly makes sense to do so after installing Windows, after installing lots of new software (and, in particular, games, which can sometimes include a huge number of files), or after a good clear-out when you may have deleted lots of files. You can set up many defraggers to defrag at a specific time and date, but I always avoid such an approach, since my PC isn't switched on 24 hours a day and I don't want defragmentation to start if I happen to be busy doing something important at the time. Others may offer to run quietly in the background, but however clever they claim to be in detecting when the user is asking the PC to perform other tasks, I still err on the side of caution when running audio applications and disable such cleverness, just in case it results in a single click during an otherwise perfect take. I personally tend to instigate a routine manual defrag of my Windows drive once every month or so, if necessary.

When To Defrag An Audio Drive Whatever your personal decision for your Windows and application partition or drive, when it comes to those used to store audio and video files even Apple are in agreement that people who create or modify large audio or video files might benefit from defragmentation (even when running Mac OS X). However, there's also a school of thought that says you're better off not tidying up huge audio files file:///H|/SOS%2005-06/Hard%20Drive%20Defragmentation.htm (3 of 11)9/28/2005 2:42:43 PM


into single neat units, as I first explained way back in SOS April 2002 (www. soundonsound.com/sos/apr02/articles/pcmusician0402.asp). When reading long multiple audio tracks, your sequencer application will access a chunk of each one in turn, before returning for another chunk of each one. So storing them in interleaved chunks of the size your audio application uses may result in less drive activity than having any existing fragments painstakingly reassembled into neat monolithic files, one after the other. At the time of writing the article mentioned above, I suggested that only the sequencer developers themselves could provide us with a suitable utility to rearrange audio files to suit their particular file requirements. However, in the absence of such utilities, I suggest that if you (for example) record 16 simultaneous live audio tracks into an empty partition and then play them back and do a little editing and mixing, you shouldn't defragment at all — your data will already have been laid down in possibly the best arrangement on the drive.

The Disk Defragmenter utility bundled with Windows XP won't consolidate your free space into one neat chunk, as shown by the large number of white gaps between chunks of data displayed here.

However, if you do lots of subsequent editing, or store various songs or projects on the same partition or drive, it makes sense to defragment, particularly if you notice any tell-tale signs of excessive drive activity. One classic sign is increased audio drive noise: a series of steady clicks indicates progressive head movements as each chunk is accessed, whereas erratic or frantic whirring suggests that the read/write heads are being thrashed to and fro and the drive may thus benefit from a defrag. Another tell-tale sign of erratic drive activity is occasional spikes on your audio application's disk meter. If these coincide with the start of a new verse or section it may simply be because your song has just started accessing lots of new audio parts simultaneously. However, if these spikes are more random, they may suggest the presence of lots of fragmented files. A defragger analysis will soon tell you. In the case of frantic drive activity you'll probably notice a drop in head noise after defragging — which would confirm the diagnosis. Defragmented partitions will probably also take less time to back up, and will probably generate audio export files or render video or animation files more quickly. Moreover, consolidating the free space on your drive into one huge chunk can also make audio clicks and pops or dropped video frames during future recordings less likely, because future recordings won't immediately end up wedged into the remaining nooks and crannies. Such consolidation can sometimes prove as beneficial as defragmenting the files themselves.



Microsoft's Bundled Defragger Having discussed the pros and cons of defragging your hard drives, we'll turn to what's on offer for doing the job. First, Windows XP and 2000 both include a Disk Defragmenter utility (a cut-down version of Executive Software's Diskeeper product), which can monitor your file activity to work out which files you access most often and then rearrange them on your hard drives to eliminate excessive head seek activity. This 'pre-fetch optimisation' only works if you have Task Scheduler enabled, but if you want to avoid the possibility of file rearrangement happening at an inopportune moment you can manually run the Defragmenter utility whenever you prefer. The Windows Defragmenter is a freebie, which is always nice, but many musicians do become dissatisfied with it, as it has various limitations compared with the full version available direct from Executive Software (more on this in a moment). These limitations include the inability to defragment the MFT (Master File Table) on NTFS-formatted drives, directories on FAT32 formats, the paging file, and certain other system Executive Software's Diskeeper 9 Pro is the files. The freeware Page Defrag utility 'full' version of the defragger bundled with from www.sysinternals.com that I first Windows, and offers far quicker and more thorough performance, but consolidates free mentioned in PC Notes October 2003 space as a separate process. will take care of the paging file, but Microsoft's bundled defragger has still more limitations: it can't defragment more than one volume at a time, or be scheduled to run at a specific time, and these latter options prevent you from easily 'defragging' all your drives overnight, for instance, if you would like to. The utility can also take a long time to defragment a volume, even if it contains minimal fragmentation, and for musicians with huge audio and sample drives this can make 'defragging' an excruciating experience. Moreover, it doesn't consolidate the free space into one neat chunk, so even after defragging your neatly reorganised files may still end up spread across several areas of the drive with space between them — making it likely that the next large file you save will immediately become fragmented! The Windows Defragmenter also requires a minimum of 15 percent free space on a drive to adequately do its job (not very helpful if you have a well-stuffed drive that could benefit from a tidy-up). However, the most annoying limitation is that it simply refuses to defragment some files, for reasons known only to itself. For instance, on my PC it avoids the 450MB of files that comprise Groove Agent's drum kit. Microsoft are quite open about these limitations (you can read the full list of explanations in their Knowledge Base at http://support.microsoft.



com/ kb/q227463), but they leave many musicians wanting to search out a better option.

Defragging Tips Ensuring that you have plenty of free space on your drive should always help to keep fragmentation levels low, but also offers another benefit: your files should end up being mostly placed in the 'outer' and therefore faster part of the drive for better performance. Some musicians even buy a drive of the 'next size up' to their immediate requirements, to ensure that this happens. It generally doesn't cost much more. Creating separate smaller partitions for Windows, data, audio files, sample files and so on allows you not only to back up each one more quickly and easily, but also to defragment them considerably more quickly. If you maintain a separate 'Current Project' partition for your audio files, you can back it up either by creating an image file to another partition or drive, using a utility such as Norton's Ghost, or simply by dragging all the files across using Windows Explorer. If you feel that the file layout created by your audio application may already be perfect, or may have resulted in a set of interleaved files that possibly provide better performance, an image file will preserve this exact layout for posterity. However, for audio backups you may prefer to use Explorer, since its method of copying all the files across in turn will result in zero fragmentation on the backup, and zero fragmentation if you later move them back for further editing. Using this approach, you may never need to defragment your audio partition.

Executive Software Diskeeper 9 Pro A widely purchased alternative is Executive Software's Diskeeper 9 Professional (www.execsoft.co.uk) at £45 including VAT, or $49 from the US web site. I've seen claims from some users that this runs up to 10 times as fast as the cutdown version bundled with Windows. Unlike that version, it also offers complete rather than partial defragmentation (although it still does its work best if you have at least 20 percent of free space left on your partition). In addition, it consolidates free space, deals with critical system files when running its Boot-Time Mode, and can defragment multiple partitions simultaneously. None of these operations are offered by the cheaper Diskeeper Home version, which, although being offered for only £23 on UK and $30 on US web sites, should probably therefore be avoided. Both Home and Pro versions of Diskeeper offer a 'set and forget' feature that defrags in the background. Most musicians should really disable this, to avoid it cutting in at the wrong moment. (Incidentally, some users have found that despite disabling 'set and forget' mode it is automatically activated after a manual defrag and remains so until the next reboot — so don't defrag your audio drive with it and immediately try recording a huge live performance!) I was impressed by Diskeeper 9 Pro's straightforward yet thorough approach to both online and offline (ie. performed on the next boot before Windows is loaded) defragging. However, its multi-pass engine may end up taking longer to complete file:///H|/SOS%2005-06/Hard%20Drive%20Defragmentation.htm (6 of 11)9/28/2005 2:42:43 PM


the task on congested drives than some rivals, since it may require several passes to achieve optimum file placement. Moreover, while it does consolidate free space, it does this as a separate, ongoing process as part of 'set and forget' mode, rather than as part of the defragging operation.

Raxco Software Perfect Disk Workstation Diskeeper 9 Pro has many contented users, but during my research two other products stood out as having particular strengths for the PC musician. The first is Raxco Software's Perfect Disk Workstation (www.raxco.com/products/ perfectdisk2k) which, for about £40, has won a lot of admirers for the thoroughness of its speedy single-pass defragmentation, which nearly always means that all your files will be placed in their new optimum positions in one run. It also works well down to five percent of remaining free space if your drives are well stuffed. Perfect Disk Workstation may also be more suitable for anyone running Windows Server 2003 or a PC network (which normally require a more expensive Server version of most other defragmenters), and will cope with several huge drives of several terabytes in size. It also supports all levels of RAID, for those with more ambitious setups. If you want the fastest results, the software's Smart Placement algorithms work well, although they may leave some tiny blocks of free space between files. While it takes a little longer, the 'Aggressively free space consolidation' option makes more sense for audio and sample drives. For the most thorough results, the offline defragmentation option can run on the next boot and (depending on the particular operating system and drive format) can deal with the Master File Table, page file, Hibernate file and directories. I found this option quick, easy and thorough.

Although it may not provide the most versatile set of options, Raxco Software's Perfect Disk proved to be the fastest and easiest-to-use defragger on test, which should please any musician with several huge drives.

However, although the user can specifically exclude certain files or folders during an online defrag, Perfect Disk deliberately doesn't support user-customised placement of files, claiming that "this provides little improvement in file system performance". I ended up having some long email correspondence with one of Raxco's System Engineers about this fact, and he maintained that file/free-space defragmentation has a far greater and measurable positive impact on file system/ drive peformance than trying to place files at specific logical clusters in the hope that that they're on the 'fastest' part of the drive. Nevertheless, he did admit that he knew little about audio/video streaming, editing and processing, or the algorithms used by audio and video applications to maximise disk performance. file:///H|/SOS%2005-06/Hard%20Drive%20Defragmentation.htm (7 of 11)9/28/2005 2:42:43 PM


Overall, I was impressed by the speed and thoroughness of Raxco's Perfect Disk and, like various other people who had downloaded the demo version, I was first offered a 20 percent and then a 25 percent discount by email, bringing the final cost down to a very reasonable £30.

O&O Defrag 6.5 The other defragmenter that may particularly appeal to musicians, and that regularly seems to top the polls in mainstream PC magazines, is O&O's Defrag 6.5. This is largely because although it can sometimes take longer to perform its magic, it tends to result in lower fragmentation overall, even when its competitors claim not to be able to reduce fragmentation on a particular drive any further. The professional edition for use on a single PC running Windows NT, 2000 or XP also costs around £27, which is probably low enough to make it an essential purchase for many musicians. However, you'll need the more expensive Server version if you've got a network or run Windows Server 2003. Like Perfect Disk, Defrag 6.5 supports any number of IDE or SCSI hard drives, up to terabytes in size, as well as RAID, and defrags the MFT, Registry and paging file. It offers plenty of background defrag options, such as automatic defragging once a particular drive reaches a pre-determined threshold defragmentation level, and neat twists such as automatically dropping into pause mode if you unplug your laptop from the mains (to preserve its batteries), then automatically continuing where it left off as soon as you plug it in again. Sadly, for the musician automatic functions such as these are mostly to be avoided. It was the software's five optimisation strategies that particularly caught my attention. Stealth mode is the fastest method, most suitable for initial defragmentation and for large file servers. It also performs some free space consolidation, although full details of its strategy aren't explained. Space mode requires far less free space on the drive than the other approaches and it maximises the contiguous free space more thoroughly, whereas the Complete/ Access, Complete/Modified, and Complete/Name methods additionally reorganise your entire file placement on the drive, according to when files were most recently accessed or most recently modified, or alphabetically.


With a variety of defragging methods on offer, O&O Software's Defrag 6.5 initially seems a perfect candidate for the musician, although on closer inspection the options may not do exactly what the doctor ordered.


I immediately started making plans to measure performance improvements after optimising my Gigastudio streamed sample partition using the Complete/Access mode, so that the instruments I used most often would end up at the beginning of the drive for quickest future access and best polyphony (remember that on most drives read performance drops from the outside to the inside by about 50 percent). It's impossible to guess the practical results of such a reorganisation, because while your most-used instruments will benefit from a faster transfer rate, this might place others used in a particular song further away, resulting in more read/write head activity to access them. However, my plans were thwarted when I read the help file more closely, since O&O organise files in the opposite way to my desired placement, putting leastaccessed files at the beginning of your drive. The rationale is that placing seldomused files near the beginning makes future defragmentations quicker, as fewer files need to be checked and defragmented. As far as I could see, this is good for the future performance of the O&O utility but not for the musician, and sadly my email enquiry for more information remains so far unanswered. I did try the Complete/Name strategy on my Windows partition, because this claims to speed up boot times, but on my (admittedly stripped-down) 5GB partition it made no measurable difference after 1.5 hours of file reorganisation. I also found setting up offline Jobs to perform the defragmentation of system files a confusing experience, and while most mainstream and business users will delight at the cleverness of the Activity Guard that monitors CPU usage, to ensure that you can always carry on working while defragging in the background, it again won't suit the musician who demands maximum performance from a PC for audio, and is far more likely to want to perform a defrag during downtime. Overall, I was impressed by Defrag 6.5, but at present I don't think it's the ideal product for the musician.

HDD Health There's not much point in attempting to squeeze the last drop of performance from your drives if they're about to fail, and advance notice of this is always welcome. HDD Health from Pantera Soft (www.panterasoft.com) is a freeware 'failure-prediction agent' for hard drives that runs on Windows 95, 98, ME, 2000 and XP. Using the SMART (Self Monitoring And Reporting Technology) built in It won't make your hard drive go faster, but the freeware HDD Health from Pantera Soft to all modern drives, it monitors could alert you to its impending failure. various aspects of performance, such as spin speed and error rates, and attempts to predict impending failure. In the past I've always disabled SMART in my BIOS, because of the tiny extra overhead it imposes, but with faster modern



drives this overhead ought to be almost undetectable. Although HDD Health can sit in the System Tray and run in the background, you needn't worry about it impinging on your audio track count, since by default it only polls your drives once every 15 minutes, and you can easily disable it when required or just run it once and exit each time you log into Windows. If it detects any changes that suggest possible problems ahead, it can inform you via a popup message, email, net message or event logging. I've been running this software for several weeks, and although sometimes its announcements of minor changes to one parameter (such as 'Seek Error Rate changes from 79 to 80') on one drive can become annoying, I'm happy to put up with this if it manages to give me notice of impending doom. One SOS Forum user has reported that one of his drives failed disastrously about a week after just such a warning, by which time he'd backed up all data and bought a replacement, just in case. Be warned!

Final Thoughts However much I'd like to provide hard and fast answers to the whole subject of defragmentation, the reality is that some of you may rarely suffer from the effects of fragmented files, while others could run into them on a regular basis. Since installing Windows XP I've run the bundled Microsoft defragger every month or so on my Windows partitions, and more rarely on my data ones. Since I don't record huge numbers of audio tracks, this makes sense for me. However, those of you who regularly record multitrack audio (and particularly those who do so at 24-bit/96kHz or higher formats) would be well advised to at least check fragmentation levels on a weekly basis. If you find that your particular regime of recording, playback and editing results in noticeable fragmentation, running a defragger utility on a more regular basis is sensible. Some may notice immediate benefits after doing so, such as audio apps no longer momentarily dropping audio or even stopping altogether during the densest part of a song. It may also result in lower drive noise. On the other hand, with already low fragmentation levels I can see little point in religiously defragging after every take. Unless your songs are beginning to push the technical limits of your hard drive you're unlikely to notice any improvement in track count if you do this. The one exception is if your drive has less than 30 percent free space. In this scenario, your maximum audio track count is more likely to drop because of fragmentation, and frequent defragging may help — although buying a larger drive is a preferable option. Having established that defragging is beneficial for most of us at some time, we come to the choice of defragger utility. I personally find the bundled Windows one incredibly slow and tedious for drives larger than a few Gigabytes, and the various limitations discussed earlier further reduce its attractiveness. O&O's Defrag 6.5 might be an ideal candidate for those who want to explore userconfigurable file placement, but although this has provided measurable performance benefits for some users, on behalf of musicians I wasn't convinced by the arguments. For me, Raxco's offering was much simpler to use, in both file:///H|/SOS%2005-06/Hard%20Drive%20Defragmentation.htm (10 of 11)9/28/2005 2:42:43 PM


online and offline modes, and was particularly speedy. For those with huge audio drives, I suspect that will be the deciding factor. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



OS X Tiger, Apple Soundtrack Pro and more...

In this article:

Introducing Soundtrack Pro Doing The iPod Shuffle Tiger Preview Autofill Me Up

OS X Tiger, Apple Soundtrack Pro and more... Apple Notes Published in SOS June 2005 Print article : Close window

Technique : Apple Notes

We offer a brief preview of the features musicians can look forward to in Mac OS X Tiger, take a first look at a major new version of Soundtrack, and examine why the iPod Shuffle is hard to resist. Mark Wherry

In recent years, Apple's presence at the NAB (National Association of Broadcasters) show, held annually in Las Vegas, has continued to become more prominent as the company continues to focus on professional products for film and digital media. The rumours surrounding what Apple would introduce this year included new versions of software such as Final Cut Pro, a new professional audio application, and the introduction of faster Power Mac G5 computers. Although this last rumour proved to be untrue, Final Cut Pro 5 was indeed introduced, and the new audio application turned out to be an update to Soundtrack called Soundtrack Pro. Both of these applications contributed to Apple's main NAB announcement, which was Final Cut Studio, a new bundle containing both Final Cut Pro 5 and Soundtrack Pro, along with a v2 update to their Motion motion graphics/animation software and version 4 of DVD Studio Pro.

Introducing Soundtrack Pro Calling Soundtrack Pro an update is a bit of an understatement, as Apple have added what seems to be a tremendous amount of functionality to this application that started life as a bundled companion to Final Cut Pro. One of the new features is a Waveform Editor for processing audio files (including audio exported from Final Cut Pro) based on action layers that allow you to make adjustments to processing and undo it at any time. Over 50 plug-ins from Logic Pro 7 are featured in Soundtrack Pro, including Space Designer, Match EQ and Linear Phase EQ.

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One nice feature in the Waveform Editor is an analysis tool that allows you to identify artifacts such as clicks and pops, phase problems, silence and signal clipping in your audio. Once identified, you can then choose to fix one or all occurrences of the artifact. In terms of audio restoration, Soundtrack Pro also includes options for noise reduction, with the ability to select a range of audio to specify as a noise 'fingerprint'.

Courtesy of Apple Soundtrack Pro's new waveform editor enables you to process your audio with over 50 bundled plug-ins and comprehensive new tools for analysis and noise reduction.

Soundtrack Pro also incorporates a Mixer window and offers support for the Mackie Control, but the real strength of the application is how it integrates with all the other members of the Final Cut Studio family, making it easy to transfer audio work to and from Final Cut Pro, for example. While Soundtrack Pro, like Soundtrack before it, is aimed at editors who want to produce their own audio for video productions, either the final result or a 'temp' for editing against, there's plenty of functionality to make it useful for audio professionals as well. We'll have a more in-depth look at Soundtrack Pro in a future issue of SOS.

Doing The iPod Shuffle I have to confess that I really do find it hard to resist new gadgets, so this month I finally gave in and bought an iPod Shuffle to complement my now two-year-old, 10GB, third-generation iPod. The iPod Shuffle was introduced by Apple CEO Steve Jobs at the Macworld San Francisco show back in January and is the first iPod to use flash memory rather than a hard drive to store your music and other data. The Shuffle naming comes from the fact that, according to Apple, one of the most popular playback modes on iPod players is shuffle, where a random playlist of music is built from all of the songs on an iPod, avoiding repetition, hesitation and deviation. However, choosing the shuffle concept for a new iPod has influenced more than just the device's name. In addition to being the first iPod that doesn't have a hard drive, iPod Shuffle is also the first iPod not to feature a display — two design choices that make it seductively small, at just 3.3 x 0.98 x 0.33 inches, and weighing 22 grams. While leaving out the display was a great way to bring down the cost, Apple figured that if you're listening to a random selection of songs, you don't really need to make specific choices about the music you want to hear, which is a fair point. So, since there's no display, the control and operation of the iPod Shuffle is kept fairly simple. The front surface features a round play/pause button (which doubles as a hold function when you press it down for a few seconds) with volume up/down and file:///H|/SOS%2005-06/OS%20X%20Tiger,%20Apple%20Soundtrack%20Pro%20and%20more....htm (2 of 5)9/28/2005 2:42:48 PM


next/previous track buttons around the circumference. The next/previous track buttons also function as fast-forward and rewind buttons when you hold them down, although I found this a bit tedious and the rewind mode didn't seem to work well at all. Finally, a twocolour LED indicates certain functions of the Shuffle, such as battery charging or play/hold modes. The iPod Shuffle, seen here plugged into a On the reverse of the iPod Shuffle is a 15-inch Aluminium Powerbook. switch to turn the player off or activate one of two play modes: either play in order, which is great for listening to a single album all the way through, or, of course, shuffle. There's also a battery indicator that lights up another LED when pressed, to show whether the battery is healthy (green) or ready to die (red).

The iPod Shuffle connects to a Mac (or Windows computer) via a USB 2.0 plug on the bottom of the device. Although an optional USB charger is available from Apple, the iPod Shuffle charges directly from the computer's USB port when plugged in and has a maximum battery life of 12 hours. Unlike some USBpowered devices, the Shuffle seems to be able to draw enough power when plugged into hubs, extenders and keyboards, which is rather useful compared to some USB devices I've used, that give 'low power' warnings when connected to anything other than the computer itself.

Tiger Preview It's almost a year since Apple CEO Steve Jobs announced Mac OS 10.4 — or Tiger, as it's more affectionately known — at the 2004 WWDC (Worldwide Developer's Conference). And by the time you read these words, this major new version of the Mac operating system will finally be on sale. While Panther (10.3) brought many improvements to the Mac platform in general, the changes specifically aimed at music and audio were perhaps less significant than in Jaguar (10.2), which introduced the Audio MIDI Setup utility and was the point at which both Core MIDI and Core Audio really matured and became usable by a wide range of developers. Courtesy of Apple Tiger is also going to be a significant operating system release for Mac-based musicians and engineers, because there are some really useful and exciting improvements to both Core MIDI and Core Audio, as well as support for standards such as OpenAL, a 3D audio API



(Application Programming Interface) that's useful for audio in games. One interesting feature for those using multiple Mac systems in their studios is that Tiger's Core MIDI features the ability to send MIDI data over a network, thanks to new options in Audio MIDI Setup that are configured in much the same way as the IAC (Inter-Application Communication) Driver. Another new feature in Audio MIDI Setup is the Aggregate Device Editor, which lets you combine multiple audio devices into one so-called 'aggregate' device, that appears to applications as a single Core Audio device. This functionality finally allows you to use multiple devices at the same time, in applications such as Logic Pro that had been missing this ability since the transition to OS X from OS 9 nearly three years ago. Tiger will also introduce the Core Audio Format (CAF) file, a new 64-bit audio file format that will allow large audio files to exist. Apple state that you can record "a thousand channels of audio for a thousand years in a single file", which should be enough even for new age musicians! CAF files will also offer metadata support for storing information about the audio, and the audio itself can be stored in uncompressed formats, such as PCM, as well as compressed formats, such as AAC. Quicktime 7 will already support CAF files when Tiger ships, and I'm sure other Mac audio applications - especially Logic, Soundtrack, and maybe Garage Band — will support this format fairly soon. We'll be taking a full, in-depth look at Tiger and how to use the new Core Audio and MIDI features in next month's magazine.

Autofill Me Up Like all iPods, you transfer music to and from the iPod Shuffle using Apple's iTunes software. The latest version has some specific options for Shuffle users. While you can drag music manually to the Shuffle, you can also use an enhanced Autofill mode that automatically fills the device (and replaces older content) with songs from your iTunes library. This can be either a completely random selection, or the highest rated songs from your collection. In the iPod Preferences window there's a great feature that I always wanted for my bigger iPod, which is the ability for any song transferred to an iPod Shuffle to be automatically encoded into a 128kbs AAC file. This is great because it allows you to keep your main iTunes library in a lossless format, without having to keep a smaller, compressed second copy for listening to on your iPod. Hopefully, Apple will implement this for other iPod users as well, someday. As an accessory for Mac musicians and audio engineers, the iPod Shuffle can be quite useful. The main reason I bought one was because I needed a new USB storage device, and the iPod Shuffle can be used as one, thanks to an option in iTunes' iPod Preferences that allows you to set how much of its memory should be reserved for music and how much should be left for data. For me, this is great for keeping all the latest plug-in and application installers in my pocket when having to maintain multiple computer systems. Secondly, while the 'shuffle' concept doesn't really appeal to me as a way of playing music, if you're working on a mix and want to listen to it on headphones and walk outside for a while — something many iPod users do, apparently — the iPod Shuffle serves this purpose as well. And it's worth remembering that, like all iPods, the Shuffle can



also play uncompressed audio, such as WAV files. The iPod Shuffle is available in two models, the first with 512MB memory for £69 and the second with 1GB memory for £99. They come with the obligatory white 'mug me, I've got an iPod' headphones, a lanyard for wearing your iPod Shuffle around your neck, and a cap that protects the USB plug when the player is in your pocket, if you don't want to make a fashion statement. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Understanding Audio Files In Cubase SX

In this article:


Of Files & Clips Workshop Regional Variations Published in SOS June 2005 Organising The Pool With Sub-folders Print article : Close window Dive In If You Want To Know Technique : Cubase Notes A Guide To The Pool's Columns Pool Management Renaming & Deleting

We take a look at the concepts of audio files, clips, events, parts and regions in Cubase, and explain how you can manage these objects in the Pool window. Mark Wherry

In April's Cubase Notes we looked at processing audio offline in Cubase SX/SL, including the use of the Offline Process History feature for undoing processing carried out on audio files. When you're doing this kind of work, it's important to understand exactly how audio is handled within Cubase — so if you're not sure about the differences between files, clips, events, parts and regions, this should be the workshop to clear up any misunderstandings.

Of Files & Clips Audio that you record or import into Cubase is stored as a regular file on your hard disk. You'll notice that recorded audio is automatically stored in the Audio folder inside a Project folder, and when you import an audio file into a Project, Cubase always gives you the option of copying that file into the Project folder, so that you can keep all the files pertaining to a given Project in the one Project folder. However, under the bonnet, rather than representing audio files within the Project Folder as a series of files in the Project itself, Cubase references every audio file used in a Project as an audio clip. The reason for this higher level reference is because of how the Offline Process History works. When you process a section of audio in Cubase, the processed section is stored in a completely different file (in the Edits folder of the Project folder) so that the original audio file isn't actually touched. The relevant clip is updated so that playback is seamless between the unprocessed and processed files — and because Cubase only presents clips to

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the user, this whole mechanism for creating extra files for edits is completely transparent and is something you'll rarely need to think about. Basically, you can think of a clip as a playlist of audio files. When you add audio to the Project window, either by recording it or by importing it, Cubase creates an audio event (which is represented by a The hierarchy of audio clips, events and rectangular box, as you'll have seen) that essentially triggers the playback of parts. In the Project window is a part containing four events. The Audio Part Editor a clip. An audio event doesn't shows those events, and in the Sample necessarily need to play the full clip — Editor one of them is ready for editing. The it's possible for the event to start and Pool shows that each clip is used by an finish at any point within the clip — and event once in the Project. you can even have multiple audio events on different tracks playing from the same clip simultaneously. In the same way that a MIDI part contains MIDI events, an audio part is a collection of audio events. You can put a group of audio events in a track into an audio part by selecting the appropriate events and choosing Audio / Events to Part. This can be useful when you want to group a series of events into one object for moving and copying, ensuring that everything stays together in the correct positions. For example, say you create a four-bar pattern from four onebar events and want to duplicate the four-bar pattern several times. Grouping the four Events into a Part makes this easier to manage. Double-clicking an audio part opens the Audio Part Editor, which allows you to work with the events inside the part just as you would in the Project window. Double-clicking an audio event in the Audio Part Editor opens that event in the Sample Editor, exactly as it would if you double-clicked an audio event in the Project window. To break an audio part back into audio events again, select the relevant part and choose Audio / Dissolve Part.

Regional Variations The Cubase manual describes a region as a section within a clip, and you can basically think of it as a kind of bookmark or internal reference point within a clip. Clips can contain many regions, and once a region has been created you can make a selection in the Sample Editor based on its start and end points, drag it to the Project window to create a new audio event, or save it to disk as a new audio file. One instance where regions can be useful is drum tracks, where you might want to define parts of an audio event that contain a good snare drum hit, bass drum file:///H|/SOS%2005-06/Understanding%20Audio%20Files%20In%20Cubase%20SX.htm (2 of 9)9/28/2005 2:42:55 PM


hit, or some section of audio you might need to use later on to patch up weaker areas of the same drum track with better hits. You could also apply this same example to any recording, of course, using regions to index bass guitar or vocal notes as well. There are many ways to create regions in Cubase, including manually in the Sample Editor. To do this, open the Sample Editor by double-clicking an audio event in either the Project window or Audio Part Editor (doubleclicking an audio part in the Project window always opens the Audio Part Editor). You can define regions in the Sample Editor The Sample Editor always shows the as reference points within an audio event. complete clip that an audio event is Notice how any regions defined in an event taken from, but the parts of the clip not are also listed in the Pool. used by the audio event you're editing are always greyed out. You can see the list of regions for the clip being displayed in the Sample Editor by enabling the Show Regions button on the Sample Editor's toolbar. This is the second button in, to the right of the pop-up menus.

To create a new region, simply mark up the area you'd like to define as a region, select Audio / Create Region (or click the Add button in the Region List), type in a name for the region and press Return. Once the Region List is visible, you can select a region by clicking in the unlabelled column to the far left of the list. Clicking in the Description column renames a region, and clicking in either the Start or End columns allows you to change these values by typing in new positions. The currently selected region is always shown by default in the Sample Editor and you can remove a selected region in the list by clicking Remove, or select the area in the Sample Editor defined by a region by clicking Select. When a region is selected, you can resize it graphically by dragging the Region Start and End flags in the Sample Edit window, just as you can change the start and end points of the audio event you're editing by dragging the Event Start and End flags. Once you've defined regions, such as drum hits, notes, or other useful sections, you can use them in your Project by dragging the region from the unlabelled column in the Region List to the Project or Audio Part Editor windows, which creates a new audio event based on that region. As a side note, regions can be automatically created when using Cycle Record Mode, to indicate different takes of the same recording if you record those takes without stopping in between each take, as discussed in November 2002's Cubase Notes (www.soundonsound.com/ sos/nov02/articles/cubasenotes1102.asp).



Organising The Pool With Sub-folders The Audio folder in the Pool can soon fill up with hundreds (and thousands) of audio clips, so in order to keep things nice and tidy you can create sub-folders inside the Audio folder to organise the list of clips. For example, you might want to organise all your imported drum loops into one folder, use another folder for newly recorded audio, and an additional folder for files you imported from another musician you're collaborating with. To create a sub-folder:

It's easy to organise clips and events in the Pool into folders and sub-folders, although these exist only in the Pool and don't affect how the audio files are organised on disk.

Select the folder in which you want to create a sub-folder. Choose Pool / Create Folder, type in a name for the folder and press Return. You can later rename a folder as you would a clip, by simply double-clicking it, typing in the new name and pressing Return. It should be noted that these sub-folders only appear in the Pool and aren't created in the Audio folder inside the Project Folder on your disk, so the files that the clips reference are unaffected.

Once you've created a sub-folder, it's possible to place clips in it by simply dragging them on to that sub-folder. Newly recorded clips are always automatically placed inside the Pool's Record Folder (indicated by the word Record in the Status column), which is the root Audio folder by default. When you import audio, however, the new clip is created within the currently selected folder. To change the Pool's Record Folder, simply select the folder you would like to define as the Record Folder and choose Pool / Set Pool Record Folder.

Dive In If You Want To Know To help you manage the audio clips and associated regions used in a Project, Cubase provides the now somewhat infamous Pool, which is basically a catalogue that lists every audio clip you've recorded or imported into the Project, along with any regions you might have created within those clips. Open the Pool by selecting Project / Pool or using Control/Apple-P. By the way, it's worth noting that you can import files into the Pool without having to place them in the Project window, by clicking the Import button on the Pool's toolbar or selecting Pool / Import Medium. This is great if you have a whole set of drum loops that you might want to use in a Project, for example, but you're not sure which ones you're actually going to use. Importing all the possible loops into the Pool allows you to organise and audition them, dragging the loops to the Project window if you hear something you want to use. This saves you having to create multiple tracks in the Project window, enabling and disabling the mutes to audition the audio. The Pool contains three folders, labelled Audio, Video and Trash. Audio clips are file:///H|/SOS%2005-06/Understanding%20Audio%20Files%20In%20Cubase%20SX.htm (4 of 9)9/28/2005 2:42:55 PM


placed into the Audio folder, video clips are placed into the Video folder (for when you're working with desktop video files), and the Trash folder works much like the Trash or Recycle bin on the Mac or Windows desktops. Expand or collapse a folder in the Pool by clicking that folder's expand ('+') or collapse ('-') button, or collapse and expand all the folders in the Pool by clicking the Expand or Collapse All buttons on the Pool's toolbar. Because regions are references within audio clips, the Pool also keeps track of the regions that have been created and groups them with the relevant clips, allowing you to work with regions in almost exactly the same way you'd work with the original clips themselves. If a clip has an expand button, you can click this to see the list of regions that have been created within that clip.

The Pool keeps track of all the audio files, clips, events and regions used in a Project, and also provides a Trash folder to help you manage deleting files from a project, in addition to keeping track of any video files and clips.

You can place a clip or region from the Pool as an event on the Project window by simply dragging it, and Cubase will create an audio track for you automatically if one doesn't already exist. Alternatively, place the selected clip or region on the selected track in the Project window (at the current location of the Project cursor) by selecting Pool / Insert into Project / At Cursor. It's also possible to use the Pool / Insert Into Project / At Origin command to place a clip back into the Project window at the point it was originally recorded. Sometimes you might want to update the Origin Time for a clip, however. Say you've recorded something but you came in slightly early. Moving the audio event that plays the relevant clip forward a little in the Project window, so that everything is in time is, of course, very easy, but doesn't update the Origin Time recorded for that clip. In other words, inserting this clip as an event in the Project window at the Origin Time will always insert it in the original, slightly early position. So to update the Origin Time of a clip or region to the current location of the Project Cursor, select the appropriate clip or region in the Pool and select Pool / Update Origin. To save a region as a new clip (which also means as a new audio file on your disk), first select in the Pool the region that you want to save as a new clip. Now select Audio / Bounce Selection (this command is also available from the Quick Menu) and choose a destination folder in the file selector that appears (you'll probably want to select the Audio folder inside the Project Folder). A new clip is created in the Pool, referencing the new file that's been created in the Project Folder based on the region you selected. Note that the region's name is used for the clip by default, although you can change this if you wish. Creating new clips (and so files) from regions in this way is useful when you want to make modifications to a region but you don't want to affect the clip from which file:///H|/SOS%2005-06/Understanding%20Audio%20Files%20In%20Cubase%20SX.htm (5 of 9)9/28/2005 2:42:55 PM


the region is derived. For example, say you have a region containing one snare drum that you want to duplicate in the Project window and apply an effect to: you don't want this effect to be heard when the original clip is played back. Bouncing the region to a clip/file, as described above, and then applying the effect to the newly created clip will leave the original clip intact. You can audition a clip or region in the Pool by selecting it and clicking the Play button on the Pool's toolbar. Clicking the Play button again while the clip or region is playing will stop the playback. If you also have the Loop button activated on the Pool's toolbar, the selected clip or region will be looped when you press the Play button, until you either press the Play button again or deactivate the Loop button. It's worth noting that the Play and Loop buttons on the Pool's toolbar are completely independent of the transport controls for the Project.

A Guide To The Pool's Columns Media This column is always visible and lists the name of the clips and regions used in a Project. Used (Use Count) The number of times an event in the Project references a clip. Status There are four possible icons, which can indicate whether: [~] The clip has been processed, and therefore plays back from multiple files. [?] The file references by the clip can't be found. (See the main text for info about renaming and deleting files in a Project to avoid this situation!) [X] The file referenced by the clip is stored externally and isn't located in the Project folder. The total number of external files is detailed in the Infoline at the bottom of the Pool when the Show Info button is active on the Pool's toolbar. [R] The Clip has been recorded since you last opened the Project, which is handy for seeing recently recorded clips at a casual glance.

In addition to these four icons, the word 'Record' is used to indicate the folder that newly recorded clips will be placed into. Musical Mode This checkbox allows you to enable and disable Musical Mode (automatic, realtime timestretching) for a clip, rather than doing it in the Sample Editor, which is actually rather helpful when you're working with a large number of clips in Musical Mode, since it saves you from opening, closing and re-opening the Sample Editor multiple times. The only catch is that before you can enable Musical Mode a tempo must already have been specified: if not, Cubase will prompt you to enter a tempo before enabling the mode. Tempo The tempo of the clip, used by Musical Mode (see above). It's possible to change



this value by clicking on it, typing a new value and pressing Return. Unless you import an Acid-format Wave file, the tempo of the audio will be unknown to Cubase and displayed as question marks. Signature The time signature of the clip, which is 4/4 by default (unless specified in an imported Acid-format WAV file) and can be changed in the same way as the Tempo value. Info Where the original file that the clip references is stored on your disk. Type The file format used to store the clip — for example, WAV or AIFF. Date The date and time when the clip was created. Origin Time The location in the Project at which the clip was originally created. When you're importing audio files, note that only Broadcast WAV files (not regular WAV or AIFF files) contain time-stamps to fill the Origin Time field. Image A visual representation of the clip or region's waveform. You can also click on this image to audition the clip or region, onwards from the point at which you clicked. Path The full path-name of the file, allowing you to see exactly where on your disk the file is stored. This is particularly useful when dealing with files that are external to the Project folder. Reel Name This field is filled only when audio files are imported from OMF files and contain a reference to a Reel Name, useful if you're importing OMF from Avid, for example, where you want to keep track of what video/film reel the clips originally came from.

Pool Management Like the list-style view for navigating computer disks and folders on your desktop, the Pool consists of a number of columns that give you different information about clips (see the screen on page 264). You can choose which columns are displayed by toggling the options in the View pop-up menu on the Pool's toolbar, and you'll notice that there are also Show and Hide All commands available from this menu, so you can show or hide all the available columns with just one mouse-click. Optimise the width of the columns to the smallest width that's useful by selecting Optimize Width from the View pop-up menu. Should you want to, you can also change the order of the columns horizontally, by simply dragging



their titles to a new position. It's possible to organise how the list of clips appears in the Pool by clicking on the column titles, just as you might on Mac or PC desktop windows. By default, the clips are listed in alphabetical order according to clip name, which is indicated by the upwards arrow in the Media column. However, you can click on any of the other column titles to list the clips according to the data represented in that column. A second click in the same column reverses the order of the list. As mentioned earlier, the Pool exists as a way for you to manage and keep track of the clips and regions used in a Project, so it stands to reason that there are many useful facilities available for keeping the Pool nice and tidy. To begin with, there's a handy Show Info button on the Pool's toolbar that toggles the appearance of an Infoline at the bottom of the Pool, to give an overview of the clips used. From left to right, the Infoline details the number of audio files in the Pool, the number of files used by events in the Project, the total size of the audio files used for clips in the Pool, and the number of external files referenced by clips in the Pool (external files are ones not inside the Project folder).

Renaming & Deleting A clip always takes the same name as the file it references in the Project folder. If, for some reason, you want to rename either clip or file, you should never rename it in the Project folder in the same way you might normally rename files on your computer. If you do, Cubase will no longer be able to find the renamed audio files and recognise them as part of your Project. Instead, to rename a clip (and the actual audio file), double-click the name of the clip in the Pool's Media column, type in a new name and press Return. You'll notice that both the clip and the file on disk are updated to take on the new name, and it's worth noting that where you have a clip that references multiple audio files the first, or original source, file is the one that is renamed. Regions can be renamed in the same way, although this has no effect on the name of the associated clip or any of the files that are stored on your disk. In the same way that you shouldn't rename audio files used in a Project from your computer's desktop, you absolutely shouldn't delete audio files from a Project Folder as you'd normally delete other computer files. Use the features within the Pool to do this instead. The Pool has a two-stage process for deleting clips (and so files) that works in much the same way as you'd delete files on your computer normally — first dragging them to the Trash or Recycle bin, then 'emptying' it. Select the clip (or clips) you want to delete and press Backspace. If you're trying to delete a clip that's used by an event in the Project, Cubase will warn you that the clip is still being used and ask whether you want to delete all occurrences of events that use the clip in the Project. If you do, click Remove; otherwise, click Cancel, which aborts the deleting process. file:///H|/SOS%2005-06/Understanding%20Audio%20Files%20In%20Cubase%20SX.htm (8 of 9)9/28/2005 2:42:55 PM


Cubase will prompt you to specify whether you want to move these clips to the Trash folder (for deleting) or just remove them from the Pool. If you want to delete the actual audio files from the disk, in addition to removing the references to the clip (or clips) from the Pool, click Trash. If you just want to remove the clip (or clips) from the Pool, without deleting the audio files on disk, click Remove from Pool. (If your intention was always to move these clips to the Trash folder, you can also simply drag the required clips to the Trash folder.) Once you have some clips in the Trash folder that you want to delete, simply select Pool / Empty Trash, and Cubase will ask you again whether you want to delete the files from disk or simply remove the references from the Pool. If you want to delete the files permanently, click Erase. To simply remove the clips from the Pool, click Remove from Pool. If you delete all the events from the Project window that reference audio clips recorded since you last opened the Project (any flagged with the 'R' record icon in the Status window), an interesting behaviour to note is that Cubase will automatically move the clips to the Trash folder for you. You can later remove them from the Pool or delete the file altogether by following the last step above.





In this article:


Timecode Commands Workshop Zoom Functions Published in SOS June 2005 Timeline & Edit Selection Commands Print article : Close window Navigation Commands Technique : Pro Tools Notes Modifying Region Boundaries, Cutting And Pasting Toggle Settings Nudging Selections Most people find that learning keyboard commands Conclusions From The 'Pain for their DAW enables them to get things done faster, Barrier' and for the advanced user, Pro Tools has a special Aides Memoires

mode which turns the entire QWERTY keyboard into a bank of one-press shortcuts. Mike Thornton

Keyboard shortcuts have been with Pro Tools from the start, and the number of shortcuts in the program has grown as the system has developed. The more shortcuts you know, the better, because you move your mouse less and become more productive. As well as conventional keyboard shortcuts involving a combination of standard keys and 'modifier' keys such as Ctrl and Alt, Pro Tools also provides something called the Keyboard Commands Focus mode. This can be a real boon in many situations, as it turns most of the keys on the QWERTY keyboard into shortcuts that don't require you to hold down a modifier key. You can enter Keyboard Focus mode by clicking on the the [a...z] button, which is located just below the zoom buttons in the Edit Window. These are available to use on all Windows- and Mac-based Pro Tools systems, with only a few timecode-related limitations on the LE systems. Good news for those who work with both Windows- and Mac-based systems that, unlike some other shortcuts, the Keyboard Focus shortcuts are the same for both platforms. There is a 'pain barrier' to go through to get the Keyboard Focus shortcuts into your way of working, but rest assured it will be worth it! If you have the chance, it might be worth going through this learning curve with a project where the deadline isn't too pressing. Also, you may find it helpful to look either at a real file:///H|/SOS%2005-06/Using%20Pro%20Tools%27%20Keyboard%20Commands%20Focus.htm (1 of 7)9/28/2005 2:43:04 PM


QWERTY keyboard or the photo on the right, as the commands are grouped together in areas of the keyboard to make them as intuitive as possible.

Timecode Commands There is a small set of additional Keyboard Commands Focus commands that relate to timecode features or the use of a stationary play head, and are available only in TDM versions of Pro Tools. Snap to timecode When you have 'parked' your machine at a suitable timecode position, these three commands allow you to spot the highlighted region in Pro Tools to the 'parked' timecode position. Pro Tools will obviously need incoming timecode from a slaved machine for this to work. Snap start of selected region to timecode: 'Y' Snap sync point of selected region to timecode: 'U' Snap end of selected region to timecode: 'I'

Snap to stationary play head This group of commands allows you to snap your selected region to the play head position. Snap start of selected region to play head: 'H' Snap sync point of selected region to play head: 'J' Snap end of selected region to play head: 'K'

Zoom Functions Zoom level 1-5: alpha keys '1' to '5' By using the numbers '1' to '5' on the main part of the keyboard (not the numeric keypad), you can access the five preset horizontal zoom settings that appear as five buttons just below the zoom buttons in the Edit window. To save your own zoom values as presets, adjust the horizontal zoom to suit and then Command +click on the Mac, or Ctrl+click on Windows, on the appropriate button to save the preset. Zoom toggle: 'E' This will zoom an edit selection (ie. a highlighted selection on a track, rather than a timeline selection) to the width of the Edit window. If you press 'E' again, it will take you back to the previous zoom setting. Centre edit selection start: 'Q'

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This will put the start of an edit selection in the centre of the Edit window. Centre timeline end: 'W' This will put the end of an edit selection in the centre of the Edit window. Zoom out horizontally: 'R' This is the same as going up to the toolbar and clicking on the zoom out button, or using the Command/Ctrl+'[' shortcut when not in Keyboard Focus mode. Zoom in horizontally: 'T' This is the same as going up to the toolbar and clicking on the zoom in button, or using the Command/Ctrl+']' shortcut when not in Keyboard Focus mode. Play to, or from, an edit point by the pre- or post-roll settings: alpha keys '6' to '9' When you have a potential edit highlighted, this group of commands allows you to: Play up to the selection using the pre-roll setting: alpha key '6' Play from the start of the selection by the post-roll setting: alpha key '7' Play up to the end of the selection by the pre-roll setting: alpha key '8' Play from the end of the selection by the post-roll setting: alpha key '9' This is very useful for checking whether you have included or excluded a particular element of audio from an edit selection.

Timeline & Edit Selection Commands These work only when you are working with the timeline and edit selections separated. You can separate the timeline and edit selections by clicking on the 'separate' button that is to the right of the Keyboard Focus button so it isn't highlighted. Copy edit selection to timeline selection and vice versa



There is a pair of commands that enables you to copy a selection from an edit on a track up to the timeline (the letter 'O'), and to copy a selection from the timeline onto an edit on a track (the alpha zero key, '0' —not the zero on the numeric pad). Play timeline selection: ']' This will play the selection made on the timeline (in the screen below, this would be the highlighted section in the Time Ruler above Region B). Using the space bar will also play this option. Play edit selection: '[' This will play the selection made on the track (in the screen above, this would be the highlighted section in Region A). If the Timeline and Edit selections are linked then either command will play the highlighted section.

Navigation Commands Move edit selection up: 'P' This will move a selection from one track to the track above, regardless of whether there is any audio on it at that point. Move edit selection down: ';' (semicolon) This will move a selection from one track to the track below. If no selection has been made, this and the 'P' command will move the cursor down or up by a track. Tab back: 'L' This will move the cursor back (earlier in time) to the previous region boundary or sync point on the same track. * Tab forward: ' (apostrophe) This will move the cursor forward (later in time) to the next region boundary or file:///H|/SOS%2005-06/Using%20Pro%20Tools%27%20Keyboard%20Commands%20Focus.htm (4 of 7)9/28/2005 2:43:04 PM


sync point on the same track.

Modifying Region Boundaries, Cutting And Pasting Trim start to insertion: 'A' If you place the cursor part of the way into a region, hitting 'A' will delete the audio from the cursor back to the beginning of that region, as shown in the top two screens above. Trim end to insertion: 'S' If you place the cursor part of the way into a region, hitting 'S' will delete the audio from the cursor forward to the end of that region. Fade to start: 'D' When the cursor is placed part of the way into a region, hitting 'D' will create a fade-in (using the default fade settings in Preferences) from the cursor position back to the beginning of that region. Fade: 'F' When a selection is highlighted, hitting 'F' will create a crossfade (using the default fade settings in Preferences) without giving you the Fades dialogue window. Fade to end: 'G' When the cursor is placed part of the way into a region, hitting 'G' will create a fade-out (using the default fade settings in Preferences) from the cursor position forward to the end of that region, as shown in the lower two screens above. Cut and paste commands: 'X', 'C', 'V' In Commands Focus mode, the usual Cut ('X'), Copy ('C') and Paste ('V') controls are available to you without the need to hold down the usual modifier keys (Command for Mac and Ctrl for Windows). Separate: 'B' This does the same as Command+'E' on the Mac and Ctrl+'E' on Windows. If you file:///H|/SOS%2005-06/Using%20Pro%20Tools%27%20Keyboard%20Commands%20Focus.htm (5 of 7)9/28/2005 2:43:04 PM


have the cursor placed in a region, hitting 'B' will create an edit at the cursor position, as shown in the top two screens below. Similarly, if you have highlighted a selection, hitting 'B' will create edits at both ends of the selection (middle pair of screens).

Toggle Settings The next two commands toggle between two states each time you click on them. Timeline insertion follows playback on/off: 'N' This allows you to turn on and off 'insertion follows playback'. With this preference on, the cursor will stay where it is when you stop playback and will continue on from that point when you hit Play. With it off, when you stop playback the cursor goes back to the point you started to play from. Before this shortcut was added to Pro Tools you had to go into the preferences to change this setting, which was a real pain. Track view toggle: '-' (minus key on main keyboard, not the numeric keypad) This will toggle the track that the cursor is on from Waveform to Volume Graph and back, as shown in the lower two screens. You can use this in conjunction with the Option or Alt keys, and if you have groups selected, this will work across them too.

Nudging Selections The next group of four enables you to nudge selections both by the Nudge value set in the menu bar — nudge back is '' — but also by the next nudge value up from the set one. The key to nudge back by next nudge value up is 'M', while the key to nudge forward by the next nudge value is '/'. Finally, perhaps the most useful key command of all is, again, available in Commands Focus mode without the need of a modifier key... Undo: 'Z'



Conclusions From The 'Pain Barrier' For me, the best features of the Keyboard Commands Focus are the ability to zoom in and out both with the Preset zooms and the incremental zoom commands ('T' and 'R'). These alone have made it worthwhile to have the Commands Focus on, and the Track View toggle, zoom toggle, Separate and Nudge keys are serious bonuses. I will probably still use the Smart Tool for fades and the Trim Tool for trimming regions, but long live Commands Focus!

Aides Memoires If you're having trouble remembering all the Commands Focus commands, you could try... Buying a set of keyboard stickers, costing £12. You can stick these on to your existing keyboard, or buy a second keyboard to attach them to, as the 'normal' letters on the stickers aren't very visible when typing. Buying a fully printed and colour-coded keyboard costing £160.

Both are available at Digidesign's web store: http://secure.digidesign.com/uk/ Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Using Automation In Digital Performer 4.5

In this article:

Basic Concepts Mixing Board Automation Using The Sequence Editor MIDI Automation Automating Plug-ins Automation Setup Snapshots

Using Automation In Digital Performer 4.5 Workshop Published in SOS June 2005 Print article : Close window

Technique : Digital Performer Notes

If you're a user of MOTU's Digital Performer sequencing software, you have access to an extensive and full-featured automated mixing environment. For hints and tips on how to use it, read on... Robin Bigwood

In Digital Performer, mix automation allows the recording of movements of on-screen controls and parameter values such as Mixing Board level faders and pan knobs, track mute buttons and plug-in parameters. By taking advantage of automation in DP Three types of automation data are visible in you can make your mixes more fluid this track, though the active layer is a 'popup' or 'switch-style' data type, in this case and reliable, deal with thorny problems allowing a Multimode Filter's filter type to be such as tracks that vary wildly in level, automated. and achieve contemporary-sounding effects such as filter sweeps and dynamic low-fi treatments. If you're using software instruments hosted in DP you can also make automation change multiple parameters simultaneously — even more than if you had a knobby analogue synth in front of you!

Basic Concepts The first thing to remember is that automation is track-based in DP. In the case of a track with an automated level fader, for example, the automation data is part of that track — there's no separate 'automation track'. Data is mostly displayed as lines, 'ramps' and breakpoints superimposed on tracks in the Sequence Editor, so it's easy to see how automation and track content align.

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Second, automation controls and facilities crop up in different guises in the DP environment, with MOTU providing lots of different ways to crack the same nut. DP's automation is not as straightforward as its simpler equivalent in Reason, for example, but that's not to say it's difficult to use. Some aspects are gloriously intuitive and require little brainwork, although, as ever in DP, there's tremendous depth and flexibility when you need it. Finally, it's worth noting that there's a fundamental difference between mix automation — that is, automation of parameters related to DP's audio architecture — and MIDI automation. Although MOTU make some attempt to harmonise their behaviour, they're really chalk and cheese. Most of what follows, then, relates primarily to mix automation, although you'll find a separate box discussing how MIDI automation works.

Mixing Board Automation For many DP users, from beginners to seasoned pros, automation means one thing: recording the movements of controls in the Mixing Board — level faders, pan controls, mutes, send levels, and perhaps the odd plug-in bypass parameter. This equates to what is possible on really high-end analogue desks, and most recent digital mixers, and it's very easy to work with. The key to Mixing Board automation is the dedicated automation section that is available on all tracks in the Mixing Board. You do need to make sure it's visible, by selecting 'Solo/Mute/Record/Auto' in the Mixing Board's mini-menu. The automation section offers separate buttons for record-enabling and playenabling track automation, as well as a mode switch, which defaults to 'Latch'. (There's more information on automation write modes in a moment.) Recording some fader movements for a track, then, is easy: Locate your sequence to the point just before you want to record the movements and click the automation record-enable button for the track.

The section of the Mixing Board that offers an easy and intuitive way of working with automation. Notice the automation mode pop-ups beneath the play and record buttons.

Play your sequence (no need to hit Record) and wiggle that fader! When you reach the end of your automated section, stop playback and de-select the automation record-enable button. Automation playback must still be enabled. Rewind to the original starting point, hit play, and you should be able to watch your fader fly about all by itself, with corresponding changes in level.



Using automation in this way is very intuitive, and it works for track faders, pan controls, mute status, send settings (so long as your sends have valid output assignments) and plug-in bypass (which you can control from the Mixing Board by option-clicking a plug-in slot). You need never leave the Mixing Board, and the visual feedback offered keeps you in touch with what's going on. If you need to re-record a section of automation that's already in place, you can simply recordenable once more and do another automation record pass — but it's here that the automation modes suddenly become important. Latch: When Latch is selected, faders and pots follow any automation data already in the track until such time as you grab the control and record new movements, but when you let go it 'latches' or 'sticks' wherever you leave it, blitzing the remaining automation data already in the track for as long as the record pass continues. This is great for when you essentially need to re-do a long section, because you can let go of the mouse mid-way. Touch: If you just need to make a momentary tweak here and there, you might give Touch mode a go. Again, faders and pots track existing automation data until you click and drag on them, but then new data is recorded only for as long as you're actually 'touching' them — the moment you let go they're back tracking existing data. Overwrite: This mode is like starting again with a track, as it ignores existing automation data and literally overwrites it from the moment the record pass begins to when it ends.

Switching a track to a 'Trim'

automation mode in the Mixing Board causes its fader to be recalibrated to a relative, not absolute, scale.

Once you're familiar with Latch and Touch modes for modifying existing automation data, you can also experiment with their Trim equivalents. These two additional automation modes allow you to make relative, not absolute, changes to existing automation data. When you select a Trim mode for a track you'll notice that its fader and pot calibration changes, with the conventional scale being replaced with a series of positive and negative values either side of a central zero point. The idea is that should you want to increase the volume of an automated section by, say, 2dB, you switch to Trim Latch, do an automation record pass, and at the right moment simply raise the fader to the +2dB mark and leave it there for as long as necessary.

Using The Sequence Editor Although it's possible to work with automation using only the Mixing Board, you get more control, and more options, when you use the Sequence Editor as well. Here, automation appears as lines and breakpoints in track lanes, and often, for audio tracks, superimposed on soundbites in the track. The breakpoints file:///H|/SOS%2005-06/Using%20Automation%20In%20Digital%20Performer%A04.5.htm (3 of 9)9/28/2005 2:43:34 PM


represent the actual automation data, and the lines between them are just the resulting 'links' between breakpoints. Exactly what automation is shown is down to which track 'layer' is currently active, and this can be changed using either the so-called Active Layer popup menu at the top of a track's Information Panel, or the Track Here, the Sequence Editor shows a single Information menu (under the Edit subtrack, with some volume automation menu item). The latter is particularly inserted. The 'active layer' is chosen with the useful when your track is sized very track lane's uppermost pop-up menu (which small (see screen at the start of this says 'Volume' here). Automation can be article). Either way you get editing record- and play-enabled with the Auto popup menu, and next to it the Insert pop-up access to any current automation data allows you to insert new data types into the types, and also soundbites track. The function of all the pop-ups is themselves, though soundbites duplicated in the Track Information pop-up effectively obscure automation data menu — the little square to the right of the when they're the active layer. The record-enable button. Active Layer pop-up menu always includes Volume and Pan as options, even when there's no automation at all in a track. However, other automation types show up here as soon as they're part of a track, so if you'd recorded the movements of a send pot, for example, this would also become selectable as an active layer. So what can you do in the Sequence Editor? To begin with, you can enter automation data graphically. To do this, switch the track's Active Layer pop-up to 'Volume' (or another suitable automation data type), hold down the 'P' key to temporarily select the pencil tool, and click in a track lane to write a breakpoint (see screen below). You might notice that as you move the mouse pointer around, the current value for the data you're about to write is shown at the top left of the Sequence Editor, so it's easy to enter a precise value. If the line that extends either side of the resulting breakpoint is dotted, this indicates that automation isn't turned on for the track. To turn it on, either play-enable automation from the Mixing Board or, better still, select 'Play' from the Auto popup menu in the Track Information panel (or, again, via the Track Information menu). You can also record-enable automation or switch automation mode from here. Having entered one breakpoint, you simply click another point on the line that extends from it to write additional ones. Depending on the relative positions of your breakpoints you can create ramps (smooth value changes between two breakpoints) and steps (abrupt changes), and you soon notice that breakpoint drags behave just like any other data with respect to DP's Edit Grid (which can be toggled on and off using the blue box at the top right of the Sequence Editor, or by holding down the Apple key). Deleting breakpoints is simple: either click an existing one to select it, or drag to select several, and then hit delete. Don't be confused by track and plug-in mute automation, both of which only allow



breakpoints at maximum or minimum values, to correspond with bypassed and enabled status respectively. When you enter or drag these breakpoints they seem to 'snap' to the extremes of the track lane, but in practice it's all quite intuitive. If you're really into working with data graphically, all DP's editing tools are at your disposal: you can use the pencil tool to insert lines, curves and periodic waveforms or, indeed, use the reshape tool to work with existing data. To enter automation data for a parameter that doesn't yet show up in the Active Layer pop-up menu, create a layer for it using the Insert popup menu, or the Insert menu item inside the Track Information pop-up menu. In both places are listed any parameter that could be automated on that track, and after you choose one DP automatically switches temporarily to the Pencil tool, so that you can write a breakpoint. Something else you can do in the Sequence Editor is edit automation data written using fader or pot movements in the Mixing Board. This often comes out looking pretty scrappy, Plug-ins get their own automation with dozens of breakpoints, so to alter it and button — the little 'A' at the top left tidy it up you might try dragging new or of their windows — which accesses a pop-up menu and changes colour existing breakpoints left or right 'across' others. DP temporarily deletes them as you do according to play- or recordenabled status. Also notice the this, and it's easy to see what effect your edit 'camera' Snapshot button in the title will have. You could also try selecting a group bar. of breakpoints, perhaps by drag- or shiftselecting or using the I-Beam tool, and choosing Thin Continuous Data from the Region menu. There are options to affect audio volume, pan and other automation data types in the window that appears, and usually only a little tweaking of the Minimum Value and Time Change parameters can tidy things up no end. Moving breakpoints is easy: just select them and drag. To copy a group of breakpoints, you might try alt-dragging them, but this becomes difficult if you need to copy to a position some way away. Instead, select the breakpoints and hit Apple-C to copy them, then move the playback wiper to the point where you want them to end up and use Apple-M to Merge them into the track. The standard Apple-V Paste also works, but you may find other, perhaps unwanted data (such as soundbite fragments) being copied to your new location as well, if you use Paste instead of Merge. It might also be worth pointing out that automation data is anchored to track location, not soundbites, so you need to be careful, if you drag a soundbite that has specific automation data associated with it, that you don't end up misaligning things. The way to keep soundbites and automation together is to zoom out to a level where you can see the soundbites you want to move in their entirety, use



the crosshair cursor or the I-Beam tool to drag-select the soundbite and a little bit either side and then drag the soundbite. The fact that you make a time-range selection means that both the soundbite and the automation data are included in the drag.

MIDI Automation I mentioned earlier that DP's mix and MIDI automation are fundamentally different, and yet in the Mixing Board their automation sections seem to have the same outward appearance. What's going on? The reason why mix and MIDI automation look the same is because MIDI and audio tracks can appear together in the Mixing Board, and it obviously makes sense to harmonise appearance and operation as much as possible. But MIDI automation is a very limited affair, and in fact is just a 'front' for track mute and two types of MIDI message, namely continuous controller number 7 (channel volume) and continuous controller number 10 (channel pan). These are messages that could end up in a MIDI track by other means — perhaps being recorded or even written in manually — but are essential in setting up a MIDI 'mix', especially on a multitimbral synth. Enabling automation playback for a MIDI track, then, is a prerequisite for channel volume and pan messages in a track to be transmitted at all. But you can also use DP's Mixing Board fader and pan controls to write channel volume and pan into a MIDI track, by moving them during an automation recording pass. Obviously, most modern synths can have nearly all their parameters remotely controlled via MIDI messages, but from DP's point of view the presence of any MIDI data in a track, other than continuous controllers 7 and 10, isn't automation — it's just data.

Automating Plug-ins If the Mixing Board is a prime candidate for automation, so are plug-ins. All sorts of creative effects are possible, including automated delay times and feedback amount, muting and enabling long reverbs, sweeping filter cutoffs and, of course, manipulating the parameters of virtual instruments. The principles behind plug-in automation are just the same as for the Mixing Board, but two things are different. First, plug-ins get their own Automation buttons, in the top left of their windows (see screen above). These are, in fact, pop-up menus allowing play- and record-enable status to be set, and automation mode to be switched. Second, some plug-ins have parameters that are not numerical in nature. Examples of this are the Multimode Filter's filter-type buttons and LFO pop-up menus. It would be arbitrary, to say the least, to try to represent data for these graphically in the Sequence Editor, so instead this sort of parameter gets a little 'pop-up' data entry that shows the parameter value by name.



Automation Setup The great thing about automation in DP is that it's largely hands-on and intuitive. However, there are times when you might need to pay the Automation Setup window a visit (accessed via Setup menu / Automation Setup). The really important thing you can do here is enable and disable automation data types for individual tracks, so if you've accidentally done an 'All Data Types...' snapshot It's not pretty, but the Automation Setup window offers you the finest control over and ended up with a track full of enabled automation data types in your largely redundant automation data, tracks, and also allows you to turn off you can disable most of it here. automation globally. Equally, if you just want to start working with a single data type for a plug-in, you could enable it here first, and that would allow you to select it in the Sequence Editor, or use the 'All Enabled Data Types' option when taking snapshots. The other important control is the global automation disable switch. Consider this scenario: you have a sequence full of carefully honed automation data but you want to change one short section by using a snapshot. With automation enabled, you can locate to the section you want to change and set up the new mix parameters, but if you were to start playback to hear what effect they had (or even just move the wiper), DP would instantly 'chase' to the existing automation values in the track. So if you want to experiment with new settings in a section of your sequence that is already automated, globally disable automation first and then you'll have free rein to experiment and take new snapshots without constantly being 'locked out' by existing automation data.

Snapshots If automation is useful to the way you work, DP's Snapshot function will extend your options, and may also let you work faster. So far, we've looked at how automation data can be written into tracks using fader and pot movements, and by entering it graphically in the Sequence Editor. Put simply, Snapshot just gives you another option — it can enter multiple fader, pot and other parameter values into tracks with just a couple of clicks. When you want to take a Snapshot you should ask yourself three questions: Which track (s)? What? Where? To put this in context let's look at a typical role for a Snapshot: capturing initial values of all your Mixing Board settings, for all tracks simultaneously.



Place your playback wiper at the start of the sequence. Then you need a Snapshot button. Only the Sequence Editor, plug-in windows and the Mixing Board have these. After clicking one, ask yourself 'which tracks?'. Using the Tracks pop-up in the dialogue box that has appeared (see screen below), choose 'All Tracks'.

This is the dialogue box that appears when you click a Snapshot button in DP. The three pop-up menus allow you to configure the 'which tracks', 'what' and 'where' of the Snapshot.

Asking yourself 'what?' (kind of data), use the Data Types pop-up to choose 'Data Types Visible in Mixing Board'.

Asking 'where?', look at the options in the Time Range pop-up. 'All time' would replace any Mixing Board automation data already present, setting the automation to one value for the whole sequence. The other options mostly deal with how the new data would integrate with existing data. So 'From Counter to Next Change (Flat)' would cause breakpoints to be written into tracks at the playback cursor's position, and for the values associated with them to stay 'in force' until the next automation breakpoints (if there were any) were reached, at which point the values of the automation data would change suddenly. For our purposes this is a good choice, so hit the 'OK' button. You may already have realised that if you want to use Snapshot options such as 'Tracks Shown in Mixing Board' or 'Current Data Types in Graphic Editor', you need to organise those things before clicking the Snapshot button, but essentially the 'which tracks, what, and where' idea still applies. When you take Snapshots from plug-in windows, options are offered in the Snapshot dialogue window pop-ups that are especially useful if you only want to Snapshot the plug-in values. For example, choosing 'Track Shown in Effects Window' means that you'll only be entering data into the track on which your plugin resides, while 'All Data Types in Effects Window' makes sure you record all of the plug-in's parameters in one fell swoop. On the flip side, if you're only working with one or two parameters for a plug-in, the 'All Enabled Data Types' option ensures that you don't bog down your tracks with automation data types you're just not going to use. (For more on Enabled Data Types see the 'Automation Setup' box above.) Perhaps the cleverest thing you can pull off with Snapshots is mix and plug-in 'morphing'. Set up a mix or plug-in with some specific settings, take a Snapshot, move some way through your sequence, dial in different values, then take another Snapshot with something like the 'From Counter to Previous Change (ramp)' option chosen. (This one is good for when you want the parameters of the Snapshot to affect the region to the left of the playback cursor but to the right of any existing breakpoints). Upon playback of the section you just defined, DP will morph continuous-type parameters from the first snapshot to the second — file:///H|/SOS%2005-06/Using%20Automation%20In%20Digital%20Performer%A04.5.htm (8 of 9)9/28/2005 2:43:34 PM


nice! Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895




In this article:

Dual CPUs In Logic Have Your Say!

Current Versions Mac OS X: Apple Logic Pro v7.1

What does Logic v7.1 offer? Logic Notes Published in SOS June 2005 Print article : Close window

Technique : Logic Notes

Mac OS 9: Emagic Logic Pro v6.4.2 PC: Emagic Logic Audio Platinum v5.5.1

After months of speculation Apple have finally announced the release of the first major Logic 7 upgrade, but what can users expect from version 7.1? Stephen Bennett

At the National Association of Music Merchants (NAMM) in January 2005, current Apple Director Of Audio Applications (and ex-Emagic boss) Gerhard Lengeling broke all the usual company guidelines regarding future product releases and announced the features of an upcoming version of Logic. This was very unexpected, as Apple policy is only to make product announcements 30 days before they actually become available. No shipping date was given, only a list of new features. Of course, the main interest was the announcement of full PDC or Plug-in Delay Compensation. Versions 5.x to 7.0.1 of the program already compensate for the delay caused by inserting Logic's own plug-ins on tracks, busses and auxes. However, delay compensation for third-party Audio Units (AU) has been for tracks only. Full PDC would remedy this situation. There had been an outcry at the release of version 7 when it was discovered that full PDC was not amongst its new features. Some had seen the unusual NAMM announcement as a cynical ploy to shut up complaining detractors by promising this much-anticipated feature in an unspecified future update. There's been a lot of rumour that PDC would be integrated into the Core Audio aspect of Tiger, the new version of OS X, due sometime in the Spring and that a version of Logic able to make use of full PDC would follow shortly afterwards. Dr Lengeling also announced that there would be a 'small fee' for this upgrade, as it was to be supplied on CD or DVD — much to the annoyance of those who had just bought the latest version of Logic.

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Just to prove that speculation can be futile, Apple have released version 7.1 of Logic, with full PDC and a plethora of other features, well before Tiger's release. There is a cost, but at £13.99 I think that most Logic users will be able to stomach it. Version 7.1 includes two 'hybrid' synthesizers, a bass amplifier emulator plug-in, the ability to drag plug-ins around directly in the channel strip, drag and drop for Ultrabeat and the EXS24, a brace of new Key Commands, improved import and export and several other improvements and bug fixes. Any audio recorded into Logic can now be time- and pitch-shifted directly in the Arrange page in an Ableton Live-like fashion. The update also includes a new version of the Waveburner CD authoring and mastering program — something hotly anticipated after the insipid release that was shipped, unannounced, with Logic 7. There's also a version of Logic Express v7.1 available with fewer features but full PDC. It really does look as if Apple has listened to all those complaints and feedback — Logic v7.1 looks like it might be the sequencer many expected v7.0 to be. I'll be bringing a full mini-report when my copy arrives. Logic 7 brought with it a slew of bugs and omissions. For example, the Key Command to open the 'hidden' automation track (so data can be edited in the Event list) is missing, as is the Catch On Start preference. The latter is particularly annoying, as it will cause Logic to start playing back a Song as soon as it's loaded — not something you want to happen in a live situation! These problems can be remedied if you have an earlier version of Logic for OS X installed on your Mac. If you open the old version, set the 'catch on start' preference to 'off' and set up a Key Command for the automation track, these parameters will be saved in the Logic Preferences file. When you run Logic 7, the Key Command will still work and Logic won't start to play the Song before you've introduced it! It remains to be seen if version 7.1 has corrected these bugs.

Dual CPUs In Logic If you have a dual-processor Mac and open the CPU monitor, you'll often find that Logic seems only to be using the first processor when playing back a song with many plugins. This will often cause the program to throw up a Core Audio error message. If you open the OS X CPU-monitor program (/Applications/ Utilities/Activity Monitor), you'll see that the operating system itself is actually balancing out processor usage evenly — even though Logic thinks it isn't! You can solve this problem by starting playback, stopping and starting again. You'll see that both CPUs are reported as being used equally in Logic now. You can also force Logic to load the second CPU with a processor

The Logic and System CPU monitors — spot the difference!



intensive plug-in by selecting the track it's on when playing back the song.

Have Your Say! If you want to suggest changes or improvements to Logic, then here's your chance! The Apple development team are inviting SOS readers to send in their suggestions of what they'd most like added or changed in Logic. Email your top five suggestions (in order of preference) to [email protected], and we'll forward your lists on to the Logic team. We'll be asking them for feedback on which changes users deem most important and how these might be addressed. Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



Yamaha AW4416 User Tips

In this article:


Track Bouncing Gating & Ducking From Aux Masterclass: Part 2 Published in SOS June 2005 Sends Effects Via Editing Print article : Close window Creative Use Of The Technique : Recording/Mixing Oscillator Dolphins & Alarm Clocks? Vocal Comping Side-chain Trickery AW Inspiration Another selection of creative and time-saving tricks Useful Web Sites for Yamaha's fully featured hard disk multitracker.

Tom Flint

Last month I looked at some strategies for automating mixes with Yamaha's AW4416, as well as discussing the operation of the Waves Y56K effects board in this context. In the concluding part of this short series, I'll be examining some of the finer points of the multitracker's routing and editing functions, and demonstrating how they can help make the recording process easier and more creative.

Track Bouncing Anyone who bought their AW4416 after the introduction of the labour-saving Quick Rec 2 page may well wonder why they should bother routing their signals to record tracks in any other way, but in the long run it's well worth learning your way around the Setup pages and getting to grips with the bussing structure so that you can get the most from the routing options. One situation where this is particularly important is when track bouncing, a process whereby you record a mix of several tracks onto a single track in order to free up the source tracks for recording extra parts. The procedure involves selecting the tracks to be combined, routing them to a common buss, and finally recording the result onto a separate track. Some recorders provide a special 'ping-pong buss' for this very purpose, but the AW4416 offers more than that by having a flexible eight-buss system.

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In the following scenario, we have a mix with just one spare track requiring a three-part harmony vocal. The first thing to do is identify three tracks which do not need to be heard while overdubbing. I usually leave myself with the drums, the lead vocal (if it is relevant), and at least one key instrument to carry the melody. Whatever tracks are expendable for the purposes of monitoring are switched for virtual tracks which can be used for recording the backing vocals. The virtual tracks occupy the same monitoring channels as the ones they replace, and this means that they inherit the same insert patches, EQ, compression, and pan positions — all of which will probably need to be changed! Before doing so, make sure the setup is saved in a Scene so that you can recall the Monitor settings after you've completed the bounce. After that it's simply a matter of recording the three harmony parts to the virtual tracks. After recording, press the Pan/Routing button followed by the F3 key to bring up the monitor channels' Pan page. Here the buss routing and panning for all 16 monitor channels can be viewed together. For this example, engage buss number seven for each of the three harmony tracks, deselect the main stereo (ST) buss so that the same audio is not monitored twice during mixdown, and then ensure that the panning of all the parts is central.

Although you can record without ever tangling with the AW4416's internal group busses, You need to use them if you're going to do any track bouncing. You can find the internal mixer's channel routing switches via the front-panel Pan/Routing button — three tracks are routed to buss seven in this picture.

The next task is to assign buss seven to the destination track, which, for the sake of this example, we'll make track 15. Press the Setup button to bring the patching pages to the screen and make sure that the Patch In page is displayed (hit the F1 key if necessary). The bottom section of the page has an informatively titled box labelled Recorder Track Input Assign, containing settings for all 16 tracks. Use the cursor keys or connected mouse to highlight track number 15 and rotate the data wheel to change its status to buss seven. Finally, press the red Rec Track select button for track 15, so that it is flashing in record-ready mode. The balance of the three tracks can now be recorded onto track 15. I usually record my BVs using the Auto Punch facility, and use the same In and Out points for the bounce too. The Auto Punch is well worth using while setting up the correct mixdown balance, as it may take several passes to get things right. After the mix is finished, the virtual tracks can be swapped back, and all the monitor-channel changes can be undone by recalling the Scene you saved earlier. The same process works for any of the busses, or for any number of tracks, and more than one buss can be used at once, which is handy if a stereo mixdown to two tracks is required.



Gating & Ducking From Aux Sends In the main body of this article, I explain how it's possible to have the left-hand of two adjacent channels gating the right-hand one. However, there is another routing possibility which is particularly useful when working on a song where the interacting channels are not next to one another — it is available from the Key In Source box, which has two further routing options, one for Aux 1 and the other for Aux 2. In other words, when setting up the dynamics for a track which requires gating, either one of the first two Aux sends can be selected as the trigger. For this reason, I like to keep the first two aux busses free if at all possible. For the sake of argument, let's say you have a synth pad which you want to gate in time with a song's rhythm, and the ideal trigger is a stereo snare and kick mix pre-recorded to tracks seven and eight. To create the patch, first make sure that the Monitor fader layer is selected by pressing the relevant Monitor button. For the synth part's Monitor channel, pick a gate processor from the channel's Dynamics Library, and select Aux 1 as the Key In Source. Next, press the Aux 1 Fader Mode button so that the hardware faders represent the send levels of each Monitor channel signal sent to Aux 1 buss. From here faders seven and eight can be raised to a decent level. Incidentally, the same thing can be done from the relevant Ch View page by selecting the on-screen aux-level fader, and then rotating the Data wheel. Having set the send level, the gate's Threshold, Attack, Release, Hold, and Range controls will need adjusting to obtain the desired effect. It's also worth remembering that the AW has a ducking processor, which can be used with the same routing but for different effects. For example, when mixing it is often useful to lower the level of some mid-range or solo instruments — such as guitars — when the lead vocal is taking centre stage. In such a case, by making vocal track the Key In trigger, choosing a ducking processor (shown as DUK in the Dynamics Library) for the guitar tracks, and setting its Range to a few decibels, its possible ensure that the level of the guitar parts automatically dips whenever the vocalist starts up.

Effects Via Editing Although the AW4416 has two internal effects processors, these are quickly used up in a complex mix, so it's useful to know that you can create nice chorus/ detuning effects using the audio-editing functions, freeing up an effects processor at the expense of an audio track or two. First identify a bit of audio, perhaps guitar or vocal, which you want to apply such an effect to, and decide where its start and finish points are. Then head to the Edit menu, choose Part and from its submenu, and select Copy to copy it to a spare track. When specifying the To Start point, add on something like 30ms to the value that was input in the From Start box. This offset will separate the copy from the original, such that when they are mixed together this will create a chorusing effect. If you are unsure as to how much delay to apply, you can either experiment by trying a delay from the effects processor first, or you can dial in some Sample Delay from the relevant monitor channel's editing page — up to 59ms can be applied in this way, which should be more than enough. file:///H|/SOS%2005-06/Yamaha%20AW4416%20User%20Tips.htm (3 of 9)9/28/2005 2:43:49 PM


To increase the impact of the effect, select the copied audio, return to the Edit menu and choose Pitch. This function reprocesses audio without altering its length, allowing for pitch adjustments in semitones and cents. To achieve If you create a copy of a track at a slightly later time position than the original, you can create a great 'static chorus' effect. This can be finea thickening tuned using the copy's channel Delay parameter (above) and can be sound rather than enhanced using the Track Pitch function (left). a harmony, it's only really necessary to pitch-shift by a few cents. After the pitch and delay edits, the part may still need a little attenuation, panning, and EQ in order to further separate it from the original audio, all of which can be done from the relevant monitorchannel editing page again.

Creative Use Of The Oscillator The AW4416 doesn't immediately appear to have too many creative 'sound warping' functions outside its effects section, but its facilities and routing options do offer a few creative possibilities to the more intrepid user. The first page in the Utility menu provides a small selection of oscillator types which can be freely routed to the eight busses, eight auxiliary channels, and main stereo buss. At first glance the 100Hz, 1kHz, and 10kHz sine waves and white noise don't promise much, and Yamaha only really included them for test purposes. However, with a little clever routing, they can be used as the basis of a variety of sound effects. Select the Patch In page from the Setup menu and navigate to the Record Track Input Assign box at the bottom. Pick an empty track and select a spare buss for its input. From the same page, change one of the effects processors from its standard send setting to the Insert status by selecting it in the Effect Patch box and then turning the data wheel. Next, press the Ch View button and go to the monitor page relating to the spare track on which you are about to record. In the middle of the page is the Effect Insert Assign box, which allows you to set your choice of processor as the channel's insert effect. Now go to the Oscillator page and engage the buss you previously assigned to the spare track, and make sure the stereo-buss routing is switched off. At this



stage your routing allows you to record an oscillator tone to the spare track via the selected effects processor. With the track in question armed for recording, turn on the oscillator, pick the 100Hz tone, and find a comfortable listening volume using the oscillator's Level control and the channel fader. To start with, select the Ring Mod effect by pressing the Aux 7 button and then following the F2 tab to the Library Before starting track bouncing, it makes page. Now press F1 to move back to sense to save the pre-bounce setup to a the Effect Edit screen and set the spare mixer scene. That way, if you change Depth to zero percent so that a your mind about the balance or processing of constant tone can be heard. Then the bounced track later on you can come adjust the Osc Frequency control until back to it and redo it. the signal is in tune with whatever else is recorded. Once this has been achieved, the oscillator depth can be adjusted to taste, as can the FM Frequency control, which more or less allows the effect to be synced with a Song's tempo. Unfortunately, the tuning may not quite be fine enough to create a totally sync'ed effect, but it could be used as the foundation for a Song. Obviously a different sound is had by using the 1kHz and 10kHz tones, so experimentation is worthwhile. Setting a slow sweep and 100-percent depth, causes the sound to rise and fall over a huge range very gradually, whereas a fast FM sweep results in a more frantic effect. Detuning the oscillator frequency away from the Song's tuning can also work well. Tremolo is the other useful effect found in the Yamaha onboard processor, which acts just as effectively on the white noise oscillator as it does on the sine waves. Set the tremolo frequency and pick a modulation waveform from the algorithm's oscillator menu. Sine and triangle are worth trying, but it's the square waveform which is most useful in the sound-effects department, as it cuts the signal more brutally, particularly when the depth of the Tremolo is at 100 percent. With these settings, selecting white noise on the oscillator will produce short bursts of noise that are very percussive.

Dolphins & Alarm Clocks? The creative possibilities don't stop there though. For the following example, set the Oscillator level to -17dB and the tremolo's controls so that the frequency is at 7.85Hz, the depth is 100 percent, and the waveform is square. The mix balance of the effect should remain at 100 percent and the rest of the EQ controls at a neutral position. Now go to the channel view page for the monitor channel the effect is being monitored through, and select its dynamics processor. From the library, pick the gate (GAT) numbered 003, and apply the following settings for



the white noise patch described above: Threshold -12dB; Decay, 23ms; Range, -70dB; Attack, 9ms; and Hold, 7.28ms. With the above setup you should be hearing a nice clear rhythmical effect which isn't too harsh. Increasing the threshold to -11dB makes the sound far softer, but taking it to -10dB should mute the effect altogether. Leave the patch at -10dB and move to the channel EQ page. Pick one of the mid controls and change its Q to the narrowest setting of 10. Then switch the EQ on and boost the mid-band by 18dB. Boosting at about 250Hz, the AW should be producing an interesting scratchy sound, not unlike someone shaking a tray of sand. The data wheel can then be used to rapidly sweep the band up and down, producing a variety of tonal effects. To make things more interesting, move to the other EQ midband, narrow its Q to 10, and this time cut by 14dB. Move the cut to somewhere around 1kHz and then try sweeping the boosted band up and down. The point where the two bands cross causes considerable cancellation, although either side there is a graduated filtering effect which is a little like the result of using a wah wah. Moving the cut to different locations shifts the articulation and changes the result in a variety of ways.

To route the oscillator through the effects, first go to its Utility page, switch it on, and assign it to a group buss using the routing switches (top). This group buss can be routed to a recorder track via the Setup Patch In page (centre left), and one of the effects processors can be switched to act as an insert effect (centre right). With the relevant recorder track armed, the oscillator now runs through its associated monitor channel, into which the internal effect processor can be inserted (bottom).

Keeping everything else the same, change the white noise to a 1kHz tone, but place the EQ cut at 1kHz. The threshold of the gate will also need to be shifted to -50dB. This time, the effect generated by sweeping the boosted EQ band is quite different, and, to me, sounds like a chirping dolphin, singing in time to an electronic alarm clock!

Leaving the 'dolphin and alarm clock' chorus going for the time being, pick another buss and, from the Oscillator page, send the signal to the second buss as well. Find another spare track, patch the buss to it from the Record Input



Assign part of the Patch In page, and then arm the track so that the oscillator can be heard. Now release the second effects processor from its send status and insert it into the new record track. The first thing to try is putting two tremolo effects together, and then slightly varying the frequency and depth. This works well, but Tremolo can also be paired with Ring Mod. I set the Ring Mod Source to Osc, with an Osc Frequency of 8.9Hz, an FM Frequency setting of 0.05Hz, and a depth of 100 percent for some seriously long sweeping effects. It's also worth trying two ring modulators together, keeping the above setting for the first, but detuning the second to 8.7Hz, reducing the depth to 50 percent, and then moving the FM Frequency to 0.3Hz, so that the two sounds twist around each other. This patch worked equally well with the 100Hz oscillator as its input.

Vocal Comping One further reason for editing between vocal lines in particular is that it can make the process of compiling a lead vocal from a number of takes much easier. I know that some AW4416 users export their vocals for editing on a software sequencer to make their life easier, and it's true that the AW4416 doesn't make editing as easy as it could do, but exporting audio requires a suitable software package and it's a pain to do, especially without USB transfer!

The V.Tr Edit screen is great for comping vocals, as you can easily see when you've whittled multiple takes down to a single line.

When I'm gathering the best vocal lines from a number of takes on different virtual tracks, I edit each take into blocks of audio and, if I have enough virtual tracks, copy each one to another virtual track. I then decide which bits of which takes I do and don't want, and delete the weak parts. Doing this is made infinitely easier when the lines have already been cut into separate blocks, allowing some bits to be edited out by eye, and the start and end points of others to be easily found from the waveform display. As soon as I am left with the optimum take for each line of the Song, I make my way to the V.Tr Edit page (by pressing Edit and then F2) so that I can see all the remaining audio. By this time, it should be clear to see from looking at the display which virtual track holds the chosen audio for each vocal line. I then copy the parts to a single master track. Sometimes there are overlaps, where the start of the line from one virtual track begins before the previous one has finished, so it's not always possible to combine all the parts. I usually use the waveform display a lot to determine if there are any problems of this nature, and if so I return to my original takes and use an alternative combination.

Side-chain Trickery file:///H|/SOS%2005-06/Yamaha%20AW4416%20User%20Tips.htm (7 of 9)9/28/2005 2:43:49 PM


The next trick is to use one channel to modulate another using the 100Hz oscillator. To do this, make sure the two channels are next to one another — say tracks 15 and 16. For this example keep the Ring Mod effect on monitor channel 16 with Oscillator Frequency at 8.9Hz, FM Frequency at 0.05Hz, and Depth at 100 percent. Change the effect inserted into monitor channel 15 back to Tremolo and set its Frequency to 4.05Hz, Depth to 100 percent and Wave to Square. Now find the dynamics page for monitor channel 16, select patch 003 again, and make sure it is set up as follows: Decay, 23ms; Range, -70dB; Hold, 1.02ms; Attack, 0ms; Threshold, around -20dB. This time though, navigate to the Key Source box and select Left (Post EQ). Having done the above, the tremolo pulse will rhythmically chop the long sweep of the ring modulator, to great effect. Practically speaking, some of the above examples as they are described here are nothing more than sound effects, however, it's possible to adapt the principles to a variety of musical situations. For example, the white noise can be fed through a noise gate which is set so that it opens according to the signal from the Key Source. If a drum track is used as the key, the white noise will become a percussive blast of noise, operating in perfect time with the Song's tempo. This would be an ideal way to add dirt to clean GM sounds, for example.

AW Inspiration Hopefully this short series has provide some inspiration to those who are still getting to grips with the many features of the AW4416, and perhaps I've also managed to convince you that even the most straightforward of this machine's audio tools have hidden creative potential.

Useful Web Sites Yamaha AW4416 Home Page

The official site for the AW4416. This is where the latest software updates and third-party products are advertised. It has a Q&A section and a downloads page, as well as an upto-date list of compatible hard disks and CD-RW drives. www.aw4416.com Social Entropy AW4416 Site

This is an unofficial user's site. However, it's more useful than the official one provided by Yamaha!



Many users who have experienced crashes or software problems have posted their questions in the forums here, making quite interesting reading. The site has a most useful news page detailing the latest updates on some of the software applications written by users. This includes some shareware written for Windows by John Kimble, called AWExtract, currently at v2.66. The program is a utility for extracting WAV files from Yamaha AW4416 and AW2816 backups. All audio tracks, regions, and sample data can be extracted. 16-bit and 24-bit resolutions are supported at both 44.1kHz and 48kHz sample rates. http://socialentropy.com/aw4416 Published in SOS June 2005 Sound On Sound, Media House, Trafalgar Way, Bar Hill, Cambridge CB3 8SQ, UK. Email: [email protected] | Telephone: +44 (0)1954 789888 | Fax: +44 (0)1954 789895



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